Sound on Sound (October 2004)

In This Issue October 2004 In This Issue Click article title to open Reviews People ART DPS II All Things Must Pass ...

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In This Issue

October 2004 In This Issue Click article title to open Reviews

People

ART DPS II

All Things Must Pass

Valve Microphone Preamp This dual-channel unit has hybrid solid-state/tube circuitry, variable input impedance, instrument-specific presets, and a flexible digital output.

Leader Editor In Chief Paul White bemoans the ephemeral nature of modern software instruments and wonders what will be the Stradivarius of our time in 300 years...

Electro-Harmonix NY2A

Business End

Valve Optical Compressor This high-end stereo compressor boasts three different flavours of optical compression.

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

Garritan Personal Orchestra

Crosstalk Orchestral Sample Library (Mac & PC) Reader Feedback Gary Garritan takes on the orchestral behemoths with a small but perfectly formed library packed with extra features. Replies to more of your emails, letters and faxes. Groove Tubes MD1b FET

Readerzone: Synthfest!

Condenser Microphone This new mic is a reissue of Groove Tubes' first largediaphragm condenser, but with solid-state electronics replacing the valve to offer a more affordable package.

Philip Taysom Walk inside Philip Taysom's studio and you will find classic synths in abundance, all connected and all ready to play...

K+H O300D & Pro C28 Active Monitors & Digital Controller Despite its small size, this new system leads the field when it comes to nearfield monitoring accuracy.

Latest Sample CDs Sample Shop Reviews/appraisals of the latest sample CDs: Luscious Grooves ***** MULTI-FORMAT; Electro Magnetic Pulse **** AUDIO+WAV+REX; SAM Solo Sessions **** MULTI-FORMAT; Saophones 1 ***** GIGASTUDIO/EXS24 MKII.

M Audio Firewire 1814 Firewire Interface (PC & Mac) M Audio's half-rack interface provides a comprehensive selection of analogue and digital I/O at a competitive price.

Sounding Off: Your CV Paul Bower Paul Bower explains why the CV may be more important than you think — even in the music industry.

Studio SOS Dorian Kelly The SOS team ride to the rescue of a budding media composer who is having trouble with his mixes.

The Engineers Who Changed Recording Fathers Of Invention Some of the legendary names in engineering and production didn't just make great records — they also invented equipment and techniques we take for granted today.

The Prodigy

Liam Howlett: Recording Always Outnumbered, Never Outgunned USB MIDI Controller Keyboard Nowadays, plenty of hit albums are recorded in bedroom The price of compact MIDI controller keyboards has come studios — but Liam Howlett of the Prodigy has gone one tumbling down in the past few years, but weighted-action full- better, by recording his latest in bed. length keyboards have remained relatively expensive —until now. We investigate the 88-note keyboard that runs off USB The Resurrection Of Brian Wilson's SMiLE

M Audio Keystation Pro 88

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In This Issue

Mark Linett & Darian Sahanaja The Beach Boys' SMiLE album was to have been the Mackie Onyx 1620 pinnacle of Brian Wilson's groundbreaking recording and Mixer production achievements — but it was never completed. Mackie combine a completely redesigned 16:2 mixer with a This year, in an extraordinary tale of emotional drama, tapemulti-channel Firewire audio interface. What more could you vault archaeology, and recording technology, Brian finished want? it, debuted it live, and then re-recorded it in the studio. SOS brings you the full story... power...

MakeMusic! Finale 2004

Technique

Music Notation Software (PC/Mac) The latest incarnation of a well-established scoring package holds few surprises, but plenty of power for the asking price.

Alesis Ion User Tips

Prosoniq Rayverb

Surround

Inverse Raytracing Convolution Reverb Plug-In (PC & Mac) There are now plenty of convolution reverbs around, but Prosoniq\'s Rayverb offers something a little different. So just what is 'inverse raytracing' anyway?

New Surroundings Binaural recordings can sound amazing on headphones, but do not work very well on conventional stereo speakers. Can they be adapted more successfully for surround systems?

Studio Electronics Omega 8

CLASSIC TRACKS: Heroes

8-voice Digitally Controlled Analogue Synth In this ever-changing world of virtual analogues, modelled instruments, and software emulations, it's a pleasant surprise to find that some companies are making synths the old-fashioned way. But is nostalgia really what it used to be?

Artist: David Bowie; Producers: David Bowie, Tony Visconti; Studio: Hansa Ton, Berlin With 'Heroes', David Bowie pulled off the rare \nfeat of having a major hit with a highly experimental piece of artrock, which featured among other highlights live synth treatments from Brian Eno, pitched feedback from guitarist Robert Fripp, and a lead vocal with level-triggered ambience.

Programming Guide Although the Ion is already well-respected for its sound Presonus Central Station quality, its synthesis architecture is also surprisingly Monitor Controller flexible. We show you how to get the best from this little This high-quality passive monitor controller can cater for five beast in your studio. stereo input sources and three pairs of speakers, with builtin cue monitoring and talkback functions. Bilocation: Binaural Recording & 5.1

Virsyn Cantor Singing Synthesis Software ( XP/ Mac OS X) If your computer could sing, what would it sound like? With the Cantor software synth from Virsyn, you can find out...

Wharfedale Pro Diamond 8.1 Pro Active Active Monitors These new active nearfields from Wharfedale offer a polished sound for studio owners on a budget. Competition

WIN Yamaha 01X Digital Mixer/Fader Control Surface/24-bit Interface Sound Advice

Q. Can dust affect microphone performance? Q. How do I compress a stereo source? Q. How do you create a sense of height in a stereo soundscape?

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

First Look: Cubase SX3 Cubase Notes This month we provide an exclusive preview of \nthe new version of Cubase — SX 3 — and offer advice on how to get the most out of Cubase's Range Selection tool.

How To Remote Control Sonar Effects Sonar Notes This month, how to use the handy Scrub tool, how to customise your track view, and an explanation of the procedure for setting up remote control of Sonar effects parameters.

Incurable Soundcard Stuttering?


In This Issue

Q. Is USB too slow for MIDI interfacing? Q. What are the correct input impedances for Guitars and Mics? Q. What do the different 'colours' of noise do? Q. What kind of stands are best for mounting monitors? Q. What mics should I use on a snare drum? Q. Why does it take so long to freeze a VST Instrument track?

PC Notes We investigate an obscure PC parameter that may be responsible for some otherwise 'incurable' soundcard stuttering problems.

Linux Jack Sound Server Ported to OS X Apple Notes The Jack low-latency sound server has been ported to OS X from Linux. We discuss its background and implications.

Logic Notes More hints and tips Learn how to combine dynamics processing with delay and reverb to breathe life into your mixes.

MIDI Sequencing In Pro Tools Pro Tools Notes While Pro Tools is widely known for its dominance in the field of computer-based audio recording, little is made of the fact that it also features fully fledged MIDI sequencing facilities.

PC Music Shareware Roundup PC Musician Shareware might be cheap, but it can also make you very cheerful. We round up some of the best of the current PC music shareware scene.

Running Autotune within Digital Performer Digital Performer Notes A bundle of tips helps you get the most from Autotune running inside DP, and there is plug-in news aplenty...

Using Mackie Tracktion Sequencer As A Rewire Host Reason Notes Mackie's Tracktion sequencer makes an ideal budget Rewire host for adding audio recording and plug-in capabilities to your Reason setup, as we explain...


ART DPS II

In this article:

Phantom Foibles Digital Facilities In The Studio But Is It ART?

ART DPS II £305

ART DPS II Valve Microphone Preamp Published in SOS October 2004 Print article : Close window

Reviews : Preamp

pros Affordable. Variable input impedance. Variable tube drive. Interesting range of vocal and instrument presets. Digital output as standard.

This dual-channel unit has hybrid solid-state/tube circuitry, variable input impedance, instrumentspecific presets, and a flexible digital output.

cons Phantom power is also applied to the rear-panel balanced jack input! No low-cut switch. Panel ergonomics could be better.

Paul White

ART's DPS II mic preamp has much in common with the ART Digital MPA that I reviewed recently insomuch as it is a dual-channel tube mic preamp with variable impedance, adjustable tube 'flavouring', and an in-built analogue-to-digital converter, but it is different in several key areas. Immediately noticeable is its 1U summary format, but its 96kHz upper sample-rate limit is also lower than that of its big A good-sounding and creative brother — the Digital MPA goes up to 192kHz — and its tube stage has been mic preamp that offers a digital output as standard and configured to offer presets for various types of sound enhancement using a at a very keen UK price. system ART refer to as V3 or Variable Valve Voicing. Amazing, isn't it, that the Watch out for that phantom only time an American company uses the British word for valve is when it powered rear-panel jack input happens to conveniently start with the right letter to make up a catchy trademark? though! information £305.49 including VAT. Sonic8 +44 (0)8701 657456. +44 (0)8701 657458. Click here to email www.sonic8.com

Photos: Mark Ewing

www.artproaudio.com

Housed in a striking, dark-blue case with a 'V'-motif grille in the centre, the DPS II comprises two identical channels of microphone preamplification, each of which use hybrid circuitry (a solid-state, discrete preamp followed by an additional 12AX7 tube gain stage) and which can accept mic, line, or high-impedance instrument-level signals. Mic input impedance adjustment is provided via a control covering a continuous range from 150(omega) to 3k(omega), and a rotary switch selects from 15 flavours of tube coloration and a neutral mode. In theory, variable input impedance means that the preamp's input stage can be more file:///H|/SOS%2004-10/ART%20DPS%20II.htm (1 of 5)9/26/2005 12:35:33 AM

ART DPS II

exactly matched to different types of microphone, but in practice most people treat it as a tone control and twiddle it until they get the best subjective result. Each channel has a phase button and individually switchable phantom power. While the Digital MPA had a semi-retro look with large mechanical VU meters, the DPS II has a somewhat more 'techy' look to it, though it still sports mechanical tube gain meters, albeit of an 'edge view' low-profile type. Above the mechanical meters are LED-ladder peak meters to monitor the input levels before the tube stage. The same type of gain structure has been employed as in the Digital MPA, where the solid-state preamp operates in tandem with a 20dB gain button to provide gain ranges of up to 48dB or up to 68dB, and the tube drive effects are created when the 20dB button is in, as this drives the tube harder. The output fader knob has another 10dB of gain in hand making the maximum available gain 78dB. Two further Digital Level gain controls at the extreme right of the front panel adjust the level fed into the converters for each channel, and there are two warning LEDs that come on if digital clipping occurs. Each channel has a Neutrik combi jack/XLR connector on the front panel that can accept either a mic-level XLR or an instrument/line-level signal via its centre jack section. Plugging in a jack overrides the XLR. The jack is high-impedance and unbalanced, so it can handle guitars and basses as well as line-level signals. Although phase switching is provided, there's no low-cut filter; given the number of project-studio mics now coming onto the market with no switchable filters, this could be a limitation.

Phantom Foibles Phantom power can be switched to the mic XLRs, but the rear panel also offers balanced inputs, wired in parallel on both jacks and XLRs, which are designed to handle anything from mic level to modest line levels. This might seem fair enough, but my tests showed that when the phantom power was switched on, it went to both front- and rear-panel XLRs, as well as to the balanced input jack on the rear panel, which could cause serious problems if a piece of sensitive linelevel equipment were to be plugged in when the phantom power was on. What's more, the manual doesn't seem to include any obvious warnings about this, though it does spell out the usual caveats on using dynamic mics with phantom power. The front-panel unbalanced jack is of course unaffected. The outputs are available on both balanced XLRs and unbalanced jacks (+28dBu and +22dBu maximum output levels respectively). Rear-panel TRS insert jacks allow other processors to be inserted into the signal path before the converters, and these may also be used as direct inputs if you simply want to feed another piece of analogue gear through to the converter.

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ART DPS II

The quoted Equivalent Input Noise (EIN) is -129dBA, which is on the good side of typical, and the frequency response of the analogue circuitry is 5Hz-50kHz, ±0.5dB. Of course the frequency response at the digital output is determined by the choice of sample rate. Whereas the Digital MPA had switchable high and low tube-plate voltages, the DPS II uses a 16-way rotary switch to offer presets based on their V3 concept that have been created for use with voices, various instruments (both electronic and acoustic), and percussion. It's not entirely clear how these settings differ, or even if there's anything else going on other than variations in the degree of tube drive and plate voltage, but the manual does explain that the second eight presets include OPL, which is an in-built Output Protection Limiter. As with all presets, you should experiment to see if some of the settings work well in situations that their name doesn't necessarily imply. In addition to processing individual sounds, it is also interesting to try the unit on stereo mixes, where experimenting with the presets and drive levels reveals different ways of adding warmth and density to a mix.

Digital Facilities Digital outputs are becoming more common as the importance of hardware mixers in studios continues to decline and more users choose to send a digital signal from a mic preamp directly to a soundcard or audio-interface digital input. Here the implementation of the digital circuitry is simpler than on the Digital MPA, with only S/PDIF or ADAT output, ADAT input, and a word-clock input BNC connector. The optical output either carries a traditional stereo S/PDIF signal or ADATformat signals, while a series of LEDs plus a button on the front panel allow the user to step around the four ADAT output pair options, all ADAT channels, or S/ PDIF. The coaxial S/PDIF output is always active. When ADAT mode is selected, anything coming in on the six unused ADAT channels is passed straight through to the ADAT output, while the two mic-preamp signals are sent out on ADAT channels one and two. However, if the all-ADAT option is selected (where all four ADAT LEDs are lit), all four ADAT channel pairs carry the same signal. Similarly, if the optical output lights are all off, any incoming ADAT signal is passed through unchanged on all eight channels. The sample rate is selected by means of a switch and a row of LEDs, with a choice of 44.1kHz, 48kHz, 88.2kHz, 96kHz, ADAT, and External. ADAT will of course only work at 44.1kHz or 48kHz. The output is at 24-bit resolution, lacking the Digital MPA's ability to dither down to 16 bits, though this shouldn't be a problem, as most people concerned about such things probably record at 24-bit file:///H|/SOS%2004-10/ART%20DPS%20II.htm (3 of 5)9/26/2005 12:35:33 AM

ART DPS II

resolution anyway.

In The Studio My main testing was done using both tube and solid-state capacitor mics, and the different tonalities that could be achieved using the various tube voicings were rather interesting. With some of the settings designed for use with instruments, background noise was audible when using a mic with low sound levels and high amounts of gain, but I imagine this would be insignificant In addition to the analogue Clip indicators which form part of the input stage's LED if the designated instruments were peak meters, the Digital Clip lights alert you being used as sources, as the signal to overloads in the onboard converter. should be much stronger. However, it does mean you have to take care when using these voicings as alternative microphone treatments. The same is also true of the acoustic guitar setting if you're miking the guitar rather than DI'ing it via a pickup. Nevertheless, the neutral V3 position is more than adequately quiet to use with a microphone, as are over half of the presets, and the tonal differences introduced by the various presets range from quite subtle to having a noticeable edge, but none of them sound unnatural unless the amount of drive used is excessive — then the sound can get a bit ragged. It's useful having a peak-limiter option on half the presets, as it can help avoid digital clipping and may also be used sparingly as a gain-reduction effect. Changing the input impedance makes a subtle difference to the sound, where the higher impedance tends to equate to a little more level and body in the sound, though how much change you get depends very much on the type of microphone you're using. The middle position is pretty safe for any mic. The converters seem to be comparable in quality to those used in the Digital MPA and I had no cause for complaint in this area. One ergonomic factor that I found a little disconcerting was the close spacing of the three knobs, which not only makes it difficult to adjust them, but also makes it almost impossible to see what's printed under and around them. It doesn't help that the two channels are set out as mirror images of each other, so what is the preamp gain control on one channel can easily be mistaken for the output fader on the other.

But Is It ART? The DPS II has at its heart a good-sounding mic preamp and surprisingly flexible file:///H|/SOS%2004-10/ART%20DPS%20II.htm (4 of 5)9/26/2005 12:35:33 AM

ART DPS II

digital interfacing as standard. The routing of phantom power to the rear-panel jack is rather silly, though, as is the unnecessarily tight knob spacing, while the omission of a high-pass filter will be seen as a cut too far by some. Having said all that, the preset tube flavours are capable of some musically useful variations on the basic sound, and I only wish the manual had gone into some detail as to what each preset actually did. Variable input impedance is a feature only usually found on more esoteric mic preamps, but it offers a little further scope for tonal change. Other than some of the design quirks already mentioned, the DPS II is actually a good little performer and capable of producing great results. Published in SOS October 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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In this article:

Overview The Signal Path Using The NY2A

Electro-Harmonix NY2A $3519

Electro-Harmonix NY2A Valve Optical Compressor Published in SOS October 2004 Print article : Close window

Reviews : Processor

pros Three distinctive sound characters. Imposing retro styling. High-quality components throughout.

This high-end stereo compressor boasts three different flavours of optical compression.

cons Significant acoustic mains transformer hum. Inconsistent thresholds between modes. Requires unusually high signal levels.

Hugh Robjohns

There are two fundamental signalprocessing tools that are almost impossible to do without in any form of audio recording or post-production: summary equalisation and compression. Both A rather quirky dual-channel are available in a wide variety of Photos: Mark Ewing valve compressor employing three different opto elements different formats, flavours, designs, to provide distinctly different and degrees of quirkiness. The device reviewed here, the Electro-Harmonix dynamic characteristics. NY2A, will score fairly highly on most people's 'quirkiness' scale: there aren't that Operation is simple if many stereo compressors that require a massive 3U-high and 14-inch deep unconventional, and the unit is built to very high standards. rackmounting cabinet, for a start! information Mike Matthews, the man behind Electro-Harmonix, declares in his introduction to $3519.13 (around the handbook for the unit that the NY2A is 'the best compressor ever made'. I £1961) including VAT. Electro-Harmonix +1 718 dare say there would be more than a few people wanting to challenge that claim, 937 8300. but how exactly do you define 'the best'? If we are talking technical +1 718 937 9111. specifications, the NY2A doesn't look so hot. The distortion figures suggest 0.37 Click here to email percent is as good as it gets, and the signal-to-noise ratio is a very modest 65dB www.ehx.com with 0dBu signals in and out. Of course, if you drive it harder, the noise performance improves in proportion, and a little 'nice' distortion is a thing of sonic beauty for most people. So maybe technical specifications aren't really what this unit is all about.

Overview The NY2A is an unusual stereo compressor in almost every sense. I have

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already mentioned its gargantuan size (and it weighs a mighty 26lbs as well!), but the controls are just as unusual. Generic compressor controls like Ratio, Threshold, Attack, and Release are eschewed in favour of Pre-gain, Compress, Post-gain, and Squash! But perhaps I'm getting a little ahead of myself here... Let's start at the beginning. The New Sensor Corporation — the parent company for Electro-Harmonix — also own and import the Electro-Harmonix, Sovtek, and Svetlana brands of Russian-made valves. Consequently, the NY2A is not only an optical compressor, but one which uses exclusively valve-based amplification. However, it doesn't have just one optical compression element per channel — that would be far too conventional — the NY2A has a trio of different opto couplers, each using a different light source to provide three distinct dynamic responses which the user can select to suit the source and musical style, as well as the desired effect. The first opto device uses an LED as the light source to provide a fast attack time (roughly 10ms — which is slow in comparison to some solid-state compressors), and a uniform response across the entire audio frequency range. The second uses an incandescent lamp, which naturally has a much slower attack time of about 50ms and a rather less even response to all signal frequencies. The thermal inertia of the lamp filament means that it tends not to respond to highfrequency level changes, with the result that it tends to sound brighter than the uniform LED arrangement. The third option uses an electro-luminescent panel as the light source, and this has a very nonlinear behaviour. A particular characteristic of this kind of device is that it changes colour with input frequency: it starts off a yellowish colour at 30Hz, and progresses through green, blue, and on to purple by 20kHz. The light output also varies considerably with frequency, giving much less light with low frequencies than with high. The result is a compressor that compresses high frequencies a lot more than low frequencies, and is thus a kind of natural deesser. The valve amplifiers in the NY2A are anything but 'standard' either. The circuitry does not employ negative feedback to linearise the signal path as most amplifier circuits do. The result is a higher distortion figure on the test bench, but a much better transient response and, many would claim, a better sound character.

The Signal Path Both channels in the NY2A are identical, and there is a linking switch to tie the side-chains together for stereo operation when required. The balanced input is interfaced with a Lundahl transformer, providing galvanic isolation from the outside world, and providing an unbalanced signal for the internal circuitry.



