Sound on Sound (July 2004)

In This Issue July 2004 In This Issue Click article title to open Reviews People Antares Filter Arif Mardin: Produce...

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In This Issue

July 2004 In This Issue Click article title to open Reviews

People

Antares Filter

Arif Mardin: Producer

Filter & Delay Plug-in (Mac/PC) Antares certainly don't believe in the 'less is more' maxim, if this new filtering effect is anything to go by.

From Aretha Franklin to Norah Jones Arif Mardin has engineered and produced an incredible array of classic records from artists such as Aretha Franklin, Dusty Springfield, Diana Ross, the Bee Gees and Barbra Streisand — yet the runaway success of his recent work with Norah Jones threatens to overshadow even these achievements.

ART Digital MPA Mic Preamp & A-D Converter This new dual preamp from ART combines imaginative valve circuitry with more digital output options than you can shake a stick at!

Business End

MPG evaluation of Readers' tracks Listen to the music while you read constructive comments Software-controlled Modular Synth from MPG (Music Producers Guild) members on SOS The Nord Modular offered a classic blend of flexible software readers' submitted recordings. and well-designed hardware in 1998. But can the improved Crosstalk G2 keep up with the soft synths of 2004? Your correspondence Digidesign Command 8 Replies to a few more of your letters, emails, and faxes. Control Surface For DAWs Learning Difficulties With fader controllers, you often have a choice of buying Leader something dedicated to one software package, or selecting a generic unit that gives versatility at the cost of tight So you think missing a lecture of your music tech course integration. Digidesign's Command 8 aims to offer the best won't harm your career prospects? Read on... of both worlds.

Clavia Nord Modular G2

Disco DSP Vertigo 2.0 Resynthesizing Additive Soft Synth (Win) Disco DSP's additive synth plug-in offers both high-quality resynthesis, with up to 256 partials per note, and a novel 'spectral display' for editing your sounds.

Focusrite Liquid Channel Recording Channel Combining Focusrite's cutting-edge preamplification and AD conversion with Sintefex's pioneering convolution technology, the Liquid Channel can mimic the most celebrated mic preamps and compressors ever designed. Could this be the only voice channel you'll ever need?

FXpansion VST To RTAS Adaptor Plug-in Format Adaptor (Mac OS X & WinXP) As the name suggests, FXpansion's utility allows VST effects and Instruments to be used in Pro Tools, but how well does it really work?

Korg Legacy Collection (Part 2)

Readerzone

Shaun Hayward & Joe Kenny Shaun Hayward & Joe Kenny of Chunc Productions reveal their gear setup and musical motivations.

Studio SOS Peter May The SOS team help Peter May to brush up his drum sounds and put more life into his mixes.

Writing Manuals Sounding Off Who'd want to be a writer of manuals? The endless showbiz parties, awards, book-signings and... hang on a minute...

Wyclef Jean Producer In 1996, the Fugees came like a breath of fresh air into a world of hip-hop that was becoming stale around the edges. Now Wyclef Jean is a star in his own right, and has deployed his production talents for artists ranging from

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In This Issue

Virtual Instrument/Hardware Controller In the second part of our coverage of Korg's new collection of software instruments, we turn our attention to the virtual MS20, and the MS20-shaped hardware controller. Just how well do these renditions compare to the original?

urban legends like Funkmaster Flex and Cypress Hill to Whitney Houston, Michael Jackson and even Tom Jones.

Latest Sample CDs

Logic Notes The Aux object might seem a little arcane, but it's the key to a variety of useful Environment workarounds.

Sample Shop Reviews/appraisals of the latest sample CDs: Alma Flamenco MULTI-FORMAT; A Funky Future MULTIFORMAT; Vienna Overdrive GIGASTUDIO/EXS24 MKII; Cuckooland Completely Bonkers MULTI-FORMAT.

Mission Pro SM6A Active Nearfield Monitors Although perhaps better known as a leading UK manufacturer of hi-fi speakers, Mission have now launched their first active nearfield specifically designed for studio use.

MOTU 828 MkII Firewire Audio Interface (Mac/PC) Three years ago, Mark Of The Unicorn were the first manufacturer to bring out a working Firewire interface. Now, they've replaced the 828 with a MkII version offering high sample-rate recording, more flexible clocking, MIDI I/O and better metering — and, what's more, they've cut the price.

Rode K2 Valve Microphone Rode have built on the technology from their respected NTK to develop this affordable new multi-pattern valve mic.

Sibelius 3

Technique

Logic's Aux object

CLASSIC TRACKS: 'The Reflex' Producers: Duran Duran, Alex Sadkin, Ian Little; Engineers: Phil Thornalley, Pete Schwier When Duran Duran began work on their third album in 1983, they were already one of the biggest bands in the world — and with eight months of studio time and half a million pounds spent, huge expectations surrounded Seven And The Ragged Tiger...

Demo Doctor Listen to Readers' own recordings Think your music is good? Listen to these tracks and read the good Doctor's analysis of their good and bad points...

Future-proofing DP Projects Digital Performer Notes This month, Performer Notes concludes its look at DP's Consoles, and suggests some ways of 'future-proofing' your Projects against the relentless flow of application and plug-in updates.

Longhorn and Harbal PC Notes The Harbal EQ-matching program benefits from some worthwhile tweaks. We report them, and discuss some new information about Longhorn, the next incarnation of the Windows OS.

Mac/PC Scorewriter Over the last 10 years, Sibelius Software have built a reputation for providing what most musicians consider to be the leading score-writing software. Now, in version 3, they have teamed up with Native Instruments to provide MIDI config problems in OS X; enhanced playback facilities — but does this upgrade live up Garageband v1.1 update to the high standards set by previous releases? Apple Notes The launch of Mac OS 10.4, codenamed Tiger, is just Soundcraft Compact 10 around the corner, but in the meantime there's the 1.1 Desktop Mixer This slim new mixer has been designed from the ground up update of Garageband to be going on with. We reveal the improvements, as well as investigating MIDI configuration to cater specifically for computer studio users. But have problems in OS X. Soundcraft got the balance of facilities right?

Steinberg Cubase System 4

PC Music Freeware Roundup

USB Interface & Recording Software (Mac/PC) Buying your hardware and software from the same manufacturer is usually a good bet for hassle-free recording. Steinberg's latest package bundles their Cubase SL 2 sequencer with a four-in, four-out USB interface.

PC Musician Thanks to the Internet and the generosity of talented programmers all over the world, it's possible to assemble a PC music software suite for no money at all. We round up some of the best download sites and freebie programs.

Studio Electronics Modmax Filter & Phaser Pro Tools v6.4 update Analogue Effect Pedals

Pro Tools Notes


In This Issue

Two new effect pedals from the celebrated synth manufacturer provide serious modulation flexibility within a stomp-box format.

Competition

WIN Camel Audio Cameleon 5000 & CamelPhat v2 plug-ins Sound Advice

Q. Do I need to compress my audio before A-D conversion? Q. How can I resolve an IRQ conflict? Q. How do I take care of my valve mic? Q. How does virtual surround work? Q. What kind of reel-to-reel tape recorder do I need?

PT version 6.4 comes with some surprising and longawaited new features.

Reason Notes starts here! Monthly column for Propellerhead Reason users SOS's 2003 Reader Survey told us that you want our help in getting the best from this popular package — and we're happy to oblige with a new monthly column.

The Key Editor; v2.2 Cubase update Cubase Notes The Key Editor is a seemingly straightforward MIDI editor, yet under its surface lie a number of features that can really speed up your editing tasks. We explain, as well as reporting on the new version 2.2 Cubase update.

The Secret Of The Big Red Button Synth Secrets After over five years, Synth Secrets reaches its conclusion (and conclusions!). Will we ever look at synthesis in quite the same way again?

Using OMF to transfer Sonar files Sonar Notes This month we look at clever envelope tricks, as well as how to transfer projects between DAWs that speak OMF.

Using Software Effects & Outboard Masterclass Get the most from your computer's CPU by learning how to put your effects plug-ins where they really count. Plus, find out how to increase your mixing power by incorporating hardware effects units into your software mixdown. Music Business

Alternative Careers for Sound Engineers New Directions The music industry's down, but your passion for audio is still high. Fortunately, there are alternatives to working in the studio that can feed your fervency and still make you a living — perhaps even a better one.


Antares Filter

In this article:

On The Outside Under The Knife The Nitty Gritty

Antares Filter £128 pros

Antares Filter Filter & Delay Plug-in (Mac/PC) Published in SOS July 2004 Print article : Close window

Reviews : Software

Very comprehensive feature set. Can produce some unique effects, especially on busy sources. Good printed manual and thoughtful preset design. Extensive MIDI control and modulation options.

cons Filters can't be overdriven, and sound too polite for many uses. Crowded, parameter-laden interface makes programming hard work. Filter curves in display aren't animated to show the effects of modulation. The lack of an HTDM version means it can't be used on aux tracks in Pro Tools TDM systems.

summary The filters themselves aren't the most characterful around, and the interface is a bit indigestible, but its extensive delay and modulation options still make Filter a powerful tool for manipulating loops and beats.

information £128.08 including VAT. Unity Audio +44 (0)1440 785843. +44 (0)1440 785845. Click here to email

Antares certainly don't believe in the 'less is more' maxim, if their new filtering effect is anything to go by. Sam Inglis

If you found a vintage synth with four multi-mode filters, each having its own envelope generator, LFO and delay line, you'd think you'd died and gone to analogue heaven. But these days, it's easy to dream up software devices with specifications way beyond what any of the hardware classics was capable of. Antares's Filter plug-in has all the above features and more, including two built-in step sequencers, a 12x12 modulation matrix, an evelope follower, extensive MIDI control and a variety of routing options. It's tempting to ask why they didn't just add a couple of oscillators and turn it into a full-blown synthesizer, but as it is, it depends on having a mono or stereo audio input, and can be used as an insert plug-in in any sequencer that supports VST, RTAS, MAS (Mac OS only) or Direct X (PC only) formats. Pro Tools users should note that there's no HTDM version, so you can't use it on aux tracks within a TDM system. This is a fairly major limitation, since as we'll see, Filter is more useful on mixes and submixes than on individual instruments. Authorisation is via challenge and response, and worked as expected in the review system. Annoyingly, no minimum spec is given, but I found that it just about ran on my ancient 300MHz G3 Mac, which is pretty good going for such a complex plug-in. In other respects, the printed manual is good, and the separate explanation of what all the presets do is equally welcome.

www.unityaudio.co.uk www.antarestech.com

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On The Outside

Antares Filter

Test Spec Antares Filter v1.00. Beige 300MHz Apple Mac G3 with 256MB RAM, running Mac OS 9.2. Digidesign Mix system running Pro Tools v5.1.3.

Antares describe Filter as 'The audio equivalent of a brilliant surgeon — who's a really good dancer.' One might add that it's the graphical equivalent of a hyperactive toddler — who's also got a migraine. Everything is crammed into one window, with lurid primary colours used to distinguish the four filters, and most of the controls extending no more than a few pixels across. None of the individual elements is too cryptic, but the overall effect is a bit overwhelming, and a multipage approach to interface design might have been more friendly. The centre of Filter's window shows a plot of level against frequency, with the responses of each of the four filters shown in a different colour. You can adjust the cutoff and resonance of each filter by clicking and dragging, but I was disappointed to find that the display is not animated to show how filter response changes with modulation. This means that the information it provides is useful only when filter frequencies are not being modulated — and modulating filter frequencies is precisely what Filter is all about. To the left of the frequency display is a set of buttons allowing you to select various series/parallel routing combinations. The individual controls for each filter and its associated delay line are at the top, whilst the modulation sources — LFOs and envelope generators — are ranged along the bottom. Immediately below the frequency display are Filter's step sequencers, while the area at the top right is where you set up modulation routings.

Under The Knife Filter is certainly bristling with features in every area. The filters themselves can be switched between low-pass, high-pass, notch and band-pass modes, with slopes variable from two-pole (12dB/octave) to eight-pole (48dB/octave). Each of them has a Frequency Link pop-up, allowing you to tie its cutoff frequency to that of any of the others. All of the filters also have their own delay lines with variable feedback and up to two seconds of delay. Delays can be switched pre or post the associated filter, and like all of Filter's time-related elements, can be sync'ed to host tempo. Output from each filter stage can also be polarity-inverted and, unless you're using the mono-to-mono version of Filter, panned. The step sequencers, or Rhythm Generators as the manual describes them, are simply two strips divided into cells of equal duration; each cell can be either on or off, and their state is switched by clicking in them. When the sequencer reaches a cell that's switched on, it sends a trigger to any envelope set to receive it, and a maximum positive value to any modulation destination it's hooked up to. Red arrows can be positioned to set repeat points for each sequencer, while the single Run/Stop button behaves differently depending on whether, and how, the plug-in is sync'ed to host tempo. With the Rhythm Generators' Sync button unlit, a single press of Run/Stop starts the sequence, a second press freezes it and a third returns to zero. With Sync on, its behaviour depends on the plug-in's Master Tempo setting, and it can either act simply to turn over start/stop duties to the file:///H|/SOS%2004-07/Antares%20Filter.htm (2 of 5)9/22/2005 8:55:00 PM

Antares Filter

sequencer transport controls, or independently of them. When Filter is sync'ed to master tempo, starting playback in the middle of a song causes the Rhythm Generator position to jump to the appropriate point in the sequence, which is a nice touch. Each of the four 'function generators' at the bottom of the screen can be switched to show controls for an LFO or an envelope generator, although in every case, both are always available as modulation sources. A wide range of LFO waveforms is available, including some neat random shapes, and LFO rate can be sync'ed to host tempo or set freely in Hertz or beats per minute. When not tempo-sync'ed, rates for each of the four LFOs also appear as modulation destinations in the mod matrix. The EGs are equally well specified, offering graphically editable ADSR envelopes with an additional Hold time parameter for the Sustain phase. They Four LFOs and four envelope generators are can be triggered by active cells in always available as modulation sources, though they are edited on an either/or basis. either of the Rhythm Generators, or by a MIDI Note On message; you can choose which MIDI channels to assign to each of the EGs. An additional MIDI Gate mode works the same way, except that the Hold parameter is ignored, and the Sustain level is held until a Note Off is received. You can also insert a delay between an EG's receiving a trigger and firing, whilst the display can be set to show envelope parameters either in milliseconds or in precentages of the step sequencer's cell duration. The modulation matrix is straightforward to use. Sources include the four LFOs and EGs, the simple envelope follower, the two rhythm generators, and a variety of MIDI messages such as notes, mod wheel, velocity and pitch-bend data. All the key parameters for the filters, delays, LFOs and EGs are available as destinations. Sadly, though, there's no way to route a side-chain signal through the envelope follower; nor can the follower output a trigger for the envelope generators. Overall, Filter's interface is not difficult to understand, but I didn't find it particularly inspiring or intuitive either. It's definitely not one of those plug-ins where you can rely on happy accidents to create worthwhile results. The size of the controls means you're often squinting at the screen to find out what you're clicking on, and their sheer number makes programming anything interesting a bit of a hard slog. This is especially the case if you start from scratch, so it's handy that Antares have supplied a decent selection of editable presets, complete with suggested parameter tweaks.

The Nitty Gritty Of course, it doesn't matter what the interface is like if a plug-in doesn't sound file:///H|/SOS%2004-07/Antares%20Filter.htm (3 of 5)9/22/2005 8:55:00 PM

Antares Filter

very good. So does Filter live up to Antares' claims for it? The best answer I can give is 'sort of'. As a surgeon, it's certainly capable enough, thanks to its 48dB filter slopes, but I can't imagine anyone buying a plug-in like this just to use it as an EQ. As a dancer, meanwhile, it's more Darcy Bussell than Patrick Swayze. Filter is certainly not the sort of plug-in to elbow its way through a nightclub, waving a glowstick with a deranged look in its eye. It's just too damn polite. The filter response is smooth and free from stepping or other artifacts, but never becomes unstable or threatens to self-oscillate, no matter how high you push the resonance. What is missed most of all is some form of soft-saturation algorithm to model what happens when you overdrive some analogue filters. There's no warm fuzziness on offer when you boost the levels, just peakiness that leads all too predictably to harsh clipping at the output. I searched in vain for settings that would beef up an analogue-style bass line or create a truly pounding kick drum. However, Filter does have some impressive tricks up its sleeve. Paradoxically, its strengths become more apparent the further away you move from straightforward filtering applications. It's definitely at its best when fed with a fairly complex source such as a complete mix, where it can supply all sorts of interesting effects. You can muck about with the spectral balance and dynamics to emphasise a particular instrument such as the bass, crisp up a dull hi-hat, bring drums to the front of the mix or push them to the back. More extreme settings can turn drum beats into resonant, metallic clangs, or warp tuneful source material into sinister, humming machinery. It can also produce some surprisingly delicate phasing and flanging effects, and if you want rhythmic delays that burble and chatter in interesting ways, you've come to the right place. Of course this is not the only filter-based effects plug-in on the market. An obvious challenger to Filter on the Pro Tools platform is Sound Toys' Filter Freak, which definitely has the edge in terms of raw sonic muscle. Filter Freak's 'Analog Mode' steamrollers Antares' design for conventional filtering tasks, whilst its neat animated display is also a useful aid to programming, and it comes in HTDM as well as RTAS formats (though there's currently no VST, Direct X or MAS version, and Filter Freak is a fair bit more expensive than Antares' plug-in). When it comes to more unusual effects, however, Filter Freak can't match the versatility of Filter, and the scope for radically altering loops and beats is far greater in Antares' plug-in. An alternative comparison here might be Ohm Force's Ohm Boyz. Despite offering multiple delay lines and two resonant filters, it still can't match Filter's wealth of parameters or MIDI controllability, but Ohm Force's quirky interface lends itself to serendipitous tweaking in a way that Filter's doesn't. Like Filter Freak, Ohm Boyz can also supply the grit and dirt that are missing from Antares' plug-in. There is some fairly strong competition around, then, but there's also no denying that Filter has a uniquely comprehensive combination of features. Users who are prepared to put a little work into programming it will find it capable of a wide range of results, some of which would be difficult or impossible to achieve with other plug-ins. It wouldn't be my first choice for straightforward filtering functions, but it provides good scope for more offbeat effects, and would certainly repay file:///H|/SOS%2004-07/Antares%20Filter.htm (4 of 5)9/22/2005 8:55:00 PM

Antares Filter

attention from anyone looking for original treatments for sampled loops. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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ART Digital MPA

In this article:

ART Digital MPA

Variable Input Impedance Mic Preamp & A-D Converter Digital Facilities Published in SOS July 2004 In Use The ART Of Preamp Design Print article : Close window

ART Digital MPA £799

Reviews : Preamp

pros Affordable. Variable input impedance. High- and low-voltage tube stage options. Digital output as standard.

cons

A new dual preamp from ART combines imaginative valve circuitry with more digital output options than you can shake a stick at!

No headphone output.

summary A good general-purpose mic preamp that offers both tonal flexibility and quality at a sensible UK price.

information £798.99 including VAT. Sonic8 +44 (0)8701 657456. +44 (0)8701 657458. Click here to email www.sonic8.com www.artproaudio.com

Paul White Photos: Mike Cameron

ART have been in the business of making affordable, effective signal processors and effects units since the mid-'80s, but in the early days their large overlapping product range and ever-changing cosmetic design may have hindered rather than helped their sales effort. Today, their styling has settled down a little, and their manufacturing has been moved to China in order to keep the retail prices low. This combination of American design and low Chinese manufacturing costs has enabled them to bring some quite sophisticated products to the project studio market at very attractive UK prices. One such is the Digital MPA mic preamp under review here, which combines discrete Class-A input circuitry with valve gain stages, variable input impedance, and comprehensive digital output options at up to 192kHz.

Variable Input Impedance Presented in a smart and ergonomically friendly 2U format, the Digital MPA comprises two identical channels of microphone preamplifier with digital outputs fitted as standard. Over the past few years, the idea of variable input impedance has gained favour as a means of more optimally matching specific microphones to the preamplifier, though most users simply adjust for what sounds most musical, rather than what is technically correct. This somewhat esoteric feature is provided here as a continuously variable control that adjusts the input impedance from 140(omega) to 3k(omega). Most mics are happy working into around 600 (omega), but increasing the impedance will often increase the gain slightly and file:///H|/SOS%2004-07/ART%20Digital%20MPA.htm (1 of 4)9/22/2005 8:55:04 PM

ART Digital MPA

also impart a sense of warmth. By contrast, a lower impedance tends to produce a slightly thinner, tighter tone. The unit has a curiously semi-retro look, with large metal knobs and mechanical VU meters on the one hand providing a counterpoint to modernistic LED illuminated perspex buttons and LED bar graphs. Class-A input stages are popular in mic preamps, because they don't suffer from crossover distortion and they don't use huge amounts of negative feedback, which has been known to compromise transient performance. Here, the circuits have been designed to give low noise, not only at the high gain settings normally quoted in the spec sheets, but also at medium and lower gain settings, where some less sophisticated designs fall down. The quoted Equivalent Input Noise (EIN) is -133dBA, which compares well against the more typical values of around -128dB quoted for the majority of mixers and mic preamps. From the input stage, the signal feeds into a tube gain stage where the user is presented with another choice, this time between high and low plate voltages. Low plate voltages impart a slightly artificial, but often musical, warmth to the sound, while higher plate voltages produce more of a traditional tube sound, which I think is far more subtle. Interestingly, the audio bandwidth extends to 48kHz in low-voltage mode and to well over twice this figure in high-voltage mode, though where the digital output is used the final frequency response will naturally be determined by the choice of sample rate. Balanced XLR and jack inputs are available on the rear panel to accept mic- or line-level sources, while a further unbalanced, high-impedance instrument jack is available on the front panel. Connecting to this overrides the rear-panel connections. There's also a TRS insert jack for placing compressors or equalisers in the signal path. The gain adjustment comprises a rotary control, with a 45dB gain range, followed by a +20dB gain switch, where the latter brings in additional gain via the tube circuitry. There's also a further 10dB of gain available using the Analogue Output control, so overall up to 75dB of gain is available. A single-pole (6dB/octave) highpass filter, variable between 10Hz and 200Hz, is available for each channel independently, as are +48V phantom powering and phase reverse.

Digital Facilities Digital outputs are not uncommon on modern mic preamps, though in many cases they are provided as plug-in option cards. Here everything is built in as standard, and in addition to the usual AES-EBU XLR and S/PDIF co-axial phono outs, there's also a dual-function optical output connector. Depending on the setting of the front-panel selector button, the optical output either carries a file:///H|/SOS%2004-07/ART%20Digital%20MPA.htm (2 of 4)9/22/2005 8:55:04 PM

ART Digital MPA

traditional stereo S/PDIF signal or an ADAT-format signal. The rear panel also includes an ADAT input, and when ADAT mode is selected, anything coming in on ADAT channels three to eight is passed straight through to the ADAT output, while the two mic preamp signals are sent out on ADAT channels one and two. When using the S/PDIF or AES-EBU formats, the sample rate may be selected by means of a rotary switch on the front panel to be 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz, or 192kHz. There's also a BNC word-clock input as well as another BNC connector to pass the word clock along to another device. When locking to the ADAT optical input, the sample rate must not exceed 50kHz, which should accommodate the varispeed range of a 48kHz ADAT machine. The signal feeding the digital converters passes through another gain control with a range of 'off' to +10dB, so that the converters can be set to operate at an optimum level as shown by the digital level meters. Where required, the 24-bit signal may be dithered down to 16 bits within the Digital MPA (selectable via a front-panel button), though in most instances it is best to record and mix at 24-bit resolution, dithering down to 16 bits immediately prior to CD burning. LED ladder meters monitor the digital output level, while the VU meters can be switched to read the analogue output level or the tube stage output level. In tube mode, there is around 15dB of headroom above 0VU before the tube saturates in highvoltage mode, or 10dB headroom in low-voltage mode, so the meters give a fair idea of how much tube flavour is being added to the sound. In high-voltage mode, the distortion is actually very low until the signal gets within 6dB of clipping, above which the tube acts somewhat like a soft limiter. In low-voltage mode, the increase in distortion is more progressive.

In Use I tested the Digital MPA with different models of large- and small-capsule capacitor microphone, recording voice, acoustic guitar, and accordion, as I had some obliging musicians on hand. I was generally impressed by the clean, transparent sound of the unit, which I used mainly in high-voltage mode, though the low-voltage option is also sensibly subtle and adds a pleasant warmth that doesn't get messy unless you really drive the tubes too hard. The variable input impedance produces a noticeable but again quite subtle effect, and I found all my mics sounded best when the impedance knob was close to, or just above, its centre 600(omega) position. Once you get beyond a certain quality of mic preamp, it can take a long time to judge exactly how good the performance is, as there are numerous factors that can influence the sound you're getting. However, the Digital MPA achieved a good sound right off with little or no effort, which is always a good sign! In fact the only feature I really missed was a headphone output, which I find helpful when setting up. The quality of the converters also seems to be good, as I made a number of recordings by feeding the S/PDIF output directly into my MOTU audio interface file:///H|/SOS%2004-07/ART%20Digital%20MPA.htm (3 of 4)9/22/2005 8:55:04 PM

ART Digital MPA

and got great results. However, it's probably fair to say that the performance at the higher sample rates probably won't give a huge increase in quality, because to take advantage of high-resolution audio you need to use extremely accurate and sophisticated converters, which invariably cost many times more than this unit does. Nevertheless, the Digital MPA works well up to 96kHz, but what was most important to me was that it sounded good at the real-world sample rates of 44.1kHz and 48kHz.

The variable input impedance of the Digital MPA allows you to achieve a variety of different tonal colours from a single mic.

The ART Of Preamp Design The Digital MPA is a very clearly presented piece of equipment that does everything you'd expect of a microphone preamplifier. The versatile tube stages are useful in shaping the tone you want from your microphone, as is the variable input impedance, but behind all these options the basic sound is good. Having a digital output as standard is always welcome, but ART have gone the extra mile here too by including ADAT support and optical S/PDIF options, so the Digital MPA should be able to integrate into just about any kind of recording system. Realistically, you're never going to get truly high-end performance at this UK price, but what ART have achieved within their design budget constraints is extremely creditable. The Digital MPA is a good-sounding all-rounder of a mic preamp with the bonus of some user adjustment to influence the tonality of the end result. I discovered no obvious weaknesses when using it on typical studio sessions, and when it is teamed up with some decent mics it is capable of greatsounding results, which is after all the bottom line. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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In this article:

Description Sounds Patches & Performances Polyphony & DSP Usage Beneath The Covers The Editor & Its Modules Morphs In/Out Group Note Group Oscillator Group LFO, Envelope & Shaper Groups What Next? Filters & Mixers Switch, Level & Logic Modules Sequencer Group Effects Group Delay Group Model Behaviour MIDI Group Conclusion

Clavia Nord Modular G2 £1595 pros Much improved performance controls. Great new modules, including DSP effects, acoustic modelling and MIDI Out! More stand-alone accessibility than its predecessor. Combines the interface and programming potential of a software synth with the portability and performance of a hardware instrument. Still sounds great. More memory locations; can store complete multitimbral setups.

cons

Clavia Nord Modular G2 Software-controlled Modular Synth Published in SOS July 2004 Print article : Close window

Reviews : Keyboard

The Nord Modular offered a classic blend of flexible software and well-designed hardware in 1998. But can the improved G2 keep up with the soft synths of 2004? Paul Nagle

When I first encountered the Nord Modular, it seemed to herald an exciting new era in synthesizer design. In 1998, it was a brave and innovative step to offer an affordable synth system which used drag-and-dropbased software to construct its patches while leaving the synthesis to the DSP in the accompanying controller keyboard. However, rather than take up the challenge, the majority of hardware synth manufacturers steered clear of this concept, until computers became powerful enough to start running synthesis software unaided by external DSP. Since then, most of the innovation in synth design has been in software-only instruments, with one or two notable recent exceptions, like Hartmann's Neuron or Roland's V-Synth.

Like its predecessor, the G2 is controlled by very DSP-light front-end software, all the synthesis calculations taking place in the hardware keyboard.

Undaunted by this, Clavia have decided to demonstrate their faith in their parthardware, part-software approach to synthesis, and have forged ahead with a 'second-generation' Nord Modular, the G2. This is just as well, as the 1998 Modular is now looking rather long in the tooth compared to some of 2004's software-only synths. The G2 therefore represents an opportunity for Clavia to

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Limited polyphony a frequent problem in many patches — that forthcoming voice expansion might be necessary if polyphony is an important purchase factor. No sample RAM — a missed opportunity in my book. Doesn't import patches from the original Nord Modular. The G2 is currently still missing some useful modules from the original Modular.

summary A remarkable synthesizer in terms of programming potential, accessibility, configurability and versatility. There are more 'ilities' I could add, but you get the picture. If you're an adventurous sound designer, or if you prefer to trust hardware for live performance, or both, the G2 is a dream come true.

bring their Modular concept up to date for sound designers, but of course with its hardware keyboard, it also has advantages to offer over software for live performers. It could be a dream instrument for both programmers and performers — so how does it fare? Although there is much that is familiar from the original Nord Modular, Clavia have opted for a drastic overhaul with the G2 — most strikingly in the areas of control and performance — and have responded to practically every criticism levelled at the earlier model. There are one or two omissions, but we'll come to those in due course. Before starting, it's important to understand that this is a complex, powerful instrument with over 150 modules currently available. Fortunately, many of these are comparable to those of its predecessor, so it would be helpful to re-read the earlier SOS reviews, starting with the initial two-parter in April and May 1998, and also the later version from July 2000 that brought the Apple Mac into the fold.

Description

There are already three types of Nord Modular G2 (see the 'Model Behaviour' box at the end of this article), but this review will concentrate on the G2 — by information happy coincidence the version I would have personally requested. This model Clavia Nord Modular G2, has a three-octave keyboard with aftertouch and velocity sensitivity (including £1595; G2X, £TBC (but expected to be in the region release velocity), plus Clavia's expressive pitch stick and modulation wheel, as of £2200); G2 Engine £795; seen on the Nord Lead series. Octave Shift buttons extend the keyboard's range Voice Expansion option by ±2 octaves when driving the internal sound engine, and in conjunction with the £295. Prices include VAT. Shift key, this transposition can be applied to externally controlled instruments Hand In Hand +44 (0) too. 1579 326155. +44 (0)1579 326157. Click here to email

www.handinhand.uk.net www.clavia.se

Test Spec G2 OS and Editor version reviewed: v1.10.

The original Nord Modular communicated with its software editor via a dedicated set of MIDI ports. Not only did this confuse some users (and some MIDI interfaces), but it felt like a workaround awaiting a better solution. Fortunately, that's just what the G2 offers — its USB connection provides all the necessary connectivity with the editor, leaving the MIDI ports to do their traditional thing. There's even a MIDI Thru socket this time! I'm pleased to report that during the review period, I never once lost contact with the modular — and I certainly couldn't say that with the last version (3.03) of the original Modular's editor. The G2 editor only supports one attached Modular — unlike the older editor, which could address up to four. The G2 has a generous four audio inputs, plus a tight-fitting XLR mic Input, complete with built-in preamp and dedicated level control. If used, this XLR connection disables Input 1. Adjacent to these, on the busy rear panel, are four audio outputs. No digital I/O is present, but the 24-bit, 96 kHz A/D and D/A converters do at least give excellent audio quality.



Gone from the front panel are the 18 unlabelled knobs and lone two-line LCD of the former Modular. The G2 boasts multiple displays and a panel endowed with rotary encoders and buttons galore. Replacing 18 knobs with eight knobs and some buttons might seem, at first glance, like a poor exchange. However, the G2 borrows a concept seen on such classic The Nord G2's user interface is a vast instruments as the Oberheim Xpander improvement over that of the original Modular, with its endless encoders and LED — a series of small displays above the collars, and displays above each of the encoders that change according to the encoders so that the control legending can assignment of each encoder. You be changed at any time. choose which controls are important and make them available for tweaking on a patch-by-patch basis; the result is that five separate 2x16 displays manage to convey more useful information at a glance than many instruments with a fixed set of controls. In conjunction with a row of Nord-Lead-3-style endless rotary encoders surrounded by a ring of LEDs, you can see up to eight values at any time. Buttons underneath the encoders can quickly toggle values and are ideal for on/off-type functions. Jumping between 'pages' of assigned controls is as quick as hitting buttons; a series of five buttons at the right of the top panel are labelled Osc, LFO, Env, Filter and Effect, and each of these has a further three sets of pages where knob assignments can be stored. When you select a different 'page' of controls, the knobs' LED collars and the LCDs update to reflect their new values. In total, 120 parameters are accessible directly in this way, and this transforms the G2 into a performance instrument par excellence compared with the old Modular. Your knob assignments needn't be confined to those of the button labels either — you are free to assign any control to any page, although it makes sense to use them as far as possible. On the G2, far more actions can be performed without ever needing to switch on your computer. Knob and Morph assignments can be performed directly from the G2 hardware (for more on Morphs, see the box over the page). As before, the complete patch can be edited from the hardware alone; if you press the Patch button (adjacent to the red Store button), the navigation keys will take you around every module, where you can make tweaks, knob assignments, morph assignments and so on. The additional displays give far greater parameter visibility than the old single-LCD version, and editing a patch this way is surprisingly painless. However, what you cannot do is change the modules or cabling — for that you need the software editor, which I'll come to shortly. Normally, the multiple displays show module names and their assigned parameters, with parameter values represented by the position of the lit LED in the encoder ring. By pushing the Display Mode button, you can read the numeric value of the parameter instead, with the parameter name replacing the module



name. The G2 architecture includes a built-in arpeggiator and vibrato oscillator plus glide and bend settings. You don't have to use up any modules to enjoy these — they are always present and accessible from the Patch Settings button. The arpeggiator is basic, as we've come to expect from Clavia, but it's great that it's there. Similarly, using the built-in vibrato LFO might be a valuable resource saver. Other Patch Settings include level, the global clock tempo, and the number of voices you've requested for your patch, although as on the previous Modular, whether you get the polyphony you want depends to some extent on the DSP load you place on the G2 when you program it (see the box below).

Sounds Factory sounds on an instrument such as this are always going to be a varied bunch; with the tools on offer it's hard to imagine any two programmers taking the same approach. Thus, a quick spin through the factory sounds reveals familiar Nord tones, acoustic guitar simulations, complete sequenced performances, many analogue emulations, electric pianos and so on. Many of these patches also include useful Variations. Here are a few choice examples, which you can also hear at www.soundonsound.com/soundbank. 'Haunted': Various vowel filters are employed in this eerie pad. 'Koto' and 'Acc Guit': These two make good use of the string oscillator. 'BackTo72': A very analogue-sounding solo patch. 'StringVariations': A highly playable collection of solo and ensemble strings. 'G2Padding': A rich, warm pad. 'Trance2_DZLW': Five sequences including synth voices and percussion. 'Alien': A simple but atmospheric patch using stereo delays and reverb. 'Transporter': Beam me up, Scotty... 'Feng Shui': An impressive series of sound effect variations. 'YetAnotherOrgan': Like the name says... 'Bells': Deep, realistic bells created by no less than 11 oscillators. 'Arpeggiaperc': An example of the arpeggiator in use (with delay). 'Fat Bass NL2': Just in case you've forgotten how good the Nord filter sounds... 'Fat Lead 1': Some lead variations that should cut through pretty much anything. 'Laika': Strummed string oscillators make up this subtle, evocative Performance. 'Beatbox': Another Performance ably demonstrating use of the sequencers and percussion. 'Jinglebellscape': A marvellous evolving Performance. I won't say any more, just listen...

Patches & Performances



Sounds are organised into 32 banks of 128 patches. If all these were filled, there would be a total of 4096 patches. However, there is insufficient memory for this number, and the manual suggests that 1200 is a more feasible maximum to expect. Nevertheless, it still makes a lot of sense for the G2 to organise its memory this way, as related patches can be grouped in the same bank for ease of access. You can select patches either by the order in which they are stored, in alphabetical order, or by category (there are 15 to choose from); you use the G2's Shift key in combination with the Down navigation key to toggle the selection criteria. The Up and Down keys then select different banks, the letters of the alphabet with which patch names begin, or categories. Patch selection is made by turning the patch encoder and hitting the Load button when you reach your choice. As on the previous Modular, up to four patches can be running on the G2 at once, each in a memory location that Clavia refer to as a Slot. This means that the G2 is up to four-part multitimbral, as each Slot is able to hold a separate patch running on its own MIDI channel. You can also layer patches by holding down several 'Active Slot' buttons simultaneously. Performances are new to the G2 — and what a welcome addition they are! Finally, a single selection can serve up a complete four-part multitimbral setup. There are 1024 Performances in total, arranged into eight banks of 128. Performance mode is accessed via the button of the same name, and you then make your selection in the same way you do for patches. Splendidly, the four patches in each Performance are not merely references to the main pool of patches — they are memories in their own right. Similarly, knob assignments can be freely made for the entire Performance — so you might create a set of mixer assignments gathering all patch volumes onto adjacent knobs, for example. Just as with patches, memory restrictions may mean you do not have enough space to fill every one of these Performances. Having worked with the previous Nord Modular for some years, I often made collections of patches based on favourite configurations of modules. So how I longed for something like the G2's Patch Variations — of which there are eight in every patch. Each Variation can have unique parameter values, knob assignments, morph settings, and so on. The only restriction is that modules and connections are fixed across all Variations. Nevertheless, this really extracts the maximum mileage out of the patch locations on offer. Variations are accessed by a row of dedicated buttons, by the left and right navigation keys, or via MIDI controller 70, and are stored automatically when the patch is saved. Unlike loading a whole new patch, Variations do not cause the DSP to be recalculated, so you can switch between them far more quickly and smoothly.



Polyphony & DSP Usage The G2 is described as 'up to 32-voice polyphonic', but typically, the patches I created in the software editor (see over the page) offered between eight and 10 voices. Once you start to add complexity and/or effects, polyphony can drop to a meagre three voices all too easily! I endeavoured to build a basic 32-voice polyphonic synthesizer with two oscillators, two envelopes, and a filter and mixer section — but as soon as I added even the most basic LFO, polyphony dropped to 28 notes. Adding one of the more complex LFOs (LFOShpA), left me with just 20 voices — and this was before I dabbled with effects or other 'frivolous' modules. In common with the older Modular, the G2 cannot dynamically organise its polyphony. As a patch is loaded into a slot, the synth's entire resources are recalculated, momentarily silencing any output. So to gain greater polyphony, it is wise to deactivate unused slots. This is done using the Shift and Slot keys together. One feature that didn't get ported from the old editor was the floating DSP count that informed you of the processing requirements for each module. This restriction may be due to the fact that many of the G2 modules are self-optimising according to the role they perform, and so no one figure for load can be applied to a given module. The G2 editor therefore requires that you learn a more trial-and-error approach: you become aware of the typical DSP needs of each module by watching the Patch Load section on the Editor's display. Fortunately, swapping one module for another is a doddle courtesy of a pull-down menu available at the top left-hand corner, and all comparable cable connections are remade automatically, which is neat. With practice, and after gleaning tips from the factory patches and excellent online user community, I began to discover better ways to use the available resources. For example, in a multitimbral setup, choosing a single slot to provide effects and then routing patches into this slot (via the internal buss system) means the available DSP stretches that bit further. Furthermore, some modules have drop-down menus (such as some of the oscillators) and these use fewer resources compared to those with radio buttons — but the payback is that when you select a new item (such as a waveform) from one of these menus, the G2 goes briefly silent whilst it recalculates the load. There is a theoretical limit of 254 modules per patch — 127 in the Voice Area and 127 in the FX Area, but in practice, you will max out the Patch Load way before you use all of those! The DSP load for the Voice Area part of the patch is represented by the 'VA' box, and that for the FX Area by the 'FX' box. There are separate boxes for Patch Load and Memory but the end result is the same — if either area goes red due to DSP overload, the patch will be muted until you delete some modules.

Beneath The Covers Internally, what has changed? Well, the G2 has RAM set aside for effects such as reverb and a delay of decent length. It has an internal buss system so that



signals can be routed internally amongst its four Slot memories and it is also now capable of generating MIDI information from its internal modules — sequencers, LFOs, envelopes and so on. Add in all manner of tweaks to the editor software (which was already a fine working environment) and you can begin to appreciate that Clavia have not been idle. Improvements are still being made. Not all the former modules have been ported over from the first generation of the Modular, and although Clavia have claimed that most will reappear or equivalents will be implemented, work is still in progress as I write this. More seriously, there is no way to import patches from the original, short of manually recreating all your patches by hand — a significant disappointment if you want to upgrade and have a large collection of your own patches. The G2 also doesn't have any sample RAM, so there is no possibility of incorporating short audio samples into oscillators. In my opinion, this would have made the G2 far more 'complete', but Clavia didn't seem terribly keen on the idea when I put it to them, perhaps because the G2 has no means to store samples on board, or perhaps because they have a specific philosophy that precludes the use of samples. But who knows what the future will hold, and whether the DSP currently allocated to reverb and delay could be cunningly reprogrammed to allow a short, volatile sampling capability? After all, we've even seen sampling turn up on some of the latest hardware modulars, such as those from Doepfer and Analogue Systems. Finally, SysEx dumping of Patches and Performances is now a reality — the function is invoked by holding down the Shift and System buttons. The data duly spills from the MIDI Out socket, at which point you can store it in your sequencer and thus ensure the correct sounds are available for each song. This was a feature much requested by owners of the previous Nord Modular.

The Editor & Its Modules The editor component of the G2 is the front-end software required to create new patches, assemble collections of modules and cable them all together. It requires a PC running Windows 98SE, 2000 or XP and should be at least a 450MHz Pentium II with 64MB of RAM and a USB 1.1 port. This is not terribly demanding by today's standards, and both my main studio PC and my ageing laptop coped with it effortlessly. Installation of both the required USB driver and the Editor went without incident, the first being handled by the usual installation wizard and the second by the installer on Clavia's supplied CD. Incidentally, you can download and install the software yourself if you're interested in seeing what it looks like, even if you don't have the synth present.



Once you're up and running, you'll notice that the currently selected patches are loaded into the editor automatically. The patch that is, as Clavia put it, 'in focus' — that is, the one that is ready to edit via the synth's panel — is also 'in focus' in the editor software (note: it is possible to play one patch and edit another if you alter the focus using the Focus/Copy button on the panel). There are 15 different module groups accessed via tabs running along the top of the window. Clicking on a tab reveals all the available modules, and holding the mouse over an individual module icon displays it in actual size. To add any particular module to your patch, you either double-click on it or drag it to the position you want on the workspace. You can set up so-called Initial Patches containing favourite modules — although only two of them — so that you can get on quickly with the business of patch construction. This is similar to the concept of Templates on the old Modular, but having just two of them seems rather stingy. In common with v3.0x of the old Modular's editor, the G2 editor's desktop workspace is divided into two areas, Voice and FX, between which you can toggle using the 'V' and 'F' keys on your computer keyboard. The Voice Area comprises the parts of a patch that are common to each voice — oscillators, filters and so on — and typically this is where you would build your main synth. The FX Area, on the lower portion of the screen, features modules that are common to all voices. As you rarely want a separate reverb on every voice of a synthesizer, this is the logical place to build your effects configurations. In practice, the only restriction is that the FX Input module must be placed in the FX Area — otherwise, you are free to build your patch how you like, available DSP permitting, of course. So if you do want a separate effect on every voice of your polyphonic synthesizer, the software won't stop you placing the modules accordingly — but you might not have enough spare processing power for the resultant patch to work! As explained in the 'Polyphony & DSP Usage' box on the previous page, it pays at all times to keep an eye on the Patch Load and Memory boxes while constructing a patch — these show the overall DSP allocation. Modules can now be coloured in a fixed selection of rather fetching, pastel hues. This is handier than it may seem, particularly in speeding up navigation through busy patches. Many modules have on/off controls that are invaluable for programming, as they effectively deactivate the module but maintain its connections. Of the other editor features, the Parameter Overview window deserves special mention — not least because it is the slickest method to assign the G2's physical controls. Using drag-and-drop techniques, this menu offers the speediest way imaginable to assign and view all 120 parameters. If you want to record parameter tweaks to an external device or sequencer, the neatest way is with the Auto Assign MIDI Controllers function. This is far neater than individually assigning a MIDI controller to each knob, although this is still



possible by right-clicking on the on-screen controls. In the next version of the editor software, this feature is apparently due to be improved still further — the plan is to incorporate it into the Parameter Overview window. I won't assault your senses by attempting to describe each and every module. Instead, I will list each group in turn, and then cherry-pick, with special emphasis on the most interesting new arrivals.

Morphs Morphs are a much-loved Clavia feature intended for use in tweaking multiple parameters simultaneously over pre-programmed ranges and in pre-determined directions. There are now eight morph groups available — the sources (Mod Wheel, Keyboard, Aftertouch, Velocity, and so on) are listed underneath the Variation buttons. Morphs can control a maximum of 25 patch parameters. In keeping with the improved user interface of the new modular, morph assignments can be made on the hardware as well as in the software editor. The Morph button transforms the Variation buttons into Morph Source selection controls, at which point you hold down the relevant button and turn an encoder to set the assignment.

In/Out Group These modules (2 Output, 4 Output, 2 Input, 4 Input, FX In, Keyboard Voice, Keyboard Mono, Device, Status, Note Detect, and Name) contain the I/O routing for audio and MIDI control signal inputs. In the G2, the physical inputs and outputs are no longer the only means of accessing audio. Clavia have provided a much-requested feature — a fourchannel internal audio buss with which you can route signals from one slot to another. This is present on each of the Input/Output modules. The FX In module is equivalent to the old Poly Area In module. It is used to route the output of the Voice area to the FX area of a patch. Particularly worthy of note in this section is the MonoKey module, offering Last, Low and High note outputs. This is far more flexible for solo patches than the old Nord Modular, which featured only last-note priority. The Device module presents a series of outputs from the performance controllers, such as the mod wheel, aftertouch, and so on. Despite having just three outputs, the Status module is enormously useful. Its first output socket, Patch Active, emits a logic signal when a patch is loaded into a slot; you could use this to trigger an action such as starting a sequencer. The file:///H|/SOS%2004-07/Clavia%20Nord%20Modular%20G2.htm (9 of 18)9/22/2005 8:55:08 PM


Var Active output produces a logic signal when a Variation is activated, and can be used in a similar way. But the most interesting is the Voice Output module, which generates a control signal corresponding to the voice number currently played. There are several modules that make use of this function specifically, such as the Multiplexer and the Control Sequencer, and the manual offers examples where these modules are used together to produce polyphonic patches in which each note has its own detune setting or waveform. Thus you can create the kind of subtle effects previously heard on synths such as the Oberheim Four-Voice, where each voice was a completely separate synthesizer. The G2 method is far easier and quicker! Finally, there is the Name Bar — a floating text box used to add documention to your patch, describe the Variations or to present information in the display windows.

Note Group The Note Group contains various pitch-related modules (Note Quantiser, Key Quantiser, Partial Quantiser, Note Scaler, Glide, Pitch Tracker, Zero Crossing Counter, and Level Scaler). The first few I'll describe aren't actually new, but when combined with others — especially the MIDI Out modules — they become more useful than previously. The Note Quantiser module takes a continuous signal and renders it into semitone steps. So an LFO could be used to generate glissando effects, for example. The Key Quantiser goes one step further; send it a range of notes and it will output notes that are confined to a specific key. A small keyboard display is used to determine the notes that will be permitted. The Partial Quantiser requires either a very short or very long explanation, but it is far more versatile than the manual's brief entry suggests. It is a control signal processor that can be used to step through an oscillator's harmonics a partial at a time, and thus create glissando or arpeggio-type effects. You can even feed the module negative control signals, thereby stepping through undertones of your oscillator's output, ie. harmonics at fractions of the oscillator's fundamental frequency. Pitch Track and ZeroCnt are modules that extract the pitch from an input signal. They function rather like the Korg MS20's Frequency-to-Voltage converter, and are equally difficult to obtain precise results from, but they're fun all the same. With clear, monophonic signals the results can be quite useable — anything else tends to produce warbly bagpipe impersonations.



Oscillator Group This group of oscillators (specifically oscillators A, B, C, and D, the Phase Modulation oscillator, Shape oscillators A and B, and the Dual, String, Percussion, Noise and Master oscillators, as well as a complete drum synth module) contains the G2's main sound sources. Some of these conform, more or less, to those of a traditional well-stocked analogue modular, but there are also several less conventional inclusions. The module offering a bank of sine waves featured on the original Modular is missing, and Clavia reveal that two A typical patch in Nord Modular G2 Editor of the modules documented in the (although note that this screengrab was done on a dual-monitor PC system). The soundmanual are not yet present either — generating modules are cabled together on which is a shame, as these will place the left, while the processing modules are the voice structure of a DX7 inside the situated under the grey bar. On the right you G2, complete with 32 FM algorithms can see the virtual on-screen keyboard and and Operator controls. I had a brief the Parameter Overview module, which is handy for making control assignments on the preview of beta versions of these G2 hardware. modules and they look impressive, but weren't sufficiently finished at the time of this review to form any definitive conclusions. The oscillators can receive pitch from the keyboard without requiring any patch connections but, if you wish, deactivating the Kbt (Keyboard Tracking) button removes this internal connection. All expected 'analogue' waveforms are present, plus several variants. Oscillator B features a double sawtooth wave. The shape of this can be altered using either a knob or the modulation input; the two sawtooth waves then shift their phase relative to each other, producing a warm, swimmy effect. The Phase Modulation oscillator produces FM-like tones, whereas the Shape Oscillators also feature a host of waveforms whose shape can be modulated. Sometimes this produces effects similar to filtering, while at other times, harder, harmonically rich timbres emerge. These modules replace the first-generation Modular's Spectral Oscillator, and have more waveforms. The Dual Oscillator produces two simultaneous waveforms plus a suboscillator. You can modulate the pulse width of the square wave and the phase of the sawtooth — so this is a pretty fat-sounding module even before you mix in the sub-oscillator! The latter's 'soft' setting reduces the harmonic content as if a low-pass filter had been applied. Surely the most interesting member of this group is the String Oscillator. This module is actually a pitch-controlled delay line with feedback. It needs to be file:///H|/SOS%2004-07/Clavia%20Nord%20Modular%20G2.htm (11 of 18)9/22/2005 8:55:08 PM


'excited' by an input signal, and will then produce tones that vary depending on the signal chosen. I found bursts of pink noise or short, percussive sources were good starting points for creating realistic plucked-string tones. The decay parameter sets the time of the internal feedback signal and the damp parameter governs high-frequency damping. Some of the factory patches, such as the kotos and acoustic and bass guitars, ably demonstrate the quality of this module.

LFO, Envelope & Shaper Groups The LFO modules (LFO A, B, and C, Shape LFO A, Clock Generator) are fairly self-explanatory and this time around, MIDI Clock synchronisation is available to three of them — LFOA, LFOB and ShpA. The LFO section also contains the Clock Generator, but lacks the random modules and the Pattern Generator of the earlier Modular. The various envelope types (ADSR, Hold, Decay, AD/R, AHD, ADDSR, Multi Stage, AHD-Mod, and ADSR-Mod) behave as you'd expect from their names, so there's no need for lengthy explanations here. Their output signal polarity and shape are set by clicking on the small Type and Shape buttons, giving bi-polar envelopes, inverted envelopes and exponential or linear curves for different envelope stages. In common with the oscillators and LFOs, the envelopes have a pre-wired connection for ease of use; they are triggered when you play a key or when MIDI notes are received (assuming the 'Kb' button on the module is active). The Shaper group of modules (Clip, Overdrive, Saturation, Exp Shaper, Wavewrapper, Static Shaper, Rectifier, and Shaper) includes various soundshapers that are familiar from the first-generation Modular. For example, there's Clip, which produces digital distortion, Overdrive for a warmer distortion, and Wavewrapper, for those FM-like distortions full of overtones.

What Next? Clavia have announced a Voice Expansion board for the G2 which doubles its polyphony, although it was not available at the time of this review. The Expansion board features four extra DSPs and RAM circuits and is user-installable. It is also compatible with the new rackmount Nord Modular G2 Engine. What's more, by the time you read this, the Mac OS X version of the editor should be well on the way. It will work in the same way as the Windows version, using USB for connectivity, and should be available for free at www.clavia.se. I was lucky enough to see details of some forthcoming modules during this review; these include pitch-shifters, random LFOs, a MIDI automation module, a new noise source and DX7-style FM modules. If the long development life of the previous Nord Modular is anything to go by, there will be plenty more to come.



Filters & Mixers The familiar Nord sound comes from the combination of its bright, clear oscillators and its distinctive filters. There are many types in the Filter group of modules: Low-pass, High-pass (each with adjustable slopes ranging from six to 36dB per octave), the Nord and Classic filter modules, a multi-mode filter, and a static 12dB-per-octave type, as well as a wah-wah, voice filter, vocoder, simple two and three-band EQs, and a comb filter, amongst others. Many of these are the same as they were on the original Modular, or have only received small tweaks, and the old fixed filter bank has been dropped altogether. The wah-wah and comb filters are new, the latter offering adjustable distance between its peaks and notches, and adjustable feedback. As filters are so integral to a synth's character, I must say that I'd have hoped for more advances in this group. The Mixer group offers a total of 12 mixers of various types, ranging from one to eight channels. There are mixers with sliders or attenuator knobs, plus pan and crossfade modules. The G2 doesn't differentiate between control and audio mixers — it adapts intelligently according to the signals present. The inputs and outputs even change colour to reflect the signal carried — red for audio, blue for control signals. As on the previous Nord Modular, internal audio signals have the highest priority, and are sampled at 96Hz. Control signals are sampled at 24kHz, so the DSP patch load is lower for mixers that do not carry audio.

Switch, Level & Logic Modules The primary function of switches is to make or break connections in a controllable way. The G2's Switch group contains various Switch modules, including momentary, toggling and value switches, all of which operate depending on the level of input control signals. There are also Sample & Hold and Track & Hold modules. If a switch has a label box, right-clicking on the box offers a 'name' option, which can be used to add text that will be meaningful in the G2's display. So if you set up a switch with four inputs as a selector for modulation sources, you could name the sources 'LFO1', 'LFO2', 'Env', 'Vel', and so on. This is yet another small but valuable interface improvement. The Window switch is highly versatile; it makes a connection and simultaneously sends a logic signal whenever an incoming control value falls between the From/To settings specified. The Multiplexer is another fun module, routing a selection of inputs to outputs (or, indeed, a selection of outputs to inputs). Its connections are determined on receipt of specific control signals at its 'Ctrl' input.



Like Switches, the Level group of modules isn't particularly glamorous, yet its constituents are vital components in any modular system, performing various transform functions on signal level. Amongst the included modules are types for adding, amplfying, multiplying and modulating the level of input signals, as well as a noise gate and envelope follower. There's also a group of Logic modules, including a Logic Gate, Inverter, Binary counter, and eight-bit A-D and D-A converters. These all have simple high or low states (corresponding to +64 or 0 values in their input signals). For more on how such binary-driven logic systems work, see April's instalment of Synth Secrets, or look up: www.soundonsound.com/sos/apr04/articles/synthsecrets.htm.

Sequencer Group Five different sequencers (Event, Value, Level, Note, and Control types), each with a maximum of 16 steps, are used to provide loop and pattern-based material of all kinds. If you need more than 16 steps, sequencers can be chained together. In such cases, the gate and control signal inputs are used to merge their outputs, and thus avoid the need for a mixer. The Park input is new; when a high logic signal is received at this input, the sequencer will stop. And if a high logic signal is received at the Reset inputs, the sequencer will restart at step one on receipt of the next clock pulse. The sequencers are very similar to those of the previous Nord Modular, and I'd have welcomed more features, such as a direction switch, the option of Morphing the length of a sequence, or the inclusion of a Step Skip option (as seen on old Moog or ARP sequencers). Yet with innovative application of what's on offer, you can still create marvellous, complex patterns.

The rear panel of the G2 is also improved over that of the original Nord Modular — there are two extra audio inputs, a dedicated XLR mic input, and a MIDI Thru. Best of all, the interaction with the G2 editing software is now handled via the USB socket, rather than by the former Modular's confusing second pair of five-pin MIDI connections.

When you assign Note Sequencer modules for control via the G2 encoders, you can see each step's actual note value on screen. The buttons underneath are used to toggle the on/off status of steps whilst LEDs adjacent to the knobs follow the sequence progression. The Control Sequencer is particularly interesting, in that its next step is determined by the values of incoming control signals, rather than by a conventional clock pulse. To step this module forwards, you send a rising sawtooth waveform to the control input, while a falling waveform steps it backwards. Additionally, the module's Xfade button slews the output between steps, smoothing the transitions.



Effects Group This group comprises a straightforward selection of effects modules (Stereo Chorus, Phaser, Flanger, Digitiser, Frequency Shifter, Reverb, and Compressor). The Digitiser module can reduce the sample rate right down to 32Hz, and can be modulated via a control source. Quantisation can be adjusted between one and 12 bits or 'off', which means full resolution (24 bits). If you want dirt and grunge with minimum effort, this is for you. The Frequency Shifter performs additive frequency shifts (unlike conventional pitch-shifters, which perform multiplicative frequency shifts) like the Bode units seen in Moog and Analogue Systems modular systems (see www. soundonsound.com/sos/jan02/articles/synthsecrets0102.asp). It produces audible effects not unlike those of a ring modulator; used sparingly, these can add considerable richness or exciter-type effects to a signal. The Stereo Reverb module is a pleasant surprise. I was starting to think Clavia would never put reverb into a synth, but thank heavens they have — even if the quality is unlikely to give Lexicon any sleepless nights, this is a secondary consideration in the context of the G2. Reverb in a modular system offers far more than just a glossy finish, because you can place it anywhere in your synth's architecture — an option you don't have with an external unit. Small, Medium, Large and Hall types are on offer, with a maximum time of over 17 seconds. A Dry/Wet level and a Brightness control are the remaining options.

Delay Group Marvellous — a whole group of modules devoted to delay, with many types (Static Delay, Single Delay, 2 Tap, 4 Tap, 8 Tap, Shift Register, Clocked Delay, Delay A, Delay B, and Stereo Delay). Clavia have sensibly provided a range selector to help conserve the available RAM, and the maximum delay time is 2.7 seconds. The delay modules range from single delays to eight-tap and stereo delays. MIDI sync is available, but if you're clocking the G2 from an external source, you need to ensure you have a very stable MIDI Clock source. The delays do not respond well if you tweak their time parameters during playback, either from a front-panel control or (if sync'ed externally) from the glitches of an unstable MIDI Clock. In fact, some of these 'adjustment' noises are quite unpleasant, so I hope Clavia will add some kind of intelligent smoothing in the future. Several unusual delay modules are present, such as the Delay Shift Register. This has eight outputs and shifts the incoming delayed material through file:///H|/SOS%2004-07/Clavia%20Nord%20Modular%20G2.htm (15 of 18)9/22/2005 8:55:09 PM


successive outputs on every clock pulse. The Clocked Delay Register works in a similar fashion, except it has only a single output and you specify the number of clock pulses after which the value signal will be output.

Model Behaviour As well as the G2 keyboard reviewed here, Clavia also offer the so-called G2 Engine — a 1U rack unit device with all the processing power of the G2 but none of the controls. It is significantly cheaper, but effectively ties you to a computer and MIDI controller or master keyboard setup. The most recent addition to the range, the G2X (announced at the recent Frankfurt Musikmesse, but not available at the time of writing) will offer a five-octave velocity- and aftertouch-sensitive keyboard, and double the polyphony of the standard G2. This last feature is available courtesy of a Voice Expansion board that will be fitted as standard on the G2X (it's an optional add-on for the G2 and Engine). The G2X will also offer two additional assignable modulation wheels with position LEDs, and a gooseneck XLR microphone will be included in the package. Thanks to Rob Hordijk — Nord Modular guru — for his invaluable advice during this review.

MIDI Group At last, the module group we've all been waiting for — or I have, anyway. This group contains modules that control the receipt and transmission of MIDI data (MIDI Ctrl/Prg/Note Send & Receive, and MIDI Note Zone). In an analogue modular system, you can easily connect the output of an LFO to a variety of other instruments — all it takes is a voltage input and a cable. But how can you connect the LFO inside a MIDI synthesizer to an external instrument? With the G2, you can! The Control Send module can transmit MIDI controller information on a specific MIDI channel. So why is this such a good thing? Well, not only can you transmit MIDI information from the MIDI Out socket of the G2, you can also route the information to any of the four G2's slots (A-D) — or to the 'This' setting, which routes the data back into the current slot. If your curiosity is still not piqued, imagine you've programmed a morph to sweep the cutoff frequency, alter an envelope attack time, fade in some noise, brighten the reverb, or modify a wave shape. All of these changes can be controlled by a single source, such as the modulation wheel. But now imagine you directed the output of an LFO into a Control Send module, simulating mod wheel events and routing them back into the current Slot — you see where I'm going? Suddenly, the LFO is capable of controlling the morph automatically. If nothing else, it generates a lovely display of moving, flashing lights on the already impressive panel, but it also opens up an exciting world of evolving patches. And because you can control morphs with the LFO in this way, you can affect parameters for which there may not even be control signal inputs. Another idea that occurred to me was to route the output of an LFO to generate MIDI Controller 70, which, you file:///H|/SOS%2004-07/Clavia%20Nord%20Modular%20G2.htm (16 of 18)9/22/2005 8:55:09 PM


may recall, is used to select Variations. It could then switch Variations automatically whilst you play. The Note Send module works in the same way, transmitting note and velocity information. Connect several sequencers to Note Send modules, assign the G2's knobs to set the note values, and in no time at all, you have assembled a sequencer of awesome flexibility. Mix in some notes generated from an LFO or Sample & Hold module (routed through that old favourite, the Note Quantiser, to make sure everything remains in key) and you can create patterns of amazing subtlety and complexity. I hope that these simple examples demonstrate the usefulness of the MIDI output modules. As is probably evident, this is the kind of stuff I just love to explore.

Conclusion Clavia's first modular is a hard act to follow. The G2 falls into the category of 'logical progression' rather than 'revolutionary creation', but there are encouraging signs of what is still to come. Certainly, the inclusion of acoustic modelling is a move in the right direction; the string oscillator sounds great, and I look forward to further, more unusual, sound sources. Perhaps, if sampling really is a no-no, resynthesis or granular synthesis modules may one day appear. With its DSP effects, MIDI Output modules and the enhancements to performance controls and its user interface, the G2 must surely fulfil most of the wishes of original Nord Modular owners. Yes, we'd love more polyphony and yes, we'd love some radically twisted new modules; yes again, we'd like it for less money, but hey, how many kidneys do you actually need? And if budgets are tight, there is always the much cheaper Engine version as an option. The main difference between the G2 and a hardware modular is that the former is essentially a closed system; if you want to add a module that does something Clavia don't provide, you can't simply cable up a module from another manufacturer. Likewise, when Clavia finally move on to developing for a replacement hardware product, users will have to accept that further improvements to this system are likely to cease (as they have for the original Nord Modular). In the light of this, it is unfortunate there is no way to import patches from the older modular. If you have invested a lot of effort programming for that platform, it may not be quite as tempting to upgrade and start again from scratch. This time around, Clavia have concentrated on making the player's modular — the underlying concepts are broadly unchanged. Sure, new modules and functions have been added, but mostly it is the ease of use that has increased dramatically. Perhaps this will banish some of the fear that the more inscrutable first generation inspired. I found that, once it's configured to taste, you can do a



hell of a lot using the G2 as a stand-alone instrument — and when you do need to delve deeper, the editor is excellent. I'd hate to think of a future where the most thrilling instruments arrive on a CD and require a computer to give them a voice. I believe Clavia have done a great job with the G2, and maybe one day it will be remembered as an important stage in a hardware renaissance. I already know several musicians who are quietly advertising their grandmothers on eBay in order to afford a G2. If you want to get deep into complex synthesis, but prefer to perform laptop-free, the G2 could be the answer to your prayers — if not to Granny's. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Digidesign Command 8

In this article:

In The Flesh Connectivity More Control Stand-alone Mode Conclusions

Digidesign Command 8 Control Surface For DAWs Published in SOS July 2004 Print article : Close window

Reviews : Hardware Controller

Digidesign Command 8 £952 pros Full length, touch-sensitive, fast motorised faders with 1024-step resolution. 'Proper' continuous rotary encoders. Completely cross-platform controller — not just for Pro Tools... ...but highly integrated if you do use Pro Tools. Built-in MIDI interface. Fader caps can be replaced! Built-in transport and monitor router makes it ideal as remote control. Banking function in standalone mode doubles number of controls addressed by the surface.

With fader controllers, you often have a choice of buying something dedicated to one software package, or a generic unit that gives versatility at the cost of tight integration. Digidesign's Command 8 aims to offer the best of both worlds. Simon Price

Digidesign already offer a rackmounting variant on their Digi 002 interface, which offers the Pro Tools LE audio interface component without the control surface, and at first glance, their Command 8 appears to be the reverse: the control-surface element of the Digi 002 without the audio interface. cons However, while the Command 8 is Fader caps feel a bit cheap indeed similar to the 002's controller, and your fingers slip off them. the comparison doesn't do justice to the new unit's scope. Command 8 provides Pots slightly too large, and hands-on control of both Pro Tools LE and TDM software, with the same high therefore close together. level of integration and sophistication that we've come to expect since Pro No ready-made templates Control, Control 24 and HUI. A significant departure for Digidesign, moreover, is for third-party software. Can't use in conjunction with that Command 8 is not just for Pro Tools: it also runs stand-alone as a MIDI controller for any other capable software. other MIDI controllers in Pro Tools.

summary Pro Tools really comes into its element when used with a dedicated controller, and this is the least expensive Digi controller to date for LE and TDM systems. Despite the low price compared to its bigger brothers, surprisingly little is lost. What's more,

In The Flesh The first thing that struck me about the Command 8 was that it's much bigger than I'd expected. From the pictures on Digi's web site I thought it would be small, light and, dare I say it, plasticky, but in fact it's a hefty and impressive piece of kit. This impression is reinforced by the way that the power supply

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Command 8 doubles up as a very fine generic MIDI controller.

information £951.75 including VAT. Digidesign UK +44 (0) 1753 655999. +44 (0)1753 658501. Click here to email www.digidesign.com

screws securely into place, a simple but reassuring touch. Despite the obvious similarities with the 002, the design of the housing is better, moving from the other surface's slightly toylike 'scooped' appearance to something more traditional. So, initial impressions are mostly good, and here's a sneak preview: I'm going to say a lot of good things about the Command 8, and it's one of the best all-round controllers I've used. That said, let's deal with my biggest criticism up front! The fader caps are weird, looking and feeling a bit like squashed beer-bottle tops. Maybe they're supposed to be quirky and different, but while their visual appeal is a matter of opinion, they fall short on a practical level. The main problem is that your fingers tend to slip off them, and you have to resort to holding them between finger and thumb, making it fiddly, and hard to control several faders at once. I know Digi are keen to maintain the status of their high-end controllers, but I don't think it would hurt for them to use the same caps as they do in their Pro Control and do justice to the rest of the unit. Anyway, the good news is that the fader caps are attached by standard metal slots so they are easy to replace. I checked with Digidesign whether you could in fact buy Pro Control fader caps, but they won't sell them to you unless you have a Pro Control. Ultimately, though, you shouldn't be too put off by this aspect of the unit, as operationally it's a professional and classy surface.