The first amplifier stage employs an EH1 dual-triode valve and provides a fixed 20dB of gain. The signal level feeding the rest of the signal path is controlled at this point with the Pre-gain control, and there is sufficient level available to force the unit into limiting or peak distortion. In effect, this control sets the compression threshold as well as the input signal level, but the threshold is also affected by the Compress control. This determines the proportion of input signal (taken after the pre-gain control) that feeds through to the side-chain to drive the illumination source, and thus sets the amount of compression. Following the opto sensor, the signal is buffered by an EH2 triode valve stage before reaching the Post-gain control which sets the output level from the device. This is followed by another fixed-gain valve amplifier, using an EH3 dual triode to provide 32dB of gain, and the Lundahl output transformer. The front-panel controls described above are accompanied by a few additional switches and features. The main controls are arranged in a row along the bottom half of the front panel with four black retro-style detented knobs. The first three adjust the Pre-gain, Compress and Post-gain parameters, while the fourth selects one of the three opto compressor light sources. Each channel also has three silver toggle switches with associated LEDs. The first operates a relay bypass to connect the input directly to the output, the mode being indicated with a green LED when the toggle is flipped down. To the left of the light source switch is another toggle switch marked Squash, and to the right a third labelled Attack — both have red LEDs to indicate when they are active. The former activates a gentle highfrequency shelf when using the electro-luminescent light source to further emphasise the de-essing characteristic of this mode, while the latter doubles the attack time for the incandescent lamp mode (raising it from 50ms to 100ms). These two switches only work when the corresponding light source has been selected, but rather perversely they operate when flipped up — the opposite direction to the Bypass switch. In fact, the panel labelling suggests this is the case (the 'Bypass' legend is written below the switch, whereas 'Squash' and 'Attack' are written above the switches), but it is confusing ergonomics nonetheless. Above these operational controls are the metering sections for each channel, delineated by a darker panel background. To the left of a large white VU meter is another silver toggle switch which selects either output-level or gain-reduction metering, and there is also a recessed screwdriver trimmer to calibrate the meter to read zero when there is no gain reduction being applied. In addition to the VU meter, there is also a 'magic eye' display valve (EH80) for each channel which always shows the output level. This responds much faster than the VU can, and so is a better guide to signal peaks and overloads. The handbook claims that the 'eye' is calibrated to 'close' at 100dBu, but this appears to be one of a rather



large number of unfortunate typographical errors found throughout the handbook. In fact, the NY2A is set up so that the 'eye' closes at 0dBu, and 0VU on the meter also appears to equate to 0dBu at the output. Between the two channel metering sections lies the toggle switch for stereo linking (up for on, with a green LED), and to the right of channel two's metering section is a power switch toggle (also up for on, but with a blue LED). Moving briefly to the rear panel, the audio inputs and outputs are all via XLR, and the two inputs each have an associated toggle switch to operate at a nominal 0dBu or +20dBu. There is also a ground-lift switch, a fuse holder, and the standard IEC mains inlet. It's worth mentioning that the unit's front and top plates become almost too hot to touch after a couple of hours of use, so the provision of adequate ventilation will be vital when rackmounting.

Using The NY2A Whichever way you look at it, this is a very quirky product, which means you'll either love it or hate it — there isn't much room for a middle-ground opinion! Initially, I was probably leaning towards the 'hate it' camp. The sensible engineer's side of my brain had me screaming in frustration at some of the quirks. The main problem was that the thresholds for the three opto circuits were radically different, which made it completely impossible to compare sonic characteristics simply by switching light source. Electro-Harmonix argue that the threshold variations have been chosen to optimise and enhance the characteristics of each opto circuit, but I remain unconvinced. For example, if I adjusted the Pre-gain and Compress controls to give a measured 8dB of gain reduction on a static tone signal with the electroluminescent setting, (although I noted the VU meter showed only 6dB), switching to the LED mode produced around 10dB of gain reduction (and the VU meter showed 20dB+). Switching to the incandescent bulb, the gain reduction fell to just 3dB (and the VU meter agreed with that one!). So in practice you have to juggle the Pre-gain, Compress, and Light Source controls in order to find a setting that suits the material being compressed — and that becomes a real handful if you are working with stereo material! Another issue which I struggled with was the operating levels. While the VU is calibrated to suggest a nominal working level of 0dBu, the device really needs to be driven a lot harder than that to maximise the signal-to-noise ratio. It's not such a problem if you are working with digital recorders and converters expecting peaks of +20dBu, but most budget analogue mixers will be struggling long before such levels are reached. The last of my major complaints was the acoustic hum coming from the mains transformer. Measured with a noise meter about three feet away, the acoustic hum was dominated by a 200Hz component which was over 20dB higher than the background noise when the unit was switched off! I wouldn't want this box humming away in my rack, that's for sure.



However, if you can accept all these quirky features, I have to admit that the NY2A can sound really pretty good. It's not the kind of ultra-transparent and subtle compressor you would want to use for mastering — it's more of a 'down and dirty' effects compressor, designed to help bring out the attitude of a sound source, or to create a dynamic edge and tension. The very lazy attack of the incandescent setting Here you can see the high-quality discrete electronics at the heart of the NY2A, suited percussion well, adding serious including three valve gain stages per character to a dull drum kit, especially channel. The optical gain-control elements the kick drum. It made a nice job of occupy the top edge of the main circuit board. vocals too. The electro-luminescent mode seemed better suited to more complex material, especially electric guitars and heavy synth parts — it made a very good job of controlling some exaggerated synthesised strings, for example. The LED mode was almost on a par with a solid-state compressor — fast, aggressive, and tightly controlling — much more like a peak limiter than a linear compressor, but still very useful in the appropriate circumstances. By juggling the gain controls it is easy to dial in a degree of gentle valve saturation which tends to thicken the sound, the harmonic distortion filling out the mid-range in particular. At the UK list price, the NY2A has a lot of very strong competition. Within £500 either way, there are the Amek 9098CL, Avalon AD2044, Chandler TG1, Crane Song STC8, Dbx 160SL, Focusrite Red 3, Manley Variable Mu, Neve 33609J, Prism Maselec MLA2, Summit Audio DCL200, and Thermionic Culture Phoenix. None offer quite the same feature set as the NY2A, but they are all extremely fine and well-respected dynamics units, some with their own distinct quirkiness too! The NY2A is a unique and colourful product that is capable of some interesting and characterful results. For me, the frustrations probably outweigh the benefits, but I'm sure many will find this a very powerful and flexible tool with which to shape their sound. It is well built, with very high-grade components, is easy to set up and adjust, and certainly looks very impressive. Published in SOS October 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:


Bundle Of Joy Orchestral Sample Library (Mac & PC) Ensemble Builder The Sound Library — String Published in SOS October 2004 Sections Print article : Close window Solo Strings Reviews : Sound/Song Library Woodwinds Brass System Requirements & Performance Keyboards & Harps Gary Garritan takes on the orchestral behemoths with Percussion a small but perfectly formed library packed with extra Personal Expression features. Compromise & Control Jamboree Bag Garritan Personal Orchestra Dave Stewart & Mark Wherry Instrument List Summary The age of the sampled orchestral super-library

is upon us, as grandly named titles like Vienna Symphonic Library's Pro Edition and East West/ Quantum Leap's Symphonic Orchestra testify. pros While these collections are bigger and better A thoroughly comprehensive than their predecessors, their huge sizes (235GB orchestral library which omits and 67GB respectively) demand more computer no instruments of note, and resources, more disk drives, more MIDI ports, throws in some rare ones. and more hunting through lists of performances Some sounds are superb. to find the right sound. For one man, this has all The library is bundled with many useful extras, and offers gone too far: US sample supremo Gary Garritan, good integration with notation the brain behind GigaHarp and Garritan programs. Orchestral Strings, adopts a 'small is beautiful' philosophy with his latest title. By squeezing the As canaries say, cheap! entire orchestra into a 2GB package and giving cons users the tools to play it effectively, Garritan Personal Orchestra (or GPO) aims The lack of multi-dynamics to combat 'sample bloat' and restore 'simplicity, sanity, and affordability'. The last renders some instruments a claim at least is beyond dispute — the library retails in the UK for only £179. little unsubtle. Garritan Personal Orchestra £179

Operating system restrictions may be a problem. The need to keep the library small, and yet still comprehensive, means that some of the samples don't sound quite natural when layered polyphonically.

summary GPO is a well thought-out package which can be used

Ironically, it was Mr Garritan himself who started the trend for large orchestral libraries — his Orchestral Strings (released in August 2001) contained 8GB of samples, a staggeringly large figure at the time. However, the US maestro still remembers his days as a struggling, impecunious musician (is there any other kind?) and has vowed to make a product within the reach of everybody, including music students and educators. As budding composers have traditionally found it almost impossible to hear their orchestrations played, this is an important development. Now, in theory, arrangements can be tested on a system that doesn't cost the earth, helping the new generation of would-be Gustav Mahlers to

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for sketching, mastering or enhancing full orchestral arrangements. Though the library doesn't attempt to replicate every performance style under the sun, its samples are well chosen, versatile and expressive. At £179, it's an absolute snip — buy it now, before someone puts the price up!

information £179 including VAT. Time & Space Distribution +44 (0)1837 55200. +44 (0)1837 840080. Click here to email www.timespace.com www.personal orchestra.com

Test Spec IBM Thinkpad T40 with 1.3GHz Pentium-M processor and 768MB of RAM running Windows XP Professional.

avoid elementary musical blunders.

Bundle Of Joy Unlike Garritan's previous libraries, which supplied sample content for a sampler you would already own, like Gigastudio or EXS24, GPO has its own sample player based on Native Instruments' Kontakt Player. This is a trend that most sample developers are following these days — East West/Quantum Leap's Symphonic Orchestra uses Native Instruments' Kompakt Player, for example — as it provides a better way of copy-protecting the sample content, tying it to a specific software application that can be registered and authorised. GPO uses the normal Native Instruments challenge-and-responsestyle registration tool, although this isn't too intrusive. For those users who worry about having to reinstall their systems at short notice, Garritan have provided a 'more generous than most' The stand-alone version of Kontakt Player 30-day grace period for registration. supplied with GPO. Aside from the copy protection, though, there's also an advantage to users in supplying a sample library linked to a player in this way, since any specific requirements of the library can be accommodated by the software, and the whole package is generally more integrated. Integration is actually quite an apt word to describe GPO, since Garritan have gone one step further than to merely provide a sample player. The GPO package also includes a MIDI + Audio sequencer, score-writing software, a reverb plug-in, and a VST plug-in host application called GPO Studio, so that as long as you have a computer with suitable MIDI or audio hardware, GPO offers everything you need to get started with computer-based orchestration straight out of the box. These 'extras' will be welcomed by composers who are just getting started in computer-based orchestration, and will surely be appreciated by schools and colleges who have limited budgets to deliver an appropriate music-technology curriculum. The Kontakt Player supplied with GPO (above) runs as either a stand-alone application or as a plug-in supporting most major formats (see the 'System Requirements' box later in this article). Running the stand-alone version will enable you to quickly assess GPO's sonic potential, and each instance of the player has eight sound slots — enough to accommodate a small instrumental group. However, since you can only run one instance of the stand-alone Kontakt Player at a time, the only way to use more than eight instruments simultaneously is to run multiple instances of the Kontakt Player plug-in, which is easily achieved with almost any modern MIDI + Audio sequencer (including those bundled with GPO), or the GPO Studio application. You can read about these applications and



their uses in more detail later, but without further ado, let's get to the heart of the matter: the sounds!

Ensemble Builder Orchestral libraries usually provide both solo instruments and ensembles — you can choose between (say) a solo violin and a 12-violin section, or a solo flute and three flutes playing in unison. But what if a particular passage requires two unison flutes, or three violins playing the same part? Using multiple identical versions of a solo instrument won't work, as unison notes played on the same samples sound totally unrealistic. With solving this in mind, Gary Garritan devised a concept called 'Ensemble Builder' which enables users to create instrument sections to their own specifications. If you want to hear two flutes playing in unison, simply load 'Flute Ens1' and 'Flute Ens2' — these two alternative solo flutes use completely different samples, so can be layered with impunity. If you need the sound of three flutes, just add 'Flute Ens3' to the mix. The three makes of solo violin each have their own set of three alternative samples, so you could conceivably construct a virtual ensemble of nine violins. The only limitation to this uniquely flexible system is that there's no mileage in layering a main solo instrument with its associated 'Ens' versions, as the 'Ens' programs are derived from the main instrument's samples!

The Sound Library — String Sections Thankful not to be staring into the bottomless depths of a 100GB library, I dived into the sample pool in search of pearls, starting with the strings. GPO's string sections (12 first violins, 10 second violins, 10 violas, eight cellos and seven double basses) are culled from the vast collection of recordings made for Garritan Orchestral Strings. Unlike GOS, there are no combined 'all violins' performances, but solo strings are included. The 12 first violins have a strong, bright sound, and perform their 'lush' long notes with fairly heavy vibrato and a hint of a volume swell, bringing to mind the undulating violins in old Hollywood film soundtracks. Muted 'lush' versions maintain the vibrato style, but produce a more subdued timbre. There are no staccatos — 'short bow' samples (lasting about a second) come closest, and 'sus+short' programs combine these with long notes in a good, all-purpose patch. The second violins have a slightly sweeter tone, and though their vibrato is equally strong, their delivery sounds


Three instances of GPO Player and the Ambience reverb plug-in running with the bundled Cubase LE, showing the Joplin GPO Demo Song.


more serene. Both violin sections play well-executed tremolos, tone and semitone trills and some great pizzicatos. These violins work well for expressive melody lines, and the 'lush mutes' are suitable for chord pads. GPO's violas sound broad, expansive and fruity — their extreme top notes have sensibly been pruned, leaving space for the violins to take over in the high register. The cellos and basses are very good indeed, the former producing a fine, dignified emotional singing tone. The lower strings duplicate the violins' performance styles, with the exception that the basses don't play trills. While these performances cover all the basic requirements for orchestral strings, it would have been nice if some fierce, emphatic marcatos had been included. There is no 'full strings' preset, so I made one myself. Working out the note ranges took very little time, and I was able to quickly construct an eight-way Kontakt 'multi' with basses, cellos, violas and second violins sustains in slots one to four respectively (all receiving on MIDI channel one). After tweaking a couple of volume settings, I had a very classy-sounding, versatile and highly playable full orchestral string section.

Solo Strings The library boasts three different makes of solo violin: the first is a Gagliano, the second a Stradivari and the third a Guarneri. These instruments cost more than your house — describing their subtly different tonal characteristics is beyond me, but suffice it to say they all sound superior to any violin you can buy at Argos. Again, strong vibrato is the order of the day; the two G-guys ease into theirs, but the Stradivari gives it full throttle right from the off. For less money than Charles Saatchi loses down the back of his sofa in one evening, you also get to play three priceless solo cellos, an expensive solo viola and an unaffordable solo double bass, all recorded fairly intimately with just the right amount of bow noise. There are only two playing styles (vibrato sustain and pizzicato), but an expressive extra touch is supplied by alternating up- and downbows (controlled by the sustain pedal). The solo strings' also offer 'ensemble', or 'Ens' versions of their sustains — see the 'Ensemble Builder' box below for details.

Woodwinds Despite its modest price, this library features some of the best sampled woodwinds I have heard. Pick of the bunch is the flute, a lovely, lyrical instrument whose slightly breathy tone is sweet, clear and piping in the higher register — an absolute knockout. The flute's 'no vibrato' option provides a slightly quicker attack. A lively piccolo and a sultry bass flute match the flute's high quality; by comparison, the alto flute sounds a bit disinterested, though its bottom notes file:///H|/SOS%2004-10/Garritan%20Personal%20Orchestra.htm (4 of 13)9/26/2005 12:35:39 AM


(arguably the most useful) are OK. I guess Gary Garritan likes oboes, as his library offers a choice of four different makes. All sound very nice indeed, but my favourite is the 'classical' model, whose evocative, slightly angular timbre sounds, well, classic. A plaintive oboe d'Amore (pitched slightly lower than a regular oboe) and two English horns contribute some beautifully played samples. As well as the ubiquitous B-flat clarinet, you get the brighter tones of an E-flat instrument, both sounding confident GPO Studio (see page 178) is a host that and assertive, if a little tonally oneenables you to run eight instances of GPO dimensional — the lack of soft, quiet Player and one instance of the Ambience samples means the timbre remains reverb plug-in, and also adds MIDI ports to your system to make it easy to use GPO with unnaturally bright throughout the score-writing software like Sibelius, Finale or dynamic range. A sinister, oily bass the included Overture SE application. clarinet and a truly evil (I mean that as a compliment) contrabass clarinet play exquisite low notes. Inhabiting the same subterranean territory, we find two bassoons and two contrabassoons, exhibiting different tonal qualities and pitch ranges, but sharing a high standard of musicality. The 'ensemble' programs supplied with the woodwinds work very well, but, apart from flute and piccolo flutter tongues, the woodwinds' performance menu is restricted to straight sustains. I was concerned this would limit their usability, but the samples are more versatile than they first appear — louder velocities produce sharper attacks, which work well for staccato notes. Other musical options are made possible by GPO's 'legato mode', as explained in the 'Personal Expression' box later in this article.

Brass GPO's principal solo trumpet is another winner. Strong and bright, it peals out its clarion notes with bell-like clarity, sounding like an instrument destined to grace hundreds of soundtracks. The second solo trumpet has a thinner, more cutting tone, and a piccolo trumpet (played with and without vibrato, but not at the same time) soars up into those rarefied pitch zones which other instruments can't reach. The addition of a mute to the principal trumpet gives it a distinctly jazzy flavour. When it comes to building a trumpet section sound, you're spoiled for choice — the principal trumpet has the usual ensemble-building elements, but there's also a terrific trumpet forte 'overlay' program which sounds like a real two-player file:///H|/SOS%2004-10/Garritan%20Personal%20Orchestra.htm (5 of 13)9/26/2005 12:35:39 AM


ensemble. This regal-sounding timbre can be mixed with the 'Ens' programs to further strengthen their sound. The only permutation that didn't work was an attempted layering of the two main solo trumpets, as their super-tight tuning produces phase cancellation! The trombones (one tenor, two makes of bass) are not half bad; in fact they measure around 91 percent on the SOS good-ometer. GPO's documentation advises using the pitch wheel to bend the notes — I followed this advice, and felt much the better for it! One of the bass trombones makes that extremely raspy, buzzsaw noise which is scarcely believable coming from a bass instrument. The trombone also gets its own powerful 'overlay' program and a mournful set of muted samples. In the tuba department, dynamic variation is achieved by one soloist playing mf and the other mp, the latter producing a nice round, warm timbre. The star turn here is the contrabass tuba player, who makes a fabulously rotund sound and distinguishes each of his notes with a very pleasing attack. Given the French horns' increasingly clichéd role in depicting cinematic scenes of drama, splendour and excitement, I imagine many film composers will instinctively reach for the horns' ff 'overlay' ensemble, which delivers the requisite blasting, brassy racket. There's also a less metallic-sounding, but still stirring forte overlay, and a set of muted samples. All the other horn programs sound fairly warm and mellow by comparison — by combining these options, users can produce a wide, dynamic range of horn timbres.

System Requirements & Performance In order to install GPO, you'll need to have approximately 2GB of free hard disk space, and to get the best out of the library, Garritan recommend using Windows XP with a 1.8GHz or better Pentium 4, Athlon, or Celeron processor, or Mac OS X with at least a 733MHz G4 — the software collection is generally incompatible on earlier operating systems, such as Windows 98 or Mac OS 9. As mentioned in the main part of the review, GPO Player is supplied in both stand-alone and plug-in versions, with support for VST, Audio Units and RTAS on the Mac, and VST and DirectX on Windows. In terms of performance, with Cubasis VST v4 on a 1.3GHz Pentium-M IBM Thinkpad, the Joplin GPO Demo Song used around 20 to 30 percent of the CPU, with three instances of GPO loading 10 instruments in total. Adding the Ambience reverb plug-in added another 20 percent of CPU usage (making 40 to 50 in total), although this could be reduced with the Quality/CPU slider. Interestingly, though, using Cubase LE to play back the same song required only 15 percent of the CPU without Ambience, and about 25 percent CPU with Ambience, proving that Steinberg definitely made some efficiency improvements to the VST engine between Cubase VST and Cubase SX-based versions. The current version of GPO Player doesn't support Native Instrument's DFD (Direct From Disk) extension, in contrast with most other Kontakt and Kompaktbased instruments, which allows samples to be streamed from disk rather than loaded in their entirety into memory. This should be available in an update at some point, but in the meantime Garritan recommend having at least 1GB of memory in the system you're using to run GPO. This is sensible, although GPO file:///H|/SOS%2004-10/Garritan%20Personal%20Orchestra.htm (6 of 13)9/26/2005 12:35:39 AM


did run quite successfully on the 768MB of memory installed in the test laptop. More RAM was needed to load larger templates, such as 'Full Orchestra + Piano', which requires 607.44MB memory in three instances of GPO. The lack of streaming isn't actually a big deal, since you'll actually get more voices by keeping all of the sample data stored in memory. This is especially useful for laptop users, as portables usually include slower drives as standard than the ones supplied with desktop machines, and accessing memory uses less power than accessing the hard drive constantly.