Connectivity All Command 8's communication with the computer is via a USB connection (another contrast with the Firewire-based 002). Additionally, there are two standard five-pin MIDI output ports, and one input. As well as enabling use of the controller in a hardware-only scenario, these connections turn the Command 8 into a general-purpose MIDI interface. During testing, I hooked up my main keyboard via the MIDI input, and declared it in OS X's Audio MIDI Setup window. This worked without a hitch, and proved a really useful addition given that I always seem to be short by one USB port. In Pro Tools, the only setup procedure required is to inform the system of the Command 8's presence via the MIDI Controller page of Peripherals Setup. Surprisingly, you can't use more than one unit at a time in Pro Tools; in fact you can't use any other MIDI controller alongside Command 8. However, you can attach a Command 8 in addition to a Pro Control, Control 24 or Digi 002 and use it for extra faders. One interesting suggested use for this is to have a unit in your live room to provide remote control over transport and monitoring levels. As well as MIDI connections, the rear panel sports a number of quarter-inch audio sockets. These do not constitute an audio interface for the computer; you still need separate audio hardware. Instead they are simply the connections for a builtin monitoring router, which can take two stereo feeds (usually your main mix and some other source). This then gives you control of main speaker and headphone monitoring levels from the Command 8's front panel.

More Control



The eight full-length faders are touch-sensitive and motorised, staying locked to the parameters they are controlling, and jumping to new positions if you switch them to address a new bank of on-screen controls. High-resolution encoding gives 1024 values, instead of the usual 128 steps you get with MIDI Continuous Controller messages. The fader tracks are marked with a dB scale, which is calibrated to reflect the brand-new feature in Pro Tools 6.4 where faders have +12dB gain above unity. This must have been a difficult decision to make, as it means that the scale is wrong for all Pro Tools Sessions that opt for the +6dB fader preference. It also means you need to ignore the scale for other software, such as Reason, where the unity points don't line up. Each channel strip has a single rotary encoder, which in Pro Tools is used to control pans, send levels and plug-in parameters. As in other Digidesign controllers, buttons down the left side of the front panel are used to select the active function for the pots. A ring of 11 LEDs around each encoder provides visual indication of the current value. Cleverly, depending on the nature of the parameter being controlled, the display is either a single LED that Tracks and plug-in parameters being shows which way the pot is 'pointing', controlled by Command 8 are highlighted in blue. The channel Bank, and plug-in Page or a continuous row of lamps lit up to Up/Down buttons switch the assignments in the current value. The pots are groups of eight. arguably a bit too big, which brings them close together and makes it easy to move adjacent knobs by accident. My other minor criticism is that when a LED is lit, its near neighbours tend to catch some of the light from underneath the panel, making the reading a little ambiguous unless viewed from directly above. Otherwise, the pots are one of the strongest features of the Command 8. For Pro Tools, they have very reliable and smooth tracking with the on-screen parameter they are controlling. They are, in fact, noticeably better in this regard than the pots on the vastly more expensive Pro Control, which sometimes suffer from poor tracking of plug-in parameters. The best thing about the pots is that they are true continuous encoders, which means that with supporting software they just pick up from the current value of the control they are addressing, without the parameter jumping to match the hardware's absolute position. With Pro Tools, the Command 8's pots always work in this 'increment/decrement' fashion, but other software can handle it too. I was particularly excited about how well this worked with Ableton Live. The advantage of this system is particularly prominent in a live situation, where you don't want to have sudden jumps, especially when mapping something like Live's tempo knob to one of Command 8's encoders. What's more, the pots are velocity-sensitive, so if you twist an encoder faster, its corresponding parameter will move further for a given angle of movement.



Sticking with the channel strips, there are three buttons above each fader, labelled as Mute, Solo and Select. In Pro Tools, they do what they say, with the Select buttons used both for track selection and for choosing options from menus in the display. The bright-orange LCD display has two rows of six characters per channel strip. During normal operation with Pro Tools, the displays show abbreviated track names, flipping to show parameter values whenever a control is moved. Command 8 shares this handy feature (which is lacking on Pro Control) with Digi's new super-high-end Icon mix surface. Directly below the display strip is a row of eight buttons, which in Pro Tools mode are not part of the channels. Instead the buttons here are used for focusing in on plug-ins and sends, reassigning the eight rotary encoders to edit those values. For example, if you press the EQ button, the display will highlight which tracks contain EQ plug-ins. Hitting Select on one of these channels will focus in on the EQ for that track, and map parameters to the The two-line LED display gives feedback on knobs and Select buttons. Where a the parameters being adjusted by plug-in has more than eight knobs and Commmand 8 controls. switches, they are divided into 'pages' that are accessed from the Page Up/ Down switches. As on Pro Control, a Flip button lets you switch things round so that plug-in parameters or send levels are controlled by the faders. This is very useful when recording changes in automation, as the knobs are not touchsensitive. A nice side-effect of the lack of a separate plug-in control section is that when a plug-in is assigned to the encoders, the Flip button automatically switches to fader control, whereas on Pro Control you have to hold down Command as well or you always get send levels. Another neat feature is the Pan/ Pre/Send button that lets you control all the panning, send levels, and pre/post settings at once for a particular channel. Command 8 uses Pro Tools' built-in controller 'banking' system for accessing more parameters than you have physical controllers for. The left and right buttons from the circular section on the right of the panel shuffle which eight tracks are being addressed by the surface. The Bank and Nudge buttons choose whether the faders are moved in banks of eight, or one channel at a time. An example of the level of integration you get with Pro Tools, compared to generic MIDI controllers, is that the cursors also have a zoom mode for use in the Edit Window. In fact, most of the switches on the right-hand side are dedicated to specific Pro Tools functions, such as window selection, Master fader display, and playback and record modes.

Stand-alone Mode



Sitting apart from most of the other controls on the right-hand section of the Command 8 is the Standalone button. Pressing this transforms the unit from a dedicated Pro Tools controller into a generic MIDI device. All the faders, rotary encoders and buttons on the channel strips can be programmed to transmit any MIDI Continuous Controller message required, with the exception of the buttons in the top row which have fixed values. This gives you 32 buttons plus the transport, so it's not that disappointing that all the other buttons cannot be used in stand-alone mode. In fact, each Preset (of which you can store eight) actually has two completely independent pages, toggled via the Bank button. All this adds up to 16 faders, 16 pots, and 56 separate buttons per preset, which should cover most situations! The transport buttons are permanently set to transmit standard MIDI Machine Control messages, so can't be used for software that doesn't support MMC. There are no factory presets provided for controlling specific devices, so getting things working can take some time. It's a shame that Digi haven't included a template that emulates the Mackie HUI or Mackie Control for instant control over Cubase or Logic. SysEx saving and loading or presets is supported, though, so hopefully some charitable people will publish some templates soon enough, or maybe Digidesign will provide these as an Extra MIDI ports, seen here in OS X's Audio update. In any case you tend to use MIDI Setup utility, mean the Command 8 can be used as a MIDI interface. the built-in controller learning functionality to set up MIDI control in software like Live and Reason. The templates in memory have default CC assignments for all parameters, so in these applications you can just start assigning controls in the software and leave the hardware alone. Programming controls is actually very easy. All that's required is to press the Edit button, then press, move, or turn the control that you want to edit. The rotary encoders are then used to set the MIDI channel, CC number, and high and low values. Switch behaviour can be set to 'latch' or 'momentary', the former for toggle switches, and the latter for controls that are only active while you are holding them down. Faders and encoders can operate at single- or double-byte precision. Command 8 worked flawlessly with Reason and Live during my testing period. Live had perfect two-way communication, so the motorised faders followed controls that were moved by the mouse or automation, and the LEDs displayed values correctly. This, combined with Live's ability to pick up from the directional movement of the rotary encoders, instead of jumping to an absolute value, meant that the hardware and software really were locked together. Reason, being unable to transmit MIDI, and unable to understand continuous rotary encoders, was more primitive, but benefited as much it could from any MIDI controller.



One feature that really stands out when running with other software is the ability to switch between Command 8's stand-alone and Pro Tools modes on the fly. This means that if you are running Pro Tools with, say, Live or Reason in Rewire mode, you can control both applications. Switching to stand-alone from Pro Tools worked fine, and I could then control Reason. However, I ran into a problem where switching back — the unit would not re-establish communication with Pro Tools properly, and I had to clear and then reset the Peripherals settings. Luckily, it turns out this problem has been fixed with a firmware upgrade, available as an updater for both Windows and Mac. This dual-operation functionality could be Command 8's greatest appeal to anyone who combines Pro Tools with other MIDI software.

Conclusions Anyone who's used to Pro Control will be able to pick up the Command 8 very quickly, as the functionality within Pro Tools is so similar, and many of the same features are present. This alone should tell you that the unit is good The Command 8 includes audio inputs and value for money. Obviously, some of speaker outputs, allowing it to be used as a the higher-end features are not the monitor controller. same, such as a separate plug-in control section, scrub wheel, or surround monitoring capability. The only things I'd miss day-to-day, though, would be Pro Control's automation enable and autosuspend buttons — and the faders of course. The Command 8 has a wide-ranging appeal. It's inexpensive enough to appeal to smaller music setups, where it will be great for controlling soft synths, and where the ability to flip between controlling Pro Tools and other software is a real bonus. Project and pro studios can either use it as their main Pro Tools controller, or as a remote control to add extra transport and monitoring control, not to mention MIDI connections. Mobile rigs, or smaller post-production rooms will also be a natural niche for the Command 8, especially where a TDM system is preferable to the integrated 002 system. I personally think it would make a great live controller for programs like Live and Reason. The thing that impressed me the most with Command 8 is that the stand-alone MIDI controller mode is not just an add-on or afterthought to the Pro Tools functionality. The unit stands tall alongside other generic controllers and should be on your shortlist even if you don't use Pro Tools. Published in SOS July 2004



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



Disco DSP Vertigo 2.0

In this article:

From The Top Filters & Effects The Spectral View Partial Power Resynthesis Factory Presets Putting It All Together Conclusion

Disco DSP Vertigo 100 Euros pros Capable of generating massive, complex sounds. Spectral editing capabilities work well in making additive synthesis less of a chore to program.

Disco DSP Vertigo 2.0 Resynthesizing Additive Soft Synth (Win) Published in SOS July 2004 Print article : Close window

Reviews : Software

Disco DSP's additive synth plug-in offers both highquality resynthesis, with up to 256 partials per note, and a novel 'spectral display' for editing your sounds. Mike Bryant

Additive synthesis seems to be an upand-coming thing at the moment in the virtual instrument market, with products Doesn't generate inharmonic components of aplenty showing how to implement the resynthesized timbres. concept in interesting and distinctive Use with the maximum ways. Now, with typical synchronicity, number of partials demands a resynthesizing VST instruments have fast PC. started to emerge. Camel Audio's summary Cameleon 5000, reviewed in April's Though its resynthesis SOS (www.soundonsound.com/sos/ Vertigo's Partial Edit section, here shown features aren't the most apr04/articles/cameleon5000.htm) has using the '2X Detune' function to simulate a sophisticated on the market, traditional two-oscillator synth tone. drawn a lot of attention, as doubtless Vertigo provides a lot of power and sonic potential for will the resynthesizing update for the money. The spectral Virsyn's much-lauded Cube. If my exhaustive Google research is correct, editing features are its however, Disco DSP's original Vertigo v1.0 for Windows PC was the first VSTi to strongest point, presenting a hit the market boasting resynthesizing features, becoming available in August of programming interface considerably friendlier than is last year. I was originally sent version 1.5 for SOS appraisal, though the release the norm for additive synths. of an enticing beta version of version 2.0 prompted the extending of the review period by some months until its final release. information cons

100 Euros (about £70). www.discodsp.com

Test Spec Disco DSP Vertigo 2.0. IBM Thinkpad T30 laptop with 1.6GHz Pentium 4-M CPU and 512MB RAM,

From The Top All of Vertigo's synthesis action is to be found in the left and centre sections of its interface, which change contextually to allow you to edit the envelopes, partial ratios, formant filter, and a spectral overview of the patch. Below the four buttons

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running Windows XP with Service Pack 1. Tested with Steinberg Cubase SX 2.01.10.

for switching between each view lurk several more nested items, for selecting, for example, which particular envelope or partial parameter is available for editing. Given Disco DSP's decision to fit a pretty complex synth into a fairly compact format, this layout is an understandable move, but it does mean that you're often a couple of button-presses away from the bit that needs tweaking. On the many occasions when I found myself momentarily flailing for the right envelope or partial parameter, I did question whether there might not have been a slightly more intuitive way to design the interface. The Edit section for configuring the partials is where the basic timbre of a raw sound begins. Vertigo can generate up to 256 partials per voice, and each bears its own phase, Vertigo's spectral view uses colour intensity amplitude, frequency and pan controls. to represent the loudness of each harmonic. The contour in the lower section allows level The central panel lets you manipulate changes to be drawn onto individual partials. these en masse with the usual bargraph-style display, providing a reasonably effective way of grappling with so many individual parameters. One nice touch when editing the graphical view is a deceptively simple mouse behaviour, which responds to the speed of your drawing motion by affecting a varying number of partials. If, for instance, you'd like to thicken up the sound by altering the phase of only an indiscriminate few, just draw a swift line, with slow motions catching more. This is an invaluable time-saver when attempting to widen the stereo image by alternately panning a few partials in opposite ways. To get you started on traditional analogue-synth-style sounds, a wave generator has been provided to conform the partial ratios to the usual saw, square, and triangle wave shapes. The '2x Detune' feature allows you to mimic a twooscillator tone by conforming odd-numbered partials to an alternate wave shape with a variable amount of detuning — a system which I found to work rather nicely. Vertigo provides two discrete additive synth engines for layering timbres or morphing between individual sounds. Though less powerful than the four sources offered by Cameleon 5000 or Cube, the morphing is implemented in quite a flexible manner, accessible via the modulation wheel, keyboard velocity, or a dedicated envelope. In addition to morphing, there are envelopes for amplitude, time-stretching, pitch-shift and cutoff frequency for each of Vertigo's two traditional multimode filters. Each can employ up to 16 breakpoints, and allow for



varying degrees of 'slope' between them, so you can fashion nice sinusoid releases and such like if desired. Creative use of the flexible envelopes is one way of at least partially redressing the absence of such traditional technology as modulation LFOs, though it would be a much more convenient technique if there was more in the way of tempo-based snapping facilities.

Filters & Effects Vertigo's two filters can be used in parallel or in series, and offer the usual comb, low-pass, high-pass and band-pass modes, with either 12 or 24 dB/octave roll-off. Like the filter on Disco DSP's Discovery subtractive synth, they lack punch when swept through the low range (something I'd describe as a disappointing lack of 'thwoomph'), though they are capable of howling like banshees at high cutoff and resonance settings. Like many synths of its kind, Vertigo also includes a 128-band formant filter to help simulate the varying frequency responses of different resonant bodies (see screen shot below). Whilst there are no preset curves to help you out, it does make targeting specific frequencies very easy, and the 'sticky' mouse behaviour makes quickly drawing densely notched patterns a breeze. Overall, the formant filter is good way of 'roughing up' a patch and introducing a little unpredictability to the otherwise very uniform repitching of its additive sound. The effects also serve a similar purpose. Eight effects are available, comprising EQ, phaser, flanger, chorus, reverb, distortion, delay and compression, and with between three and eight parameters available, they provide a fair amount of flexibility. My favourites were the pleasantly spacey phaser and the stereo delay, whilst the reverb is also very usable in this context. All in all, they add a welcome extra dimension to Vertigo's sound.

The Spectral View One important feature of an additive synth is the ability to shift the harmonic content of sounds over time, by varying the relative level and frequency of their constituent partials. With lots of partials making up a sound, you might imagine that programming these tonal shifts could be difficult to accomplish in a hands-on and intuitive manner. Disco DSP's attempt to tackle this problem is found in the spectral view, new to Vertigo 2.0 and replacing the rather unwieldy envelopebased system used in previous versions. As shown in the accompanying screen shot, the spectral view presents the timbral evolution of a patch by using colour intensity to represent the loudness of each partial, resulting in a pleasing sonogram-like trace. Reorienting this overview sideways, the bottom-most window allows level changes to be drawn onto individual partials in high-resolution 'time slices'. Because it's possible to see at a glance which partial you're editing and its place in the overall spectrum, this serves as an excellent way to get to grips with detailed elements of a sound, making it very easy to tackle, for example, unwanted artifacts or overly prominent harmonics. The pitch contours of file:///H|/SOS%2004-07/Disco%20DSP%20Vertigo%202.0.htm (3 of 8)9/22/2005 8:55:16 PM


individual partials can be adjusted in a similar manner by switching to the Frequency view. For broader sound-shaping, Vertigo lets you modify the spectral trace directly, by dragging a marquee around a section and applying boost, cut, or fade controls. Though not as slick as would be the provision of bona fide Vertigo provides a 128-band formant filter, drawing tools, this system works nicely along with two traditional multi-mode types, which can be used in parallel or in series. in practice, and combined with the facility to solo individual partials, it's possible to be very diagnostic in chopping out or boosting certain parts of the overall sound. The spectral view also allows you to set loop points and to globally stretch or compress sounds using the Length parameter. As some might already have anticipated, a feature that ties in nicely with this spectral arrangement is that which allows you to import your own bitmaps (of any size or bit depth) onto which Vertigo's partials are plotted. As a novel way of generating patches, this feature provides a great deal of scope for experimental fun, even if you may end up disappointed by the translation of your loved one's visage into a scary, undulating drone. All round, I consider the spectral view and the in-depth and relatively easy editing power it provides to be one of Vertigo's most powerful and appealing features. It really goes a long way to overcoming the often overwhelming parameter density that can deter from programming additive synths. Like most of its breed, however, carving out a basic timbre from scratch isn't the easiest of undertakings, which is where the resynthesis section comes in...

Partial Power Potential users would be wise to bear in mind that Vertigo 2.0 can be a very thirsty synth. Though Disco DSP have done a remarkable job in boosting the CPU efficiency since version 1.5, the reality is that those with PCs slower than 2.0GHz (the recommended spec) are likely to find themselves wanting a faster machine to really take advantage of all 256 partials Vertigo can generate. As a rough guide, I found that my 1.6GHz Pentium 4-M laptop (not to be confused with the very different Pentium-M) was capable of only three notes of polyphony at the 256-partial setting. With a more modest 64 partials I could just about approach the maximum 16-note polyphony, but with no power left in reserve for anything else. Since maxing out your CPU sends Vertigo into a bit of a tizzy until you press the Panic button, I often found it best used in conjunction with Cubase SX v2's Freeze function.



Resynthesis As you'll notice from the accompanying screen shots, the rightmost section of Vertigo's interface contains a few less-than-musician-friendly controls. These relate to the various 'Windowing' sizes and types available for configuring the resynthesis engine, and must be set prior to importing a Wave file. Though the manual describes the Windowing size choice as a trade-off between accuracy and time resolution, I found in practice that the larger sizes almost invariably produced a more faithful rendering of imported timbres. Conversely, smaller sizes sounded rather more coarse, but were generally less CPU-intensive to synthesize, as fewer partials were being used to cover the full frequency range. Which of the four windowing types — Bartlett, Hamming, Hanning or Blackman — will be most appropriate for your sample is harder to discern in advance, so trial and error is often the best option. The actual wave import process only takes a second or two if your sample is reasonably short, so it's not too much bother to try out a few different configurations, but nonetheless, I'd prefer it if the synth did more of the work for you in finding an optimal setting. By way of comparison, Vertigo lacks the aforementioned Cameleon 5000's clever use of noise to reproduce inharmonic elements of imported timbres, and is rather less versatile as a result. The percussive elements on the attack phase of certain sounds (think the flute 'chiff' or the piano 'thunk') don't survive the resynthesis process, though in some cases a confluence of high-frequency harmonics can help mask their absence. On the other hand, Disco DSP's approach of endowing Vertigo with a larger-thanusual number of partials (four times the quota available with Cameleon 5000) can certainly yield some uncommonly rich and complex tones, and it scores highly on rendering huge string sounds and complex pads. It is, however, often a worthwhile trade-off to reduce the number of partials at the expense of some high-frequency harmonics if your PC has lived through a couple of winters (see the 'Partial Power' box for more details). With these reservations in mind, I was pretty impressed with Vertigo's resynthesis efforts. It's capable of producing highly usable renderings of a wide range of common instrument sounds — in particular, I found that basses, strings and organs all came out sounding lively and playable after a little tweaking of the amplifier envelope.



Factory Presets Vertigo 2.0 includes 32 preset patches, some of which do a good job at showing off its facility for rich, ambient pads and offbeat, spacey oddness. This, however, is actually a regression from the 64 patches that came with version 1.5, which were both more sparing with the number of partials they required, and of generally higher quality. Due to the major changes in the way it works, version 2.0 isn't able to load patches made with its predecessor, though Disco DSP were reportedly studying this problem at the time of writing.

Putting It All Together When you're aiming for more than simple resynthesized tones, Vertigo 2.0 rewards patient programming with unique and interesting patches. It's not a onebutton magic module, nor a synth where programming from presets is necessarily a prudent option. As you might imagine, importing your own samples is the best place to start new sounds, and from there you might find them evolving to places you didn't really expect. Vocal sounds and phrases are an interesting case, since they retain their distinctive characteristics even when stripped of a lot of harmonic detail. Consequently, one of the synth's biggest strengths is in making sounds with the submerged-but-recognisable stamp of a vocal formant somewhere within. Rhythmic loops also make effective fodder for unusual patches, with Vertigo's attempt to consolidate a melange of untuned noise into a reproducible form often yielding interesting (and useable) results. And, of course, pitch and time characteristics are independent where transposition is concerned, so unlike a traditional sampler, rhythmic elements will remain consistent up and down the keyboard. The ability to morph between a pair of separately resynthesized tones opens up some of the most interesting possibilities. There's only so many times you can morph Aled Jones into a bassoon before the novelty wears off, but putting the envelope to imaginative uses yields many a sonic delight. The time-stretch envelope is also an invaluable tool in producing interesting, dynamic morphs between a pair of contrasting tones.

The envelopes provide up to 16 breakpoints, with variable curvature between them.

Programming with the spectral window lends itself to the creation of some great sounds; I particularly enjoyed making small melodic loops by adding little bleepy transients to certain harmonics. These can then be stretched into pad-like



textures that evolve in unpredictable ways. Using the horizontal and vertical fade controls to introduce harmonic shifts and patterns that bleed into one another also works a treat in generating interesting and unusual timbres. From a programming perspective, the spectral window is another area of Vertigo that would benefit hugely from the addition of some tempo integration; something similar to the snap-to-grid facility found in the controller lanes in Propellerhead's Reason would make life much easier. This regret notwithstanding, taking into account the effects and filters (see box) means there is no shortage of ways to completely mangle imported timbres, or just add a little garnish on the top.

Conclusion I have to admit that after spending a good bit of time with the 1.5 version of Vertigo I was originally sent for review, I was tending towards something of a 'Yes, but...' conclusion. I felt that Vertigo's undoubtedly great sonic capabilities did not quite make up for either its slightly awkward interface, nor the whopping CPU usage that that version exhibited. I'm glad to report, therefore, that whilst it could certainly use some further improvement in places, version 2.0 has substantially redressed the balance. Several elements of the synth don't quite match up to its (rather more costly) competitor from Camel Audio, notably the morphing facilities and the sophistication and efficiency of its resynthesis features. On its own terms, however, Vertigo 2.0's additive resynthesis engine is undoubtedly very powerful, and it possesses overall a kind of quirky usability that, once accustomed to, can yield fabulously attention-grabbing, offbeat sounds. The spectral editing capabilities play the biggest role here, and really help 'open up' resynthesized timbres to user manipulation in a way that's more hands-on and accessible than anything else I've yet seen. This is a synth that, more than most, is about coming up with your own unique sounds, and as such it won't suit everyone. For those who are interested in sonic invention — particularly those who like an organic edge to their synth programming — Vertigo is a worthwhile choice and good value at only 100 Euros. Published in SOS July 2004






Focusrite Liquid Channel

In this article:

A Toe In The Water The Convolution Solution Recent Developments More Than Just A Preamp Front-panel Controls The Preamp & Compressor Replicas Using The Liquid Channel Even Better Than The Real Thing? Liquid Control Software & USB Facilities

Focusrite Liquid Channel £2344 pros 40 preamps and 40 compressors in the one box. The promise of future additional Replicas as free downloads. The Replicas can mimic the control restrictions of the units which inspired them, but they can also allow much more flexibility if required. Two units can be linked for accurate stereo operation. Elegant remote-control software. High-quality preamp in its own right. Flexible configuration and I/ O facilities.

cons No DI input. Potentially too subtle for many home-studio applications.

summary An impressively versatile machine that recreates the nuances and subtleties of a range of rare and esoteric preamps and compressors with remarkable fidelity.

Focusrite Liquid Channel Recording Channel Published in SOS July 2004 Print article : Close window

Reviews : Recording Channel

Combining Focusrite's cutting-edge preamplification and A-D conversion with Sintefex's pioneering convolution technology, the Liquid Channel can mimic the most celebrated mic preamps and compressors ever designed. Could this be the only voice channel you'll ever need? Hugh Robjohns

The Focusrite Liquid Channel has been one of the most keenly awaited products of the year (or last year for that matter!), and at recent trade shows the Focusrite booth has always been jam packed with people keen to Photos: Mark Ewing see what all the fuss is about for themselves. It is the result of three years of joint research and development between Focusrite and Sintefex. I previewed of the product back in SOS December 2003, but the completed product has only now been launched. Part of the delay in bringing this product to market has been the sheer time it has taken to create all of the necessary convolution data required for the 80 simulations of mic preamps and dynamics units. The company have also been working hard to complete the remote control software. Another issue which is still to be fully addressed, but which has slowed progress slightly, is that of copyright. A convolutional processor is, in theory, able to faithfully recreate every sonic nuance of any chosen product. It could be argued, therefore, that by replicating a manufacturer's product with a convolution processor, their copyright may be infringed. Looking at it another way, had that manufacturer's product been reverse engineered and the circuitry copied component for component, the breach of copyright would be fairly obvious — yet the convolutional approach, in theory, produces indistinguishable results. It's

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Those with the budgets and space may still prefer the individual characters of original units, but the Liquid Channel affords the rest of us the chance to share and enjoy the qualities associated with so many classic preamps and compressors.

information £2344.13 including VAT. Focusrite +44 (0)1494 462246. +44 (0)1494 459920. Click here to email www.focusrite.com www.ffliquid.com

certainly a thorny and untested area of law, and it will be interesting to see how things resolve in the future. Of course, this is wholly irrelevant to the user — you buy the box, dial up the effects you require, and get on with making music — but there are implications as far as publications are concerned. In reporting on direct comparisons between the Liquid Channel's simulations and the original hardware units, legal advice suggests that it is prudent for us only to identify by name units which are out of production or which are from Focusrite's own stable. Having said that, it is logical to infer that the accuracy (or otherwise) of any one emulation should translate to all the others...

A Toe In The Water

So what does the Liquid Channel actually do? The brief answer is that it 1GHz Pentium III PC laptop employs dynamic convolution and other with 512MB RAM, running very sophisticated technologies to Windows XP Home. mimic a wide range of classic Liquid Channel OS v1.0.2. preamplifiers and dynamics processors Liquid Control software v0.9g. — these simulations are referred to by Focusrite as Replicas. There is also some digital equalisation thrown in, some very high-quality digital conversion, and the ability to process line and digital inputs as well as microphone signals. Test Spec

The 2U rackmounting box is painted white, with chunky mirrored chrome rack ears. It is a single-channel processor, catering for analogue mic and line inputs (but no DI input, sadly), as well as AES-EBU digital input. There are analogue and digital outputs, word-clock in and out for synchronisation, and a USB port. The digital interfaces and clocking support sample rates up to 192kHz, with full 24-bit resolution. The USB port is used both for remote control (using the Liquid Control software) and data transfers, to and from a Mac or PC. It is possible to link units together for stereo or multi-channel applications as well, to ensure the dynamics processing maintains a stable image. A pair of phono connectors provide the link input and output connections, and each unit is simply coupled to the next in the chain, with the last unit being connected back to the first. The factory set of Replicas contains 40 preamps and 40 compressors, including rare and long-discontinued products alongside many more modern and familiar devices — a complete list is given in the 'The Preamp & Compressor Replicas' box. It's pretty obvious what most of the Replica names refer to, and most identify the country of origin as well as a model number. file:///H|/SOS%2004-07/Focusrite%20Liquid%20Channel.htm (2 of 11)9/22/2005 8:55:20 PM


The Convolution Solution The audio characteristics of a signal processor — the frequency response, phase response, nonlinearities, and so on — can all be measured in a variety of ways. Once defined, these characteristics can then be recreated. One approach is usually called modelling, where fairly conventional digital signal processing techniques are used to create a sound character similar to the original unit. A more accurate approach is what Sintefex and Focusrite call 'replication' where the exact characteristics of the original signal processor are measured precisely, and then that data is used to modify a new input signal through convolutional signal processing. The result should be a totally accurate recreation of the original unit's characteristics. A particularly useful technique for analysing a signal processor is that of measuring the impulse response. An impulse is a simple click, but the right kind of click can be shown mathematically to contain all signal frequencies (within a defined bandwidth), at the same level and at the same time. If such an impulse is fed into an audio signal processor, the output signal will be different in some way, according to the processing applied. In most cases the duration of the impulse is extended and there may also be level changes, but the precise shape of this 'timesmeared' impulse is unique for the processing applied, and so defines every relevant aspect of the signal processing in a totally precise and repeatable way — it is the sonic equivalent of a fingerprint, if you like. Impulse response measurements are very useful when working with digital equipment, because a single digital sample is essentially similar to a single impulse. For each input sample you need to generate a string of output samples of various amplitudes to replicate the same impulse shape. This imposes the sound characteristics of the wanted device on the input sample, and this stage of the process is the convolution. Each input sample is convolved with data representing the impulse response, which defines the characteristics of the wanted device, to produce a string of new output samples. To make it slightly more complicated, the string of new samples generated for each input sample have to be added to the strings of samples from all the other input samples that came before and after. It's not difficult to do, but it does take an awful lot of sums, and this is what makes convolution a DSP-hungry technology. In fact, although the maths behind convolution has been around for a long time, it is only the recent phenomenal advance in DSP technology which has allowed the concept to be turned into a practical and cost-effective reality for the professional audio market. Things really start to get complicated, though, if you want to be able to change the sonic characteristics on the fly — for example, to change an EQ frequency control. In this case, the shape of the impulse response will also change, because each response is unique to the specific processing applied. Consequently, the signal processing has to calculate or load new impulse responses as the control is altered, and convolve the new data — all without glitches and clicks. In the case of a dynamics processor, things become more complicated still, because the impulse response changes not only as the result of control changes, but also with the changing amplitude of the signal itself — and this is why the Liquid Channel uses dynamic convolution to replicate the processing for its compressor and limiter simulations.



Recent Developments One important change from the prototype I played with last year is a rationalisation of the way in which the compressor controls work. There are now two options: a totally authentic As Original mode, and a more flexible Free mode. By default, the unit operates in the As Original mode, whereby the controls and displays recreate the actions and parameter ranges of the simulated unit as closely as possible. Thus, if the original unit doesn't have all the controls provided on the Liquid Channel, those functions are simply labelled Fixed and the controls disabled. If the original model's controls worked in the reverse direction, the Liquid Channel mimics that exactly, and on devices where it is possible to press several buttons at once to create odd (but often useful) effects, the Liquid Channel matches those facilities too. On the other hand, there may be occasions where you want to be able to adjust a parameter even though the original didn't allow that, or adjust a control beyond the original's normal range. The Free mode allows just that — all of the Liquid Channel's controls are allowed to affect the corresponding parameters of the compressor Replica over the full range afforded by the processor, with intelligent interpolation for settings that lie between or beyond those of the original unit. The nature of vintage processors is that no two ever sound exactly the same — component ageing, repairs, and replacement parts all alter the sonic character. Clearly, it isn't practical to measure every classic preamp and compressor and provide all the alternatives in the Liquid Channel, so Focusrite have instead included the ability to adjust the level of harmonic distortion for each Replica. In the preview prototype the control only adjusted the second-harmonic component, but it has been improved considerably in the release software to control the third and fifth harmonics as well, and in realistic proportions related to the input gain setting and the preamp Replica selected. The control range typically spans 0.01-20 percent total harmonic distortion. There will also shortly be a range of 'hot' versions of many of the preamp Replicas with far greater levels of distortion, all available free to registered users, intended to cater for those who want a really cranked-up sound. I've heard many of these 'hot' versions, and they offer very useful alternatives. In addition to the initial factory Replicas, it will be possible to change and update the library of convolution data from a computer using the USB port. Focusrite have already set up a dedicated www.ffliquid.com web site, and the Liquid Assets page will apparently be offering brand new Replicas for free download on an ongoing basis, although nothing new was yet available there as SOS went to press. User settings of each and every parameter — including mic gain — can be stored in any of 99 user memories, and these may in turn be archived to an file:///H|/SOS%2004-07/Focusrite%20Liquid%20Channel.htm (4 of 11)9/22/2005 8:55:20 PM


external computer via the USB port. The microphone preamp stage is the most sophisticated and complex design that Focusrite have ever come up with (see the 'More Than Just A Preamp' box), and is followed immediately by a very high-quality A-D stage. From here on, all of the signal processing is digital, but it essentially starts with the convolution of the selected preamp characteristics. This is followed by the convolutional dynamics processing and the digital equalisation, the latter based on the Focusrite D2 plugin. The dynamics and equalisation can be reversed in their order if required, and the equaliser can be allocated to the dynamics side-chain for frequencyconscious compression. The Liquid Channel can also be used entirely in the digital domain. Clearly, the convolution used to emulate the various dynamics processors is a digital process, and so it is easy to see how a digital signal can benefit from that aspect of the machine. The clever bit is that a D-A stage is included to route the digital input through the analogue preamp's line input. Although this misses out on the characteristics imposed by the matching of a microphone to the preamp, the remaining analogue signal-processing artefacts and characteristics can still be applied to a digital input.

More Than Just A Preamp Replicating the characteristics of a reverberant space, equaliser, or compressor are far from trivial, as already discussed, but once the measurements have been made it becomes a well-defined problem to solve. Replicating the characteristics of a preamp seems, at first sight, to be equally straightforward... until, that is, you consider the interaction between the microphone and preamp.

The Liquid Channel's preamp board is

Equalisers and compressors are contained in its own heavily shielded metal generally line-input devices, and the compartment, and the lid of this compartment has been removed in this signal levels and impedances for picture to show the complexity of the line-level signals are well defined. circuitry. The special relays are the silver Provided the device has a canisters in the top right-hand corner of the reasonably high input impedance, shielded enclosure, and the custom input and the source has a reasonably transformer is in the bottom right-hand low output impedance, the actual corner. The separately screened area in the interface plays a negligible part in bottom left-hand corner contains the digital the sonic character. The same is not clock circuitry and converters. true for microphone preamps, where the relationship between its input impedance and the output impedance of a microphone is more complex, with far greater variations between products. Whereas line inputs tend to be predominantly resistive, microphone circuits are inherently more reactive, with file:///H|/SOS%2004-07/Focusrite%20Liquid%20Channel.htm (5 of 11)9/22/2005 8:55:20 PM


significant capacitive and inductive components to consider, particularly where the preamp has a transformer input circuit. As a result, the same microphone may sound different when connected to alternative preamps, and vice versa. These effects play a very significant role in creating the overall sound character, as anyone who has played with a switchable-impedance preamp can confirm, and they cannot easily be created by convolution technology. To do so would require each and every preamp to be measured with each and every microphone, which is clearly an impractical thing to do. The solution which the Focusrite engineers came up with was to simulate the preamp's front-end characteristics in hardware — although even this means that considerable additional measurements are needed to analyse the interface properties. The result, though, is that when the user dials up a preamp Replica the convolution processing takes care of the preamp's own sonic characteristics — the effects of discrete solid-state circuitry, valves, op amps, and so on — while the configurable hardware ensures that the microphone 'sees' exactly the same input impedance and properties as if it were connected to the actual product being simulated. The overall sound character and tonality should therefore be exactly the same as if the identical mic were connected to the original preamp. Making a configurable input stage is not simple, and goes against the normal practice of keeping the microphone signal path as short and simple as possible. Focusrite have had to come up with a complicated arrangement of very highquality relays to switch various inductors, capacitors, and resistors into the input circuits, in order to provide the appropriate impedance characteristics. Furthermore, a special custom-made transformer can also be switched into the input circuitry when necessary, because it was found that some microphones react with the transformer in a way which can't be captured by convolution.

Front-panel Controls The Liquid Channel has a clean and tidy front panel, with clear legends and meters and a good-sized LCD window. The preamp section occupies the left of the panel and contains a vertical bar-graph meter displaying a 20dB range with a separate Digital Clip LED. Level with the top of the meter, a button cycles through the analogue mic/line and digital inputs, while a fourth LED in the same block illuminates when the selected preamp Replica is using the input transformer. The lower half of the preamp section contains a rotary encoder to set the input gain, the setting indicated with a ring of LEDs around the base of the knob. All of the Replicas benefit from an enormous +6dB to +80dB of gain, even though few of the originals offered anything like that much. Line inputs can be trimmed by ±10dB, and the usual phantom-power, polarity inversion, and high-pass-filter facilities are all controlled by three buttons below the encoder. The filter can be adjusted via a Setup function to turnover at 75Hz or 120Hz, and both positions have a second-order (12dB/octave) slope. To the right of the input section is an area devoted to digital clocking. There are three buttons and various LED arrays, the middle of which steps through the internal clock rates from 44.1kHz to 192kHz. The button below switches the clock file:///H|/SOS%2004-07/Focusrite%20Liquid%20Channel.htm (6 of 11)9/22/2005 8:55:20 PM


source between internal word clock, external word clock, or the digital input, while the top button engages the Session Saver mode. This is a simple automatic gain control which monitors the signal level and reduces the input gain if there is a danger of digital clipping. The middle part of the panel is SOS Technical Editor Hugh Robjohns lines dominated by a very wide, two-line up the Liquid channel against the Pultec LCD window, below which are six MB1 and Focusrite Red 1 preamps, as well rotary encoders. Like the input gain as the celebrated Fairchild 670 compressor, control, these all have a ring of LEDs in order to critically evaluate the quality of the convolution processing. around their bases to indicate the current settings, although their precise numeric values are also displayed along the bottom line of the LCD. The first encoder controls the level and nature of harmonic distortion, as described earlier, and the following five controls adjust the various standard compressor parameters. In order, these are Threshold, Ratio, Attack, Release, and Make-up. The top line of the LCD window shows Replica and preset names, as well as preamp gain, and the divisions between data are not always entirely clear. Above the LCD is a pair of buttons, one at each end, which is used to select alternative preamp and compressor Replicas — the audio is muted briefly while the new Replica is loaded. To the right of the LCD are two more buttons and another vertical bar graph. The buttons enable a Stereo Link mode and switch the compressor into circuit. The meter shows the amount of gain reduction being applied, but the scaling remains the same as on the prototype preview model: a quarter of the range covers just the top 1dB of gain reduction, and the top 3dB occupies half of the meter! However, the effect I commented on in my preview, where the meter display appeared to lag significantly behind the audio signal peaks, has been corrected.

The Preamp & Compressor Replicas Microphone Preamps: CH D19 D SI 72 DEUTSCH 72 DEUTSCH 76 DK MEC1A FF GREEN 5 FF ISA 110

Each band of the three-band equaliser is controlled by a pair of rotary encoders, one to adjust the turnover frequency and one to dial in a generous ±18dB of boost or cut. The high shelf can be tuned from 200Hz to 20kHz, the low shelf from 10Hz and 1kHz, and the mid-band between 100Hz and 10kHz with the two switchable Q values of 0.8 and 2.5. The final front-panel section on the extreme righthand side carries all of the unit's housekeeping


FF RED 1 STT1 VTTX UK 1073H(omega) UK 1073L(omega) UK 1081H(omega) UK 1081L(omega) UK 1960 UK 33114


facilities. The first of the top three buttons is used to bypass the entire machine, while the next two provide Compare and Revert functions. The Compare button switches between the current and previously saved program settings, allowing them to be compared nondestructively. The Revert button recalls the previously saved settings, discarding the current parameter adjustments. The lower half of the panel contains the usual memory Save and Recall buttons, along with two more buttons for naming programs. The last button here is labelled Setup, and this accesses the various machine configuration options.

UK CAD UK HELI UK PA1 UK PURE UK SL4K UK TRI A UK VR US 2022 H(omega) US 2022 M(omega) US 3124+ US 3124HOT

Using The Liquid Channel The Liquid Channel is an impressive piece of equipment in it's own right, even before you take its convolutional elements into account. The A-D converter has a noise and distortion figure of 109dBFS, and the D-A is specified as -103dBFS, with less than 20ps of jitter from the internal clock. The mic preamp's EIN figure is a respectable -126dB at 80dB of gain and a 150(omega) source — remarkable given the very complex input switching circuitry.

US 737 US 786 US 786 L(omega) US AT1100 US FLAM US HV3D US M DUNK US M610 H(omega) US M610 L(omega) US MB1 US STT1VT

Among the preamp Replicas are two that provide a totally unprocessed preamp stage with either a transformer input or an electronically balanced input. Critical auditioning in this mode with the compressor and EQ stages bypassed demonstrated the inherently high quality and transparency of the preamp circuitry. It sounded wonderfully neutral and clean to my ears, with plenty of headroom (+16dBu, in fact) and a low, smooth noise floor.

US VIPERL(omega) US VIPERM(omega) US VT2

Compressors & Limiters: DK CL1B DK LCA2B FF GREEN 5

Dialling in various alternative preamp Replicas, their sonic signatures were immediately obvious in most cases, particularly when switching to valvebased preamps and those with transformers. Just about every topology is represented here: valves, discrete solid state, op amps, Class A, fully balanced, transformer coupled, direct coupled — you name it,

FF ISA 130 FF RED 7 STT1 SSSS STT1 VTVT UK 1960 UK 221X UK 2254 UK 33609 UK C2



there's a preamp that uses it here somewhere. It's only when you can switch directly between different preamps that you realise that, although a good preamp is always a good preamp, there are sometimes significant differences in the way the source is portrayed. Often, the differences are subtle and easily masked in a mix, but sometimes choosing the most complementary preamp can make the difference between a great recording and a magical one. The same is true for the compressor Replicas, but the differences here are even more marked because, in addition to the sound character of different circuit topologies, there is also the dynamic behaviour — the way different compressors respond to transients and sustained sounds, and the way they recover from gain reduction.

UK C2CRUSH UK CHIS UK DP402 UK FX384 UK PIE UK SC2 UK SL4K UK SL510 UK TL C1 UK VR US 160S US 160S OE US 165 OE US 2A US 2500ONH

The options provided cover all the bases: optocompressors, solid-state FET compressors, VCA designs; transistor, op amp, valve and digital gain stages; soft and hard knees; and peak and RMS sensing. If you are already familiar with the particular unit emulated, you'll know what to expect, but even if you don't it is very easy just to scroll through the options until you find the one that sounds best for the particular situation.

US 2500ONS US 737 US BF 3A US BF 76 US DCL200 US FC 670 US LA100A US M FET

Even Better Than The Real Thing?

US M OPTO US MU US SF 76

I remember being impressed when playing with the US SF 4 preview prototype unit because of its apparent sonic US STRESS accuracy in conjuring up the sound of the Focusrite ISA110. So when the time came to review the production unit I decided to source a number of classic vintage preamps and compressors from our friends at FX Rentals for further comparisons. A car load of substantial flightcases later and I had a Pultec MB1 valve preamp (complete with separate 110V step-down transformer), a Fairchild 670 dual-channel valve compressor (with integral step-down transformer in a huge and very heavy rack!), and a Focusrite Red 1 solid-state quad mic preamp (one of my favourites) all stacked up. (FX Rentals offer single-day equipment rental for UK home studio owners on a budget. For example, the charges for the three comparison units used here would be £35.25, £146.88, and £82.25 per day respectively, including VAT.) By using an Audix control box, I was able to switch directly between the outputs file:///H|/SOS%2004-07/Focusrite%20Liquid%20Channel.htm (9 of 11)9/22/2005 8:55:20 PM


of the Liquid Channel and each of the original units in order to make direct comparisons. I used most of my collection of large- and small-diaphragm condenser mics (Neumann TLM103 and KM180 series, Sennheiser MKH series, Microtech Gefell M930, and Audio Technica AT4040), as well as a couple of pretty naff dynamics I had kicking around (just to see what reaction there was to the impedance switching). Sources were some acoustic guitar, a bit of cello, and several spoken and singing voices, as well as line-level signals from a CD player with a variety of solo-instrument and test recordings — the latter proving particularly useful for evaluating the compressors. The crucial question is, of course, how well the Liquid Channel compares to the original units. Well, to my ears I have to say it compares extremely well indeed. In fact, I couldn't tell the difference between the Fairchild 670 Replica and the original at the same settings. There can't be many 670s around in the UK and I wondered if Focusrite had hired the same machine from FX Rentals to make their measurements! It was much the same story with the preamps, although I found I had to dial in some Harmonics to match the rather rich and warm character of the original MB1. Switching to the solid-state Red 1, again I found it very hard to tell it apart from its Replica, and the intrinsic qualities and character of the preamp were readily identifiable. Randomly switching around the other preamps and compressors I was never disappointed, although often overwhelmed with the sheer variety on offer. It was a very wise move to include some 'hot' preamp versions, as sometimes you just need that extra drive and power, and the Harmonics control is a very useful and powerful tool in shaping the sound even further. I was occasionally frustrated when some unfamiliar compressors refused The Liquid Assets area of Focusrite's to let me adjust parameters I felt I dedicated Liquid Channel web site contains downloads of all the existing factory wanted to change, but it would have Replicas, and will host any new Replicas that been the same had the real box been are developed. in front of me. And, of course, switching over to the Liquid Channel's Free mode immediately gave me all the extra control I needed. I'm not so sure this product, impressive though it undoubtedly is, is the best possible investment for the smaller home studio. The quality and range of preamps and compressors here is sublime, but this is a finessing tool and I wonder if the differences might be a little too subtle in circumstances where there are bigger hurdles to overcome than choosing between different preamps. To appreciate many of the subtle qualities and variations available here requires good recording and monitoring environments, and for the UK price of the Liquid Channel you could make significant improvements to your studio's acoustic treatment or purchase high-quality monitors!



Having said that, the Liquid Channel is a fabulous and unique product, and I can see it becoming as much a standard studio tool as a Lexicon reverb or an Eventide multi-effects processor — although a dedicated stereo version would be rather nice, especially for mixing and mastering applications. This is an ideal machine for mid-sized and larger studios, either to provide or supplement a flexible range of esoteric and characterful preamps and compressors. This is definitely worth a private audition, but not at a trade show or in a noisy showroom — you need to take the time and trouble to listen to the Liquid Channel somewhere where you will be able to appreciate it fully. Thanks to FX Rentals for loaning the comparison units used in this review. +44 (0)20 8746 2121.

Liquid Control Software & USB Facilities The Liquid Control software requires Windows XP Home or Professional for the PC, or Mac OS 10.2.8 or higher for the Mac. Both systems require a USB 1.1 port and 5MB disk space for installation. The software worked very well in practice and it hardly needs to be described — the front-panel controls and displays are exactly the same as the machine itself, and there are fully bidirectional commands between them so that both displays show the same things, and all the controls do the same things. For this review I stuck with a late pre-release version of software (v0.9g) simply because there seemed no need to update it — everything appeared to work perfectly for me. One handy feature is that the preamp and compressor Replicas can be changed by clicking on dedicated menu drop-downs. I found it much easier to navigate a drop-down list than to scroll through the options on the hardware. Updating the Liquid Channel's firmware and the Replica sets is trivially simple through the appropriate Liquid Control menus, but updating all 80 Replica files takes quite a while! Although the interface to the Liquid Channel is via USB 1.1, I understand that Focusrite can supply Cat 5 USB converters to enable the unit to be controlled from a remote computer instead of one at the end of a shortish USB cable. This will be a far more attractive option to many studios. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



FXpansion VST To RTAS Adaptor

In this article:


In Practice Plug-in Format Adaptor Tested Plug-ins Performance And Limitations Published in SOS July 2004 Conclusions Print article : Close window

FXpansion VST To RTAS Adapter £60

(Mac OS X & WinXP)

Reviews : Software

pros Brings hundreds of effects and instruments to Pro Tools, including many free ones. Non-intrusive and highly integrated inclusion of VST plug-ins in the Pro Tools environment. Practically no performance overhead above that of the plug-in. Compatible with UAD1 and Powercore DSP effects. Great value for money.

cons Any problems will require some technical ability to resolve. Can be unpredictable, so not suitable for critical situations. May not work with your favourite plug-in.

summary Successfully converts most VST and VSTi plug-ins into working RTAS versions, opening up a whole new world of goodies to Pro Tools users. While it doesn't work with everything, and needs some technical competence to manage, you get big rewards for your efforts.

information £59.99 including VAT. Sonic8 +44 (0)8701 657456. +44 (0)8701 657458. Click here to email

As the name suggests, FXpansion's utility allows VST effects and Instruments to be used in Pro Tools, but how well does it really work? Simon Price

VST plug-ins running in Pro Tools, thanks to FXpansion's adaptor.

Both Pro Tools LE and TDM systems use Digidesign's proprietary RTAS (RealTime Audiosuite) format for host-based plug-ins. While this supposedly keeps Pro Tools' mixing environment within Digidesign's sphere of influence, it has limited the range of virtual effects and instruments available to Pro Tools users. The situation has been improved by Rewire support, as well as Native Instruments' adoption of RTAS, but there is still a great number of VST-format plug-ins (many free) that we've missed out on. It seems quite hard to believe, but FXpansion's VST To RTAS Adapter can take VST effect and Instrument plug-ins and alter them so that they work as RTAS versions. (They also have a VST To Audio Units Adapter, for OS X users of Logic who want to incorporate VST plugins.) What FXpansion's adaptor does is 'wrap' the plug-ins in an extra layer of code that mediates between Pro Tools and the plug-in, making the plug-in appear to Pro Tools as if it was in the RTAS format. This 'wrapping' takes place off-line, using a configuration application, and once it's complete, FXpansion claim that the adaptor code uses next to no CPU power, so the plug-ins should be nearly as efficient as in a native VST host like Cubase. No extra background programs have to be running when you use the converted plug-ins — they just appear in Pro Tools alongside your existing effects. Installation creates a folder containing the VstRtasConfigure utility and a local VST plug-ins folder. The Configure application scans your computer for VST plugins, creates the 'wrapped' versions and places these in your Pro Tools Plug-ins

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www.sonic8.com www.fxpansion.com

Test Spec VST To RTAS Adapter v1.02. Apple iBook 800 running Mac OS 10.3.2. Tested with Pro Tools LE 6.2.2 and Digidesign M Box hardware.

folder. Even with a large number of plug-ins this only takes a few seconds. After this, you only need to use the utility when you add new plug-ins, or wish to remove them all. The important option in VstRtasConfigure is the 'Scan only local VST plug-ins' toggle. While the utility can scan your whole computer and convert any VST plug-ins it finds, it's highly recommended that you copy the ones you wish to use into the local folder. The best method is to convert in small batches rather than all at once. The reason for this is that some plug-ins are not going to work, and introducing a small number at a time makes it much easier to weed out problematic ones. After the configuration program has finished, all that remains is to quit it and launch Pro Tools. Converted plug-ins appear in the normal RTAS plug-in list, available at insert points in audio tracks, and also aux inputs on LE systems. Adapted plug-ins are to easy to differentiate from the others as their names appear with the prefix 'VSTW', which tends to clump them at the bottom of the list. Pick a plug-in, and hold your breath. With any luck the plug-in will pop up looking for all the world like it's perfectly at home in Pro Tools. As well as having full use of Pro Tools' standard Save and Load preset functionality, the VST preset system is available at the bottom of the plug-in window. Automation is also implemented.

In Practice Now you're probably thinking there must be a catch with all this, right? Well, that depends on your expectations. There is a certain amount of uncertainty in using the adaptor (as you might expect), which means that you need to approach it with a willingness to get your hands dirty, and without expecting it to work with everything. There are certain technical differences between the RTAS and VST formats that are hard to reconcile, such as RTAS's lack of support for multiple outputs from a virtual instrument. Also, the huge range of possible system configurations and plug-ins makes testing difficult, especially for a relatively small independent software house like FXpansion. Taking all this into account, I set aside some time to install a range of effects and instruments, a few at a time, and get a feel for what can be expected. I downloaded around 40 free and demo plugins to add to my own. You can see a summary of my results in the 'Tested Plugins' box on the next page. To begin with I was using an Apple iBook 800, running OS 10.2.8 with Pro Tools 6.1 and a Digidesign M Box, and had about a 50 percent success rate. This, I later learned, was an unusually poor result caused mainly by my Mac. Even so, many VST Instruments worked fine, including some large and demanding ones such as Gmedia's Oddity and ImpOSCar. Internal MIDI functionality worked fine, with the plug-in instruments automatically seen as destinations in MIDI tracks. One thing that didn't work in this configuration was tempo sync with Pro Tools' timeline, though this turned out to be one of the things that improved when I updated to Pro Tools 6.2.2 (see below).



Failures tended to fall into two categories. Some plug-ins, such as Bitshift Audio's Phatmatik Pro and Linplug's RMIV, would crash immediately upon loading (or shortly thereafter). A number of other plug-ins appeared stable, but had graphics problems that rendered them unusable. This category included the huge MDA collection of free effects. Strangely, these plug-ins had worked with an earlier beta version of the adaptor that I'd been testing, so I suspected something was amiss. VST plug-ins are converted prior to use with FXpansion suggested that I really the Configuration Utility. ought to be running Panther (OS 10.3) and Pro Tools 6.2x, even though the published requirements are OS 10.2 and any version of PT6 — and after dutifully upgrading, most of the problems I had did indeed disappear. It actually turns out that some minor filing system corruptions on my hard drive had been preventing the FXpansion installer from installing some supporting files, causing my graphics problems. Unfortunately, instruments that had caused crashes like RMIV were no better after the upgrade, and obviously fall into the category of stuff that's just not compatible at the moment. However, everything was suddenly faster, more stable, tempo sync started working, and my success rate was up to at least 80 percent. Also on the positive side, FXpansion can be seen to be actively involved with the on-line user forum, and if a particular incompatible plug-in is popularly requested there's a good chance that they will look into updating the adaptor to accommodate it.

Performance And Limitations VST plug-in performance tends to vary widely even in native environments due to differences in programming quality, so it's difficult to judge how transparently VST To RTAS Adapter does its work. Testing with the ImpOSCar and Novation V-Station instruments, performance was only marginally weaker in Pro Tools compared with using unwrapped versions of the plug-ins in Cubase SX, and to be frank I'm tempted to put this slight difference down to Pro Tools 6

Tested Plug-ins Plug-in - Did it work? 4Front E-Piano Module - Yes BIAS Vbox - No Bioroid Turntablist Pro - No (crashed config utility) Bitshift Phatmatik Pro - No (crash) Camel Audio Camelphat Free - Yes CM DS404, CM303, SR202, CM101 - Yes CM CM505 - No (crash) Creakbox Bassline - No (no sync) Destroy FX EQ Sync - No



LE itself, which seems to struggle a lot on less-than-cutting-edge systems. Inevitably, there are glitches here and there. Although upgrading to PT 6.2.2 got tempo sync working in most plugins, it was still a bit flaky in others such as PSP42. My review version (1.02) had a bug where automation would only get played back for plug-ins whose window was open. I'm assured this is now fixed in version 1.04, which came out as the review was to be filed. Other points conceded by FXpansion are lack of support for control surfaces such as Digi 002, no delay compensation, and the performance limitations imposed by RTAS's fixed 32-sample buffers. However, Pro Tools LE does have plugin delay compensation up to a point, and this worked perfectly with wrapped plug-ins on my system. I also suffered no worse audio break-up problems with Pro Tools than with Cubase, so the small buffers don't seem to be an issue even with very demanding instruments. One issue I did encounter with the CM instruments was that the sound would cut out while adjusting on-screen controls. Apparently this can be a symptom of low CPU power, although I didn't have this problem with any other plug-ins.

Destroy FX Rez Synth - No (crash) Digital Fish Phones Blockfish, Floorfish, Spitfish - Yes DMA collection (21 plug-ins!) - Yes DMI Hammer - No (sound but can't edit) Elemental Audio Inspector - Yes Gmedia Oddity, ImpOSCar - Yes Green Oak Crystal - Yes Linplug Free Alpha - Yes Linplug RMIV - No (crash) Muon Mdrive - Yes NdcTrem+ - Yes Novation V-Station - Yes Ohm Force OhMyGod - Yes PSP Lexicon PSP42 - Yes (but no tempo sync) PSP Spring Reverb, Vintage Warmer - Yes ReFX Beast, Claw, QuadraSID - Yes Rumpelrausch Täips Crazy Diamonds - Yes Smartelectronix Bouncy, Crazy Ivan, Cyanide Shaper, H2O, Madshifta, Supaphaser, Supatrigger, Ambience, Analog Delay, KT Granulator - Yes

An exciting development in recent versions, which is enjoying continued special development, is compatibility with certain DSP hardware products. Incredibly, the adaptor can wrap plug-ins that work with TC Electronic's Powercore and Universal Audio's UAD1, allowing you to use these DSP-accelerated plug-ins in Pro Tools, which must be ruffling a few feathers at Digi HQ! DSP support is at a preliminary stage, and the performance is not as good as when using these plugins in their native environments, but early reports sound encouraging. There is the problem of latency in these systems which Pro Tools can't adjust for automatically, but it can at least report the delay time (as with TDM plug-ins), allowing you to nudge audio to compensate.

Conclusions



Technically, VST To RTAS Adapter is quite an achievement, and politically is one up for the consumer (remember, those people who fund the industry) against the large manufacturers' format wars. If my tested plug-in list is anything to go by, the adaptor works with the majority of VST plug-ins out there. Where it does work it's remarkably transparent, doing all the things you'd expect from a real RTAS plug-in. There are plug-ins that just Adapted plug-ins appear in the standard plugdon't work, so if you are thinking of in list with the prefix 'VSTW'. buying the adaptor in order to use something specific, you should check out the compatibility first. The best place to do this is the online FXpansion forum at www.kvr-vst.com. There are other things you should consider before making up your mind. You should expect a certain amount of tweaking and experimentation to get a stable working system. I went through my share of crashes in the early stages, including several occasions where I had to strip out all Pro Tools' settings files to get up and running again. You need to be prepared for this possibility. Also, the fact that you are introducing potentially unstable third-party plug-ins into the system means you should think twice before installing this stuff in critical situations such as professional recording or mixing sessions. However, for writing, sound design or home Pro Tools studios it really is worth any inconvenience. With all the free instruments and effects VST To RTAS Adapter makes available, I can't think of another way you could spend this little money and get so much extra gear. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

Getting Started Component Modelling Technology The Oscillators The Filters The Envelope Generators Modulation Saving & Loading The Patch Panel & ESP Coarse Observations MS20FX Manuals Extending The Model Legacy MS20 Spec Playing Legacy MS20 Conclusions

Korg Legacy MS20 & MS20iC pros An excellent recreation of the sound and feel of the MS20. It's even better as a polysynth. The MS20iC has the 'wow' factor. The documentation is clear and comprehensive. Neither MS20 nor MS20FX crashed in a month of testing. It really is very good.

cons The Edit View window is too narrow, and unadjustable. There's no MIDI channel selection — the stand-alone version is always in Omni Mode. It's power-hungry, so polyphony is relatively low.

summary Not only does the Legacy MS20 software recreate a much-loved monosynth, it

Korg Legacy Collection (Part 2) Virtual Instrument/Hardware Controller Published in SOS July 2004 Print article : Close window

Reviews : Software

In the second part of our coverage of Korg's new collection of software instruments, we turn our attention to the virtual MS20, and the MS20-shaped hardware controller. Just how well do these renditions compare to the original? Gordon Reid

Last month, I reviewed the Wavestation component of the Korg Legacy Collection, and found it to be an excellent recreation of the original synths. In many ways, this isn't surprising, because it shouldn't be too difficult to duplicate the function and sound of a 'vintage' digital synth in a new digital form. But what of the other components of the Collection, MS20 and Polysix? Can Korg recreate their success when called upon to model the distinctive sounds and vagaries of two of its most famous analogue synths? This month, it's Legacy MS20's turn for a thorough examination. Original photo: Mark Ewing

Getting Started As I described last month, loading and authorising Legacy Collection is straightforward, although for indefinite use it requires that you obtain a licence number from Korg's web site. Once you have done so, you can use Legacy MS20 as a stand-alone program, or as a VST or AU plug-in. You can also use it within the Legacy Cell environment, which we will look at next month.

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extends it in very useful ways that in no way detract from the character of the original. Together with the MS20FX software and MS20iC hardware controller, it is an imitative software synth of almost unparalleled desirability.

information £399 including VAT. Korg UK Brochure Line +44 (0)1908 857150. +44 (0)1908 857199. Click here to email www.korg.co.uk www.korg.co.jp

Test Spec 1GHz Apple Titanium G4 Powerbook with 512MB of RAM, running Mac OS v10.2.8. Korg Legacy Collection version reviewed: v1.0.0.

The MS20iC controller connects to your computer (and draws its power) via a USB cable. If, like me, you're connecting it to a Mac, you need do no more; the controller will be recognised by the OS X Audio MIDI Setup software. Windows XP users, on the other hand, will need to log on as administrators, and perform a 10-step procedure to install the Korg USB MIDI driver. Although somewhat smaller than a vintage MS20 (it's an 84-percent scale recreation, apparently), the MS20iC hardware controller is an exact recreation of the original synth, with one major exception: the sockets are merely physical representations of the software patching in Legacy MS20. In other words, the front-panel patch points are not real CV connections, and you should not attempt to input control voltages from other synths. The program offers two modes, 'Play View' and 'Edit View'. These are somewhat loosely named, though, because you can play in Edit View and edit in Play View, and both views display the parameter values when you hover over the knobs. However, Play View shows only the front-panel controls that were present on the original synth, while Edit View shows all the additional facilities available. I'm rather frustrated by Edit View, because as with all the other Legacy MS20 screens, it's fixed at just 804 pixels wide, and yet the full control panel is 1491 pixels, necessitating endless scrolling back and forth. With a bit of graphical trickery (see below), I can show you what the panel would look like if you could fit it all on screen at once. Given that my Powerbook displays 1440 horizontal pixels, I would be much happier if I could use this version to program and patch Legacy MS20...

Component Modelling Technology Korg make a big deal of the physical modelling used to recreate the MS20. They call this 'CMT', which stands for Component Modelling Technology, and claim the following: "CMT takes an organic approach to modelling, duplicating the circuit path of a synthesizer by recreating each individual component — transistor, capacitor, resistor, etc. CMT's cloning of the entire instrument ensures that every nuance is replicated." I have doubts about this — specifically about the suggestion that the software synth is constructed in a single operation from models of individual components. Certainly, there is software that does what Korg claim; it's used by circuit designers to 'prototype' boards without having to build them. However, the processing requirements are considerable, and modelling even uncomplicated analogue circuitry remains, even on today's computers, an off-line, non-real-time process. The Macs and PCs on which the Legacy Collection runs do not offer sufficient power to model every component in a synthesizer individually in real time in this way. It might be possible if CMT was a 'linear' representation that ignored all the non-linear responses of the circuits, but this would generate a very sterile sound, not at all similar to the original synth. For this reason, I feel that Korg's description of CMT was probably invented by their marketing department, not their engineers. If this is the case, I can't imagine



why; no matter how Korg have achieved it, the recreation of the MS20 is impressive enough not to need such 'help'.

The Oscillators Korg made some interesting choices when designing the MS20's oscillators. Firstly, they chose different ranges for the two, with VCO1 offering 32', 16', 8' and 4' options, while VCO2 was offset to 16', 8', 4' and 2', with a fine-tuning control offering a range of ±12 semitones. Secondly, they gave both different waveforms, with VCO1 providing triangle, sawtooth, a variable-width pulse and white noise, and VCO2 offering sawtooth, square, a tight pulse with a duty cycle of less than 10 percent, and a so-called ring modulator. As expected, Korg have reproduced this configuration on Legacy MS20, so I would like to compliment the company's engineers on their willingness to recreate the original synth without modifications. It must have been tempting to add the MS10's pulse-width modulation to VCO1, or to add oscillator sync, but they didn't, and decisions like this make the emulation far more accurate and eccentric than it might otherwise have been. I started by testing the triangle waves produced by the two synths. Sonically, the results were similar, although it was not hard to differentiate one from the other. Legacy MS20's triangle wave is a tad brighter than the original MS20's, sounds less 'hollow', and lacks some of the apparent 'bottom end'.

Legacy MS20 in its entirety.

This was not simply a matter of EQ. I tried to equalise the character of one to match the other, but couldn't, so I decided to measure the outputs using an analyser. I discovered that the waveforms are similar, but not identical. Indeed, the spectrums of the two waves are markedly different; Legacy MS20 has stronger mid- and high- frequency components, while the original MS20 demonstrates stronger alternate harmonics. Moving on, Legacy MS20's sawtooth wave sounds fuller than the original MS20's rather rounded waveform, with the original synth being much the buzzier of the two. The magnitude of this difference is clearly visible in the graph plots shown on the next page. Likewise, the square waves and pulse waves exhibit significant differences between the emulation and the original. It is, therefore, remarkable how similar the two synths sound, although Legacy MS20's square wave is clearly the more 'hollow' of the two, and its pulse wave is somewhat louder at high pitches. Next, we come to the device that Korg called a ring modulator, but isn't. This accepts the pulse wave from VCO1 (no matter which waveform you select on the panel) and the square wave from VCO2, and passes them through a device called an XOR gate. This treats the waveforms as '1's and '0's, and creates a



new waveform that has some of the characteristics of a ring-modulated signal. The limitations of this approach are manifold: you can't use any other waveforms, you can't modulate external signals, and you can't use the modulator as a VCA. On the other hand, this 'logic' modulator is very cheap, and has a distinctive, harsh sound that many people like. What's more, the programming required to emulate a gate is straightforward, so it comes as no surprise to find that Legacy MS20's 'ring modulator' is indistinguishable from the original's. The final controls in the oscillator section are simple ones: VCO1 and VCO2 Level controls, Portamento Legacy MS20's oscillator Time, and Master Tune. To sum up the oscillators, it's section. fair to say that there are significant variations in the waveforms and spectrums generated by the emulated and the original MS20. However, the sonic similarities are far greater than the differences, and it would be difficult to quantify them without making a direct A/B comparison.

The Filters Considered thin, weedy and uninteresting in 1978, the MS20's filters have since become icons in analogue synthesis, spawning an industry of blatant copies and thinly disguised imitations. On the surface, there's nothing special about them; they're simply a pair of resonant high-pass and low-pass 12dB-per-octave filters, which will oscillate if the Peak (resonance) exceeds a certain value. But in the late 1980s, when the world rediscovered the squelchy sounds of the Roland TB303, a new dawn for the MS20 was not far behind. My measurements show that the lowest cutoff frequency for both of the MS20's filters is 26Hz. The maximum values are 17.2kHz for the low-pass filter, and 17.1kHz for the high-pass filter. This is not a bad range, but Legacy MS20 puts it to shame, with both filters ranging from 14Hz at the low end to 21kHz at the upper. These differences may not seem huge, but The MS20's sawtooth wave. the low frequency extension is almost a full octave, which is very significant because the high-pass filter does not have a 'hard-knee' response, and so attenuates signals well above the notional cutoff frequency. Consequently, bass sounds are deeper on the software MS20 than they are on the original. Similarly, a low-pass filter with an upper frequency of 21kHz is going to pass more of the file:///H|/SOS%2004-07/Korg%20Legacy%20Collection%20%28Part%202%29.htm (4 of 13)9/22/2005 8:55:35 PM


high-frequency components, and will sound both crisper and brighter. The result is a synth that can sound slightly richer and crisper, or less fizzy, depending upon your point of view. Ignoring the extended range, Legacy MS20 must still imitate the original's distinctive filter characteristics if it is to sound genuine. For example, the MS20's self-oscillation is not a sine wave; the waveform has numerous harmonics, and a distinctive character. Legacy MS20 recreates this extremely accurately. Likewise, if you make the MS20's low-pass filter self-oscillate and then add some oscillator output at a lower, unrelated pitch, you do not obtain two clean pitches: the result is distorted. Legacy MS20 recreates this well, and although it is possible to distinguish it from my hardware MS20, I would be hard-pressed to tell you which was which in a blind test. Another trick with the original MS20 involved placing either filter on the edge of self-oscillation and mixing the oscillators at or near maximum level. If you then backed off the oscillators' volumes, the filter would begin to oscillate. Again, this is imitated with remarkable accuracy.