Keyboards & Harps You wouldn't normally expect to find a grand piano in an orchestral library, but GPO generously lays on a Steinway concert grand. This remarkably finesounding, 244MB piano has not been miked too closely, which allows its sound to breathe. The celeste was also recorded from some distance, a naturalsounding hall ambience giving a nice 'bloom' to its pristine chimes. Keyboardists will be delighted to find a harpsichord and a full-blown Baroque pipe organ (boasting 13 glorious stops) for those Hammer House Of Horror or church wedding (same thing?) moments. There are two harps, made by Venus (the goddess of love) and Wurlitzer (the god of, er, electric pianos). As one might expect from the man who brought us Gigaharp, the two instruments sound absolutely superb, and practically justify the price of the package alone. There are straight notes, and luscious harmonics. If you want to sequence a heavenly harp glissando, simply sweep your fingers up and down the white keys, then use one of Garritan's 144 harp 'data packets' (a small MIDI SysEx file) to automatically edit the pitches to the requisite scale and key.

Percussion The library's percussion covers all the orchestral bases without getting too exotic. Timpani, the traditional purveyors of orchestral grandeur, are mapped chromatically over nearly two octaves, offering left and right-hand straight hits (but no rolls). The timps are clean, resonant and tuneful, and pack a reasonable amount of punch. A decent set of tubular bells sound a little distant, which gives a nice perspective and adds to their mysterious effect. The glockenspiel and crotales are delicate, bright and very pretty, and the hand bells might come in useful at Christmas. GPO has two marimbas — the 'grand symphonic' model has a delightful, melodic, almost liquid tone, while the other has a slightly harder attack. In the jazz corner, the vibraphone has been nicely recorded — pressing the sustain pedal turns on its distinctive tremolo, a simple but authentic touch. The xylophone shares the marimbas' attractive mellow quality, and while that's preferable to a brittle, unmusical tone, some may wish it was more percussive. file:///H|/SOS%2004-10/Garritan%20Personal%20Orchestra.htm (7 of 13)9/26/2005 12:35:39 AM


The only truly esoteric sound in this section is the so-called 'glass harmonica', which is neither glass, nor a harmonica — it's a synth sound which emulates the sound of a fabulous 18th-century instrument (a chromatic set of rotating glass bowls, stroked on their rims to produce an ethereal singing sound). Danger! GPO's orchestral bass drum has an adjustable 'fundamental' bass tone — turning it up adds a lot of extreme low-end energy, which could do nasty things to your speakers if you're not careful. Accompanying this is an excellent, explosive tam-tam gong which sounds as though it could also do a lot of damage, clean orchestral snare drum rolls and hits, piatti, crash and ride cymbals. The library also contains a useful selection of hand percussion, as detailed in the 'GPO Instrument List' box at the end of this article. Most items are rationed to one or two hits — all sound fine, although the mark tree is slightly clonky. For some reason, there's no wood block, but the maker has made space for that most essential and expressive of orchestral instruments, the wind machine. This infernal canvas and wood contraption sounds like a medieval hairdryer. It may turn out to be the most useless sample ever included in a sound library, but it's there if you need it! Incidentally, the library offers its presets in 'dry' and 'wet' versions — the same samples are used for both, but the latter adds a splash of Kontakt reverb to the sounds. However, this uses a lot of CPU power, so users are advised to reserve the wet versions for auditioning instruments.

Personal Expression We take it for granted that note-on velocities control volume, but in GPO, the volume of the brass, woodwinds and most of the strings is controlled by the mod wheel, with velocity determining the amount of attack. This scheme could result in a deathly silence when the library is used to play back existing MIDI arrangements, but the maker has pre-empted the problem by supplying a MIDI file translator which converts all velocity data in the file into mod-wheel information. Using the wheel for dynamic expression immediately makes one aware of the importance of swells and fades in imparting a 'singing' style to melodies, and it doesn't take long to develop an expressive mod-wheel technique. However, one downside is that when you first load a sound, it's virtually inaudible till you push the mod wheel up! There are other potential pitfalls: instead of the traditional pitchbend and mod wheels, some keyboards use an all-in-one joystick, which springs back to a default 'zero' position after use. If your keyboard has one of these, you may have to use a sequencer to send GPO a 'mod wheel full up' message (ie. MIDI continuous controller 1, with a value of 127) on all MIDI channels when auditioning its sounds. This will also be necessary whenever you play with both hands! GPO has a very effective legato playing mode, which basically works by cutting off the samples' initial attacks. This simple technique makes consecutive notes sound more 'joined up', giving melody lines a convincingly mellifluous, flowing character. In the interests of realism, the solo wind instruments' presets are monophonic — this prevents notes from overlapping, and thus makes fast runs and even trills sound authentic. Legato mode is selected simply by holding down the sustain pedal while playing, but it's easy to switch the pedal back to its conventional purpose if necessary. Similarly, the solo wind instruments can be



made polyphonic by a simple parameter adjustment.

Compromise & Control Altogether, GPO supplies a total of 60 types of instrument and ensemble, which rises to 76 when you factor in the alternative makes of solo instrument. How can such a comprehensive set of sounds fit into 2GB? The answer lies in the way the instruments have been sampled. There are relatively few performance styles per instrument, and no staccatos, marcatos, grace notes, crescendos, diminuendos, glissandi, slides, col legnos, brass rips or falls, effects, runs, arpeggios, phrases or chords. All the sustained notes are looped, which is musically convenient, and also conserves sample memory. The biggest savings have been made in the area of dynamics: it's now common to find orchestral instruments with three or more dynamic layers, but GPO seems to consist mainly of one-dynamic performances. Users must therefore seize the initiative themselves, and use a combination of mod wheel and keyboard velocity to breathe dynamic life into the samples. The straightforward design of the GPO Kontakt Player should encourage users to create their own 'multis'. A set of clearly labelled rotary controls govern portamento (glide), volume, pan, reverb and tuning; a couple of additional controls (Var 1 and Var 2) can introduce random variations of intonation and timbre, in an attempt to ward off the dreaded 'machine-gun' effect of repeated samples. Newcomers to Kontakt will find it simple to use, while anyone who already owns the full version of the player will be able to load GPO's samples with full editing facilities. Another nice touch is that the player provides eight stereo outputs, so it's possible to route each instrument in an instance of the player to an individual channel on the mixer of your host application, or hardware output if you're running the stand-alone version.

Jamboree Bag If Garritan had stopped here, with GPO no more than a well-rounded orchestral library built into the Kontakt Player engine, it would still have proved to be a successful product. However, as mentioned in the first part of this review, the GPO package consists of a suite of applications to help you make the most out of the library, covering everything from sequencing to creating notation. There's even a bundled reverb plug-in, a version of Smartelectronix' Ambience (http:// magnus.smartelectronix.com), a competent reverb algorithm that includes presets specifically designed for GPO's instruments. GPO was initially supplied with an OEM version of Steinberg's Cubasis VST v4, a popular cut-down consumer and education-oriented version of Cubase VST 5 with the ability to record and edit MIDI and audio Parts, and utilities like a fullyfunctional VST Mixer with VST Instruments and effects. A further nice touch was



the inclusion by Garritan of a collection of template Cubase Songs such as 'Full Orchestra + Piano' and 'String Quartet', which are empty songs with preloaded instances of the GPO Player plug-in with the Ambience reverb plug-in. Unfortunately, the Cubasis VST 4 application was only available for Windows users, which meant Mac users were a little left out; but as we were finishing this review, Garritan were changing the GPO package to include Cubase LE instead. Cubase LE runs on both Windows and Mac OS X and is very similar to the first version of Cubase SL, a fully featured MIDI + Audio sequencer offering a large percentage of the functionality described in the SOS review of Cubase SX. GPO users will probably be pretty surprised at just how powerful this sequencer is, with its support for video, eight VST Instruments, four send effects, two insert effects per audio channel, Key The bundled Overture SE software comes and List editors and a basic score with a selection of templates, such as the String Section and Piano template shown editor, making Cubase LE easily worth here, which correspond to the templates a good part of the asking price of GPO supplied with GPO Studio to preload the on its own. Some of the more correct sounds on each channel. advanced features which are not present include the Logical Editor (presets are included, though), the Drum editor, MIDI plug-ins, the advanced score layout editor, and macros. While most MIDI + Audio sequencers support virtual instrument plug-ins, most score-writing packages you would want to use with GPO don't offer this functionality. One way around this would be to use a MIDI loopback utility with the stand-alone GPO Player, but, as mentioned earlier, this would limit you to playing back only eight instruments simultaneously. So for those people who want to use GPO with a score-writer, or those who want more than eight instruments with the stand-alone version, Garritan have thoughtfully provided a dedicated host application called GPO Studio for both Windows and Mac OS X users. Based on technology from Plogue's Bidule (www.plogue.com/bidule), GPO Studio provides access to eight instances of GPO Player (64 simultaneous instruments) and a single instance of the Ambience reverb plug-in, and adds MIDI ports to your system so you can choose GPO Studio as your MIDI output in applications such as Sibelius or Finale. There's also a further utility you can run to route MIDI data from your computer's MIDI input ports into GPO Studio, which makes it possible to run GPO on a separate, dedicated computer, if you should need to do this. Even if you don't have the budget for Sibelius or Finale, that still doesn't mean you won't be able to use a score-writing package with GPO, as the package file:///H|/SOS%2004-10/Garritan%20Personal%20Orchestra.htm (10 of 13)9/26/2005 12:35:39 AM


includes Geniesoft's Overture SE (www.geniesoft.com). Overture is reasonably intuitive to operate, and can handle anything from a single stave of music to a full orchestral score, with a full range of symbols and layout facilities, including the ability to hide or extract certain parts. Scores created in Overture SE will play GPO's samples via the GPO Studio application, as previously described, and there's even a GPO menu in Overture SE, enabling you to name the sounds used in each instance of GPO Player in GPO Studio, with preset Soundsets included for the Multi-programs and other templates included with GPO.

Garritan Personal Orchestra Instrument List Note: numbers in brackets indicate different makes of instrument. Instruments marked with an asterisk include three extra sets of samples, designed for building 'virtual ensembles'. STRINGS

WOODWINDS

12 first violins.

Piccolo.

10 second violins.

Flute.

10 violas.

Alto flute.

Eight cellos.

Bass flute.

Seven double basses.

Oboe (1*, 2, 3, 4).

Solo violin (1*, 2*, 3*).

Oboe d'Amore.

Solo viola.

English horn (1*, 2).

Solo cello (1*, 2*, 3*).

E-flat clarinet.

Solo double bass.

B-flat clarinet.

KEYBOARDS & HARPS

Bass clarinet.

Steinway grand piano.

Contrabass clarinet.

Celeste.

Bassoon (1*, 2).

Harpsichord.

Contrabassoon (1, 2).

Pipe organ.

DRUMS & CYMBALS

Harp (1, 2).

BRASS

Bass drum. Snare drum.

Piccolo trumpet.

Crash cymbals (piatti).

Trumpet (1*, 2*).

Suspended cymbals.

Trombone. Bass trombone (1, 2). French horn (1*, 2*). Tuba (1, 2). Contrabass tuba.

TUNED (PITCHED) PERCUSSION Timpani. Tubular bells. Glockenspiel. Crotales. Marimba (1, 2). Xylophone. Vibraphone. Hand bells. 'Glass harmonica'.

UNPITCHED PERCUSSION Tam tam. Gongs. Mark tree. Triangle. Cowbells. Sleigh bells. Castanets. Claves. Shaker. Guiro. Ratchet. Tambourine. Wind machine.



Summary Packed with demos, templates, MP3s, PDF documents and MIDI files, GPO is quite an elaborate affair, but its 60-page booklet throws light on its technicalities. Pitch ranges, performance techniques and stage positions are also explained, reminding us there's a musical mind behind all the science. And the library's music-first stance and commitment to education has already led to a close collaboration with Berklee College of Music, resulting in a GPO 'Berklee Edition'. While GPO has an obvious market for beginners, it also offers a degree of professional appeal for anyone looking to sketch out orchestral arrangements with a single computer. It's becoming a little bit of a cliché to talk about laptops these days, but GPO will definitely have value to mobile musicians, since the player and small samples make it an efficient solution for undertaking reasonably large-scale work without the library consuming your laptop in terms of hard disk space or battery power and processor usage. GPO is also a good alternative for Sibelius users considering the Gold version of the Kontakt player for Sibelius 3, as some of the instruments are of a slightly better quality. The only caveat is that using GPO with Sibelius requires a little more care in setting up with GPO Studio (this is described in the GPO manual) and in creating instrument definitions for Sibelius' more advanced performance abilities. As a package, GPO is pretty unique, but East West/Quantum Leap's Symphonic Orchestra Silver Edition is another product worth considering if you're looking for the ultimate condensed orchestra, although it costs slightly more at around £199. This is a stripped-down version of the full Symphonic Orchestra package, is also based around Native Instruments' Kontakt Player, and offers a similar selection of instruments (including some choir samples), although it doesn't include any of the extras that make GPO appealing. Silver Edition users have an upgrade path to the other members of the Symphonic Orchestra family, although Garritan have also worked out a deal so that GPO users have an upgrade path to products from the Vienna Symphonic Library. Decisions, decisions! In the commercial world, Goliath usually tramples David, but this intelligent orchestral package may well buck that trend; its affordable price, comprehensive nature and user-friendly attitude are sure to attract many musicians. Although some orchestral sample users will miss the subtly differentiated performance options of larger libraries, others will be glad to work with a simpler set of sounds that do the job quickly and efficiently. For composers and arrangers chasing deadlines, and musicians who like to get fast results, GPO could be the perfect solution. Gary Garritan may not yet have cured the world of 'sample bloat', but by instigating a brave new anti-expansionist policy and keeping the price low, he has struck a blow for all those scorewriters who don't yet have a collection of gold discs in their lavatories. Published in SOS October 2004






Groove Tubes MD1b FET

In this article:

Hardware Overview Standmount Confusion Listening Tests

Groove Tubes MD1b FET £120

Groove Tubes MD1b FET Condenser Microphone Published in SOS October 2004 Print article : Close window

Reviews : Microphone

pros Full, rich sound quality. Higher output level than most comparable mics. 'Love it or hate it' looks. Gentle power-up/down feature.

This new mic is a reissue of Groove Tubes' first largediaphragm condenser, but with solid-state electronics replacing the valve to offer a more affordable package.

Useful novice recording information in handbook.

Hugh Robjohns

cons Relatively noisy by modern standards. No specifications or qualitycontrol plots supplied. Fiddly standmount. Proper shockmount not included.

summary This solid-state reissue of Groove Tubes' Model 1 largediaphragm valve condenser mic shares the same extended bandwidth and generally flattering sound character, but with the cost and reliability advantages afforded by solid-state ClassA FET electronics.

information £119.99 including VAT. Digital Village +44 (0)20 8440 3440. Click here to email www.dv247.com www.groovetubes.com

The American manufacturer Groove Tubes was founded towards the end of the '70s by Aspen Pittman — a man with a long-standing fascination for valve audio equipment. From its early beginnings in a garage workshop, the company have grown into an international business manufacturing and distributing a wide range of high-quality valves, alongside a range of valve-based guitar amps, speaker systems, and recording equipment including microphones, compressors, preamps, and equalisers. Among an impressive lineup of microphones, the company offer both valve and solid-state medium- and large-diaphragm models, with either fixed, interchangeable, or variable polar patterns. The GT55, GT57, GT66, and flagship GT67 models were reviewed in SOS April 2003, but the subject of this review is an 'enhanced' reissue of the very first Groove Tubes valve mic, the MD1 or Model 1, with solid-state electronics in place of the valve to bestow a significant price reduction.

Hardware Overview The original Model 1 valve mic has been completely reworked and has been reissued as the MD1b Vacuum Tube. However, its solid-state sibling is the MD1b FET. Both mics share exactly the same capsule, output transformer, and mechanical construction (albeit with different finishes), but the FET version employs a Class-A FET-based impedance converter and preamp in place of the original valve. The review model was supplied as part of a pair (serial numbers

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302023 and 302024), each mic shipped in its own cardboard protection case and the two kept together by an overall sleeve. Inside each foam-lined cardboard shipping box is a fabric-covered, foam-lined wooden box which contains the microphone, standmount bracket (complete with an adaptor for converting from 3/8-inch to 5/8-inch threads), warranty card, and handbook. The latter — Choosing & Using Microphones — is well written, with good background technical information and practical mic-placement advice, but offers little specific information on the MD1b FET, and there are no quality-control test plots or even specifications included at all. The microphone is housed in a cylindrical metal case, measuring 191mm long (7.5 inches) and 50mm (just under 2 inches) in diameter. It weighs roughly 1400g, and the steel case is painted matt black (the valve original is silver), with a dual-layer wire-mesh grille over the top and on the front and rear faces around the capsule. The case is held in place by two screws at the base, and has various holes, cutouts, and etched markings on it. The forward axis of the cardioid pattern is identified by the etched model name at the bottom and a cardioid graphic at the top of the case front, along with a grid of twelve holes. These were intended to allow the user to view the valve inside the sibling Model 1B tube mic (as well as help to promote a cooling airflow), but serve no real purpose here. The rear side of the mic's case features an inverted cutout Groove Tubes logo designed to look right when the microphone is suspended upside down. The mic's serial number is not visible externally — it is printed on a label glued to the output transformer, and can only be seen after removing the microphone's outer case.

Standmount Confusion The microphone is supported by (or suspended from) a simple bracket which screws to the base of the mic. A chunky plastic ring cup extends from the mic stand adaptor and a (separate) threaded metal collar passes through this ring to screw into the microphone's base, leaving the mic's XLR output connector accessible through the centre of the construction. Unfortunately the correct method of assembly is not entirely obvious, and the complete lack of any instructions doesn't help, so it is very easy to fit the parts together incorrectly! Fortunately, if the adaptor is assembled incorrectly the microphone will still be held reasonably securely, but it's far harder to tighten and adjust. The difference may appear fairly subtle in the photographs, but the correct form is significantly easier to use. Note that, when fitted correctly, the 'cup' of the collar sits around the microphone's base, and the knurled locking ring ends up slightly proud of the collar. If the plastic collar is fitted the opposite way around, the knurled part of the locking ring sinks into the cup and therefore becomes difficult to adjust. Sadly, the adaptor also has to be removed before the mic can be put away in its case. The supplied adaptor incorporates no shock protection at all, although there is an optional basket shockmount design that supports the mic from elasticated loops. file:///H|/SOS%2004-10/Groove%20Tubes%20MD1b%20FET.htm (2 of 6)9/26/2005 12:35:41 AM


However, the microphone capsule is mounted on a pliable rubber internal suspension, and this does offer a degree of protection from mechanical shocks anyway. No technical specifications are given at all, but an Internet search eventually provided the following figures. The frequency response is claimed to extend between 20Hz and 20kHz, and a response trace I found suggests the mic is fundamentally flat within ±1dB up to about 8kHz, above which there is a peak in the response reaching +4dB at about 12kHz. This high-frequency peak is a common characteristic of most of the Groove Tubes mics, although it would appear to be better controlled here than in some of the other models.

The difference between correct (left) and incorrect (right) assembly of the standmount's plastic cup and knurled metal ring is easily confused, and there's no indication in the handbook as to which is the correct configuration.

Output sensitivity is quoted as a very generous 32mV/Pa, with a maximum SPL of 134dBSPL for one percent distortion. Groove Tubes claim that their quality control is such that every mic is 'matched to the appropriate 'golden microphone' specification within ±1dB across the entire frequency range', and that even more accurately matched stereo pairs are available. The microphone's self-noise figure is a little disappointing by modern standards, at 22dBA — a full 15dB noisier than the Neumann TLM103, for example. Having said that, the Neumann produces almost 4dB less output level, which mitigates the situation slightly. The mic has a transformer-balanced output with a nominal 200(omega) output impedance, and requires 48V phantom power. The capsule itself is a traditional 1.1-inch design constructed from brass, with a three-micron evaporated-gold diaphragm. The capsule is mounted on a metal post supported by a flexible rubber cone to provide some mechanical isolation from the microphone body. The electronics are carried on two printed circuit boards mounted either side of the microphone's internal frame, and conventional components are used throughout. PCB markings suggest the same boards are used for the GT55 and GT77 mics as well. The front board appears to carry mainly the Class-A audio circuitry, with a metalcanned FET at the top handling the capsule's signal and a second transistor evident further down the board. The rear PCB appears to carry the power-supply components, and a compact output transformer sits at the base of the microphone. A nice feature is that the phantom supply circuitry has been file:///H|/SOS%2004-10/Groove%20Tubes%20MD1b%20FET.htm (3 of 6)9/26/2005 12:35:41 AM


designed to ramp up and down slowly (over about twenty seconds) to minimise the risk of loud pops and bangs on the mic's output when connecting things up.

Listening Tests As would be expected, the MD1b FET benefits from the use of a decent pop shield — the internal gauze grille offers little protection from plosives when used for close vocals. The mic seemed slightly more resilient to stand-borne vibrations when suspended from above (rather than below) using the supplied standmount adaptor. However, I found mechanical noise remained a problem on occasions, and with no integral high-pass filter a decent elastic shockmount would certainly be a very useful purchase. A suitable device is available as a cost option, but I really think Groove Tubes or their distributors should consider including one with the MD1b FET as standard. After all, most of the Chinese microphone manufacturers seem able to include free shockmounts with even their least expensive mics these days. My first impression of the MD1b FET was of a very nice and versatile mic for the money, and I have been using the supplied pair alongside the rest of my permanent mic collection for some time. I was particularly interested in how the mic fared against my most comparable mics: the AKG C414B ULS, the CAD M179, the Neumann TLM103, and the Audio-Technica AT4040. All of these are large-diaphragm, true-condenser mics with solid-state electronics, but all of them are more expensive — between 1.5 and five times the UK price of the MD1b FET. Interestingly, they all exhibit a high-frequency presence peak — although with varying degrees of peakiness and bandwidth — and similar variations in polar response with frequency. The MD1b FET had by far the highest output level of the collection, and needed almost 10dB less gain than the quietest of its review rivals, the AKG C414B ULS, but it was also clearly the noisiest too. However, its self noise wasn't an issue when used for close-miking applications, and only became an irritation when I tried to use a pair of MD1b FETs as a distant stereo pair to cover a small string group. The AKG and Neumann mics sounded the smoothest and most refined of the collection, the AKG seeming to have a better bass response when used at a respectable distance too, but these two mics are significantly more expensive that the MD1b FET. The obvious presence peaks of the CAD and Groove Tubes mics sounded particularly similar to each other, while the AudioTechnica's presence hump seemed to start a lot lower — around 6kHz — and was broader than the Groove Tubes and CAD mics, which appeared to have narrower peaks starting around 10-12kHz. This characteristic tended to give more a sense of 'air' instead of the sharper presence 'edge' typical of the Audio-Technica mic. The AKG's peak was the most subtle, followed file:///H|/SOS%2004-10/Groove%20Tubes%20MD1b%20FET.htm (4 of 6)9/26/2005 12:35:41 AM


closely by the Neumann's — although neither were as flat as my reference Sennheiser MKH40s. However, I should stress that these are all relatively minor differences when heard in isolation, and which will tend to favour different mics for different sources. Overall, I found the MD1b FET to have a typical large-diaphragm character with a well-behaved polar response and a reasonably balanced off-axis response, which means that spill from off-axis sounds won't become too badly coloured. The general sound quality is fairly rich and full — tending towards the 'larger than life' sound typical of good valve mics, but not quite reaching that lofty goal. The mic is suitable for a broad range of close-miking applications, but it wouldn't be my first choice for more distant mic techniques because of the relatively high noise floor. The airy top end and rich mid-range suited most vocals well — although not all, especially sibilant female singers — but male voices in particular were portrayed with a nice, full bodied character and good articulation. The high maximum SPL figure also allowed the mic to be used for miking up guitar cabinets, percussion, and brass sections. For the latter I found I often had to tilt the mic down to help reduce the effect of that high presence peak, but the same peak proved beneficial in drum overhead applications, where transients were delivered with clean, crisp attacks. Overall then, I'd put the MD1b FET in a similar category as mics like the CAD M177 and Audio Technica AT4040 in terms of the general performance and sonic abilities. Its price is not too dissimilar to the quieter CAD M177 mic, and it is worth noting that the CAD M177 VP version at £188 is supplied with a suspension shockmount and pop shield as standard. Unsurprisingly, the MD1b FET is not quite as smooth and detailed as the Neumann TLM103 or AKG C414B ULS — although they both have their own idiosyncrasies anyway — and not as rich and involving as Groove Tubes' more expensive valve mics. However, it does provide a very good and usable sound quality at a very attractive price which competes closely with many of the better Chinese offerings, and its vintage styling and sound character will have strong appeal for many. It's certainly well worth auditioning, particularly for anyone considering a new condenser vocal mic. Published in SOS October 2004






K+H O300D & Pro C28

In this article:

Infinite-baffle Design Amplifier & Electronics Pro C28 Digital Controller Using The O 300D Pro C28 Comparisons Verdict

K+H O300D & Pro C28 Active Monitors & Digital Controller Published in SOS October 2004 Print article : Close window

Reviews : Monitors

K+H O 300D & Pro C28 £3090/£2719 pros Superlative sound quality and performance. Extremely compact dimensions. Comprehensive roomtailoring EQ. Flexible mounting options. Facility for external digital crossover option.

Despite its small size, this new system leads the field when it comes to nearfield monitoring accuracy. Hugh Robjohns

The German manufacturer Klein +Hummel — named after founders Horst Klein and Walter Hummel — have been making studio monitors and cons PA loudspeakers at their factory near Digital input probably Stuttgart since 1945, and have built an unnecessary for most users. enviable reputation for reliability and Pro C28's relatively high cost and limited sample-rate quality. One of the company's smaller Photos: Mark Ewing capability. studio monitor systems — the O 100 and its O 800 subwoofer — was reviewed in SOS November 2002 and, a small summary The O 300D is an active three- niggle about the supplied handbooks being in German aside, I had nothing but way nearfield monitor which, praise for them. despite its unfeasibly small box, produces a huge sound in every regard. It provides an extremely clean and neutral sound across the entire spectrum, with a very high degree of resolution and transparency. However, the Pro C28's restricted sample rates and very high cost will limit its attractiveness, given that it will mostly provide only incremental improvements to an already very fine monitor.

The subject of this review is an altogether more serious studio monitor: the active three-way O 300D, along with an optional digital crossover unit called the Pro C28. Sadly, the handbooks supplied with the review equipment were still written entirely in German, which didn't impress, but with the aid of the Internet I was able to find an English-language version and eventually glean what I needed to know.

Infinite-baffle Design

information O 300D monitors, £3090.26 per pair; Pro C28 controller, £2719.10. Prices include VAT.

The O 300D is a surprisingly small active three-way nearfield monitor, clearly intended for use on the meterbridge of a large-format studio console. It measures 253 x 383 x 290mm (hwd), weighs in at 13.5kg, and features full magnetic shielding, so placement near moving-coil meters and computer VDUs is not a

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K+H O300D & Pro C28

Beyerdynamic +44 (0) 1444 258258. +44 (0)1444 258444. Click here to email

problem. The sealed (infinite-baffle) cabinet is constructed very solidly from MDF finished in a charcoal-grey enamel as standard, although there is also a flockfinished version (recommended if the speakers are likely to be moved a lot), and various other special finishes are available to order. There are also various builtwww.beyerdynamic.co.uk in options and kits for mounting the monitors in various ways if the console www.klein-hummel.com meterbridge is not appropriate. Each cabinet has an 8mm threaded socket on each side to which various optional mounting adaptors and brackets can be fitted. There are also heavy-duty brackets with quarter-inch holes fitted at the rear of the speaker for wall or ceiling mounting. As standard the monitors are not supplied with any kind of protective front grille, although a fine wire-mesh grille is available as an option, if required. The driver complement comprises a 210mm (eight-inch) polypropylene-cone woofer, a 76mm [three-inch] soft-dome mid-range unit, and a 25mm [one-inch] titanium-dome tweeter. The tweeter is mounted above the mid-range unit — both being the same components as used in the company's larger O 500C monitor — and the pair are located beside the woofer, which occupies the remaining baffle space. The red K+H badge below the woofer illuminates when mains power is applied, and flashes if any of the amplifiers clip or if the protection limiters are triggered. K+H prefer to use a three-way format even in modestly sized speakers like the O 300D because of the numerous performance advantages it affords — even though there is a corresponding cost penalty. Sharing the frequency range across three drivers reduces the demands on each individually, permitting much better control and accuracy. The bass driver no longer has to generate mid-range frequencies, avoiding problems with resonances, breakup modes, and harmonics, as well as reducing the problem of standing waves in the cabinet and Doppler distortions (mid-range frequencies being modulated by LF signals using the same cone). The mid-range and tweeter domes are housed in their own sealed enclosures and thus are completely independent of the woofer and the acoustic energy contained within the main cabinet. The front baffle is made from a material referred to as 'Low Resonance Integral Moulding' or LRIM, and this can be machined to produce complex threedimensional shapes such as the waveguides used for both the tweeter and midrange drivers. These waveguides help control dispersion, as well as allowing the two drive units to be mounted in the correct vertical plane for accurate time alignment. An elliptically shaped waveguide is used for the tweeter to provide wider horizontal dispersion than vertical, and a modest horn shape is used for the mid-range dome. The narrower vertical dispersion is intended to minimise HF reflections from the console surface, which would otherwise cause detrimental comb-filtering effects. The wider horizontal dispersion ensures a large 'sweet spot', making it easier to hear the spatial imaging over a wider proportion of the console. The O 300D monitors are 'handed', meaning that they are available as left- and right-hand models — K+H recommend placing the speakers with the tweeters on the outside edges when the speakers are used as a stereo pair. However, there


K+H O300D & Pro C28

is no dedicated centre version, and in 5.1 surround systems either a left- or righthand monitor can be used for the centre channel. In this application it should be set up vertically, with the tweeter and mid-range driver at the top. For the LFE channel, one or two O 800 subwoofers should be used.

Amplifier & Electronics The amplifier chassis at the rear of the monitor incorporates three discrete MOSFET power amplifiers providing 150W RMS (250W peak) for the woofer and 65W RMS each for the mid-range unit and tweeter. The mid-range amp can manage 75W peaks, while the tweeter amp can produce 110W peaks. Each amplifier channel has built-in protection limiters. The peak SPL measured at one metre is quoted as 112.8dB (for three percent distortion), although at more realistic listening levels (95dBSPL) the distortion is specified as being below 0.5 percent for all frequencies above 100Hz. The overall frequency response is given as 40Hz-20kHz (±2dB) and the supplied traces are remarkably flat. The crossover points are set at 650Hz and 3.3kHz, both with fourth-order (24dB/ octave) slopes. There is also a protective subsonic filter which rolls off at 6dB/ octave below 30Hz. There are two audio inputs available: analogue and digital. The Lundahl transformer-balanced (and floating) line-level analogue input is connected via an XLR socket and has a nominal sensitivity of +6dBu — although there is also an input attenuator which ranges from zero to -24dB to accommodate higher levels. Digital signals can be connected in either the conventional AES3 format via the same XLR socket used for the analogue input, or via a BNC socket in either AES3id (unbalanced) or S/PDIF formats. The balanced AES signal is routed through a transformer, the secondary of which is wired in parallel with the BNC socket. This enables the latter to be used as a loop-through if required (to convey the stereo digital input signal on to the other stereo speaker, for example), or to connect a 75(omega) termination either directly or via a BNC 'T' piece, as appropriate. The handbook provides comprehensive instructions about how to connect inputs of various formats correctly. A screwdriver-operated rotary switch selects the type of input signal in use: analogue, left digital channel, right digital channel, or a mono sum of both left and right digital channels. The digital input is handled with a 24-bit delta-sigma converter that can accept any standard sample rate between 32kHz and 96kHz. As with most active monitors, the O 300D features a range of room-correction equalisation facilities, all adjusted through recessed screwdriver-operated controls. The bottom end can be tamed to compensate for placement in free space, near a wall, in a corner, or in a corner near the ceiling — each successive setting introducing 3dB more attenuation to the woofer's output. There is also a mid-range equaliser which attenuates a broad band of frequencies extending from about 50Hz up to 1kHz to compensate for reflections from a console when mounted on the meterbridge. The options provide from zero to 6dB of attenuation in 2dB steps. Finally, there is a high-frequency level control to accommodate the


K+H O300D & Pro C28

variabilities of control-room HF absorption and reflection. This provides level trim from +1dB to -2dB in 1dB steps, and it affects frequencies from about 2kHz upwards. The handbook provides a complete set of traces and measurements for the O 300D, which appear very impressive. One plot is of the system group delay, and this is interesting because it shows negligible delay for any frequency above about 100Hz, and only a gently rising delay below that to about 10ms at 50Hz. This is linked to the use of a sealed enclosure, and results in a very tight and fast bass response with good transient accuracy. Other rear-panel facilities include a ground-lift switch to separate signal and chassis grounds, a mains rocker switch, and an IEC mains inlet socket. One particularly interesting feature is a seven-pin XLR socket and its associated slide switch. The socket provides direct inputs to each of the three power amplifiers, bypassing all of the internal input handling, crossover, and protection circuitry (apart from a peak limiter for the bass amp). The slide switch disables this internal circuitry and enables the external three-channel discrete inputs, the aim being to drive the monitors from the optional Pro C28 digital controller unit. This unit provides input-signal handling, digital conversion, crossover, sophisticated room equalisation, and protection limiters, all performed in the digital domain with high-resolution processing — see the 'Pro C28 Digital Controller' box for more details.

Pro C28 Digital Controller The Pro C28 is a rackmounting stereo digital loudspeaker controller which occupies 2U of rack space. It is primarily designed for use with the O 300D studio monitor — although it can also be configured to operate with a wide range of monitors and speakers systems — and creates up to four audio bands per channel to drive a tweeter, mid-range driver, woofer, and subwoofer. The crossover uses specially developed Finite Impulse Response (FIR) filtering algorithms for precise and independent adjustment of the overall system's amplitude and phase responses, and the system is similar to that built into the company's large O 500C monitors. The linear-phase FIR filter characteristics allow much more accurate in-room response and transient behaviour than is normally possible with conventional analogue crossovers and electronics. In addition to the crossover FIR filters, the unit also incorporates delay facilities for each input channel (up to 1000ms) and an Infinite Impulse Response (IIR) filter section that enables real-time variable equalisation. This caters for both standard boundary correction and very sophisticated room-correction EQ — the latter being accommodated with ten bands of fully parametric EQ per input channel. To protect amplifiers and drivers, the C28 controller incorporates separate digital limiters for each output, and these take into account both transient peak and sustained RMS levels, with a look-ahead facility to ensure every transient is file:///H|/SOS%2004-10/K+H%20O300D%20&%20Pro%20C28.htm (4 of 8)9/26/2005 12:35:43 AM

K+H O300D & Pro C28

caught. The software is also able to model the power amplifier's load performance to ensure effective protection, and can anticipate overheating and over-excursion of each driver by modelling their voice coils. One particular advantage of the FIR approach is that the crossover slopes can be as steep as 96dB/octave, and so the crossover region spreads over a much smaller region than is possible with analogue filter designs. This is claimed to benefit the speaker's overall dispersion and directivity characteristics. The maximum SPL of the speaker is also increased because of the accuracy of the amp and voice-coil modelling and the precise control exerted by the digital limiters. The Pro C28 can either be operated from its front panel, with an optional RC55 infrared remote control, or from a PC via RS232 or MIDI ports. In complex surround-sound systems several Pro C28 units can be linked together in one control group using MIDI. The front panel provides an input level meter, overload/ clip lights for each output, and a two-line 24-character alphanumeric display. The unit can store up to 70 different parameter sets — which can also be transferred to/from a PC — and settings can be password protected to prevent unauthorised tampering. The unit's internal flash memory is pre-programmed with the relevant FIR settings for a range of K+H monitoring systems, enabling a very fast and hassle-free basic installation. Once the unit is hooked up to the monitors via supplied seven-pin XLR cables, and the input signal connected to the controller, the user simply has to select the appropriate monitor type and adjust the room EQ as necessary. The IIR EQs can then be used to provide more complex and sophisticated room correction, ideally based on the results of accurate room analysis and measurement equipment. Once established, K+H can also translate a customer's IIR filter settings into FIR parameter sets, to maintain the linear phase attributes of the overall system, if required. One of my concerns with the digital input on the O 300D is the potentially reduced resolution which occurs if the input signal is attenuated in the digital domain. To address this issue, the Pro C28 uses a 'stacked configuration' for its 24-bit deltasigma A-D and D-A converters, with a 'gain ranging' technique which is claimed to provide a usable dynamic range of more than 130dB. The two analogue XLR inputs are electronically balanced (an input transformer can be switched into circuit if required), and can accommodate peak levels of +24dBu. The four balanced analogue XLR outputs per channel can be switched to operate with nominal levels of +16dBu, +10dBu or +4dBu. Each channel is also equipped with a seven-pin XLR for single-cable connection to K+H monitors. Internal digital signal processing is performed with 48-bit precision, but can only operate at sample rates of 44.1kHz or 48kHz. The unit is also equipped with a separate digital input and output, which can be configured to operate as an insert point if you like.

Using The O 300D I'm a firm believer in the advantages of three-way loudspeakers myself, and use the rather large passive three-way PMC IB1s as my main reference monitors. So it was with great enthusiasm that I rigged the O 300Ds alongside to let battle commence! I hooked the K+H speakers up using the balanced analogue inputs, checked that all the EQ settings were flat, switched them on, and started playing


K+H O300D & Pro C28

some familiar reference CDs. I have to say my initial reaction was one of shock! These are modestly sized nearfield monitors, yet they were producing an enormous sound — not only were they capable of going loud effortlessly, but there was also a deep bass output and a huge stereo image. I was so surprised, I quickly checked that I wasn't inadvertently driving the PMCs at the same time! I wasn't... these monitors really do produce a staggeringly big sound — and the fact that they do it from such small boxes is astounding. Given the cabinet size, the amount and depth of bass is very impressive — as is the naturalness of it. I can quite believe the claim that the response is 2dB down at 40Hz, and because of the sealed cabinet the response falls gently, so that even the lowest musical octave is still conveyed into the room. There is nothing hyped about the bass at all, either. It is very accurate and believable, without any of the boomy, one-note quality from which so many small-box ported designs suffer. It is also very tight and well controlled — there is no obvious overhang, and kick drums and bass guitars are delineated perfectly, allowing you to hear exactly who is playing what and when. The most obvious strength of all welldesigned three-way systems is the midrange clarity — the result of having a bass driver that only has to worry about bass — and in the O 300D the clarity and resolution of the mid-range is superlative. The dispersion of each driver through the crossover regions is also very well matched, so that the entire system is phenomenally well integrated — both on and off axis. The top end is as well balanced as everything else, although I found that in my room I preferred the balance with the HF trimmer set to -1dB. An advantage of a small cabinet, and having the mid-range and tweeter mounted in waveguides, is that the imaging is excellent and stable over a wide area. Simple coincident mic recordings produced lifelike imaging with huge width and depth information, seeming to fill the room with the soundstage. My everchallenging spoken-voice recordings were handled with great aplomb, delivering a totally believable and extremely neutral character. A variety of singing voices were also handled superbly, with excellent natural dynamics and every little detail laid bare for inspection — but without any hint of the slightly shouty or overly forward character that many monitors seem to employ to appear 'revealing'. The O 300D comes across as a really high-class, accurate monitor in every way, despite its diminutive size. For its intended nearfield applications, there is plenty of monitoring level available, and the sound doesn't harden with increasing levels until you get close to near-deafening limits. Complex orchestral recordings were handled with the same consummate skill as file:///H|/SOS%2004-10/K+H%20O300D%20&%20Pro%20C28.htm (6 of 8)9/26/2005 12:35:43 AM

K+H O300D & Pro C28

a wide range of rock, pop, and jazz: quality recordings stood out as such, while mediocre recordings were revealed for what they were. Bass-heavy tracks were delivered without any sign of strain, and even after being glued to these monitors for an entire day I felt no hint of fatigue at all. The O 300D is without question one of the very best active three-way monitors I have ever used — and undoubtedly the best monitor bar none for a cabinet as small as this. It is quite simply astonishing. Switching briefly to the digital input, I felt the quality of the internal converter was pretty good, but it didn't quite match the degree of resolution afforded by my Apogee PSX100, and became the quality 'choke point' in an otherwise exceptional signal path. Part of the quality reduction is probably due to the need to control the monitoring level in the digital domain — thus, at modest listening levels several of the most significant bits remain unused and resolution is inherently compromised. Using the analogue input, the converter was always working at optimum resolution and the analogue signal attenuated to suit, with minimal loss of quality in the process. Consequently, I would prefer to see the digital input provided as an option rather than a standard feature — hopefully with a corresponding reduction in the price of the monitor (albeit a small saving). I would also like to see built-in and switchable 75(omega) terminations provided for the BNC socket, as many users won't realise the importance of correct termination (despite the clear instructions in the handbook) or have suitable BNC 'T' pieces and terminations available.