The Envelope Generators The envelope generators on the MS20 are not ADSRs, nor are they ARs, ADSDs, or any of the other configurations that were common in 1978. One is an HADSR (an ADSR with a Gate Hold) and the other is a DAR (an AR generator with an initial Delay stage). Let's deal first with EG2. Hard-wired to the VCA, this generates the contour that you would expect. However, when you use it to sweep the filter, changing the Sustain Level seems to have no effect on the filter cutoff frequency. Well, that's not quite true. It has a brief effect, but then the filter settles back to its previous cutoff frequency. Legacy MS20's sawtooth wave.

What's happening is this... EG2's sustain level is always set to be 0V at the filter's input, and therefore does not affect the cutoff frequency. This means that, when modulated by EG2, the cutoff frequency starts below where the cutoff frequency would be if not affected by the envelope, rises above this point in the Attack phase, then drops back down to 0V in the Decay phase, in time for the Sustain phase. When you release the key (or when the Hold time is completed) the cutoff drops at a rate determined by the Release to where it started. Consequently, the Sustain Level control determines the height and depth of the frequency shifts in the Attack, Decay and Release stages. Aside from a couple of rare Yamaha synths, I'm not aware of any other manufacturer implementing a file:///H|/SOS%2004-07/Korg%20Legacy%20Collection%20%28Part%202%29.htm (5 of 13)9/22/2005 8:55:35 PM


filter contour in this way, and it is one of the oddities that sets the MS20 apart from the crowd. Happily, Korg have recreated it correctly on the Legacy MS20. What is not identical, however, is the calibration of the envelope times; the slowest Attack on my MS20 is much slower than that on the software version, although the Decay and Release are much the same. EG1, although the less flexible of the two, offers four stages: Delay, Attack, Sustain (at the maximum level), and Release. This acts almost identically on the two instruments, so we need say no more about it.

Modulation To the left of the original MS20's envelopes lies a simple Modulation Generator that offers just two controls: frequency and wave shape. Two basic waveforms are available: pulse, with widths ranging from 0 percent to 100 percent, and sawtooth/triangle/ramp. These are recreated on the Legacy's MS20iC, but the software also adds LFO MIDI sync. When this is switched on, the LFO is synchronised to the tempo of incoming MIDI Clock, the tempo of the host application, or the Tempo in Legacy Cell, depending upon the way in which you're using the program. You determine the association between the external clock and the LFO rate by setting a combination of the Base Note relationship (which varies from 1:1 and 1:32, including triplet steps) and the Frequency/Times knob, which multiplies from one to 16 the number of clock pulses needed to complete a single LFO cycle. It's a simple system, and it works well. The six knobs to the left of the Modulation Generator allow you to direct various combinations of EG1, EG2, and the triangle output from the LFO to the control inputs of the oscillators, the high-pass filter and the low-pass filter. These knobs allow you to create a basic range of vibrato, tremolo and 'sweep' effects, and to shape notes in the usual fashion. But if you want to step beyond the limitations of the pre-patched architecture, you'll need to turn to the patch panel...

Saving & Loading Legacy MS20 has a simple patch save/load system, accessed either via a menu or via the Prog List screen. Firstly, you can rename and write the current patch to any of the 32 locations in the memory bank. Next, you can save a single patch to disk (although this file will not contain the Configuration parameters — see page 190) or save the whole bank. When you wish to load patches, you can reload single patches or complete banks. It couldn't be simpler.

The Patch Panel & ESP For many players, the MS20 patch panel (below right) is little more than a file:///H|/SOS%2004-07/Korg%20Legacy%20Collection%20%28Part%202%29.htm (6 of 13)9/22/2005 8:55:35 PM


modulation controller, allowing you to select and modify a wider range of sources and route them to the destinations of your choice. Sure, it offers an S&H module, but it has no mixers, no multiples, and just a single patchable VCA, which means that you will only get the best from an MS20 by combining it with other modular equipment. Unfortunately, Legacy MS20 does not allow this, so you are limited to using it as a stand-alone instrument. Operationally, I particularly liked the fact that physical patching on the MS20iC controller is echoed on screen so that, when you save and recall patches, you can see exactly what's going on. With 10 physical cables provided, you should have enough for any patch, but even if you run out, you can continue to make connections on-screen by dragging virtual cables from one socket to another. The software does not preclude 'wrong' connections, including the naughty but nice trick of feeding the output from the headphone socket into the Ext Signal In jack. It's just a shame that on-screen patching is not echoed by the magical appearance of a patch lead on the MS20iC controller! Pleasingly, almost everything on the software version seems to work as on the original, including the unusual 'Track and The envelope Hold' behaviour of the mis-named Sample and Hold module generators. (for more on this, see the August 2000 instalment of Synth Secrets, at www.soundonsound.com/sos/aug00/articles/ synthsec.htm). The same — for better or worse — is also true of the External Signal Processor, or ESP. When I bought my first MS20 in 1978, I tried everything to get the ESP to act as a monophonic guitar synth. I experimented endlessly with signal levels, external filtering and compression, and with the band-pass filtering within the ESP itself but, try as I might, the best I could obtain was a glitchy noise that refused to track correctly. In this respect, Legacy MS20's ESP — whether running in stand-alone mode or as a plug-in — is no different from the original MS20's. The ESP works like this. If you're running Legacy MS20 as a stand-alone program, any audio presented to the computer's audio input is directed to the Signal In socket of the ESP. Once the ESP is receiving audio — and the key here is to discover that you must switch it on by clicking on the Audio In legend — you can set the signal level, and use the band-pass filter to isolate the range of fundamental frequencies (if you do not do this, the ESP may try to track harmonics, which is one source of glitching). You can then take the output from the frequency-to-voltage converter, using this as a CV to control other parts of the synthesizer, and the output from the Envelope Follower and trigger generator (which is actually a Gate generator) to do likewise. In practice, this works exactly as it did on the original ESP. In particular, you'll discover that it remains impossible to use the derived CV as an even-tempered



pitch CV, no matter how you adjust the CV Adjust within the ESP itself, or the Ext knobs in the modulation section. No change there, then — but of course you can use the CV to control parameters other than just pitch, and at least the emulation remains faithful to the original, warts and all.

Coarse Observations The MS20iC is lovely, and I particularly like the fact that when you switch it off and on again the program and the Edit screen jump to the physical values of its knobs and inserted cables. Unfortunately, the quantisation of its knobs is too coarse for careful programming. To obtain precise adjustments, you must use the Shift key and mouse to change parameters on screen; programming in this fashion offers much finer control. This method allows you to program an Osc2 detune of 0.02, which results in a rich, slow 'chorusing' between the oscillators. In contrast, the smallest division available from the MS20iC's front panel is 0.19, which is far too great to create this effect.

MS20FX If you're using Legacy MS20 as a plug-in rather than as a stand-alone software synth, you'll notice that a second program appears in the effects list of your host application. Called MS20FX, this has all the facilities of the MS20 software plus stereo inputs from within the host application (note, however, that MS20FX is unable to process audio presented to the computer's audio inputs). The stereo inputs connect to the Ext Signal In sockets in both the main patch panel and the ESP, so you can use MS20FX to apply filter and amplitude modulation to the sounds of other computer-based instruments and/ or use (or attempt to use...) the ESP to derive pitch and amplitude information from their signals.

Legacy MS20's patch panel.

The benefit of this becomes apparent when you experiment with the 32 MS20FX patches (or Templates) supplied by Korg. I have owned an MS20 for 26 years, but I still hadn't tried some of these, especially the trick of overdriving the ESP for some excellent distortion effects. Thank you, Mr Programmer! Directly comparing the two synths, I found that the ESP in the Legacy Collection works slightly better than that of the original MS20, but it will still glitch and become unstable in all but the most favourable of circumstances. Again, this is a testament to Korg's desire to produce an exact replica of the vintage synth. I suspect that it would have been easier to create something that works well than file:///H|/SOS%2004-07/Korg%20Legacy%20Collection%20%28Part%202%29.htm (8 of 13)9/22/2005 8:55:35 PM


to model something so erratic, but the only concession the company has made to the 21st century is to make the ESP stereo, which you will discover if you increase the Polyphony and Unison to values greater than '1', and set the Unison Spread to greater than zero. All of which brings us to...

Manuals Using Legacy MS20 and MS20FX is a doddle for anyone who has experience of the original MS20, and for anyone prepared to read the comprehensive (but not overly long) documentation. Of particular benefit for novices are the original MS20 manual and patch charts, both of which are provided on CD-ROM in PDF format — a nice touch.

Extending The Model As you can see in the screenshot above, Legacy MS20's 'Edit' screen possesses seven sets of controls not found on the original MS20, and these extend Legacy MS20's capabilities way beyond what was possible on the hardware MS20 back in 1978. The first of these (Voices) allows you to determine the polyphony, and how many voices will be overlaid in Unison. The former accepts figures from one to 32, and the latter from one to 16, meaning that polyphony can be as high as 32 voices, or as low as one, with up to 16 voices under each note. In addition to this, there are the Mono and Poly assignments, the former of which overrides the numeric Poly setting and allocates the total polyphony to a single key whether or not Unison is The Modulation section. greater than '1', and Single/Multi triggering, which refer only to monophonic modes. Underneath these, you'll find the Unison detune and spread knobs which, as you would expect, determine the amount by which the pitches of the unison voices are detuned and spread across the stereo field. Unfortunately, there is nowhere to set the key priority. The original MS20 used single triggering and high-note priority, but Legacy MS20 has last-note priority, so there is no way to get the software to replicate the playing style of the original synth. You also have to be careful of the interaction of these six controls, because they can lead to at least one unexpected result. If you set the polyphony to be greater than '1', the voice mode to Mono, and the Detune and Spread to '0', you'll find that the tone of your patches changes when you change the polyphony. At first, I



thought that this was a fault, but it isn't. Given that the software is emulating an analogue synth, the voices are not identical, but they are so close that, with no detune or spread, you obtain phase cancellation. Every time you change the amount of polyphony, you are changing the nature of the cancelling, and thereby the tone. So beware. You must also pay attention to the amount of power demanded by higher polyphony and unison settings. Like Legacy Wavestation, Legacy MS20 is a power-guzzler and, if you're not careful, you're going to exceed your CPU's capabilities. I found that, depending upon the nature of the patch, I could typically get as high as 18 voices before the sounds started to glitch, or acquired a grungy quality. This is quite low, particularly when unison is '2' or higher, but the consequence of setting the polyphony even lower is note stealing. If you want to stop the computer maxing out, you can set a CPU loading threshold that, if exceeded, will cause the MS20 to stop producing audio. This means that there will be no nasty digital noise if the processor overloads. The next panel has just one control, marked 'Analog'. This adds a random CV variation to the pitch and filter cutoff frequencies every time you press a key. Some people will like this, but I found it most pleasing when set to The External Signal Processor, or ESP. zero. I've not noticed random tuning errors on my vintage MS20, and I don't see that I should insert some where none previously existed. Alongside this, you'll find a pan-pot that affects the relative loudness of the left and right channels in the final output and, above this, the Pitch section which determines the pitch-bend range (using a MIDI pitch-bend wheel, joystick or whatever, not the MS20's wheel) and a global transpose of ±24 semitones. Between these, the Ext Audio Gain settings allow you to set the input and output gains of the ESP. The final two additions are very important, because they allow you to control the synth using your choice of two MIDI controllers from a list of eight: key velocity, pressure, pitch-bend, mod wheel, breath controller or MIDI MIDI continuous controller (or CC) numbers 16 or 17. Source 1 offers five destinations (the filter cutoff frequencies, VCO1 pulse width, VCO2 pitch and audio VCA gain), while Source 2 offers a different set: the filter cutoff frequencies (again), FM depth, high-pass filter modulation depth, and low-pass filter modulation depth. Independent bi-polar knobs determine the depth of the effect at each destination, so although the number of controllers and destinations seems somewhat limited by modern standards, the amount of control available is considerable. What's more, you can remap the source controllers to CC numbers of your choice from #1 to #95. You do this in Configuration view, which is also where you determine the polarities of the MIDI CCs, the master tuning, the temperament (equal, pure major, pure minor, or any one of 13 User scales) and set the MIDI filters on or off. However, be warned that these settings are lost each time you close the



program, unless you save them in an FXB bank. Again like Legacy Wavestation, Legacy MS20 has no SysEx capabilities, but its MIDI capabilities are still extensive. Most obviously, you can assign new MIDI CCs to any of the knobs and switches on the front panel. The simplest way to do this is to hold down the right-hand mouse button (on a PC) or the Ctrl key (on a Mac) and click on the control of your choice. You can then assign any MIDI CC number from a drop-down list, or tell the parameter to 'learn' the next incoming CC. However, if you change the CC assignments in the software, there will no longer be a correspondence between the control legending on the MS20iC, and the effect of these controls on screen. If you are courageous enough to change any assignments, you can save the new controller map using a menu hidden behind the Korg logo. This saves and loads maps, allows you to display all the assigned CCs, and provides a clipboard to cut and paste patches from one memory to another.

Legacy MS20 Spec Patches: 32. Oscillators: Two. Maximum polyphony: 32 notes. Number of patch points: 27 plus signal and headphone output. Formats: Audio Units (AU), VST, or stand-alone (no MAS). Legacy Collection minimum computer requirements: 1.5GHz Pentium 4, 1.3GHz Centrino, Athlon XP/2000+, or 1.8GHz Celeron-based PC running Windows XP Home or Professional Edition with 256MB of RAM (512MB recommended), or 800MHz G4 Apple Macintosh running Mac OS 10.2.6, with 256MB of RAM (1.25GHz G4 with 512MB of RAM recommended).

Playing Legacy MS20 Like many software synths, Legacy MS20 offers an on-screen keyboard, which is useful as a test-bed when creating new sounds, but is pretty much useless for anything else. Likewise, although the MS20iC points the way forward for all 'soft' instruments, its 37-key mini-keyboard lacks cop for serious players. It is velocity sensitive, but it's also spongy, and feels to me like a procession of black and white springs. That's a great shame, and I would be prepared to see the package cost a little more and include a full-size recreation with a proper keyboard. Then you could program sounds from the hardware controls and play from the full-size, velocity- and pressure-sensitive MIDI keyboard, all on one unit... now that would be something! But what I really like about Legacy MS20 is its polyphony. Back in the 1970s, a handful of manufacturers tried to manufacture polyphonic versions of their file:///H|/SOS%2004-07/Korg%20Legacy%20Collection%20%28Part%202%29.htm (11 of 13)9/22/2005 8:55:35 PM


monosynths, but it wasn't until recently that software synths demonstrated what it would be like to hear genuine, polyphonic versions of revered monophonic instruments. The results can be stunning. Increase the polyphony, add Unison, spread and detune, add suitable effects and... oh wow! Rich ensemble strings, huge cathedral organs, and all manner of pads, brass, and other traditional polysynth sounds just pour forth. Of course, there are some sounds for which the MS20's architecture is not best suited. For example, the lack of oscillator sync makes the synthesis of acoustic and electric pianos difficult. But I was still amazed by the quality of sounds I The extra controls in Edit view. was obtaining quickly and easily. As for nonimitative sounds and effects, a polyphonic MS20 is a dream. Oh yes, and the MS20FX incarnation deserves another mention, because the results of playing other software synths through it can be quite remarkable. Turning to sounds I don't often use, I discovered that a polyphonic MS20 is also a dance musician's dream, at various times producing the bass, stabs and percussion effects that are so useful in these genres. However, this was also where I discovered a significant shortcoming... Attempting to launch multiple instances of the stand-alone Legacy MS20, I realised that there was no way to allocate each to a single MIDI channel. In other words, the stand-alone program is always in Omni Mode. Bah! The only other problems I encountered occurred when using the package as a VST plug-in. Firstly, it had a tendency to produce stuck notes when the polyphony/unison was high and I played fast runs. This did not occur in standalone mode, so I suspect that it may have been a problem in my chosen VST host. Likewise, the VST version did not recognise patch cables when I inserted or removed them from the MS20iC. I could still use 'virtual' cables on screen, and everything worked correctly, so again, this may have been down to a problem with my host package. But to place this in context, it's worth noting that, while it's impossible to test every possible situation, I put the MS20 and MS20FX through some very stiff tests and they steadfastly refused to crash. This reflects very well on Korg, especially since I was using v1.0.0.

Conclusions Korg have done an excellent job duplicating the timbre and response of the vintage MS20, recreating many of its subtleties with remarkable success. Indeed, the sound is close enough to fool anybody without the opportunity for a direct A/B comparison. Sure, some patches are not exactly the same, perhaps because



Legacy MS20 is somewhat better-behaved than my 26-year-old bunch of circuits, but the differences are perhaps no more than you might expect had Korg built a new, analogue MS20 and placed it alongside the original. Bearing in mind that there were also two hardware versions of the original, and that these are reputed to sound slightly different from one another, the differences between Legacy MS20 and my 'Blackboard' MS20 become even less significant. Inevitably, the feeling of déjà vu is particularly evident when you use the MS20iC, but its importance is even greater than that. By including a dedicated hardware controller, Korg have made Legacy MS20 genuinely desirable for reasons other than cost or convenience. As for the cost... at £399, the Legacy Collection is stunning value. Had Korg released just MS20, MS20FX, and the MS20iC for the same price, it would still not have been overly expensive. So, what's the catch? To be honest, there isn't one. The MS20 has more than enough character to deserve recognition as one of the important synthesizers of the 1970s, and if you're after a straight recreation of this, you'll be very happy. If you're hoping that sensible enhancements will extend the MS20's usefulness, you'll be more than very happy. Even if you're not a fan of the original synth, you'll be impressed. It's a winner. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



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In this article:

Alma Flamenco **** A Funky Future *** Vienna Overdrive ***** Cuckooland Completely Bonkers ****

Latest Sample CDs Sample Shop Published in SOS July 2004 Print article : Close window

Reviews : Sound/Song Library

Star Minority Extreme Sports ***** Helibungy **** Street Luge *** Wet Zorbing ** Extreme Ironing * Wheelbarrow Freestyle

Alma Flamenco **** MULTI-FORMAT This library kicks off with the kind of fast nylon-strung guitar strumming traditionally associated with Flamenco, and these examples cover a range of chords so that they can easily be stitched together to create something credible. The recording quality is extremely good (as is the playing skill), and in addition to strums there are lots of fast finger-style passages and chordal arpeggios, as well as instantly recognisable Spanish-style chord progressions. Fitting these together is pretty easy, and could be used to knock up a couple of holiday TV themes or advert soundtracks quite quickly. Where more authenticity is desired, you can layer on other Spanish sounds and voices from later in the CD, but do try to avoid the worst of the film clichés, please! After the guitar parts comes a wide selection of impassioned male vocals, some that work a cappella and some that could be slotted in over or between the guitar flourishes. Again the recording standard is very high, and the performances are professional and authentic sounding. The main vocal lines are followed up by a few harmony phrases and then the obligatory angst-filled female vocal phrases — why is it that Spanish women always sound cross? Once again, these are dripping with authenticity. Once the ladies have given voice to their woes, there's bongo percussion (loops and hits), hand drums, claps, finger snaps, and of course castanets. These layer beautifully with the guitar parts, or you can use them to underpin your own compositions. Finally, the guitar makes a bit of a comeback with some seriously virtuoso playing and some single-note multisamples. While single-note multisamples are very useful to create your own riffs to link the included phrases, not having them arranged as ready-to-go sampler programs means that you'll have to do a little work before you can use them. However, they shouldn't need

file:///H|/SOS%2004-07/Latest%20Sample%20CDs.htm (1 of 6)9/22/2005 8:55:40 PM

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looping, so it's not too daunting. In all, this is a great Flamenco construction kit that can add depth and authenticity to any music that has a Spanish flavour. Paul White Audio CD and Acidised WAV/REX 2 CD-ROM set, $55 (around £31). Click here to email www.discoverysound.com

A Funky Future *** MULTI-FORMAT As the title suggests, the loops within this library are intended to provide a modern take on funk, and they are split into four elements: drum loops, bass loops, various percussion loops, and 'guitarscapes' (of which more in a moment), all originally recorded at tempos between 80bpm and 120bpm. For me, the highlight of the collection is the basses. These are divided into Double, Warwick, and Rubber types, and in the main they are left very much untreated with little or no processing. Both playing and recording seem to be of a high standard and the licks themselves have plenty of funky attitude. In use, it is very easy to combine loops to form a complete bass line suitable for a full song structure. There is also plenty of finger noise and the occasional sound of strings slapping on the fretboard (particularly noticeable on the double bass loops) and this adds to the sense of realism when the loops are placed within a full mix. There is also plenty of bottom end, especially on the Rubber loops. Things are a noticeably more processed when it comes to the drum loops. While the playing is certainly funky throughout, there are very few untreated, straight acoustic kits here. Almost every loop has been tweaked or filtered in some way to give the sound a more contemporary feel. With a total of about 100 two-bar loops, there is a reasonable selection, but each loop tends to be quite distinctive. Mixing and matching therefore takes a little more care, and if you don't have any luck here then you'll have to turn to beat slicing of a single loop in order to program in some subtle variations. The percussion loops, also heavily processed, can add some variety. These are split into Percussive and Hi-freq groups. Many of the loops in the latter category are based on hi-hat or shaker loops, but they've been filtered to provide additional movement in the sound — these can be made to work really well over some of the main drum loops. Odd man out is the 'guitarscapes' section. While these loops were (obviously!) recorded on guitar, the processing means they tend to have a more bed- or drone-like feel. I'm not sure this is what most people would think of as 'funk guitar', as the pad-like textures lack any real sense of file:///H|/SOS%2004-07/Latest%20Sample%20CDs.htm (2 of 6)9/22/2005 8:55:40 PM

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rhythm. I could imagine them working well in an electronica context, but perhaps some straighter funky wah-wah, or even a collection of processed keyboard loops, would have fit the bill a little better here. Overall, it's obviously nice to get all three sample formats within the same twoCD set, and the loops are all very well recorded and played, but it would have been nice to see just a little more material in order to shout 'bargain!' with greater conviction. John Walden Audio CD and WAV/REX CD-ROM set, 99 Euros (around £67) including 25 percent VAT. MI-7 +46 406 306 970. +46 406 992 509. Click here to email www.mi-7.net

Vienna Overdrive ***** GIGASTUDIO/EXS24 MKII Kerrang! The makers of the Vienna Symphonic Library rip off their orchestral dickie bows, pull on their Iron Maiden tee-shirts, fluff out their mullets, and indulge in some serious rock head-banging with this foray into the world of amplified instruments. Featuring a seven-string solid-body electric guitar played through vintage Marshall 4x12 cabinets, the 5.8GB library was performed by a small team of guitarists under the watchful ear of VSL's Michael Hula. Like Vienna Concert Guitar, the library was recorded in the company's Silent Stage environment, but rumours that the Vienna team constructed an acoustically treated mosh pit within the space are almost certainly untrue... As the name suggests, these guitar samples are mightily overdriven and distorted throughout, recorded fairly dry, but with release samples adding a hint of room ambience as well as authentically crunchy noteoff noises. The basic tone is broad, cutting, and raunchy, and the extra low 'B' string takes you down into bass guitar territory. You can select non-vibrato samples and wail away Jan Hammer-style on the pitch wheel, or play the vibrato samples chordally to create a big, singing Brian May 'guitar choir' effect. For a more histrionic delivery, Whammy-bar vibrato and dives are available. Long notes last about five seconds, but if you want Nigel Tufnell-style infinite sustain ('Just listen to it!'), there are looped versions of the long notes. Sustains and medium-length notes come in one dynamic only — LOUD, dude! — but staccato samples (available in two lengths) have three dynamic layers, which adds a lot of feel to funky rhythm passages. Vienna Overdrive offers an enormous selection of guitar techniques: harmonics, 'noisy' grace notes which leap theatrically to a sustained note, tone and semitone file:///H|/SOS%2004-07/Latest%20Sample%20CDs.htm (3 of 6)9/22/2005 8:55:40 PM

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bends, exciting feedback and string scrape effects, mandolin-style tremolos, trills ranging from a semitone to a major third, 220bpm speed-metal repeated notes, and even samples played with an E-bow (a magnetic gadget held over the strings to produce eerie violin-like sustains). All these variations are performed with panache, and given a very aggressive attitude by the heavy amp distortion. Heaviest of the lot are the power chords; played in full major and minor voicings, fourths, or (my personal favourite) fifths, they sound satisfyingly full and fruity, without getting too middly or 'fizzy'. With the addition of the Performance Tool, the library gets even more tasty, expressive, and real-sounding. A very full implementation of performance styles includes repeated series of single notes, power chords, 'palm muted' short notes, and thrashy rhythm noises, in a choice of dynamics and tempos. Dial up the performance legato straight notes, hammer-ons, and pull-offs, and you can do a passable imitation of the legions of guitarists who impersonate Allan Holdsworth, the king of fluid legato guitar. I also greatly enjoyed using the 'legato slide' samples to play the slithery, 'raga'-style melody of Jimi Hendrix's 'Third Stone From The Sun', aided by the performance legatos' Second Track Sustains, a dual-MIDI-channel system which automatically tacks a long sustained note onto the end of each slide. This mega-sampled guitar has impeccable rock credentials. Its fast octave runs (played with great aplomb in all keys, up and down from every scale note, by Konrad Schrenk) evoke US guitar virtuosos Steve Vai and Eddie Van Halen, and detuned 'sub' multisamples stretching down to C0 add a contemporary Korn/ Meshuggah-like flavour. Like the Vienna Concert Guitar, Overdrive enjoys luxurious, up-market packaging and an extensive booklet packed with precise musical detail. Slash and Yngwie would snort at the idea, but thanks to a bunch of classical-music enthusiasts from Vienna, keyboard players can now transform themselves into virtual guitar heroes while avoiding the concomitant hair, booze, drugs, tattoos, Spandex, hearing damage, and negative-lifestyle issues. Dave Stewart Gigastudio or EXS24 MkII DVD-ROM, £130 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.viennasymphoniclibrary.com

Cuckooland Completely Bonkers **** MULTI-FORMAT

This library claims to bring you 'probably the most inspiring and weird sample collection ever produced', and is arranged into four sections: Ambiences, Bits, Beats, and Musical Loops. The WAV files can be dragged into most sequencers file:///H|/SOS%2004-07/Latest%20Sample%20CDs.htm (4 of 6)9/22/2005 8:55:40 PM

Latest Sample CDs

without the need for a sampler, and it's easy enough to loop the rhythmic examples. Ambiences is a great section for instant gratification, as it contains some really nice evolving samples, each of which lasts around 30 seconds. There's everything from lush angelic choirs to creepy backgrounds — I was immediately tempted to fire up my sampler and start composing a score for an imaginary film. The Bits section contains a number of short samples that could be useful in the construction of wild bass lines or for adding effects to film soundtracks. Titles such as Loud Feedback and Electronic Monster probably give you the general picture here, and there is some interesting use of filtering in some of the bass samples. The Beats section is divided into five subsections of Slow (below 100bpm), Faster (101-120bpm), Quick (120bpm and over), Strange, and Funny. All of the drum samples appear to have been treated to some degree. The beats in the first three sections have a clean sound, and would probably work well layered with programmed drums to give a more realistic feel to your rhythmic parts. The drum samples in the Strange and Funny sections, on the other hand, are clearly electronic in origin, and it would be worth dipping into these sections if you're interested in the more ambient side of music making. I suffered a bit of a sense of humour failure when I reached the Funny section, which is at the bleepy end of strange and might be considered the least inspirational of the samples contained on the CD-ROM, unless you are inclined towards minimalist electronic music, in which case some of the samples could be highly appropriate. The last section of the CD-ROM offers a number of rhythmic loops with both descriptive titles and tempo indication. Firing up my copy of Propellerhead Software Rebirth and running it against some of the samples on offer immediately created interesting textures that grooved along very nicely indeed. I am always slightly wary of the use of terms such as 'completely bonkers', so even though the Cuckooland series to date has something of a reputation for sonic weirdness, I was still a little wary. However, when I listened to the samples I was, in the main, impressed. I think this sample collection would be a useful addition to your sample library if you like your samples slightly off-the-wall, and it might just spark off some creative ideas or provide that extra piece of ambience needed to add depth to the track you're working on. Tony Beech Audio CD, £60; Akai, EXS24, Gigastudio, Halion, Kontakt, Reason Refill, REX, or WAV CD-ROM, £60. Prices include VAT. AMG +44 (0)1252 717333. +44 (0)1252 695680. Click here to email www.amguk.co.uk www.samples4.com


Latest Sample CDs

Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Mission Pro SM6A

In this article:

Drivers & Design Amplification & Controls In The Sweet Spot

Mission Pro SM6A £599

Mission Pro SM6A Active Nearfield Monitors Published in SOS July 2004 Print article : Close window

Reviews : Monitors

pros Great basic sound. Good dynamics and highlevel performance. Useful range of input connections.

cons Bass not perfect — should have been a closed-box design. Slightly bright balance.

summary A usefully competitive product from Mission Pro, and at this UK price you can't go wrong.

information £599 per pair including VAT. Sound Technology +44 (0)1462 480000. +44 (0)1462 480800. Click here to email www.soundtech.co.uk www.mission professional.com

Although Mission are perhaps better known as a leading UK manufacturer of hi-fi speakers, they have now launched their first active nearfield specifically designed for studio use. Phil Ward

Paul White had a look at the SM6A's passive sibling back in SOS September 2003, describing a successful first foray into the studio market for hi-fi major player Mission. Now, with the launch of the active version, Mission Pro have pretty much the entirety of the entry-level monitor market covered. And without wishing to let the cat out of the bag and reveal my conclusions in only the third sentence of the review, they have it covered pretty effectively too!

Drivers & Design

Photos: Mike Cameron

The SM6A looks like a typical small active nearfield monitor, and in many respects it is just that. It does however include a few features and functions that mark it out in an increasingly large crowd. As ever, there's an amplifier module bolted to the rear panel of the enclosure that carries mains and input sockets, an indented gain control knob, and a power switch. Beneath the module there's a reasonably well flared reflex port — although not, unfortunately, flared on its inner end. Moving round the front, there's a 150mm bass/mid-range driver and a 25mm fabric-dome tweeter mounted on a profiled MDF front baffle. While the tweeter is a pretty ordinary-looking device, the bass/mid-range driver features Mission Pro's proprietary Paramid cone material, described as 'a sandwich of

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Mission Pro SM6A

aramid fibres and pulp paper to form an optimum combination of low resonance and high rigidity'. Loudspeaker designers have, since time began, been searching for cone materials that display that 'optimum combination of low resonance and high rigidity', along with the vital, but missing from Mission Pro's description, low density. Using paper reinforced with fibre, as in Paramid, is one that generally works well and offers an effective compromise between the many conflicting cone material requirements. It's the one I'd look to first if designing a monitor of this type. At 32mm in diameter, the bass/mid-range driver's voice coil is significantly larger than would be typically found in a hi-fi speaker of similar size, and bodes well for the SM6A's thermal power handling and compression performance. The polished aluminium-look bass/mid-range driver chassis is not, however, all it seems. The visible part is actually a painted plastic bezel, the driver chassis itself being a separate item screwed independently to the front panel beneath. At best, the bezel offers some extra clamping force to the driver chassis; at worst it's just for looks. Mission Pro introduced inverted drivers (tweeter below bass/mid-range) on their hi-fi speakers many moons ago, and the SM6A is primarily intended to be used in this orientation. However, thanks to a rotating badge it can be mounted either way up (and sideways if convenience is really more important to you than potential monitoring accuracy). Inverting the drivers is one of those ideas that it's a wonder is not more popular. It's of benefit because, with a listening position typically just above the top of the enclosure, it equalises the path length between each driver and the listener — approximately time-aligning the drivers and ensuring that the integration intended by the crossover design is actually delivered at the listening position. Next time you see a 'time-aligned' speaker with a stepped or angled front panel, just ask yourself why the designers didn't simply invert the drivers and leave the front panel flat. Another sign that Mission Pro have worked hard at optimising the integration of the drivers is the steeper than average crossover filter slopes — fourth-order lowpass on the bass/mid-range driver and third-order high-pass on the tweeter. Steep filter slopes such as these are likely to result in a speaker that shows good frequency-response consistency off axis. Not only will this mean that there'll be less variation in sound at different listening positions than is likely with a more primitive crossover design, it also means that the basic character of the speaker is likely to be relatively uncoloured, as the tonal balance of the early side-wall reflections will be reasonably similar to the direct sound.

Amplification & Controls Not only is the SM6A blessed with some sensible electro-acoustic engineering, some worthwhile thinking was also put into its use. The amplifier panel carries a comprehensive set of input sockets — balanced XLR, balanced/unbalanced quarter-inch jack, and unbalanced RCA phono. For once, with an active speaker, file:///H|/SOS%2004-07/Mission%20Pro%20SM6A.htm (2 of 4)9/22/2005 8:55:45 PM

Mission Pro SM6A

there's no excuse for not having a workable signal cable to hand! There's a signal-sensing automatic switch-on feature, and the amplifier panel also plays host to two knobs: a Bass Contour knob that dials in one of four low-frequency response shapes; and a Mid EQ knob that provides Studio and Hi-fi options. The latter control is intended to provide either the tonal accuracy of a nominally flat frequency response (the Studio option) or the attenuated mid-range said to be typical of a hi-fi speaker. I have to admit to a little scepticism about the real-world value of these EQ fiddles. Leaving aside the fact that it's a pain to have to reach around behind the speaker and flick a switch, and that these days it's so easy to insert a global EQ on the monitor output, the complexity behind the perception of tonal balance with a particular monitor located in a particular position in a particular room is such that simple, broad-brush equalisation can't really hope to do anything but make things sound different — not really better, or more accurate, just different. Having said that, it may well be that one of the SM6A's low-frequency response shapes will suit a particular listening environment slightly better than others, but there's no way it will rescue inappropriate speaker positioning or solve room-acoustic problems. Similarly, it might be in some applications that the sound of the SM6A is preferable, or even more useful, with the EQ switch set to Hi-fi, but my experience of the tonal balance of hi-fi speakers is that they range from seriously recessed in the mid-range to outrageously boosted. There is no 'typical' hi-fi speaker mid-range balance.

In The Sweet Spot Enough carping! I wrote in the opening paragraph that Mission Pro have the small active monitor market covered effectively, and so they do, because the results when you actually get to switch on and use the SM6A are good. The speaker's most impressive attribute is its mid-range character, where it pulls off the unusual trick of being at the same time well-balanced, uncoloured, and analytical. I felt I could work with it right from the start, with judgment of midrange balance, while hearing right down into the tail of a reverb, made easy. Further up the bandwidth, the SM6A's top end seemed balanced slightly on the bright side (ironically, high-frequency balance is the one area where simple EQ fiddles can work) and if I were to use the monitor full time I'd probably set up a simple HF shelf EQ (perhaps -1dB from 4kHz up). The basic quality of the tweeter is good though, and I'd have no problem listening to it for extended periods. I was impressed too with the SM6A's dynamic performance and ability to play loud without falling apart, both in terms of performance and construction. My needs in this respect are never particularly extreme, but the SM6A exceeded them by no small measure. It appears unafraid of hard work.


Mission Pro SM6A

If the SM6A disappoints in any area, then it's at the bottom end of the audible range. I'm no great fan of reflex-loaded speakers and, whatever the position of the Bass Contour control, I felt I could hear the slight bass transient blurring and indistinct pitching typical of the technique. The effect was mild — although the SM6A's bass isn't perfect, it's still perfectly competitive — but I was reminded once again of why I believe sealed-box speakers work better, and I was left wondering why manufacturers are so wedded to choosing bass extension over bass accuracy. Pulling all the strands together, I was left with respect for the SM6A. It's not easy to manufacture and sell a well-engineered and good-looking active monitor at this UK price — especially one which works as well as this. So congratulations Mission Pro, and if I can make a request, please leave the reflex port out of the SM6A MkII... Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



MOTU 828 MkII

In this article:

On The Outside The 828 MkII On The PC I/O Silver Lining? In Use A Small Step For An Interface...

MOTU 828 MkII Firewire Audio Interface (Mac/PC) Published in SOS July 2004 Print article : Close window

Reviews : Computer Recording System

MOTU 828 MkII £695 pros Cost-effective. Excellent I/O support with high sample-rate capability. Good sound quality. Can be used as a standalone mixer. Good bundled audio software.

Three years ago, Mark Of The Unicorn were the first manufacturer to bring out a working Firewire interface. Now, they've replaced the 828 with a MkII version offering high sample-rate recording, more flexible clocking, MIDI I/O and better metering — and, what's more, they've cut the price.

cons Optical S/PDIF and ADAT I/ O share the same socket.

Paul White

summary

The original MOTU 828, launched in The MOTU 828 MkII is a 2001 and reviewed by SOS in July of significant improvement on that year (www.soundonsound.com/sos/ the well-respected original jul01/articles/motu828.asp), has proved 828 offering better metering, word clock in and out, SMPTE to be an incredibly popular Firewire support, enhanced I/O audio interface for both Mac and PC capability and good hardware users wanting multiple channels of metering, all for £100 less analogue and digital I/O. It has now than the older model. been superseded by the more sophisticated 828 MkII, which brings with it 24-bit, information 96kHz recording capability, improved connectivity and hardware metering. As £695 including VAT. before, MOTU's Mac-only Audiodesk multitrack recording software is bundled Musictrack +44 (0)1767 with the interface, and offers a multitrack recording and processing environment 313447. not dissimilar to the audio side of Digital Performer. You also get the Cuemix +44 (0)1767 313557. utility, which can set up the mixer for computer-controlled or stand-alone Click here to email operation, and this has evolved to include some original features added since the www.musictrack.co.uk manual was written, so you need to download the latest PDF files to find out how www.motu.com these work. Perhaps most important is the addition of control surface support for devices such as Mackie Control, though Talkback and Listenback have been also been added, which is welcome news for anyone operating a separate control room and studio. The best news is that despite the many improvements, the price is now £100 lower than the old 828.

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MOTU 828 MkII

On The Outside Housed in a rather more stylish 1U rack case than its predecessor, the 828 MkII features eight analogue inputs on balanced jacks, 10 analogue outputs on balanced jacks and two combi jack/XLR mic/line/instrument connectors (with individually switchable phantom power), which are now on the front panel for easy access. Insert send jacks for these two channels are fitted to the rear panel, and any free outs or ins can be used as aux sends and returns. Gold-plated connector contacts are used throughout and the converters are 24-bit, 64x oversampling with 96kHz capability. The analogue input jacks can be individually switched to either +4dBu or -10dBv sensitivity and up to 6dB of additional digital gain is available if needed. Further digital I/O is catered for by a pair of Toslink/ADAT lightpipe input and output connectors, which provide New to the 828 MkII are MIDI I/O, word clock eight channels at 44.1kHz and 48kHz support and SMPTE sync. or four channels at high sample rates, and stereo S/PDIF in and out on dedicated RCA phonos, which can also be duplicated on the optical I/O. A standard ADAT 9-pin sync socket is provided for the sample accurate sync'ing of an ADAT or ADAT-compatible recording/playback device and there's now both word clock in and out on standard BNC connectors, so if you have a master clock unit, there's now somewhere to plug it! Usefully, the word clock circuitry can also follow and generate high and low sample rates so that, for example, you could be running at 44.1kHz but still lock to a double-speed 88.2kHz clock, or vice versa as selected in the MOTU Firewire Audio Console. The other welcome addition is a second Firewire socket, which makes life easier when you need to chain peripherals. Other niceties include MIDI In and Out, on-board SMPTE sync for those rare occasions when MTC isn't what you need, and the ability to use the unit as a stand-alone mixer. As with the original 828, there's a rear-panel footswitch jack that can be used to punch in and out in Audiodesk and other compatible audio applications (this can be set to emulate any keyboard keystroke), a headphone output that can follow any output pair, a master level control that governs the headphone and main out level and Cuemix zero-latency monitoring. Having a physical output level control is particularly useful for anyone using active monitors without a mixer, as this provides them with a means to adjust their monitoring level. The main outputs themselves are equipped with 128x oversampling converters and may be used to provide a main stereo output feed, or they can carry the zero-latency cue mix. Cuemix can be used to send the signals being recorded directly to the outputs rather than monitoring them via the computer, so that they can be heard without any delay, albeit with no added plug-in effects. Having said that, MOTU have made it easy to connect a hardware reverb unit for monitoring by adding an effects loop controlled by an aux send. Cuemix is also useful for monitoring MIDI hardware instruments directly rather than via aux channels set up in your file:///H|/SOS%2004-07/MOTU%20828%20MkII.htm (2 of 7)9/22/2005 8:55:51 PM

MOTU 828 MkII

sequencer, though of course you don't get zero latency with software instruments as their sound is generated inside the computer. As supplied, the 828 MkII comes with multi-channel ASIO drivers for Mac and PC as well as a Mac OS X Core Audio driver.

The 828 MkII On The PC While the main review text demonstrates that the MOTU 828 MkII is an excellent solution for the Mac-based musician, the unit is, of course, also aimed at PC users. Although the Audiodesk recording software is only available for Mac, all the other key features outlined in the main review text still apply within a PC context. In addition, as the majority of users would probably be integrating the 828 MkII into an existing recording/sequencing environment with applications such as Nuendo, Cubase SX or Sonar, the loss of Audiodesk will probably not be a major issue for most PC-based potential purchasers. I was able to test the 828 MkII with both desktop (2.6GHz Pentium 4 with 1MB RAM) and laptop (2.0GHz Pentium 4 with 512MB RAM) systems, both of which were running Windows XP. Driver installation, which has to be done prior to connecting the MOTU to the PC, proceeded without a hitch. The review unit was supplied with version 3.1 of the drivers but a newer version (3.3) was available from the MOTU web site as a 2.6MB download, and I used the latter in testing on both systems. Using the laptop, the only (very trivial!) problem I encountered was the need for a six-pin to four-pin Firewire cable, as many laptops feature the more compact fourpin Firewire connectors. MOTU do supply a healthily long 10 foot six-pin to six-pin cable, and this worked fine with the Firewire PCI card in my desktop PC. The performance of the 828 MkII was excellent when tested with both Reason (v2.5) and Wavelab (v4.01b). The system felt very responsive and there were no glitch issues even at a very comfortable 128 samples-per-buffer setting. Real-time performance of soft synths within Reason via the MOTU MIDI In was perfectly acceptable. I could easily imagine combining the 828 MkII with a Firewireequipped PC laptop to form the core of a very respectable mobile recording setup. With the desktop PC I was able to test using Cubase SX (v2.0.1.10), Sound Forge 7, Wavelab (v4.01b), Acid Pro (v4.0d) and Reason (v2.5). In using all these applications, I only experienced one minor problem. At the 128-sample buffer setting, playback within Wavelab produced some minor glitching using the ASIO drivers. This disappeared at higher buffer settings and was also not an issue when using the MME-WDM drivers. As Wavelab had functioned fine on the laptop system, I'm inclined to think this might have been something specific to the hardware or OS setup on the desktop PC, rather than an issue with the 828 MkII or its drivers. In all other respects, the MOTU system performed flawlessly. Indeed, once hooked up, use within SX was totally transparent. All the various inputs and outputs were available via the SX VST Connections dialogue. Configuring the various analogue outputs to feed a surround sound monitoring system within either SX or Acid also worked without a problem. Playback of commercial recordings, from jazz through to nu metal, demonstrated that the quality of the analogue output hardware is very good. Recording both vocals and acoustic guitar with a mid-priced condenser mic via the front-panel XLR inputs also produced very pleasing results — crisp and clear highs and solid lows. For all but the most demanding of studio environments, the audio quality of the 828 MkII is very respectable indeed and, with all the usual due care and attention in other file:///H|/SOS%2004-07/MOTU%20828%20MkII.htm (3 of 7)9/22/2005 8:55:51 PM

MOTU 828 MkII

areas of the signal chain, the 828 is perfectly capable of producing commercial/ broadcast-quality recordings. I can only agree with Paul White's comments in main review that the MOTU 828 MkII offers some significant improvements over the original 828. While one would expect that the audio quality has moved forward in the three years that separate it from the original, perhaps more significant is the improved functionality — and in particular the additional controls available via the front panel. These really do make the unit much more flexible. While the Audiodesk software is not available to PC users, the useful Cuemix application is, and functions in the same fashion as on the Mac. Although I did not have the opportunity to test the GSIF drivers, the WDM and ASIO drivers seem very solid indeed, making the 828 MkII suitable for use with all the current crop of major PC audio applications. Essentially, whatever you want do to, the MOTU does its best not to get in the way, whether you're working on PC or Mac. As Paul commented about the original 828 back in his July 2001 review, this is a 'does exactly what it says on the tin' type of product. If the combination of analogue and digital connectivity fits your particular needs, I'd have no hesitation in recommending the 828 MkII, and if I could justify upgrading my own rack-based audio interface at present, the MOTU would most certainly be on my shortlist. John Walden

I/O Silver Lining? If you total up all the 828 MkII's I/O possibilities, you get a maximum of 20 inputs and 22 outputs at up to 48kHz, though the ADAT I/O channel count is halved when working at high sample rates. The original 828 had no MIDI I/O, whereas the MkII has one MIDI In and one MIDI Out, which communicate with the host computer via the Firewire cable, though on the Mac platform, MIDI is supported under OS X only. MOTU claim that the MIDI timing is as accurate as the host software allows. The two front-panel combi sockets accept XLR mic connectors, balanced or unbalanced jack line connectors and high-impedance instrument connectors (usually unbalanced). Each input has its own Trim control for level setting and a miniature toggle switch to turn on phantom power for use with capacitor mics or active DI boxes. Also on the front panel is the headphone jack, which by default shares the same level control as the main stereo outs. This is not always what you want, but you can also assign the headphone out its own signal source and level control if you need to. The SMPTE input is on a standard quarter-inch jack, and a similar jack provides a SMPTE output for when the 828 MkII is operating as master. A DSP-powered phase-locking system is used in combination with filtering to achieve fast lock times and sub-frame timing accuracy. One of the included pieces of software is called MOTU SMPTE Console, and provides the means to control the SMPTE functionality of the unit when, for example, sending SMPTE to stripe tapes. SMPTE sync works with any host applications, Mac or PC, that support the ASIO 2 sample-accurate sync protocol. Note that the 828 MkII has been designed to file:///H|/SOS%2004-07/MOTU%20828%20MkII.htm (4 of 7)9/22/2005 8:55:51 PM

MOTU 828 MkII

allow you to use multiple units together, but whether this is possible depends on the operating system of the host computer and/or the audio software you're working with. In theory, under Mac OS X, users can slave the MOTU driver to another Core Audio driver — either MOTU or third-party, which includes the Mac's built-in audio The Cuemix software is, in effect, a control panel for the 828 MkII's on-board DSP, — for accurate sync. However, certain which runs your mixer setups without audio programs, including the current imposing any load on the host CPU. versions of Logic and Cubase, don't as yet allow users to select multiple audio interfaces under Mac OS X, a situation that both Emagic and Steinberg need to sort out without further delay. The Cuemix DSP section of the control panel controls the internal DSP mixing facilities and also allows Global settings to be adjusted. If you're hooked up to a computer and you try to use the front panel to change something that you should be adjusting via a software control panel, the two-line, 16-character backlit LCD window prompts you. As a stand-alone mixer, the 828 MkII can sum all the inputs to a stereo pair and up to four simultaneous mixes can be run, each feeding a different output pair. This could be useful in setting up a couple of alternative monitor cue mixes, for example. Computer-free use is made possible by the six rotary encoders on the front panel where their values, levels or parameter names are shown in the display window. All six DSP section encoders and the level knob have integral push switches for confirming operations or toggling between modes. Using the DSP section, up to 16 mix presets can be stored and recalled. The 828 MkII's LCD can be switched between Setup and Mix modes by pressing the Setup knob, and in Setup mode you can choose whether the optical I/O is ADAT or S/PDIF, you can choose the digital sync mode, set the headphone assignment and so on. Back in mixer mode, the LCD shows the settings for the current mix, where the two lines of characters become two lines of tiny faders. When something is adjusted, the display temporarily 'zooms' in on this parameter while it is being adjusted and the time the display takes to revert to normal after adjustment can also be changed by the user. The Cursor knob scrolls through the 20 possible inputs while the Value knob adjusts their settings. Because the 828 MkII stores Cuemix settings when disconnected from the computer, you can save desired mixes, complete with default pan and gain settings, then make changes from the front panel if necessary when the interface is being used as a stand-alone mixer. At its simplest, this would allow you to rehearse via the 828 MkII in your studio without needing to boot up the computer. The ability to run four separate mixes simultaneously adds to the 828 MkII's flexibility. Mixes may be named and the settings include levels, pan positions, mute and solo settings. Mix 1 can be routed directly back into a computer input, which makes it easy to mix back to a spare stereo track of the software. This is a useful option where you may be using external MIDI synths that need to be


MOTU 828 MkII

added back into the final mix.

In Use I've always appreciated that MOTU hardware generally installs and runs without any problems, and though there was a little initial delay in supporting the Mac OS X platform (which was as much due to the changing nature of OS X as anything), the drivers MOTU have come up with are rock solid in my system. The current OS X driver is v1.09 (which also works with the old 828), and this is also happy with dual-processor G5 machines. The halving of the ADAT channel count at high sample rates is a function of the way the ADAT interface works, as it was never originally designed to handle anything higher than 48kHz — any other piece of gear with ADAT I/O has the same limitations so this it not a MOTU issue. The mic amps, while not in the 'silly money' quality bracket, are the equal of what you'd expect to find in just about any well designed small mixer, and being able to get at them on the front panel makes life a lot easier when the 828 MkII is mounted in a rack. It's also handy being able to plug a guitar directly into the interface, especially if you have a software modelling amp plug-in to process it with. The instrument inputs work well and are creditably quiet in use. I always liked the sound of the original 828 and the MkII's converters sound, if anything, fractionally better still, so there are no complaints in that department. Obviously you can get better results using audiophile converters that cost around the same as a family hatchback car, but these are pretty good. I have to admit that while my old 828 is still perfectly adequate for most of my needs, the 828 MkII's metering is a lot nicer, with LED signal-present monitoring of the analogue, mic and S/PDIF outputs, and either four or five-segment bargraph metering of the analogue ins, S/PDIF in, main stereo out and mic inputs. Also in the display section is an indication of the current clock rate, the digital lock status of the interface and SMPTE activity. And of course the old 828 didn't support high sample rates. Cuemix Console is a simple but visually stylish and functional piece of software that handles mixing within the 828 MkII, either in conjunction with, or independently of your host sequencer program. It can take care of latency-free direct monitoring, and as it mixes using the DSP chips inside the 828 MkII, it doesn't tax the host software. Although modern computers are now so fast that latency is seldom an issue, it's still nice to be able to set up zero-latency monitoring for the benefit of those performers who are very sensitive to even a few milliseconds of delay, and the ability to mix without a computer is great if you just need to plug in a mic and a guitar for rehearsals. Setting up a mix using small assignable controls isn't as quick as on a conventional mixer, but it's pretty straightforward. The only obvious compromise in the 828 MkII's design is that the optical S/PDIF I/ O uses the same connector as for ADAT-compatible devices, so you can't use file:///H|/SOS%2004-07/MOTU%20828%20MkII.htm (6 of 7)9/22/2005 8:55:51 PM

MOTU 828 MkII

both at the same time. Most studio gear seems to have co-axial S/PDIF connectors, in which case this won't be a problem, but if you do have S/PDIF equipment with optical I/O and need to use it at the same time as the ADAT I/O, then you'll need to buy an optical-to-co-axial adaptor to work around this limitation. Though I can't see anybody buying an 828 MkII who doesn't already have software to run with it, the bundled Audiodesk is actually pretty sophisticated if you only need audio recording and mixing capabilities, and because it runs without a physical dongle, it is ideal for live laptop recording. It is based on the audio recording and editing side of MOTU's flagship Digital Performer package, so it's not a toy by any means.

A Small Step For An Interface... The 828 MkII is a definite step up from the original 828, adding to the original feature set considerably while trimming the price. The front-panel mic amp inputs, improved metering and high sample-rate support are all welcome, and while not everyone will need the DSP mixing facilities, it certainly doesn't hurt to have them. The word clock input will also be welcomed by those users who were frustrated by the lack of it on the original 828, there's now full SMPTE support and of course the I/O count is slightly higher. If you need a bomb-proof audio interface that will work with almost anything, then the 828 MkII will not disappoint, and if you need more analogue I/O, there are now several choices of third-party eight-channel 'mic amp to ADAT' boxes from which to choose. As a satisfied 828 user myself, I'm now looking for an excuse to upgrade! Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Rode K2

In this article:

From NTK To K2 Recording Performance Peak Of Achievement?

Rode K2 £570

Rode K2 Valve Microphone Published in SOS July 2004 Print article : Close window

Reviews : Microphone

pros Warm, smooth sound. Affordable. Nicely engineered and styled. Uses a readily available tube.

cons

Rode have built on the technology from their respected NTK to develop an affordable new multipattern valve mic.

No low-frequency cut switch.

summary The Rode K2 is a very nice multi-pattern tube microphone, with a warm, classic sound. Although there are cheaper mics coming in from various Far Eastern locations, it offers a very good balance of price and quality control.

information £569.99 including VAT. HHB Communications +44 (0)20 8962 5000. +44 (0)20 8962 5050. Click here to email

Paul White Photos: Mark Ewing

It's almost two years since I reviewed the Rode NTK valve mic, and it turned in such an impressive performance that both SOS's Debbie Poyser and myself ended up adding one to our respective mic collections. Valve mics inevitably cost more than their solid-state counterparts, but they are far more affordable today than ever before and they remain popular because of their flattering musicality. At first, fixed-pattern tube mics such as the NTK were about as much as the home user could be expected to afford, but as Rode have developed more sophisticated means of production in their native Australia (the new K2 is entirely designed and manufactured in Sydney), they are now able to offer the multipattern K2 at a very attractive price.

www.hhb.co.uk www.rode.com.au

From NTK To K2 Physically, the mic is almost exactly the same size as the NTK, but has a different grille design with a heftier support frame. It features a large (one-inch diameter) dual-diaphragm capsule with gold-sputtered, edge-terminated mylar diaphragms. As with the NTK, the onboard circuitry is based around a selected 6922 dual-triode tube operating in a Class-A configuration, working in conjunction with bipolar output transistors so that no output transformer is required. The tube is held in place in its porcelain socket by means of a plastic spring clip, and the standard of mechanical and electrical engineering is up to Rode's usual very high standard.

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All the electronic components are mounted on a glass-fibre circuit board, while the capsule itself is shockmounted. Such wiring as there is is very neatly executed. The double-layer, stainless-steel grille mesh provides the necessary screening from radio frequencies, while also protecting the fragile capsule from physical damage, and the all-metal housing is beautifully machined and finished with a matte nickel plating. Access to the tube compartment is achieved by unscrewing the main body sleeve, and a knurled locking ring is used to secure the included SM2 shockmount. The mic comes in a tough moulded RC2 plastic travel case of the type normally associated with power tools, and includes not only the shockmount and power supply, but also the necessary multi-pin XLR cable to connect it to the power supply. The mic itself has no switches or other controls and the mic pattern is adjusted using a continuous rotary control on the power supply. This provides everything from omni, through various widths of cardioid, to figure of eight. A goldplated stud indicates the 'hot' side of the capsule, in Rode's usual tradition.

The K2's internal valve can be replaced without any soldering.

There are no pad or filter switches on the PSU, but this shouldn't be a problem, as most mic preamps and mixers have the necessary low-cut filters and, with a maximum SPL handling of an incredible 162dB, you'd probably have to shoot this mic to get it to distort! I always like to remind potential purchasers that the paper spec tells you precious little about the sound of a mic, though it can help to establish how quiet or sensitive it is. The quoted sensitivity figure is -3dB (reference 1V/Pa) and the equivalent noise figure is just 10dBSPL (which means there's a signal-to-noise ratio of 81dB) — rather better than some solid-state mics, and a little quieter than the NTK. The frequency range is 20Hz-20kHz, with little deviation other than a very deliberate, but still suitably subtle, presence peak centred at around 12kHz in omni mode. Another lesser peak at around 5kHz is evident in cardioid mode, which is very reminiscent of the NTK's response shape. This is about the only part of the spec that might give a clue as to how the mic might sound, as this type of presence peak often results in an open and airy high end.

Recording Performance If anything, the Rode K2 is a hint sweeter-sounding than the NTK, and it definitely does the 'subtle flattery' thing — which is, after all, why we tolerate tubes inside our microphones in the first place. The way in which it flatters is hard to describe, but, in addition to adding weight to what I call the 'chest' frequencies


Rode K2

of the voice, it also captures the high-end detail in a way that sounds noticeably smoother and less aggressive than is often the case with solid-state mics. As with the NTK, the lack of a bass roll-off switch means you'll probably need to engage the low-frequency filter on your preamp or mixer when recording vocals, and you'll certainly need a pop filter. In fact, pop filters are such essential components of any vocal recording system that I'm surprised more mics don't come with them — they're just as important as shockmounts, yet we seem to get those thrown in with nearly every mic we buy these days. As a vocal mic, the K2 sounds very classy indeed, and it has been deliberately engineered to be reminiscent of classic studio mics. It is, however, rather quieter than some of these tube classics, and probably a lot cheaper to service given that the tubes used in the Rode mics don't belong to an endangered species. This is a great mic if your voice needs a bit of filling out, or if the high end of your voice needs rounding off. I also tried the mic on the usual acoustic guitar and hand percussion, where it turned in a solid performance, combining warmth with evenness of tone. I particularly liked this mic for strummed acoustic guitar played in a pop style, as you get a punchy, no-nonsense result that sounds almost as though it's been slightly compressed, with no ragged edges.

Peak Of Achievement? Rode's K2 is a great performer, offering multi-pattern functionality at a very affordable UK price. Rode have risen well above the bargain-basement dog fight that seems to be going on at the moment, so the K2 isn't the cheapest tube mic on the market by a long stretch, but it offers quality, both mechanically and sonically. Anyone in the market for a large-diaphragm multi-pattern mic should audition the K2, if only to find out what the others have to live up to. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Sibelius 3

In this article:

Sibelius 3

Exposition Development I: The Kontakt Mac/PC Scorewriter Published in SOS July 2004 Player Kontakt Kontent Print article : Close window Development II: Flexi-time Reviews : Software First Kontakt Development III: Live Playback Plug It In, Plug It Out Recapitulation Over the last 10 years, Sibelius Software have built a No Manual Labour Required reputation for providing what most musicians Coda consider to be the leading score-writing software. Sibelius Reviews In SOS Now, in version 3, they have teamed up with Native Cadential Extension

Sibelius 3 £595 pros The inclusion of an integrated sampler provides better quality playback for those users without extra hardware or software, and new features such as 'Save As Audio CD'. Newly enhanced Flexi-time and playback features gives the user more control over the MIDI output from Sibelius.

Instruments to provide enhanced playback facilities — but does this upgrade live up to the high standards set by previous releases? Mark Wherry

Nearly 10 years ago now, I fondly remember inserting a blue 3.5-inch floppy disk with a printed Avery label into an Acorn RISC-powered computer to run a demo version of Sibelius 7, a brand new score-writing application Sibelius still offers the that promised to revolutionise music quickest and easiest-to-use notation technology. It blew my mind. tool set for creating and Compared to the Mac, Atari and DOS editing notation, which has or Windows applications of the day, been fine-tuned for version 3 with better support for guitar Sibelius 7 was simply the fastest scorenotation, Optical spacing and writer around — the original advertising Sibelius 3 in all its glory. And yes, the notes much more. literature of the time claimed you could are meant to be funny... cons reformat the entire score for Wagner's Nothing worth mentioning as Ring cycle in just 0.1 seconds! The fact that Sibelius 7 was hand-coded in a reason not to buy Sibelius if assembler probably didn't hurt matters, but it wasn't just the speed that made you need an application to Sibelius different: its user interface was remarkably intuitive and uncluttered, and work with notation on a you really felt like the program was there to make your life easier when preparing computer. scores. summary In my opinion, Sibelius is simply the best notation software available, and the new playback features are a

The original choice to develop Sibelius for Acorn's RISC OS was perhaps inevitable: not only were the ARM processors great chips (they still are), but Acorn computers were used almost exclusively in every school across the UK at

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logical evolution to close the gap between what you hear in your mind, what appears on the page, and what the computer plays back.

information £595; upgrade from version 1 £175.08; upgrade from version 2 £151.58; competitive crossgrade from Finale, Encore or Mosaic £386.75; Kontakt Player Gold £175.08. Prices include VAT. Sibelius Software +44 (0) 20 7561 7999. +44 (0)20 7561 7888. Click here to email www.sibelius.com

the time — a big market for Sibelius. And since the developers (originally just two brothers, twins Ben and Jonathan Finn) were from Cambridge, where Acorn were based, it was the computing platform they had grown up with. However, technical brilliance couldn't save Acorn from having to get out of the desktop computer business as Windows-based computers began to dominate the education market after the release of Windows 95, so it's perhaps a testament to Sibelius that the application remains one of the most successful originally Acorn-based applications to live on past its original platform. A decade on from my first Sibelius encounter, the number '7' has long since been dropped from the name, and two major releases for the Mac and Windows platforms later, Sibelius Software have released version 3 of the company's now established and universally revered scoring software. Sibelius is one of those applications that always seems to do everything you require, but at the same time, when a new version is released you always end up with a whole host of functions you wonder how you ever lived without. So as a long-time Sibelius user (and fan), I couldn't wait to get my hands on the latest version — could they make the perfect score-writer better for a third time?

Test Spec Sibelius version 3.1.1. IBM T40 laptop with 1.3GHz Pentium-M, 768MB RAM, ATI Mobility Radeon 7500 graphics, with Echo Indigo and RME Digiface audio interfaces, running Windows XP Professional. AMD dual Opteron 246 system with 4GB RAM and an Nvidia Quadro4 380 XGL, running Windows XP Professional.

Exposition Installing Sibelius is a straightforward affair, and the process is clearly described in the helpful manual (see 'No Manual Labour Required' box for more information). During the installation process you'll be prompted for your serial number, and this number is used as part of the copy-protection mechanism when you register the application, a process which is pretty much the same as in previous versions of Sibelius. An unregistered installation will lose the ability to save after five days, but the simplest thing to do is simply register via an Internet connection the first time you run the application. Alternatively, you can also register by fax or phone and, again, the manual helpfully describes what you need to do in order to complete the registration process. One feature I liked in version 2 of Sibelius was the ability for the user to transfer registrations between computers, which is great when you want to authorise your laptop from your desktop and vice versa, and leaves the user somewhat in control of the copy protection. This was absent in the first release of version 3, but later put back in version 3.1. For one reason or another I have ended up reinstalling my computers more than once in the last few months, but I have never had a problem with Sibelius refusing to be registered, even after several attempts.


That was then: Sibelius 7 running on an old Acorn Risc OS computer. Not so different to the Sibelius we have these days, really.

Sibelius 3

Sibelius Software's registration server remembered my details and warned me that I wasn't licensed to run Sibelius on multiple computers simultaneously, which is fair enough. Sibelius 3's baseline system requirements are pretty low, starting with a just a Pentium or G3 processor. Windows 98/ME/NT4 users should have at least 64MB RAM (Windows 95 isn't supported), while 2000 or XP users need at least 128MB, and Mac users running Mac OS 9.1 or 10.1.5 (or later) should have at least 40MB free RAM to run the application. While Sibelius Software recommend having more memory if you're going to be scanning music with the bundled Photoscore Lite software (and a TWAIN-compatible scanner on the Mac), it should be pointed out these requirements are just to run the basic score-editing features, and not the more advanced playback features discussed in the next section. See the 'First Kontakt' box for more detailed information concerning system requirements. I have to confess that I didn't spend too much time running Sibelius 3 on a Mac, although it's worth mentioning that Sibelius Software claim the Mac OS X version is now 50 percent faster when compared with Sibelius 2. It definitely seemed snappier on a friend's 17-inch Powerbook, and whether or not you'll notice a 50 percent improvement, Mac users will certainly benefit from the new optimisations. The first change you'll notice when you run Sibelius 3 is the new Quick Start window, which by default is displayed every time you start the program unless you specify otherwise, and incorporates the old 'tip of the day' window. The Quick Start provides easy access to the most common tasks you would require when launching Sibelius: opening a recent score from a pop-up menu, opening another score from a file selector, opening a MIDI file for conversion, starting a new score from scratch, or scanning a piece of printed sheet music. It's a small but welcome change — one of these five options would be the reason you launched Sibelius in the first place, so it makes sense to be presented with them from the outset.

Development I: The Kontakt Player The biggest set of new features in Sibelius 3 concern its playback abilities, and in collaboration with Native Instruments, Sibelius now includes as standard a builtin version of Kontakt, Kontakt Player Silver, which can play eight different instruments simultaneously from a palette of 19 instrumental sounds, along with a full complement of 114 drum and percussion sounds. The quality of these sounds is a definite improvement over using most computers' built-in sounds, such as the Windows GS software synth, and is perfectly adequate to get a good representation of what you're writing. For those who have slightly more demanding playback requirements, Sibelius Software and Native Instruments also offer Kontakt Player Gold, which provides 63 instrumental sounds (including the 19 from the Silver version) and offers the ability to play back 32 of these instruments simultaneously — see the 'Kontakt file:///H|/SOS%2004-07/Sibelius%203.htm (3 of 13)9/22/2005 8:56:03 PM

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Kontent' box for more details. Kontakt Player Gold is available from Sibelius Software or Native Instruments for £175.08. While Sibelius has always been pretty good at using external MIDI devices, the real benefit of Kontakt Player is the seamless integration. Although you can mix and match your playback from any number of external devices, in conjunction with Kontakt Player if you like, using Kontakt Player alone means that the score always plays back pretty much as intended automatically. This might sound a little obvious, but using external MIDI devices requires a pre-prepared profile to be The new Quick Start window gives present for that device so that Sibelius knows immediate access to any task you how to pick the appropriate program changes would need to carry out when you launch Sibelius. to perform the score correctly. For example, one particularly neat feature in Sibelius is that it can interpret technique instructions so that violin parts switch between arco and pizzicato playing styles automatically. If your MIDI device wasn't supported by Sibelius, you'd have to set this up yourself, whereas, as I mentioned earlier, Kontakt Player just performs these technique changes automatically straight out of the box. The real killer feature that the Kontakt Player facilitates, however, is 'Save As Audio Track', which provides a simple way of answering what was previously a complicated question: namely, how to I get my Sibelius composition onto an audio CD? In version 3, you simply click the CD icon on the toolbar (making sure that nothing is selected so the export starts from the beginning of the score), choose an appropriate file name, select a location, click OK, and the recording is made to a file in real time. This feature alone is going to be a really big deal in the education market, making it very easy for students to produce audio CDs of their compositions. The only aspect of Kontakt Player I found mildly annoying is that the sounds required to play the score only load when you reset the sounds in the Mixer window, click on a note, or press Play for the first time — MIDI Thru, for example, won't work until you've done one of the three moves. While it only takes a couple of seconds to load each required sound, I'd rather have the sounds load when I first set up a new score or add a new stave, instead of having to wait the first time I expect to hear something. It's also possibly a shame that the Kompakt Player isn't more open-ended, meaning that you can't put your own sounds into the player — imagine importing sounds from the Vienna Symphonic Library, for example. But this would add a huge layer of complication to the player, which is probably why it was deemed (quite correctly, in my opinion) unnecessary. In terms of playing back more complicated orchestrations, using a separate sampling application like file:///H|/SOS%2004-07/Sibelius%203.htm (4 of 13)9/22/2005 8:56:03 PM

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Gigastudio with, say, the VSL is still a good option (the subject of a future technique article), and this is probably still my preferred playback method. You could also use East West's Symphonic Orchestra via a suitable plug-in host (such as V-Stack) and a MIDI loopback utility (to get the MIDI data from Sibelius to V-Stack if both applications are running on the same computer), Garritan Personal Orchestra, or indeed any software sampler and library. The only catch is that when you're not using the built-in Kontakt Player, you lose the ability to create an audio track so conveniently with the sounds you're using — although there would be nothing to prevent you switching back to Kontakt Player for saving an audio track.

Kontakt Kontent The Kontakt Player sound set comprises contributions from many sound developers, such as Sonic Implants (SI), Peter Siedlaczek (PS), Zero-G (ZG), Dan Dean (DD), Tabspace (T), Narrators (N) and Northstar (NS). Below is a list of the content supplied with the Gold and Silver versions of Kontakt Player. Strings Gold and Silver: String Section (SI), String Section Pizzicato (SI). Gold only: Violin Section (SI), Solo Violin (SI), Solo Viola (NS), Cello Section (SI), Solo Cello (SI), Contrabass Section (SI), Solo Contrabass (SI), Solo String Pizzicato (SI), String Section Tremolo (SI), String Section Col Legno (SI), String Section Sur Ponticello (SI).

Brass Gold and Silver: Solo French Horn (PS), Solo Trumpet (PS), Trumpet Section (SI), Trombone Section (PS). Gold only: French Horn Section (SI), Solo Trumpet with Straight Mute (SI), Solo Trombone (N), Solo Flugelhorn (ZG), Solo Euphonium (DD), Solo Tuba (SI).

Woodwinds Gold and Silver: Solo Flute (N), Solo Oboe (PS), Solo Clarinet (SI), Solo Bassoon (SI), Solo Tenor Saxophone (SI). Gold only: Solo Piccolo (SI), Solo Recorder (NS), Solo English Horn (SI), Solo Bass Clarinet (PS), Solo Alto Saxophone (NS), Solo Baritone Saxophone (N).

Pitched Percussion Gold and Silver: Timpani (SI), Grand Piano (PS), Glockenspiel with Hard Sticks (SI), Orchestral Harp (NS). Gold only: Harpsichord (SI), Celesta (SI), Vibraphone with Medium Sticks (SI), Marimba with Medium Sticks (SI), Xylophone with Medium Sticks (SI), Tubular Bells (NS), Handbells (NS).

Other Gold and Silver: Solo Male Voice 'Ah' (SI), Choir Ensemble 'Ooh' (SI), Church Organ (N), Nylon-string Guitar (ZG). Gold only: Solo Female Voice 'Ah' (NS), Male Choir Ensemble 'Ah' (SI), Female Choir Ensemble 'Ah' (SI), Rhodes Piano (SI), Drawbar Organ (SI), Reed Organ (SI), Steel-string Guitar (SI), Clean Electric Guitar (SI), Overdriven Electric Guitar (SI), Fingered Electric


Sibelius 3

Bass Guitar (SI), Picked Electric Bass Guitar (SI), Slap Electric Bass Guitar (SI).

Drums & Percussion The 114 drum and percussion sounds (too tedious to list!) are supplied in both Gold and Silver versions and are an extension of the normal General MIDI set. In other words, if you've ever played around with a General MIDI module on channel 10, you'll get these sounds and quite a bit more, covering drum kits, bass drums, snares, cymbals, wood blocks, bongos, triangle, and so much more, including the all-important vibraslap and lion's roar. And we all know how difficult it is to find a good session lion. "Once more with feline..."

Development II: Flexi-time Sibelius have also improved Flexi-time, the recording feature introduced way back in the Acorn days that allows Sibelius to follow your tempo when entering notes in real time, as opposed to the other way around. Flexi-time is designed to compensate for controlled tempo fluctuation in a performance, such as a the ritardandos or accelerandos a player may naturally perform while recording notes into Sibelius, rather than following an erratic or badly timed performance, and it has always achieved the former rather well. A welcome new feature in version 3 is the ability for Flexi-time to interpret two different polyphonic voices for each stave in the notes you're recording, which basically means that overlapping notes are automatically assigned to different voices. When you do mean for a passage to be notated with different voices, this feature will save you a great deal of time — it might not be 100 percent accurate every time with your intentions, but it at least always gives you a good starting point — and can be disabled if you don't mean for a stave to contain polyphonic voices.

In collaboration with Native Instruments, Sibelius 3 features an embedded version of Kontakt Player to provide integrated sample playback and the ability to create an audio track from a Sibelius score with just a couple of mouse clicks. Above you can see the optional Kontakt Player Gold, along with the standard Sibelius mixer.

The 'two voices per stave' ability means that if you're recording a piano part with both hands, you do indeed get four voices in total while you're playing — effectively, two voices for each hand — which will be great for preparing SATB-style choral parts. And, as an aside, the ability for Sibelius to interpret two voices per stave also applies to MIDI file imports as well, which will surely be a feature welcomed by those sequencer users who later prepare their scores with Sibelius. Due to the timing and scheduling differences that can exist in dual-processor or hyperthreaded systems, Windows users working with Flexi-time can occasionally


Sibelius 3

have problems with such configurations which result in the recording becoming progressively out of time. For these situations, an 'Enable MIDI Time Stamps' toggle has been added to the Flexi-time Options window, which presumably uses an alternative method for time-stamping MIDI events — perhaps Windows' own time-stamps via Direct Music rather than those calculated by Sibelius. However, there didn't seem to be an issue working with Flexi-time when running Sibelius on my dual Opteron machine. And Mac users needn't worry, as Core MIDI's method of handling time-stamps in OS X is able to take care of this on dualprocessor Power Macs. On the subject of timing, Sibelius have been thoughtful enough to include a warning (which can be disabled) if you try using Flexi-time with the Kontakt player, pointing out that Flexi-time may not work very well if the latency of the sound output is causing the player to perform slightly late. Another improvement to Flexi-time in version 3 is the inclusion of replace and overdub modes, allowing you to specify whether existing music gets replaced or merged when you record over a passage already containing notes.

First Kontakt Like all Native Instruments products, Kontakt Player supports ASIO, Direct Sound and Windows Multimedia-compatible audio interfaces; but the ASIO support is definitely the best option for those who want low-latency performance, especially important if you're recording notes in real time. Aside from suitable audio hardware, to make use of Kontakt Player Silver it's recommended that you have at least a Pentium III or 500MHz G3 processor and 128MB RAM (or 64MB for OS 9, or 196MB for Windows 2000/XP) to play back a couple of sounds. For scores that use eight sounds, the ideal requirements list a 1GHz processor and preferably a G4 or G5 for Mac users, along with 256MB RAM. For Kontakt Player Gold to reach its full potential, Sibelius Software advise a 2GHz Pentium 4, Athlon XP or G4/G5 processors, between 512MB and 768MB of RAM, and a 7200rpm hard drive to take advantage of Kontakt's DFD (Direct From Disk) streaming. In use, though, I found I could play back fairly big scores on my IBM Thinkpad (as detailed in the Test Specs box), even with the Gold version, and Kontakt Player is certainly a great tool for obsessive laptop users like myself, helping to make the perfectly integrated and mobile Sibelius setup even better.

Development III: Live Playback Sibelius has always been pretty good at making an attempt at interpreting a performance of the score during playback, with features such as Espressivo and the ability to perform dynamics and other articulations contributing to make the music sound a little less mechanical. However, version 3 introduces a special Live Playback mode, enabling Sibelius to remember the timing and velocity information for notes recorded using Flexi-time. This means that although Sibelius will still clean up the score so that it looks correct, when Live Playback file:///H|/SOS%2004-07/Sibelius%203.htm (7 of 13)9/22/2005 8:56:03 PM

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mode is enabled the application will play back the notes exactly as you played them. If you want to fine-tune the velocities of notes for Live Playback mode, there's a handy View Live Playback Velocities mode that displays a note's velocity as a small bar-graph beside that note on the score. This could potentially be quite messy, but it's been implemented in a very clear way; and where there's more than one note on a given beat, Sibelius shades in the velocity bar for the other notes so you can see there are indeed multiple velocities used for the notes in a chord. Where polyphonic voices are used, the standard voice colours are used for the velocity bar (blue for voice one, green for voice two, and so on) so it's very easy to see what you're editing.

With View Live Playback Velocities enabled, velocity bars are drawn besides the notes, allowing you to see and edit the velocities that were played during Flexi-time input. Note also the new coffee-stained paper texture...

The Live Playback note velocities can be edited by simply dragging or clicking the velocity bar, and the only slight confusion occurs when you want to edit the velocity of a note where there are other notes on the same beat. In this case, editing the velocity bar sets all notes on that beat to the new velocity, although if you first select the note whose velocity you want to change, only that note's velocity will be altered when you edit the velocity bar. It's also worth noting that the velocity bars only show up for notes that have been entered using Flexi-time input, although it's perfectly possible to add velocity attributes to notes entered with the mouse using the Transform Live Playback feature. To conclude the new playback-related feature set, Sibelius Software have also dramatically enhanced the application's ability to play repeats, and Sibelius now fully interprets text instructions such as Da Capo al Fine, and supports even the most complex structure with signs, codas and segnos. Taking repeat playback even one step further, new properties in the Playback section of the Properties window allow you to specify on which repeat passes symbols, such as dynamic markings like 'p' or 'f' (or indeed any other objects — even notes) should be used. The improved handling of repeats in Sibelius is particularly useful when you're creating an audio file of your work using the 'Save As Audio Track' feature. And it's doubly useful when exporting MIDI files (oh yes, MIDI files now contain the repeats fully written out), which are also now exported with any Live Playback timing and velocity information if this feature is active. A really nice touch worth noting for those who work with film is that Sibelius compensates for the repeats in the timecode in displays; and if you opt to have timecode displayed at every bar, the application will stack the timecode positions for each repeat pass above the stave. Mind you, it's still a little bit frustrating that file:///H|/SOS%2004-07/Sibelius%203.htm (8 of 13)9/22/2005 8:56:03 PM

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Sibelius can't send out MIDI Timecode to external devices, since this would be really useful for slaving video if you're using Sibelius as your main composition tool, meaning that you wouldn't have to export a MIDI file into a sequencer as the manual suggests.

Plug It In, Plug It Out One of the most intriguing features in post-Acorn versions of Sibelius has been the inclusion of plug-ins. While most plug-ins used by software companies are precompiled components, Sibelius plug-ins can be edited and created using the interpreted ManuScript language with the Plug-in Editor window, which are both built right into Sibelius. ManuScript is based on the Simkin (www.simkin.co.uk) embedded scripting language created by Simon Whiteside and extended with the help of Graham Westlake, and it really gives the most advanced user complete control over Sibelius' functionality for manipulating and analysing all the elements on a score. Sibelius Software have been adding more plug-ins with each new version, and Sibelius 3 now includes nearly 80 plug-ins as standard, which provide some of the more advanced tools within the application. Highlights in the new version include plug-ins to realise parts from figured bass or chord symbols, remove unison notes, add accidentals to notes and export lyrics to a text file. There's also a plug-in for creating just about any scale or arpeggio you can imagine, from a straightforward major scale to a Japanese pentatonic and beyond. The latest implementation of ManuScript can also access all the new Sibelius 3 features, include the Live Playback parameters (such as velocity) and the object colour information. As mentioned earlier, it's perfectly possible to create your own Sibelius plug-ins using the built-in editor, and a separate 58-page PDF file detailing the ManuScript language gets installed into your Sibelius application's Extras folder. For anyone with experience in high-level programming languages such as C or C++, ManuScript is fairly easy to get along with, although it's not a feature designed for beginners and, as such, this is the only area of Sibelius where the company doesn't provide full support to the user. There is however a mailing list you can join at www.sibelius.com/ helpcenter/resources/plugins.shtml if you're interesting in developing plug-ins, and this page also contains many extra plug-ins for download. A third-party site that also has links to some additional plug-ins and Sibelius resources can be found at www.musicprep.com/sibelius.

Recapitulation In addition to the headline-grabbing new playback-related features, there are also plenty of editing improvements that even those Sibelius users who aren't so concerned with its inbuilt performance abilities will appreciate. For example, perhaps the biggest improvement to Sibelius' note-engraving algorithms in this new version is the development of a new note-spacing technique referred to as Optical spacing — and yes, that's optical with a capital 'O', since it's also a Sibelius trademark.


Sibelius 3

The new Optical spacing algorithm was apparently the result of assistance provided by 'several of the world's leading music engravers', and is designed to make sure that the spacing between notes takes into account obstacles that might make the score look cluttered, such as accidentals. This is especially The new Optical spacing algorithm is a big noticeable where you might have improvement when you have objects such as multiple accidentals preceding a chord, tuplets and accidentals. although Sibelius is intelligent enough in more complex passages to not introduce more space than is required to make the elements look tidy and prevent collisions. The algorithm will also re-space elements where space was added if it's no longer necessary after certain elements have been deleted, and in addition to avoiding obstacles in the score, Optical spacing improves the way notes of different time divisions (such as tuplets) line up against each other, even on different staves. While Sibelius has always been great at producing beautifully laid out scores with a degree of intelligence, Optical spacing definitely offers dramatic improvements in fairly dense scores and clears up many of the problems identified by Mike Senior in his review of Sibelius 2 back in March 2002, especially when creating piano reductions of full scores with Sibelius' arrangement features. However, on occasion I still found it necessary to force a few updates to the spacing using the 'Layout / Reset Note Spacing' command, although Sibelius got it right most of the time. Sibelius Software have also revised the rules used for beaming notes together into groups, and they are now based not only on the time signature, but also on the context in which the notes are beamed. And if you want to alter the beaming on note groups yourself, there's a new, more advanced Reset Beam Groups window. A neat new feature that makes working with larger-scale arrangements more manageable is Focus on Staves. Clicking this new button on the toolbar redraws the score to display on the selected staves, which makes it easier to focus (literally) on one or more parts without the screen being cluttered by parts you don't need to see, such as those obscure triangle tremolos that only occur towards the end of the piece... This is especially useful if you're working on a smaller screen, and it also doubles as a quick way to print a single part from a score without needing to use the Extract Parts command. But a nice touch is that the full score still plays when you're using Focus on Stave mode, which means you can still hear the parts you're focused on in context, and you could use this feature (especially in conjunction with Scorch, the web browser plug-in that displays Sibelius scores in web pages) to just show the melody, while the score plays the full arrangement.


Sibelius 3

No Manual Labour Required It's a widely acknowledged fact that most manuals supplied with computer software are pretty unhelpful, and as a writer, there's nothing I find more frustrating than a poorly written manual. Sibelius is one of the best exceptions to rules I've ever seen: the manual is impeccably well-written, and after a brief Start Here section covering the installation and registration processes, there's a Quick Tour chapter to get new users acquainted with a minimum of effort. After the Quick Tour, there's a short How To section, pointing out the features of Sibelius you'll need to use in order to present certain styles of notation, such as a early music or for keyboard instruments, or how to achieve common tasks, such as extracting parts. The rest (which is the majority) of manual is a reference section organised alphabetically by the musical tasks you actually want to achieve. Bliss! No wonder that previous versions of Sibelius, back in the Acorn days, actually won awards for their good English! The manual is supplied as a perfect-bound and clearly laid out 592-page book, and rather than being an outdated tome covering version 1 with a handful of pamphlets to describe the latest version, has been completely revised for version 3.

Coda Another nice touch is the shadowed notes that appear when you hover the mouse over a stave; in retrospect, adding notes to the stave was often a bit like playing 'pin the tail on the donkey' in earlier versions, especially when leger lines were involved. Not any more: the shadowed notes (which appear both above and below the stave with the appropriate leger lines) enable you to see exactly what note will be added when you click the mouse button. As minor as this feature is, it's worth the cost of the update alone the first time you add notes to a stave in Sibelius 3! Between Sibelius 2 and 3, Sibelius Software also introduced G7, a version of Sibelius aimed specifically at guitarists that added many features aimed specifically for those producing guitar notation. As you might expect, Sibelius 3 is compatible with G7 and benefits from most of the features created for this other application, such as improved tab input using the QWERTY keyboard and the ability to load and save to and from ASCII tab files, with which Internet-savvy guitarists will be familiar. You won't get the graphical fretboard from G7 (which is either a good thing or a bad thing,


Sibelius' ManuScript plug-in development windows showing the guts of the Scales and Arpeggio plug-in — a result of using this 'new for version 3' tool can be vaguely seen in the background.

Sibelius 3

depending how you look at it!) but Sibelius Software have included over 50 new fretted instrument and tuning profiles, which means that writing parts for lutes, dulcimers and sitars is now much easier than before. Phew! The Photoscore Lite scanning software included with Sibelius also adds the ability to scan in tab and guitar frames in version 3, so guitarists are now catered for much better than in previous releases. There are many other numerous minor improvements that show the true attention to detail Sibelius Software have: for instance, you can now change the font for all text styles in one pass thanks to the new Edit All Fonts window, and you can now change the colour of pretty much everything in the score, which will be reflected on both the screen and the output from your printer. While the value of colour is debatable for some engravers, there's no doubt it can be useful to highlight certain elements for editorial decisions, or when producing scores and worksheets for education. Sibelius Software have also thoughtfully added the ability for Sibelius 3 to save out files in Sibelius 2 format (which is really going to help compatibility in organisations where not every copy will be upgraded at the same time), and Sibelius can now also export your score as a TIFF file. Finally, the Keypad returns to its more traditional look, with the various other property panels introduced in Sibelius 2 moved to a separate Properties window and, as with Sibelius 1 and G7, buttons to select and indicate the currently selected voice for a stave put back into the Keypad window.

Sibelius Reviews In SOS Sibelius 2, March 2002: www.soundonsound.com/sos/mar02/articles/sibelius2.asp. Sibelius 1, May 1999: www.soundonsound.com/sos/may99/articles/sibelius.htm. G7, July 2003: www.soundonsound.com/sos/jul03/articles/sibeliusg7.asp

Cadential Extension You might notice from that this review is written from the perspective of someone who really likes Sibelius, and I can't deny that this is absolutely true. It's simply the finest notation tool I've ever used on any computer, and a rare example where the developer's attention to detail really shines through to reveal a wellpolished product. I experienced a few Kontakt Player-related crashes and restarts when I was first started using 3.0, but the application has become increasingly stable again with updates to Kontakt Player, coupled with the Sibelius 3.1 and 3.11 updates. I could carry on gushing about how intuitive Sibelius is, how easy the application is to learn and how quick it is to work with, how amazing all of the Internet-related features (covered in the review of Sibelius 2) are, and so on, but instead I'll offer file:///H|/SOS%2004-07/Sibelius%203.htm (12 of 13)9/22/2005 8:56:03 PM

Sibelius 3

one final anecdote. Just I was finishing this review, I was talking to a friend who had been working on the score for Shrek 2 and had just returned from the orchestral scoring sessions in London. We ended up talking about Sibelius 3, being the musical geeks that we are, and he couldn't praise it enough. They had just got the new version before the scoring dates and he claimed it had literally saved certain parts of anatomy during the late nights between sessions, preparing parts and making amendments for the next day. It goes without saying that Sibelius' abilities as a score-writer are second to none, and while many people use it alongside a sequencer, exporting and importing MIDI files to prepare a final result, its merit as a composition tool is also significant. For this reason alone, it's almost impossible for anyone studying music these days to avoid Sibelius, especially in the UK, and the application's widespread use in professional circles, coupled with so many high-profile endorsements, mean that it's almost certainly going to be equally popular on whatever computer we're all running 10 years from now. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Soundcraft Compact 10

In this article:

What Does The Computer User Need From A Mixer? Soundcraft Compact Series Channel Features Master Section In Practice Marks Out Of 10

Soundcraft Compact 10 £212 pros Easy to operate. Compact without being cramped. Quiet mic preamps.

cons

Soundcraft Compact 10 Desktop Mixer Published in SOS July 2004 Print article : Close window

Reviews : Mixer

This slim new mixer has been designed from the ground up to cater specifically for computer studio users. But have Soundcraft got the balance of facilities right? Paul White

Mic preamp gain control law is very bunched up at the top end of the pot's travel. No built-in talkback facility.

Now that computer workstations have evolved such powerful and comprehensive mixing facilities, there's summary rarely any need for large analogue Despite a couple of minor consoles unless you prefer the sound or omissions, the Soundcraft the tactile experience of mixing through Compact 10 deserves credit for trying to meet the needs of them. Indeed, most systems can the typical computer studio operate quite happily with a small user at a sensible UK price general-purpose mixer, though that and without being overoften involves compromise and you still complicated. may be paying for features you don't information need. What Soundcraft have done with Compact 10, £211.50; their new Compact range is develop a Compact 4, £105.75. Prices small desktop mixer specifically for use include VAT. with computer-based systems, leaving Photos: Mike Cameron Soundcraft +44 (0)1707 off all the frills that aren't needed and 665000. adding some dedicated features that actively benefit computer users. +44 (0)1707 660482. Click here to email www.soundcraft.com

What Does The Computer User Need From A Mixer? So, what does a typical computer user need from a mixer? Firstly, you need a small number of mic preamps which can be routed into the input of an audio interface or soundcard. Most users with smaller systems rarely record more than one or two tracks at a time, so only a handful of mic channels are necessary.

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While software instruments are getting better and computers are becoming powerful enough to run several at once, most people still have a few favourite hardware synths or samplers that they'd like to run alongside their computer tracks, so the ideal mixer needs to be able to accommodate these too. And then there's the latency issue. Most modern computers will run with a latency small enough that most users won't notice it. However, there are many older computers still in service, and some musicians are also particularly sensitive to latency, so a zero-latency monitoring option is still good to have. Soundcraft provide this by allowing the user to route the channel being recorded directly to the monitor outputs while recording (this works only for recording audio, not for software instruments, of course). The mixer also has other tricks up its metaphorical sleeve, but first lets get an overview.

Soundcraft Compact Series Available in a four-channel version as well as the ten-channel model under review here, the Soundcraft Compact mixers are sturdily built from steel, and have a flat profile with rounded front and rear edges, moulded end cheeks, and all connections on the top surface where they can be easily accessed. Power comes from an external PSU, which is less of a concern in the studio than when mixing live, and because most users will be doing their mixing within the computer the channel level controls are on knobs rather than faders. The secret to this little mixer's effectiveness is that, although it looks like a basic 'something into two' mixer, it actually has multiple busses, but to avoid scaring the user these aren't identified as such. One buss carries the record outputs feeding the soundcard, one carries a pre-pan, pre-fader monitor mix (hence it is really a mono pre-fade send on the mic/line channels and a stereo pre-fade send on the stereo channels), and one is a main left/right mix. Record and Monitor buttons are provided at the bottom of each channel so you can decide where the channel signal should be routed — to the soundcard inputs only, to the monitor buss only, or to both. In addition, the channel signal goes directly to the main stereo mix when the Record button is off. The Monitor outputs feeding your speakers get their signal from the Monitor buss, and any Monitor button activated adds that signal to the buss. Note that the Level controls affects only signals fed onto the main stereo mix buss, as the Monitor switches are wired before the Level controls. For zero-latency monitoring, the channel Monitor button can be used to audition the signal being recorded at source, and by disengaging the Record buttons on channels not being recorded, the recording can be kept as clean as possible with no risk of accidentally adding contributions from other channels, though of course you can opt to press Record on several channels at once if you're recording a mix of signals. Having got the basic theory out of the way, it's time for a brief tour of the facilities.



Channel Features As with many small mixers these days, there are two types of channel — mono mic/line and stereo line-only. The mic/line channels (there are four on the Compact 10) utilise Neutrik balanced combi jack/XLR inputs, and phantom power is switchable per pair of channels rather than globally. The first two channels have switchable low-cut filters, while the second two use a similarly positioned button to select a DI mode (with high input impedance) for the jack input, optimising it for use with electric guitars and basses. All the mic/line channels have a TRS jack insert point directly below the input. While you may not need to insert anything while recording (most DAWs have perfectly adequate compressors and EQ), the insert points can also be used to provide direct, preEQ outputs when used with appropriately wired cables. This is handy if you need to record more than two separate tracks at a time. The channel gain trim control is followed by a three-band EQ without bypass switching, and this offers ±12dB of adjustment at 60Hz, 600Hz, and 12kHz. After that there are channel Pan and Level knobs, but there are no aux send controls because, in the intended application, they aren't necessary. In fact I'd argue that even EQ isn't necessary on the record channels, as it's invariably better to record flat, but at least you have the option. All the remaining channels are stereo, the last two having phono inputs only — these are individually switchable to match turntables, complete with the necessary RIAA equalisation. The remaining stereo channels offer both jack and phono inputs and a mono/stereo button. When Mono is selected, the same signal feeds the left and right busses in equal amounts. All stereo channels have the same EQ, pan and level facilities as the mic/line channels. Record and Monitor buttons are located at the bottom of each channel strip. Normally the channel signal feeds the main stereo mix buss, but when Record is pressed, it is rerouted to feed the Record Out sockets instead.

Master Section The Master section is also very straightforward, and includes master level controls for the monitor and mix outputs, as well as two sets of headphone outputs, one intended for use by the artist and one for use by the engineer, where the engineer can switch between the artist's phones mix and the regular monitor mix. The Artist output also has a further control that allows the artist to set the desired mix of the main stereo mix buss (and hence the soundcard output plus any other sources feeding the mix buss) and the signal being recorded, so the performer can set his or her own balance of voice/instrument against the recorded parts. Below the Mix level control is a Monitor button that allows the stereo mix to be sent to the Monitor outputs. Normally the Mix output would be used to feed the final mix to a master stereo recorder, where any live inputs



(such as synths) can be combined with the soundcard playback. Of course it could also be patched back into the soundcard input if you wished to record directly back into the computer. Further level controls are provided for the Record Out and Playback In connections, and a Playback Monitor button enables this source to be added to the monitor mix. A nice touch, missing on so many small mixers these days, is a simple Mono button on the control-room Monitors outputs, so that you can check the mono compatibility of your mix, and of course there are stereo output level meters as well as power indicators and a low-battery indicator. This battery indicator may suggest that the mixer can accept batteries as well as a mains adaptor, and it can. However, these batteries must be connected via an optional external battery pack that plugs into the PSU socket — there's no provision for onboard batteries. In terms of connectivity, it's nice that the mixer doesn't present you with balanced XLRs when what you really want is unbalanced phonos or simple jacks. In fact, the Playback In and Record Out connectors, which would normally by patched to the ins and outs of your soundcard, are on both balanced jacks and unbalanced phonos, which are the most common forms of soundcard connector. The Mix and Monitors outs are on balanced jacks, and the headphones are on stereo quarterinch jacks as you'd expect. In fact, there's only one unexpected connector, and that's a chunky ground terminal in a recess of one of the end cheeks. Soundcraft suggest that you might want to use this to earth any turntable connected directly to the mixer, but it could also be used to provide a technical earth if the mixer was being run from batteries and was connected to another non-grounded device such as a laptop computer.

In Practice Considering what a simple mixer this is, Soundcraft have managed to cram a lot into it, but there are still some useful features missing. One might be a Bounce button, where the stereo mix is fed directly to the Record Out connectors when mixing back to a spare stereo track on the computer. The other is an integral talkback mic feeding the Artist headphone mix, because unless you're working in the same room (and most computers are too noisy to allow that), you may have problems communicating with the performer. Other than that, they've covered a lot of ground without making life too complicated for the user. In a typical computer-based system, the Playback Ins and Record Outs would be connected to the audio interface or soundcard, the Monitor Outs to active monitor speakers or to a power amplifier feeding speakers, and the Mix output to a stereo recorder such as a Minidisc or CD-R recorder. The artist being recorded would file:///H|/SOS%2004-07/Soundcraft%20Compact%2010.htm (4 of 6)9/22/2005 8:56:10 PM


wear headphones plugged into the Artist headphone socket, while the sources to be recorded would be plugged into the input channels (and their routing buttons set to Record). Normally, this would require you to disable software monitoring in order to prevent both the direct and delayed sound being audible at the same time. Any synths playing along with the mix could be connected to stereo inputs with the routing set to Monitor. However, if you're using the Artist headphone mix rather than the main monitor speakers you don't need to press the Monitor buttons, as these sources feed the main stereo mix anyway. If there's already material recorded that the performer needs to hear via the studio monitors, the Playback Monitor button would also need to be down, otherwise the soundcard outputs would only feed the main stereo mix buss. Using the Record routing buttons, any connected synths could also be recorded to the computer one at a time to convert them to audio tracks. My tests on the unit confirmed that it was straightforward to use and the mic amps were subjectively clean and reasonably transparent sounding, even though the audio bandwidth of the mixer only extends from 20Hz to 20kHz (±0.5dB) where the current fashion is to have audio bandwidths extending to 30-40kHz or greater. I saw no reason to doubt the quoted -128dBu EIN figure. However, there is an anomaly on the mic Gain controls that I found annoying. The last 10dB or so of gain comes in suddenly in the last millimetre or two of the gain pot's travel, and it is so abrupt that I was unable to set any interim gain settings — it really is a distinct step. With capacitor mics used at a normal working distance, this amount of gain shouldn't be necessary, but this characteristic may cause some gain-setting problems when working with more distant mics or low-sensitivity dynamic mics. The RIAA inputs seem well suited to typical magnetic pickup cartridges and are ideal both for DJ-style recording and for transferring vinyl samples into your computer. There seems to be adequate level to drive the majority of soundcards, and the jack outputs can cope with up to +20dB of level. The EQ also seems suitably benign, though the mid-range is best used for cut only, as the frequency chosen sounds really boxy or honky when used for boosting. Having said that, you may never need to use it if your computer system has decent EQ. The routing system covers most desktop recording eventualities, and of course you could also use the Compact 10 as a regular live sound mixer provided that you accept its limitations, specifically the lack of any effects sends. Some effectssend capability would have been useful in studio applications too, as there's no straightforward way to add reverb from a hardware unit while recording a vocal using zero-latency monitoring. On the whole, though, the Compact 10 is a good compromise between flexibility, simplicity, and cost, and I like the fact that the monitoring is in stereo rather than mono as it would be when set up with a simple pre-fade send. It's a shame that, although the mixer is actually very quick and easy to use, the manual doesn't do a very good job of guiding you through the process of running a session. Descriptions of what the controls do are all very well, but the user also needs to know how they all work together in practice.



Marks Out Of 10 I feel that a mixer specifically for computer recording has been long overdue, and Soundcraft seem to have got it about right for the less complicated end of the market, especially at this UK price. Having said that, if you don't need the RIAA turntable inputs, but you do need to record more than a couple of sources at once on a regular basis, then you can get a lot more flexibility (at the expense of more complex operation) by spending just a few more pounds on a conventional multi-buss mixer. Where the Compact series scores is simplicity, and it's also a good shape for desktop use, as it's not too deep. In some ways its a pity a battery compartment was not built in, as it would have made mobile use with a laptop very convenient. While I would have liked to have seen a built-in talkback mic and maybe some additional routing for mix bouncing back into the computer, the Compact 10 does a great job of streamlining the business of both recording to and monitoring from a computer DAW. Definitely a step in the right direction.





In this article:

Tour The MI4 Soft SL The Soft Underbelly Performance System Requirements Rock The System?

Steinberg Cubase System 4 USB Interface & Recording Software (Mac/PC) Published in SOS July 2004 Print article : Close window

Reviews : Computer Recording System

Steinberg Cubase System 4 £400 pros Versatile audio feature set with good 'plugability' for the compact studio. Good subjective audio quality. Cubase SL 2 software shares the stability and a good deal of the power of its big brother SX2. No additional copyprotection dongle required.

Buying your hardware and software from the same manufacturer is usually a good bet for hassle-free recording. Steinberg's latest package bundles their Cubase SL 2 sequencer with a four-in, four-out USB interface. Mike Bryant

What's the attraction of buying a single 'system' to meet your computer cons recording needs? For one thing, it Using the MI4 at low gives the combination of software and latencies can put a lot of strain on your PC. hardware an unmatched stamp of Usual USB bandwidth approval in terms of interoperability, limitations to take into account which is always good for peace of where 24-bit/96kHz mind. Another factor is the price: you performance is concerned. might expect to get a bit more for your Photo: Mike Cameron Direct signal monitoring money with a bundle compared to limited to two simultaneous inputs (panned in stereo), and equivalent products bought separately from different manufacturers. excludes the S/PDIF input.

summary

Steinberg are no strangers to this idea. They've previously aimed at the cheap Whilst those seeking the most end of the market with their Cubasis VST Project Pack, and have worked hard at efficient low-latency the not-so-cheap end to conjure up the aura of the 'system' by re-branding RME performance might want to hardware for the Nuendo line. Now they're looking more to the middle ground, look elsewhere, Steinberg's with a package that compares readily with Digidesign's M Box/Pro Tools LE excellent Cubase SL 2 bundle. This is the System 4, which comprises a pairing of the mid-range Cubase software combined with an attractive and versatile piece SL 2 sequencing software and an all-new USB 1.1 audio interface called the MI4. of USB hardware makes for a Though early shipments were marked as Windows only, I'm informed that great package at a decent support for Mac OS X should be included in the same box by the time this review price. hits the shelves. information £399.99 including VAT. Arbiter Music

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Technology +44 (0)20 8970 1909. +44 (0)20 8202 7076. Click here to email www.arbitergroup.com www.steinberg.net

Test Spec

Tour The MI4 The MI4 is a solid, half-rack-sized unit with an attractive brushed-metal top and front, and comes jammed into the System 4's box along with a one-metre USB cable, installation CD, a 60-page hardware manual, and three separate copies of the 240-page Cubase Getting Started guide in English, French and German. (Being almost entirely monolingual, I would've preferred a single paper copy of the comprehensive manual, but never mind.)

MI4 driver version 2.3.7. IBM Thinkpad T30 with Pentium 4-M 1.6GHz CPU and 512MB RAM, running Windows XP SP1a, with a Lacie D2 USB2 external hard drive and Echo Layla 24 audio interface. Tested with Steinberg Cubase SL 2.02 and SX 2.01.10, Wavelab 4.01b, Propellerhead Reason 2.5.

All in all, the MI4 is — you guessed it — a four-in, four-out unit, with four analogue inputs, two analogue outputs, and co-axial S/PDIF and MIDI I/O. The first two analogue inputs are equipped with mic preamps, and round the back you'll find they boast connectivity in the form of separate XLR and balanced jack sockets (as opposed to the cooler-looking but less practical Neutrik combos). Next up are a pair of TRS insert points for inputs 1 and 2, followed by jacks for inputs 3 and 4 (unbalanced) and the main stereo output (balanced). A tour of the MI4's front panel reveals a single high-impedance instrument input, gain knobs for the preamps along with associated four-LED meters and phantom power indicator, peak and signal indicators for inputs 3 and 4, a direct signal monitor mix knob, a headphone output with its own dedicated volume control, and a master output level knob. There is also a bright blue power LED, which flashes if no valid clock source is received from the digital input, and MIDI activity indicators.

The System 4's software component is Cubase SL 2 — little brother to Cubase SX 2, but still mightily powerful. The MI4 serves as copy protection, so there's no need for a separate dongle.

It should be apparent by now that Steinberg have intended the MI4 to be a versatile solution for mobile and compact studios, catering for electric guitars, condenser mics and line-level studio gear at professional and consumer levels. It's also handy to be able to make use of any external processors you might have lying around, either as inserts, or as send effects via the S/PDIF I/O. The inclusion of MIDI I/O rounds out what I consider an extremely well-judged featureset for the sometimes-mobile musician. Another point to note if portability is an important consideration is that the MI4 is powered entirely by its USB connection, and there's nowhere on the unit to plug in an external adaptor. The possible inconvenience here is that USB hubs or PCMCIA adaptors won't be able to supply enough juice to get the MI4 working without being powered themselves. On the subject of USB practicalities, those not overflowing with USB ports will be



pleased to note that they won't have to allocate an extra one for the usual Cubase copy-protection dongle. With the System 4, the MI4 is the dongle — albeit a rather more useful one than we're accustomed to — and the bundled version of Cubase SL 2 won't start without it plugged in. That's not to say it's the only interface you can use with the software (nor vice versa), and I had no problems switching between the MI4 and my normal Echo Layla 24.

Soft SL Cubase SL 2 is, of course, the little brother of Cubase SX 2, and unlike some of Steinberg's OEM 'lite' versions, uses the same code base as their flagship product. Since SX2 has received plentiful treatment in the pages of SOS, I won't expound too much about it here, but make sure you check out the review from last November's SOS, available on-line at www.soundonsound.com/sos/nov03/articles/ cubasesx2.htm. Many of the differences between the two products are likely to be fairly academic to most users, such as the reduction of the maximum track count from 256 to a mere 128, and the absence of a surround-sound audio path, which the MI4 wouldn't support anyway. The best way to make sure the System 4 caters to all your software needs is to download the PDF comparison chart from Steinberg's web site. For my way of working I really didn't find much wanting in Cubase SL 2 compared with SX2. Unless the score element is important to you, the biggies are likely to be the lack of off-line processing with VST and Direct X plug-ins, and the more limited configurability of the mixer. Otherwise, most of the best new features from the SX2 upgrade are present and correct, most notably the multiple-take track view function, which Mark Wherry rightly praised to the heavens in his original review.

The Soft Underbelly Installation under Windows XP is as easy as USB has always promised to be. All you have to do is run the device installer and follow the prompts as to when to plug, unplug and re-plug the device into your PC. You don't even have to reboot afterwards, and the unit appears clearly labelled in your Control Panel hardware applet as MI4 Audio and MI4 MIDI (as opposed to the irksomely anonymous 'USB Audio Device'), so full marks to Steinberg. The first thing you'll notice when all is done is the presence of a new '4' icon in your Windows System Tray. This opens the MI4 Panel, which illustrates the internal routing of the interface and allows more in-depth configuration of it than is possible with the hardware alone. The mic preamps can be set up by enabling 48V phantom power or a -20dB pad, and you can toggle between the XLR and the jack connectors. In a similar manner, the second input pair can be switched between S/PDIF and analogue circuits, and the gain on the latter adjusted in four discrete levels between -8 and +10 dBV.



Finally, the MI4 Panel lets you switch the direct monitoring source between input 1 on its own (for the highimpedance instrument connection), and each of the two analogue pairs. This is a less effective solution than allowing the user free rein to set up a monitor mix of all four input signals, and it's annoying that there's no way of hearing inputs 2, 3 and 4 other than panned as part of a stereo pair. I was also disappointed to find it's not possible to directly monitor the digital input.

The MI4 Panel provides a nice overview of the unit's internal routing, and allows you to configure various settings not accessible with the hardware, such as phantom power, input selection, and the direct monitoring source.

The other software side of the MI4 is its ASIO driver control panel, a customised version of that sold by Propagamma and licensed for many different USB products from many different manufacturers. This driver offers a choice of six latency settings, ranging from a 'relaxed' 138 milliseconds down to a 'high speed' 7 milliseconds, measured on the input and output combined at 44.1kHz.

The remaining controls concern the device's clock rate and bit depth. Whilst the MI4 is capable of 24-bit, 96kHz performance on both the inputs and outputs, the limited bandwidth of USB 1.1 means that certain compromises have to be made when using the high-quality modes. At 24-bit a total of six I/O streams can be enabled (four in, two out, or vice versa), whereas in full 24/96 mode this falls to just two, with a choice of stereo in, stereo out, or one mono channel each way. For a reason I can only imagine to be a bug, the mono 24/96 setting would only let me record via the XLR input on channel 2 — despite it being deactivated in the MI4 Panel! Though Steinberg obviously envisage the hardware being used in conjunction with their software, it's worth pointing out that the MI4's four-in, four-out capability is only available if you use the ASIO drivers — to Windows it appears as a standard stereo device. In this case, the digital output mirrors the main analogue stereo outputs, a feature that could come in handy if you want to record to a Minidisc or something.

Performance I had expected the MI4's low-latency performance to be broadly similar to that of the last product I tested that used the Propagamma ASIO driver (Audiotrak's Maya 44 USB), and in general terms, it was. On my 1.6GHz Pentium 4-M Thinkpad, the lowest 7 millisecond 'high speed' setting was usable only under very light load, and at a 'rapid' 13 milliseconds the device showed a processor use between 10 and 20 percent higher than the nearest equivalent latency on my file:///H|/SOS%2004-07/Steinberg%20Cubase%20System%A04.htm (4 of 6)9/22/2005 8:56:16 PM


Echo Layla 24. Nevertheless, I found that stability at low latencies was rather better than I'd anticipated, either due to differences in the hardware or improvements made to the driver since I last looked at it. Except on the 'high speed' setting, the dreaded clicks and pops were entirely absent, and although real overloads still tended to have a truly stultifying effect on Windows, the MI4 was well behaved. In terms of audio quality, my tests with Rightmark's Audio Analyser very much agreed with the results listed in the MI4's technical specifications. Loopback recording at 24-bit/44.1kHz returned a residual noise level of about 93.1dB(A), along with total harmonic distortion of around 0.007 percent on inputs 1 and 2. Recording the MI4's outputs into my Echo Layla 24 gave a respectable noise figure of -95.6dB(A), which again concurs with that quoted in the specs. Where day-to-day use is concerned, I could find little to complain about on the quality front. The instrument input is clean and provides plenty of gain, while the mic preamps appear perfectly neutral and quiet. The outputs also sounded good to these subjective ears, comparing well with the Layla 24, though the maximum output level was about 6dB lower than that (more expensive, mains-powered) device at its 10dBV setting.

System Requirements PC Processor: Pentium or Athlon 800MHz (1.4GHz or faster recommended). RAM: 384MB (512MB recommended). Operating system: Windows 2000, Windows XP Home or XP Professional.

Rock The System? All round, Steinberg's System 4 comes across as a good way to leap nimbly into the world of computer recording, or replace an existing setup with a more compact and mobile one. The package is certainly very comprehensive, with a versatile hardware component and software that — albeit a 'mere' cut-down version of Cubase SX 2 — is still massively powerful.

Mac Processor: Power Mac G4 867MHz (Power Mac G4 dual 1.25GHz or faster recommended). RAM: 384MB (512MB recommended). Operating system: Mac OS 10.2.5 or higher.

The decision to choose the System 4 over its most obvious competitor, Digidesign's M Box, will rest largely on whether the look of Pro Tools LE floats your boat more than that of Cubase, and I think it fair to say that Steinberg's is the more open and flexible bit of software. With twice the number of simultaneous I/O channels, plus the inclusion of MIDI, the MI4 also boasts a more generous paper specification, and whilst its retail price is a bit higher than the M Box currently sells for, things are already starting to even out on the high street. On a more general level, it's worth considering whether the MI4 — in common file:///H|/SOS%2004-07/Steinberg%20Cubase%20System%A04.htm (5 of 6)9/22/2005 8:56:16 PM


with other USB 1.1 interfaces — can meet your needs in terms of low-latency performance. For a lot of musicians this will be a non-issue, but soft-synth aficionados and those who require live-input monitoring should be mindful of the CPU penalty they're liable to pay when taking advantage of the speedier settings provided by the MI4's ASIO driver. The longstanding question mark over general USB compatibility is also a justified concern, though I experienced no such issues with the MI4 on my system. With these caveats in mind, it's easy to recommend the Cubase System 4 as a complete solution into which you can plug a mic, a synth, or a guitar, and record using one of the best software packages around for a decent price. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Studio Electronics Modmax Filter & Phaser

In this article:

Controls & Facilities How Do They Sound?

Studio Electronics Modmax Filter & Phaser Analogue Effect Pedals Published in SOS July 2004

Studio Electronics Modmax Filter & Phaser £330

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Reviews : Effects

pros Good basic filtering and phasing effects. Extremely flexible envelope and LFO modulation. Rugged.

cons

Two new effect pedals from the celebrated synth manufacturer provide serious modulation flexibility within a stomp-box format.

Control layout not particularly intuitive.

summary Both pedals are flexible and are capable of some good effects, but I felt they were better as creative sounddesign tools than for straight processing of instruments.

information Modmax Filter, £329.99; Modmax Phaser, £329.99. Prices include VAT. Turnkey +44 (0)20 7419 9999. +44 (0)20 7379 0093. Click here to email www.turnkey.uk.com www.studio electronics.com

Paul White Photos: Mike Cameron

Studio Electronics seem never to do a job by halves, so when they went into the pedal business they ended up building something closer to a series of analogue studio rack processors in a pedal format. Their Modmax pedal range comprises Filter, Phaser, and Ringmod, the first two of which are reviewed here. Both follow a similar physical format, with an included mains PSU, a 13-knob control panel, a three-position waveform switch, and a stomp-box-style bypass switch. All models can be used with electric guitars, line sources, or other electronic instruments via the unbalanced jack input, and in all cases the output is mono on an unbalanced jack. Although structurally simple, the pedals are very sturdily built from folded sheet steel, with plenty of clearance between the switch and the knobs to allow for imprecisely placed feet! The bypass switch is of the traditional latching type that responds with a firm thunk when pressed. I felt the switch legending wasn't particularly clear, and the arty typeface didn't help much either. On a practical note, a sticker under each unit also warns that the silk screening depicting the power supply polarity is wrong, so it's probably safest to stick with the power supplies that come with these pedals.

Controls & Facilities The Filter pedal is essentially a resonant filter that has switchable modes (band-

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pass, low-pass, or high-pass) and that can be modulated either by an envelope follower, an LFO (with switchable triangle, square, or random waveforms), or a combination of both. The envelope follower's behaviour is set using the Input control (which affects the trigger threshold) and the Attack and Release controls. The LFO section has its own Depth and Rate controls, as well as a Fast/Slow range switch, and there's a four-LED meter in the centre of the unit that shows when the input is high enough to trigger the envelope follower. Ideally, the red LED should light only on the signal peaks. The secret to the flexibility of all the units in the Modmax range is that you can dynamically modulate four different effect parameters, independently and simultaneously. There's the sweep of the filtering or phasing effect, the effects mix, the LFO rate, and the LFO depth. Both positive and negative modulation are available. The Input gain control has enough range to accommodate guitars, electronic instruments, and line-level signals, and it feeds directly through to the output when the bypass switch is operated. Although it's a clunky mechanical switch, there's no noticeable electronic switching noise when operating it. The Output control sets the output level to match line-level devices or instrument amplifiers, while the oddly named Drench slide switch adds extra gain that can be used to overload the effect output stage in a creative way. The effect level may then be re-balanced using the wet/dry Mix control. The associated Dynamics control modulates the wet/dry mix from the sweep envelope. The filter section has the three switchable modes described earlier, plus the usual Frequency and Resonance controls. Three trigger modes are available for LFO control over the filter sweep: Norm, Trig and One-shot. In Norm mode, the LFO is free running, while in Trig mode the start of the LFO cycle is re-triggered from the input signal whenever it is high enough to light the red LED at the top of the centrally located bar-graph meter. In One-shot mode, the LFO triggers in single cycles, producing an envelope-like effect. Further Dynamics knobs provide real-time control over the filter sweep's range and the LFO's rate and depth. The LFO's own Depth control determines the depth of the LFO signal before any dynamic control is applied. The control layout of the Modmax Phaser is very similar to that of the Filter except that the filter mode switch is replaced by one that selects two, four, or six stages of phasing, corresponding to the analogue phase-shift stages employed by the original phaser pedals. As a rule, the more stages you have, the more notches you get in the frequency response and the stronger the phasing effect. In this instance, the Resonance control makes the phased sound more peaky in character by accentuating the notches and peaks, while Frequency affects the tonal range. I found that moving Frequency to either extreme tended to seriously dilute the phasing effect — it worked best to leave this control in the middle of its range.



How Do They Sound? Filter produces the characteristic swept-filter sound in resonant low-pass mode, while the band-pass mode is more like an auto-wah effect. The high-pass option generates the characteristic thin, cutting sound that you hear when all the low frequencies have been removed, so what you get in the filter department pretty much covers all the bases without getting too silly. I couldn't make much sense of the Drench switch on the Filter model, as it seemed over-aggressive and difficult to control, not to mention noisy. However, it fared somewhat better on the Phaser, adding to the range of available effects. The types of modulation effects on offer, as you might expect, range between envelope-modulated filter sweeps and LFO-controlled filter modulation, and you can get interesting combinations, such as filter sweeps where the LFO control only takes effect once the effect of the envelope follower has died away. Deep LFO modulation of the filter frequency with high resonance settings can cause very audible thumping effects, though, while in the case of Phaser some settings reveal very obvious sweep modulation noise. The quality of both the filtering and phasing effects is warm and generally musical, though whether you need this degree of modulation control to improve your music only you can say. I also found the control operation to be somewhat confusing and the results were not always predictable. As you've probably picked up by now, I'm in two minds about these pedals. The basic quality of the effects on offer isn't in dispute, but the controls aren't as intuitive as they might be and it's also rather too easy to end up with a noisy or thumpy-sounding end result if you're not very careful about how you set the controls. Being able to dial up effects that respond dynamically to the way you play is generally a good thing though, and being able to dynamically blend envelope- and LFO-modulated effects makes it possible to set up sounds that would be impossible by any other means. While I can appreciate the creative potential of these units in some areas, I know I'd probably never use them myself as straight guitar processors, as I can get all the effects I need from more conventional effects units or plug-ins. However, there are other musicians who would probably go for this combination of flexibility and quirkiness in a big way, and I can see lots of applications in processing sounds to create dance grooves or loops, or in mangling the sound of a synth. If you are adventurous with your filter effects, you might find this range of pedals gives you the tools you need to create new and wonderful treatments, but that's something you'll have to decide for yourself. Published in SOS July 2004






Q. Do I need to compress my audio before A-D conversion?

Q. Do I need to compress my audio before A-D conversion? Published in SOS July 2004 Print article : Close window

Sound Advice

I've been told that I should use an analogue compressor or voice channel with compression as a front end when recording digitally, as this will reduce the dynamic range of the input signal and make better use of the A-D converter's available bit depth. Is this true? Nathan Henley Features Editor Sam Inglis replies: This is a widely held belief, but shouldn't be adhered to blindly. For one thing, it's important to consider the role of the compressor's attack time. In lots of sources, the highest signal levels occur during transient peaks, for instance at the start of guitar chords or drum hits. Many compressors are not capable of reacting instantly to these peaks, even with their attack time at its fastest setting. If this is the case, the highest peaks in the signal will be passing straight through the compressor whilst the quieter portions of the signal are attenuated (as shown in figure 2 below). With no make-up gain applied, the compressed signal will therefore tend to present lower overall levels to the A-D converter — yet because the transient peaks have not been attenuated, there is no additional headroom to apply make-up gain, and doing so risks clipping the converter. If you do want to raise the average level of the signal prior to A-D conversion, then, you need to make sure you're using a compressor with an attack time fast enough to squash these transients, or teaming your compressor with a peak limiter that can catch what the compressor misses (see figure 3). Some input channels, such as the Focusrite ISA430, do offer both compression and limiting, whilst some A-D converters also feature 'soft limiting'. Other signals, such as vocals, present fewer problems because they contain fewer sharp transients. However, it's worth thinking about why you would want to muck about with the dynamics of your input signal in this irreversible way. Modern A-D converters offer plenty of dynamic range — enough to capture the most demanding sources and still leave plenty of headroom to avoid clipping, especially if you're recording at 24-bit. It should therefore be unnecessary to compress a signal in the analogue domain simply in order to capture it at acceptable quality in a digital system. On the other hand, lots of engineers just like the sound of analogue compression, and feel that it can't be reproduced using plug-ins or other digital compressors. In digital recording systems, the most convenient place

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Q. Do I need to compress my audio before A-D conversion?

to apply analogue processing is on the way in, since this avoids sending recorded signals through another stage of D-A and A-D conversion. So if you want to use analogue compression because of the way it sounds, it often makes sense to do so at the input stage. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q. How can I resolve an IRQ conflict?

Q. How can I resolve an IRQ conflict? Published in SOS July 2004 Print article : Close window

Sound Advice

My PC uses an Abit KR7A-133 motherboard (not RAID) running an Athlon Thoroughbred XP 2400 processor with 1GB of RAM. The expansion slots currently contain the following devices: AGP slot — Matrox G450 dual-head

Some PCI devices are happy to share an IRQ, with little impact on their performance.

PCI slot 1 — SCSI CD writer PCI slot 2 — empty PCI slot 3 — M Audio Audiophile PCI slot 4 — empty PCI slot 5 — empty PCI slot 6 — TC Powercore My motherboard manual says PCI 4 shares an IRQ with the two USB ports, and that PCI 1 shares with the AGP. My Audiophile currently has an IRQ to itself and so I'm not touching it. However, I need the Powercore to not share an IRQ. In my BIOS, it's not immediately clear how to specify which IRQ is for which slot, since it only allows me to set each IRQ to 'PCI Slot' or 'Reserve'. I don't understand what reserving means, but I have all the IRQs assigned to PCI, so why do some PCI slots need to share? When I put the Powercore in slot 5 it shares an IRQ with the AGP and SCSI, and in 6 or 2, it shares with the USBs. IRQs 2, 4 and 7 don't seem to be in use at all, so how do I assign IRQ 7 to PCI slot 6? My motherboard manual doesn't mention slot 6 sharing with anything; it states that only slot 1 and 3 share IRQs with other things. This is what baffles me. I've read Martin Walker's PC Musician article on PCI slots and interrupts from May 2003, but my IRQ table is set out differently to the way Martin describes. I've got a PIRQ table in my manual but I can't make head nor tail of it. If anyone wants to explain what to do I'd really appreciate it. SOS Forum post file:///H|/SOS%2004-07/Q.%20How%20can%20I%20resolve%20an%20IRQ%20conflict.htm (1 of 3)9/22/2005 8:56:43 PM


PC music specialist Martin Walker replies: I'm not surprised that you're confused, since, as I explained in the PC Musician article you refer to (you can find it at www.soundonsound.com/sos/may03/articles/ pcmusician0503.asp), what determines how the various slots and other PCI devices share their interrupt resources is down to the motherboard manufacturer, and here we enter a mysterious world where each model of motherboard can be different. If a particular expansion slot shares an internal interrupt with another device, this cannot be changed — it will share regardless of what you change in the BIOS or in Windows. However, remember that the majority of modern expansion card devices will be quite happy to share an IRQ, and if this is the case there's normally only a tiny performance hit if your soundcard (or, in this case, your Powercore card) ends up sharing with something else. Having said that, it is preferable for them to each have their own unique IRQ if possible. Since your motherboard's Interrupt Request Table is obviously in such a different format from my own, I downloaded it to shed some light on the problem (the IRQ table is reproduced below), especially since this may also help other musicians in deciphering their own IRQ tables. Unfortunately some Abit tables have subsequently proved to have printing errors (the AT7-MAX for instance), but let's assume yours is correct. The PIRQ signals in the left-hand column are those from the VIA VT8233 chip set on your motherboard — only four unique interrupts are available to PCI slots (PIRQ 0 to 3), but as explained elsewhere in your motherboard manual, you can if you wish assign whatever IRQ number you like to each one using the four 'PIRQ_X Use IRQ No' settings in your BIOS. The various 'INT' entries in the other columns relating to each of the PCI expansion slots are exactly the same as those I discussed in my feature — each slot has four physical connections to be allocated to up to four interrupt lines, generally given the letter-based names Int A, Int B, Int C and Int D rather than numbers to avoid confusing them with system IRQs. As I also said previously, if an interrupt is required by any expansion card, like your Powercore, Int A is nearly always assigned, leaving the other three for possible use in a multi-function device. So, for instance, the Audigy 2 soundcard uses A for its main audio/MIDI portion, and B for its IEEE 1394 Host Controller. However, since the majority of cards only require one interrupt, in nearly all cases Int A is the only one to look for in the table. When we do this, we discover that: PCI slot 1 has Int A assigned to PIRQ 0 (whatever Windows or the BIOS assigns to this). No other PCI slot has Int A assigned to PIRQ 0, but, as you say, and the manual confirms, PCI slot 1 shares an IRQ with the AGP slot, so both your audio expansion cards should avoid this slot if possible; PCI slot 2 has Int A assigned to PIRQ 3, as does PCI slot 6, so these two will normally share an IRQ as you've already found; PCI slot 3 has Int A assigned to PIRQ 1, as does PCI slot 4, so these two will share, so leave your Audiophile in PCI slot 3 and keep slot 4 empty if possible; and PCI slot 5 has Int A assigned to PIRQ 2, and while according to the notes accompanying the printed file:///H|/SOS%2004-07/Q.%20How%20can%20I%20resolve%20an%20IRQ%20conflict.htm (2 of 3)9/22/2005 8:56:43 PM


table this shares an IRQ with the Highpoint HPT 372 chip set, according to your motherboard manual only the 'R' version with RAID support has this chip, so yours won't share. Even though you may have one or more 'spare' IRQs not allocated to anything, it's the hard-wiring of the slots that determines what shares with what, and you can't avoid this. The two USB ports provided by each USB Host Controller always share an IRQ, and the notes beneath the table suggest that this is shared with PCI slot 4, but, since slots 3 and 4 share, this isn't borne out by your Audiophile in slot 3 getting its own unique IRQ. Your tests instead suggest that both USB ports share with slots 2 and 6, so perhaps this is a printing error! So, my best suggestion for your Powercore card is slot 5, since this doesn't seem to be sharing at hardware level. The fact that you've found that this ends up sharing with AGP and Slot 1 might be due to another factor — elsewhere in the manual it states that your motherboard has an APIC (Advanced Programmable Interrupt Controller) so do make sure this is set to 'Enabled' in the Advanced BIOS Features page — in the 'Disabled' mode only six IRQs will be available for all motherboard devices. Remember also that you can disable unwanted motherboard devices such as serial and parallel ports to release their IRQ allocations to the pool. If you still can't get a unique IRQ for your Powercore card, try setting your 'PNP OS Installed' BIOS parameter to 'No', so the BIOS will allocate IRQs, and then force the appropriate PIRQ to use your choice with the four 'PIRQ_X Use IRQ No' settings mentioned earlier. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q. How do I take care of my valve mic?

Q. How do I take care of my valve mic? Published in SOS July 2004 Print article : Close window

Sound Advice

I've just got hold of an Oktava valve mic. Having not used one before, I was wondering if there are any general rules or guidelines I should be aware of when using valve mics. Is it harmful to leave the mic powered up for long periods of time — a whole day, say — or should it be used just for the times you need it? SOS Forum post Technical Editor Hugh Robjohns replies: I have a couple of 'do's and don'ts' for you. Never disconnect the mic body from its dedicated power supply unit without turning the power supply off first, and never switch on the supply unit with the mic disconnected. Never drop the mic (obviously!), and be very gentle with it for a few minutes after it has been switched off. It's usually best to leave the mic to cool off completely before disconnecting, dismantling and packing it away! When you first power it up, allow up to five minutes for the microphone to warm up and fully stabilise before making any critical recordings. Valves do age over time and will eventually 'wear out', although this is an extremely slow (and often subtle) process. Changing valves is not particularly difficult in most cases (it's best to refer to the manufacturer's guidelines for instructions on how to do this) and new valves needn't be expensive, but it pays to use high-quality valves. If you are using the mic on and off throughout the day then I would leave it on the whole time. If you are using it for a recording in the morning and then maybe again late at night, I'd probably turn it off in between times. Published in SOS July 2004

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The Oktava MKL2500 valve mic uses a 6C315P vacuum tube.

Q. How do I take care of my valve mic?



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Q. How does virtual surround work?

Q. How does virtual surround work? Published in SOS July 2004 Print article : Close window

Sound Advice

Can you tell me how virtual surround systems work? How is a '3D' effect achieved with two speakers? Ivo Eames Technical Editor Hugh Robjohns replies: The more effective 3D-from-stereo systems, such as those from Q Sound Labs and those incorporating Roland's RSS technology, make use of complex comb filtering of the signals in the two channels, with the idea of simulating the natural comb filtering that occurs as sound enters the ears. The term 'comb filter' refers to the characteristic shape of the frequency response which is created when a signal is combined with a slightly delayed version of itself — if you were to plot the frequencyresponse graph of the combined signals, you would see regularly spaced deep and narrow notches that look a little like the teeth of a comb. The resulting effect is often described as sounding 'phasey' or like it's coming down a tube. Our hearing system primarily makes use of level and time-of-arrival differences between sounds arriving at the listener's two ears to help locate the direction of sound sources, but there is also a third technique that relies on comb filtering which is, as yet, still not completely understood. Q Sound's QSys/AX Direct X plug-in produces 3D-from-stereo effects. Essentially, as sound enters the ears it also reflects off the curves of the pinnae (the curved cartilage that forms the outer ears) and possibly also the shoulders, and those reflections combine with the direct signal, create very subtle but unique interference comb filtering.

From a very early age our brains gradually learn how to relate specific patterns of comb-filter notches to the corresponding sound source directions, and it can be a remarkably accurate system. However, everyone's ears are shaped differently and so everyone has slightly different associations between comb-filter pattern and corresponding source direction. This makes it virtually impossible for any system generating artificial combfiltering artefacts — such as RSS and Q Sound — to produce reliable and repeatable 3D positional impressions for more than one person.

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Q. How does virtual surround work?

As a result, although both Q Sound and RSS systems can create the impression of sounds located in 3D space, different people often perceive the same sounds as coming from entirely different places — and sometimes with wildly differing directions. Sound reflections from room boundaries and mixing console surfaces, for example, can also affect the spatial imaging quite dramatically — the deader the listening environment, the more stable and believable the effect will be. So, if you just want a spatial effect these kinds of systems can work quite well. However, if you want accurate 3D imaging from two speakers that will be heard the same way by lots of different people listening in different environments, current systems are probably not the way to go — binaural techniques via headphones are probably a better solution. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q. What kind of reel-to-reel tape recorder do I need?

Q. What kind of reel-to-reel tape recorder do I need? Published in SOS July 2004 Print article : Close window

Sound Advice

A friend of mine has a couple of old stereo tape reels which he'd like to transfer to CD. I've seen a number of stereo reel-to-reel tape machines for sale on Ebay and I'm considering investing in one, probably a Studer or a Revox, not only to help out my friend, but also because I'm interested in experimenting with tape loops and tape saturation effects. Are old machines by these manufacturers reliable? Unfortunately, he doesn't know the make of the reel-to-reel recorder the recordings were made on, or the tape speed at which they were recorded. Are all recorders and players compatible? What different tape speeds are there? Finally, if I want to try editing tape, what equipment do I need? Can you give me some pointers? SOS Forum post Technical Editor Hugh Robjohns replies: Reel-to-reel tape comes in lots of different flavours, and there are even more recording formats. A lot depends on the age of the tapes you're dealing with, but, generally speaking, quarter-inch tape was the norm for domestic mono or stereo tape recorders, while professional mono, stereo and multitrack machines used quarterinch, half-inch, one-inch or two-inch tape — and still do! In the case of stereo recording, 'professional' machines are usually stereo 'halftrack' recorders, meaning that the left channel occupies (almost) half the tape width, and the right channel the other half. Domestic machines tend to be 'quarter-tracks', recording two channels in one direction and two in the other when the tape was turned over, each track occupying slightly less than a quarter of the total tape width — hence the terminology. The same arrangement is used in compact cassette tapes.

A Revox B77 stereo tape recorder, minus tape reels.

Clearly, a half-track machine won't be able to play quarter-track tapes, and vice versa, unless it is fitted with additional heads to accommodate the alternative format. A few recorder models offered this facility as an

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option. Then there is the issue of tape speeds. The usual professional speeds for quarter-inch tape are 7.5 and 15 ips (inches per second). 30ips is rare on quarter-inch machines and tends to be used only on the bigger multitracks and half-inch mastering machines. The greater the width of tape you give to each track, and the faster the tape speed, the higher the quality of the recording in terms of the signal-to-noise ratio, although higher speeds also reduce the low-frequency extension of the recording slightly. The typical tape speeds for domestic machines are 7.5 and 3.75 inches per second, although a lot of machines had the option of a slower speed of one and seven eighths (1.875ips) too. So a tape machine might have settings of 3.75 and 1.875 ips rather than 7.5 and 3.75 ips. Indeed, some tape recorders offer three speed options instead of just two. There is also an even slower speed, used by many radio stations for logging the broadcast output, of 15/16ths of an inch per second! So before you can buy a machine to replay these tapes you really need to find out more about them to make sure your machine has the right type of heads for the tape format, and the correct speed options. There were lots of different tape recorder manufacturers in reel-to-reel tape's heyday, but the Studer machines, and their domesticated siblings, the Revox recorders, are perhaps the best known and most commonly available. The Revox B77 and the professional version of it, the PR99, are both widely available and have better electronics than the older A77, although this is still a reliable and good-sounding machine when set up well. Other variants include the A700, and older valve machines like the venerable G36. Although Studer/Revox no longer make these tape machines, maintenance is still quite easy, parts are easy to get hold of, and there are plenty of specialist repair people around. Revoxes come in all the usual flavours with quarter- or half-track heads and standard (7.5/3.75) or high-speed (15/7.5) versions. The Studer A807 was the last quarter-inch-format machine the company made, and the very last ex-factory stock was sold only a few weeks ago. I reckon the A807 is the best machine of its type that Studer ever made, both in terms of the audio electronics and the superb transport mechanism. Obviously, a lot of second-hand Studers have had a very hard life — particularly those that have come from broadcast environments — so be careful when buying. New heads and motors are available, but a bargain recorder can quickly become a very expensive toy if it needs a lot of tender loving care to get it to perform as well as it should. As to your final question, for tape editing equipment, check out www.canford.co.uk or www.studiospares.com. You'll need a decent editing block, some single-sided razor blades, some splicing tape, a white or yellow Chinagraph (wax) pencil and possibly some leader tape. Standard editing blocks (also called tape splicing blocks) have the three splicing guide grooves set at 89 degrees, 60 degrees and 45 degrees to the central channel which holds the tape. The 89-degree guide is intended for editing out clicks and isn't normally used for music or speech editing. The 60-degree guide in the middle is the one to use for normal stereo editing work, but it is worth noting that the angled cut means that one channel will fade in fractionally before the other. This becomes more noticeable with the 45-degree slope. This provides the longest crossfade between outgoing and incoming audio segments and so tends to give the smoothest edits. But with stereo material the fact that one channel changes before the other can become very



obviously audible with some types of material — something known as a 'flashing edit' because the sound appears to flash from one side of the stereo image to the other. The basic technique for editing is to move the tape over the machine's replay head (usually the one at the right-hand side of the head block,) to find the 'in' point at which you want to edit, and mark the back of the tape carefully in line with the centre of the replay head using the chinagraph pencil. A cutting block for quarter-inch tape with 45Do the same for the other edit point (the 'out') then remove the tape and 60-degree guide grooves. from the machine's heads and tape guides. Place it into the channel of the splicing block, magnetic recording side down, aligning the first chinagraph mark with the middle of the three cutting guides (the 60 degree guide). To save time, you can overlay the second edit point (so that a single swipe of the blade will cut both bits of tape in the right places), or you can cut them one at a time — whichever you prefer.