Pro C28 Comparisons With the Pro C28 controller configured and hooked into the O 300D monitors, and with all the IIR equalisers set flat, I restarted my listening tests using the same material. Initially, I was hard pressed to tell any difference, although when I moved on to material with a fast-moving and dynamic bass content I became aware of a slightly greater sense of reality and, if possible, crisper transient edges to most instruments. I didn't notice any significant improvements to the stereo imaging, and as I wasn't driving the system to ear-splitting levels any improvements in the overload protection were wasted on me! While I did perceive some incremental improvements in overall quality — particularly at the bottom end — I have to say that I don't think I heard more than £2500-worth of improvement! In 'difficult' control rooms, the Pro C28 would certainly allow precise and elaborate equalisation to be applied, potentially enabling a far more linear response to be obtained than would ever be the case with conventional analogue electronics. However, sophisticated measuring equipment (and the experience to interpret the results) are required to optimise the filter settings and it wasn't practical to assess this aspect of the system for this review. However, I have heard the improvements such equalisation can bring to a system when used properly (during my review of the similarly equipped Tannoy Ellipse iDP monitors), so I have no doubt that the Pro C28 would be a file:///H|/SOS%2004-10/K+H%20O300D%20&%20Pro%20C28.htm (7 of 8)9/26/2005 12:35:43 AM

K+H O300D & Pro C28

very powerful and beneficial tool in the appropriate circumstances. I have some concerns over the limited range of sample rates — many potential users might well prefer to work with 96kHz or even 192kHz digital sources, for example. The balance of cost to benefit is a major concern of mine too, and it is not helped by the fact that even if an O 300D installation uses the Pro C28, the customer still has to buy a complete set of utterly redundant I/O, crossover electronics, and limiters within each O 300D monitor. Perhaps K+H should consider supplying a stripped-down, lower-cost version of the monitors for exclusive use with the Pro C28 controller, and update the DSPs and interfaces to accommodate the higher sampling rates that are becoming commonplace in many studios now.

Verdict Even without the Pro C28, these K+H O 300D monitors are expensive, but I found them astonishingly good and very easy to work with. For a similar financial outlay in the UK, the obvious competition would include the ATC Active 20, Dynaudio Air 15, Genelec 1032AM, JBL LSR6328P, and Tannoy Ellipse 10. These are all good reference-quality monitors, but all apart from the Tannoy Ellipse are also two-way designs, and even in the Tannoy you don't have a traditional woofer/mid-range/tweeter complement such as in the K+H design. The resolution and detail of the O 300D, especially through the mid-range and upper bass, is exceptional, as is the quality and quantity of bass. Both easily justify the professional price tag, before you even consider the compact dimensions, which make the monitor ideal for location work or when working in unfamiliar studios. When the time comes to upgrade my nearfields these K+H monitors will be at the top of my auditioning list, and I strongly recommend you put them at the top of yours too! Published in SOS October 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



Latest Sample CDs

In this article:

Luscious Grooves ***** Electro Magnetic Pulse **** SAM Solo Sessions **** Saxophones 1 *****

Star Unused Mac OS X Cats

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

Reviews : Sound/Song Library

***** Caracal **** Ocelot

Luscious Grooves *****

*** Serval

MULTI-FORMAT

** Oncilla * Kodkod

Behind the dodgy artwork of this new release from Sample Lab lurk almost 300 24-bit drum and percussion breaks. These are organised into 13 tempo groups covering 80-135bpm, and the booklet claims that all the loops are sample accurate, allowing you to layer any in the same tempo group without further processing. Ever the sceptic, I gave this claim the once over in my sequencer, but there was no knocking it. That said, the loops do swing differently, so layering isn't entirely a no-brainer. Sonically, there's a lot of variety here. Some loops are obviously electronic, others are straight drum kits, others are built from ethnic percussion; the sound quality is by turns big and hi-fi, small and crispy, warm and retro; and the processing has its share of subtle vintage flavours and real room/plate treatments, but also periodically goes to town with extreme dynamics, distortion, and filtering. What is clear is that care and attention have been lavished on these loops, which is a breath of fresh air when many 'yet another bunch of loops' libraries seem satisfied just to run boring samples through the odd distortion plug-in. This library doesn't really fit very well within any specific style, so hip hop producers are as likely to benefit from this library as house 'choon'-smiths or the more technological emo rockers. The strength of this approach is that almost anyone is likely to get something from Luscious Grooves, so I'd suggest that it would make an excellent starter pack for the beginner, as well as being a solid workhorse for the jobbing producer. Of course, the ability to layer samples so easily also means that you can get a lot more mileage from this collection than you might think at first — there are as many different shades of any given loop as there are other loops to mix in with it.

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Latest Sample CDs

One thing which will endear this release to professional producers is that the documentation not only includes the audio CD track number, WAV file name, and tempo indication, but also adds a few well-chosen words of description. Why can't every manufacturer of sampled loops do this? First of all, you get some idea of which files are most likely to contain the kind of sound you're after. Plus, if you come across a loop you like but don't want to use straight away, it's a lot easier to remember which one it was at a later date. If you have even a few different loop libraries then this kind of information is invaluable, because it lets you spend more time making music and less trawling through sample CDs. It's been tough deciding on the star rating here, because, thoroughly solid and professional though Luscious Grooves is, it somehow left me more impressed than inspired. This is probably because it covers such a wide stylistic range — no matter who you are, only some of the material is going to press your particular buttons. However, I reckon I could rely on finding a slick, useable break for just about anything I was working on, and this shouldn't be undervalued, especially in a professional environment where time is limited; so five stars it is! Mike Senior Audio CD and WAV, EXS24, Halion, and Reason Refill CD-ROM 3-CD set, £59.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.samplelab.com

Electro Magnetic Pulse **** AUDIO+WAV+REX Created by Perry Geyer of Cybersound, Electro Magnetic Pulse has a title which could place it in a number of musical styles. However, the emphasis here is on breakbeats with attitude, aimed at styles such as Electro, Big Beat, and the more experimental end of House. In testing, I auditioned the WAV files using Acid Pro v4 and the set contains over 400 individual loops (approximately 270MB in total) spread across nearly 100 construction kits. In the main, each kit consists of between two and seven loops — although a few contain more — and a demo WAV is included with each kit showing one way the loops can be put together. The construction kit titles suggest that they are aimed at those with a more experimental side to their musical production; Insectoid, Atomize, Dr Demento, Mind Sweep and Offworld are typical examples — this is more Bill Laswell territory and not for those looking to complement their chart-orientated R&B or pop loop collections! These loops would suit a club/dance setting and, given the hard-edged, high-technology tone, they would also work well within a film or advertising context featuring urban, industrial, or futuristic images.


Latest Sample CDs

While there are exceptions, the vast majority of the loops are dance-orientated drum loops, originally recorded within the 110-140bpm range. However, this is far from being just another straight drum-loop collection. As well as a solid kick/snare-driven beat, the bulk of the loops contain other percussive elements and many feature some filtering or other processing. From synthy bleeps, vinyl scratches, over-the-top snare reverb, big booming kicks, or beatsynchronised filters, these loops generally feel quite busy, especially when used at high tempos. Indeed, in some contexts many of them would work as a musical bed with only a modest amount of anything layered over them — just enough not to worry about the terms of the licence! Within each construction kit, many of the loops provide variations around a consistent style of rhythm and, given how full-sounding many of the loops are, it is actually easier to sequence them rather than layer them. However, if you do want to layer some material over these beds, there is little in the way of strong melody or chord sequences within the collection — some mixing and matching from other libraries or live playing would be required to add these elements. While I'm all for the multiple formats that many modern sample sets seem to be supplied in, I'd hesitate to say that the quantity of material supplied in Electro Magnetic Pulse represented a bargain in the current market. However, the quality of what is here is really high and, having experimented with the loops for the purpose of this review, their creative potential is obvious — good stuff! John Walden Audio CD and WAV and REX CD-ROM 3-CD set, £62.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.bigfishaudio.com www.cybersoundmusic.com

SAM Solo Sessions **** MULTI-FORMAT Project SAM's high-quality, low-cost orchestral brass libraries have consistently put smiles on composers' faces. Their debut release, the versatile SAM Horns, was followed by two five-star titles, SAM Trombones and SAM Trumpets, forming a comprehensive trilogy of concert-hall ensemble recordings. To complete the full orchestral set, six solo brass instruments (including a tuba) have now been sampled and released under the title SAM Solo Sessions. file:///H|/SOS%2004-10/Latest%20Sample%20CDs.htm (3 of 6)9/26/2005 12:35:45 AM

Latest Sample CDs

With its Giga edition weighing in at 9.68GB, this is Project SAM's biggest library yet. An intensively sampled 'B'-flat solo trumpet sets the pace, working through a list of deliveries which includes non-vibrato long notes with a subtle volume swell, soft vibrato sustains, crescendos, and diminuendos. The player also performs a selection of Project SAM's trademark 'intervals' (slowish, upward legato movements), short, fanfare-like phrases, machine-gun-like 16th notes, and upward octave runs. Some of these samples are duplicated in SAM Trumpets; all are strong, controlled, and confident sounding. Sounding pure, bright and clean in the big hall, the well-played piccolo trumpet's staccatissimos steal the show — these lively articulations are a joy to play, and work very well in rhythmic passages. The solo trombone's four-dynamic long notes and staccatos sound rich and punchy when layered together in the 'hard sustain' program. SAM Horns' four-horn ensemble is impressive, but this library's solo horn is more precise and focused — its steady, polished delivery is ideal for melodies and chords, and its marcatos make a big, commanding concert-hall noise. The cimbasso, a rare, mad-looking contraption, turns in some insanely low, ferociously powerful bass notes, as well as more subtle and mellow horn-like sonorities. Last up is the tuba, and this one's a great all-rounder. For some reason its 'Klezmer staccatos' reminded me of Vivian Stanshall's masterpiece 'Do The Strain', but on a less scatological note its four-dynamic sustains can deliver both expressive, mournful melodies and apocalyptic bass trumps. The tuba's top notes suffer from some coarse tuning errors, which I imagine Project SAM will fix in a web update. As in SAM Trumpets, the samples are presented in a choice of three listening perspectives. The Utrecht concert hall sounds great, adding a delectable, floaty reverb throughout. However, there's a potential problem using sustains (none of which are looped) with release triggers — if notes are held too long, the samples will expire, creating an embarrassing gap between notes and their reverb! There are no trills or glissandi, but there are six useful programmed ensembles, plus some effects: not the unhinged sonic treatments found in SAM Trombones and SAM Trumpets, but low-key, low-pitched instrumental noises. Composers can keep on smiling — this is a worthy companion to Project SAM's brass ensembles. Dave Stewart 24-bit EXS24, Kontakt, and Halion 2-DVD-ROM set, 149.99 Euros; 16-bit Gigastudio DVDROM 149.99 Euros. Prices include VAT. Project SAM +31 (0)30 231 4500. +31 (0)30 230 4658.


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Click here to email www.projectsam.com

Saxophones 1 ***** GIGASTUDIO/EXS24 MKII The sax is a notoriously difficult item to sample — its sound is fluid, mobile, and dynamic, its timbre and repertoire unpredictable and dramatically varied. Worse still, the instrument's traditional exposed solo role renders any shortcomings horribly obvious. All this makes the saxophone a very difficult moving target for samplists; several sound companies have taken aim in the past, but most have missed by some distance. VSL's Saxophones 1 (6.1GB) trains its sights on the soprano and tenor saxes. Played by Robert Bernhard, the multisamples were pristinely recorded in the company's Silent Stage, and duplicate the elaborate performance variations of VSL's orchestral solo winds. (For details, see the review of VSL Pro Edition in SOS March 2004.) The soprano's long notes are beautifully played, with a gorgeous warm tone and lyrical vibrato infusing the samples with real feeling. You'll want to improvise for hours with these beautifully in-tune legato notes, which sound equally poignant delivering melody lines or quiet chords. My only criticism is that, while the instrument's soft-sounding, lower three dynamic layers are tonally homogeneous, it's quite a shock when the fourth layer of bright, piercing loud samples leaps in. Bridging this timbral gap would require the insertion of another couple of dynamic layers. Mr Bernhard plays his tenor sax just as nicely as the soprano — the tenor's bigger sound offers more opportunity for fruity staccato honks and free-booting blasts, but the quiet notes remain sensitively played, warm, and inviting to the listener. Overall, the delivery is quite controlled; there are no screaming 'Roadrunner'-style outbursts, licks, or phrases, but one cute programming touch adds long or short 'falls' to the end of the saxes' long notes in the form of release triggers — very effective for jazzy brass stabs. Making VSL's brilliant Performance Tool work in conjunction with their Horizon titles requires some tedious fiddling about with 'on-line activation wizards', but it's worth the effort. Hearing the two saxes liquidly moving from note to note with realistic slurs and slides is very gratifying — thanks to the Tool, this is the first time a sampled sax has been able to satisfactorily 'join up' legato notes. The huge selection of note repetitions, also managed by the Tool, is another bonus. Only one downside: inexplicably, the soprano's loud performance legatos were file:///H|/SOS%2004-10/Latest%20Sample%20CDs.htm (5 of 6)9/26/2005 12:35:45 AM

Latest Sample CDs

delivered with built-in portamento slides, whereas the quiet versions were played straight. A rare mistake. While one could quibble at a couple of other stylistic decisions (conventional trills are omitted in favour of accelerating ones; fast octave runs use chromatic and whole-tone scales rather than major and minor), the inescapable conclusion is that this is a laudable, great-sounding piece of work of enormous musical depth. Dave Stewart EXS24 MkII and Gigastudio DVD-ROM, £177 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.viennasymphoniclibrary.com Published in SOS October 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



M Audio Firewire 1814

In this article:

On The Outside Firewire 1814 Brief Specifications Drivers Rotary Controller Console Utility Hot-plugging Not So Hot In Use Final Thoughts

M Audio Firewire 1814 £449 pros Versatile I/O including eight analogue inputs and four outputs, plus eight-channel ADAT support. Balanced option on analogue outputs. Separate switching of digital in and out between S/PDIF and ADAT formats. Handy A/B headphone cueing for live sets. Aux buss can be used to patch in an external effects unit.

cons No +4dBu option to connect more professional gear. No balanced option on analogue jack inputs.

summary M Audio's Firewire 1814 is a straightforward yet versatile interface that provides a unique set of features in a compact package at a good price, with some handy extras.

M Audio Firewire 1814 Firewire Interface (PC & Mac) Published in SOS October 2004 Print article : Close window

Reviews : Computer Recording System

M Audio's half-rack interface provides a comprehensive selection of analogue and digital I/O at a competitive price. Martin Walker

M Audio proved that Firewire audio interfaces don't have to be expensive with their £349 Firewire 410, which I reviewed in SOS March 2004 (read the review on-line at www.soundonsound. com/sos/mar04/articles/maudio410.htm). This features two analogue inputs with mic/instrument/line level options, eight analogue outputs, plus both co-axial and optical S/PDIF and MIDI in and out, and has proved very popular. They made the point even more forcibly with their £229 Firewire Audiophile, which was reviewed by Mike Watkinson just two issues later in SOS May 2004 (www.soundonsound. com/sos/may04/articles/maudiofirewireap.htm). This again provides two analogue inputs, but this time without the mic/instrument preamps, plus four analogue outputs, co-axial S/PDIF and MIDI In and Out. With its 18 inputs and 14 outputs, you might expect their rather more ambitious Firewire 1814 to be substantially more expensive, but it's still a very reasonable £449. This is largely because eight of those ins and outs are devoted to ADAT I/ O, although you still get eight analogue inputs, four analogue outputs and coaxial S/PDIF in and out, plus word clock and MIDI I/O.

information £449 including VAT. M Audio +44 (0)14 4241 6590. +44 (0)14 4224 6382. Click here to email www.maudio.co.uk

On The Outside In appearance the Firewire 1814 could easily be mistaken for the 410. It's the same height (1U) and depth (160mm), but slightly wider at 168mm. Some musicians might question why M Audio didn't adopt a full-width rack case,

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Test Spec Firewire 1814 Windows XP driver download version 1028. Intel Pentium 4C 2.8GHz processor with hyperthreading, Asus P4P800 Deluxe motherboard with Intel 865PE chip set running 800MHz front side buss, 1GB DDR400 RAM, and Windows XP with Service Pack 1.

especially since six of the digital sockets emerge from a rear-panel breakout cable rather than fitting on the panel itself. I suspect this is because M Audio sell the Firewire 1814 as part of their mobile interface range and want to keep it compact. However, you can bolt it to any universal rackmounting kit using two holes provided on the bottom of the case, and the appropriate screws are thoughtfully provided.

The front-panel layout is also very similar to the 410's. The mic/instrument inputs are on the same Neutrik Combi sockets, with a switchable 20dB pad, rotary gain control, signal and clip LEDs, plus globally switched +48V phantom power. Next to them are twin stereo headphone sockets, each with its own rotary encoder level control, and then signal 'blinkies' that show S/PDIF and ADAT in/out activity. Tested with Steinberg Next up are a further 12 'blinkies' for the eight analogue inputs and four outputs, Cubase SX 2.2 and Wavelab 5.0, Native Instruments Pro 53. and beneath these a software-assigned rotary encoder just like the 410's, intended for control of monitor levels. There's also a momentary A/B switch with associated LED that I'll come back to later, and the front panel is completed by a power switch and status indicator. The mic/instrument preamps can be switched in to defeat the first two unbalanced line input sockets on the rear panel, and there are a further six Like many Firewire interfaces, the 1814 can unbalanced analogue inputs on the be powered over the Firewire buss if your rear panel, making the 1814 far more computer provides the six-pin version of the versatile than the 410. However, it only interface, or using an optional adaptor. has half as many analogue outputs, although this time all four can be used balanced or unbalanced, which is a huge improvement for anyone who's ever run into ground-loop interference problems. All the rear-panel analogue I/O uses robust quarter-inch jacks, although like the jack inputs, these can only be run at the consumer -10dBV standard, making the FW1814 not the ideal interface if you have gear that uses the 'pro' +4dBU standard. The rear panel also features a pair of Toslink optical sockets for either ADAT or S/ PDIF use, and a pair of Firewire ports — one to attach the 1814 to your computer, and the other to chain further Firewire devices. The rear panel is completed by a 15-way D-type connector, from which a one-foot-long breakout cable provides in-line sockets for MIDI In and Out, co-axial S/PDIF and word clock I/O. Overall, the 1814 provides a versatile I/O complement for the price. With a small mixer or additional mic preamps, it would be suitable for recording a small band live, or you could use it to connect several hardware synths plus a vocal mic and guitar. It also provides plenty of opportunities for further expansion — you could, for instance, add an eight-way ADAT-compatible A-D/D-A converter to build the analogue I/O up to 16 in and 12 out. Like the Firewire 410 and Firewire Audiophile, the Firewire 1814 model is also bundled with both six-pin to six-pin and six-pin to four-pin Firewire cables, each file:///H|/SOS%2004-10/M%20Audio%20Firewire%201814.htm (2 of 8)9/26/2005 12:35:49 AM


one metre in length, so you can either plug it into a desktop or laptop computer. It can be powered from the Firewire buss if your computer provides a six-pin port, but the four-pin variety found on most laptops doesn't supply power, so a 12V, 1A DC wall-wart PSU is also supplied. M Audio's generous Delta Live software bundle is also included with the 1814, containing Propellerhead's Reason Adapted, Ableton's Delta Live!, plus a Pro Sessions sampler CD-ROM. You also get a printed Quick Start guide, and although the main manual is in PDF format on the driver CD-ROM it's an excellent read, describing loads of alternative setups.