Having cut the tapes, remove the unwanted section and ensure the remaining tape is aligned neatly at the new edit. Don't throw the removed tape away yet — you may need to splice it back in if the edit doesn't work! Next, cut off about three-quarters of an inch of splicing tape. I like to stick the end to the edge of the razor blade to move it around as this avoids contaminating the tape with finger grease and makes it easier to align with the splicing block. Place the splicing tape over the edit with the join between the outgoing and incoming sections in the centre of the splicing tape. Take care to align the splicing tape with the recording tape so that it doesn't overlap the edges (proper splicing tape is slightly narrower than the recording tape to make sure that the glue can not spread to the tape guides or parts of the tape reel when placed correctly. Press down firmly, then carefully remove the newly spliced edited tape from the block and thread it back onto the tape machine to listen to your edit. If the edit doesn't work for some reason, it is possible to carefully peel off the splicing tape and have another go if necessary — just as well you didn't throw that unwanted bit of tape away too soon! Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

The Red Or The Blue? Cocktail Hour Favourite Places Smooth Grooves Back In The Days Soft And High In An American Studio Creating The Stutter Rap The Benefits Of Digital

Arif Mardin: Producer From Aretha Franklin to Norah Jones Published in SOS July 2004 Print article : Close window

People : Artists/Engineers/Producers/Programmers

Arif Mardin has engineered and produced an incredible array of classic records from artists such as Aretha Franklin, Dusty Springfield, Diana Ross, the Bee Gees and Barbra Streisand — yet the runaway success of his recent work with Norah Jones threatens to overshadow even these achievements. Tom Doyle

There are very few septuagenarian record producers operating in the music industry, and there is surely only one who can claim that his biggest successes are amongst his most recent work. But having taken charge of both of the award-scooping, unit-shovelling albums by Norah Jones, Arif Mardin — five days short of his 72nd birthday when he spoke to Sound On Sound — can reasonably make that boast. Not that this self-effacing, Turkish-born veteran of hundreds of recordings over five decades, many of them modern classics, is likely to ever boast about anything. Yet Mardin admits he was slightly taken aback by how quickly Jones's Feels Like Home began flying out of the shops in the first few days of its initial release. "I emailed her," he says, "and I wrote 'Hey, this is a first in my career: over a million records in one week!'"

The Red Or The Blue? Mardin's remarkable studio career began when he moved to New York in the late file:///H|/SOS%2004-07/Arif%20Mardin%20%20Producer.htm (1 of 9)9/22/2005 8:57:15 PM


'50s and secured a job assisting Atlantic Records' chief engineer Tom Dowd. As a result, he found himself witnessing the heyday of the R&B label and working with the likes of Aretha Franklin, Ray Charles and Dusty Springfield on some of their most famous recordings. By the early '70s, he'd branched out as a producer in his own right, overseeing key releases by the likes of the Bee Gees, the Average White Band, Chaka Khan, Diana Ross and Barbra Streisand. Nevertheless, despite his firsthand experience of the developments in recording technology over the years, Mardin claims to be anything but an equipment-head. "So this is more of a technical interview?" he wonders, a slight touch of trepidation in his voice. "The only problem is if you ask me about 'Which blah blah do you use for this?' I can only tell you it's the blue one or the red one. I know what they can do, but I don't really follow every technical magazine." That's very encouraging to hear from someone who's won 11 Grammies... "Yeah, but it's true. It's like, Give me that blue box!"

Cocktail Hour In 2000, when Mardin was first approached by Norah Jones's label Blue Note to work on the sessions for what would become her 14-million-selling debut album Come Away With Me, the record had already been made once with another producer, Craig Street, though neither the label bosses nor Jones herself were particularly enamoured with the results. "It was more of a folky, guitar-orientated approach," Mardin explains. "Both Norah and the Blue Note people thought that it should actually go back and capture the original piano-flavoured demos. So the band knew the tunes and we went in and I produced about 80 percent of the album. It was actually very quick — I think we did it in less than a week." In the wake of the first album's success, it's reasonable to assume that the sessions for what would become Feels Like Home began with a mood of high expectation. Not so, says the producer. When in April 2003 Jones and her band assembled in the rural retreat of Allaire Studios in Woodstock for a week of recording, the atmosphere was apparently, like the music, remarkably laid-back. "That was a very pleasant week," Mardin recalls. "The view was beautiful, the air was clean and after the sessions, I used to be their barman, mixing up martinis. file:///H|/SOS%2004-07/Arif%20Mardin%20%20Producer.htm (2 of 9)9/22/2005 8:57:15 PM


We cut quite a number of songs and most of them, maybe about eight of them, made the album." While the surroundings were certainly convivial, the producer admits that there were some acoustic challenges to tackle before any music could be committed to tape. "Even though it was a marvellous place to be in," he explains, "their main room, the ceilings were very high. We had to use isolation and put Norah under something almost like a tent so that her voice wouldn't hit the ceiling and come down and reverberate. There are two studios — one is the big one where the ceiling is very high. But in late November, early December, we went there again with Levon Helm and Garth Hudson [formerly of the Band] and we were in the smaller room because it was a nicer drum sound, nicer vocal sound. But the desk was in the middle of the room, there was no control room. So the engineer Jay Newland and myself would put earphones on to monitor the sounds. But we had to make sure we didn't drop a cup of coffee on the floor or something!" Norah Jones's only request when it comes to recording is that tracking is done to analogue two-inch, in this instance using a Studer A827 with Dolby SR in conjunction with an SSL 9000J desk. "We eventually go to Pro Tools," says Mardin. "but only after we've done all the session in analogue. I obey the artist's wishes and she wanted that. If Arif Mardin puts Norah Jones (back, third you ask me, I'm all for using the latest from left) and her band through their paces. technology. I'm not a those-were-thegood-old-days person. The only thing is, if I'm using the newest technology, the sound has to be sweet. There were a lot of new technologies in the '80s and '90s, digital tape recorders and things like that, and the sound was terrible because the converters weren't good. "Back when I was working with the Bee Gees in Miami, we had the first Mitsubishi digital 32-track and it was very, very brittle. Lately I've been working with the 96kHz Pro Tools and I like the sound very much. But I don't use Pro Tools to go in and get into syllables and tune everything up. That's not my style. I use it as a tape recorder." As someone who's worked with analogue for so many years, Mardin still enjoys the benefits of recording to tape, but admits that it sometimes involves patience and perseverance. "With analogue you have that certain sound, of course. But sometimes there are problems. Last year I worked with an artist and she wanted to be recorded on analogue multitrack. But first of all we had a bad batch of tape, then I'd totally forgotten that you can't record heavy stuff on track 23 because it interferes with the SMPTE on 24, and then you have this crosstalk of the bass drum going through other tracks. I said 'We must've been in analogue hell in the '70s!' Don't forget, with Norah's albums, we're talking about studios where file:///H|/SOS%2004-07/Arif%20Mardin%20%20Producer.htm (3 of 9)9/22/2005 8:57:15 PM


maintenance is fantastic. But usually these machines are 30 years old. So that's another factor that, yes, they were great then. But now, you've got to find a good machine, so it still plays well."

Favourite Places After more than 40 years in the business, Arif Mardin has seen the inside of a lot of studios. So what does he look for in a recording venue? "It depends on what kind of recording it is. If I'm going to record a 40-piece orchestra, say, 25 strings and woodwinds and this and that, it has to be a sound stage, like the Sony one in LA where The Wizard Of Oz was recorded. That studio is still intact. I did some Barbra Streisand there about four years ago. And of course in New York we have Right Track, a brand-new big facility for film recording. Or there's the Hit Factory, which is well equipped and maintained and is a great orchestral studio. "Then of course, if it's a small group I'm recording, the room must be intimate and there must be at least three iso booths so I can put the guitarist and the piano separately and so on. It all also depends on how the engineer wants it, but I like to have iso booths. I don't give much consideration to what the desk is. The way I work is, What do we have here? SSL. What do we have here? Neve. Fine. I don't even worry about it."

Smooth Grooves Following the Allaire sessions for Feels Like Home, Jones and her band returned to the road and, as a result of playing the new songs live, developed the material further, particularly when it came to fixing tempos. Recording recommenced in November 2003 at Avatar Studios in New York, formerly the Power Station. "But we weren't in the main room," Mardin points out, "we were upstairs in the smaller room and it was so cosy and excellent. We recorded there for about a week, then we moved onto Sear Sound, where we'd worked during the making of the first album. Sear Sound is owned by Walter Sear and he loves vintage equipment, old Telefunkens and whatever. So we did some tracks and then mixed the whole album there." Of course one of the consistent features of Jones's records — indeed most of the records on Mardin's CV — is the live, soulful feel to the performances. "Oh yeah, they're mostly live," the producer insists. "If there was any blatant mistake, like a wrong note or something, we would correct that. But it's definitely live. That's why with Norah I really feel like I'm at Atlantic Studios in the late '60s. This had that same feeling: very organic, with musicians listening to each other and playing together, which is very rare nowadays. I'm a jazz fan, so I'm for smooth grooves." In terms of mics, Mardin doesn't list any particular favourites. "I rely really on the engineer. Also, mics are like bottles of wine. We had one Telefunken here in New York that we used on the first album and Norah loved it, so we ordered the same in Allaire and it didn't sound the same. So she ended up buying the one that she liked. They're like bottles of wine — one is good, one is... I wouldn't say bad, but file:///H|/SOS%2004-07/Arif%20Mardin%20%20Producer.htm (4 of 9)9/22/2005 8:57:15 PM


different." The mixing of Feels Like Home was completed in a two-week period at Sear Sound in December 2003 and Mardin remembers it as a relatively painless experience. He's keen to point out, however, that unlike some other top-line producers, he still maintains a very hands-on approach to the process. "I'm not a record producer who goes home, saying 'Send me three versions and let me choose one.' I'm there all the time. Working with Jay Newland made it very easy because he's really great. We were probably mixing three songs every two days. Totally analogue, but we were also going to eight Pro Tools stems, so the analogue quality is preserved 100 percent. We'll have the vocal with its reverb as a stereo, a drum stereo with the bass, guitars on another stereo and the keyboards on another. If we say 'Well, we need a little bit more vocal,' or something, then the stems are there and they can be also used for 5.1 mixes in the future."

Norah Jones's debut Come Away With Me is the biggestselling album in Blue Note's history, and her follow-up Feels Like Home sold over one million copies in its first week of release.

Interestingly, Mardin says that his mixing technique is still very much informed by his time back at Atlantic in the '50s and '60s. "Tom Dowd used to say 'Voice and bass, make 'em zero. Then the kick maybe half a dB softer and the rest just sprinkled around.' That's why when Ahmet Ertegun [head of Atlantic] heard Tommy's work when he was 16 years old, he said 'This young boy has the correct bass and drum level to my liking, hire him!' That was the R&B sound."

Back In The Days As an engineer in his Atlantic days, Mardin found himself working with equipment that was remarkably advanced for the time. In 1958, the studio took delivery of what he remembers as being the first Ampex eight-track machine, 10 whole years before even the Beatles were able to take the leap up to the format. "It was one-inch tape and the machine was like a skyscraper. Most producers at that time wouldn't even want to hear about multitrack. So Tommy would go in and do it the old-fashioned way — record whatever was happening onto a mono tape with the correct reverb, correct EQs, mix the master live. But he would also record everything to the eight-track simultaneously. So when stereo started to happen, Tommy didn't have to do fake stereos, he went back to those eighttracks and made real stereos. "But we used it for multitracking too. I remember in '58 or '59, when we were doing 'I Believe To My Soul' with Ray Charles, he didn't like the girls singing harmony and Tommy said 'I have the means — you can do four parts.' So the eight-track was an incredible thing. But as I say, it wasn't used for mixing until file:///H|/SOS%2004-07/Arif%20Mardin%20%20Producer.htm (5 of 9)9/22/2005 8:57:15 PM


later. It was when Tom and I worked with the Young Rascals that we used the eight-track as it's supposed to be used — record first, then mix later." Mardin is keen to point out that as a young engineer, he wasn't afraid to take an editing blade to a two-inch master. For Aretha Franklin's gospel take on 'You'll Never Walk Alone', he Photo: Michael Ochs/Redferns recalls having to go to some lengths to A younger Arif Mardin at Muscle Shoals with mask a splice in the live church Aretha Frankin (back) and Tom Dowd (right). recording. "She didn't like one part of the song, so she came to the studio and played it and sang and said 'Make your edit there.' I said 'How are we gonna make an edit into the live church sound?' So I assembled a lot of people and they would talk and hum and clap and everything to create that atmosphere. Then I took a room murmur of the church and made a long loop of it. On the splice, I put a cymbal and things like that and it worked out fine." Out of all the sessions he did with Franklin, Mardin singles out the recording of 'Respect' as a particular high point. "I was assisting Jerry Wexler, who was the main producer, and the atmosphere in the studio at Atlantic on 60th Street was electric. It was like making a soup. You know, the guitar player plays a lick and we all say 'Keep that, but do this.' Aretha would go next door and rehearse the background parts with her sisters. I would write down chord changes and be the liaison between the control room and the musicians. It was incredible."

Soft And High When pushed, the ever-modest Arif Mardin admits to having been partly responsible for creating the layered falsetto vocal sound that helped to revitalise the Bee Gees' career. "I didn't say 'Hey, sing falsetto, I'd like to invent this sound for you,'" he insists. "But there was a melody in one of the songs — it might have been 'Jive Talkin' — and I said 'Can you take it up an octave, please?' And Barry said 'If I take it up an octave and shout like an opera singer, I'll sound like a fool. Let me sing it in a soft voice.' So he sang it in that voice and the brothers and everyone were saying 'That's great, that's great, keep that.' So that's how it came about. But I didn't say, I want you to sing falsetto here. I just said 'Kick it up an octave.'"

In An American Studio The producer recalls the atmosphere at American Studios in Memphis during the sessions for Dusty Springfield's 1969 classic Dusty In Memphis as being similarly charged, even if it was for very different reasons. "Dusty Springfield was very intimidated because she was saying 'Hey, I'm singing in the same vocal booth file:///H|/SOS%2004-07/Arif%20Mardin%20%20Producer.htm (6 of 9)9/22/2005 8:57:15 PM


that Wilson Pickett sang in.' She would get the key and sing a little bit, but refuse to sing final vocals. So we got the grooves there and in New York she finished her vocals. And after the vocals, we did the strings and horns. "The only thing I remember about American Studios was that it was small. I did many, many recordings there — the drummer would be the same drummer, Gene Christman, so the drum sound would be there, you didn't have to spend hours on it. It was one mic on the kick, one overhead, one near the snare, that's it. The microphones were always in place and everything. It was four tracks and anything that you could equalise together was put on the same track — drums and organ, fine, bass and tambourine, fine, but bass and guitar, a little difficult if you make a mistake in the balance and you want to enhance the guitar. We'd divide the instruments over two tracks and leave one for the vocalist and one for strings or horns or background vocals or whatever." Ever modest, it seems, Mardin doesn't credit his contribution to these records as having been important to their success. "You know, I was blessed to have worked with incredible artists. Sound is not that important if you work with Donny Hathaway or the Bee Gees. They're just incredible singers and they inspire you. With the Bee Gees, when we were doing the Main Course album in the mid-'70s, there was a total stream of creativity in the studio, everybody excited. But we were using state-of-the-art machinery then. We had the ARP 2600 synthesizer — mono, but we created so much modern stuff at that time." Mardin also looks back on some of the proto-MIDI equipment that he used back in the '80s with real horror. "Well I don't know if you remember, but there used to be a lot of MIDI delay in the '80s. There was a device called the Russian Dragon and we used to call it Rush And Drag. You would measure the delay on tape and mark it with a grease pencil. For example, if the kick drum was behind the bass 30 milliseconds or something, you'd adjust that to bring them together. That was the kind of horrible things we had to do. It took hours."

Creating The Stutter Rap Even into his 50s, Arif Mardin was renowned for being a producer who moved with the times, and his studio treatment of Chaka Khan's 'I Feel For You' in 1984 inadvertently created one of the most memorable production hooks ever. "Hip-hop had started and it was incredible. We recorded that in New York and LA and it was a trial-and-error situation. I would put a bass line in and it'd be 'Take the guitar out,' and so on. Finally, when it came about, we were amazed at what had happened. But of course, accidents occur. With Reggie Griffin, who was the producer of Grandmaster Flash's 'The Message', I said to him 'I'd like to have a love rap, very short, no hip-hop boasting about my gold chain, I just want a ChakaI-love-you thing.' So with that instruction, he came back with Melle Mel's rap. In those days we had an AMS sampler, so we put Chaka's name in and my hand slipped on the key. So that's how that Chaka-Chaka-Chaka thing happened. Believe me, it was an accident!"

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The Benefits Of Digital Although he appreciates the sonic virtues of some analogue gear, these days Arif Mardin is very much a digitally-minded record producer. "Well, I can tell you one thing that's great if you're totally in the digital domain. Working on a [Neve] Capricorn a few years ago on a Barbra Streisand project, she was in LA and we were in New York and we would mix the song, play it to her over the fibre-optic telephone line, she would critique it, we'd do the changes, and then she would say 'Oh, what about yesterday's mix?' If it wasn't a Capricorn, the assistant would have to set the mix up, look at his notes, set the EQs and everything. With the Capricorn you press one button and everything is recalled. So with Barbra Streisand, who's known to be very particular, we mixed five songs in seven days. Would you believe that? She was very happy. So in that domain, it was important that we worked on a digital desk. "However, for example when I record analogue and you capture it with Pro Tools, I'm working at 96k, so I don't want to go to a digital board where my sound might be downgraded. So when I work with Pro Tools we go to an analogue board. I did Diane Reeves last year and we used a 192kHz sampling rate — y'know, maybe for a future SACD or things like that — but even so, the CD sounded fantastic, there was something of the 192 in the sound. The silkiness was there. Maybe it's psychological, I don't know." One development in vocal production that Mardin says he only ever uses carefully and sparingly is Auto-Tune. "Sometimes I use tuning if the artist is out of town or whatever — you know, logistical stuff. But I work with artists who can sing, and if a vocal track is perfected by tuning a few notes here and there — a few, I'm saying — I don't mind it because the artist, he or she will take that vocal and sing it live better than the recording. That becomes a blueprint for the artist to sing live. But of course they shall remain nameless. There are so many handsome or beautiful young artists who cannot sing and the vocal is concocted in the studio and they don't even sing live, because they can't. My criteria is if the artist can sing live, then in the studio, I don't mind correcting a few notes." Most importantly, the producer acknowledges that using Pro Tools has made his job a lot easier, particularly when it comes to creating orchestrated arrangements based around one singer. "I did Jewel's Christmas record four or five years ago and we prepared synthesizer tracks of the rhythm section and the orchestra. I'd first do a piano/voice rehearsal with her and a piano player in LA, put the voice on one track, the piano on another track, and then my son Joe would write an arrangement. The arrangement contains that vocal that she did, but we squeeze it, time-compress it, whatever, according to the final tempo. But it's there to inspire us. "Then we'd go to LA with that Pro Tools drive and she may say 'Y'know, I'd like to add eight bars,' and it's like 'No problem.' So then the arrangement is finalised and we're ready for the orchestra. Jewel was touring all the time and we were on



the phone saying 'Ask your piano player to give you a key on "Silent Night".' So we get it, I do a little mock-up using a session singer to do a guide vocal, send it to her, she likes it. Then Jewel came in July in the heat of 95 degrees and finished the vocals. We had to decorate the vocal booth with Christmas decorations! "It's interesting right now because Joe and I are co-producing Queen Latifah and it's a very intricate musical situation, big-band jazz standards, wonderful stuff, wonderful singing. So it's the same thing — we have five new songs, fly to LA, we hone the synth-based arrangement, she does a beautiful vocal line and now we're ready to go. Also, it's less expensive because can you imagine if you've got 40 people in the studio and the artist says 'Can you cut this out?', it's a nightmare and a lot of money. There Pro Tools is so important. You wanna add eight bars? Repeat something? Do it. The flick of a button." At 72, you'd imagine that the pressures of Arif Mardin's bi-coastal recording career might be beginning to take their toll upon him. But then, hearing just how enthusiastic he is about his current projects, there seems to be little evidence that the producer is thinking about taking life a little easier. "I am slowing down a little bit," he laughs. "My wife doesn't believe it though!" Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Business End

In this article:

Business End

The Uterus MPG evaluation of Readers' Women Published in SOS July 2004 Triviul Slaughterhouse Print article : Close window This Month's People : Miscellaneous MPG Panel

tracks

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'. This month's industry panel is drawn from the MPG (Music Producer's Guild).

The Uterus Women Track 1 Nikolaj Bloch (NB): "I thought this was going to be quite interesting to start 2.3Mb with — the first track seemed quite promising, but I don't think they kept that momentum going. I think the first song could be made to be very interesting, it could be quite fun, but I don't think any of the other songs really grab me. The first song's not especially strong but it's quite fun in an irreverent kind of way. It does have a bit of a hook, which I don't think the other songs have. Some of the intros were quite exciting but then, afterwards, the tracks don't really live up to them. I think the production could be much more exciting but it's fine for a demo — it shows off what it could be and that's what a demo's supposed to do."

Magnus Fiennes (MF): "It's hard to be critical of something that's basically trying to be naive and oddball — the whole strength is that it's supposed to be lo-fi and home-made. They're wearing that well though, he's doing it with aplomb. I mean obviously this could benefit from some sort of sparkling, banging mix, but they're not doing badly as it is. "It's very voguish, I like the combination of influences and his slightly whiny voice; I think the fact that it is slightly whiny and annoying probably helps the thing actually. It's got that early '80s, new-wave delivery. I think they're maybe just slightly too postmodern though, they look like they're just slightly too old to be doing it and it's all a bit knowing, like they've read all the right magazines and that. It doesn't feel dangerous enough, it just feels a bit calculated, a bit too styled." Haydn Bendall (HB): "I like the first track. I think they need to take a few more chances and make it a bit braver though. The concept is nice but it's quite safe and that's not very appealing. All the distortion on some of the tracks makes it sound less honest, like they're putting it on and trying too hard. They say in their letter that they only started doing this this year though, and,

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Business End

that considered, I think it's a promising start." Sam Stubbings (SS): "I think this is all fairly formulaic punk — in the song structures and with what they're singing about. It's all been done so many times before and it's a bit tired. The way they're using synths in some of the later tracks was maybe starting to get somewhere interesting. The guitar is very distorted, I know it's supposed to be like that, but I like it when they take the distortion off and you're left with a more jangly sort of sound, I think the tracks start open up and groove a bit when they do that. I think it'd be interesting to see them live. "I think the thing to remember is that if they're sending their CD in to be reviewed in this session, then they're probably looking for management and a record deal, I think if that is what they are looking for, then they're not going to get it with this as it stands. There's not enough that's different or new or edgy. It just seems a bit lost at the moment. I think if they get going and push it to the extremes of what they're doing then they could make it work."

Triviul HB: "I don't like the first or third tracks on this at all but I love the humour of the second one. It's got a very grandiose feel to it, and that really shows off his vocal style. It reminds me a little bit of the musical fireworks of something like Moulin Rouge — it has that theatrical element to it.

Track 1 2.9Mb

"From a production aspect I think it would have been nice if he'd gone more with the humour of it. I think you could have a huge amount of fun doing this sort of thing, and sometimes it feels like he's taking it a bit too seriously. I think there's a way of doing these things where you can enjoy the humour of it and still be very committed to it. I think you could have a fantastic time in the studio doing the second track. The quality of the production's very high as well, I can see from his letter that he's had experience in professional recording studios and I think that really shows." SS: "I think the production is superb to be honest, I haven't heard anything this clean for ages. The second track is easily the best song of the three; it's got a nice hook and that makes it stand out from the others. I don't really find anything grabbing me with the other tracks but maybe a second or third listen would help that. I found it engaging just because of the quality of the production actually; I love the harmonies and the layered vocals. I like his voice a lot in the second song; he has got some Matt Bellamy in him, especially when he gets to the higher end. When he lets go it sounds very good, the rest of it sometimes feels a bit too controlled, and sometimes it sounds like a piece of production music because it's so clean." NB: "I agree, it doesn't breathe, when he's singing loudly the music doesn't get louder with him. I don't like the first one very much, it doesn't do anything for me, the third one seems to be in a completely different style — more funk and American blues, and I don't think that's where his strength is. "The packaging is very simple, just a standard CDR case and a simple typed letter, but I think that's great — it's got enough information, it's clear and easy to read. A photo might have been nice though. The only thing I would criticise about the letter is that it has a lot of references in it. Maybe that's just me, but when I'm listening to something for the first time I like to be able to make my own references. The other thing that bothers me about it is that I couldn't see any similarity between his music and some of the things he was referencing — Dido and Nelly Furtado for example." SS: "I think the one problem with this, and it's indicative of a lot of people doing music on their own file:///H|/SOS%2004-07/Business%20End.htm (2 of 5)9/22/2005 8:57:18 PM

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these days, is that there are no real dynamics to it. I get that with the things that I write by myself, and as soon as you get a band involved it's so different. It would be great to hear this stuff with live musicians playing it." NB: "I think a live band could work really well with this. It's already got a theatrical element to it and a band could really bring that to life. He mentions in his letter that he's got a classical background, and some string parts or other live instrumentation could really lift this." HB: "If a record company phoned me up, played me the second track and said 'would you like to go into the studio to do this?' I'd say yes immediately. Personally I think he's got enough there to take it further and he doesn't need a band. He does need more than just one good track to get a record deal though, and I just wonder if he's got more stuff like that."

Slaughterhouse Track 1 MF: "The first problem with this is a sonic one — it's just not fat in any way, 2.2Mb it just sounds like soft synths. The beats aren't tough enough, the flow of his rapping is really a bit pedestrian and his diction's not great. I quite like the way that they're embracing a more electro sound with some of the tracks though; the third track is actually quite innovative in some ways. The second track's dreadful, sort of second-rate R&B with a guitar, you've just got to move away from that, it's embarrassing.

"I think it's so difficult to find anyone who's doing anything interesting in UK hip-hop. This doesn't sound like somebody doing their own thing, sometimes original ideas come through but mostly they're just copying American hip-hop. I don't think they understand what the beats need to be doing at all, it's very dissipated across the stereo field and it's very thin. There's just a kind of reediness with this that I associate with soft synths, you really need something like an MPC or an SP12 to get that tight low end. A lot of the time there's too much going on; too many delays going off. That distracts from the main thing, which is the beats. They need to stick at this and hone it down, make it simpler and more direct." HB: "The delayed double-tracking thing is a real mistake. It seems to me that there's a real lack of commitment with this — the instrumentation, the sounds, the vocal, the treatment. It all sounds like they don't really mean it. "On a technical level, I think these guys could really do with getting some decent monitors. There's a bass line on the second track which is really out of control and just swamping everything else. You really do need an accurate view of the bass end when you're doing this kind of music. "The UK does seem to have a real problem with hip-hop, it's really weird because there's some really good French hip-hop out there, there's even good Russian hip-hop. The British seem to find it very difficult though." NB: "In their letter they use the phrase 'tried and tested formula' and they're comparing themselves to American artists. To me the UK music industry seems so fashion-led, one thing happens and everybody else runs after it. I think that's a big problem with UK hip-hop — that it's just aping the Americans. One of the things I really like about hip-hop is that it's one of the only forms of pop music where people can talk about contemporary subjects in their lyrics and actually express something file:///H|/SOS%2004-07/Business%20End.htm (3 of 5)9/22/2005 8:57:18 PM

Business End

original — obviously that's not going to happen if you're just copying someone else's style and ideas. I think they really need to try harder not to follow formulas, just doing what they want would give them a much better chance of creating something new and interesting. SS: "I think the first track on this is really good. I haven't heard any good UK hip-hop that's based on American hip-hop, it only gets interesting when you get people like Dizzee Rascal and The Streets who are doing their own thing and not trying to emulate the Americans. I think the first track really has potential though, I know people who would absolutely love that. "I think the production is very amateur, it doesn't groove and it isn't hard enough. The vocal mix on the first track is much better than on the others, it seems to sit much better within the rest of the mix. I think if they were to carry on in the same vein as the first track then they could do alright, they just need to keep trying to develop their own ideas and be original."

This Month's MPG Panel

After working as a piano tuner for Steinway & Sons, Haydn Bendall was employed at Abbey Road studios for over 17 years, including 10 as senior engineer. in addition to working with artists such as Fleetwood Mac, George Martin, Elton John, Damon Albarn and Hans Zimmer, Haydn has collaborated on several musicals with Eric Woolfson and has made extensive recordings with all the major London orchestras. Today, Haydn is involved with Schtum Ltd and The Firebird Suite. Sam Stubbings is the Senior Producer for the DVD division of Metropolis. He began his career five years ago at Abbey Road and has since worked with artists ranging from Paul McCartney to Muse. More recently he has produced both the first DVD single (Bjork's 'All Is Full Of Love') and the first commercial DVD-Audio disc (Holst's The Planets). He also has his own act, Redstar, who are currently recording an album and gigging in London. Nikolaj Bloch is a freelance engineer, writer and programmer with almost 20 years' experience in the music business. As the guitarist in the band Subcircus, he played all over the world until their split in 2000. Since then he has worked as a programmer and soloist on several major Hollywood films. He has also written for a range of artists varied enough to include American country singers and Jimmy Somerville. He enjoys spending time writing and collaborating in Nashville throughout the year. Magnus Fiennes is a composer, producer and songwriter. His production career has taken in a diverse range of artists and bands, from the Spice Girls and All Saints to Pulp and Tom Jones. Recent co-writing credits include Massive Attack, Kylie Minogue, Sugababes and Liberty X. He has also composed several film scores and written for numerous TV shows and commercials. Many thanks to The Firebird Suite who hosted the session. Their web site is at www.thefirebirdsuite.com. The MPG's web site is at www.mpg.org.uk. Published in SOS July 2004


Business End




Crosstalk

In this article:

Professional Pride Garage Band Has A Fan The Price Is Right!

Crosstalk Your correspondence Published in SOS July 2004 Print article : Close window

People

Professional Pride In response to Chris Marsh's letter in SOS May 2004, it is disappointing to hear yet another horror story of budget recording sessions gone badly wrong. From working day in, day out with musicians, writers and producers both amateur and professional over 17 years I can truly empathise with how soul-destroying a botched session can be for both artist and engineer. But to suggest that professionals are not keen to complete their work to the highest standards possible, being interested only in separating their clients from their money, is idiotic, antagonistic and wrong. My initial suggestion to Chris is to look in his record collection, check the credits of his favourite 50 recordings and see how many of them were executed in professional studios, by dedicated (and evidently talented) people exchanging their time and skills for a living. The evidence will speak for itself. As for aligning the craft of the professional engineer with that of the witch doctor — their only power coming from the ignorance of those they rule — I'm afraid this simply doesn't wash. Anyone who cannot provide consistently reliable results will not last in a position of responsibility, and none more so than the engineer whose gaffes are recorded on tape as permanent evidence for all to hear. Also, as is borne out by the vast majority of the interviews in SOS, a good engineer is never afraid of passing his/her knowledge on to the 'ignorant'. Why? Well, contrary to what Chris thinks, recording is a black art, so much so that a technique that achieves the magic you and I aspire to capture is not replicable with the next artist, the next studio, the next piece of equipment or even the next day. Greatness is achieved from the chemistry between those present, the place and the moment. It comes when you aren't expecting it and it largely eludes you when you are. And it rarely comes to those who have not put in the blood, sweat and tears it takes to make themselves bloody good at their craft, and, in this field, that means professionals. Unfortunately Chris, if you did not employ a producer, you only have yourself to

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blame for your woes. In the sub £500-per-day range, it's a minefield, but at the end of the day you were the one who chose to use the studios you did. And without that producer, you alone are responsible for directing your engineer. It sounds like your big mistake was not asking to hear records the engineer/ studio had made, not checking references from their previous clients, not spending enough time establishing the ability of those involved to fulfil your remit within your means, nor having the nous, experience and commitment to shop around until you found a suitable place. Whilst I can sympathise, don't use your obviously limited experiences as the basis upon which to denigrate my profession. Joe Leach

Garage Band Has A Fan After reading Mark Wherry's Apple Notes column about Apple's iLife software in April's SOS, I was compelled to put pen to paper. A fortnight ago I was lucky enough to take delivery of a 12-inch G4 Powerbook to replace my ageing G3 — until then I hadn't considered upgrading my copy of Logic. After learning what the Logic upgrade would cost me, I was disgruntled to say the least. However, as I spend more time with the new Powerbook, I am becoming less and less concerned about upgrading. This is purely down to Garage Band. For experienced Logic users, Garage Garage Band — for some readers, it's good enough to rival Logic. Band is a breeze, as it has adopted all of Logic's GUI conventions — it 'feels' like Logic. Within an hour, I was completely at home running my collection of Audio Units synths and effects — I didn't need to refer to the help pages once. Additionally, as a Final Cut Pro 4 user, if I really feel the need, I can bounce individual tracks out of Garage Band and apply further Emagic effects in Soundtrack [Apple's audio-for-video application, included with Final Cut Pro 4 — Ed]. Whilst I realise that some musicians will consider the Garage Band feature-set to be too restrictive, I think it's more than capable of cutting it in a professional situation.


Crosstalk

And the best bit... it's totally free [with the latest Macs, that is — Ed]. Who needs Logic any more? Not me! Pete Gill

The Price Is Right! Simon Price is bang-on with his views on GUI design [Sounding Off, SOS May 2004]. As someone who started MIDI sequencing with Cubase Score on a Macintosh LCIII, I have watched as the studio has slowly transformed from the actual to the virtual. About two years ago, I was messing around with Cubase VST when I realised that I hadn't been able to finish a tune in over a year. Worse than that, I was suffering from repetitive strain injury and was beginning to lose my eyesight! The tactile studio space that had once surrounded me had receded all the way into a small square (the monitor) in front of me. And it was a total mess, with each window partially overlapping the next, like a pile of lecture notes in a student flat. All this jerking around with the mouse in order to locate windows and make small knob adjustments had allowed RSI to set in, and, on top of this, I'd got into the habit of squinting into a fixed distance and hardly ever blinked. This was straining my eyes and causing them to dry out. I was temporarily banned from mouse and CPU work by my specialist, ultimately leading to the loss of my day job as a sound designer. Then one day I was introduced to an early version of Ableton Live, which a colleague of mine (Stefan Robbers) was beta-testing at the time. I thought the context-senstitive interface was amazing. Clear mixer and arrange views, no cluttered menus, simple graphics and no damn pop-up windows (unless you want to use a VST plug-in's GUI). And the price was good too! To date, the only other software interface that has impressed me like this has been that of Mackie's Tracktion. I now work with a 17-inch Powerbook and an additional monitor, leaving me free space for the odd Reason sequence or Spark session running alongside Live. Any hardware synths get sequenced using a Yamaha RS7000 and are recorded into Live. The only other outboard gear I have now is a MOTU 828 interface, a TC Triple C processor and a Mackie 1402VLZ mixer for monitoring. This simple little setup is portable, and has allowed me to do the best work I have ever done. Also, I no longer suffer from double vision or arm and shoulder pain, because this setup has broken those bad habits.


Crosstalk

When I think about Simon's examples of bad GUI designs, I not only agree, I shudder. Long live Ableton, and all those who follow. Rob Bantin Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Learning Difficulties

Learning Difficulties Leader Published in SOS July 2004 Print article : Close window

People : Industry/Music Biz

Over the past couple of years, I've visited quite a few schools and colleges running music-technology courses, and I've served as an external examiner for one university course. What I've discovered is that on pretty much all of these courses, regardless of the educational establishment involved, some students consistently don't show up for lectures, some don't hand in work on time (or at all) and some seem to be on the course only because they think music technology is an easy option. It seems that everyone either wants to be a recording engineer or a record producer (mainly for dance or hip-hop), but few really have much of an idea how tough it is being a studio engineer, or of how few opportunities there are for new producers. Come to think of it, few realise how many tens of thousands of students are currently on courses aiming for what are, realistically speaking, no more than a couple of dozen serious studio engineering vacancies per year. And at every Q&A session that Hugh Robjohns and I hold, we're asked the same question — what's the best way to get a job in a studio? To answer that, I think it's easier to identify the people who won't make it. If you're on a music-technology course and you miss a lecture or practical session without being close to death's door, then this isn't the career for you. If you don't make a serious stab at every piece of coursework and hand it in on time, you should consider a different career path. And if you think you can get good grades by copying and pasting chunks of information from the Internet without actually understanding it, forget it, because both your lecturers and prospective employers will catch on soon enough. And most importantly, if anyone can drag you out of the college studio without leaving a trail of bloodied fingernails embedded in the furniture behind you, you're just not made of 'the right stuff'. The successful student will spend every possible moment in the college studio, will trade food, cash and body parts for anybody else's spare studio time and will talk about little else other than recording to the extent that all their non-musical friends will lose the will to live. The same person will walk into their friends' homes or into record stores and notice instantly if the speakers are out of phase, and regardless of their own musical interests, will absorb any information on every aspect of recording. This same person will go home from college and, barely pausing for food, will boot up his or her computer and try to put some of this new information into practice. The main bathroom reading will be Sound On Sound, and our candidate will know all about the technical issues associated with recording as well as the artistic ones. Assuming the above conditions are met, the successful applicant will also need extremely good peoplehandling skills and exceptional stamina. Musicians are notoriously volatile, especially if things aren't going well in the studio, and you're the one who's going to have to keep everyone calm and keep the session moving along, even if you've been working for 12 hours without a break. This probably narrows the field down to a few

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Learning Difficulties

hundred candidates, after which you still need an element of luck to be in the right place at the right time. Nobody said it would be easy, and nobody can guarantee you'll like the job if you do manage to get it, but at least you've been warned. If in any doubt at all, then keep recording as a hobby and find another career — apparently good gas fitters and electricians are in short supply these days, and they earn rather more than the vast majority of studio engineers! Paul White Editor In Chief Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Readerzone

In this article:

Beginnings The Big Step Important As Your Fridge The Studio The Good, The Bad & The Ugly Main Equipment Working Methods Chunc Projects The Future

Readerzone Shaun Hayward & Joe Kenny Published in SOS July 2004 Print article : Close window

People : ReaderZone

Debbie Poyser Photography: Jyoti Mishra Joe Kenny (left) and Shaun Hayward.

Sometimes hi-tech music can be pretty hard work. Creating a studio is expensive and time consuming — choosing the gear from a confusing array on the market, getting the necessary money together, fitting out a suitable space, headscratching over wiring and setup, then spending possibly months becoming properly familiar with hardware that's sometimes inscrutable and software that's nearly always deep. And all this before you can get down to serious music production. Shaun Hayward and Joe Kenny, aka Chunc Productions, recognise this catalogue of potential difficulties. Three and a half years ago they got together to find a house where they could begin creating a studio. The setup would be used to record their own album and would also serve as the engine room for their production partnership with other bands and artists. There was one tiny extra obstacle: both Shaun and Joe are blind. Joe lost his sight at five years old, after an unsuccessful operation for congenital Glaucoma, while Shaun's sight failed progessively into his 20s, due to another genetic condition, Retinitis Pigmentosa. If you've been (rightly) patting yourself on the back for having slogged through the process described above and come out the other end with a working setup, think for a minute about how much harder it would be if you couldn't see what you were doing. It's tempting to think it would be impossible, but Shaun and Joe have proved that assumption wrong. Not only do they have a very nice basement studio centred around MOTU 24 I/O audio hardware, Sonar software for recording and a Soundcraft Ghost analogue desk, they are indeed part-way through recording their album. What's more, they've been working with two separate vocalists, one of whom will be the subject of a Channel 4 documentary this summer, and they also act as producers for local bands who take their fancy.

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Beginnings While Joe and Shaun met while studying music technology at a Hereford college for the blind, their routes to that point were very different. Joe was fresh from school, having decided that music and not 'A'-levels was the way to go. He'd played guitar since he was a child, adopting a unique 'left-hand-over-the-top' style because when he first picked up the instrument that seemed the way to play it. Shaun, on the other hand, had spent years playing solo and in bands in the Birmingham area, majoring in vocals and keys. At Hereford, Shaun and Joe both became inspired by lecturer Paul Cobbold (who had previously worked with the Pogues and Clannad, amongst others). Both were aiming for a career in the audio industry, and while Joe went off to continue studying at Leeds Metropolitan University and then Leeds College of Music, coming out with an HND in Music Production, Shaun landed a job engineering at Forge Studios in Oswestry, at the same time playing piano and singing at a local restaurant. The studio provided valuable practical experience, although, as Shaun relates, learning the gear without being able to see it was initially "very stressful. They had an 80-input Amek Rembrandt desk and Bantam patchbay. I was used to a normal patchbay, with holes in horizontal rows, and when you go Bantam they're sometimes in vertical rows. Bantams have much smaller holes too. It's not a job I'd like to do now, because it does stress you out. I wasn't only engineering, I was producing, keeping people happy, smiling, telling jokes, and at the same time I'd be thinking 'where the hell is that wire going?' Sometimes I got a tape op, sometimes not. When I did I'd send him to read the DA88 meters and tell me if we were in the red yet. Otherwise I'd use these [indicates ears]. I'd do a little test run and as soon as I heard any distortion, pull it back." He seems to have managed admirably well, no doubt helped by good ears and a phenomenal memory — desk navigation had to be done entirely by touch, with layout and channel assignments memorised — and became a valued part of Forge. "Lee Monteverdi, who used to work with Pete Waterman, would come in and do a lot of mixes where I'd worked on the tracking. We did some sessions where Phil [Forge owner] would pay me just to come in and sit at the back of the room and listen, to say if I felt something was happening that was wrong, if they were pushing a compressor too hard, or something needed lowering in the mix, or a hi-hat frequency was too bright against the rest of the track. Just to get it all to blend. A lot of vocal production, too, because singing is my main thing."


Readerzone

The Big Step While working at Forge, Shaun kept in touch with Joe and would frequently visit Leeds, where Joe was at University making a discovery of particular importance for the future prospects of the partnership. He explains. "That was when I first got into operating the Cakewalk 6 software. The speech software we use [called Jaws, by American developers Freedom Scientific] worked with Cakewalk 6. "A company called Musicians In Focus had a plan to bring this speech software to blind people who wanted to record their own music. For me, it really was an eye-opener. Before that I'd been using Tascam DA88s, reel-toreels, ADATs... I could use them, but you could never read what was on the screen, and if you got it wrong, if you hadn't un-armed a track or whatever, there was no second chance. But as A view of the Chunc Productions basement studio. soon as you get onto a sequencer you can do anything anyone else can do. When I came across it, I thought... 'That's it!' "The basic function of the speech software is to read what's on the screen and speak it to you. It's made for Windows and it works with common Windows apps, like Microsoft Word and Excel, and Internet Explorer. But then it started to be developed for Cakewalk. Little tweaks have to be made to optimise it for different software. You have to tell it how it's supposed to look at the screen — for example, whether it's meant to read graphics." Shaun takes up the story. Having heard from Joe about using Cakewalk with the speech interface his first thought was: "This is great, we could work together and be the producers we really want to be, get on a level playing field with everybody else. So I said let's look for a property and put a studio in it, and Joe was really up for it. From there I went around buying stuff. I bought the desk, he had a PC, I already had my Roland keyboard, so we started from there. We found this house. The cellar was absolutely minging, so we spent £3500 on tanking it out. Luckily the landlord gave us £1300 towards that. Everyone kept saying we were mad, putting money into a place we didn't own, but you have to follow your dream. "When we first got Cakewalk going, that first year, we were making mental sounds. We were so excited to be able to make grooves ourselves. A lot of stuff we still couldn't do, like the MIDI wouldn't work properly, so we taught ourselves to play everything in by hand and copy and paste, and that's how we got our unique sound.


Readerzone

"We don't use commercial loops. We make our own, drumming them in. Joe has his own style entirely and he gets sounds out of the guitar that no-one else can get. As for me, I was taught on Hammond organs so in a way I'm self-taught on the piano and I use a lot of bass end and chords down the bottom. Since we've been here we've pursued originality, trying to keep going our own way." As Cakewalk advanced, Joe and Shaun moved on with it through its mutation into Sonar, aided by Jaws 'scripts' written by an American programmer that enable the speech software to read what's in Sonar. Joe: "Cakewalk communicate with people who are prepared to write the scripts. Sonar was a bugger for script writing. Apparently it looks a lot more like VST than the previous Cakewalk software, very graphical, so it was harder to write the scripts. But people have kept up with it, thank God."

Important As Your Fridge Standing quietly in a corner by the racks I notice an Alesis Air FX mounted on a mic stand. Shaun and Joe are immediately enthusiastic: "That's the best bit of kit in the house," says Joe. "It's awesome. A lot of studios won't have them, because of the quality, but sometimes quality is irrelevant, it's about getting a vibe. That's got 50 different effects on it. Some of them are useless, but some of them are great — the filters, for example. You've got four points of infra-red rays and you move your hands about and you can produce effects you'd never be able to produce with just two knobs, make shapes with your hands and get some great sounds going. We usually put it through a Penta set for limiting so that it doesn't crash too much, and mess around. We use it on drums." "And it's good for intros and outros," adds Joe. "Put a stereo tune through it and mess it up." Shaun continues: "We had one guy in here and he played with it and then he turned around and said 'that is as important as your fridge!'."

The Studio The basement studio at Chunc HQ in Leeds is a cosy and comfortable little space, although it can get rather warm in summer. There's a portable airconditioning unit vented through the back wall, but it's too noisy to use during recording or critical listening. The walls are mainly covered in thin carpet, though the back wall is wood panelled and the wall behind the desk has a pleated curtain in front of it. The main building work was done by a local firm, but Joe and Shaun did much of the rest themselves. They also undertook the wiring (balanced throughout), including the patching from the patchbays to the desk, all using touch and memory. Shaun explains: "My Dad made all the looms and we bought one of those £10 testers from Tandy. We'd pick up one end of a lead, put the tester on it and keep picking up leads until the tester went 'bleep'. Then we'd know we had the other end. Then we taped numbers to the leads, in braille, and we kept doing that until we'd finished all the looms." file:///H|/SOS%2004-07/Readerzone.htm (4 of 10)9/22/2005 8:57:27 PM

Readerzone

Choice of gear came down to the usual balancing act between needs and finance. Shaun was determined that although they would be recording digitally, to hard disk via Cakewalk/ Sonar, the desk must be analogue. "We use a mixture of analogue and digital. Analogue desk, analogue outboard, and digital hard disk recording. That is the best way to go. When you hear bands recorded via digital desks, they're very clean. It's great for dance music, but for real bands there's no warmth."

The rack patching took days to wire, but now gives full and easy access to the outboard processors.

The assignable nature of digital desks doesn't go down well with Joe and Shaun, either. Joe explains: "I had a bit of experience with the Panasonic DA7 and I thought it was a horrible console to use. You've got three knobs and a button that changes what the knobs do — ie. top, middle or bottom EQ bands, or they're aux sends. That's too many things for three knobs to do, if you ask me." Moving on to digital recording, the pair have nothing but praise for the MOTU 24 I/ O they chose as their audio interface, with Joe observing that "MOTU are the best. The quality and dynamic range are just great. We get such a fat sound recording at 24-bit, 96kHz." Shaun adds: "A lot of our old stuff was done at 44.1, but now we record at 96k. You can really hear it on things like acoustic guitars." In answer to my question about what exactly is added to the sound by the extra bits and sample depth, the pair have an interesting viewpoint. "It's very hard to describe," responds Joe. "It's a height thing." Shaun enlarges on this statement. "Nobody else seems to have figured this out. If you have a sound in front of you so that the vocal is there in the centre [indicates something placed at around chest height], at 96k it will be up here a bit [moves hand up towards head height] — a bit bigger. If that transfers 24 times you're going to get a much fatter sound. Unfortunately, with Sonar 2 you can't mix sample rates in one file. With Sonar 3 you can, so we really want to get that. On the tracks we haven't finished for our own album it would be great to be able to re-use old material, as well as recording new stuff for the same songs at 96k."

The Good, The Bad & The Ugly The studio's two well-stocked racks contain some great favourites, such as two Focusrite Penta preamp/compressors. "They are brilliant," says Joe. Shaun agrees. "Their mic preamps are fantastic. We've got a Neumann TLM103 mic that we used to put straight into the desk and it was a bit naff, but you plug it in to a Penta and it's clear, in focus, warm, full. Plug any mic in and it helps. file:///H|/SOS%2004-07/Readerzone.htm (5 of 10)9/22/2005 8:57:27 PM

Readerzone

"The presets are on little individual switches in a row with their own LEDs, and we manage to work with those just by counting. For example, number nine is one of the acoustic guitar presets. Then if the preset is not quite right you edit it. But there are no rules... We'll sit there and hit the buttons till we find the right sound. Maybe it's a kick drum preset and a guitar is going through it, but that doesn't matter if it's the sound you want for that moment." Not all the rack gear comes in for the same praise. Joe: "The TC Triple C is hopeless for a blind user, because it's so menu-driven. There are 100-odd presets, but we can't get to them, because of the menus." "We bought that because it's a multi-band compressor, the only 'finaliser' we could afford," adds Shaun. "We've got a friend who we sat and worked with for a while, getting our own presets Joe's unique guitar style evolved because as a blind person he had no preconceptions together, and when we want to master about how to play the instrument. something we ask him to access a certain preset for us. The other thing with the Triple C is that the knobs don't click and they don't have a pointer you can feel." "And they're very light, which means that the slightest touch makes them move," adds Joe. "I'm sure even sighted people have had the same problem," says Shaun. "They brush against something and 'oh shit...' it's changed in the middle of a mix." A strong presence in the racks is Behringer: Shaun and Joe own no fewer than seven Behringer units, about which they have mixed feelings. The two Behringer preamps, Ultragain Pro Tube models, see perhaps the most action. Joe: "I wouldn't use them for a nice vocal or a nice piano, but if you bang a full-distortion guitar solo through them... bloody great!" Joe also likes the Ultra Q Tube EQ, primarily for basses and guitars. "It generates warmth. You put that on a very clean sound and the warmth becomes noise, but on a bass it's just warmth." The Behringer gear clearly has its uses, then, though Shaun calls the Ultrafex Pro enhancer "the worst buy I ever made" and freely admits that he finds the Intelligate so inscrutable that he's only turned it on once! Well designed knob-driven interfaces are a favourite with Joe and Shaun — as they surely are with the majority of SOS readers. They particularly like their Lexicon MPX100 effects processor — although, as Shaun reveals, "we only use that for delay. We don't use the reverbs because they're not very good. But it's got tap tempo and is very tactile. It would be brilliant if Lexicon came up with something that used an engine from one of their really high quality machines, and stuck it in a box with that interface. If you could get the real quality with those knobs and that layout, it would be fantastic. Those two knobs in the middle where you can extend stuff and change it... it's so instant. But I would think a lot of people wouldn't want that. They'd want the menus to mess around with." I


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observe at this point that I'm not so sure they would! Monitoring is taken care of by Tannoy Reveals and a pair of tiny PC speakers. "The Reveals are very good, you can hear everything with them," offers Joe. "The bass they can handle never ceases to amaze me." Shaun adds that "in this small room you get a 'main monitor' sound out of them. They're fantastic workhorses. And for nearfields we use those PC speakers. If you can make it sound good on there and it isn't cracking up — because the top end cracks up on them — it'll sound good anywhere. We try to match between the Reveals and the PC speakers. If we can be flicking between the two — and obviously there will be a bass disparity because the Reveals are much bigger — and there's not that much of a difference in the sound, we know we're on to something."

Main Equipment RECORDING Altec Lansing PC Speakers MOTU 24 I/O Audio Interface and PCI 424 card Neumann TLM103 Microphone Pentium 4 2GHz PC with 1GB RAM, 2 x 40GB drives, Seagate 120GB drive Pioneer DVD writer Samson Power Amp Soundcraft Ghost 32:8 Analogue Mixer Tannoy Reveal Monitors

OUTBOARD Alesis Air FX Processor Aphex 204 Aural Exciter Behringer Intelligate Noise Gate Behringer Tube Composer Compressors x2 Behringer Ultragain Tube Pro Preamps x 2 Behringer Ultra Q Tube EQ Behringer Virtualizer Multi-effects Dbx 266 XL Compressors x2

Studio sound sources include the Roland RD500 digital piano and selected rack modules.

Focusrite Platinum Penta Compressors x2 Focusrite Platinum Compounder Compressor Lexicon MPX100 Effects Processor Line 6 Pod Modelling Guitar Preamp/Processor Line 6 Bass Pod Modelling Preamp/Processor Midiman 8x8 MIDI Interface


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TC Electronic Triple C Multiband Compressor TC Electronic M1 Effects Processor

SOUND SOURCES Emu Proteus World module Emu Proteus 2000 module Evolution MK361C Controller Keyboard Line 6 Guitar Pod and Bass Pod Processors Novation A-Station module Roland RD500 Digital Piano

Working Methods To get an idea of how Shaun and Joe operate in the studio, I ask them to talk me through a typical working session. Shaun begins. "If we take, for example, a fourpiece band, drums, bass, guitar, singer, the first thing is to get the drummer set up in the booth and mic up the kit." The miking up is done by touch, as Joe confirms. "It's just experience, you get to know where to point the mics and how far away and so on." The drums are then routed through preamps to the desk. Joe explains. "The typical thing would be to put the kick through a Penta. The overheads, too, would go through a Penta, and we could then use its limiting. It's imperative you get a good overhead sound and a good kick. The hi-hat and the snare would go through the Behringer preamps, because sometimes the cracking sound of the valve works quite well on a snare. The Behringers are a bit noisy but they're ballsy. They'd come up at the desk, we'd route them to the groups, they go straight to the MOTU, we'd open up Sonar and select the inputs and arm tracks for record. Then we'd plug the bass and guitar into the Pods, route them, and so on, then give the vocalist a mic to hold, which we usually don't record." "At this point it's all about getting a good drum track," says Shaun. And getting the best from the players involves more than just the gear. Psychology comes into play too. "Half the time, on that first pass we don't even record the bass and guitar. Nine times out of 10 they don't play the part right. People come in and say they want to do it live, for that live feel, but every band we've ever produced in here has left with a great live feel, because of the way we work. We motivate, we encourage, we turn up the monitors and really get the player into it. We're listening for feel all the time, to make sure that energy is created. We like to work with people as individuals, so once we've got that drum track nailed, the session changes. Now it's 'everybody out, and where's the bass player?' It's his time then, and we want to make him feel important. There's a lot of politics in bands, so the best way psychologically is to get each member on their own, make them feel comfortable and safe and tell them they're doing great. In the end, that's what makes the better product from them. They've not got the worry of the guitarist giving them a look that says 'you made that mistake again'... We send the others upstairs to make tea and watch a DVD. Then it's the guitarist's turn. It does seem to be the best way to work" file:///H|/SOS%2004-07/Readerzone.htm (8 of 10)9/22/2005 8:57:27 PM

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While Shaun and Joe are making each musician feel comfortable they're also taking care of business with Sonar, routing inputs where they want them, arming tracks for recording, and so on. Joe explains how it's done. "We've got a template we use that opens up 24 tracks ready to go. We use the arrow keys to get around. Imagine a grid: up and down is going between tracks, left to right is going through the columns in that track: Mute, Record, Solo, Pan, Volume, Input, Output... I don't know if that's how it looks on screen, but to us it appears mentally as a grid system." Every time the cursor reaches a significant point, a computerised voice gives audio feedback. For example, if Joe arrows through a track strip, the voice might read out something like 'Mute off, Solo off, Record off'. Joe: It'll give us feedback at every keystroke. If I'm on a Pan button and I want to pan something, the voice interface says 'Pan', I hit Enter, type in my new value, hit Enter again — and there's the pan position changed for that track." Once everyone has finished track-laying, one of the most time-consuming parts of the process begins: editing. The pair clean up guitar tracks, chop breaths and coughs out of vocal tracks, and so on. Joe: "It's all based on your 'from' and 'to' time. I might select from bar two to bar four, because I know I'm going to do something with that space — cut it or put a plug-in on it, for example. That's the way it works." If a band has elected to record without a click track, this system of using bars becomes more difficult and entails Shaun and Joe doing some careful listening and remembering of edit locations. Fortunately, the speech interface gives audio feedback of bar/beat/tick positions when asked to. Not being able to see to use onscreen wave editing means that the process is slower, but it gets done. Shaun points out that "music is a listening, not a seeing medium. I've known people to edit badly, thinking they can see everything on screen, but I can still hear little clicks, because they've edited with their eyes. People are so caught up in seeing what they're doing rather than listening." Next up is the mix, which is largely a hands-on process at the desk, although Shaun and Joe will sometimes use some Sonar level automation to lighten the mixing load. They swap roles constantly at this point. "We have similar skills with mixing," says Joe. "Shaun will be at the desk and I'll be wandering around behind him, going 'louder', 'quieter', and so on, or I'll be here adjusting the compressors... or vice versa." "That's the beauty of me and Joe as a team," adds Shaun. "There's differences in terms of our personalities, but when it comes to working we both play drums, bass, guitar and piano, to different degrees and in different styles, and we both can use all of the equipment at the same level. So we share responsibility. We trust each other. If he says 'that kick's too loud,' I reduce it straight away. And he'll do the same with me." Some tracks are then 'finalised' using the TC Triple C, sent out of the audio hardware, through the patchbay into the Triple C, then back to the MOTU again, without going through the desk, before being topped, tailed and faded (if necessary) in Sonar. Shaun comments that they'd really prefer to have someone else master for them. "It would be great to get some help along the way, and someone else's perspective." file:///H|/SOS%2004-07/Readerzone.htm (9 of 10)9/22/2005 8:57:27 PM

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Chunc Projects The Chunc Productions website, www.chunc.net, gives details of Shaun and Joe's projects, including work with singers Sarah Caltieri and Joanna Laporte. Sarah, also an actress and featured in a Channel 4 documentary this summer, is using the two tracks she completed with Joe and Shaun to seek a publishing deal. The pair have now returned to their own album project. Shaun: "That's why we set up the studio. I think our music fills a gap in the market. We listen to a lot of rap — we don't use rap but we're influenced by the beats. The music is very melody-driven pop, but it isn't pop. It's not exactly political, but it says something about life and the way we are as people. It's kind of soulful punk."

The Future Next on the agenda for Shaun and Joe is finishing their album, but at the same time they'll be continuing to pursue their aim of establishing themselves as a production team. They've already made a good start, laying the groundwork of creating a studio, making contact with other artists, and refining their working methods. Shaun is matter-of-fact about the determined attitude that's allowed them to get so far already. "If there's a problem, we work around it. We have a saying here: there are no problems, only solutions." Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Studio SOS

In this article:

First Things First Vocal & Guitar Miking Studio Ergonomics The Return Visit Fixing The Mixes Peter's Comments All's Well That Ends Well

Studio SOS Peter May Published in SOS July 2004 Print article : Close window

People : Studio SOS

The SOS team help Peter May to brush up his drum sounds and put more life into his mixes. Paul White

Although the toms and snare sounded acceptable on the first visit, on his return Paul optimised the sound with some careful tuning and damping.

Peter May's studio occupies a room at the rear end of the ground floor of his house in a space known as 'the void', a room half set into the ground, which certainly helps with the sound isolation. Peter is a school teacher, but was fortunate earlier in his life to get a teacher placement at Abbey Road studios, which gave him the bug for recording and lead him to question his choice of career! Having worked his way up through TEAC four- and eight-track machines, Peter has been recording since the late '70s, but has now moved over to a computerbased system comprising a Windows machine running Steinberg's Cubase SX. file:///H|/SOS%2004-07/Studio%20SOS.htm (1 of 9)9/22/2005 8:57:30 PM

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He has also recently invested in a new monitoring system comprising Genelec 1029As with a matching subwoofer. Although at the time of our first visit he still used his Soundcraft Spirit powered mixer for monitoring his computer mixes and for playing back CDs and so on, recording was done through a recently acquired Allen & Heath GS3 analogue console, which feeds his 10-input Terratec audio interface. The room itself is separated into two areas by a supporting buttress, and all the walls have been covered using thin wood-effect panelling on battens with lightweight rockwool behind. Although this makes the surfaces very reflective, the sheer amount of stuff in the studio provides a lot of dispersion to keep this nicely under control, and the panelling (more by accident than design) seems to act as quite effective bass trapping. Peter was having trouble getting a good sound from a live drum kit, which he had been recording with just four mics — kick, snare, and two overheads. As he'd recently bought a set of AKG clip-on drum mics, he wanted to explore ways of using these more effectively. He was also having some problems with his mixes sounding lifeless.

First Things First The first item on our agenda was to check out the monitoring system. This was not set up symmetrically, nor would it have been practical to do so because of the shape of the room, but we heard a problem with the lower midrange that wasn't attributable to this. A quick check around the back of the 1029As revealed that Peter hadn't set the DIP switches on the back of the speakers for use with a subwoofer, so Peter had been positioning his overhead mics much too close to the kit, so Paul and the necessary low-end roll-off wasn't Hugh moved them higher to get a more happening. Hugh adjusted the DIP cohesive sound. However, the AKG C1000s switches with the aid of a long thin Peter had selected for the task were rather screwdriver and, once they were dull-sounding, so Paul recommended correctly orientated, the sound replacing them with a pair of the new generation of inexpensive condenser mics to improved noticeably. Playing a track get a brighter sound. with a busy bass line showed a few small subjective discrepancies in bass level between notes, but they were not nearly as severe as we expected in this room, so we decided to leave the subwoofer where it was (beneath the table supporting the console) and press on. I offered to tune the Premier drum kit lurking at the other side of the buttress, but Peter rather guiltily confessed that he'd lost the tuning key. Fortunately, the snare file:///H|/SOS%2004-07/Studio%20SOS.htm (2 of 9)9/22/2005 8:57:30 PM

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and toms were fairly well tuned anyway and I only needed to apply a small pad of tissue (using masking tape in this instance, though gaffer tape is better) to the upper and lower tom heads to tame the ring. The kick drum was a different matter, as the head was too worn to get any depth in the tone, but I retuned it as best as I could (via the hand-operated tuning lugs), replaced the lightweight pillow inside the drum shell with a heavier sleeping bag (a thick blanket would have been better still), and taped a plastic card to the head to try to get more snap from the felt beater. As a rule, wooden or plastic beaters give better definition to the tone for rock and pop music. Peter had been setting up the AKG kick mic around a foot in front of the drum, so we repositioned it inside the shell on a boom arm and then experimented with the fine positioning to get the best tone we could. The outcome was a sound with much better definition, but still without any real depth — we weren't getting any depth acoustically either, so a new head has been added to Peter's shopping list. After raising the AKG C1000 overheads to a more suitable positioning, I played some particularly sloppy drum rhythms so that Hugh could make a test recording on Cubase SX. One of the clip-on mics was used for the snare, but we recommended that Peter experiment using his other clip-on mics on the toms after listening to the test recording. The overall sound wasn't bad, but close-miking the toms gives you more control over balance and also gives a more solid sound that's well-suited to rock and pop work.

Peter's original kick mic positioning several inches outside the body of the drum didn't give nearly as much definition to the recorded sound as miking close to the batter head inside the drum.

After recording, we used the gate in Cubase to clean up the kick drum, and also applied some EQ. A boost at 80Hz accentuated the (sadly underwhelming) thump of the drum, and a little more boost at around 4kHz emphasised the smack of the beater. A further dip at 150-200Hz stopped the sound becoming boxy. It is worth noting that drums recorded in small or dead rooms won't sound great unless the right reverb is added, and in many cases the reverb plug-ins that come with most sequencers don't really cut it. What you really need is not a washy or splashy effect, but the sense of being in a real room. The newer convolution reverbs do an excellent job of this, but they are very processor hungry. The best settings to use tend to be ambience programs or responses

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taken in real studio rooms, and the host-powered Cubase SX reverb wasn't really doing the job. Peter said he was in the market for a better 'in-computer' reverb, and as his studio was almost midway between where Hugh and I live, we agreed to a second visit after he'd made any recommended changes and added some new equipment. I told Peter that I liked the reverbs that came with TC's Powercore, and as the Powercore Element card is now so cheap he said he was seriously considering buying one.

Vocal & Guitar Miking Early on we noticed a Rode NT1 mic set up on a boom stand with a pop shield positioned over the end, rather than at the side — the NT1 is a sideentry mic, so the end of the mic is 90 degrees off axis. When we asked Peter about this, he said that he recorded this way because when he had tried singing into the side of the mic it had produced a dull sound. It transpired that he'd been using the back of the Paul and Hugh quickly noticed Peter's unusual vocal-miking setup! Given that the mic rather than the front, which is NT1 is a side-firing mic, a much clearer perhaps understandable when you've sound was immediately obtained by having been used to dynamic mics all your the singer address the correct side of the life, where there is only one obvious mic, as marked with the gold dot. business end. In fact, the Rode mics use a gold stud to denote the correct side to yell into, and as soon as this was explained to Peter, he had no further problems. Nevertheless, as the room had so many reflective surfaces, we did suggest hanging a heavy duvet behind the singer when recording vocals, as this would help soak up any excessive mid-range and high-end liveness. Peter had also tried using this same mic for recording the acoustic guitar, but with disappointing results. He had felt that perhaps he should be recording in a more dead-sounding area, but I thought the studio room was fine and might even benefit from a sheet of hardboard or MDF on the floor (beneath the guitar) while recording, so as to reflect some sound back up to the mic. We set him up playing his Takamine acoustic guitar and got him to listen to the sound through headphones as I moved the mic around. Peter was surprised at just how much the sound changed with different mic positions and the best sound we achieved was with the mic a couple of feet away looking in the direction of where the body meets the neck. It was generally agreed that the range of sounds available from the NT1 and his guitar provided plenty of scope, and that excellent results could be achieved this way. For his electric guitar work, Peter had had the good fortune to pick up a Line 6 Pod, in excellent condition with PSU and manual, very cheaply at a local car-boot


Studio SOS

sale. He'd gone through the presets, but hadn't found much he liked, so I spent a few minutes setting up some '70s rock and blues sounds for him using his old faithful '70s Les Paul Standard. On our second visit, I set up four further sounds to use as generic bass DI settings for his newly acquired Aria bass, and these were based on the Pod's Brit 30 amp setting and on the The action of Peter's Gibson Les Paul Standard was set too Marshall-inspired amp high, so Paul used an old junior hacksaw blade to deepen setting played through a the nut slots slightly, but suggested that Peter also have the 4x12 speaker model. The instrument set up professionally as soon as possible. Pod Clean amp model is also capable of some descent bass sounds, as is the Tube DI setting, though I recommended that Peter buy the update chip to convert his Pod to version two software, as that allows you to mix amp models with different speaker models. While playing his Les Paul, I noticed that its intonation wasn't great, the reason being that the nut slots had never been cut to the appropriate depth. This meant that fretted strings went slightly sharp, and also that the action was higher than necessary. This was compounded by a slightly excessive bend in the neck, which could be redressed by a minor truss-rod adjustment. As this was potentially such a lovely instrument, I suggested Peter treat it to a proper set up at the local guitar shop, as it had evidently never been set up since it was bought, but I did deepen the nut slots for him using a ground-down junior hacksaw blade, which improved the playability immeasurably. Now when he gets it set up properly, the slots will only need a little extra dressing with the correct slotting files. We did manage to find a box spanner to fit the truss rod, but it seemed reluctant to turn and I didn't want to be the one to snap it! Maybe a good soak in WD40 would have loosened the recalcitrant nut?


Studio SOS

Studio Ergonomics The recording system was arranged so that Peter had the mixer and monitors in front of him, where they should be, but his computer was directly behind him. This meant that when he was setting up mixes and effects in the computer his back was to the monitors, which is less than ideal. We felt he would be better off moving the computer to the bench to his right so that at least he could border="0" alt="sos Peter@Monitors. remain facing the monitors as he s"> mixed. This would be possible if Peter's studio layout was less than ideal for the large CRT monitor was mixing in software — when seated at the replaced by a flat-screen model, computer, the main monitors were set up and this would also have the directly behind him. advantage of avoiding interference from the monitor scan coils causing hum on his guitar pickups. On our return visit he had moved the monitor screens to the bench on his right as suggested, which made much more sense from an ergonomic point of view and also left him with more free work-surface space.

The Return Visit Our second visit had to be postponed once, because Peter had managed to contract a particularly unpleasant computer virus, but when we finally arrived Peter had replaced his large VDU monitor screen with a pair of 17-inch flatscreen displays, and had also replaced his empty chocolate Hob Nob packets with full ones! We suggested that the system would work better if the Powered mixer was removed from the setup altogether, as the Allen & Heath mixer had more than enough inputs and facilities to handle recording and monitoring. As Peter currently does virtually all his mixing within Cubase SX anyway, the mixer doesn't actually have to do much in the way of mixing at all. He'd rewired his system along the lines we suggested, with the direct outs on the first eight channels feeding his soundcard inputs, while the remaining channels were used to monitor the stereo soundcard outputs plus any other sources. Peter had originally been sending the mixer channels to the interface inputs via the buss faders, but, as he invariably records one channel onto one track, the direct outs offered a shorter and theoretically cleaner signal path. Peter had also bought a Powercore Element system since our first visit, but was having some problems installing it. This turned out to be due to a faulty card, and thanks to the efforts of Jim Motley at TC Electronic UK (who diagnosed the problem) and the guys at Digital Village who supplied the card, a replacement was organised without delay.


Studio SOS

However, Peter wanted to work on some mixing while we were there, so I suggested that we use his existing Alesis Midiverb, and feed it from a spare output buss in Cubase SX so that it would become part of the effects send/return loop within his sequencing software. The reverb returns could be taken back to a spare stereo pair of inputs on the audio border="0" alt="sos Paul@Bass interface and then mixed back into +Pod3.s"> Cubase, but just to prove the principle we brought the reverb return back into Although the Line 6 Pod is associated more with guitar sounds, Paul showed Peter how two spare channels on the Allen & you can set up the unit to create perfectly Heath mixer, as Peter tends to mix to good bass sounds as well. a stereo DAT recorder connected to the mixer anyway. This worked fine, and though not an esoteric reverb by any means, we still felt it sounded a lot better than the built-in Cubase reverb, which after all had to be designed to minimise CPU usage. The channel aux sends in Cubase could be used to control the reverb level in the usual way, and provided that Peter makes a note of the reverb settings and return levels on the analogue desk, everything should be repeatable.

Fixing The Mixes Since our first visit, Peter had obtained a large acoustic office divider screen which he'd placed between himself and the drum kit area to improve separation and to preserve his hearing. He'd also recorded some new material which was rock/pop in style, and the recording seemed much improved over what we'd heard before, though there were still some areas that we felt needed attention. For example, the drum overhead mics lacked brightness, as Peter was still using his AKG C1000s for this job and they tend to have a fairly subdued high end. Adding 'air' EQ by applying a wide parametric boost at around 12kHz helped. The ideal solution would have been to assign the C1000s to duties better suited to them and to use a pair


Peter's host-based reverb processing was letting his mix down, so Hugh suggested that

Studio SOS

of basic, flat-response, smalldiaphragm capacitor mics as overheads. Where budget is a concern (and where isn't it?), there are numerous low-cost mics that will work perfectly.

he should plumb his Alesis Midiverb into the system, fed from an output of the computer's audio interface and returned to the mix via the Allen & Heath analogue mixer.

Although Peter still hadn't managed to replace the kick drum head, he had invested in a new tuning key for the rest of the kit! The continued tenure by the aged head meant the kick still needed generous amounts of 80Hz and 3-4kHz boost to add any sense of weight and impact, and even that didn't compensate for the state of the head. One tip here is that, when faced with this kind of sound at a stage where it is too late to re-record the part, you can sometimes add the desired depth by using the sub-octave plug-ins that come with many sequencers, just to add an extra layer of depth to the sound. This is really a salvage measure rather than good recording practice, but it can get you out of a tight corner when all else fails. The snare sound was also quite dull, and needed a lot of high-end EQ. Unfortunately Peter didn't have a harmonic enhancer plug-in, as I've had some success using the one in Emagic's Logic to add life to dull drum sounds — and indeed to lifeless overhead mics. On one track, Peter had used a synthesised bass sound instead of a bass guitar, and to add energy and depth I tried patching it through Steinberg's tubeemulation plug-in. In isolation it didn't sound exactly like a bass guitar, but once in the mix, it played its part a lot better than previously and actually sounded pretty good. That left the vocals, which sounded absolutely fine now that Peter had sorted out his NT1 mic orientation. However, because this was a rock track the voice needed some fairly strong compression to help it sit in the mix and to give it a high-energy character. We used the Cubase SX dynamics for this, using a 6:1 ratio, automatic time constants, and a threshold setting that gave 10dB gain reduction on peaks. This is more than I'd normally use, but it suited this vocal part extremely well. Peter also played us some tracks featuring acoustic guitar, where he'd tried some of the mic techniques we'd explored on our first visit, and I have to say the results were pretty good and needed little in the way of EQ.

Peter's Comments "You would have thought that I would have seen it all and done it all, having been involved in the world of home recording for 30 years. However, when Paul and Hugh visited I was amazed at how much there was still to learn and think about — they went through the studio with a fine-toothed comb! I think I was expecting, having read previous Studio SOS articles, that I would need to cover the walls with acoustic foam, yet by the end of the session I was pleased that Paul felt the studio didn't need that treatment. I felt really embarrassed about the vocal mic being the wrong way round, but it just goes to show that, even with all that experience, you can still get something wrong! The mixing techniques I was shown were really valuable and with the new Powercore system set up I started remixing old files to bring them back to life.


Studio SOS

"Home recording must be potentially one of the most expensive hobbies around — I imagine many readers will have felt dissatisfaction with their gear at some point, and the temptation to upgrade can seem irresistible. Replacing my monitors with flat-screen versions and buying the Powercore have burnt a hole in my pocket, and yet I've still got the drum head to replace and I must seriously think about new mics for the overhead drum positions. Oh, and the Les Paul still needs to be set up... I'm beginning to see that I need a long-term plan for buying gear, with a list of priorities. I wonder if there are any Russian football millionaires out there who would like to sponsor a small home studio? They probably couldn't afford it..."

All's Well That Ends Well Once again, the big picture turned out to be the sum of a lot of little parts. Rearranging the studio layout and simplifying the way the mixer was wired helped a lot, and Peter is planning to add another wiring loom so that he can mix all the separate outputs from his soundcard in the Allen & Heath mixer where necessary. Plus he has added a couple of capacitor mics to his list so that he can get a brighter overhead sound on the drum kit. His future plans include moving his control room into the adjoining garage so that he has better separation and more room for the musicians, but that's still some way off. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Writing Manuals

In this article:

About The Author

Writing Manuals Sounding Off Published in SOS July 2004 Print article : Close window

People : Sounding Off

Who'd want to be a writer of manuals? The endless showbiz parties, awards, book-signings and... hang on a minute... David Greeves

Spare a thought for the instruction-manual writers of this world. Like the poor souls who stick stickers on bunches of bananas (saying 'Chiquita' or whatever else) or go round amending the timetable at every bus stop on the route of the number 23, we pass by their life's work without ever wondering who they are. And their work is surely more notable, with all due respect, than that of the timetable putter-uppers or banana-sticker stickers — writing a 1200-page manual is not to be sniffed at, even if it is the work of more than one person. After all, reading a 1200-page manual is no mean feat, and it's one that is rarely attempted. I wonder if this offends them? I wonder if, one day, our manual writer, moonlighting in the customer support department, takes a call from a confused customer who's having trouble with the software's quantising features, and realises this can only mean that his lovingly written explanation of this subject ('Appendix F: MIDI Timing — Quantising, Grooves and beyond!') has gone unread. Does he shed a silent tear, and die a little more inside? Perhaps the general neglect of the manual-writer's craft contributes to the slight lack of enthusiasm that one detects in many manuals. In any case, there's little room in writing one for self-expression. And that's the way it should be — creativity and enthusiasm have no place in the instruction manual! I'd rather have a manual that's thorough, straight to the point and dry as dust than one which tries to be clever or funny. I may be on my own

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About The Author David Greeves is Sound On Sound's News

Writing Manuals

here, but a manual which includes light-hearted interjections and humorous captions is the paper-andglue equivalent of that odd man down the pub who thinks he's your friend and is always laughing at his own jokes. Why not go the whole hog and add an animated jack plug which pops up in the corner of the screen and says things like, "Hi there! I see you're creating a virtual instrument track. Can I help you with that?"

Editor. He believes that everybody has one good manual in them. Someday, he'll write his.

That aside, when you consider what manual writers have to put up with, I think you'll agree that they do an excellent job — most of the manuals you see these days are decent enough. Some are very bad and nearly all are fairly ugly, but most are very good. Nevertheless, they all seem to follow the same conventions. Take, for example, the obligatory 'troubleshooting' table at the back. I'm sure they can be useful sometimes and, from the manufacturer's point of view, they probably pre-empt a few calls to customer service (perhaps preventing a few more knocks to our poor manual writer's fragile self-esteem), but I can't help feeling that their format limits their usefulness somewhat. Most problems that can be solved in 15 words or less shouldn't really need explaining, and most things that you really need to be told couldn't possibly be explained in 15 words. And there are always a couple of entries which don't tell you anything at all. You know the sort of thing I mean. 'Problem: Unit has no power. Cause: Power is not connected. Solution: Connect power'. There's something satisfying about a concise explanation of the head-slappingly obvious — I'm sure philosophers have a special name for it. Perhaps some form of mystic religion could be built around the reciting of these mantras — repeat after me: 'The cables are faulty. Replace the cables'. Some manuals seek to further bemuse the reader with a web of indices, appendices and cross-referenced footnotes that threaten to trap you in an endless loop. But others are striking by the sheer lack of information contained within, or by the indecipherable way they present it. I recall a particular Russianmade mic (admittedly not a difficult thing to operate) which was accompanied by a small, unevenly cut booklet containing what appeared to be hand-traced frequency graphs and a few hand-written notes... in Russian. Perhaps they took the word 'manual' to mean 'done by hand'. It's just a shame they didn't have a stab at translating it into English — the results would have been priceless. I'll leave you with an excerpt from a manual, the rest of which has long since been lost, which hangs on a notice board in a corner of the Sound On Sound office, a benchmark in the field of incomprehensible technical writing which has seen off many a challenger over the years. If anyone can produce a better example, I'd like to see it, and if anyone can work out what the following actually means, I'll be really impressed! "Part 1: Be able to adjust the revolution of knob. Using adjustment metal not spring or cushion as usual, the revolution smooth. Use the fitting adjustment tool.


Writing Manuals

"Part 2: It is housed part of the gear into case. As grease is not leak, part of the shaft and gear is greased safficiently [sic]. Therefore it is low in rotary resistance and surpass others in endurance. "Part 3: Winding shaft and gear is unification processing. It is new manufacture whose screw is not loose and whose knob is fixed with the shaft." It's a dirty job, but someone's got to do it! Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Wyclef Jean

In this article:

Wyclef Jean

The Fundamentals Producer Waiting For The Ears To Published in SOS July 2004 Twitch Wyclef Jean's Tech Support Print article : Close window Going Platinum People : Artists/Engineers/Producers/Programmers Getting Together The Hard Disk Pro Tools Generation

In 1996, the Fugees came like a breath of fresh air into a world of hip-hop that was becoming stale around the edges. Now Wyclef Jean is a star in his own right, and has deployed his production talents for artists ranging from urban legends like Funkmaster Flex and Cypress Hill to Whitney Houston, Michael Jackson and even Tom Jones. Dan Daley

Wyclef Jean is the quintessential American immigrant success story: he moved with family to New York City from Haiti, the buckle of the central Caribbean's notorious island poverty belt, at the age of nine, ending up on another island of sorts — the Marlboro Houses council flats in Brooklyn's Coney Island. Don't be misled by the neighbourhood's fame as a late-19thcentury resort area; in the 1980s, when Jean was growing up, this area around the last stop on the F train subway was still like Moss Side with a beach. It was also a place where much of the new wave of Caribbean immigration was settling, Photos: Dave King. where Puerto Ricans had been replaced by Wyclef Jean (left) and Jerry Haitians and Dominicans, and its musical and Wonda grew up together, and now form an in-demand cultural eclecticism fit Jean's inclinations well. Hip-hop and rap blared from car boots filled with production and writing team. 18-inch subs, rattling auto and apartment windows alike. This sonic palette, in which everything a 100dB sound wave hit would become a resonant instrument itself, probably reminded Jean of his earliest encounter with a musical instrument, the tambu, a conga-like drum he file:///H|/SOS%2004-07/Wyclef%20Jean.htm (1 of 9)9/22/2005 8:57:37 PM

Wyclef Jean

played as a child in the Haitian town of Croix-des-Bouquets. It fit well with the musical influence conferred by his father, a minister, in whose church Wyclef and his cousin Jerry Wonda — who is still Jean's collaborator on production and composition to this day — would direct (and eventually record) the choir. However, Jean doesn't think of himself as a Haitian-American, and though he has been quoted as referring to himself as "100 percent Haitian... Haiti till I die!" and has been dubbed by the press as the 'ambassador to the world for hip-hop,' in conversation he will tell you that he is neither a diplomat nor a hyphenated American, but rather "first and foremost a human being." He is also one hell of a musical savant, as well. His Grammy wins and nominations are out of proportion for someone barely in his '30s, and his discography is equally extensive. After buttressing his native musical talents at high school in New Jersey, where he learned guitar and studied jazz, his first group, the Tranzlator Crew, evolved into the Fugees, named for the slang for 'refugee' — a status Jean had emotionally long since left behind but still empathises with. Also featuring Wyclef's cousin Prakazrel Michel (aka Pras) and Michel's highschool classmate Lauryn Hill, the Fugees would go on to blend jazz, rap, R&B and reggae into a multiplatinum career whose second record, The Score, hit number one on the pop and R&B charts in 1996. Lauryn Hill's vocal on the remake of 'Killing Me Softly With His Song' and Wyclef's cover of Bob Marley's 'No Woman No Cry' demonstrated a remarkable courage and cleverness: besides making these classics contemporary again at a time when rapidly consolidating radio was looking for sure things, the songs' familiarity allowed the production chops Jean and Wonda brought to the recordings to shine through — simple, clear, just a hint of urban edginess and, most importantly, the elusive and difficult-to-define 'vibe', the Holy Grail of urban record-making. What attitude is to rock, vibe is to hip-hop, and Wyclef Jean has supply to spare, as is amply evidenced byy his recent album The Preacher's Son. The album was among the first to be recorded in Jean's new Platinum Sound studio, whose rooms are split between Jean's work and outside clients. It's located on the second floor of the building that formerly housed Warehouse Studios on West 46th Street, once Manhattan's nightclub row, next door to Joe Allen's bistro and on the fifth floor, three levels up from the New York City offices of Solid State Logic. Small wonder that Platinum's two studios are populated by 9000J and 9000K consoles. In a New York studio, it's rare that a spare capacitor is closer at hand than a pastrami sandwich.

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Wyclef Jean

Wyclef Jean's own roots in Haiti are firmly embedded in his memory and his music. "What I recall are the acousticals," he says. "The fundamentals. That everything was an instrument without being electrical, man. When we got technical later, that was just an addition. Percussion, guitars, everything was acoustical. You could hear the reality of the sound right in front of you. In you. It was Caribbean music. When I came to Brooklyn I listened to whatever they were playing in the trucks and cars, hip-hop mostly. And reggae. All that music I make now comes from that background. I'm a musical person and it all just went into making me who I am... Hold on, I got Clive Davis on the other line..." Wyclef takes a few minutes to talk with Davis, former head of Arista Records who now heads the wildly successful J Records, developing artists such as Alicia Keys. Jean has a joint venture label deal with J Records. He returns from the call, saying "I gotta make a change to my record for Clive. There's a track with me and Wayne Jean's Platinum Sound studio features two Wonder and Elephant Man — 'Doctor'. SSL rooms, both with SSL consoles. This is Clive wants to make sure the vocals the J9000. are up far enough and the drums are louder. I don't consider [Clive Davis] a producer. He 's a music man. Our conversations are about chords, major and minor, Mixolydian [scales]. They're very healthy music conversations. Any producer knows Clive Davis has been around a long time, around way before me. I don't take his comments lightly. We talk about pushing the envelope of a record." As to the influence of radio on those decisions, Jean says, "I don't know how my stuff gets on radio, 'cause it doesn't sound like any other records. My records get on all the charts in all formats, all over the place. 'Maria Maria' [with Carlos Santana] got on all the charts. '911' [a hit duet with Mary J Blige] got on all the charts. It's my eclecticism that does it." The Fugees, and before that the Tranzlator Crew, were Jean's developmental vehicles, as a composer, performer and producer. 'We were Haitians in America and we felt like we were translating the music for here. We rapped in five different languages. World slang for 'refugees' is 'fugees', and that's who we represented. I wasn't doing nothing different from what Miles Davis or Dizzy Gillespie was doing. They travelled the world interpreting music, too. They used all the music they found. I just happened to be growing up in the time of hip-hop, but I could take a breakbeat and fuse it with Bitches Brew. My edge is hardcore street edge, but I'm aware of jazz, of country music, of classical. It can all be fused. When I'm chillin' I listen to everything. I like Bob Dylan's 'The Joker Man', it's a kind of obscure cut of his. And Johnny Cash. I cut a record on his 'My Delilah'. I listen to Miles Davis. I'm a music man."

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Wyclef Jean

Waiting For The Ears To Twitch Wyclef Jean came of age at an interesting time in technology, when certain key items, such as the drum machine, were poised to become the instruments of urban music, just as the Stratocaster had for rock & roll. "The way I learned was with a single format — a Linn 9000 [sampling drum machine]," he explains. "Kalise Bayon from Kool & The Gang taught me how to use it. It was the fastest way to make music. All you needed was a Linn 9000 and a keyboard. That's where I got my recording style. My cousin Renell bought me and Jerry [Wonda] a little Tascam [multitrack] tape recorder and we started cutting to that with the Linn and an [Ensoniq] VFX keyboard. Than I needed to plug my guitar into something and I just plugged it right into the Tascam, so we were always recording live. What I was learning was to make it happen as quickly as possible and find the sounds we wanted as quickly as possible. That's where I got my recording chops. I'll tell you a secret about 'Killing Me Softly'. I wanted a Fender Rhodes piano on that track but we didn't have one. So I took a sound in an S900 sampler and put a delay on it and detuned the sampler and played it like a Rhodes and it sounds like one. We just had to invent shit as we went along. "In the studio, I have two [engineers] I work with, Andy Grassi and Serge Tsai. Andy did a lot of drugs in the '60s and he's of the hippy generation, and he has mixed for everyone and their mother, so if I want a Nirvana guitar sound, he knows how to get it for me. Serge is from Amsterdam [Rotterdam, actually; see sidebar], so he's a pothead, but they have a certain style of production over there. He comes up with weird delays and things that only a guy on Amsterdam weed can come up with."

The live area for the Platinum Sound J-series room.

Still, Jean can be very hands-on when he wants to. "I start by taking every effect off of everything. I just want to hear the sound raw and pure. When you put effects on a the sound right at the beginning, you lose the natural elements of the sound, the elements that inspire you. These are the raw elements that you build from. The human ear can hear three things at a time. If you're blessed with a gift like mine you can hear dog whistles, so maybe I can hear four things at a time. But I just get in front of the Genelec or Yamaha speakers and wait for my ears to twitch. That's what you wait for. I can't explain it other than that. It twitches. It tells me where to put things. I'm looking forward to checking that out when I do some of my next CDs in 5.1." If sampling is integral to urban record production, ripping is its dark side, and Jean has gone on record stating that he is unhappy about how artists lose out file:///H|/SOS%2004-07/Wyclef%20Jean.htm (4 of 9)9/22/2005 8:57:37 PM

Wyclef Jean

when songs are downloaded illegally. How does that jibe with the fact that much of rap is based on samples of other people's records? "Sampling came from us listening to old records," he replies. "We grew up with the Stylistics, so we would loop a Stylistics beat and use it. There's nothing new under the sun — everyone is influenced by someone else, there's only so many chords in a musical scale. What we were doing with sampling, we were doing out of love, out of respect for the artist. We weren't thinking that we were ripping him off. That's opposed to you taking my whole CD and downloading it for free. That's different." Given the wide range of artists that Jean has worked with over the years, how he chooses who to collaborate with is interesting. "I have to be a fan, or else I look for groups that are just coming up, not famous, like Chen of Rough Riders, who is a Chinese rapper. Destiny's Child was like that. They were new when I gave them their first hit. Same with Mya; we gave her 'Ghetto Superstar Surprise'. With Destiny's Child's first hit, 'No No No', I had them sing the melody in triplets, like a rap. Radio thought the singing was too fast but the next thing you know it was a hit. That was a very low-key production. We made it somewhere in Texas, very low-key."

Wyclef Jean's Tech Support Serge Tsai has worked with Wyclef Jean for the past five years, alongside the producer's other regular engineer, Andy Grassi. The Rotterdam native is the detail man when Wyclef is deep inside the vibe. But part of the speedy approach to production that characterises Jean's records is having a set pattern of production techniques. "I'll always have the Neumann U67 set up," says Tsai. "That's our main microphone. We'll try an 87 or a 251 on occasion, but it almost always goes back to the 67. It just works for everything. The same for the Pro Tools HD rigs we have. They're big — 24 inputs and 64 outputs. We just use a lot of the same gear for most of what we do. Everything is always plugged in, the MPC for percussion, the guitars, and a DAT machine is always running, to make sure we catch every idea as it goes by. The studio is always in 'ready' mode. Like Wyclef says, the vibe will take care of the rest." One of Platinum Sound's two studios, the one fitted with the SSL 9000K, was designed originally by Tom Hidley, and his trademark front-wall monitoring system still dominates the control room's front wall, though it's now loaded with TAD components. The second studio, with a 9000J console, was designed by Frank Commentale, who did several other rooms in New York that have appealed to Jean over the years, including Chung King and some of the rooms at Hit Factor. It's fitted with Augspurger monitors, and both rooms use Genelec 1032/1074 and KRK E8 nearfields. Commentale also tweaked the original room's acoustics, while Andy Grassi did the wiring designs for both rooms. "When I first met Wyclef and Jerry Wonda, they were definitely tape guys," Tsai says. "I was with them as they transitioned to computers and to Pro Tools, and they were not immediately down with that approach to recording. But once they


Wyclef Jean

saw everything they could do digitally, they were totally hip with it." A Studer A827 multitrack is on hand, a remnant of the Booga Basement days, but is hardly touched lately, Tsai says. "It's not like I don't like tape, but I know what tape sounds like, and I can simulate it pretty well when we're recording digitally. It's all in using multi-band compression wisely. I'll send the signal out from the buss on the 9000K to various compressors and return them through different EQs. You can make the sound bounce a little, like it does on tape, and this also adds a little bit of noise to the signal — the kind of grunginess that you get from analogue tape." Andy Grassi was a staff engineer at the Hit Factory in Manhattan when he met Wyclef Jean and Jerry Wonda in 1998, when the producers came into the studio with singer Joan Osborne to work on a song for a movie soundtrack. He's been their regular engineer ever since, a post gained, he says, because of his experience with live rock records as well as hip-hop. He brings that combined experience to their sessions, as well. "This is hip-hop very often, but there's also a lot of live recording of instruments going on, as well," Grassi explains. "The trick to what I do is to find ways to balance hip-hop's need for spontaneity — someone will just grab a microphone and wail at any given moment — and getting everything recorded clearly and without distortion. And that balancing act has gotten more interesting since we put Pro Tools in, because the sessions move a lot faster now." Grassi says he likes to set up microphones around the recording room as through he was preparing a live tracking session, even though most of the percussion is created by Wonda on his Akai MPC3000. "Jerry has a lot of acoustic drums that we recorded sampled to the MPC," Grassi says. "It's a good sampler when you go in through the digital input; the analogue ones tend to sound kind of grainy. So I try to get Jerry to record digitally. I also keep an amp head set up in the control room with a cable running to a speaker cabinet in the studio. I always want to try to have things that actually move air on the tracks. Even when we're using digital drums — which is most of the time — I use a few reverb tricks, like very short setting and fast slaps, to give the sense that there's some air around the drums. The [Roland] Dimension D unit is especially good for making a digital kick drum sound like it has four walls around it." Tsai and Grassi split mixing duties, which Jean acknowledges are not his cup of tea. "I just come in now and then and say change this or that's OK," he says. Basically, says Tsai, a day will often start with a beat, a bare skeleton of a song, and by day's end it's a finished track. "We don't record tons of stuff," he says. "Wyclef usually knows what he wants and he'll get right to it. If we record horn parts, he'll pick and choose a bit, but mostly we go right for what he has in his head the first time. Track it, record vocals, a few overdubs, and mix it. Next."

Going Platinum Given the scale of Wyclef's production schedule, it was inevitable that he and Wonda would eventually need an expanded version of the studio they had built several years before across the Hudson River, in New Jersey. Platinum Studios opened last year. "We used to have a studio in New Jersey, Booga Basement," he says. "But I decided a while ago that, as a businessman, I'm not going to give [commercial studios] millions of dollars that I can use to build my own studio and where I don't have to watch the clock. We have two rooms, one with an SSL J file:///H|/SOS%2004-07/Wyclef%20Jean.htm (6 of 9)9/22/2005 8:57:37 PM

Wyclef Jean

and one with a K. But I like Neve sounds, too. I love the mic pres, EQs and compressors. I love the SSL compressors. We have a lot of digital gear, and we record to a Pro Tools system, but I like the warmth of an analogue console. But the important thing was that we made a place where the artists feel as comfortable working as we do. That's why I like it that we have the entire floor, for privacy. We also have a policy of not disclosing who is working here. "Before this, I worked in just about every studio around New York City. Hit Factory, Chung King, Sound On Sound, House Of Music in New Jersey. I'm a studio whore. Different studios have different vibes, and that can affect the music one way or the other. But the risk is always that it can slow you down when things aren't flowing, when people have to be told to do things instead of just knowing that this is what you're going to want for a vocal The other Platinum Sound control room, with or for a guitar part. Nothing's more its K9000 SSL console. important than keeping the flow going. That's part of vibe. Especially since I'm writing everywhere, 24/7. All the time. In the street. On the bathroom. On the motorcycle. The key to the way I work is that everything always be ready. The microphones set up, the sounds ready. If I pick up a guitar and get into a vibe, they know it in the control room and they know to turn on the guitar amp microphone. Vibe is all about spur of the moment. Being ready when the thing happens. You don't want to miss it when it happens." Vibe. The magic word and elusive state of grace in urban music. "That's it," he says. "There's no second takes for vibe. If you can't get it after 10 minutes, don't even bother any more. It's about the moment. When Mick Jagger worked here with us, when Carlos Santana worked here, it was the same thing."

Getting Together Wyclef Jean's collaborations have produced many memorable tracks, such as 'Maria, Maria' which he created with Carlos Santana. "That song was inspired by West Side Story, which I had seen the movie of," he recalls. "West Side Story, East LA, it's all the same. I wanted it to be a sexy story, and Santana is a guitar god, and he can play anything sexy. Carlos knew how he wanted his guitar amp miked; my job was to make sure that we captured what he did and translated it to what the song needed. He would play a part, it might go for four minutes. I'm listening for the part I want, it might be a minute of that. I find it and loop it. We do that a lot — find the parts out of longer passages and then move them around, arrange them into a song. Now, we could have Carlos play the parts over and over again and put them in the right places, but we use technology instead. We


Wyclef Jean

make loops and move them around. Look for the parts that strike you as the theme of the song, in terms of its vibe. We don't want to cut and paste the whole track, but the thing is to find the parts with the vibe and capture them, then put into the places you want them. I don't want to get between him and the energy of the recording. Then the magic is lost." The K-series room also boasts an impressive live area.

Jean takes a similar approach to producing vocals. "Record everything, never erase anything, when it's going let it go — I rarely stop it in the middle," he exclaims. "Then you f**k it up. Some people like to cut vocals in parts, but I like to try to get one full pass, then fix it as needed. I really don't like to use AutoTune. You work with great singers, you'll get what you're looking for if you give them space and the right vibe to play off of. That's why rap records are so great, so full of vibe. Because they're not perfect. You can hear someone yelling in the background. It's there, on the track. So be it. That's what Thelonious Monk would say — so be it. The dirt just goes along with everything else on the record. Only Patti LaBelle got her part on my record down perfect in one take. One single take." As he said earlier, Jean needs to be a fan of an artist before he can commit to working with them. But when it comes to artists such as Mick Jagger, Santana, Michael Jackson or Tom Jones, does he ever give himself a chance to just sit and listen and enjoy their presence? "I do," he replies. "I did with Santana. I realise this is the legend sitting here. I want to sit and listen, but then I also know I have a job to do. We have to make a record, so I move my head into that space and get to work. Mick Jagger is incredible. We did work on his last solo album. He would play a run on the guitar and I'd love it and make loop of it and play it back to him and he'd make up another part. He's another guy who looks for the theme to build a song from. He understands vibe. Michael Jackson is larger than life, like a star should be. It's fun to watch how he interprets rhythm. He moves his head in one way, that Egyptian thing, side-toside. But at the same time, he moves his feet and makes a drum part. It's like his mind is analysing the rhythm and his feet are performing it. It's as if he's turned his whole body into an entire drum kit and is playing it."


Wyclef Jean

But it's the ultimate Caribbean musical ghost that infuses Jean's own spirit, an artist he can now never collaborate with: Bob Marley. "It's automatic. He's someone I grew up on," he says. "I felt like I had met him when I listened to his music. Music is an appreciation of history. Every piece of music links you to another. Quincy Jones is like my father in music. He came from Cab Calloway's band and then he flips over and produces Michael Jackson. You have to know where music is coming from to know where it's going."

The Hard Disk Pro Tools Generation Wyclef Jean's take on the future of production is that technology will continue to progress and we'll all just keep getting used to it. But don't count on limos for everyone, like it used to be. "We're in the hard disk Pro Tools generation now," he says. "You can be in Tennessee and I can be in New York, and if you get an idea you can play the part and email it to me and I can add to it and send it back to you. I do that with Missy [Elliott] all the time. We're getting into the computer with music. I don't think that's a bad thing. I don't think you'll be able to make a complete record that way, but you can build the song. But the business is changing. The record labels are getting more strict. They'll tell you now, 'You got 10 hours to make me a record, f**kface!' Ha! Like it used to be. No more rock star stuff. But that gets you to the essence of the song very quickly. Maybe it's a good thing." Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Logic's Aux object

In this article:

The Different Types Of Audio Object Logic Tips Dealing With Multi-channel Virtual Instruments Have Your Say! Using Aux Objects With Rewire Extract The Timing Of A Drum Loop As MIDI Combining Aux Objects & Buss Objects Using Aux Objects As Input Objects

Current Versions Mac OS X: Logic Pro v6.4 Mac OS 9: Logic Pro v6.4 PC: Logic Audio Platinum v5.5.1

Logic's Aux object Logic Notes Published in SOS July 2004 Print article : Close window

Technique : Logic Notes

The Aux object might seem a little arcane, but it's the key to a variety of useful Environment workarounds. Len Sasso

The Aux Audio object is one of Logic's best-kept secrets. While there is only one situation in which you must use it, there are many for which it is the most convenient alternative. For separating the outputs of multi-channel plug-in instruments, the Aux object is the only choice. As a buss return, it opens a whole slew of new effects routing possibilities. Aux objects are also the easiest way to manage stereo Rewire busses. So let's see how it all works.

The Different Types Of Audio Object Audio in Logic comes primarily from four sources: external input via your audio interface, playback of audio files, the output of virtual instruments triggered by MIDI data (live or recorded), and other audio applications whose output is bussed in via Rewire or another third-party bussing scheme. In Logic, all audio, whatever the source, is managed and routed using Audio objects in the Environment. All Audio objects are created equal and start out with no specific purpose. When you create an Audio object from the Environment window's New menu you get a small icon labeled '(Audio Object)'. If you double-click it, it will expand into a vertical channel strip with no controls and the word Off in the middle. To give it a purpose, you need to select it and then choose what kind of Audio object it should be by using the Cha parameter in its Parameters box (located at the left edge of the Environment window). Audio objects fall into seven categories: Track, for audio file playback;

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Logic's Aux object

Input, for external audio; Aux, for routing audio within Logic; Instrument, for playing virtual instruments; Rewire, for receiving audio from other applications; Output, for sending audio to your audio interface; Buss, for routing audio within Logic; and Master, for controlling overall output level.

Linplug's RMIV synth/sampler is capable of individually routing its sounds to different mono or stereo audio outputs. In this example RMIV's first 12 virtual pads are assigned to four stereo outputs and eight mono outputs. The first stereo output is handled by Logic's Instrument Audio object, whereas the remaining outputs are handled by eleven Aux objects.

Your Autoload Song may have Audio objects of some or all of those types. Typically you will find them all on the Audio Layer of the Environment window. If you have little Environment experience, a good way to follow the examples here is to delete all the Audio objects, then create new ones as needed. (Note that deleting Environment objects in one Song will have no effect on your other Songs, new or old.) I'll start by using Aux objects to manage the outputs of a multi-channel virtual instrument.

Logic Tips You can drag the cursor in any window without bringing that window to the top. The trick is to make sure you hold the mouse button for a moment — quick clicks are what top the window. If you use Markers to delineate Song sections, you can use the Key Commands Set Locators By Previous/Next Marker and Enable Cycle to quickly jump back and forth through the Song by section. If Logic is playing when you use the Key Command, playback will immediately jump to and cycle the new section. You can assign a mixer channel (actually the associated Audio object) to more than one Group by holding Shift while selecting additional Group assignments. The Groups act independently, allowing you to, for example, assign the same channel to several different mute Groups.

Dealing With Multi-channel Virtual Instruments Virtual instrument plug-ins fall into three categories, depending on the number of audio outputs they support: mono, stereo, and multi-channel. Multiple outputs are


Logic's Aux object

often used for different drums in a drum synth, different samples or sample groups in a sampler, or different instruments in the same multitimbral synth or sampler. In each of those cases, one use for multiple outputs is to apply separate processing to each output. Another is to allow separate outputs to be bounced to separate audio files. A third is to route different virtual instrument outputs to different outputs on your audio interface. The Aux object is the key to each of those options. To use Aux objects for managing the outputs of multi-channel virtual instruments, you need to create an Aux object for each of the instrument's mono and stereo outputs. Typically the virtual instrument will allow you to set how many outputs of each type it has, and you will need to quit and restart Logic after changing the configuration for the changes to take effect. Once you've set up your instrument's outputs and created Aux objects for each output, you simply need to assign the Aux objects' inputs to the various instrument outputs. You do that using the top drop-down menu in the I/O section of the Aux object's channel strip. Be sure to make the Aux objects mono or stereo as appropriate, by using the button at the bottom left of each object's channel strip. One thing to notice here is that the instrument's 'main' outputs (often simply labelled as outputs one and two) are not available as Aux object inputs. Those outputs are always routed to the Instrument Audio object's output. In fact, any virtual instrument outputs that are not used as an input to an Aux object are also automatically routed there — the purpose being, of course, to ensure that none of the instrument's multiple outputs are simply dead ends. Another thing to keep in mind is that using different virtual instrument outputs does not necessarily result in their going to different Logic outputs. The ultimate destination of each of the virtual instrument's outputs is determined by the output assignment of the Aux object to which it is routed. For example, you can use different Aux objects for different insert effects and different buss routings, but still have all the virtual instrument's outputs routed to a single Logic stereo output.

Have Your Say! If you want to suggest changes or improvements to Logic, then here's your chance! The Emagic development team are inviting SOS readers to send in their suggestions of what they'd most like added or changed in Logic. Email your top five suggestions (in order of preference) to [email protected], and we'll forward your lists on to the Logic team. We'll be asking them for feedback on which changes users deem most important and how these might be addressed.

Using Aux Objects With Rewire Although Rewire busses are technically mono, some applications automatically group them into stereo pairs. However, Logic is not one of them — the Rewire file:///H|/SOS%2004-07/Logic%27s%20Aux%20object.htm (3 of 6)9/22/2005 8:57:52 PM

Logic's Aux object

Audio objects are always mono and each mono Rewire input requires its own object. This means that the stereo outputs of Ableton Live and Propellerhead Reason, for example, require two Rewire Audio objects for processing when Rewired into Logic. If you want to treat them as the right and left channels of a stereo buss in Logic, an Aux or Buss Audio object is the solution, and Aux objects are more flexible. Logic's busses are stereo by default. The trick is to pan the Rewire object carrying the left stereo channel hard left and to pan the one carrying the right channel hard right, then set both Rewire objects' outputs to the same Logic buss. Next, create an Aux object, ensure it is in stereo mode, and set the Logic buss carrying the Rewire channels as its input. The Aux object now functions as a stereo Rewire return. You can insert effects in the Aux object, and you can create sends to route the Rewire audio to other Logic busses, which is something you can't do using Buss Audio objects. This allows you, for example, to use a single send for routing stereo Rewire inputs to a master reverb.

Extract The Timing Of A Drum Loop As MIDI Once you've sliced up an audio file into regions corresponding to individual musical events, as shown in SOS May 2004, it's a simple matter to capture their relative timing in a MIDI sequence. You might then use the MIDI sequence as a Groove Template for other MIDI sequences in your Song or as the rhythmic basis for another instrumental part. Doubling a hi-hat or ride-cymbal pattern with a bass line can, for example, produce interesting results. To capture the groove, first create an empty MIDI sequence starting at or before the position of the first region and extending to or beyond the end of the last region. Open the new sequence in the Matrix Edit window and turn on MIDI step input by clicking the Matrix Edit window's In button to turn it red. Select the audio track containing the regions in the Arrange window and move Logic's Song Position Line (SPL) to the beginning of each region, switching to the Matrix Edit window and playing a note at each new SPL position. Three of Logic's Key Commands make this repetitive task fast and easy. Use the Select Next Event Key Command to move from region to region on the Arrange track. (I have the right arrow key assigned to that.) Use Goto Selection to then move the SPL to the beginning of the selected region. (I have the Tab key assigned to that.) Then use Select Next Window to switch between the Arrange window and the Matrix Edit window. (I have Option+Tab assigned to that.) The choice of pitches and velocities for the notes you're entering depends on your intended use for the MIDI sequence. If you're simply making a Groove Template, the pitches don't matter and, unless you're applying the groove to velocity, neither do the velocities. If you're going to use the MIDI sequence to play another instrument, the pitches are critical and, depending on the instrument, the velocities are probably important too. You can adjust the pitches after the fact


Logic's Aux object

using MIDI Step Edit, by double-clicking the In button so that it is red with a lock icon. That also changes the velocities, so you will probably want to go back and edit those manually using Hyper Draw or the Event List window. Another use for the MIDI sequence is to play a sampler which has the slices mapped to consecutive notes. In that case, the pitches in the MIDI sequence should correspond to the sampler mapping, and the velocities should all be equal (and probably maximum) to play the slices at their original volumes.

Combining Aux Objects & Buss Objects An important thing to remember when using Buss and Aux Audio objects for the same buss is that the Buss object comes first. That means its level and pan settings, as well as the processing by any effects inserted in the Buss object, happen before the Aux object gets the signal. It also means that if you change the Buss object to mono, its left and right channels will be mixed to mono — the buss effectively becomes mono. On the other hand, if no Buss object is present in the Environment for a buss feeding an Aux object, the buss signal goes directly to the Aux object. In short, Aux objects can be used as buss returns with or without having Buss objects for the same returns. If you do use both objects for the same buss, you will typically want to turn the Buss object's output off. There are two reasons to use Aux objects as buss returns: they have buss sends of their own, and several of them can be fed from the same buss. That allows you to create complex effects paths which would be impossible without Aux objects. For example, if you want to apply separate insert delay effects to the left and right channels of a stereo Logic buss, create two Aux objects panned hard left and right and use the same Logic buss as input to both. You could then use the Aux objects' sends to route the processed signal to a master reverb. The more busses and Aux objects you have, the more complex your signal processing configurations can become, of course. If, as in the above example, several Aux objects are fed by the same buss, it is often convenient to use a Buss object as a master level control for that buss.

Using Aux Objects As Input Objects In addition to taking their inputs from Logic busses and multi-channel virtual instruments, Aux objects can also take their inputs directly from Logic's input busses. They can, therefore, be used exactly like Input Audio objects to destructively process incoming audio when using Logic's real-time Bounce feature. The Aux object offers the slight advantage that you don't need to have Software Monitoring enabled in Logic's Audio Driver Preferences to hear and bounce the results, as you do when using Input objects. At first, adding the Aux object to the already confusing array of Environment Audio objects might appear to be adding unnecessary complexity. But in fact, with its many uses and great flexibility, the Aux object actually represents a file:///H|/SOS%2004-07/Logic%27s%20Aux%20object.htm (5 of 6)9/22/2005 8:57:52 PM

Logic's Aux object

simplification in Logic's audio management. Published in SOS July 2004




CLASSIC TRACKS: 'The Reflex'

In this article:


Going South Roxy's Songwriting Methods Producers: Duran Duran, Alex Head Honcho Thornalley, Pete Schwier Contrasting Abilities Published in SOS July 2004 The Days Before Auto-Tune Print article : Close window Genius Of The Peas Technique : Recording/Mixing Reflex Response Out Of Tune A Faceful Of Garbage Taking It Seriously

Sadkin, Ian Little; Engineers: Phil

When Duran Duran began work on their third album in 1983, they were already one of the biggest bands in the world — and with eight months of studio time and half a million pounds spent, huge expectations surrounded Seven And The Ragged Tiger. Richard Buskin

It was during a three-year stint at the Gallery Studio facility of Roxy Music lead guitarist Phil Manzanera in Chertsey, Surrey, that Ian Little got the call to work with Duran Duran. Having already engineered Manzanera's Primitive Guitars album and served as assistant producer on Roxy's Avalon, Little appeared to fit the bill when Britain's leading New Romantics were searching for a new producer to collaborate with them on their third album. After 1982's highly successful Rio, which had spawned hit singles 'Hungry Like The Wolf', 'Save A Prayer' and the catchy synth-rock title track, the Roxyinspired, Birmingham-based quintet of Andy Taylor (guitar), John Taylor (bass), Roger Taylor (drums), Nick Rhodes (keyboards) and Simon Le Bon (vocals) opted to part ways with producer/engineer Colin Thurston before embarking on Seven And The Ragged Tiger. Initially, Ian Little was asked to remix a track off Rio at Tony Visconti's Good Earth complex in London's Soho, and when he realised that no members of the band were even bothering to show up he was, as he now recalls, "really pissed off". The result was that he completed the mix in double-quick time, making it sound as rough-eged as he could, and departed the studio thinking that he'd file:///H|/SOS%2004-07/CLASSIC%20TRACKS%20%20%27The%20Reflex%27.htm (1 of 12)9/22/2005 8:58:02 PM


never hear from Duran Duran ever again. Wrong... Over the course of the next week, each of the band members contacted him to say that their material had never sounded so exciting, and this in turn led to his co-production assignment on the ensuing album. "They'd left me to mix that track alone because they wanted to see where I was at," Little explains. "They wanted to know what I would do given a free rein, and I suppose that was the right approach and my own reaction was also right — I did exactly what I wanted without any concern for what they might think. I mean, when I left the studio that evening I thought 'That's the end of that,' because to my mind the mix that I had done was like giving them the finger. It was so punky compared to how their music had sounded before — Colin Thurston's production was very musical, whereas I didn't approach things from that direction. My technique was much more rhythmic, much more percussive; a snare that cut your ears off, distortion on Andy's guitar, taking John's stereo chorused bass and wacking it into mono. I wasn't interested in pansy music, so I got sounds that I myself liked, and my attitude was also a little dismissive because I was so pissed off, proving that things can be a lot easier if you're true to yourself." After co-producing the single 'Is There Something I Should Know?', which debuted atop the British charts (and would eventually be included on the CD release of Duran Duran's eponymous debut album), Ian Little and the band commenced sessions for Seven And The Ragged Tiger. Leaning more towards a synth-dance sound than their previous two records, this album would eventually achieve double platinum status while featuring another trio of hit singles: 'New Moon On Monday', 'Union Of The Snake' and transatlantic chart-topper 'The Reflex'.

Going South Initial songwriting and demo sessions for Seven And The Ragged Tiger took place during a three-month spell in the South of France that began in March 1983, where everyone lived and worked in a three-storey chateau while a 24track mobile studio rented from North London's RAK Studios was parked outside. With the band set up alongside a 14-channel mixer in a large, open room on the top floor with a pitched roof and bare floorboards, Ian Little sat behind the truck's API console and recorded the jam sessions to a 3M M79 two-inch tape machine running at 15ips, while monitoring on Tannoy Red speakers. "The mobile came with a load of outboard gear, including Drawmer gates, AMS delays, Lexicon reverbs and an Eventide Harmonizer," he recalls. "When it came to the miking, I used Neumann U87s for Simon's lead vocals and Andy's BVs; an file:///H|/SOS%2004-07/CLASSIC%20TRACKS%20%20%27The%20Reflex%27.htm (2 of 12)9/22/2005 8:58:02 PM


AKG 414 inside the bass drum and a Shure SM57 in front; an AKG C451 pencil mic on top of the snare and an SM57 underneath to capture the rattle; two flat black PZMs behind the kit as well as a pair stuck up on the ceiling, mainly for drum ambience; and SM57s on the guitar and bass amps — the guitar was DI'd and Andy's effects were like an insert on that line."

Roxy's Songwriting Methods With feedback from Ian Little as to what parts of their jam sessions he considered worth developing, the band members wrote songs for the new album. And they did so while adhering to a deadline — as the project was geared towards a Christmas release, they knew that the longer they took to write the material, the less time they'd have to record it. "Nothing had been written in advance, so the biggest starting point they'd ever have would be another song," Little remarks. "For example, 'Union Of The Snake' was built and written on the bass drum pattern of David Bowie's 'Let's Dance'. Everyone loved that track, so Roger, Nick and I analysed it and realised that the bass drum pattern was quite unusual — although you think it's just a regular old two-bar loop, it actually doesn't repeat for eight bars. It's quite tricky. However, it's a great beat, so Roger started playing it, got it right, John joined in and came up with the bass riff, and then the song was pounded on top of it.

Ian Little with his present-day writing setup.

"I didn't know how the band members had written in the past, but I did introduce them to some of Roxy Music's songwriting methods. I mean, Bryan Ferry would often come up with a chord sequence and record 15 or 20 seconds of that sequence, put it on a cassette and number it. After a few weeks, he'd give a cassette to their producer Rhett Davies containing, say, 75 20-second sequences of just two or three chords, and Rhett would then say 'Well, we could merge number 32 with number 74 and make a song.' That taught me a way of optimising your output — whatever you come up with, keep it, look at it again and use it. And if you're able to combine some of those elements to make a song, all well and good. As soon as you've got a structure, you'll then write parts that blend those sections together and there will be a homogeneity to the entire song. "The other method that both Phil Manzanera and Bryan Ferry employed in terms of their writing was to have me program a Linn Drum, put some delays on it and create a groove. Bryan would then vamp on the keyboard and produce what he file:///H|/SOS%2004-07/CLASSIC%20TRACKS%20%20%27The%20Reflex%27.htm (3 of 12)9/22/2005 8:58:02 PM


called a 'moody synth' sound, which was like a pad sound with plenty of movement and character. That would enable him to get a lot of feeling out of a couple of chords, and Duran Duran did the same thing. They initially used a drum machine when composing 'Is There Something I Should Know?' before Roger took over, and we also had a Linn in the South of France. I don't recall using it that much, but once in a while, when ideas dried up and everyone was sitting around scratching their nuts, I'd create a groove and they would just bounce off it."

Head Honcho After three months of using the RAK mobile, Little and the band moved to AIR Montserrat for a six-week-long round of recording sessions. And Alex Sadkin, who had previously taken over the mix of 'Is There Something I Should Know?' was brought in as the head production honcho on Seven And The Ragged Tiger. A protege of Island Records founder Chris Blackwell, the American-born, Londonbased Sadkin produced and engineered recordings by Bob Marley, Grace Jones, Robert Palmer, Talking Heads, Joe Cocker, James Brown, the Thompson Twins, Foreigner and Simply Red. He would die in a car accident just four years later at the age of 35. "I'd helped the band write 'Is There Something I Should Know?' and so by time of the mix I didn't know it from Adam," Little admits. "It could have been a sausage. You lose any objectivity when you've been listening to something that long, and so it needed Alex to come in and get the thing mixed properly. When we'd recorded it, I hadn't overseen this Photo © Lynn Goldsmith/Corbis young engineer who was really just a Nick Rhodes, Simon Le Bon and John Taylor tape-op, and he'd recorded the drums (left to right) at AIR Montserrat during the without compression and with no recording of Seven And The Ragged Tiger. consistency of level. That's why nobody was able to mix it — we did a mix at Good Earth, one at Eel Pie, one at the Gallery and one at the Power Station in New York with Bob Clearmountain, before Alex and Phil Thornalley finally did the job at RAK by replacing the drums with samples triggered out of AMS's. Well, can you imagine how much all that cost? "EMI was really beginning to get nervous about me and the band co-producing the album, and to be honest I was a little bit out of my depth, so the decision was taken to use Alex. I was devastated. After all, can you imagine what that would have done to any career hopes I had? After producing one single, someone was brought in to mix it with me and then I was gone? I'd be dead in the water. I went to the band's management and said 'You can't do this to me. I'll be finished,' and

file:///H|/SOS%2004-07/CLASSIC%20TRACKS%20%20%27The%20Reflex%27.htm (4 of 12)9/22/2005 8:58:02 PM


they said 'We're sorry, but this is a big project and we can't risk it, so you're out.' Then I canvassed the band members and they all appeared to want me involved, but they weren't prepared to stand up to their managers or to EMI, so I went to Alex and he said 'From what I can see you're like a sixth member of this band. I won't do it without you.' He needn't have done that — from a financial standpoint he would have been better off if I wasn't involved, but that shows what kind of person he was; an absolute gentleman. But he also wasn't an idiot, and this signalled a recognition on his part that I was an integral part of Duran Duran's work process at that time. I didn't get writing credits and I don't believe I ever deserved writing credits, but I know I helped them to come up with the ideas during that time. "Alex didn't lord it over me, and he never thwarted my creativity or my input one little bit, but I relied on him to make sure that everything was being done to the right technical standards. For instance, when we started work on 'The Reflex' and adopted the drum pattern from the demo, I knew the part much more intimately than Alex. I therefore told him what the groove was about, what I felt we needed to do with the bass drum, the snare and the hi-hat, and how I heard the groove working — when you're producing a drummer, you've got to relate to what he's doing with each of his limbs, because the groove's success depends on how he interweaves the various elements. So, I conveyed that to Alex, who was quick on the uptake, and he then took charge of getting the right sounds. Since I was more on the band's side of the fence, making all of them more aware of the rhythmic quality of their performances, he provided us with a perfectly balanced production approach."

Contrasting Abilities While Ian Little loved every minute of his interaction with Alex Sadkin, he also enjoyed working with band members whose respective musical abilities stood in sharp contrast to one another. "Nick was a soulmate of mine, insofar as he could play something totally out of tune and like it just as I could," Little says. "If, for example, you listen to a lot of the synth parts on 'The Reflex', they're borderline in tune, yet they work. He was more concerned with the effect a part had on the track than whether or not it was perfectly in tune, and that has always been my approach. You can't have things that stick out like a sore thumb, but in the same way as a blue note is bent off what it should be, you can play notes that aren't traditionally meant to be there. And if they create a bit of tension and have the right effect, then so be it. I suppose what I'm saying is that Nick was not a classically trained pianist. His ability was in his taste. He knew what he wanted to hear and I think that's what it's all about. It's about having the vision and the ideas, and knowing when something does and doesn't work. "Andy, on the other hand, was the most traditional of the bunch. He had served his time touring air force bases as part of a rock & roll outfit, so as far as musical ability was concerned he could have played in any number of bands, whereas I don't think that was necessarily the case with the others. They needed each



other, they needed the vehicle that Duran Duran presented them with. Roger was a very competent drummer, he could keep good time, and John was somewhere in between Andy and Roger. He could sometimes amaze in terms of the bass lines he'd come up with, but I don't think he'd ever stand up and say 'I'm a great technician.' He could do what he had to do. "One of the things that intrigued me about all Those were the days: Ian Little as of them was that they didn't have an he appeared in the pages of Melody encyclopedic knowledge of rock & roll. They Maker at the turn of the '80s. knew what they liked and they knew what was going on, but it was clear that their knowledge of music history had plenty of holes. My image is of Nick as the arty-farty one, Andy as the rock & roller, Roger as your down-home boy who had learned how to play the drums, John as the no-nonsense musician, and Simon as something of a misfit in that regard, with a less of a musical background."

The Days Before Auto-Tune Indeed, while the band's rhythm section tracked various parts, Simon Le Bon would often sit around until a song's melodic structure took shape and he felt comfortable hearing a particular chord sequence. At that point, he would vocalise and contrive his own melody lines before later writing the lyrics. "Out of the five of them, Simon was the least involved and the least active," Little confirms. "Although he was always chucking in his two pennies' worth while the others were playing, he'd tend to be ambling around the room. He'd 'la-di-da' and maybe have a few words or the odd line here and there, but he wasn't terribly keen on committing himself to his melody until he knew what lyrics he wanted to sing. If he could avoid tying it down, he would, and I think that was probably the best way for him to work. He wrote all the lyrics on his own, nobody else was involved, even though the others would tell him if they thought something was terrible. "There are some vocalists — including Ferry and, at the most extreme, Bob Dylan — whose voices aren't very tuneful but have plenty of character, and Simon falls somewhere between the two. He doesn't have a very characterful voice although it's certainly recognisable as him. It's a bit nasal and a bit forced, but I admire the way in which he stuck at it. He had to really work hard to develop a style, and eventually people grew used to him — he knew what he could and couldn't do. He is not a naturally gifted singer, and as I'm sure he himself would admit, he doesn't have great pitch. It's not unusual for him to sing out of tune, so when I worked with him we would use quite a lot of effects on his voice; mainly Eventide Harmonizer with a very small percentage pitch-shift up or down or both, file:///H|/SOS%2004-07/CLASSIC%20TRACKS%20%20%27The%20Reflex%27.htm (6 of 12)9/22/2005 8:58:02 PM


in addition to the normal step delays and reverbs and possibly even some chorus. Remember, these were the days before Auto-Tune. By making the pitch ambiguous, the Harmonizer helped disguise the fact that his singing was flat. "A major issue for Simon was that if nothing on a track really defined his melodic line, he would struggle. For instance, with some of today's R&B material or even Prince's stuff from Sign O' The Times where there's very little to pitch against, Simon would be lost. He needed something like a pad to which he could link himself. I mean, I can't sing in tune to save my life, so the fact that he could pull himself close enough to be acceptable The writing sessions for Seven was pretty amazing. That's not an easy thing to And The Ragged Tiger took do. I don't think he lacked objectivity in terms of place in a French chateau, and were recorded to an RAK mobile his own output, and I don't think he would have truck, as Ian Little's diagram been at all musical if he couldn't hear his own shows. limitations, but one of the things about Simon as a person was that he didn't want to show any sort of vulnerability, so he'd never come into the control room after doing a take and say 'Wow, am I out of tune!' Instead he'd go 'Cool, how's that?' He'd always try to put a positive spin on it, and yet if you said 'Simon, go and do it again,' there was never a problem. He was realistic. "We'd do three or four takes of his vocals and comp words, lines and half-lines. Still, listen to the way in which Nile Rodgers sampled and used Simon's 'Reflex' vocal when he did the mix for the single. Samplers were very, very new back then and we weren't using them, but for the intro on that single version Nile sampled Simon's singing and pitch-shifted it really low. I'd never heard a vocal pitch-shifted in such an overt way. What's more, we got some black session vocalists in from New York for 'The Reflex' — BJ Nelson and Michelle Cobbs — and I told them to sing like little girls: 'La-la-la-la-la, la-la-la-la-la...' Well, Nile picked on that part and stuck it right at the front of the song. That was all part of the funky vibe."



Genius Of The Peas In addition to tracking the drums, bass, rhythm guitars, key melodic guitars, synths and backing vocals for 'The Reflex' at AIR Montserrat, percussion was also provided by way of Nick Rhodes' keyboards and the ingenuity of one Raphael Dejesus. "Raphael was the genius," asserts Ian Little. "He had a rubber wrist, and he could do things with a film-canister shaker that you just wouldn't believe. He would take those little 35mm plastic film canisters and fill them to different levels with variable-sized dried peas and beans to produce a variety of sound and pitch, as well as different speeds of shaker, because the more peas there are inside, the quicker the recoil, and vice versa. This guy had it all down to a fine art. Depending on the beats-per-minute and how busy we wanted him to be, he would choose his weapon, and it was just magical. He would be in some sort of zen state when he was doing his thing and it was just like applying glue, covering up all the jerkiness and all the stiffness in the rhythm track. I've been an addict of shakers ever since, but I've never found anybody like him."

Reflex Response Harbouring a desire to be taken more seriously in the clubs, Duran Duran liked the idea of being a more danceable band by the time of their third album rather than just a synth-based outfit. To this end, just as 'Union Of The Snake' kicked off with a 'Let's Dance' bass drum pattern, so 'The Reflex' lent itself to a much more rhythmic feel. "If there was something really irreplaceable on the two-inch tapes that we had used in the South of France, we'd build up the tracks on there," Little states. "However, 'The Reflex' was started from scratch because it was a song that needed excellent technical quality with really sharp drum sounds and so on. That is, aside from the steel-drum part in the bridge section of 'The Reflex', a synth part that Nick found. He had a Jupiter 8 and I know we tried a Fairlight at one point, tooling through all these presets and not liking any of them. Neither of us could be bothered getting into editing, so we just left it. That's what he was like and I was quite happy to do that. What we would tend to do, rather than spend hours and hours editing a sound, was find a sound that we liked but that had something lacking. We'd then try to find another sound that would make up for the deficiency, before combining the two and layering things up. "Some people learn how they can edit synths down to the nth degree and then they try to sculpt the sound that they want to perfection, but I never had the patience for that and Nick certainly never had the patience for anything like that. I remember spending hours with him, going through these sounds, until we found something, and there'd be unrestrained joy. It was just wonderful. We'd get all excited, jump around like kids and then figure out where we were going to use it. Or how we were going to use it. Well, whenever I hear that steel-drum part it always brings a smile to my face because it's so out of tune. Steel drums always are, but it was exactly right in terms of rhythm and tone. So a wood-block sound



was mixed in to make it even more percussive and, Bob's your uncle, it did the job. "One of the things that I did was get a pad synth sound, put it through a Drawmer gate on an external trigger, and then have the gate triggered off a percussive part like a side-stick which was already being played or programmed specifically to trigger the Ian Little today. gate. Sequencers were still pretty new and cumbersome in those days, so this provided an alternative way of getting a pulsed synth sound, but the type of sound was also very, very different. If it's sequenced, each note is being triggered individually, whereas a pad that is held down for a sustained sound that is then gated and triggered is a totally different animal, because there's no attack at the front of each note other than that provided by the gate opening. The Drawmer was the first gate I used that had frequency selectivity, and that meant you could really isolate the part of the rhythm that you wanted to accent. What's more, the attack and release times on the gate could totally change the character of the pulse that you heard. This was something that I introduced to Nick and you can hear it both on 'Is There Something I Should Know?' and 'The Reflex'; more like an organic version of a sequenced line, being more rounded while boasting a certain degree of randomness."

Out Of Tune Once Roger Taylor's drums and John Taylor's bass had both been recorded for 'The Reflex', it was time to track Andy Taylor's lead guitar. However, a few minutes into the first run-through Alex Sadkin and Ian Little looked at one another across AIR's console and stopped the take because the tuning was out. Andy quickly retuned his guitar and began playing from the top, only for the same thing to happen, and it was only after this routine was repeated several times that the true cause of the problem was discovered: the tape machine was varying its pitch as it ran through the reel. "Anything that was in tune at the beginning was out of tune by the time we got to the end, and if Andy then tuned up at the end he would be out of tune when we returned to the beginning," Little explains. "We went on like this for a long while before we told the studio that their machine was knackered. Their technicians then tested the machine and insisted it was fine, before we eventually persuaded them to sit down and listen to it carefully with us, at which point they had to concede that, although their equipment illustrated it was consistent across the reel, this just wasn't the case. As the reel decreased in size on one side and increased on the other, the tension arms should have adjusted to keep the reel tight and to keep the playback speed consistent, but this wasn't happening, and



eventually the studio had to fly someone out from MCI in Miami to fix it. "It meant we had a torrid time trying to get the guts of that track down, and in a sense it relates to the in-joke about the steel drums: the girls' backing vocals were tuneless at the very best but they were quirky and they did the job — theirs was a rhythmic part — and the steel drums likewise. I often think the gestation period of the track was so full of these tuning issues, they just became part of it. There were tuning issues all the way through it. Even Simon's vocal, some of the harmonies that he did were very dubious, but it all seemed to come together in the end. Just as I think of a blues bend as being sideways, 'The Reflex' to me is a horizontal track. Still, I have to say that I personally don't give a toss about tuning if the character and atmosphere are right. The thing about Duran Duran is that they knew what they were doing, and they were all fantastic individuals and fantastic characters. The band was driven by that quirkiness, and I personally think that when they started to break up and work with other people is when they lost that magic."

A Faceful Of Garbage After Pete Schwier had perfomed the engineering chores in Montserrat, Alex Sadkin wanted his RAK sidekick Phil Thornalley to fulfill that role during the final sessions at EMI 301 in Australia, where Simon Le Bon's vocals, Andy Hamilton's saxophone, and a few synth and guitar overdubs were to be recorded prior to the mix. Nevertheless, the cost-conscious record company initially balked at this suggestion. After all, £20,000 had been eaten up simply flying Sadkin, Little, the five band members and their two managers first-class one-way to Sydney, and EMI didn't want to spend £900 a day for Thornalley's services. Instead, a succession of three local engineers were hired to sit behind 301's Neve console, none of whom fit the bill. "The first guy was useless, so we got rid of him within a couple of days," Little recalls. "However, the guy who replaced him would hear us talking, saying 'It would be good to hear that through an amp,' and instead of setting up a mic he would just sit there looking at us, scratching his balls. We'd then have to wait for three minutes while he tried to organise things. So, we got fed up and told him to get lost. Then the third guy came along and his habit was that he'd play the track through to the end and hit Stop, and when we'd discuss it and ask him to play it again he'd say 'Oh, I haven't rewound it.' We'd say 'Don't you think we're going to want to hear it again? Weigh up the odds!' This kept happening, so that same night Alex and I stayed behind in the studio, attached a cardboard box to the ceiling directly above where the guy sat, filled it with the smelliest, wettest, dirtiest rubbish we could find, and attached the closed lid to a piece of string that went across the ceiling and hung down in front of the sofa where Alex and I sat behind his chair. The next day we came in and we sat there, the bloke played the track and he stopped it, and we said 'Play it again.' Then, just as he said 'I've got to rewind...' I pulled the string and saw him look up and get a faceful of garbage. That was him out of the door. He was furious, but at least EMI agreed to get Phil Thornalley."



Taking It Seriously After Seven And The Ragged Tiger, Ian Little went on to produce Belouis Some's debut album Some People — including the classic 'Imagination' — and Sparks' Pulling Rabbits Out Of A Hat, before in the late '80s family concerns kept him out of the music business for a number of years. Since his return to production last year he has joined forces with manager David Ravden of MGR and has been working with Jez Larder at Skyline Studios in Epsom, developing new bands and songwriters including Nathan Quist. Yet he still values the lessons learned from the Ragged Tiger sessions, and in particular, the example set by Alex Sadkin. "The reason a producer gets royalties is because he takes responsibility, and Alex was one of the people who taught me that during the course of the Ragged Tiger album," Little remarks. "He came to me and said 'Look, you and I are responsible for how this turns out. You've got to take this seriously.' For a while I hadn't done this. I'd been having a laugh, I had really enjoyed myself, but I hadn't taken it seriously in that repect. Alex, meanwhile, had the weight of the world on his shoulders, and I did eventually share that with him. The budget was half a million quid, and at the end EMI came to us and asked 'Is the record any good?' How did I know? I'd been listening to it for eight months." Towards the end of the Seven And The Ragged Tiger sessions, the impending release date necessitated Ian Little working in one studio at EMI 301 in Sydney (see box, left) while Alex Sadkin toiled in another. Indeed, one record company deadline had already been passed by the time they got around to mixing at the same facility. "In those days, EMI had to lock out the pressing plant for three days to create enough product for the stores by the scheduled release date, and it was therefore a big deal for them to make that booking. Well, we went through two deadlines, and by the time we were approaching the third one EMI said 'If you don't deliver on time, the record won't be released for Christmas, and that is not going to happen.' Having been quite flippant about the whole thing, Alex and I had sailed through the first two deadlines, but we now knew the third one was for real, so we actually flew back to London with the compiled masters and headed straight for the cutting room. It was a close call, but we made it..." Published in SOS July 2004






Demo Doctor

In this article:

Demo Doctor

Relay Doctor's Advice: Peaks And Listen to Readers' own Published in SOS July 2004 Compression Black Water Echo Print article : Close window QUICKIES

recordings

Technique : Recording/Mixing

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

Relay Venue: Home Equipment: Fostex D2424 multitrack recorder, Soundcraft Ghost mixer, Audio-Technica AT4033 microphone, Focusrite Penta channel strip, AKG drum kit microphone set. After a favourable mention two years ago in the Quickies section, this heavy rock outfit deserve a bigger review for their latest efforts. Since the last demo, they've ditched their trusty old Alesis D4 drum machine and got a real drummer and, more importantly to their fans, they've lost a male vocalist and acquired a female one! However, the emphasis is still on classic rock in the style of bands like Rainbow and Whitesnake. The guitar parts on this CD are especially wellrecorded. Sticking with a tried-and-tested Marshall JMP1 guitar preamp, guitarist Steve Tsoi coaxes some marvellous classic rock lead and rhythm sounds from his setup. The second track, for example, has a fantastic lead guitar sound, full of sustain and rich in harmonics, which an echo effect suits perfectly. On most of the CD the rhythm guitar is double-tracked and panned hard left and hard right, giving the mix a lot of stereo width. I would, however, take issue with the clean guitar sound at the start of the second mix, which is rather thin. I'm guessing that all the bass has been attenuated to allow the clean, picked guitar to cut through when the fat, overdriven rhythm guitar comes in later on. However, it doesn't follow that the file:///H|/SOS%2004-07/Demo%20Doctor.htm (1 of 5)9/22/2005 8:58:05 PM

Demo Doctor

picked guitar has to have the same sound at the beginning of the track — you can bring in the bass cut when the second guitar comes in, either by using EQ automation in a software sequencer, or by hand on a hardware mixer. Listening to the drum sounds, too much compression has been applied to the snare on the first mix, which is crushing the life out of it. I suspect that this heavy compression has been used in an attempt even out the level of the snare hits. Instead, I would suggest importing the drum tracks into a software audio editor, scanning the snare track and bringing down the level of any prominent peaks before applying any compression. Triggering a sampled snare from the original snare recording and layering the two together is also an option. Singer Alison's vocals work well with the rest of the band and could help set them apart, as there aren't many female rock vocalists around at the moment. Her voice is powerful, but it's occasionally mixed a bit too low. At the beginning of the second song the lead guitar obscures the vocal when it comes in, when the voice should be as loud, if not a touch louder, than any of the lead guitar parts. However the interaction of harmony lead guitar and vocal parts at one point in this song is simply excellent and perfectly balanced. As to the other mix levels, I think the keyboards are generally too loud, but they don't spoil what is essentially a decent rock album. www.relay-online.com

Doctor's Advice: Peaks And Compression A few demos this month seem to be struggling with the various processes of mastering and post-production, whether it be the use of compression, limiting or equalisation. Remember that until recently, these skills were the best kept secret in the recording industry, so you can't expect to get it right overnight. The kind of equipment used by professional mastering engineers is well beyond the reach of the home studio user, but nearly all computer-based sequencing packages are supplied with multi-band compressor and EQ plug-ins. However, an oftenoverlooked asset is a good set of PPM meters to watch your mix levels. They can be very useful for spotting the occasional, brief peak signals that compromise your dynamics processing later in post production. The most common sources are errant snare drum hits, percussive or fast-transient vocal sounds and highly resonant synths. By adjusting the level of these peaks before applying compression to the whole mix, more even results can be achieved.