Firewire 1814 Brief Specifications Sample rates: 44.1 to 192 kHz on all analogue outputs and analogue inputs 1/2; 44.1 to 96 kHz on analogue inputs 3 to 8. Analogue inputs: two, balanced XLR with switchable global +48V phantom power and 1.8k(omega) impedance, or unbalanced TS quarter-inch jack instrument with 500k (omega) impedance, both using mic preamp with up to 66dB gain plus optional 20dB pad, or unbalanced line-level TS jack at fixed -10dBV sensitivity and 10k(omega) impedance, plus six further identical unbalanced line-level TS jacks. Analogue outputs: four, balanced/unbalanced TS quarter-inch jack at -10dBV level, two headphone with individual level controls. Digital I/O: ADAT optical in and out switchable to S/PDIF optical, co-axial phono S/PDIF in and out (both S/PDIF outputs support AC3 and DTS formats), MIDI In and Out, two sixpin Firewire ports, word clock in and out (will sync to external word clock at sample rates up to 96kHz). Frequency response: 22Hz to 22kHz ±0.1dB at 48kHz sample rate, 22Hz to 80kHz ±1dB at 192kHz sample rate. Dynamic range: inputs 105dBA, outputs 107dBA. THD + Noise: less than 0.003% at -1dBFS. Dimensions: 10.5 x 7 x 1.875 inches. Weight: 3.44lbs.

Drivers Like many other soundcard manufacturers, M Audio have now abandoned Windows 98 with new products, and their Firewire interfaces require Windows 2000 SP4 or Windows XP SP1. On the Mac both OS 9.2.2 and OS 10.2.8 or later are supported, with the caveat that native Firewire ports are required (G3/G4 accelerator cards are not supported). When I visited the M Audio web site for the latest PC drivers, which are shared by all three Firewire products in the range, version 1.0.2.8 was just a few weeks old, and I'm pleased to report that M Audio are still active in providing feedback about driver issues to their users. Since I reviewed the Firewire 410 in the March issue, various improvements have already been implemented.



One specific item is worth mentioning in the case of the 1814: its ADAT ports rely on an 'interleaved' multi-channel driver to avoid potential problems with large numbers of individual ins and outs in some versions of Windows. Most modern audio applications don't have any problem with this, but some using the MME or WDM modes might only let you access ADAT channels 1/2. To install the Firewire 1814, then, you run M Audio's driver software file first, then power down your computer, connect the 1814 to a convenient Firewire port, and then power back up. The new hardware is then detected automatically and the already existing drivers for it are installed, and a new icon appears on the Windows Taskbar to launch the Control Panel utility. I experienced no problems during the install.

Rotary Controller As with the other members of the M Audio Firewire range, the FW1814's frontpanel level controller can be assigned to various functions — 'sw return bus', 'output bus', 'input bus' or 'aux send' — but the 'phones' option has gone, since the 1814 already provides dedicated level controls for each headphone output on the front panel. Additional 'ctrl' buttons appear on all the relevant channels as you change this assignment so you can choose which channels follow the level controller's position, and your choices are remembered between assignments. The controller itself is far easier to use than its predecessors, being placed on the front panel so you can get at it easily, and while still a little jerky, the feedback of its current position to the software faders it's controlling is still a big improvement.

Console Utility The 1814's FW Control Panel is identical in many respects to those of both the 410 and Audiophile models already described in some detail in SOS March and May 2004, with each stereo channel in the Mixer page providing a pair of level faders with associated peak-reading meters, a pair of Aux sends, Solo, Mute and Link buttons, plus various routing options. This time there are just two pairs of analogue software return channels, routed by default to their corresponding physical output socket pairs (although you can send them to either or both using the two routing buttons below the level faders), plus four pairs of analogue input channels with additional pan controls. On the Output page you'll find two pairs of analogue outputs that can listen to either the main mixer output or the Aux buss, the Aux buss master controls, and two separate pairs of headphone outputs, which can listen to outputs 1/2, 3/4, or the Aux buss. The Aux buss will commonly be used to set up a separate monitor mix, but it can also be pressed into service to add effects to live input signals. This allows you to audition input signals with effects while using ASIO Direct Monitoring, provided you have an external hardware effects unit that can be patched in using S/PDIF or a spare pair of analogue ins and outs. M Audio



provide detailed setup instructions in the well-written PDF manual. The Hardware page provides the option to switch the optical I/O between ADAT and S/PDIF formats; clicking on the indicator to switch from one to the other didn't result in any visible change for about nine seconds on my PC, but The Firewire 1814 Mixer provides plenty of eventually the indicator does change versatility, especially since the Aux buss can state and display your choice. not only be used to set up a separate However, as you might expect, you monitor mix, but also to patch in an external hardware effects unit. In this screenshot you can't change any of these settings at can also see the eight-channel metering of all if an audio application is using the the ADAT option. interface. When the digital input is set to S/PDIF you can choose whether to use the optical or co-axial input port, and when either the input or output is set to ADAT an additional ADAT S/MUX option appears so you can switch from eightchannel to four-channel operation, which you must do when running at 88.2 or 96 kHz sample rates — you should select your sample rate first via your audio application, and then enable S/MUX. The Sync Source section has two additional options for 'word clock' and 'internal — digital signals muted'. The latter means that the 1814 is using its internal clock and ignoring any incoming signals on the digital input socket, and is M Audio's answer to the intermittent clicks that a few users experienced with the 410 in this situation. The original 'internal — digital signals unmuted' mode is now only for use when you want to lock an external device to the 1814 clock and also receive a return signal through the digital inputs. Since I last looked at the range there's also a new section for 'monitoring settings', where you can if you wish disable ASIO Direct Monitoring altogether, and set up the controls associated with the front panel A/B button and indicator. This button can either toggle direct monitoring on/off, or switch headphone output 1 between two sources, which are separately set after clicking on the 'set source' button, the options being 1/2 out, 3/4 out or aux. This is great for A/B cueing when working with two decks (or one deck and a CD player), or when mixing live. As with all DSP-based interfaces, some functions inevitably disappear as you raise the sample rate and processing overhead increases. In the case of the 1814 the entire Mixer page apart from its meters becomes greyed out at 88.2kHz and above, while the Aux buss similarly goes from the output page.



Hot-plugging Not So Hot M Audio include a red printed sheet in six languages in the FW1814 box, plus related messages during the driver install and in the PDF manual, warning that in some isolated cases connecting a Firewire device into a system that's already powered up (hot-plugging) has caused permanent damage to either the peripheral or the host computer's port. M Audio are in no way to blame for this (they adhere strictly to the Firewire industry standards), but because of this possibility you should always power down your computer before plugging in any Firewire device, just in case. This is a huge shame, as hot-plugging should be one of the strengths of the Firewire standard.

In Use Because the 1814 has balanced outputs, I could use balanced cables to connect it to my mixing desk, so there were no ground-loop problems — as a check I tried replacing them with unbalanced ones, and I could immediately hear mouse and graphics card noises in the background, just as I could with the FW410. Unlike some unscrupulous manufacturers, M Audio quote a real-world test spec rather than simply reprinting the theoretical figures for the converters, and using Rightmark's Audio Analyser I measured a very good dynamic range of 109dBA at both 32-bit/44.1kHz and 32-bit/96kHz, and a frequency response flat to within ±0.1dB from 16Hz to 20kHz, opening out to -0.5dB at 41kHz with the 96kHz sample rate, in line with M Audio's own figures. The Firewire 410 only has 192kHz-capable outputs, but the 1814 can do 192kHz in and out, so I was also able to perform loopback tests at 24-bit/192kHz. Once again the dynamic range measured 109dBA, while the frequency response extended to -1dB points of 9Hz and 61kHz. To audition audio quality I tried some blind tests against my Echo Mia and Emu 1820M. After matching the output levels of all three interfaces to within 0.1dB using my mixer, I used the ASIO Multimedia drivers inside Cubase SX 2.2 to run all three side by side, created three identical audio tracks, each routed to a different card, soloed one of them, and then after hiding the Cubase Inspector so I couldn't see their output routings, got someone else to thoroughly 'shuffle' the tracks by repeatedly dragging them up or down.

The Hardware page of the 1814's Control Panel utility provides individual switching of the digital input and output formats, plus an extremely useful A/B switching function in conjunction with the associated front-panel button, for headphone cueing when recording or in live sets.

Then I sat back and auditioned my normal wide range of audition material covering vocals, classical, jazz, rock and dance, to see if I could pick out any file:///H|/SOS%2004-10/M%20Audio%20Firewire%201814.htm (6 of 8)9/26/2005 12:35:49 AM


differences blind. All three interfaces sounded very good, but although the differences were subtle, each time the cards were reshuffled I could eventually pick out Emu's 1820M as having a slightly more 'focused' sound, while I couldn't tell the Firewire 1814 apart from the Mia. The difference was particularly noticeable with distant reverbed sounds, and presumably due to the 1820M having a lower jitter clock. Nevertheless, M Audio's Firewire 1814 still provides high audio quality — on a par with most other interfaces under £500 — and as with the 410 I had no problems running it with Cubase SX 2.2 using its ASIO drivers set to an incredibly low 1.5ms latency at 44.1kHz, while Play Ahead settings within NI's Pro 53 on my PC were a good 25ms for Direct Sound.

Final Thoughts The Firewire 1814 is a versatile interface that should find many applications, including live recording of small bands, and as the nerve centre of a small studio where you need to interface a wide variety of digital gear. Its balanced outputs are also very handy, although some potential users may still run into ground loop problems when connecting multiple hardware synths to its unbalanced inputs. One obvious Firewire-based competitor is the widely advertised yet sadly still invisible Hercules 1612 FW, at a very similar £439. Once this does finally get released it will provide 10-in/eight-out balanced analogue, a similar pair of mic/ line input options, two MIDI Ins and Outs rather than one, plus co-axial and optical S/PDIF I/O and word clock in and out, though there's no ADAT support. However, its performance is as yet untested, so I'll make no further comment. If you want to buy now, Edirol's FA101 doesn't have ADAT or co-axial S/PDIF I/ O, but provides eight analogue ins and outs including two mic inputs (one with an optional high-impedance setting for guitar use), plus S/PDIF optical and MIDI I/O for a very similar price of £429. Like the Firewire 1814 it can run up to 192kHz with a reduced number of channels, but all its analogue I/O can be balanced and run at +4dBu levels. Anyone looking for a compact Firewire audio interface is almost bound to place M Audio's Firewire 1814 on their shortlist, and if you want eight analogue inputs and ADAT support, it's the only option I've come across at under £500. Little extras like the Aux buss for monitoring and adding external hardware effects plus the A/B cueing facility should also raise a few smiles, and the Firewire 1814 will appeal to anyone who wants more analogue inputs and far more versatility than its 410 stablemate, for not a lot more money. Published in SOS October 2004







In this article:

Librarians & Editors No Driver Required More Than Just 88 Notes Factory Presets Zoning & Grouping In Short...

M Audio Keystation Pro 88 USB MIDI Controller Keyboard Published in SOS October 2004 Print article : Close window

Reviews : MIDI Controller

M Audio Keystation Pro 88 £399 pros Great price for what's on offer. Class-compliant plug-andplay with OS X and Windows XP. Full complement of different hardware controllers, with refinements like Drawbar mode and Snapshot.

The price of compact MIDI controller keyboards has come tumbling down in the past few years, but weighted-action full-length keyboards have remained relatively expensive — until now. We investigate the 88-note keyboard that runs off USB power... Paul Wiffen

cons

In recent years, the main focus for makers of master keyboards has been on smaller controllers with between 1.5 and three octaves. Perhaps the number of classically trained piano summary players out there is diminishing, or An authentic full-length perhaps there are more younger weighted-action keyboard in sturdy stylish case at a plastic musicians who have not trained on 61-note price — and it acoustic pianos, but whatever the integrates well with your cause, the small MIDI keyboard with a studio computer, too. host of assignable knobs which allow Photos: Mike Cameron information you to twiddle and edit your synth £399 including VAT. parameters has become one of the M Audio +44 (0)1442 strongest-selling items in most music stores over the last 10 years. 416590. Manufacturers like Midiman, Edirol and Evolution have built much of their recent +44 (0)1442 246832. success on the back of products like the Oxygen 8 and the MK249C, which Click here to email combine a limited-size non-weighted keyboard with octave-transpose buttons www.maudio.co.uk and a bunch of assignable knobs, sliders and buttons. No master volume control. Footswitch polarity must be reversed at every power-up (if necessary).

www.m-audio.com

In contrast, the majority of manufacturers seem to be steering away from larger, weighted-action master keyboards. However, I am happy to report that M Audio, the US outfit which recently acquired UK company Evolution, is bucking this trend by releasing the Keystation Pro 88, a fully weighted 88-note master keyboard with USB MIDI class-compliance as well as traditional MIDI hardware connectivity. Housed in a smart silver casing, with a large blue LCD screen to

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keep you informed of everything that is going on as well as a forest of knobs, buttons and faders, it really looks the part, too. In the past, many keyboards designed for use with a computer have had a rather cheap plastic look which belies their excellent feature set. The same cannot be said for the Keystation Pro 88, which has a very sturdy construction and looks as though it should stand up well to the rigours of life on the road. It has five rubber feet (two at each end and one in the centre of the base), so if you don't have a keyboard stand available, you can still put the Keystation on a table without fear of the screw heads in its base causing damage to the tabletop.

Librarians & Editors The Keystation's 10 programs (see the 'Factory Presets' box at the end of this article) may seem a little limited for a keyboard of this size and flexibility, but if you have a computer, you can use it to increase the number of sets you have available. The supplied CD-ROM comes with a PC librarian only, but I was delighted to discover that M Audio have now posted a v1.1 update for the Enigma Mac OS X librarian on their web site, and this M Audio's existing Enigma editing/librarian allows the software to be used with software has now been updated to work with the Keystation Pro 88. There are the Keystation Pro 88. now a couple of extra windows (see below) to display the many additional parameters and the new zones which the Keystation implements. It's great to be able to see the Zones at a glance on screen, and similarly, the Zone Editor window allows you to see all the parameters for each zone instead of having to dial them up one at a time in the Keystation's LCD. The two windows interact in real time, so if you drag the upper limit of a zone on the graphic display, the box in the parameter display updates. I strongly recommend programming all Zone stuff this way and then transmitting it across to the keyboard.

No Driver Required Small MIDI keyboards which take their power from their USB connection have become quite common of late, but it is quite something to play an 88-note weighted mother keyboard that doesn't need to be plugged into the mains. As soon as I plugged the Keystation into my Powerbook, the backlit LCD display sprang to life. You should however make sure that your laptop's power supply is plugged in, as the Keystation, like many peripherals for portable computers, drains the laptop battery much faster than the computer does on its own. I suspect that the majority of purchasers of the Keystation will be using it with a



computer, and therefore don't require a separate power supply. M Audio certainly don't supply an external PSU with the unit, even though there is a 9V DC Input around the back for a standard PSU, if you should need one (for use with an older non-USB-compatible computer, for example). Those of you who read my review of the Evolution MK461C master keyboard and its brethren in the June issue of SOS (see www.soundonsound.com/sos/jun04/ articles/evolutionmk461c.htm) will be aware that Evolution are the masters of the Plug and Play class-compliant USB MIDI device — no drivers are needed to make their products work with OS X on the Mac or Windows XP. You may also remember my heartfelt wish that the acquisition of Evolution by M Audio might lead to M Audio's own products becoming as easy to set up and configure. I am happy to report that the Keystation Pro 88 is just such an M Audio product. Drivers are supplied for Windows 98/ ME and Mac OS 9.x users, but OS X and XP users don't need them. Plug the Keystation Pro in via USB with Logic open under OS X, as I did, and a dialogue box immediately reports an extra available two MIDI Ins and one MIDI Out. This is because in addition to the keyboard itself, the Keystation also features one of the few things I felt was missing from the Evolution MK400-series controllers, namely an additional hardware MIDI Input on the back panel. This means that you can plug another MIDI device into the Keystation, and have all its MIDI data output going into the computer as well as the amazing number of controllers from the Keystation itself. This is actually done as a second USB MIDI Port, allowing the MIDI In data and that of the keyboard to be routed to different destinations inside your computer. The keyboard itself has a very nice feel; it's not too heavy and fatiguing to play, but nor is it too light, so you still get that 'resistance' from the keys which is so necessary for triggering a real piano sound. The feel has a major impact on how you play a piano sound; within hours of receiving the review unit, I was using it very successfully for a rehearsal at which the rest of the band said how good the piano was sounding. The following week, when I used the unweighted 461C instead, there were comments about it not sounding so good! On both occasions, I was of course triggering the same EXS24 sample in my Powerbook. The only master keyboard I have ever played which feels more natural for piano than the M Audio is the Kurzweil PC88, and on that instrument, the piano sound is built in, and has been fine-tuned to respond perfectly to the action. But the Keystation's natural feel is remarkable, especially at its £399 price point. If you wish to fiddle with the response, there are nine different velocity curves which can be selected, but I have to say I found the default setting (a simple linear one) was just fine.

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In addition to the 88 weighted wooden keys, the Keystation features no less than 24 fully assignable rotary knobs, 22 assignable buttons and nine assignable 60mm faders (which have the same drawbar-invert function as on the MK461C and MK449C keyboards and the UC33 controller box). Add to this two wheels (pitch-bend and modulation), five transport buttons, two footswitch inputs and a volume (expression) pedal input, all of which are also fully assignable, and you end up with a pretty impressive package. With 24 rotary knobs, and 22 buttons, I felt as though I had returned to the old days of keyboard playing, when every parameter had a control of its own — even when playing virtual synths. Previously with controller keyboards, I have always found that I either had to pick which 10 or 16 parameters I would need to access most often, or create several different banks of controls, for example one for editing the filter and LFO and another for the envelopes. With the number of controls on the Keystation, you can have everything you need at your fingertips, whatever virtual device you are triggering. The controls of the instrument are essentially an expanded and refined version of those on the MK461C. Above your left hand are the nine Function keys labelled Global Channel, Channel Assign, Control Assign, Program, Data LSB, Data MSB, Store, Recall and Zone Range above the corresponding button. A single press activates each of these, and then you can use the numeric keypad or the '+' and '-' buttons to enter or adjust the value shown in the display for the next three seconds (after which the unit returns to normal operation). In addition to these most commonly used nine functions, another seven are available via dual key-presses, and these combinations are clearly labelled below the buttons as Control Mute, Control Select, Drawbar, MIDI Out from USB, Velocity Curve, Device ID and Memory Dump, so you don't have the brainache of trying to remember which dual key-presses activate which functions. Most of these functions are selfexplanatory, but a few are worth looking at in a bit more detail. Drawbar, in particular, is worth its weight in gold to organ players, as it instantly inverts all the 60mm faders so that they send their lowest value at the top of their travel and the highest at the bottom, thereby simulating exactly the way drawbars functioned on Hammond organs. Although I have seen other products offer this function, in my experience you have to invert each slider individually.

Despite its affordability, the Keystation Pro 88 has plenty of assignable hardware controls on offer (on both rotaries and faders — see right and far right), plus comprehensive zoning facilities and MIDI message transmission options.