Black Water Echo Venue: Home Equipment: Pentium 133 PC, Creative Labs Soundblaster soundcard, Syntrillium Cool Edit Pro editor, Alice desk, Marantz amp, PMC TB1 monitors, Red 5 vocal microphone, PZM and various capacitor/ dynamic microphones, homebrew 'opto' mic channel, Drawmer M500

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Demo Doctor

dynamics processor, Alesis Quadraverb GT, Nanocompressor and Microverb, Klark Teknik spectrum analyser. The big-sounding mixes that this duo create owe a great deal to the expansive delay and reverb treatments applied to the guitars. These are handled with great confidence by guitarist Steve Satori from the opening of the first track, with its ping-pong delay, to the final fade-out. On the first track, the echoed guitar sets up the whole rhythmic swing of the piece, and this delay has been fed through a reverb to add depth. In the bad old days, when spare recording tracks were at a premium, I always liked to print the guitar with the effect to a stereo track for this style of playing. Of course, this meant getting the effect's dry/wet mix balance just right. Nowadays, with many more tracks available, when using outboard effects it's safer to record the stereo effect return and the dry signal separately. This keeps your options open for the final mix, provided that you've stored your presets on the effects unit! In other areas, the guitar is reminiscent of Robert Fripp and his penchant for harmonising overdriven, sustained lead guitar lines in a rather beautiful and melancholy fashion. One trick which Steve takes from the old school of recording is to apply loads of reverb to the guitar using a prefade send and then begin the song with only the effects return faders turned up on the desk, gradually fading up the dry guitar signal. Steve asks me to point out any glaring errors in his mixing and production, and the most obvious one for me is that the lead vocal is mixed too loud throughout. I also find the vocal delivery a little too relaxed, but it's hardly suprising given the mellow nature of the backing tracks. Mixing the vocal at a lower level with a slightly more edgy EQ (a slight boost at, say, 5kHz) will help in two ways. Lowering the level of the vocal will prevent it being the sole focus of attention and enable the listener to engage more with the songs as a whole. Secondly, the EQ on the voice will add energy to the performance without losing the clarity of the voice at this new, lower level in the mix. Having compared his mixes with other commercial releases, Steve thinks this CD is a bit lightweight and was thinking of applying multi-band compression. However, it compared favourably against the three commercial CDs I played on my monitor system. Interestingly, all the other recordings had more upper midfrequency content and sure enough, if you boost this recording around 2kHz by only a few dBs, it seems to bring the mix image forward, adding energy and subjective loudness in the process. This is probably all that's necessary.


Demo Doctor

QUICKIES

Matt Gibson Matt has just finished a HND in music production and is now pursuing a career in songwriting. His music is electronic and pretty experimental in nature, showing a flair for production techniques involving the use of filters and a good ear for samples. In commercial terms, this means he's placing himself outside the mainstream pop market but well within the soundtrack genre, where his atmospheric work would be suitable for advertising, film or theatre. He says that some post-production work has taken place on the mixes, and I found them all a bit lowfrequency-heavy at higher monitoring levels. This is something which a simple lowfrequency roll-off around 60Hz could rectify. Matt might also consider checking the placement of his speakers and the general acoustics of his studio to make sure he's getting an accurate impression of his mixes. Sometimes a problem with the monitoring can be solved simply and cheaply, as Paul White and Hugh Robjohns' Studio SOS features repeatedly demonstrate. The subject of monitor placement is also covered by Hugh in the May 2004 issue's Q&A section of SOS, which is well worth a read. Matt's friendly web site (www.matthewsmusic.com) features a picture of him playing the guitar — which he slots into the electronic music rather well — plus some MP3 samples.

Dare To Dream Heavy reverb has been correctly identified by this synth pop duo as a trademark '80s production technique. It's used on everything, but especially on a lightweight vocal that lends the music a certain vulnerability, which adds to the retro feel in quite an appealing way. However, this is a genre which also relied heavily on the dry, up-front kick drum and it's in the bass frequency range that Dare To Dream prove to be inconsistent. The third song is a good example of the lack of continuity here. It begins with plenty of power provided by a bass synth with a full-bodied sound, but as soon as the main backing track starts, this bass is swapped for a lightweight substitute and the whole mix drops in energy and becomes weak. The other two songs seem to lack depth in the bass frequencies and are surprisingly kick-drum light. Yet good points emerge in the choice of other sounds and general songwriting skills. On the third mix I especially liked the accurate Simmons drum sound, which was also a feature of the early to mid-'80s, and some of the synth sounds are very close to the Juno 60 and 106 keyboards of the era.

Kenneth Christ Kenneth (aka Brighton-based breakbeat merchant Dan Heath) feels that there is a lack of separation and detail in his mixes. On the evidence of this demo I can hear too many sounds vying for space in the crucial 800Hz - 2kHz frequency range,


Demo Doctor

which indicates that the keyboard arrangements need a bit more care. In addition, the widespread use of open filters is brightening the sounds up so much that they fight against each other to make an impact. Remember that a filter is a sort of equaliser and can be used subtractively to take the edge off a sound as well as liven it up. Some of these parts could have been more easily layered in the mix if only one or two key sounds were allowed to dominate over the others. The breakbeat sound itself is odd and there's none of the low-frequency depth to the kick drum that you'd find on a professional mix. I think this is the result of no more or less than his choice of drum sound — a better quality kick-drum sound or sample would make a bid difference. It might sound like I'm giving poor Dan a pasting here, but in fact this CD isn't far off the mark at all. It's tight, has some excellent moments and simply requires a little attention to the mix.

A Perfect Remedy With an opening track entitled 'Ground Breaking', this band are setting themselves up for a fall, but the quality of the vocal performance wins through. In fact, the vocals are at their best on the second song, where a tight harmony vocal lifts an already catchy chorus. However, there is the occasional audible dip in level caused by compression cutting in during the loud sections. Because the whole mix dips a little, rather than individual elements of it, this would seem to be the result of excessive compression applied in post-production, so this could be corrected if the master is taken back a stage or two. But elsewhere, this is a competently produced slice of rock in the style of bands like the Red Hot Chili Peppers. The beats have a strong groove and the guitarist teases some interesting sounds from his guitar setup, largely with the aid of overdrive, wah-wah and echo with a large amount of feedback. However, I did find the guitar sound a bit fizzy in the high frequencies and surprisingly lacking in those lower frequencies which help to knit the guitar in with the bass. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Future-proofing DP Projects

In this article:


Group Therapy Digital Performer Notes Recording & Playback Published in SOS July 2004 Quick Tips Re-using Consoles Print article : Close window Future-proofing DP Projects

Technique : Digital Performer Notes

This month, Performer Notes concludes its look at DP's Consoles, and suggests some ways of 'futureproofing' your Projects against the relentless flow of application and plug-in updates. Robin Bigwood

Last month, I explained how DP's Consoles could be used both for remapping MIDI data and as a source of data in themselves, as well as some of the ins and outs of buttons, sliders and knobs. This month, it's time to round things off, beginning with Consoles' Groups function.

Group Therapy Guess what? Console Groups let you group Control Items together so that they can be governed simultaneously by a Master Item. For example, you could assign three sliders to transmit controller messages to change filter cutoff on different channels of a multitimbral synth module. After grouping them and, optionally, creating one master slider for the group, all three sliders could be made to move simultaneously whilst dragging just one.

For dedicated Console makers, the sky's the limit. This Console, by Pieter Van Der Wal, controls aspects of a Yamaha 01V.

The way 'slave' Items react to a master is pretty flexible too — they can track a master Item's values absolutely, matching them exactly, or instead offset or scale the values, and even work with negative polarity, so that as the master's value increases, the slave's decreases. As is the way with DP's Consoles generally, the sky's the limit as regards complexity, and dedicated tweakers should be able

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to dream up some wild and wacky setups that have all kinds of MIDI controllers, note events, plug-in operations and external data sources linked to each other. Adding a Control Item to a group is simple: select it, choose Group from the minimenu and type a group letter (A to Z) into the dialogue box that appears. It's here that you set positive or negative polarity and decide whether to match, offset or scale the master value. In the latter two cases you also get to set the 'strength' of the relationship, as a percentage. All these settings can be made in the standard Control Assignment dialogue box, too, which always appears when you first create a Control Item. Setting an Item to be a master is also simple. Select it, choose Control Assignment from the Console's minimenu, assign a group letter, then click the Master radio button. There's one important consideration for master Control Items when they're sliders: the so-called 'null point'. You can think of this as a kind of reference point, and it basically allows relative values of masters and slaves to be set up. For example, if you set a master Item's null to a value of 100, and then set one of its slaves (in 'offset' mode) to a value A Grouping dialogue box, accessed via a Console's mini-menu. Groups allow Console of 20, moving the master to a value of 127 would result in the slave moving to elements to interact with and control one another. a value of 47. Nulls make master items send 'changes' in value, relative to the null, rather than absolute values, as measured across the entire range of the slider. Tricky stuff, maybe, but important for many Group setups. Incidentally, nulls can be set either by option-clicking the slider at the desired point or by typing a numerical value into the Control Assignment dialogue box.

Recording & Playback It's in the nature of some Consoles to just sit there, being used occasionally to send or re-route a particular type of data, while you get on with the serious business of putting together a track. Consoles don't have to be so passive, though — they can be integrated right into the heart of your sequence, interacting with it as a means to record data into tracks, or modifying data already present in tracks as they play back. It's important to note, though, that while Consoles aren't tied to individual tracks, their Control Items often are, acting as a source of data for tracks, and also following (and perhaps modifying or re-routing) the data during playback.



There are two ways to initiate data recording from a Console. The first is to record-enable any tracks assigned to your Control Item(s), put DP into overdub record mode, then hit Record. The second is to leave all tracks un-enabled but make Setting a Console slider's null (which is indicated by the little vertical lines) the Console the uppermost window before allows the relative values of master and hitting Record. In either case the outcome slave Control Items to be governed is the same — new data is recorded into reliably. appropriate tracks from any Control Item whose output is a track itself, or a MIDI target that is also, somewhere amongst your tracks, a track output. This sounds more complicated than it is. Just assume that the Console automatically 'connects' itself to tracks associated with any of its Control Items and you won't go too far wrong. When you're playing back a sequence, any Console Control Items associated with data types in play-enabled tracks will animate to reflect the current values of those data types. This not only gives excellent visual feedback of what's going on, especially with controllers, but allows sliders (or other Control Items) to be 'grabbed', at which point data in the track is ignored and the new 'live' values generated in its place. Let go of a Control Item and DP immediately starts dishing out the track's data once more.

Quick Tips Compared to many of DP's other windows, a hastily constructed Console can look pretty awful, making even a Pluggo plug-in interface seem stylish! However, Consoles can be made to look half-decent by using their built-in layout tools, accessible from the mini-menu. Certainly, a little effort spent on ergonomics and aesthetics should be worth it for Consoles you'll end up using a lot. When you set up button Control Items in a Console, you may notice the little 'M' (mute) option next to some of the value fields. This makes a button behave more like a mixer's mute button in that when it's released it doesn't send an absolute value but instead allows the data to return to the current value of its target controller type. This can work well during playback, when the button is set to control volume, filter cutoff, or some other fairly 'noticeable' aspect of a synth's sound (for example), allowing for momentary interruptions to the sound, very typical of many modern fragmented or glitch-oriented styles.

Re-using Consoles Having spent time honing your perfect Console, how do you save it? And how do you call it up in another sequence? Well, Console windows themselves don't have Load and Save functions, and there's no such thing as a DP 'Console file' that you'd ever come across in the Finder. Instead, Consoles are actually saved as part of a normal DP file — 'embedded' in it, if you like. This has some file:///H|/SOS%2004-07/Future-proofing%20DP%20Projects.htm (3 of 6)9/22/2005 8:58:09 PM


important consequences for how you can re-use Consoles that you've designed in different projects, and also how you can best create a 'library' of Consoles accessible at all times. Loading a Console from another project is done via the File menu's Load command. You navigate to the DP file that contains the Console you wish to open, click Open (or doubleclick the filename) and then choose 'Consoles' in the Load dialogue box that appears, before finally clicking OK. The Console is loaded into your current project, but may need some tweaking, depending on the degree of similarity of track names, MIDI destinations and so on, between the two projects.

Blockfish, from the digitalfishphones freeware Audio Unit bundle, is a great little plug-in that I've found forgets its settings. A quick screen-grab can help you to dial in those killer settings once more.

While this is perfectly straightforward, remember that to Load a Console the 'donor' Project must always be available. If you compress, archive or delete it, you'll lose access to any Consoles it contained. For this reason, it's worth thinking about building up a library of Consoles, and this can easily be done by creating a Project that you might call 'Consoles Project' (for example), and that you never use for anything other than Console 'storage'. Every time you created a Console in another Project you could then open 'Consoles Project', load into it the newly-created Console, then hit Save. A slightly irksome solution, maybe, but one that should help to ensure that your favourite Console creations are always available in one place when you need them.

Future-proofing DP Projects Applications such as DP, which use host-based processing to do their stuff, give to home- and commercial-studio owners alike a level of performance, flexibility and value for money that was unimaginable only a decade ago. What they don't provide, though, is either totally reliable, crash-proof operation, or very much guarantee of backwards (or, for that matter, forwards) compatibility. We run the gauntlet not only of operating-system conflicts, but of relentless upgrades and third-party software development efforts, and as a result life can sometimes get tricky, especially when all you want to do is make music. I doubt there are many Performer Notes readers who haven't at one time or another opened a lovinglycreated DP Project (even one they've recently worked on!) only to find that, well, it won't open, or comes up with plug-ins and sound files missing or wrongly set up. As we all know, this is a thoroughly depressing experience, so how do you avoid it, or at least get going again more quickly if it does occur?



There are a range of strategies you can employ — depending on how 'bomb-proof' you'd like your Project-recall chances to be — ranging from the sensible to the neurotic. It probably goes without saying, for example, that freeware and some shareware plug-ins are amongst the most disruptive and least reliable bits of computer code that swim into our lives, however good they sound. You perhaps shouldn't expect to always be able to open a Project and find your 'little' plugins in the same state as they were last time you saved — although it's certainly nice when they are! Although they only reference Consider trying to get to grips with plug-ins' own data, Clippings windows are patch Save and Load facilities, and be prepared nevertheless a superb to use them, both before closing projects, and organisation tool for all the files after re-opening them. You might also want to that DP Projects frequently collect. consider using OS X's screen- and windowgrabbing facilities to take snapshots of plug-in user interfaces, which could then be saved in the Project folder along with the Project file itself, and used as a reference if a plug-in fails to initiate correctly. Shift-Command-3 takes a picture of the whole screen, while Shift-Command -4 gives you a crosshair that you can drag over a smaller area, or, after hitting the space bar, a dedicated window-grabbing tool that snaps individual windows. Grabs are saved to the desktop as a default, and can then be dragged to a folder of your choice.

Another option for storing plug-in information, and also track-output assignments, patch information and locations and so on, is the Tracks window's Comments column. You can store a surprising amount of text here, and the odd annotation made whilst developing a Project could potentially save hours in the long run. Of course, all sorts of data can be referenced by DP's Clippings windows — pictures, photos, lyric sheets, you name it. See Performer Notes from way back in August 2001 for more on this superb DP feature. A more belt-and-braces solution, especially useful if you intend to put away a DP Project for a year or two, is to to try to render as many elements of your Project as possible into future-proof formats. Plug-in-laden tracks can be turned into 'straight' audio in DP in a number of ways, including via Freeze Tracks (see July 2003's Performer Notes), Bounce to Disk, or even a manual, real-time bounce to a new track. If, at some later point, you needed to load these audio tracks into another sequencer, you could either rely on the audio's time stamps to put them in the right place, or for even greater security make sure that individual track bounces always take place from the start of the sequence, so that manually lining up multiple tracks becomes easy. Finally, consider saving MIDI tracks as a MIDI file too, as they will then be much more widely compatible than if they were in DP's own file format.






Longhorn and Harbal

In this article:

Harmonic Enhancements Seeing Is Believing Even Better Balance? More Longhorn News

Longhorn and Harbal PC Notes Published in SOS July 2004 Print article : Close window

Technique : PC Notes

The Harbal EQ-matching program benefits from some worthwhile tweaks. We report them, and discuss some new information about Longhorn, the next incarnation of the Windows OS. Martin Walker

Back in SOS March 2004 I reviewed version 1.02 of the Harbal Visual Mastering System for Windows users. Harbal (which stands for 'Harmonic Balancing') is a stand-alone utility that runs under Windows 95, 98, ME, NT, 2000, or XP, and is designed to match the frequency response of a target track to that of a reference. I was impressed by its combination of resizeable main graphic display for Version 1.5 of Harbal offers a host of new detailed examination and editing of the display features, shelving as well as existing frequency response of any parametric equalisation, loop monitoring, popaudio track, automatic loudness up tips, and support for ASIO drivers. compensation to make EQ choices easier, and a well-chosen set of analysis files for comparison with high-quality commercial tracks.

Harmonic Enhancements Four months on and Har-Bal International have just released Harbal 1.5, with a set of new features and improvements. One of the most noticeable is the twin vertical sliders down the left-hand side of the main display. These control playback Volume (a new feature) and overall filter Gain (for raising the final level

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Longhorn and Harbal

of a track or making it match others in the context of an album). The latter was previously only available as a text box on the toolbar. However, far more useful is the set of three new buttons on the status bar that let you toggle the display of the peak, average and geometric-mean power plots of frequency response for both target and reference tracks. With all six curves displayed, Harbal can become cluttered and therefore confusing, so these options make a huge difference to screen legibility. You can still edit the frequency response on any of the three curves. Sometimes it's easier to work with the peak plots, which (for example) may show momentary harshness at a particular frequency more obviously (perhaps because this instrument is only heard for a short time in the track, and therefore won't show up in the average plot). Whichever curve you choose to edit, it controls a single overall filter response. Just as useful is a further button to toggle the three curves displaying the original non-EQ'd response, to show what's changed, and while this is visible a fifth button adds dots to the curve currently being played back, so you're sure which one you're listening to when performing A/B tests. Above the main display there's now a read-out of track statistics: source fileWindows Longhorn will feature windows with name, reference file-name, filter filevariable translucency so that you can still name (you can save your EQ see what's going on underneath. responses as filters, to use again as a basis for other tracks), plus overall Average and Peak levels for both source and reference tracks. These statistics really should have been in the initial release, as they make things a lot clearer. Harbal users can still start monitoring playback at any point during the track, by moving the horizontal fader above the main display, but there's now a new toolbar Loop button that, when pressed during playback, jumps back to your chosen start point and continues looping to the new loop point indefinitely, until you toggle it off or stop playback. I found this very useful for auditioning a specific section of a track, although it would be even easier if the slider were replaced by a slim graphic bar showing the track's waveform, in which you could define a start and end point, as you can in most audio-editing applications. The filter controls themselves have also been improved, with two further new toolbar buttons alongside the original parametric one, to let you switch the cursor to low or high shelf options. These make tilting a portion of the response in either direction much easier than before — perfect for taming excessive bass or treble, or for adding air or warmth to a track. Beware, though, of adding loads of bass to match that of a target track if yours has no significant level in that region. This is a weak point of automatic 'EQ rippers' and often results in distortion. file:///H|/SOS%2004-07/Longhorn%20and%20Harbal.htm (2 of 5)9/22/2005 8:58:15 PM

Longhorn and Harbal

Seeing Is Believing Also new is a handy set of 'tips' that pop up as you move the cursor across the spectrum, offering details about the various frequency bands, problem areas, and what to listen out for (see screen below). In addition to the default General tips, a further selection can be loaded encompassing a range of brass, percussion, strings, voices, and woodwind instruments, and you can also create your own tip files to remind you of particular problem areas in your mixes. Overall, these tips are useful, particularly for the novice, and my only reservation is that the longer ones often obscure the graph's frequency markings. Less immediately useful, but handy for analysis, are two new main display views. 'Frequency Response' shows the current EQ curve rather than your track's response, and can be edited using the various cursor tools, while the Impulse Response provides a fixed display that's only really useful in showing the difference between the original Linear Phase and new Minimum Phase filter options — the audible difference between these two Alt-Tab in Longhorn will offer a tilted 3D view is extremely subtle by the way, but of currently open applications, so you can see their contents. some people find the 'pre-ringing' of linear phase filters objectionable when they're using high Q settings. Now they have another option. Harbal now supports ASIO and DirectX driver formats as well as MME, and you can now choose a soundcard output within the utility, rather than using the default Windows one. Compared with version 1.02, the time the software takes to switch between the untreated audio and the EQ'd version has also been halved, which makes the utility feel more responsive. An 8192-point EQ algorithm will always involve considerable latency, but 93ms at 44.1kHz is still far from instant, making subtle A/B comparisons somewhat frustrating, since you can't always hear exactly when the changeover occurs. Although this latency could be reduced at the expense of higher CPU overhead, a visual indication of the exact changeover point would provide welcome feedback, and the developers have taken this suggestion of mine on board for possible inclusion into a future version. Other, smaller, improvements include background recording of the final EQ'd file, so that you can start analysing and editing another track; recent file history lists; support for mono files; a Match Loudness function to instantly bring the overall level of the source track to that of the destination, using the Gain slider control; Batch Analysis so you can grab a load of tracks and have them all converted into reference responses in one go; and the cure of a few small bugs.


Longhorn and Harbal

Even Better Balance? Overall, although it's still initially harder to get to grips with than some rival 'EQ ripper' utilities, I still prefer Harbal, with its more considered approach to improving the final sound of an audio track, the detail of its manual and tutorials, and its helpful on-line user forum hosted by Har-Bal's professional mastering and research engineers. It's a serious tool that can teach you a lot about why your mixes don't sound like commercial tracks, at the same time as improving them. It's a shame that the promised plug-in support isn't yet in place, so that, for example, it would be possible to load a multi-band compressor and normaliser and use Harbal's EQ as part of a mastering plug-in chain, but apparently so many new features were added to this release as a result of user suggestions that this has been delayed. However, that support is still promised in due course, as is a VST plug-in version, and a Mac port will possibly become available by the end of 2004. The version 1.5 upgrade is free to existing users, while the program itself is still a very reasonable $95 for new ones. An 8-bit demo version is also available as a free download from www.har-bal.com.

More Longhorn News I've mentioned Microsoft's Longhorn operating system a couple of times already in PC Notes (most recently in August 2003), but the latest indications are that we won't see it until at least mid-2006. However, its much-vaunted transparency and 3D graphic engine (code-named Avalon) — which, for the first time, mean that for the full experience even office software users will require an up-market graphics card — have been confirmed as optional. Three different graphic interfaces will be offered, for a tiered 'user experience'. If you have a high-end graphics card with at least 64Mb RAM you can try the full 'Aero Glass' experience, with variable-transparency windows and 3D shading features. I can see this being popular with PC musicians who, for the first time, will be able to view their sequencer arrange pages scrolling smoothly by while they're working on superimposed and semi-transparent editing screens 'above'. The mid-level 'Aero' is more suitable for cards with 32Mb of RAM, and both Aero graphic options will also require DirectX 9.0, AGP 4x and a Longhorn graphics driver. However, those whose graphic cards lack these features will still be able to use the 'classic' interface, designed to mimic the corporate look of Windows 2000. Microsoft are also hoping to incorporate new laptop-friendly features, such as an additional user interface that allows instant access to film and audio playback without booting up Windows itself, to make the laptop as easy to use as a file:///H|/SOS%2004-07/Longhorn%20and%20Harbal.htm (4 of 5)9/22/2005 8:58:15 PM

Longhorn and Harbal

portable DVD player. Industry experts are already predicting that by the time Longhorn is released the vast majority of PCs will be able to run it, although that does seem to presuppose that they incorporate a graphics card that runs about three times as fast as those available today, a minimum of 2Gb of RAM and a 4-6GHz 64-bit processor. Mind you, most PC processors being sold by the end of 2005 are likely to be 64-bit capable, so Microsoft are encouraging hardware developers to start writing 64-bit drivers for their wares early. With two years to wait, hopefully our Longhorn transition will be comparatively smooth. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



MIDI config problems in OS X; Garageband v1.1 update

In this article:

Garageband 1.1 iTunes 4.5 When MIDI Goes Bad If It Bends, It's Funny. If It Breaks...

MIDI config problems in OS X; Garageband v1.1 update Apple Notes Published in SOS July 2004 Print article : Close window

Technique : Apple Notes

The launch of Mac OS 10.4, codenamed Tiger, is just around the corner, but in the meantime there's the 1.1 update of Garageband to be going on with. We reveal the improvements, as well as investigating MIDI configuration problems in OS X. Mark Wherry

As this year's Apple World Wide Developer's Conference (WWDC) draws closer, there is much anticipation surrounding the hardware and software developments that will be announced. It was widely expected that, at least on the software front, Apple would present something along the lines of Mac OS 10.4; and, indeed, perhaps as bait to draw more developers into the conference (tickets start at $1295 with early registration, by the way), the company announced on May 4th that Jobs' WWDC keynote would feature a preview of Mac OS 10.4 — codenamed Rusty-spotted Cat (http://members.aol.com/cattrust/rusty.htm). I'm joking, of course, but Apple really are sticking with the feline codenames and 10.4 will officially be referred to as Tiger. Roar. Other than this crumb from Steve Jobs' table, there's currently no other information available about what the feature set of 10.4 will include, although, as usual, there are plenty of rumours. Aside from software, though, the most interesting aspect could be improvements to the Power Mac G5 series, and whether Jobs' promise of 3GHz G5 processors one year on from last year's conference will hold true.

Garageband 1.1 Apple released a version 1.1 update to Garageband this month, and to find out

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what's new and improved in the release I spoke with Apple's Product Line Manager for Audio Applications, Xander Soren. Apple had consumer-oriented intentions for Garageband, but Soren commented that they had been surprised at the positive reaction to the program in the pro audio community. While Apple have products such as Logic Express and Pro for customers who need more functionality than Garageband offers, there are apparently many professional customers using it because of the simplicity of the interface. The purpose of the 1.1 release was really to "polish the corners" of the application; so, instead of any significant new features in Garageband, the update provides allround improvements to the existing feature set. One notable area this applies to is support for Rewire, which Garageband 1.1 enables audio previews for is officially part of the 1.1 update, MIDI loops to be used as actual audio loops. having first been introduced in 1.01, as Here you can see a MIDI loop dragged from mentioned in last month's Apple Notes. the browser, and, below it, the audio version, created by Option-dragging the same MIDI Apple have apparently worked with loop from the browser. Ableton and Propellerhead, amongst other companies, to ensure smooth operation with Live and Reason, and are keen to point out that they have one of the simplest Rewire implementations for users to deal with, since it just works: no configuration required. Apple also worked with third-party Audio Unit developers to ensure that Garageband works well with the most popular Audio Units. The user interface for track handling has been improved, and it's now possible to drag tracks in the Track List to change the order in which they are listed. Also, if you ever wanted to copy the settings of one track in a Project to a new track you've just created, you can now use the Duplicate Track command, which duplicates all the settings of the selected track into a new one, but without copying the actual content of the track you're copying. Performance was an issue that was widely discussed concerning the original Garageband release, presumably because of the number of effects active by default when you create new tracks (many people have confirmed that Garageband is no more or less efficient than Logic). However, since the MIDI loops in Garageband always included an audio preview anyway, these audio 'loop previews' can now be added to the song instead of the MIDI version, by Option-dragging the MIDI loop. The advantage here is that the effects are already bounced into the audio, saving overhead, but the disadvantage is that the MIDI loop becomes just like any other audio loop in Garageband, meaning that you won't be able to perform any additional MIDI editing on it. Unfortunately, there's no way to switch between audio and MIDI versions of loops once they've been added to the song, but this is still a welcome new feature for those who use Garageband's MIDI loops intensively.



iTunes 4.5 Just prior to the release of Garageband, Apple also updated iTunes to version 4.5 and added a handful of new features to the iTunes Music Store, which celebrated its first birthday on 28th April. The Music Store has definitely been a success for the company in terms of marketing and publicising the idea of legitimate music downloads on the Internet. With sales of over 70 million songs in the first year, it will be interesting to see if Apple's one-vendor solution continues to appeal to the market, with increasing competition from the likes of Microsoft and Sony. The most recent incarnation of the iTunes Music Store now incorporates music videos and film trailers — and, in conjunction with the new iTunes 4.5, users can now share playlists (but not the content, mind) with other users. But perhaps the most intriguing addition to iTunes 4.5 for audiophiles is the new Apple Lossless encoder, which manages to store audio data in half the space normally required. The Apple Lossless encoder is featured in QuickTime 6.5.1, meaning that you can take advantage of this new codec even if you don't use iTunes. Staying with the subject of codecs for a moment, iTunes 4.5 can also now import unprotected WMA (Windows Media Audio) files, although rather than supporting WMA playback natively, iTunes transcodes the files into AAC. Citing Apple's commitment to "discourage music theft while preserving fair personal use rights," the company have reduced the number of times you can burn a single playlist from 10 to seven, although you can still burn single songs an unlimited number of times (in different playlists) and copy them to an unlimited number of iPods. Still, if Apple taketh away with one hand, they giveth with another, since you can now play content purchased from the iTunes Music Store on five Macs, up from three. We'll be looking more closely at the FairPlay copy protection system, as well as the new lossless encoding, in a future Apple Notes column.

When MIDI Goes Bad In an effort to win a prize for the largest Core MIDI configuration, for the past few months I've been trying to help tame a system comprising 12 MOTU MIDI Express XT USB MIDI interfaces. These devices were connected to a PowerMac G5 via three USB hubs: one hub connected to the actual computer, with two additional hubs branching from this to connect the MIDI interfaces. After plugging these devices in one by one, to establish the order in which they would be identified by Core MIDI, as shown in OS X's Audio MIDI Setup utility, I always wondered what would happen if the order somehow changed — and this month I found out. The system in question ran Nuendo 2 as the sequencer, and a side problem we'd been experiencing was that when Projects consisted of a fairly large number of MIDI tracks — 250, for example — the Project could take several minutes to load. While I initially suspected Nuendo to be at fault here, switching off the MIDI interfaces would cause the same Project to be loaded in a matter of seconds. And creating a smaller Project with fewer MIDI tracks also resulted in more bearable loading times. The hypothesis is fairly obvious: a larger number of file:///H|/SOS%2004-07/MIDI%20config%20problems%20in%20OS%20X%3b%20Garageband%20v1.1%20update.htm (3 of 5)9/22/2005 8:58:19 PM


connected MIDI tracks in a Project results in longer loading times — and Core MIDI itself probably isn't to blame since Logic doesn't have this issue.

Because Emagic's Unitor and AMT MIDI interfaces chain together with only a single USB connection to the computer, Core MIDI sees this setup as a single 64-port MIDI interface.

Getting back to the first point, the system in question underwent a hiccup this month, such that the order of the MIDI interfaces mysteriously changed. But more than this, the interfaces were seen by Core MIDI as if they were different interfaces than those previously connected. Why would this be a problem, I hear you ask? Well, Core MIDI assigns a unique identifier to each MIDI interface in the system, which an application can make use of to identify specific interfaces. If these identifiers somehow change, an application, such as Nuendo, won't re-load your Projects correctly because as far as it can tell, the interfaces used by the Projects no longer exist! This is the point at which the long Project loading times really became a problem, with loading times roughly doubling. This was probably because the Project first looked for all the interfaces that didn't exist any more, before trying to connect to the new interfaces. When the Project did eventually load, Nuendo's Pending Connections window revealed that, indeed, it could no longer see the same MIDI interfaces; but because the interfaces were listed as a series of cryptic numbers, there was no way (other than the tried and tested method of poke-and-hope) to remap the Project to the new interfaces. And even if we could have discovered the best way to remap the interfaces, the average Project loading time would have been four minutes for the remapped Projects!

If It Bends, It's Funny. If It Breaks... Fortunately, Core MIDI architect Doug Wyatt was kind enough to offer some advice about what could be causing these problems, and it definitely seemed to point to an issue with the drivers for the MIDI interfaces. In theory, if each USB device has a unique identifier (such as a serial number), the order of the interfaces should never change, no matter what their location on the USB interface. To confirm that the MOTU interfaces could indeed be uniquely identified, I switched off interfaces five through seven and then turned on interface six again. Audio MIDI Setup confirmed that interface six was active again, meaning that the system could identify the correct interface, rather than finding the next empty slot. While I've yet to find a solution with the MOTU interfaces, after swapping back to eight Emagic MIDI interfaces (one Unitor8 and seven AMT8s) and sacrificing the extra 32 ports of MIDI, there were two interesting points to note. Firstly, the Emagic driver makes Core MIDI see the multiple interface setup as one giant 64port MIDI interface, since instead of connecting each interface via USB, you connect only the first via USB and then chain the other interfaces via the Unitor/ file:///H|/SOS%2004-07/MIDI%20config%20problems%20in%20OS%20X%3b%20Garageband%20v1.1%20update.htm (4 of 5)9/22/2005 8:58:19 PM


AMT's serial connections. So far, this seems to have eliminated the chance of the interfaces changing order. Secondly, loading times in Nuendo Since each MOTU MIDI interface has its own have gone down from two minutes to USB connection to the computer, CoreMIDI 20 seconds, which is obviously a big sees each individual interface separately. difference! However, the big question is whether this simply because the system now has fewer ports, or if it's something to do with the MOTU interfaces taking longer to initialise when Nuendo communicates through Core MIDI. I would certainly be interested in hearing of other users' experiences with multiple MIDI interfaces where the driver model sees each interface individually, especially those using sequencers other than Logic and Nuendo. In the meantime, one workaround for worried users of Nuendo or Cubase is to use the application's MIDI Device Manager and always map your MIDI Tracks to these virtual devices within the sequencer. In this way, even if the state of your MIDI interfaces change so that Cubase or Nuendo no longer identifies them correctly, it's easy to remap the virtual devices to the correct physical devices in the MIDI Device Manager window. Published in SOS July 2004





In this article:

Where To Look Synthedit SIR Convolution Reverb Useful Music Utilities Impulse Responses Trackers Freebies From Commercial Developers Audacity Final Thoughts

PC Music Freeware Roundup PC Musician Published in SOS July 2004 Print article : Close window

Technique : PC Musician

Thanks to the Internet and the generosity of talented programmers all over the world, it's possible to assemble a PC music software suite for no money at all. We round up some of the best download sites and freebie programs. Martin Walker

There's an amazing selection of PC software available for free download, but with so much to choose from, and the fact that many musicians trying it don't have the chance to compare it with commercial packages, it's sometimes hard to tell what's good and what's not. Moreover, I've noticed a lot of snobbery about, of both varieties — some professional musicians look down their noses at freeware, dismissing it all as rubbish, while on the other hand there are beginners who use freeware downloads almost exclusively to get started, and who think the vast majority of commercial releases are not worth the money.

With sometimes stunning graphics and a huge range of possibilities, these are a few of my favourite stand-alone freeware VST plug-ins and instruments created using Jeff Mclintock's excellent SynthEdit.

As always, the truth falls somewhere in between these two extremes. Certainly, while there are some freeware 'my first plug-ins' to be found that are likely to be quickly discarded by most people, there are also some absolute gems in the freeware world. So this feature will be a roundup of free PC software that doesn't quite merit a full stand-alone SOS review and doesn't fall neatly into Plug-in Folder, but deserves more than a quick mention in PC Notes.

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Where To Look One of the biggest problems with freeware is knowing where to look for it on the Internet. With millions of web sites worldwide and the majority of freeware tucked away on its creator's personal pages, it's hardly surprising that many excellent releases go almost unnoticed. However, there are, fortunately, some Internet portals — web sites acting as themed gateways to many others — that can help us. One of the best for soft synths and plug-ins is K-v-R (www.kvr-vst.com), which posts regular information about new commercial, shareware, and freeware music software releases of all types, and maintains a database of instruments, effects, and their host applications, with an advanced search engine to find specific types — so you can, for instance, choose 'synth (wavetable)' or 'Exciter/Enhancer' to narrow your findings. It also hosts about three dozen forums supporting the users of products from a wide selection of the smaller developers. If you're specifically looking for freeware, the best site I've come across in my travels is one called Database Audio (www.databaseaudio.co.uk), which not only covers Mac and PC Windows offerings, but also Linux. There are plenty of plugins and soft synths to explore here, and you can also find some intriguing standalone applications. I personally like its list format, with a few sentences plus one tiny image of each release, which often leads you to finds that you might not otherwise have made. The main page shows a list of the latest additions, while buttons across the top lead you to the full lists of plug-ins (including soft synths), applications, and online instruments (in formats such as Flash, Java, and Shockwave). A user chart shows the top 20 most popular downloads of the moment. If your interest lies in multimedia players, MPEGX (www.mpegx.com) is a useful port of call, as it has lists of audio and video rippers, encoders, players, recorders, editors, CD and DVD burners, and utilities, as well as links to full versions of related applications, such as Ahead's Nero. A more general-purpose site is HitSquad's Shareware Music Machine (www. hitsquad.com/smm), which has a huge categorised list of software ranging from audio editors to wavetable emulators for a variety of platforms, not only including Mac and Windows, but also BeOS, Linux, OS/2, and Atari. However, I always find this site frustratingly slow to use, partly because its policy of including lots of time-limited and restricted demos of commercial software makes searching for freeware very time consuming, and partly because of the number of pop-up ads that slow your progress. Nevertheless, since there are a vast number of items to download, you'll probably find things here that you wouldn't elsewhere. Yet another one to try is The Sonic Spot (www.sonicspot.com). This site has a far smaller collection of plug-ins and soft synths, but a good range of older software



such as hardware synth patch editing and stand-alone utilities. It's also quick to use, with clearly displayed file sizes and software status (demo, shareware or freeware).

Synthedit There are now various commercial DIY soft synth creator packages available, including the very popular Reaktor from Native Instruments and Tassman from AAS. Users of both of these can download lots of additional instruments and effects from a variety of web sites, designed by both their developers and other users, but of course you do need to buy the package in the first place to take advantage of these, and to run it as a host before loading in your modules. A rather different approach is taken by Jeff Mclintock's SynthEdit (www. synthedit.com), which is a very reasonably priced ($20) shareware package for designing your own soft synths and VST effects, playing them live via MIDI, and recording, playing back and treating live audio. It's primarily aimed at advanced users and is available in two versions to suit Windows 98/ME or Windows XP, NT Database Audio is one of the best gateways and 2000. The 2.1MB download file to a whole new world of new Mac, PC contains a tutorial to help you get Windows and Linux audio freeware, started. There's also an optional SDK providing links to many unusual items that (Software Development Kit) for those I've not spotted elsewhere. who want to go the whole hog and create additional low-level SynthEdit modules to perform new functions. Over 50 of these have been contributed to the web site by talented users. There's a thriving SynthEdit Yahoo newsgroup, a dedicated SynthEdit community, and a K-v-R users' forum. So far, SynthEdit is similar in concept to Reaktor and Tassman. The big difference is that its designs run without SynthEdit being present, so its users can distribute them as freeware, donationware (where those who want to show their appreciation of the work that's gone into the end product can make an optional donation to the author via Paypal), or as full commercial products, like Boomedia's Studio Weapons, reviewed in SOS August 2003. Moreover, since the host application and its various toolbars isn't required, the user interface can be as conventional or radical as your graphic skills permit. Users have already made knob-creation tools, a 'skin' assembler and font editors to help others with the design process, and the results speak for themselves. Of particular merit are the stunning freeware designs by Spanish developer group Elogoxa (www.elogoxa.net), including a virtual recreation of Bob Fripp's file:///H|/SOS%2004-07/PC%20Music%20Freeware%20Roundup.htm (3 of 12)9/22/2005 8:58:28 PM


'Frippertronics' delay using two Revox B77 tape machines running a continuous tape loop. Elottronix XL looks wonderful (see the screen at the start of this article) and offers up to 80 seconds of delay, plus LFO pan, Biquad X filtering and a tapenoise generator. Other offerings include the highly unusual distortion and feedback synth effects offered by 'The Devil Inside', a mastering X-Cita inspired by BBE's Sonic Maximiser, and the vintage style Baxxpander to add warmth and saturation effects. However, for me the highlight is Sun Ra, a free-running or keyboard-controlled VST Instrument that creates complex ambient textures from samples originating from real solar sources. Wonderful! At the other end of the scale, but nevertheless extremely useful, is Wally Cescato's MIDI Data Monitor (www.freewebs.com/wallyaudio), which loads as a VSTi and displays incoming MIDI note, aftertouch, pitch-bend and controller data on four 'LEDs', as well as providing a scrolling data display in text form. This is ideal when you're trying to sort out why there's no sound coming from your VSTis in Cubase, for instance, or for examining incoming MIDI controller data to see why it's not altering the synth parameter you expected. In all, there must be well over a hundred freeware offerings from a wide variety of SynthEdit developers, and I must mention those of SOS reader and forum contributor Oli Larkin, who first brought the SynthEdit range to my attention. He has 14 on his web site (www.oli.adbe.org). My favourite is the complex yet subtle Dronebox, with six tuned resonators and sub-oscillator that can turn a guitar into a sitar or a drum track into chordal Eastern trance. Overall, apart from the slightly greater CPU overhead inevitable with all modular designs compared to hard-wired creations, SynthEdit designs can and do look and sound wonderful. Jeff Mclintock doesn't even demand a credit alongside products created with it, although he does appreciate one. Consider it done, Jeff!

SIR Convolution Reverb Most musicians will, by now, have heard about convolution reverb, the technique that captures the sound of a real acoustic space as an impulse response (IR) file and then uses it as a digital audio effect, either in hardware or plug-in form. This results in significantly more realistic reverbs than those created using an algorithmic model, albeit normally at the expense of rather more processing power. This same technology is also now routinely used to capture the particular non-linearities of preamps, microphones, compressors, speakers and their cabinets, and even the sound of software-designed virtual spaces. However, while the 'big boys' release extremely clever but expensive plug-ins that can modify the initial impulse files in various ways, the maths for straightforward impulse reverb playback is well known, and many users are beginning to find that the quality of the original impulse responses is arguably more important than the playback engine. In other words, a significant part of the



cost of commercial convolution reverbs is down to the huge effort involved in creating IRs from world-class acoustic spaces for the bundled library. With this in mind, PC users are lucky to have a freeware VST plug-in available for IR playback. After over a year in Beta form, during which time it gained an enthusiastic following, Christian Knufinke's SIR (Super Impulse Response) has recently jumped to version 1.005. You can download it from the author's web site at www.knufinke.de/sir. Using SIR is simplicity itself. You just click the Open File button on the righthand side to load in an impulse file in standard 16-, 24-, or 32-bit WAV format. Its waveform then appears in the graphic window and you can start auditioning it. Beneath the window is a set of sliders. These control pre-delay, apply a basic graphic envelope to the impulse response (to remove part of its Attack for a smoother effect), add Envelope decay characteristics to simulate a smaller space, or change the Length of the impulse using gating, either to minimise CPU overhead or for special effects.

SIR provides PC owners with a free convolution reverb player that they can use with lots of high-quality impulse responses downloaded from around the world.

Further versatility is provided by the Stretch control. This alters the original sample's length, and hence the effective room size, between 50 percent and 150 percent. There are also width controls for both the input signal and IR output, and a Reverse button for those backwards effects. There's even a free-form eightpoint graphic filter at the bottom to let you draw in your own EQ for the wet signal. Mixing is handled by a pair of Dry and Wet sliders with peak-reading meters alongside, whose caps flash if you run into clipping. There's also an Auto Gain option and a 12dB boost to cope with low-level impulse files. Very handy Reset buttons are provided for the IR and EQ sections, to return all their controls to the default positions. After downloading some impulse files to accompany SIR (see next section), I was most impressed by the results. Like all convolution reverbs, it's incredibly versatile — if you don't like what you hear you just try another impulse, and keep trying until you find the one that's most suitable for your track. I find the Stretch control effective, within limits, for fine-tuning the apparent size of the space, although metallic and lumpy artifacts can kick in at more extreme settings. Used carefully, the Attack control also proves handy for reducing early reflections, while the Envelope effectively shortens the Decay Time. The Length control must be used carefully to avoid artificial-sounding truncated tails, but it's perfect for gated reverb effects that make drums sound huge without swamping them in reverb. file:///H|/SOS%2004-07/PC%20Music%20Freeware%20Roundup.htm (5 of 12)9/22/2005 8:58:28 PM


For straightforward playback of IR reverbs, SIR sounds very good to me, and the CPU overhead on my P4 2.8GHz PC was also modest, rarely exceeding six percent with 16-bit/44.1kHz impulses, compared with about four and half percent for the Waves Rverb and up to nine percent for PSP's EasyVerb. If your PC is much slower than this, SIR has a Dynamic CPU Consumption mode that seems to reduce overhead once the reverb tail has dropped to silence and calculations are therefore no longer required, without audible degradation. If you still find the CPU overhead too taxing on your PC, you can always use the 'freeze track' function of your sequencer to capture audio tracks complete with reverb. The only limitation is that SIR imposes a fixed 8960-sample latency, which equates to 203ms at 44.1kHz. While this is huge, multitrack applications such as Cubase SX 2.0 compensate for it automatically, and latency won't normally worry anyone running a mastering application such as Wavelab. Only those whose applications are without automatic compensation should find this latency a problem, and even then it's possible to tackle it with a utility such as the AnalogX Sample Slide, as I described in SOS April 2004. However, SIR is not suitable for use as an effect during the recording process, and a few users have also apparently found instability problems while using it, although you can always remove it from your VST Plug-ins folder if this happens.

Useful Music Utilities AnalogX (www.analogx.com) provides a wide range of really handy utilities including DXMan for managing DirectX plug-ins, BitPolice for analysing what a DX plug-in is doing to your audio stream, MIDI Mouse Mod for mouse control of up to four simultaneous MIDI controllers, and DriveTime, which sits on your taskbar providing a read-out of remaining hard disk space in hours and minutes at a particular sample rate and bit depth. Anvil Studio (www.anvilstudio.com) from Willow Software is a sequencer offering comprehensive MIDI support, with staff, lyric, piano roll, drum and event editors, plus limited stereo audio support, although an optional $19 add-on extends this to eight audio tracks. It runs on all Windows versions right back to Win 95. Converter, from urr Sound Technologies Inc (www.urr.ca), is a DOS-only application designed as a system for sophisticated real-time MIDI performances, with an advanced MIDI input processor, audio to MIDI converter, gameport or joystick to MIDI converter, and mouse or touchpad to MIDI converter. You can use it with most elderly PCs, from an absolute minimum of a 486DX 33MHz, with the addition of an MPU401 interface or a soundcard such as a SoundBlaster or Gravis Ultrasound. MusicGraph, by Paul Nelson, (http://gigue.peabody.jhu.edu/~pnelson/welcome.html) is a visual tool for exploring the content of modern compositions. It takes a MIDI file and produces coloured graphs of interval content, total volume, notes per second, intensity and dissonance. Nero's CD-DVD Speed (www.cdspeed2000.com) is a handy 471K benchmark utility that tests the most important features of your CD or DVD drive. Various other related freeware utilities are also available on the same download page. Steem (http://steem.atari.st) is a freeware Atari STE Emulator that's claimed to run the vast majority of ST software without problems, including games and MIDI music applications. file:///H|/SOS%2004-07/PC%20Music%20Freeware%20Roundup.htm (6 of 12)9/22/2005 8:58:28 PM


Impulse Responses Impulse files themselves are also freely available for download at various web sites, the most notable being the Noise Vault (www.noisevault.com). This now has hundreds of offerings from lots of different contributors. Understandably, given that it's far easier to get access to old hardware than to set up mics and speakers in halls, studios and churches, these largely consist of impulses captured from reverbs and other devices from the likes of Alesis, EMT, Eventide, Kurzweil, Lexicon, Quantec, Roland, TC Electronic and Yamaha, plus a small range of real spaces, mics, preamps and speaker cabinets. The downloads are generally packs of several 16-bit/44.1kHz IRs totaling several Megabytes in size. They vary in quality considerably, with some being clean and clear while others are noisy. As might be expected, there are more Lexicons than anything else, but gems from other manufacturers can also be found in the collection. Noise Vault also has active forums, with lots of helpful information on capturing your own impulses, as well as how to convert other formats to the With thousands of original sample-based standard WAV used by SIR, among songs already available for free download, others. This data will be particularly easy-to-use editors such as MadTracker 2, useful to users of Sonic Foundry's shown here, or simpler playback-only applications like the ModPlug Player (lower (now Sony Media Software) Acoustic right), the tracker format is still popular with Mirror tool, who already have a good lots of musicians. library of impulse files in SFI format but often experience much higher CPU overheads when using Acoustic Mirror than they do using SIR as a plug-in in their chosen MIDI + Audio sequencer. Other packs of impulses are available free from various web sites, including Echo Chamber's German-only site (www.echochamber.ch). There's a large library from Prosoniq (www.prosoniq.com) in HQX format for Mac users (although the free Stuffit Expander utility, downloadable from www.stuffit.com, will happily expand these files for PC users); the complete set of Acoustic Mirror Impulses from Sony Media Software (http://mediasoftware. sonypictures.com); and four banks from Voxengo (www.voxengo.com), this time of virtual (imaginary) spaces created with the Impulse Modeler application that I first mentioned in PC Notes October 2002.



Trackers For the last 17 years, various Tracker applications have helped generate hundreds of original sample-based songs in a compact file format, and PC users can now access them free of charge, as well as creating their own. Way back in 1987, Karsten Obarski came up with his SoundTracker software for Commodore's Amiga platform, which provided hardware support for playing back up to four samples simultaneously at differing sample rates. By today's standards, SoundTracker was a fairly basic step sequencer with four channels. New samples could be triggered at any step on each of the channels, with data displayed in a four-column format and entered using the computer keyboard. The resulting four-channel Patterns were then chained together via a master list to form songs in MOD (module) format, consisting of the song data plus the various samples that had been used. Despite these limitations, complex drum tracks could still be built up by interleaving samples in the same channel, leaving the other three channels for bass, melody line, and other accompaniment. For much music this proved perfectly adequate, although many musicians resorted to sampling a selection of chords to fill the sound out. Many other special commands were also available to enter into a step position in addition to the note data, such as tempo changes, volume changes and pitch bends, to provide more expressive capabilities. This step-editing process might sound tedious, but a huge number of Amiga musicians rose to the challenge of making the most of this 4-voice polyphonic sampler package to add music to demos and computer games, in the process perfecting techniques for incorporating effects such as echo and gating. When PCs first came of age for music making in the mid '90s, ScreamTracker (www.hitsquad.com/smm/programs/Scream_Tracker) was one of the first music software packages to appear with sample support, followed by the still popular Fast Tracker 2 (www.gwinternet.com/music/ft2), Impulse Tracker (www. noisemusic.org/it), and MadTracker (www.madtracker.org). Fortunately, by this stage MIDI support had been added to most trackers, so you could play in your tunes in 'real time' from a music keyboard rather than entering them as step data from the computer keyboard, while the four-track limit was removed in PC trackers; a greater number of tracks could be mixed down to stereo using a software re-sampling engine. ScreamTracker supports up to 16 channels and sample sizes up to 64K, while FastTracker offers 32 channels and Impulse Tracker offers 64, as well as supporting much larger sample files, of up to 2MB. Other new features also appeared, such as software reverb and surround sound effects, although using them will make your songs sound different when played back on other trackers.



Despite the number of loop-based sequencers available today — Acid and Live, for example — trackers are still immensely popular, because they are compact and easy to learn and use. They're also often free to download, and there are many existing MOD files with relatively small file sizes available for free download (it's possible to create melodies from one single-note sample and complex songs using just a few hundred kilobytes of samples). MOD files can be downloaded from quite a few sites. One of the largest must be the Mod Archive (www.modarchive.com), which houses nearly 12GB of files, as well as hosting various tracker-related forums. Music can be found in a huge variety of styles, although electronica abounds. I found a nice collection of modern dance, trance, rave house, and synth pop music by 'Otis' at www.xs4all. nl/~perseus during my travels, but you are never likely to run out of MOD files to download — a Google search for 'mod file downloads' turned up over a million links. If you're specifically interested in soundtracks ripped from computer games, Game Music Base (www.mirsoft.info/gmb) has a huge collection in many different formats, including MOD, although some of the earlier Amiga ones don't sound quite the same on the PC, to my ears at least. If you just want to play back these archive songs, you don't even need a full tracker — MOD Players such as the 334K MODPlug (www.modplug.com) are perfectly adequate for this task (you can download yet more MOD files at this web site), while the popular WinAmp player (www.winamp.com) will also play back MOD files, as well as other formats. MOD Players are also generally more stable than trackers, in my experience.

Freebies From Commercial Developers If you're concerned about installing freeware plug-ins and soft synths from unknown developers, in case they have any effect on the stability of your PC, remember that those in VST format can simply be deleted from your VST plug-ins folder if they cause any trouble. However, one way to ensure that you don't have any such problems is to download the 'freebies' kindly provided by some commercial developers. Here are a few that I've found useful: IK Multimedia (www.sampletank.com): SampleTank Free 1.1 plus 20 free instrument files. Native Instruments (www.native-instruments.net): SoundForum soft synth, Traktor DJ Player. OtsZone (http://otslabs.com): Ots CD Scratch 1200, Ots Turntables Free. PSP (www.pspaudioware.com): PianoVerb, VintageMeter. Sony Media Software (http://mediasoftware.sonypictures.com): ACID XPress Voxengo (www.voxengo.com): OldSkoolVerb, EssEQ, Tempo Delay, Tube Amp VST. Yamaha (www.yamahasynth.com/download/twe.html): TWE Wave Editor v2.31.



Audacity Although there are some wonderful commercial stereo audio editors and mastering packages available for PC users, not everyone needs their vast arsenal of functions; nor can they afford their price tags. Enter Audacity (http://audacity. sourceforge.net), a free audio editor for Windows, Mac OS9/X and Linux users. Originally started in 1999 by Dominic Mazzoni while he was still at college, its code is now open-source freeware and several dozen developers around the world collaborate to add new features. Audacity is only a tiny 2.6MB download, but despite this it has lots of features and offers a choice of 20 languages for its menu text. The program can import and export WAV, AIFF, AU, Ogg Vorbis and MP3 files, and supports 16-bit, 24bit, and 32-bit float formats at up to 96kHz. It provides a familiar graphic editing environment, with a main waveform display and a set of toolbars across the top, although these can be floated if you prefer. The Control Toolbar contains attractive transport controls, plus a set of tools that includes the usual Selection, Draw and Zoom tools, plus a well-coded Envelope tool for changing volume over time, TimeShift for moving tracks left or right, and a clever Multi tool that lets you access any of these, depending on the location of the mouse and the keys you hold down. Audacity offers support for various audio

The Edit toolbar has 11 buttons, all formats, a fully-featured graphic editing with familiar functions. The unlimited environment and lots of integral effects — in Undo/Redo functions, in particular, are short, it's a totally free audio editor. welcome. The Mixer toolbar lets you set playback level, along with input source and recording level if your soundcard uses the standard Windows mixer. So far, so good. But when you really start examining the various menus, you realise just how much Audacity has to offer. Loop enthusiasts will appreciate the way the Zero Crossings function modifies the start and end points of a selection to avoid clicks, as well as the variable-threshold Beatfinder, while the Effect menu is a revelation, with several dozen treatments on offer, including compressor, EQ and FFT filtering with click-and drag graphic interface, pitch, speed, and tempo changing, plus many others with a simpler slider-only interface. There's even a Noise Reduction treatment that uses a noise profile. While this produced fairly obvious chirpy artifacts with the samples I tried, it proved to be a



wonderful effect at extreme settings! As with Sound Forge, you can preview each effect before applying it off-line. In a similar way to Wavelab's Montage functions, Audacity's Project menu lets you import, record and play back multiple audio tracks for subsequent mixdown into a single stereo stream. You can even import raw data, add or import text labels and load in and display MIDI files, although MIDI playback, recording and editing are still destined for a future release of the software. The Generate menu can create 30-second files containing test tones, white noise, silence and a handy audio click-track at any bpm rate. I was extremely impressed with Audacity, but it has a few limitations worth mentioning. Firstly, it doesn't import audio tracks from audio CDs, although there are various other freeware utilities that can do this if it's something you need to do, and to export MP3 files you'll need a suitable encoder, such as the freeware LAME, which can be found via links from www.mp3-converter. com/encoders/ lame_encoder2.htm. For most musicians it will also be frustrating not to be able to use any of their DirectX or VST effect plug-ins, although there is a partial workaround — the optional VST Enabler plug-in for use with Audacity isn't built in, for licensing reasons, but it can be downloaded separately and dragged into Audacity's own Plug-Ins folder, along with any VST plug-ins you want to use. It certainly works, but the current version of the Enabler ignores the graphic interface from each plug-in, instead replacing it with a generic 'slider and text box' window. Simple plug-ins are still usable under these circumstances, but for more complex ones it's mighty frustrating. Let's hope the promised upgrade with full graphic plug-in support will appear soon.

Final Thoughts Although I was fully expecting to report plenty of interesting free downloads when I first started working on this feature, I was still amazed at some of the things I found in my travels. I hope this roundup has whetted your appetite, as well as proving that free software can sometimes be surprisingly capable. I'm certain much of the software here could actually be sold on a commercial basis if its developers so wished, and given the huge amount of time it takes to develop (I know — I used to be a software author myself in a former existence) it would be great if users would support them, where appropriate, with a donation. Since I deliberately restricted myself to freeware on this occasion, there's still a whole world of excellent low-cost PC shareware out there that I haven't yet mentioned, and I hope to cover some of the best in a future feature. Published in SOS July 2004






Pro Tools v6.4 update

In this article:

Quick Tips It Goes To 12 Compensation Culture End Of The Line For Mix Hardware Recording Improvements

Pro Tools v6.4 update Pro Tools Notes Published in SOS July 2004 Print article : Close window

Technique : Pro Tools Notes

Current Versions 6.4: HD, Accel and LE systems for Windows XP and Mac OS 10.3. 6.1: Mix systems on Windows XP and OS X.

Pro Tools version 6.4 comes with some surprising and long-awaited new features. Simon Price

It's been an incredibly busy couple of months in the Digidesign universe, and we've finally found out what they've been ferreting away at over in San Francisco. All in a flurry Digi have announced the amazing-looking Icon mixer, the small-format Command 8 control surface, a proper surround reverb (Revibe), and plans for live gear. It's perhaps not surprising, then, that Pro Tools version 6.4 slipped out fairly quietly in April, at first looking like a simple support release for the new hardware controllers. However, it's a pretty significant upgrade that introduces some new features, including one right from the top of the wanted list: automatic delay compensation. Many of the other additions show that Digidesign have been listening to some of their customers in the TV and film production worlds, but there's plenty of stuff in there to make it worth upgrading for everyone.

Quick Tips A shortcut to bypassing plug-ins (without having to open their window) is to command-click on their slot in the Inserts section of the mixer. Command-clicking on break points in automation graphs deletes them. This is often one of the quickest ways of cleaning up and smoothing automation. Sometime it's useful to know what the loudest point of a region of track is, especially if you want to find out whether the level reaches 0dB (and is therefore probably clipping). The Gain Audiosuite plug-in has a Find Peak command that will scan a selection and report the top level. It's often best to leave out the Master Bypass parameter when enabling automation for a plug-in. This way you can easily bypass the plug-in manually during playback without fighting any automation. Also, plug-ins often click, pop or otherwise make unhappy noises when you automate their bypass. It's better to file:///H|/SOS%2004-07/Pro%20Tools%20v6.4%20update.htm (1 of 4)9/22/2005 8:58:32 PM


just automate the controls to a 'flat' or inoperative state where you don't want the plug-in to do anything.

It Goes To 12 There are a couple of things that are obvious when you fire up 6.4, especially if you open up an existing Session. You're immediately asked if you want to switch the Session to the new +12dB fader mode, or leave it alone. Yep, Pro Tools' faders now provide +12dB gain above the unity point, where they used to have only 6dB. In rock and pop music production you tend to be working with loud signals most of the time, but in other areas like TV and film, there's much more variation in levels, and the +6 faders have been a long-term bugbear. This change should reduce the need for adding gain at the plug-in stage, which always risks clipping the channel before you've even got to the fader. Many of the other tweaks are also angled towards post-production, but the biggest change is near the top of the list for many music-based Pro Tools users...

Compensation Culture Automatic delay compensation aims to remove problems caused by the small delays introduced into signal paths by routing through plug-ins, sends, and outboard gear. As described in Pro Tools Notes February 2003, these small delays can cause phase problems when mixing signals, or in some cases sync problems when working to video. We'll come back to ADC in detail in another Pro Tools Notes soon, but here's a quick Pro Tools 6.4 has a new tidy plug-in list overview of what it does: the system structure, and faders that go up to +12dB. That's one louder than 11. But you'll need a looks at every signal path in the mixer, HD TDM system for the new automatic delay and identifies the largest latency compensation feature. between any track and the final mix outputs. It then applies the appropriate amount of delay (in real time) to all other paths so that everything comes out the same. This is a sophisticated and resource-hungry scheme, but is capable of much more than the some other ADC methods that simply nudge recorded audio about. For example, the delay is applied to aux input tracks, which means that signals coming through the mixer live are taken into account, as are hardware inserts and send/return paths that go outside the virtual mixer. The power comes at a price: ADC is only for TDM systems, and HD/Accel ones at that (see box about Mix hardware). Another persistent feature request that's been heeded is for a menu structure in file:///H|/SOS%2004-07/Pro%20Tools%20v6.4%20update.htm (2 of 4)9/22/2005 8:58:32 PM


the plug-ins list. Until now of course the plug-in list has been one long jumble, with an often bemusing ordering. The most common request has been for users to be able to create sub-folders in the plug-in folder, which would then translate into a hierarchical list in Pro Tools' insert selectors. Well, instead Digi have opted for a menu structure that is predetermined by the plug-in developers. The plug-ins list is now subdivided by effect type, with the type Both LE and TDM versions now have being flagged by the manufacturer. No transport controls in the Edit Window's toolbar, and a new method for navigating doubt this will infuriate a few people around large projects. who want to be able to manage this manually, but let's be honest, life's way too short to be poking around in your computer's library directories unless you really have to! Some customisation is possible, however, in the form of a 'favourites' system. Plug-ins designated as favourites are listed at the top of their categories. Like the plug-in list improvements, a new track-numbering scheme comes to both LE and TDM, and provides an extra way to navigate across large mixers. As you can see from the screen shot, all the tracks have been designated a number, so you can scroll directly to particular places in a similar way to using memory locations. Similarly small but useful improvements have been made to signal clip indicators in version 6.4. I mentioned before about the annoyance of plug-ins clipping a channel pre-fader, a problem can that be irritating and difficult to track down. TDM software now indicates that a plug-in has clipped in the inserts section of the mixer, so you can identify the offender straight away. Any tracks that have clipped will also now light up in the track show/hide list.

End Of The Line For Mix Hardware At the same time as announcing the new version of the Pro Tools software, Digidesign confirmed what has been becoming more and more obvious to many users: there will be no further development for Mix hardware, except for one final software release. The company cite the increasingly complex testing and development required for multiple platforms as the reason for this decision. It's certainly a tricky one, because the reason Mix was such a milestone was that for many people it represented the first DAW that could pretty much do everything they needed. Now those people are faced with a choice between upgrading to HD, or keeping their computer frozen at its current configuration to continue with Mix. There is a third option of course, which is to get a really top-of-the-pile G5 or XP workstation and go the LE route. To sugar the pill (a little), Mix owners can upgrade from 6.x to the 6.4.1 release free, and there are extra discounts on HD upgrades until the end of June. Version 6.4.1, which will be released some time in the summer, brings most of the new 6.4 TDM features to Mix, with the notable exception of automatic delay compensation, file:///H|/SOS%2004-07/Pro%20Tools%20v6.4%20update.htm (3 of 4)9/22/2005 8:58:32 PM


which Digi say requires too much DSP memory for Mix cards to handle.

Recording Improvements The other major area of tweakage apart from ADC is the addition of some advanced recording and monitoring features, again just in the TDM version of Pro Tools. The previously global input monitoring status ('Auto Input Monitoring' vs 'Input Only Monitoring') can now be set differently from track to track. The status is now displayed and controlled in the channel strips, making it easier to swap on the fly between what's recorded on a track and what's coming in live. Punch recording has been improved along similar lines. It's now possible to punch different tracks in and out of record independently during the same record pass. Previously, punching into record acted globally on all record-enabled tracks. This new functionality is called TrackPunch, because it's important that new features get their own two-words-joined-together name. Being able to change input monitor status is called, of course, TrackInput (look it up if you don't believe me). Apparently, in early alphas of Pro Tools, the on-screen fader concept was known as VolumeChange. TDM upgrades to 6.4 are £95 unless you got your system this year, making it the first paid upgrade since 6.0, while the LE version is free (or £50 from v5) as it doesn't gain so much. The most encouraging thing about 6.4 is that it is largely concerned with changes that really help some day-to-day work situations, rather than flashy features that look good at trade shows. Well, they hardly need to try too hard with that enormous Icon thing gleaming away in the booth! Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Reason Notes starts here!

In this article:

First Principles Teacher's Pet Bass Place Screaming For Bass Stage, Right?

Reason Notes starts here! Monthly column for Propellerhead Reason users Published in SOS July 2004 Print article : Close window

Technique : Reason Notes

Sound On Sound readers who use Reason have spoken: they want our help in getting the best from this popular package — and we're happy to oblige with a new monthly column. Derek Johnson

The results of Sound On Sound's recent readership survey have not yet been fully analysed, but preliminary results revealed that a decent percentage of you are confirmed users of Propellerhead's Reason virtual electronic studio software. What's The SubTractor bass patch, after not many knob tweaks: it's subby, but it cuts through a more, a recurring request from these mix. users is for more coverage of Reason in SOS. No sooner said than done: welcome to the first instalment of our new monthly Reason Notes column, which takes its place alongside our existing columns for the other leading sequencer packages.

First Principles Where does one start with a program as complex and fulfilling as Reason? The software is actually nowhere near as obtuse or difficult to learn as some other packages, yet newcomers — especially those with little experience of the hardware being emulated in Reason's rack — can find it overwhelming. It simply offers so much. One good guideline is to start small, perhaps by limiting your rack to just a handful of devices. Another is to direct your attention to any SOS article on analogue synthesis: Propellerhead's design choice means that such articles can be digested and the techniques applied within Reason. Practically

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any book on analogue synthesis and electronic music studio techniques written in the last four or five decades could help you develop a sound-design and composition strategy. This column will also try to help: despite the fact that it's accessed by a mouse, Reason is a very-hands on package that invites tweaking, and that's the clue to really getting the hang of the software. I'll be offering some ideas about where to start with your tweaking, and exploring sound design with what is, after all, an integrated electronic music studio in your Mac or PC. We'll also look at lateral approaches to things the software can't, or appears to be unable to, do.