MIDI Out from USB is another useful function. This allows the Keystation to act as a USB MIDI Interface for other devices by sending out whatever MIDI file:///H|/SOS%2004-10/M%20Audio%20Keystation%20Pro%2088.htm (4 of 7)9/26/2005 12:35:56 AM


messages come down the USB cable from the computer via the Keystation's MIDI Out. The reverse is also true — data coming to the MIDI In port is automatically sent up the USB cable to the computer. Without this option activated, the data sent from the MIDI Out is what the Keystation itself is generating (so that it can be used as a Master keyboard even when there is no computer in the system). When it is activated, the Keystation no longer sends its own controller information out of the MIDI Out. This may seem confusing at first, but if you think of it in a similar way to Local control on a keyboard which does make sounds, it becomes simple. Whenever you are using a computer, you should set the MIDI Out from USB to On, and whenever there is no computer, you must set MIDI Out from USB to Off, or the Keystation will not trigger external modules. The Octave Up and Down buttons double as a transpose parameter when pressed together and the Snapshot function (which should be familiar to MK461C users, and sends out the current position of every knob, slider and button) is activated by pressing the '+' and '-' buttons together. Again, good panel labelling means that you do not have to hunt in the manual to find out how to do these things. In fact, the only thing which isn't on the front panel is what to do if you have a footswitch with the opposite polarity to the Keystation's programmed default. I needed the manual for this because it isn't done via the front panel! All you have to do is hold the offending footswitch down as you power up the Keystation and it takes this as the default off position. This does mean, however, that you have to remember to do this every time you power up the Keystation, because it can't be memorised.

Factory Presets The Keystation comes with its 10 memories pre-loaded with factory setups, which can be reloaded at any time by holding down the '+' and '-' buttons as you turn the unit on. These contain typical setups which are likely to be the most useful to the user. So 01 is a General MIDI Preset, 02 a GM mixer preset, and 03 a GM2/XG preset. Then we get away from GM; memory 04 is a Propellerhead Reason Mixer and 05 is a Reason Instrument. My favourite is the Native Instruments B4 in 06. 07 and 08 contain Cubase SX/Nuendo and Logic presets respectively, and 09 has a Soundblaster synth setup. While it is unlikely that more than one or two of these will prove instantly useful, the amount of work that has gone into those one or two will save you a lot of programming time. Even if you don't have the instruments concerned, it is a lot quicker to adapt an existing preset than program one from scratch. For example, I was able to adapt the drawbar-based B4 preset into one for Emagic's EVB3 in about 10 minutes.



I had some trouble getting the transport controls to work with Emagic's Logic, even when I loaded the 'Pro 88 Logic Preset.LSO' file on the supplied CD-ROM into Logic. I contacted the guys who designed and set up the Keystation here in the UK at the old Evolution offices, and between us we managed to work out what was going wrong. First of all, you need to enter the appropriate MIDI key commands in Logic's user preferences. These cannot be updated with an LSO file, so the commands have to be entered using the Learn Key function in Logic (see the screenshot below for the list we used). Then we discovered that the transport keys were toggling, which meant that only each second press of each key was actually working. It turned out that I had an early version of the Logic preset before this had been fixed, but the more recent one has been put on the M Audio web site for anyone else that needs it. It took about 30 seconds to change the output commands so that the buttons were no longer toggling.

Zoning & Grouping Below the main Function buttons are additional buttons to allow control of the Zones and Groups which are a new feature on the Keystation Pro. Once you get to 88 notes, it is rare that you need to use the entire keyboard to play one sound except for the grand piano, which means that when triggering other instruments, you probably want to split the keyboard into various zones to control different timbres on different MIDI channels or even instruments. The Keystation Pro 88 allows for four different zones on the keyboard, each of which can have its own associated range, MIDI Channel, program change and transpose settings. Additionally, you can decide whether controllers like the pedals and wheels affect each zone individually. These Zones can be activated and deactivated during performance without the need to change programs. It means that you can essentially keep four different sub-programs in each main program (say a piano over the whole keyboard, a bass sound in the left hand and strings and brass layered in the right hand) and then mix and match them as required. Deciding which are active at any given time is the job of the four buttons just below the main function buttons. Each has an LED associated with it so that you can see at a glance which zones are active at any time. The Zone buttons have a secondary function which I have not seen before — that of activating the three different Groups of controllers. Group A comprises the wheels, pedals and transport controls, B covers the 24 rotary knobs and the nine buttons beneath them, and Group C includes the nine 60mm sliders and the nine buttons beneath them. Each controller is labelled on the panel with the group letter, as well as its own individual number in the format A35, C3 or B46, so can you see immediately which group a controller belongs to. Now you might wonder why you would want to deactivate groups of controllers. The main reason is that when you recall a program, you can choose to only load part of its controller set, leaving other controllers still left on the previous assignments. For example, you may be only using the slider and buttons beneath them to control an organ sound for Native Instruments' B4, but may want to use file:///H|/SOS%2004-10/M%20Audio%20Keystation%20Pro%2088.htm (6 of 7)9/26/2005 12:35:56 AM


the 24 rotary knobs for a parameter-heavy synth like Gmedia's impOSCar. If you deselect Group C (the sliders and buttons) once you have the B4 program loaded before recalling the program for impOSCar, then the synth controller assignments for the 24 knobs and their nine buttons will be loaded in, but the nine sliders and their buttons will remain assigned to their B4 parameters. Then, by deselecting both Groups B and C, you could load the transport controls for Logic into Group A, but keep the synth and organ controls on B and C respectively. Of course, it does require a fairly organised mind to keep track of what you are controlling where, but you are not forced to use this facility if you don't wish to. An extra bonus is that when you switch the unit off, it retains whatever is in the current setup as an additional memory, so you can have a complicated multi-Group selection available to you at power-up.

In Short... The only thing I don't like on the Keystation Pro is the same thing I mentioned at the end of my MK461C review — there's no hard-wired master volume control here, either! But as I said in that review, this is really only an issue if you do a lot of playing live, and need to turn down an incorrect sound or stuck note in a hurry. In the studio, this should never be a problem. However you look at it, the Keystation Pro 88 respresents tremendous value for money. Whilst Fatar master keyboards of this size have been available for well under a thousand pounds, they have not had the numbers and flexibility of controllers that this unit has. In fact, no keyboard this size has been able to work directly with computers via USB before, so that in itself makes it unique. Its primary raison d'être, as a hammer-action 88-note keyboard, more than justifies the purchase price on its own, but when you add in all the knobs, sliders, buttons, wheels and pedals, you really do have the ultimate controller keyboard at a steal of a price. Published in SOS October 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



Mackie Onyx 1620

In this article:

Construction & Design Mono & Stereo Channels Direct Outputs The Sound Black Beauty

Mackie Onyx 1620 Mixer Published in SOS October 2004 Print article : Close window

Reviews : Mixer

Mackie Onyx £769 pros Warm, musical sound. Low noise and wide dynamic range. Fuss-free Firewire interface.

cons No mono button. Analogue direct outputs on D-Sub connectors.

summary The Onyx is a very nice little mixer as it stands, but with the addition of the Firewire card it provides a very worthy front end and monitoring controller to any DAW that supports the Firewire protocol and that runs one of the supported operating systems.

information Onyx 1620, £769; optional Firewire interface module, £439. Prices include VAT. Mackie UK +44 (0)1268 571212. +44 (0)1268 570809. Click here to email

Mackie combine a completely redesigned 16:2 mixer with a multi-channel Firewire audio interface. What more could you want? Paul White

This new straightforward but highquality mixer incorporates A-D and D-A conversion plus a Firewire audio interface to transmit both the channel direct outputs and the main stereo output to your computer. However, in order to avoid making the mixer prohibitively expensive for those users who didn't need digital I/O, Mackie have made the Firewire interface an optional extra that can apparently be Photos: Mark Ewing installed by the user simply by plugging it in and installing the driver software. This is a very sound idea and, although when I started this review no firm date had been set for the release of the Firewire card, Mackie managed to get one to me just before we hit deadline. This interface works at up to 24-bit/96kHz and can handle 16 inputs and two outputs simultaneously.

www.mackie.com

Test Spec Apple G5 dual 2GHz with 2GB RAM, running OS 10.3.4.

The clear advantage of having a built-in Firewire interface is that you no longer need to worry about your computer's audio interface having enough capacity to handle all the inputs you might wish to record. Up to 16 inputs from the mixer (plus the main mixed output, taken pre-fader) are converted into 24-bit digital audio and then sent directly to your audio software's input via your computer's Firewire port. A gain trim control on the interface card allows the user to adjust the recording level sent to the DAW from the pre-fader stereo output. ASIO for Windows XP and Core Audio for Mac OS 10.3 or above are supported (although Windows PCs can connect to multiple desks where Macs can only handle one), and the Mackie Tracktion sequencer is bundled with the interface. Although the

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Mackie Onyx 1620

return path from the computer is only stereo, the vast majority of people now mix within their DAW software so as to be able to make use of processing plug-ins, so this isn't a serious limitation.

Construction & Design As you'd expect from a Mackie mixer, the Onyx has a solid-steel construction and uses all-metal jack sockets, where the majority of the connectors are mounted on the top panel for easy access — D-Subs, inserts, aux sends, and power inlet are on the rear, as are the XLR main outputs and the external talkback mic input. A long covered slot allows for the installation of the Firewire card, and power comes directly from the mains rather than from an external PSU, accepting any standard international supply voltage without the need to switch. The overall styling, though a departure from the traditional Mackie livery, is still an exercise in clarity and functionality. The rounded edges, light fader area, and perforated rear section make the appearance much less square and formal than usual, while the rubber-edged end cheeks give the mixer a very purposeful look. Although Mackie products are now built outside the US, the design of this particular mixer was apparently supervised by Greg Mackie and, overall, I have to say I like this new style. The mixer has a simple dual-buss 16:2 design that uses a new generation of Mackie mic preamp claimed to be their best design yet. Certainly the paper spec is good, with a frequency response extending from 20Hz to 50kHz within 1dB, and it is only 3dB down at 100kHz. The mic preamp noise is also low at -129.5dB with a 60dB gain setting, while the mic amp distortion is below 0.0007 percent. The design has evolved from the existing XDR mic preamp used in the VLZ Pro mixers. The gain range is wide and can accommodate inputs as high as +22dB line level. To maintain maximum flexibility, channels one and two include switchable highimpedance instrument inputs (selectable via buttons in the preamp panel) as well as mic inputs. Each of the eight mono mic/line channels has individual phantompower switching, as compared with the more usual budget solution of making this global, or at best having it switched for groups of channels. All channels have an input gain trim control, and gain trim adjustment can be undertaken using the Solo switches in PFL mode as usual, where the signal level shows up on the main meter section.


Mackie Onyx 1620

Mono & Stereo Channels As is becoming increasingly common, some of the channels (nine to 16 in this case) are arranged as stereo pairs, and as stereo EQ sections are more complex and expensive than their mono counterparts, the line-only mono channels have four-band EQ with sweepable mid-bands, while the stereo channels have threeband EQ, rather like the smaller mixers in the existing Mackie range, but based on new circuitry. EQ bypass is available on all channels, and the mono channels also have switchable low-cut filters and pre-EQ insert points, but no phase buttons. The EQ circuit is a new design by Cal Perkins, who I've often thought of as Mackie's 'fifth Beatle', as he was responsible for many of Mackie's circuitdesign innovations. All the EQ bands have a ±15dB gain range with high and low shelving filters at 80Hz and 12kHz. The stereo channels have their mid-bands set at 2.5kHz, while the mono channel midbands can be swept over the ranges 100Hz-2kHz and 400Hz-8kHz respectively. I'm glad to see that the lower mid-band control can be taken down into the upper bass regions where EQ is often required, and there's The first two jack inputs can accept guitar signals directly, but you need to make sure plenty of overlap between the two midthe Hi-Z mode is selected with the adjacent bands. Cal was the engineer behind button to provide a suitably high impedance. the XDR mic preamps, the HR-series monitors, and some aspects of earlier Mackie mixers, but in this instance his aim was to capture the classic EQ sound of British mixing consoles of the '60s and '70s. The design is based on the traditional Wein Bridge filter circuit, which may seem to offer less range than a modern EQ, but it should sound sweeter and more musical. At least, that's the plan! The circuit also includes new combining filters to reduce the amount of phase shift. Both mono and stereo channels have four aux sends (each switchable globally for pre-fade or post-fade operation) and simple routing or muting via Mackie's now familiar Alt button, which sends the channel signal to the stereo Alt buss as an alternative to the main stereo mix buss. If nothing is connected to the Alt outputs, then this is the same thing as having a Mute button. This Alt feature also makes the mixer more flexible in a conventional (analogue I/O) recording context, as it allows some mixer channels to be used as recording sources via the alt outs and direct outs while others, routed to the main mix, are used for monitoring. All channels include a 60mm fader, four-LED level metering and a Solo button in addition to the Alt/Mute button.

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Mackie Onyx 1620

One unusual feature of this mixer is that the channel direct outputs emerge as balanced line signals on a pair of 25-way D-Sub connectors on the back of the mixer, and not as separate jacks. While this undoubtedly saves both cost and space and puts all the outputs in one place for when the Firewire card is fitted, DSubs are a right royal pain in the proverbial if you decide to wire them yourself, and it means you're pretty much forced to use a patchbay with them. My best advice is to buy them ready-made, but at least the wiring protocol is the same as for Tascam's analogue D-Sub cables (not the TDIF digital ones) so they can be bought off the shelf. The Firewire option fits into the rear of the unit (a five minute job involving just four screws and a ribbon cable), where a Firewire button in the master section allows the stereo output from the computer audio system to be selected as a monitor source — other alternatives are the main stereo out, Alt, or Tape. A further button allows these sources (other than the main mix of course) to be routed directly into the main stereo mix. The tape inputs and outputs are on phono connectors, as is usual on mixers of this size, and right next to these connectors is a powered BNC socket for connecting an optional 12V lamp such as a Mackie's own Little Lite. The rest of the master section is pretty straightforward, with masters for all four aux sends and returns, pre/post switching for all four sends, and the option to route effects to monitor using aux three as the source. The main outputs are all balanced on TRS jacks, with the left/right mix outputs also being duplicated on balanced XLRs on the rear panel.

The Onyx's optional Firewire interface (below) allows you to transfer audio directly to and from your computer. Once the necessary drivers have been installed, the mixer transmits 18 analogue inputs and can also access a stereo monitoring signal from the computer via the master section's Monitor Source buttons (right).

An integral talkback mic with non-latching button and level control can talk to the control-room/headphones outputs and/or to the aux one and aux two outputs (useful for talking into stage or studio live-room monitors). There's also the option to use an external talkback mic if needed — an External Mic selector switch is included in the talkback controls. When talking to the control-room outputs, the control-room signal level drops by 20dB to make sure you are heard over the music. The master fader is a single 60mm, stereo fader which, like all the channel faders, has 10dB of additional gain above unity. That leaves a Solo Mode switch for selecting PFL or AFL, Mackie's famous Rude Solo lamp, and a stereo 12-section LED meter that also shows the level in PFL mode for gain trim setting. file:///H|/SOS%2004-10/Mackie%20Onyx%201620.htm (4 of 7)9/26/2005 12:35:59 AM

Mackie Onyx 1620

The Sound There's nothing at all complex or unusual about the way this little mixer operates, but how does it sound? To find out, I compared it with Hugh Robjohns' Mackie VLZ Pro mixer. I could detect very little subjective difference between the preamps when used with the same studio capacitor microphone — certainly both were very quiet, though if anything the Onyx had a marginally less hard sound. I'd have no worries about using these preamps for any normal kind of recording — to get something of this quality in a relatively inexpensive mixer is very good news indeed. It also means that, with the Firewire card installed, the Onyx would make a seriously decent front end for a computer-based recording system. Evaluating any EQ section is always more subjective, but it didn't take me long to come to the conclusion that the Onyx sounds noticeably sweeter than the EQ in the regular Mackie range. It's also more flexible than most of their smaller mixers, having two sweepable mid-bands, and I found that cranking the upper mid-band up to 8kHz and then adding just a little boost gave a nice sheen to vocals. I'd have happily foregone some of the overlap between ranges in exchange for the upper mid-band being extended further, up to 10-12kHz. However, there's still a lot of range, and the low mid-band goes down far enough so that the troublesome part of the lower mid-range (what I call the 'boxiness zone') is well covered. The shelving high-frequency control sounds perfectly smooth and predictable, but a 'bell' EQ is often nicer for adding air and breath to a vocal or instrument sound, hence my desire for more upper mid-band reach. At the low end, it was possible to use gentle boost to thicken the sound and add weight without clouding the end result in the way that the original Mackie EQ could if it was overused. While I'd recommend most small-desk EQs be used for cutting rather than boosting, this one sounds musical in boost mode too, provided that you don't get too carried away. A few tweaks with a budget capacitor mic enabled me to add air without sibilance, add weight to the low end, and clean up a mid-range vocal honk. This required only very small cut and boost adjustments, so Mackie saying that the EQ range has been curtailed in exchange for a more musical quality isn't really an issue, as there's still far more range than you could reasonably use in most situations. It's also good to have a proper hard EQ bypass so that you can see immediately if your settings have improved things. Tests with the EQ set to neutral showed negligible change in sound quality when the EQ was switched in and out. Installing the Firewire part of the system was surprisingly painless, and sound arriving through this interface sounded virtually indistinguishable from that arriving through the analogue outputs, to my ears. A driver CD-ROM comes with


Mackie Onyx 1620

the necessary PC driver plus a copy of the Mackie Tracktion sequencer as well as PDF manuals, but Mac users (I used Mac OS 10.3.4) don't even need drivers, as Core Audio welcomes the interface like an old friend. Onyx shows up as 18 input sources and two outputs, where inputs 17/18 carry the main left/right mix and one to 16 are the direct channel outs. However, you can only select 16 sources to record simultaneously, even though there are 18 possible sources. Buffer size is set as usual in the Core Audio panel and I had no problems inputting all 16 sources with a buffer size of 256. Smaller settings may well be possible, but much depends on the speed of your hard drive, especially if you are using a laptop — laptops tend to have slower drives than desktop machines. It should also not be overlooked that, because this is a mixer and not just an audio interface, it handles all the monitoring and source switching requirements of a typical small studio as well as providing basic talkback.

Black Beauty There's not much to dislike about the Onyx. With the Firewire interface installed it should make a great front end for any computer DAW system, but even without it manages to cram a lot of sonic class into a small package. Maybe some purists would like 100mm channel faders instead of 60mm models, but overall the mixer feels uncluttered and easy to navigate, which is also very important. The new mic amps are technically as good as or better than those in the VLZ Pro mixers, and have a slightly smoother, more flattering sound. However, the big subjective difference is in the EQ section, which, as well as having the benefit of two sweepable mid-bands, does have a smooth vintage character that flatters more overtly. I really like this EQ, but, equally, I like the hard bypass switch for when I don't need it! I feel that four aux sends is enough on a mixer of this size, especially as all can be switched pre- or post-fader in the master section. Although the master section itself is very basic, it seems to do just about everything you need, including allowing you to add effects to the monitor mix and providing effective talkback facilities. In fact, the only thing I really missed was a mono button, which, like the monitor Dim button, seems to have been left off to save on cost and space. The availability of a properly integrated Firewire card means that the Onyx can be patched into a DAW system using just a single Firewire cable, rather than using a bunch of analogue feeds, and the quality of the converters seems


A new master talkback section is located conveniently on the front edge of the mixer. A momentary action switch routes the output of a built-in microphone to the headphones and control-room outputs and/ or the first two aux busses, via a

Mackie Onyx 1620

dedicated level control. comparable with other decent, nonesoteric soundcards and interfaces I've tested. In fact the only trick Mackie seem to have missed is in not including S/ PDIF I/O, which could have been useful when recording from a digital source or mixing to a DAT or MD machine. I like the way the Onyx looks and feels, and, as you spend a lot of time staring at your mixer when working, that's also important!

If you need plenty of inputs for your computer system but tend to mix and process internally, then the combination of Onyx and its Firewire interface offers a very neat and flexible alternative to a separate mixer and audio interface. Ultimately, the Onyx is a very nicely designed, musical-sounding mixer that belies its compact size and heralds a slightly new direction for Mackie, both in terms of audio philosophy and styling. The designers have also recognised the importance of integration with computer-based digital workstations without forcing you to pay for this capability if it isn't required. Published in SOS October 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:

Evolution Not Revolution Data Entry System Requirements Palette Power Supporting Players Now On X Smartmusic Final(e) Thoughts The First One's Free...