Teacher's Pet Propellerhead really pay attention: Reason has become quite an educational tool wherever music technology is taught, and in response to this they've released the Teaching Music With Reason package. Aimed at secondary school and higher music technology classes, the course helps to develop musical and composition ability as much as it does to teach electronic music skills. The integrated approach includes the teaching of mixing techniques, improvisation and using loops, and finishes with the compilation of a CD of coursework. Along with the software, the package includes lesson-preparation material, teaching plans, student worksheets, 'How Teaching Music With Reason: the to' guides and assessment sheets. There's package. apparently enough material to take students through a couple of terms of weekly lessons. Teachers who are unfamiliar with Reason are supplied with background information so that they can stay one step ahead of their students!

Bass Place The root of most music is the bass, and when I first started working with Reason it was very important for me to be able to produce bass patches with the SubTractor synth that had a real subby, woofer-shaking low end — I like to feel bass in my tracks as much as hear it. So here's an approach you might like to try if you have trouble achieving a full bass sound. Create a SubTractor, highlight it and create a Matrix sequencer. Now input a simple pattern in the Matrix and press play. Move your mouse back to SubTractor and start playing with the parameters.



A good starting point for a bass sound is a square waveform selected in SubTractor's Oscillator 1 section. What you hear if you play the Matrix pattern is a hollow sound, rather like an unfiltered TB303. The next step is to select the 'x' setting for Oscillator 1's Mode button, which engages Phase Offset Modulation. This creates a copy of Oscillator 1, and (in this case) multiplies the copy waveform with the original. Adjust the offset between the two waveforms by moving the Phase knob slightly to the left (say, a value of 50). The sound is now a lot more solid and rich, with almost a sub-bass feel, but it doesn't lack definition and attack. If the sound is too short and blippy for you, move on to the Amp Envelope and adjust the Release parameter. To add even more definition to the sound — maybe you're after a bass sound that plays a more up front part — engage Oscillator 2. Set it up exactly as you did Oscillator 1: square wave, 'x' in POM, Phase knob set to 50. The sound is now using four oscillators, and taking up more of your computer's CPU resources. It should have a random phasiness, which is interesting, but now try it with Osc 2's Octave setting at 3. The sound is still solid and bassy, but it has a cutting top note. An even more cutting sound can now be produced by moving Filter 1's Frequency slider down (to about 45) and the Resonance slider up to about 50. Listen closely: there's still a low frequency rumble, but with an aggressive filter-led blip to the sound. See the 'Screaming For Bass' box for an extra bass-boosting tip.

Screaming For Bass Since V2.5, I often take a cheating approach to adding what night almost be called bass enhancement, by way of the The settings for a quick Scream 4 bass Scream 4 distortion device, which enhancer patch: we'll look at using this can be used quite subtly for this versatile device for mix processing in future. task. First, highlight a SubTractor and create a Scream 4 (it'll be patched in line with the Sub's audio output). Turn the Damage control down a little so it's not distorting so much, select the 'Tape' Damage algorithm, and tweak P1 (Tape Speed in this case) fully left and P2 (compression) as far to the right as feels good to you. If you want to go even further, engage the 'Cut' section, and move the Lo slider up (the section is labelled 'Cut', but doing this actually boosts the bass frequency of this simple but effective EQ). Even more resonance can be added by playing with Scream 4's Body section (especially Body types D and E). Indeed, tweaking these parameters takes us further into Reason's potential for integrated sound design: the effect is as much part of the sound as what was produced by the original SubTractor.



Stage, Right? Reason, as originally conceived, was meant to have more in the way of live performance facets than it did in the final release. There are a few oddities in its control set that hark back to this, and Propellerhead occasionally hint about adding the features at a later date. But perhaps Italian software house Petertools have beaten the Props to it. Their Liveset software aims to be a configurable link between your MIDI hardware controller(s) (keyboards and control surfaces) and Rewire-compatible applications. Currently, the application in question is more specifically Reason (as Liveset was developed for it), though other apps will be directly supported in future. Graphically, Liveset has clearly been inspired by Reason: it's presented as modules placed in a virtual rack, each device can be named via a scribble strip, and interconnections between modules are made with virtual MIDI leads. Functionally, Liveset's modules are rather like MIDI plug-ins — similar in feel to those provided with Steinberg's Cubase SX, say — that are placed in line between your MIDI controllers and the Reason rack. Reason has no such data-manipulation tools, so in one way this is a back-door approach for a third party to add new features to the program. Liveset was created because one of the developers, who uses Reason as part of his live setup, wanted the tools it provides.

Liveset aims to provide Reason users with live performance tools.

The Petertools software provides two MIDI inputs, so at least two performers (or one performer with two controllers) can play Reason as if it were a rack full of real synths, with none of the restrictions that would be entailed by playing it directly. Particularly conspicuous are the note and controller multiplexers, which each provide eight separate MIDI outs from one MIDI In (or two sets of four MIDI outs from each of two MIDI Ins), so that many Reason devices can be easily played at once. (Using Reason's Hardware Interface only lets you play four devices simultaneously.) Easy-to-use key splitting on one MIDI channel — something else Reason can't do — allows complex splits of multiple Reason devices to be created, and the remaining devices provide some useful note and data manipulation including note transposition and controller crossfading. Liveset, in its beta format, is very much a real-time live performance tool — which is highlighted by the fact that it wouldn't be possible to record a Liveset-driven performance into Reason's MIDI sequencer. As powerful as Reason is, it can still only record one thing at a time, and in any case there is currently no way to route MIDI data from Liveset (or any other Rewire application) directly to a Reason



sequencer track. The software talks to Reason devices directly, in the same way that Cubase SX does when it's Rewired to Reason. Liveset also offers an alternative transport bar, complete with song position, playback loop points and tempo controls. Usefully, for on-stage use in lightcompromised situations, there's a big song position display! Currently, the software is PC only, requiring a 633MHz P3 or better processor, Windows 2000 or XP, a MIDI interface and an audio card with ASIO drivers. Liveset is going through beta testing as I write, but progress is quite advanced and the software should be available later this year, at a price of around 149 Euros (£99). I'll have another look at Liveset in a future column. In the meantime, check out www. petertools.com. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



The Key Editor; v2.2 Cubase update

In this article:

Nudge, Nudge, Wink, Wink Cubase SX 2.2 Update Something's Cooking

The Key Editor; v2.2 Cubase update Cubase Notes Published in SOS July 2004 Print article : Close window

Technique : Cubase Notes

The Key Editor is a seemingly straightforward MIDI editor, yet under its surface lie a number of features that can really speed up your editing tasks. We explain, as well as reporting on the new version 2.2 Cubase update. Mark Wherry

It has to be said that, compared to Cubase VST, the Key Editor found in Cubase SX/SL version 1 was a little under-nourished for the liking of serious Cubase MIDI fiends. So it's perhaps no surprise that it received a great deal of attention in version 2 from Steinberg's developers, and although some users still feel it lacks a few of the finer points from the previous generation of Cubase, there's no doubt that the Key Editor remains one of the program's strongest assets.

The Key Editor has many useful key commands for changing and manipulating the current selection of MIDI note events, without your hands ever having to leave the keyboard.

Learning key commands is always a great way to speed up workflow in any app, and it's possible to achieve some quite detailed editing in the Cubase Key Editor using nothing more than cursor keys and a handful of modifiers. Using the left and right cursor keys, for example, enables you to step forwards and backwards through the note events in the editor. When you have more than one note event on the same beat, the order in which these are selected depends on the order in which they were entered into Cubase. This order is the same as displayed in the List Editor and, other than opening this window at the same time, there's no real way to know in what order events will be selected, unless you're dealing with a monophonic line.

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The temptation when navigating through the Key Editor with the cursor keys is to use the up and down keys to select the different notes in a chord. However, as you'll soon realise, these keys are reserved for transposing the selected note (or notes) higher or lower, in semitone steps. To move the current selection up or down by an octave, press Shift at the same time as either of the cursor keys. This begs the question: what happens if you hold down shift while pressing the left and right cursor keys? Well, this works for selecting notes, in much the same way as described earlier, except that the previously selected note (or notes) stay selected as you extend the selection of notes further forwards or backwards in the event list. For example, with a note selected, holding down Shift and pressing the right cursor key three times will select the next three notes in addition to the one that was already selected. On the subject of selecting notes, here's an interesting (and brief, don't worry!) digression. When you want to select notes of the same pitch, there's a neater way than by using the commands in the Edit / Select menu. The problem with this command is that it always selects onwards from the currently selected note, meaning that if you have notes of the same pitch earlier in the song, these won't become selected. While this can be useful in some cases (and can also be achieved by Shift-double-clicking a note), you can select every note of a given pitch in the Key Editor at any point by Control/Apple clicking the pitch on the keyboard that runs up the left of the window.

Nudge, Nudge, Wink, Wink Once you've selected some notes in the Key Editor, you can also carry out quite a number of edits without your fingers ever having to leave the keyboard. We've already looked at transposing — that is to say, adjusting pitch — but you can also adjust the position of notes, in addition to the start or end points. Holding down Control/Apple allows you to nudge the selection forwards or backwards in time, based on the current quantise setting. With a quantise setting of 1/8, for example, notes will be nudged forwards or backwards in eighth-note steps (or quavers, for European musicians!). And remember, assigning quantise values to key commands via the 'MIDI Quantise' group isn't a bad idea either: I usually use Alt+1 to Alt+7 for whole notes through 64th notes respectively. This operation works the same irrespective of whether or not Snap mode is enabled. If you want to adjust the length of a note (or the currently selected notes), press Alt and Shift together while pressing either the right or left cursor keys to increase or decrease the note length (or lengths) — this time in steps according to the current Length Quantise setting. I usually leave Length Quantise set to Quantise Link, to make things easier, especially where key commands are concerned, although you can assign the Length Quantise values to their own key commands via the 'Set Insert Length' group. To adjust only the start point of a note, leaving the end point where it is, just press Alt (without the Shift key) while using the left and right cursor keys instead. This command uses just the regular quantise setting. file:///H|/SOS%2004-07/The%20Key%20Editor%3b%20v2.2%20Cubase%20update.htm (2 of 5)9/22/2005 8:58:44 PM


Two things are worth noting about the key commands we've discussed for nudging and adjusting the length and start points of notes. Firstly, these commands are also available via the Nudge toolbar commands, which are usually hidden by default. Simply right-click on the toolbar, and make sure Nudge is checked, to reveal them. Secondly, key commands also work for manipulating events in the Project window. The left and right cursor keys select the next or previous event on a track, while the up and down keys select the next or previous track in the track list. Holding down Shift for either direction selects multiple objects or tracks, and although the selection of objects on a track works the same as the selection of notes within the Key Editor, handling of multiple track selection is a little different. Press Shift and the up cursor key to select the tracks above the selected track, one by one, and press Shift and down to select the tracks below it. The difference here is that selecting additional tracks works independently in the two directions, basically meaning that there's no way to decrease the number of selected tracks without starting again (by pressing the up and down cursor keys without Shift).

Cubase SX 2.2 Update Steinberg recently released Cubase SX v2.2, an update that adds many long-awaited features to the application, such as support for Mackie Control XT. Mackie Control XT units are added to the Device Setup window just like regular Mackie Control devices, but when Cubase detects that multiple Mackie Control devices have been set up it spreads the channel selection across the available devices, assuming that the XT units are Cubase SX 2.2 includes three new plug-ins: placed, in order, to the left of the Monologue, Embracer and Tonic. main Mackie Control unit. This behaviour also extends to modes where one channel's (or a plug-in's) controls are spread across the available encoders. Staying with control surfaces, the Generic Remote has also been enhanced with the ability to import remote description files defined in v1.x versions of Cubase. While users with control surfaces will no doubt appreciate the improvements to that aspect of Cubase, one of Steinberg's highlights for the 2.2 release is the addition of three new plug-ins developed in collaboration with Wizoo. Monologue is a two-oscillator virtual analogue synth based on physical modelling techniques. Being a monophonic instrument, it's ideal for bass, lead and sequenced sounds, as demonstrated by the selection of presets. Embracer is a pad synthesizer that can work either in stereo or surround, thanks to an additional second stereo output added to the Mixer to deal with the rear channels. The width and spatial positioning of the two pad wavetable oscillators (offering a choice of 12 waves) can be controlled via an intuitive interface. Finally, Tonic is a filter that models the



characteristics of analogue filters with variable envelope triggering modes, and has a step sequencer for drawing your own LFO shape. A new option that may be the most significant part of 2.2 for Windows users is the 'Use System Timestamp' flag at the bottom of the DirectMusic panel in the Device Setup window. Enabling this option forces the time stamp given to each MIDI event as it passes through Window's DirectMusic sub-system (the part of the DirectX multimedia Windows API that handles MIDI) to be used rather than the time stamp generated by Cubase when it records the event. This solves what has become an increasingly common problem whereby (with certain MIDI device drivers) MIDI events are shifted backwards in time during real-time MIDI recording. Cubase SX 2.2 is available for download from www.steinberg.net, or, if you prefer navigating FTP sites manually, ftp://ftp.pinnaclesys.com/Steinberg/download.

Something's Cooking One of the great things about key commands in Cubase is the ability to combine a sequence of them into a Macro, which we've looked at in previous Cubase Notes columns. Using this technique with the commands we've just been looking at, you can create some really useful tools to achieve Logical Editor-style operations without using the Logical Editor. This is useful since you still can't assign key commands to Logical Editor presets in Cubase, although this is hopefully not too far away. As an example Macro, how about creating something that copies the currently selected notes and moves the copy an octave above or below the original? In the Key Commands window, click the Show Macros button, followed by 'New Macro', to start creating a new Macro. By selecting the appropriate key command in the upper part of the window and clicking 'Add Command', add the following commands to the Macro: 'Edit > Copy', 'Transport > Locate Selection', > 'Edit > Paste', and 'Navigate > Add Up'. The Transport command is necessary because Cubase always pastes the contents of the clipboard at the current position of the Project cursor. It's necessary to set the Project cursor to the start of the selection in this Macro, so that the notes are pasted back into the Editor at the correct time. For adjusting the length and position of notes, a nice idea (especially if you use a separate keypad device for triggering key commands) would be to build Macros to automatically select a different quantise setting before the required Nudge command. In this way you could have different keys to adjust note length by eighth notes, quarter notes, and so on. Published in SOS July 2004






The Secret Of The Big Red Button

In this article:


The Secret Of Synth Secrets A Different Way Of Thinking Synth Secrets Published in SOS July 2004 Sources Modifiers Print article : Close window Controllers Technique : Synthesis Source, Modifier Or Controller? Controlling Controllers Adding A Bit Of Interest Strengths & Weaknesses After over five years, Synth Secrets reaches its Why Bother? conclusion (and conclusions!). Will we ever look

synthesis in quite the same way again?

at

Gordon Reid

Every few weeks, my colleagues at Sound On Sound and I receive a telephone call or email demanding to know which is the best synthesizer. When these requests come directly to me, I have often replied, asking the enquirer what he (it's invariably a he) wants to achieve, and what criteria he deems to contribute to the term 'best'. And, every time I do, the response is, "Look, never mind all that... just tell me which one is best. I know you know — why won't you tell me?" Sometimes, I have tried to help the caller by asking him to think about what is most important to him — huge polyphony, sequencing, complex arpeggiation, suitability for live use, ethereal floaty sounds, heavy bass, rhythm sections, and so on. And the reply is invariably, "Look, never mind all that... I'm not interested in all that technology. Just tell me which one is best. I know you know — why won't you tell me?" You might think that this attitude is rare, but it isn't. If anything, it's becoming more common, perhaps because manufacturers are determined to convince you that you will become the latest, biggest, and richest musical phenomenon the world

Figure 1: The Big Red Button — surely it's in there somewhere?

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has ever seen if you simply buy the latest version of their Argon Megastation with the added wotsit and optional thingies. There's even a second stage to this delusion. It's the feeling that, hidden somewhere on the Argon Megastation, there's a Big Red Button marked 'Number 1 hits, free drugs, and more sex than you can handle'. Few musicians admit to searching for the Big Red Button, but many believe that it exists, and some are really sore that I won't tell them where it is.

Figure 2: A simple synthesizer architecture.

The Secret Of Synth Secrets Somewhat over five years ago, the very nice people at Sound On Sound and I were discussing the belief in the elusive Big Red Button, and we agreed that it would be interesting to publish a series that explained some of the less frequently discussed aspects of sound and synthesis. The idea was to avoid the introductory tenor favoured by other such series (your VCO-bone's connected to your VCF-bone, your VCF-bone's connected to your VCAbone... and so on) and to tell you things that you didn't even know you wanted to know. Even at the start, it was clear that no single approach would be suitable for all readers. One respondent described the first two parts of the series as 'condescending', even comparing them to a nursery rhyme, while others simultaneously claimed that the same parts were far too intense and mathematical. Nonetheless, over the ensuing five years, we have covered more ground than any previous SOS series on sound generation and synthesis.

Figure 3: Thinking about the synthesizer in a different way.

Figure 4: The Prophet 10 oscillator section offers a range of common waveforms.

Figure 5: The external signal processor (ESP) from the Korg MS20.



But now I find myself just a handful of paragraphs from the end of the final part, and I would like to leave you with one overriding idea that, in my opinion, sums up everything we've discussed. I hope that this will help you to become a better sound designer and player, although this seems a tall order for just one article.

Figure 6: Thinking in terms of sources and modifiers.

To attempt this, I'm going to offer you an alternative to the manner in which most people approach their synths. I call this 'modular synthesis', but you won't need a £10,000 wall of vintage modular synthesizer modules to take advantage of it.

A Different Way Of Thinking Many books on synthesis begin by explaining that sounds can be characterised at each point in time by three qualities: pitch, timbre and loudness. This idea is then extended to state that you can partition most synthesizers into oscillators that determine the pitch, filters that determine the timbre, and amplifiers that determine the loudness. Just throw a couple of contour generators and low-frequency oscillators into this brew and you have (so some would say) everything that you need to synthesize every sound imaginable (see Figure 2 below). However, I find this approach to be rather limiting, so, instead of viewing synthesizers in this way, I'm going to propose that you consider them in terms of three major classes that I call Sources, Modifiers, and Controllers (see Figure 3, above).

Figure 7: Viewing a simple synth as a set of sources, modifiers and controllers.

Figure 8: Two oscillators operating as sources.

Figure 9: Two oscillators; one a source, the other a controller.

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The most common sources on synthesizers are audio-frequency oscillators (such as those shown in Figure 4, above) although there are others, including noise generators and the external signal inputs offered by, for example, the Korg MS20's ESP section (see Figure 5, above).

Figure 10: Using the Juno 60 filter as a source in an organ patch.

The range and quality of the sources in your synth will place tight constraints upon the sounds that you can create. Clearly, you can't easily synthesize a hollow sound without an initial waveform that has that characteristic, nor can you create brassy sounds without waves (or, for that matter, samples) that offer the appropriate harmonic structure. So it's a good idea to learn to recognise the sounds of various waveforms, especially the ones commonly found on synths, such as sawtooth, square, pulse, and triangle waves. If you can hear a sound — whether produced acoustically or on another synth — and judge the waveform or combination of waveforms that is most appropriate to emulate it, you've taken a huge step toward earning the title of synthesist. Likewise, if you can imagine a sound, and choose the right waves to make it a reality, you're well on the way to becoming a sound designer. Of course, it becomes harder to identify waveforms as the patches that contain them become more complex. You may think that if you envelope them, pass them through a filter or two and apply various forms of modulation, it would become impossible to identify the waves themselves. But you might be surprised. The ear/brain combination is a remarkable tool, and once you have learned to recognise waveforms in isolation, you'll be

Figure 11: A modifier, some controllers, and lots of modifiers controlling the controllers.

Figure 12: Representing Figure 11 in modular fashion.

Figure 13: A five-stage sforzando envelope.



amazed at your ability to identify them in more difficult circumstances.

Modifiers Modifiers are those parts of the synthesizer that modify the signals produced by other parts of the synth. Most obviously, these are the filters and amplifiers illustrated in Figure 6 (below). Unfortunately, many synth users seem to have become obsessed with the class of modifier known as the low-pass filter. Every discussion about synths seems to centre on similar questions: 'What filter does it have?' or 'What is its cutoff slope?' To be fair, the characteristic sound of a synth's filters is a major arbiter in the sounds that you can obtain from it. Nonetheless, we should not concentrate only on low-pass filters, because there are many other modifiers. High-pass filters, band-pass filters, band-reject filters and comb filters all have their place in shaping the timbre of your signals. Less obvious modifiers include sample & hold generators, ring modulators, frequency-shifters, slew generators, reverbs, and even chorus units, all of which have been described in detail during this series. There are no rules that say where these should lie in the signal chain, and if you want to modify your audio signals using, say, reverb and chorus before any form of filtering or loudness shaping, there are plenty of synths that will let you do so, especially affordable software-based ones.

Controllers The third major class in this scheme is that of the 'controllers'. These generate the signals that determine how the other parts of the synthesizer are operating. Again, there are obvious examples of these, including contour generators, LFOs, pitch-bend wheels, and joysticks... all of which can transmit control signals to modifiers such as amplifiers and filters, as well as to sources such as oscillators. We can therefore reconsider the synth architecture in Figure 2 and draw Figure 7 (above), in which each of the constituent parts is viewed in a much more general sense as either a source, a modifier or a controller. I like this representation because, although it is explicit about the shape of the patch, it does not fix the exact natures of the building blocks within it.

Source, Modifier Or Controller? Up to this point, I've kept everything simple, and although I've introduced the concepts of sources, modifiers and controllers, I haven't stepped far beyond the ideas of oscillators, filters, amplifier, envelopes and LFOs. But now I want to file:///H|/SOS%2004-07/The%20Secret%20Of%20The%20Big%20Red%20Button.htm (5 of 9)9/22/2005 8:58:51 PM


make everything more interesting by demonstrating that things are not always what they seem, and that there are many synthesizer modules that do not fit snugly inside a single classification. Take, for example, the simple architectures shown in Figures 8 and 9 (below), both of which contain two audio frequency oscillators and a mixer. In Figure 8, the outputs from the two oscillators pass as audio to the mixer, so we can say that both oscillators are acting as sources. But in Figure 9, the output from the second oscillator is being fed into the pitch control input of the first, so we can say that while Osc1 is still a source, Osc2 is now acting as a controller. This means that the classification of the oscillator is not determined by its operation, but by its position in the patch! You don't need a huge modular synth to encounter this. A perfect example exists in the world's first pre-patched, integrated synthesizer, the Minimoog, on which a knob in the Controllers section of its front panel determines whether Osc3 is acting as a modulator (a controller) in addition to, or instead of, its role as an audio oscillator (a source). Here's another example. Anyone who has read this series will be conversant with the fact that filters with two poles or more (ie. with 12dB-per-octave or steeper cutoff slopes) can exhibit 'resonance' and that, if pushed to the limit, the accentuation of the gain at the cutoff frequency will turn into sine-wave oscillation. This idea is illustrated very clearly in Figure 10 (above), which shows how you can use the oscillating filter to obtain a third pitch from a synthesizer — the Roland Juno 60 — that has just one oscillator. In other words, the filter is no longer acting just as a modifier, it is now a source. Clearly, you don't need a modular synth to think in a modular fashion, but thinking in these terms becomes even more useful as your synthesizer become more sophisticated. Take, for example, the trapezoid generator in the EMS VCS3. In normal operation, this acts as a contour generator; that is, as a controller. However, there is also a repeat mode in which the contour becomes a low-frequency cyclic waveform and acts as an LFO, which is another form of controller. But if you make the contour brief enough, the oscillation starts to move into the audio spectrum, at which point you could use it as either a controller or a source. It's now up to you to decide which is most appropriate, and to patch it into your sound in the way you feel to be most suitable. So here we have three very different synthesizers, the groundbreaking Minimoog, the simple Juno 60, and the complex EMS VCS3, all of which contain one or more modules that can act ambiguously as sources, modifiers or controllers. The same also applies in the modulation matrices of modern digital workstations; oscillators become controllers, modifiers become sources... and so on. As you can see, 'modular synthesis' can be a valid idea on almost any type of synthesizer.



Controlling Controllers Now, there's nothing stopping you from modifying control signals, just as you modify the audio signals generated by the sources. Indeed, placing controllable amplifiers in the signal paths between controllers and the things they are controlling is the basis of almost all synthesis, and you will find many VCAs (or their digital equivalents) in even a modest synthesizer. But there's no reason to limit yourself to using amplifiers... you can filter control signals to create new control signals, or shift their frequencies, or chop them up using S&H units, and apply a thousand other ideas. A simple example to illustrate this idea of controlling the controllers lies in Figure 11 (on the next page), which shows the filter section and contour generator of an ARP Axxe. In this case, the filter is a modifier, the ADSR envelope generator is a controller, and the three additional faders in the centre of the diagram (marked Kybd CV-S&H-Pedal, LFO, and ADSR) control the gain of three amplifiers (which are themselves modifiers) that determine how much of three controllers are applied to the modifier's cutoff frequency. Yikes! That's a heck of a sentence, so let's look at Figure 12 (also on the next page), which is the block diagram for this patch. This is much clearer, and if you learn to think of your patches in this way, you can be very much more explicit about what is happening when you create a sound on a synthesizer.

Adding A Bit Of Interest Even if you have a clear idea how to use the building blocks of your synthesizers, there are still a few other things it's a good idea to consider when designing and playing sounds. For example, it's sensible to think about timing signals, and be clear about the differences between triggers and gates. You should also make sure that you know when it might be preferable to use single-triggering, multitriggering, and the various types of key priority, as well as when monophony is superior to polyphony. You'll find all of these discussed if you look back over the course of this series. But not everything is cerebral. Just as important as your understanding of sounds is your ability to play them. If you want to sound good, it makes sense to develop good techniques for using controllers such as pitch-bend and modulation wheels, joysticks and aftertouch. Unfortunately, the number of synthesists who ignore articulation and expression has led to an oft-quoted statement, usually made by guitarists of the teeth-clenching, groin-thrusting variety, who believe that wringing the neck of a bit of dead tree constitutes a purer form of music. They claim that "synthesizers are just a big collection of soulless switches." They're wrong, of course, but the inability of many keyboard players to coax even a modicum of expression from a synth lends credibility to their views. You can't solely rely on contour and modulation generators to do the hard work for you. While their great strength is that they make everything consistent and repeatable, so that a sound is recognisably the same from one note to the next, their great weakness is that they make everything consistent and repeatable, so that every



note sounds the same from one to the next. So, next time you're making music, why not try using the physical controllers on your synth to adjust the way in which a note 'speaks', or add expressive vibrato (or tremolo, or growl) using the pitch-bend wheel or ribbon, rather than (say) relying on an LFO which always produces the same effect. It doesn't matter whether you're playing in an orchestral style, or prog-rock, or dance, or industrial techno... you might be surprised at the possibilities this opens up.

Strengths & Weaknesses If you are going to apply modular thinking to a range of synths, you have to become aware of each one's limitations as well as its strengths. Let me give you an example... 'Sforzando' is the name of a type of sound in which the loudness peaks very quickly, dies away, and then swells later in the note. Generating this contour (shown in Figure 13, right) requires a minimum of five stages. Unfortunately, ADSR contour generators have only four stages. Perversely, each of the three envelopes on the cheap, often denigrated Korg EX800 has five stages, and many cheap digital synths have envelopes with five, six, or even eight stages, meaning that they are capable of producing a sforzando brass patch, whereas the 'mighty' Prophets, Jupiters, and Memorymoogs can not. Consequently, you can't assume that all the facilities available on a cheap synth will be available on an expensive one, or that an expensive one can generate all the sounds that you can obtain from a cheap one. Nor can you make sweeping statements like 'analogue synths are better than digital ones'. If you want to create the sound of an overdriven low-pass filter sweeping through the spectrum of a sawtooth wave, it's probably true. If you want to imitate a brass instrument or an electric piano closely, or create a lush, evolving texture, it's probably not. You must therefore choose your instruments carefully so that you can produce those sounds if you want them.

Why Bother? Consider the following statement: the best way to develop a new recipe for a meal is to take random ingredients, throw them in a pot, and see if anything edible ensues. It's rubbish, isn't it? If you were invited to a dinner prepared in this fashion, you wouldn't return for seconds. So, whether you want to imitate existing sounds or program interesting new ones, it's vital to be able to select the appropriate sources, the right sort of modifiers, and the controllers that can shape the sound and let you control it in the ways that you want. If you do not, it's likely that your random twiddlings will generate another filter sweep that sounds little different



from a billion other filter sweeps created over the past four decades. That's why I've spent the past three years showing you how it's possible to analyse sounds as diverse as brass instruments, guitars, orchestral percussion, electronic percussion, pianos, strings, woodwind and electromechanical organs, and how you might use the building blocks of common synths to emulate them. I've tried to make everything as general as possible, and have provided examples using a wide range of synths. Unsurprisingly, a handful of readers objected... they simply wanted me to show them how to set this voltage-controlled wotsit to that value, and thereby obtain the sound used on the latest chart-topping smash. In short, they wanted to know where the Big Red Button was. However, the point was not to show you the settings that synthesize a particular sound, but to teach you how to think in a modular fashion, no matter what synths you might own, and thereby allow you to work out the settings for yourself. Of course, you may not be interested in understanding how and why synthesizers do what they do, and you may be perfectly content with serendipitous experimentation. If so, that's fine. But if you want to get the best from your instruments, I think you ought to go further than twiddling and hoping. Even modest synths allow you to think about synthesis in a modular fashion, so why not try it? So — which is the best synthesizer? That's easy. It's the one that allows you to obtain the sounds you want. But whichever one you use, you should remember one last thing... there is no Big Red Button. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Using OMF to transfer Sonar files

In this article:


The Magic Of OMF Sonar Notes The Export Process Published in SOS July 2004 Quick Tips The Import Process Print article : Close window Automation Envelope Tricks Technique : Sonar Notes Copying & Pasting Envelopes Easy Clip Envelope Copying Getting Advanced Too Many Envelopes!

This month we look at clever envelope tricks, as well as how to transfer projects between DAWs that speak OMF. Craig Anderton

You work with Sonar, but your keyboard player uses Cubase SX and you want to collaborate. So you go through the tedious process of rendering all your soft synths as hard disk tracks, saving each audio file as a WAV that starts at the beginning of the song, then saving a standard MIDI file for any MIDI parts. But wait! It doesn't have to be quite that complex, thanks to OMF (Open Media Format). Sonar can export all its audio information as a single OMF file, which other OMF-compatible applications can then import. In the process, the target program recreates the same project, puts all the audio files along the timeline in the proper order, and even grabs the correct track names.

The Magic Of OMF This process works very well, but there are a few cautions. First, OMF deals only with audio tracks — not MIDI, soft synths or plug-ins. You'll need to render any soft synth tracks to hard disk audio files, and if you're using MIDI tracks they'll have to be exported separately as Standard MIDI Files and imported into the target program. OMF also doesn't deal with plug-ins, so you've three options: Hope that your collaborator has the same plug-ins. In that case, you may be able to save presets from within the plug-in, and hopefully they'll load into the target's plug-ins. (Note that Sonar's Plug-in Manager allows trading presets only with other Sonar users.) Apply the plug-in, then render the track as an audio file, which can be exported within the OMF file. file:///H|/SOS%2004-07/Using%20OMF%20to%20transfer%20Sonar%20files.htm (1 of 6)9/22/2005 8:58:56 PM


Provide the raw audio track within the OMF file, but also include a text file with detailed instructions as to plug-in parameter values (as an example, for a delay you'd specify the delay time, amount of feedback, delay/ straight mix, and so on).

The Export Process

The export process is quite straightforward. Just remember that OMF deals only with audio tracks, so you'll need to export MIDI data separately.

Once you have the situation with soft synths and plug-ins sorted, it's time to export. Go File / Export / OMF, to bring up a window with a variety of options. You can save as OMF Version 1 or OMF Version 2, but I found that Version 2 worked and Version 1 didn't, so go with Version 2. Next, choose your Audio Packaging. It's most convenient to embed the audio within OMF, as that creates a single, simple file with everything you need — rather like Sonar's Bundle format. If you choose 'Reference Audio Externally', that creates a separate folder containing the audio that the OMF file references. Tick 'Include Archived Tracks' to include tracks within the project that are archived, and if you use Groove Clips, I suggest ticking 'Mix Each Groove Clip as a Separate Clip'. For Audio Format, I usually tick WAV, unless there's a compelling reason not to — for example, if you're trying to export to a Mac program that doesn't understand WAV files (which is unlikely). When you click on 'Save', Sonar commences processing. After watching various progress bars fly across the bottom of the screen, you eventually end up with your OMF file.

Quick Tips To see which MIDI output corresponds to a physical MIDI device, go Options / MIDI Devices. The first selected output device goes to Port 1, the second to Port 2, and so on. Miss the Sonar 2 color scheme? No problem. Go Options / Colors, and under Presets, choose 'Sonar 2'. To record on multiple tracks, press the Arm button for each track. Just make sure the correct input is assigned to each of them. To change note properties in the Staff or Piano Roll view, right-click on the note and edit the desired option.

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The Import Process Importing, at least with Cubase SX, is simple, and most other programs follow a similar protocol. Just go Import / OMF, navigate to your OMF file and click on 'Open'. You'll see your project tracks sitting just as they should, with their proper names. Isn't technology wonderful?

Automation Envelope Tricks Like most other hosts, Sonar uses envelopes for automation. You can either draw these 'from scratch' or create them by arming parameters for automation and moving their associated controls. These motions are translated into envelopes, which control the parameters on playback. As to which is preferable, it depends on your application. When envelopes are used to create time-related effects (for example, volume changes that occur on the beat), you can generate far more precise envelopes by drawing them in. But if you're going for something like expression or volume changes, you probably want to do these in real time to capture the desired 'feel'. Double-clicking on an envelope creates a 'node'; you can click on it and drag it to the desired level and place on the timeline. But you can also change the shape of the segment between nodes, which is quite cool. Right-click on the line between two nodes and you'll see a menu offering four curve options: Jump, Linear, Fast Curve (concave shape), and Slow Curve (convex shape). All of these are fairly obvious, except Jump: it means that the level stays at the first node's value until the Now line traverses the second node, at which point the value jumps abruptly to that of the second node. I find this option very useful for Here a periodic tremolo envelope has been creating 'sample and hold'-type effects drawn for one measure, then copied, and is when applying the envelope to filter about to be pasted to cover 15 additional cutoff. Just place nodes at the desired measures. rhythmic points (note that you can snap notes to rhythmic values), drag them around to various values, and select the Jump slope. As working with individual nodes can get tedious, Sonar offers several tools that allow you to edit groups of nodes, or portions of an envelope. The simplest of these is as follows: click on the envelope segment between two nodes, then drag to raise or lower the level of both nodes simultaneously. To edit multiple



segments, shift-click on additional nodes. If they're all contiguous, clicking on any of the segments will raise or lower all the curves between these nodes. However, you can also shift-click to select non-contiguous segments. For example, if you shift-click on two nodes at the ends of one segment, then shiftclick on two more nodes at the ends of another segment, raising or lowering one will raise or lower the other, even if they're not adjoining. There's also a faster way than shift-clicking to select contiguous nodes: click and drag the cursor in the timeline above the envelope. When you release the mouse button, all nodes within that region (for the track that has the focus) will be selected. Deleting nodes is simple: you just click on a node (or shift-click on multiple nodes) and then hit the keyboard's Delete key. It's also possible to right-click on a node and select 'Delete Node', but note that this affects only the one node on which you right-clicked, not all that are selected.

Copying & Pasting Envelopes It's possible to copy and paste envelopes independently of any other track parameters; we discussed this in the May 2004 Sonar Notes, in response to a reader query about how to have the exact same envelope on two different tracks. Briefly, you copy a track by selecting it, right-clicking on a Clip or empty space in the track and choosing 'Copy' (make sure all boxes are unticked except for Track/ Buss automation), then clicking 'OK'. You then paste the envelope where desired. It's also possible to copy just the section of the envelope that passes over a particular Clip. The process is almost identical. As long as the entire track isn't selected, you can right-click on a particular Clip, then select 'Copy' and 'Track/ Buss Automation' only. When you paste, only this portion of the envelope will be pasted. Clip Envelopes can also be copied and pasted independently of, or along with, track envelopes. For example, suppose there's a Clip with a Gain envelope and a track envelope for Pan that passes over the Clip. If you right-click on the Clip and select 'Copy', you can tick 'Clip Automation' as well as (or instead of) 'Track/Bus Automation'. Then, when you paste, both envelopes will be pasted. However, note that if you select Clip Automation, the Clip itself will be copied too (as indicated by the 'Events in Tracks' box being automatically checked). This makes sense; you can't expect Sonar to copy a Clip envelope unless there's a Clip involved.

Easy Clip Envelope Copying As a Clip envelope is often an integral part of the Clip, you probably don't want to



go through the hassle of doing a multi-step cut and paste — especially since there's a tool designed expressly for this purpose. Go to the Track view Toolbar, click on the downward-pointing arrow next to the cursor icon (second from the left), and choose 'Select Track Envelopes with Selected Clips'. Now when you copy and paste Clips, any associated envelopes will go along for the ride. (As with the standard envelope-copying procedure, you can choose whether to copy/paste Clip Automation, Track Automation, or both).

Getting Advanced I've often enjoyed the Cubase feature that lets you draw envelopes with repetitive waveforms — just the thing for tempo-sync'd effects such as tremolo. Sonar doesn't have this feature per se, but you can come close using the advanced Paste features. You'll need to draw one complete cycle of the repetitive-envelope-to-be, and copy it. But when you go to paste it, make sure the Paste Advanced dialogue is showing. Here you can specify a particular number of repetitions (as well as where you want the envelope to start, if somewhere other than the current cursor position, and whether it will replace an envelope or slide it to the right to make room for the new one). You can even almost 'morph' two envelopes together, if you paste an envelope into a track that already has an envelope of the same type and select 'Blend Old with New' as the paste option. See the screen above,

Too Many Envelopes! Sonar overlays all track envelopes in the same track. This has advantages and disadvantages: It makes the overall workspace less cluttered (seemingly a Cakewalk design goal), but makes the individual tracks more cluttered if there are a lot of envelopes. Fortunately, there are several show/hide options. The most basic one is to rightclick on an envelope and select 'Hide Envelope' (logical enough). But you can also right-click on the track itself and go Envelopes / Show Track Envelopes. From here, you can untick envelopes you want to hide. To show them again, follow the same path but this time, tick them. Published in SOS July 2004






Using Software Effects & Outboard

In this article:


Combining Plug-ins Masterclass Effect Or Processor? Setting Up An Effects Loop Published in SOS July 2004 Using Hardware Effects For Print article : Close window Your Software Mixdown Technique : Effects/Processing Delay Compensation

Get the most from your computer's CPU by learning how to put your effects plug-ins where they really count. Plus, find out how to increase your mixing power by incorporating hardware effects units into your software mixdown. Paul White

Those of us who started recording back in the days when hardware was used for everything soon got the hang of where signal processors could or could not be connected. However, the software world isn't always quite so intuitive, especially if you've never used a hardware mixer before. In the virtual world, effects boxes are often replaced by plug-ins, and the first thing you need to know when patching in a plug-in is whether it is an 'effect' or a 'processor'. This basic designation is something that we've explained many times in the past in relation to hardware, but it also applies to plug-ins. When it comes to new users, it's one of the first places they come unstuck, so I make no apologies for recapping.

Combining Plug-ins My rules for using effects and processors have to be adapted slightly when you want to combine an effect and a processor. For example, you may want to place an EQ, which is a processor, before a reverb plug-in, which is an effect. In most cases, where effects and processors are combined in this way, the resulting combination can be considered as still being an effect, so placing compressors and equalisers before or after an effect being used in an aux send is quite OK. As a rule of thumb, go by the status of the last plug-in in the chain.

Effect Or Processor?

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Although pretty much any plug-in that changes the sound in some way can be thought of as being a signal processor of some kind, the 'effect' and 'processor' designation helps us divide these devices into their two main categories, which in turn tells us where we can connect them. Put simply, a processor passes the whole signal and imparts some change to it, the most common examples being EQ, compression, limiting, gating, expansion, distortion, and filtering. Effects, on the other hand, are designed to be combined with the untreated signal, which is why most feature a wet/dry mix control. Prime examples of effects are delay, echo, reverb, chorus, and flanging — in fact pretty much anything that uses delay as part of its means of operation. Each channel on a hardware mixer (at least, any half-decent one) includes something called an insert point, a physical connection that allows you to interrupt the signal path and send the signal through some external effect or processor. Virtual mixers also have insert points, but instead of these feeding external hardware boxes they allow you to place a plug-in into the signal path. Insert points are very flexible, so you can use either effects or processors in them — the only thing you have to remember is that when using effects the amount of effect added is adjusted using the mix control on the effect plug-in itself. Most virtual mixers allow you to add multiple insert plug-ins, in which case the signal passes through them sequentially, usually from the top of the screen to the bottom, as shown in Figure 1.

Setting Up An Effects Loop

Figure 1. Using multiple insert effects on one channel.

The other way to connect effects is to use a post-fade aux send, sometimes called an effects send. Each mixer channel will have one or more aux send controls, usually numbered. All the aux-one sends are mixed together and, on a hardware mixer, this mixed signal is sent via the master aux-one output socket to an external effects box. All the aux-two sends are mixed and feed the aux-two master control and output, which allows you to add a second effect. The output from an effects box fed from an aux send is then added back into the mix via a special input called an aux return, and these are often stereo simply because many effects boxes have stereo outputs. Although an aux return has a fancy name, it's still just another type of mixer input and it is added to the mix of all the other mixer channels feeding the stereo mix. Note that you don't have to return effects connected to aux send one to aux return one, though it sometimes helps keep your thoughts in order to do so. Figure 2 shows an effects send and return setup as it relates to a conventional mixer. The insert points are also shown, along with the pre-fade send normally used for foldback. Virtual mixers employ a very similar system of aux sends, but, rather than feed an external effects box, the channel aux send controls are routed to an effect plug-in and the output from the plug-in is routed back into the stereo mix. Steinberg's Cubase does this by using a virtual effects rack, which is



conceptually similar to the way things work with a hardware mixer, though Emagic Logic does things slightly differently. With Logic, the channel post-fade aux sends can be routed to either physical soundcard outputs or to mixer busses, so the normal approach is to use a buss as an aux return and then route its output back into the stereo mix. The sends are then set up to feed this buss, and the desired effect slotted into the buss insert point. In terms of signal flow, this is again very close to what happens on a hardware mixer. If the effect is a reverb plug-in, turning up the aux-one send control on mixer channel one will send some of that signal to the reverb unit. When the output from the reverb is added back to the main mix, the audio signal on channel one will be heard with reverb added to it. Because the untreated part of the signal is still going via channel one, the plug-in's mix control should be set to 100 percent wet so that the only thing the plug-in contributes is effect, and not more of the original dry signal. The great advantage of the aux send system is that a single reverb plug in (or other effect) can be used to treat as many mixer channels as you Figure 2. An effects send/return like. The character of the effects will of course be setup working with a the same in all cases, but the amount of effect conventional mixer. added can be adjusted independently for each channel using the channel's aux send control. Incidentally, post-fade sends are used for effects, so that when the channel fader is adjusted, the effects send level is automatically adjusted by the same amount, thus ensuring a consistent amount of added effect. If a pre-fade send were to be used, the effect level would remain constant even if the channel fader were turned right off. While effects can be fed from the pre-fade aux send system, the same is not generally true of processors, as these are designed to work with no dry signal added. There are some exceptions, but for the most part these should be left to advanced users who know exactly what they are doing and why. If you learn nothing else from this article, it is important that you remember the rules for effect and processor connection: effects can be connected either via channel insert points or via the aux send system, but Processors should normally only be used in channel or subgroup insert points, not via an aux send loop.

Using Hardware Effects For Your Software Mixdown One common problem is working out how to use an external hardware effect or processor with a virtual mixer. While most plug-ins are perfectly adequate for the task, hardware reverbs still tend to be better than many of their software file:///H|/SOS%2004-07/Using%20Software%20Effects%20&%20Outboard.htm (3 of 5)9/22/2005 8:59:02 PM


counterparts, mainly because there's a limit to how much of a CPU's resources a plug-in can decently ask to tie up. The newer convolution reverbs are an exception, but they are very CPU hungry. If you have an audio interface with more than two outputs, you can use an external hardware device, but exactly how you do this depends on whether you also have an external hardware mixer. I imagine most computer studio users will have at least a small hardware mixer to combine their soundcard outputs with the outputs from any external MIDI synths, in which case all you need is a pair of spare mixer inputs and you can use a hardware effects unit. The way this works is to set up the post-fade send in the virtual mixer as usual, but instead of routing it to an effects plug-in, you send it to one of the spare outputs on the audio interface. This output feeds the input to your reverb unit and the reverb unit's outputs go into the hardware mixer panned left and right, as shown in Figure 3.

Figure 3. Setting up a send to a hardware reverb when using Emagic Logic with an external mixer.

If you have spare audio interface inputs and your sequencer can add live inputs to the mix (as can the current versions of both Steinberg Cubase and Emagic Logic), you also have the option of feeding the hardware reverb outputs back into the mix directly, as shown in Figure 4. This is useful if you don't use a hardware mixer. In some cases, working this way will add a little delay to the reverb because of the system latency, but as reverb is traditionally pre-delayed by several tens of milliseconds anyway, this is unlikely to cause problems. It's less easy to use external processors as insert effects, especially if you have a limited number of additional audio outputs on your interface card, but it can be done. The usual method is to route the track to be processed to one of the spare audio outputs, then feed it into a hardware mixer channel with the desired processor connected via its insert point. If there is no hardware mixer, then you can route the soundcard output directly to the processor before feeding the processor's output back into a spare soundcard input. If you run this setup, you may find that the system latency delays the processed input by too much, in which case you can either record the processed signal as a fresh audio track, or apply negative delay to the audio track being processed to bring the processed input back into line with the rest of the song. Of course you can only use as many hardware insert effects as you have spare audio interface inputs and outputs.

Delay Compensation file:///H|/SOS%2004-07/Using%20Software%20Effects%20&%20Outboard.htm (4 of 5)9/22/2005 8:59:02 PM


Every plug-in takes a finite time to process audio, so the output is delayed slightly relative to the input. This can be a problem when using sequencers that don't offer automatic plug-in delay compensation, as any processed tracks will play slightly late. Even where the host software does compensate automatically, it relies on the plug-in reporting its latency figure correctly — and not all do. Furthermore, not all software handles Figure 4. Using a hardware reverb with a plug-in delay compensation in the virtual mixer. same way — in Emagic Logic, for example, you only get delay compensation in the channels, not in the busses or outputs. You may also have to check that plug-in delay compensation is active in the software's preferences. Where you're feeding a reverb from a send and adding only the wet signal back into the mix, this isn't usually a problem, because a few milliseconds of delay simply adds to whatever pre-delay value you've set anyway. However, with other effects it may become significant, so when you next contact your sequencer manufacturer to suggest new features, add full plug-in delay compensation to your list! Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Alternative Careers for Sound Engineers

In this article:


Forensic Audio New Directions Expert Witnesses Published in SOS July 2004 System Requirements Only In America... Print article : Close window Audio Archiving Music Business In The Corporate Reel World Looks Are Everything Sound Design

The music industry's down, but your passion for audio is still high. Fortunately, there are alternatives to working in the studio that can feed your fervency and still make you a living — perhaps even a better one. Dan Daley

Looking at the music sales charts these days is a bit like looking at the dentist's bill. You know what's there, you know it's unpleasant, you know you have to deal with it, but still you avert your eyes. Don't, because there are more ways than one to employ your audio skills and desires to work with sound than working the increasingly tenuous landscape of recording studios. No, we're not talking about becoming a pirate. But we do want to point out that there has always been more than one way to skin the cat of a career in audio. The alternatives may not be as glamourous (although some actually are more so) and they may not be as readily apparent as the goal that set you down this road in the first place. But alternatives are there aplenty. This article looks at just a few of the possibilities, examines what brought certain people to them, and considers what it takes for others to get there.

Forensic Audio Let's start with what might be the most offbeat but coolest possibility. Forensic audio is the stuff TV docudramas are made of. Specifically, it's the application of audio engineering skills to analyse, clarify, enhance, edit and detect sounds recorded onto magnetic, digital and or other types of media. It also consists of other elements, including transcriptions, recording depositions, providing expert witness testimony, doing identifications of and authenticating recorded audio, and various other services. Think it sounds a bit dry? Listen to Arlo West recall a notuntypical case and compare it to a session spent doing 100 takes of the same file:///H|/SOS%2004-07/Alternative%20Careers%20for%20Sound%20Engineers.htm (1 of 11)9/22/2005 8:59:15 PM


line of a vocal. "A police department in Texas sent me a tape of a lady who dialled police emergency 911 when someone broke into her house, and she stayed on the line while the intruder was in her house. She never hung up the phone. Not even when he killed her." West owns a one-man forensic audio operation in the small town of Auburn, Maine. Nearly two decades ago, he Arlo West at work — it's not often you see an was a regular client of Dallas Sound audio engineer in a suit and tie! Labs in Texas, an audio recording studio in the Lone Star state, and slowly evolved from being a musician to being an engineer. "I just kept coming into the control room during sessions and changing the sounds from in there," he says. "I liked getting behind the board, to the point where I decided to make a career out of it. I made a lot of good records there over the years, and worked on films like Titanic and with artists like U2." However, personal circumstances changed and in the late 1990s, West migrated back to Maine, where he found more work as a musician than as an engineer. In the course of a conversation with his attorney, to whom he owed money for services rendered, he recalled that he had done some forensic audio work at Dallas Sound Labs, enhancing the sound of police emergency tapes so attorneys and investigators could figure out exactly what had happened on a particular case. "So on a whim I asked my attorney who did his forensic audio work," says West. "And he told me he had to send it out of state, that no one in the entire state of Maine did that. The he asked if I could do it. I told him about the work I'd done and he jumped all over me. He told me to start that business now."

Expert Witnesses That timely advice has paid off handsomely in the two years since West took it. Compared to the $600 or so that most in-the-trenches engineers in the US earn for a day's labour, West's card rate for work such as audio enhancement and track cleaning is $125 per hour, with a minimum $500 retainer required just to walk in the door. He gets $1000 a day for in-court testimony, plus expenses, and charges $50 for a CD-R copy of any work he does. What's perhaps more astounding is that he rarely gets any dickering on costs — "Attorneys just want to get the job done, and they're used to paying pretty steep fees for expert work and testimony," says West — and that he considers his rates on the low side of what similar service providers get, noting that his relatively remote location won't allow him to charge what someone in, say, New York might get.

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West has a relatively Spartan equipment complement; most of his work is done in Sound Forge v6, running on three home-made computers. "I can hot-rod them as I need to that way," he says. He also uses Enhanced Audio's Diamond Cut Live 32 software, which was made with forensic audio applications in mind. "Diamond Cut Live is both a recording and an editing system, and it's being increasing used for recording emergency calls by police departments that are going digital," he explains. "Most emergency calls are recorded to tape, which is played at a very slow speed and the tape itself is used over and over again until it's falling apart. That's why so much of forensic audio work is simply making those tapes intelligible again." West has a Mackie 1604 mixer but says he rarely uses it; most audio subjected to forensic analysis consists of a single Philip Harrison's working area at JP French mono track, anyway. Aside from an array of Direct X plug-ins for EQ and de- Associates. essing, he'll occasionally employ a Focusrite Platinum Voicemaster Pro for outboard processing. Sometimes audio can't simply be brought in on media. When he has to retrieve audio from a digital voicemail box, of the type offered as a service by local telephone companies, for instance, he'll often get the client's password, then play the messages to be analysed back through a speakerphone and record it to the hard drive using a microphone instead of a DI. "You can choose a microphone for that application just as you would for a vocal or a drum," he says. "The type of microphone can help you get the sound back so it's listenable. After that, it's mostly a de-hiss and de-click job, generally looking for the frequencies that are most apparent in the voice in question and eliminating everything else." Another regular project is removing the muffled sound from tapes when clients hide recorders under layers of clothing. "Ninety percent off the time, it's muffled with bad signal-to-noise and tape hiss," he explains. "It sounds like people are talking over the noise of a jet engine. You need to whittle down the unwanted noise but not remove the desired spoken words. It can be tricky." Other facets of forensic audio are more subtle. "Sometimes unidentified sounds on a tape can have significant meaning — they can perhaps place a person somewhere at a certain time or point to a location from the environmental background sounds. These sounds can have many other meanings as well: were



they using an AC adaptor or batteries? Sometimes the speed of the tape is different when you use the AC adaptor, suggesting that it was done near an electrical outlet and not in some remote location as may be proclaimed. "Also, has the original tape recording been tampered with? When the tape head engages the tape by pressing Record, it leaves a distinct impression. This impression can be found everywhere on the tape where the individual has engaged the Record button. Gunshots, thumps, bangs — any sound will have a unique waveform pattern, a waveform 'fingerprint', so to speak. Waveform examination as well as spectrographic scrutiny and other means of investigation, including physically looking at the tape, can disclose a whole scope of evidence. Then there's audio restoration. I remember doing a job for a person who had sent his cassette through the washing machine. You would think that that would have completely destroyed any audio on the tape. It actually did damage the recording but with careful analysis it was restored. Not to its original state, but well enough to be transcribed and used." Forensic audio can be as intense as any studio session. West recalls a client, a doctor, who was suspicious of her fiancé. She planted a digital recorder under the seat of his car to document any dalliances that might take place there. The resulting tape was garbled and muffled, as might be expected, but the voices of a man and woman were recognisable. The medic, frantic that her betrothed was stepping out on her, pleaded with West to unmask her rival. After working on the tape for a few hours, he was able to figure it out for her. "It was the radio, and there were two DJs on, a man and a woman, and they were like 'shock jocks', getting a little raunchy on the air," he says. "When she found out that that was what it was, you never saw a more relieved or thankful client."

System Requirements Each of these alternative audio pursuits has its own special needs in terms of gear and expertise. But each is attainable. Here's a nutshell: Forensic Audio Capital investment: Relatively small — a computer-based DAW and signalprocessing plug-ins. Expertise: Ability to digitise and then manipulate audio to isolate specific data; understanding of basic legal standards and events. Ability to approach audio obliquely — what does a gunshot sound like in a thunderstorm? — much as a sonar operator learns to identify ships by their acoustical signatures. Marketing quotient: In the US at least, business can be built up by approaching local lawyers and private investigators, which can be easily researched in local phone directories. Potential return on investment: Depends on country, region and level of work done, but lawyers are used to paying the asking fee for a range of technical forensic services. Archive Audio



Capital investment: Relatively small — a computer-based DAW and signal processing plug-ins. An array of various tape formats to access sources. Portability preferred, since much of the work will be done on site. Expertise: Ability to stabilise and restore decaying analogue audio formats, digitise them and then help maintain them through evolving digital format generations. Marketing quotient: Approaching local academic institutions, as well as historical societies and large companies. Potential return on investment: As a guide, archivists in US academia make between $25,000 and $50,000 per annum. However, as the need to digitise and manage large volumes of data increases, the potential in the commercial end can be considerably more. Corporate Audio Capital investment: Can be significant — in addition to enough equipment to outfit a capable studio, the facility will need aesthetic as well as acoustical design to make it acceptable to top-tier clients. Expertise: As much creative as technical. Marketing quotient: High. As much time needs to be spent on cultivating business in a narrow slice of the world as on creating content for it. Few people can operate as a one-person shop. A partnership between a creative type and a business type is common. Potential return on investment: Very high. Successful corporate events audio professionals can earn into the low hundred-thousands — dollars or Sterling. Sound Design Capital investment: As little or as much as you can. The sector is moving towards an all-laptop environment. Expertise: As much creative as technical. Marketing quotient: Mostly networking, but also scanning the landscape for opportunities for sound design applications that potential clients aren't aware exist, such as local radio commercials and web sites. Potential return on investment: Difficult to discern. Sound designers compete with musicians and particularly creative engineers in the perception of many. Revenue will likely be directly related to how well you create an identity for yourself and your sonic signature, as well finding niches for them that more traditional audio folks don't see.

Only In America... Thanks to the different legal system, there are fewer opportunities for private forensic audio specialists in the UK, but they do exist and their numbers are growing. Philip Harrison, forensics consultant with JP French Associates, has seen his work used on a number of high-profile cases, including ongoing work with the Bloody Sunday enquiries and the UN War Crimes tribunal stemming from the former Yugoslavia. Less globally critical but just as interesting was his work on the infamous Who Wants To Be A Millionaire? cheating scandal. When it



was alleged that one of the contestants had been receiving leads via coughed signals, Harrison was asked to investigate. "We found first that the coughs were coming from within the circle of contestants, one of the people wearing [lavalier] microphones," he recalls. "A particular contestant was in the 'hot seat' and when asked a question would ponder the responses out loud, saying something like, 'It could be A or perhaps it's B but then again it could be C,' and so on, and when someone would cough when he said the right choice. The microphones had been mixed down to mono for the Digibeta track of the video, however. So we used the stereo audience microphones — four on each side of the studio — to determine directionality of the source of the suspect coughs. And we were able, using analysis of the audio, to determine that pretty precisely." Precisely enough to contribute to the conviction of the contestant and two other defendants. Harrison, who did his first multitrack recording at age nine (and who says he spent his teens reading Sound On Sound but was constantly disappointed because "I could never afford to buy anything advertised in it"), took Acoustics Studies at Southampton University. When he decided to take a year's time off between classes, he sent out numerous letters to prospective employers from a list provided by the Institute Of Acoustics. The first positive reply he got was from JP French Associates, and he's loved the work ever since. "You never know what you're going to get from one day to the next," he says. "A bit different from setting up a drum kit every day."

Audio Archiving They're at work every day, in the British Museum, in the Smithsonian Institute and the Library Of Congress. They are a small army of audio archiving specialists, many of whom began their careers making music. The irony is perfect, considering how critical to the music business has become the ability to restore old music, repackage it in new formats and resell it. There's music in those archives, everything from old Edison cylinders to the music of the spheres — celestial creakings recorded for the still-ongoing SETI (Search for Extra-Terrestrial Intelligence) project. But there's even more spoken words — oral histories recorded to document not just the specifics of a life but its emotions, as well. Historians estimate there are literally tens of millions of hours of such oral histories stashed at governmental, organisational and academic institutions globally. Most are on analogue tape, and many of them are deteriorating faster than they can be

A large part of audio archivist Curtis Peoples's work consists of stabilising decaying analogue media and transferring it to digital formats.



saved. Projects such as the Ellis Island Oral History Project, and the EUsponsored Echo Project, are challenges to audio archivists globally. It may not be as lucrative as forensic audio, but it is equally emotionally and professionally demanding and rewarding. Curtis Peoples was working as an intern in a studio in Lubbock, Texas, in 1994, when he graduated from South Plains College with a two-year Associate degree in audio engineering. After a few years of scavenging for assistant engineering jobs and some live sound work on the side, he decided he needed to secure his future and went back to college to get a degree in another area of personal interest: history. At Texas Tech University in Lubbock, as an undergraduate, he volunteered to work on the university's Vietnam Archives project, which was collecting and organising artifacts from the war. As it turned out, these would eventually include thousands of hours of oral histories recorded on analogue tape, much of which was decaying. Peoples applied his knowledge of audio, and researched 'sticky shed' syndrome — the breakdown of the binding emulsion which holds the tape formulation to the tape backing. "I discovered that that tape emulsion formulations had changed in the 1970s and 1980s, mainly due to more intense demands by the music industry to increase tape's technical characteristics," he explains. "The newer tape formulations were more prone to collect ambient moisture, especially if they weren't stored in climate-controlled conditions. Moisture unlocks 'sticky shed' syndrome, and other problems of field recordings, like uneven tape winding, also contribute to moisture getting into the tapes and starting decay. Much of this stuff was stored in basements and attics, full of dust and mould." Peoples eventually became the chief archivist for the Vietnam project, and since then has moved over to the same position for the university's Southwest Collections archive, which chronicles the region's history. (You can listen to some of the oral histories of both projects at www.vietnam.ttu. edu and www.swco.ttu.edu.) Using a recently acquired Quadriga system from German manufacturer Cube-Tec, With tens of millions of hours of recordings in which digitises audio into the EBU need of rescue worldwide, audio archivists Broadcast Wave file format (BWF) as aren't going to run out of material for a while... well as automatically logging metadata and audio artifacts for later cleaning, Peoples is now digitising oral histories by the dozens. He's already looking to the future, realising that what's going on in academia now is the tip of the iceberg in terms of the oceans of analogue information that will have to be digitally archived. "The commercial potential for this is enormous," he says. "It's going to be nothing but a growth area." Indeed, growth at companies like Iron Mountain, which archive corporate and governmental sites globally, reflects the fact that what was once considered so much junk is now being recognised in the digital era as valuable 'content'.



The Quadriga system costs nearly $20,000, but much of the rest of the technology Peoples and other audio archivists use is relatively inexpensive; besides a hard disk-based workstation, he suggests a good A-D converter and an array of tape (and if possible, wire) recorders. The more arcane the playback source, the better. "Fewer companies make open-reel recorders, so having them is a valuable tool in commercialising audio archiving," he says. That applies to turntables, as well, since there are hundreds of thousands of recordings that never went beyond that format. "It's a very good start-up type of business, since it doesn't require a lot of capital," he says. "But it does require some very specific skills."

In The Corporate Reel World The concept of corporate audio is perhaps the most antithetical to the notion of passion for sound. But it takes a similar dedication to launch a new product as it does a new band. In fact, with globalisation the buzzword in the corporate world, corporate sound is perhaps truly the most international of audio avenues. Paul Guzzone, producer and owner of Triple Z Music in Manhattan, found that out when he flew to Japan for IBM to create the music for the global computer maker's sponsorship efforts for the 1998 Winter Olympics there. He worked with the Japanese band Kotoza to create a theme track for IBM based around the traditional Japanese instrument the koto. "We wanted to take koto music and create a song Paul Guzzone of Triple Z Music specialises in creating music for corporate events. using it in a western pop music form," Guzzone explains. "Ultimately, I took a track of a solo koto played by the woman in the band — the koto is traditionally played by women — and brought it back to my studio on a DAT and built a new track around it. Then I sent the track back to Japan where she played the solo part again but to the new backing track." Guzzone, a bass player by trade, a jingle writer by ambition and an engineer by necessity, as many musicians are these days, sought to augment his income about 15 years ago by going to work for Brielle Music, then a leading force in corporate event production — corporate theatre, as it's known in some circles. Guzzone learnt the trade and gave it his own spin a dozen years ago, doing productions for new auto introductions and Time magazine's 75th anniversary party at Radio City Music Hall. "I got into it at a good time," he says. "I hit my stride as corporate events were getting bigger and more elaborate."



Looks Are Everything It's not a piece of cake, however. Guzzone can count on only a couple of major events a year, plus other gigs that diversity demands, although those tend to feed the corporate work. He plays bass with and co-produced the last album for the Bacon Brothers, the band fronted by actor Kevin Bacon and his brother, and is just embarking on producing pioneering alt-folk-rockers Aztec Two Step's comeback album. "The albums give you musical credibility with the corporate world, and the corporate work tends to make you look reliable and productive to music clients," he explains. Business comes via word of mouth, mainly, in the close-knit world of corporate events coordinators. Having one's own studio is necessary, and while it doesn't have to be a palace, it does have to look good. Guzzone says that a decent portion of the estimated $100,000 he's spent over time on gear and facility has gone into making his studio/office suite in Manhattan look inviting to clients. It's where the deal is When you're dealing with high-flying corporate types, it's crucial that your studio often closed, though not before a lot of looks as good as it sounds. work, mostly meetings, trying to suss out what the client is looking for. "You play a No Doubt record, they say no, you play Linkin Park, they say yes," he says. "It's finding out what they're looking for." A personal studio is necessary because clients want revisions and they want them quickly. It also permits demos for prospective projects to be done costeffectively and with the studio always available. Locating in New York puts Guzzone's studio close to the city's truncated but still-exceptional pool of musical talent. "The good part is that you don't need a lot of gear," Guzzone points out. "A hard disk recording system, a small iso booth, a couple of nice microphones and mic pres — I have Avalon and Drawmer — and not much else. It's not so much the equipment as it is understanding what the client wants and being able to be personable." However, synchronisation is important — many corporate events are multimedia, and live music and dancers will perform in synch with prerecorded tracks. "Pro Tools pretty much takes care of that using the Universal Slave Driver, and I convert the video pre-records from VHS to Quicktime movies for working purposes," he says. The rewards can be good — as much as $40,000 to $50,000 per major event, plus between $3500 and $4500 per demo. But that includes sometimes seemingly endless meetings with clients and other media producers on the event, and often as many rewrites, plus the time spent waiting for approval of lyrics and melodies, less the cost of additional musicians on tracks, and time spent on site before and during and event. And not every bid will get accepted — there is competition out there. "It can be a good living, but you will have to work file:///H|/SOS%2004-07/Alternative%20Careers%20for%20Sound%20Engineers.htm (9 of 11)9/22/2005 8:59:15 PM


for it and at it," says Guzzone.

Sound Design The term 'sound design' has been current for 25 years, starting with Star Wars and the first Star Trek film, though as yet it lacks its own Oscar category. Sound design is that place between music and sound effects that seems so hard to define but now seems impossible to live without. Frank Serafine was one of its first practitioners, on that first Star Trek film in 1979, and he's gone on to do sound design for films including Hunt For Red October, Virtuosity, and the cinetech classic Tron. Serafine's Venice Beach studio is about as good as it gets, though he recalls that he started out with a four-track and a Stevens console a quarter of a century ago. Friends working on the Star Trek film gave the young composer a shot at pitching music for the film. Veteran Jerry Goldsmith got the gig instead, but it brought Serafine to the attention of the producers, who gave him some space on the soundtrack to create sonic ambience with his array of synths and sequencers. He got his first screen credit as a sound designer on Tron and the niche took off after that, becoming a staple of cinema sound. Now, sound design is moving into other areas of the media, including television, commercials, the Internet, computer games and documentary films. Serafine agrees that it's another growth area for audio, but one that will require expertise beyond sound. "Media are now migrating to the laptop, and in the future more of the sound design and editing is going to be done Photo: Sara Beugen by the picture editor," Serafine says. "So anyone getting into this field would Triple Z Music provided audio for this Xerox corporate presentation. do well to learn the platforms that sound and picture are converging upon, like Final Cut Pro. When you look at the progression of the last 20 years, you see that technology has cut the sound complement on a film's post from a large crew to just a few people. No reason to think that trend won't continue." In terms of the technology of sound design, Serafine says he is surrounded today by scores of vintage analogue and digital synths that have all been rendered obsolete by the Emagic EXS24 software sampler he uses on his Logic Audio system, which he runs in conjunction with Pro Tools hardware. "It emulates all the synths I used to use," he says — including the one he employed to create the 'transporter' sound for Star Trek, which in turn echoed the sound effect from the original television series, which was produced by a Farfisa organ. "Sound design today means staying at the edge of the technology," he says. "That's what will make it easier to interface with all the other media that are now file:///H|/SOS%2004-07/Alternative%20Careers%20for%20Sound%20Engineers.htm (10 of 11)9/22/2005 8:59:15 PM


using sound design." Reality TV is another budding sound design field, and one which also reflects the realities of current production budgets. "Working from a laptop fits in with shows like Cops," he says, referring to the grandfather of all reality shows, shot with a handheld camera with an attached condenser mic and the occasional wireless and which, after over a decade on the air, generates far more revenue than it costs to produce. But Serafine also points out that sound design has other possibilities for revenue streams. One of the biggest is the reuse of sound elements that comprise sound design. Serafine has several volumes of library sounds on the market which he says act as a residual source of revenue, an annuity in a field that, like most of audio, is work for hire.

Frank Serafine in his studio. These days, the vintage keyboards he used to design sounds for classic films have largely been superseded by software.

As we can see, there are a lot of possibilities out there that go well beyond the walls of recording studios. Identifying them, learning about them and seeing how they fit into you future plans, is the key to a successful career in audio, one that will go long-term. Published in SOS July 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



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