MakeMusic! Finale 2004 £299 pros Speed Entry really is fast. Built-in Soundfonts for good audio proof. Exceptional value. Mac OS X compatible.

cons Would be nice if tempo was tracked during real-time recording. A matching 'human playback' that matched the above would also be welcome.

summary Finale 2004 may not be packed with total innovations, but it is a restatement of what the package is capable of. It's always tricky to choose between two market leaders, but if all else fails, check out the price tag!

information £299 including VAT. Et Cetera Distribution +44 (0)870 873 8731. +44 (0)870 873 8732. www.etcetera.co.uk

MakeMusic! Finale 2004 Music Notation Software (PC/Mac) Published in SOS October 2004 Print article : Close window

Reviews : Software

The latest incarnation of a well-established scoring package holds few surprises, but plenty of power for the asking price. Derek Johnson

Within any software milieu, there will be the main players, the enthusiastic wannabes and the also-rans. If this is true of MIDI sequencing software, it becomes even more so when we move into the relatively rarified atmosphere of scoring/notation software. Notation programs often seem superficially similar to standard MIDI sequencers, and many sequencers offer note input via score editors and various layout and printout options. But this aspect of MIDI sequencers isn't nearly fully enough developed for composers, arrangers, engravers and educators who simply don't need the audio processing and data manipulation typically offered by such programs. They'd rather see software development and processing power channelled to aiding their goal of getting the notes and allied graphic elements into a score in as fast and intuitive a way as possible, arranging the layout on the page, and outputting a readable or publishable result, whether direct to paper or to some form of exportable graphic file.

www.finalemusic.com

Test Spec Finale 2004 v2004.r.3

There have been a surprising number of scoring packages over the years, and what's even more surprising is that many of them have actually lasted for years. Leland C Smith's SCORE, for example, is still in development and started life in 1967. It's easy to think of Sibelius (version 3 of which was reviewed in July

file:///H|/SOS%2004-10/MakeMusic%21%20Finale%202004.htm (1 of 8)9/26/2005 12:36:03 AM


PC with 3.06GHz Pentium 4 and 512MB RAM, running Windows XP.

2004's SOS) as a newcomer, and it's now nearly 10 years old! Perhaps Sibelius's closest competition is the subject of this review, however: the latest version of software that was launched by Coda Music as a Macintosh-only package in the late '80s, Finale 2004.

Evolution Not Revolution Now marketed by MakeMusic! (with exclamation mark), Finale was last reviewed in SOS in its 2003 livery back in January 2003. To be honest, not a lot seems to have changed, though without completely gutting the program and starting from scratch, not a lot has to. Finale will always benefit from the newest developments in computer technology, but not at the expense of maintaining a familiar and comprehensible notation environment. MakeMusic! simply add refinements and new tools rather than offering major changes with each new upgrade. Let's have a quick look at Finale 2004, alluding to the new features when they arise. First of all, you need to decide what sort of document you'll be putting your notes into, and the software starts Amongst the many arrangement plug-ins is helping right away. Choose one of the one that creates automatic piano reductions, 30-plus template scores, including jazz such as this example derived from the score featured in this review's main screenshot. It band, full orchestra, concert band, needs a little editing to remove redundant piano and voice plus various slurs. educational score types. Once you've spent some time with Finale, you may find that you'll be saving your own template scores, for recall later. The templates are ready to go in all ways, including the correct handling and playback of transposing instruments. The next easiest setup option is the Setup Wizard, and as you might expect, Finale 2004 guides you through the process of naming your score, choosing its page size, choice of instruments from a huge collection, their order, initial tempo and time signature, pickup bar, font and so on. Of course, it's possible to start with a more or less blank page and add one staff at a time as you decide you need them. The choice is yours. You're fairly free to decide how many lines a staff is made up of, and can work entirely with highly configurable tablature (userdefineable tunings for each tab string, for example), if fretted instruments are your thing.

Data Entry Next, you'll want to input some notes. There's flexibility here, too. Any or all of the note input methods can be used at any one time, so its possible to mix and match note input via MIDI keyboard, mouse or computer keyboard. The new-forfile:///H|/SOS%2004-10/MakeMusic%21%20Finale%202004.htm (2 of 8)9/26/2005 12:36:03 AM


2004 Simple Entry tool speeds up note entry no end, using computer keyboard shortcuts to choose notes and note values, and customise your preferred parameters. Chord entry, transposition, and temporal moves are all logically undertaken. Articulations, key changes and time signatures can also be input using Simple Entry. The feel is very much of a 'word processor' for music, and if it's not quite as fast as that would imply, it's a move in the right direction. Real-time performances can be recorded from MIDI keyboard (or any other controller, such as a MIDI guitar) using the Hyperscribe tool, the performance appearing almost automatically, and fairly accurately, on screen. Monophonic lines can also be transcribed from audio sung or played into a mic — this option has been improved for 2004, and it can be an eerie experience to see it actually working! With either of these choices, The Smart Page Turns plug-in offers tons of you'll need to tidy up afterwards; the control over this new automatic feature. tempo during recording is also fixed, so a certain amount of expression will be lost that is only slightly compensated for by inputting tempo data on a second pass. Finale is also good at turning scanned music into a workable document: basic facilities, which have been improved for 2004, are built into the software. If you do work with other sequencers, sequences can be saved in MIDI File format and imported into Finale; the software is very helpful in how that file is interpreted before it hits the page. One thing you won't have to notate is individual parts from a multi-part work: the software extracts them automatically. So, while inputting notes might not seem as fast as doing the job with pencil and paper, when you add copying time to create individual sheets for all the performers in even a small-scale work, Finale 2004 is a time-saver! Inputting expression markings, articulations, performance notes and text is also straightforward. It's even possible to design your own graphic elements and define how they'll play back. Editing and moving on-screen data is much enhanced with the new version, and managing complex music is handled well. Staff spacing and other page-specific manipulation is easily and tidily handled. On-screen music can be as complex as you like, with multiple layers per staff that can be treated as individual voices, right down to being assignable to different MIDI channels and dynamics. And yes, Finale 2004 can function as a sequencer, complete with on-board sound source for audio proof. The Smartmusic software synth and General MIDI Soundfont are new for Finale 2004. Of course, the software can already transmit its performance via MIDI, file:///H|/SOS%2004-10/MakeMusic%21%20Finale%202004.htm (3 of 8)9/26/2005 12:36:03 AM


either to an internal or external sound source, but this new development means that audio output will sound identical when moving files to other Finale 2004 workstations. Sibelius 3's built-in sample player — based on Native Instruments' Kontakt — may have the edge here, but you need a more powerful computer to run it, and more money must be spent to upgrade to the best implementation. Finale 2004 is less ambitious, but the proof offered by the GM Soundfont is The new built-in scripting is controlled via perfectly adequate for most situations — and if you like the performance a lot, these windows. It's a bit arcane, but the use of plain English speeds up the process. the software can now bounce the performance to disk as an audio file. The sound quality is adequate, being little more than an expanded General MIDIstyle sound set, but is absolutely fine for quick checking. On the MIDI and performance side of things, Finale makes a move towards traditional sequencing with drum groove capabilities, Band In A Box-derived autoharmonising, and other data manipulation. MIDI data can be turned into score elements, and vice versa.

System Requirements PC: Windows 98, 2000, NT, Me or XP; import and export of EPS files supported under 98, NT 4.0 and Me only. Mac: Mac OS 10.1.5 or higher (Mac OS 10.2 or higher for some features) or Mac OS 9.0.4-9.2.2.

Both require 800 x 600 monitor resolution, at least 128MB RAM and 200MB hard disk space. To input notes via MIDI you'll need a MIDI interface and MIDI keyboard or similar; to hear a performance, you'll need to add headphones or monitors and possibly audio hardware; and to utilise the audio input, add a microphone. Scanning in requires a scanner, and equally obviously, a printer is required to generate hard copy of finished work.

Palette Power There are a couple of things to be aware of when using Finale 2004. First of all, there are a lot of palettes — collections of tools for notes, clefs, expessions, articulations and so on — and they can all live on your desktop, scattered around the page you're working on. You won't need them open all the time, so get used to focussing on what you actually use. Secondly, the menu options, at the top of the page, can change depending on which tool is selected. This is particularly



relevant with the Mass Edit tool. Experienced MIDI sequencer users may wonder why they can't just highlight, cut and paste notes to move around the page. The answer is that Mass Edit, the tool dedicated to just these functions, has to be selected first. Remember also the drop-down PlugThe Instrument List box shows which GM Soundfont voices correspond to the ins menu. This hides a lot of Finale's instruments assigned to your score. power, as well as offering some unexpected creative tools. There are plug-ins for adding 'smart page turns' (which works out how to format an entire part so that a performer won't be lumbered with inconvenient page turns), another automatically adds cue notes to sections with long rests, and yet another offers search and replace options. If you work with words, Finale can now add hyphens and word extensions automatically. You have the The Drum Groove plug-in window, new for 2004. The 'groove' can become part of the final say, though, and can override a result that doesn't lie the way you want score. it to. Other plug-ins create new music from chords or melodic material in ways that ordinary sequencers don't. And if you find yourself doing a particular operation, or operations, repeatedly, you might like to investigate 2004's built-in scripting. With this, a series of changes to a document — or in batches to a series of documents — can be quickly automated. The finished result can be printed out immediately, or saved as a TIFF or EPS file for exporting to any proper DTP package. And on the subject of DTP, it is perfectly possible to combine blocks of text on a Finale 2004 page with music, as one might for exercises in the classroom, or when creating handouts. The program is flexible enough to help you create a very tidy job — and the music can always be played back while on-screen! However you do it, you can be assured of professional-looking results that musicians will be pleased to play from.

Supporting Players Finale may be one of two or three high-profile and well-known scorewriting packages, but there will definitely be more out there than you think. Have a surf over to http://ace.acadiau.ca/score/others.htm if you need any confirmation. This is as comprehensive a list as I've seen, with links to commercial, shareware and freeware packages for all sorts of platforms, ancient and modern.



Now On X Just a few more features to point out amongst the list of remaining 2004 tweaks. First, it's worth noting that the new Finale is now Mac OS X compatible, though it will still happily run under Mac OS 9.02 and higher. Windows users are, of course, supported, with 98, 2000, NT, ME and XP capable of running the latest version. If you need to run the software on two computers, this can be done: each package has two authorisations. Playback performance has been enhanced: the Human Playback option interprets on-page expressions, articulations and other markings in response to a preset style — jazz, baroque or reggae, for example — that can be further refined using a good collection of playback controls. This can be convincing, and helps you visualise (audiolise?) what a real performance might sound like. If the presets don't impress you, try to do better by tweaking the playback settings yourself.

The Simple Entry options dialogue together with the 'Edit Shortcuts' window for Simple Entry. Once you're familiar with the software, nearly everything can be accessed form the computer keyboard.

When it comes to manuals, Finale 2004 is well-equipped. In the box, you're supplied with an installation and tutorials guide, running to over 200 pages. You may need no more, since even newcomers will get their head round the basic, and some not-so-basic, concepts, if they go slowly. A 'Quick Reference Card', 10 pages long in the Windows version, neatly summarises Finale 2004's tools and palettes, keyboard shortcuts and the Maestro font character set. Topping it all off is a very handy visual index that offers an even tighter summary of what's going on screen via a heavily annotated example score. Users with experience of music software, composers with some computer exposure, or anyone who hates reading manuals, may well find this chart helps get them going in the shortest time. As if this wasn't enough, there's the exhaustive complete user manual, provided as a collection of excellently crossreferenced PDF files. In general, I'm not keen on PDF manuals, but this is a doddle to navigate and very well written — and at least a lot more than the basics are covered by the supplied printed material. To top it all off, there are the quickstart videos, accessible from within the program, so if the idea of software really does not compute, these videos are the next best thing to having a Finale 2004 expert sitting next to you.



Smartmusic Auto-accompaniment features have been available for a couple of versions of Finale, and as the main text notes, Band In A Box technology joins a wealth of various automatic arrangement and playback tools. But Finale 2004 also features a neat twist on this theme, in the form of a link to Smartmusic. Though essentially an auto-accompaniment system, Smartmusic's special focus is that it's an educational tool, providing learning musicians with interactive accompaniments to test pieces, scales and exercises. They don't need a patient pianist or tame orchestra at home: their computer, and the Smartmusic system, fill the gap. The system is sold on subscription, via the www.smartmusic.com web site. Understandably, the vast majority of subs so far have gone to schools in North America. A sub gives you access to the software and a library of 20,000 accompaniments and accompaniments to a further 50,000 scales and exercises. Finale 2004 users can save files in Smartmusic format, and anyone with an SM sub can use that file as an accompaniment. So teachers with Finale 2004 can provide classes full of students with standardised or individual accompaniments to suit whatever curriculum or regime is in place. And Finale comes with hundreds of exercises as instantly accessible templates for teachers to check out, tweak, and hand out amongst pupils.

Final(e) Thoughts I started this review off thinking that there are currently two scoring packages vying for your attention. Of course, there are others, but I think it's fair to say that Finale 2004 and Sibelius v3 are the main choices. These days, there's very little that one can do that the other can't, whether on the film scoring stage, in the sheet music production department or in music classes. In fact, it's quite straightforward to move files from one to the other. The differences mainly lie in how each package helps the composer produce a result. Latterly, there may have been some arguments about the speed of these operations, Sibelius migrating from a very efficient platform with some very efficient coding to the Mac and PC. But in real terms, Finale 2004 can easily hold its own on modern, and not quite so modern, computers.

The new Smartmusic accompaniment option is controlled from this window, accessed from the Save Special menu item.

Basic MIDI Setup and soft synth settings — very simple, but very useful in practice. file:///H|/SOS%2004-10/MakeMusic%21%20Finale%202004.htm (7 of 8)9/26/2005 12:36:03 AM


What surprised me was having a quick look at pricing. The fact is, Finale 2004 is the latest revision of an established, respected package suitable for professional environments as much as all levels of education, and it costs a little under £300. Its main competitor, Sibelius 3, lists at double that, and even the 2003 version of Finale was £479 when it was reviewed last year! Perhaps the state of the American dollar is working in our favour here. As an aside, Finale, back in its earliest incarnation retailed for a smidgeon under 1000 late 1980s pounds. And we think music software is expensive today! But I digress: composers, arrangers and educators at all levels continue to use, and continue to discover, this efficient and accessible software. You'll be in illustrious company if you make it your notation environment of choice.

The First One's Free... Want to sample the Coda/MakeMusic! approach to notation, for free? Check out Finale Notepad: it's available as a free download from www.finalemusic.com/ notepad. It doesn't rival Finale 2004 as such, but does provide basic notation facilities on eight staves, in a form that's completely compatible with Finale 2004. In fact, in spite of the eight-stave limit, Notepad can still load complex Finale files, and edit notes. It could be a good choice for school situations, professional facilities, or for distributing your music to people who don't own the full package — they can also play back and print out your work, as well as tweak it. Published in SOS October 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



Presonus Central Station

In this article:

All Aboard Optional CSR1 Remote Control Layout & Operation Source Selection Cues & Talkback Monitor Control The BBC Way The Central Station In Use

Presonus Central Station £500 pros Totally transparent main signal path. Balanced and unbalanced analogue inputs. Two digital inputs. Separate recording and cue outputs. Optional remote controller. Built-in talkback facilities.

cons No phase-reverse facility. No single-speaker mono output. 'Line lump' mains transformer. No interlocking logic option for the third speaker output.

Presonus Central Station Monitor Controller Published in SOS October 2004 Print article : Close window

Reviews : Miscellaneous

This high-quality passive monitor controller can cater for five stereo input sources and three pairs of speakers, with built-in cue monitoring and talkback functions. Hugh Robjohns

With the continuing rise to domination of the computer workstation, and the corresponding demise of the hardware mixer, one of the major practical hurdles to overcome is that of monitoring — source selection, speaker control, talkback, cue feeds and so forth. A lot of people solve the problem by using a compact budget mixer, but these rarely incorporate all the facilities that are really needed and usually end up being an ergonomic nightmare.

summary An extremely transparent and neutral stereo monitoring controller, featuring an entirely passive main signal path with logic-controlled relay switching. The unit also incorporates a D-A for two digital inputs, plus independent cue, headphone, and talkback facilities. An optional remote controller is available.

Photos: Mark Ewing

When I had to address this problem several years ago I was able to persuade a broadcast mixer company, Audix Broadcast, to cobble a bespoke unit together for me based on long-established BBC monitoring functionality. This unique unit has worked flawlessly for me ever since and I'd be completely lost without it, but clearly this particular solution is not an option open to most people.

However, several well-known manufacturers have realised the need in DAWbased studios for a versatile monitoring controller, and there are now several information alternatives around across a wide price range. Some address the specialised Central Station, £499.99; needs of surround-sound monitoring environments (Tascam DSM7.1, SPL Model CSR1 optional remote file:///H|/SOS%2004-10/Presonus%20Central%20Station.htm (1 of 10)9/26/2005 12:36:07 AM


control, £165.41. Prices include VAT. Hand In Hand +44 (0) 1579 326155. +44 (0)1579 326157. Click here to email www.handinhand.uk.net www.presonus.com

2380, and Audient ASP520/510, for example), but there is a growing number of units aimed at the stereo monitoring market. For example, Samson offer the C*Control, Mackie have their forthcoming Big Knob, SPL have announced their Model 2381, and Presonus have just introduced the Central Station. The Central Station has been designed as a very high-quality monitoring control unit, and is one of the most expensive in the short list of alternatives I gave above. In general terms, it provides much the same range of facilities as the others: outputs to three sets of speakers with the usual dim and mono controls, five stereo inputs, metering, and separate cue outputs with talkback. However, this Presonus offering is unusual in that the main signal path is completely balanced and entirely passive, and all the switching is performed with sealed relays (34 of them to be exact). There are electronics in the box, of course — a rather surprising amount, actually — but these only perform essential functions like mono summing, headphone amplification, talkback microphone preamplification, and stereo 24-bit/192kHz DA conversion. So the manufacturer's emphasis is firmly on the side of transparent sound quality with very little in the way to add noise, coloration, or distortion. They're talking my language!

All Aboard The unit is contained in a 1U rackmount box measuring 5.5 inches deep and weighing a modest 5lbs. Sadly, it employs an external 'line lump' PSU rather than having an internal mains power supply. Although I'm sure there are perfectly sensible technical arguments for taking the mains transformer out of the box in this way, I would still prefer an integral supply from the point of view of neatness and convenience. The transformer supplies two 16V AC feeds and a single 9V AC feed, all of which connect to the main unit via a five-pin female XLR plug. The front panel is brushed aluminium, while the rest of the case is painted black. The majority of the connections are on the rear panel, but there are two headphone sockets on the front. All the switches are illuminated with blue LEDs, and the control knobs are a nice knurled metal design, again painted blue. The panel markings are very clear and easily readable, even in dim lighting situations. The rear panel is pretty crowded, but is clearly subdivided into the various sections. Starting at the left (as viewed from the rear), there is the five-pin XLR file:///H|/SOS%2004-10/Presonus%20Central%20Station.htm (2 of 10)9/26/2005 12:36:07 AM

Any of the stereo pairs of speaker outputs can be trimmed by using a screwdriver to adjust the appropriate multi-turn pot on the front panel.


power inlet, and then a pair of digital inputs: an optical Toslink socket and a phono S/PDIF input. Next are six TRS quarter-inch sockets which provide balanced line-level outputs to feed three stereo pairs of active speakers or speaker amplifiers. The next section boasts another four TRS sockets which provide the Main and Cue monitor outputs, again stereo and balanced. Analogue inputs are catered for with another four TRS sockets for two stereo balanced inputs, plus a pair of phono sockets for an unbalanced stereo input. A 15-pin D-Sub socket allows a remote control unit to be connected (see the box for more details), and the final section contains a three-pin female XLR for an external dynamic talkback mic, with a TS socket for a footswitch to activate talkback remotely. There simply isn't space on a unit this slim to house XLR connectors for everything, so the TRS sockets are a good compromise and a lot more convenient than a row of D-Sub connectors.

Optional CSR1 Remote Control The Central Station can also be controlled from an optional remote control unit, the CSR1. This is a really neat little controller, intended to sit on a desktop beside your DAW keyboard and mouse. It measures 50 x 130 x 140mm hwd, sloping down slightly at the front, and connects with the Central Station via a single 15-pin D-Sub socket. The unit is powered entirely from the Central Station and no other connections are required. The remote is styled in the same way as the main unit, and shares the same illuminated buttons and knobs. To control the Central Station, the remote must be activated by pressing a small button on the rear of the main unit. When this is done, a blue LED illuminates adjacent to the main volume control on the Central Station to indicate that the remote control is active, and a similar LED on the remote illuminates to confirm that it is now in control. At the same time, the buttons illuminate to reflect the current source and output settings. The CSR1 is equipped with the six source-selection buttons for the main output (TRS1, TRS2, Aux, and Digital, plus S/PDIF and Toslink), plus the monitoring buttons (A, B, and C speaker selectors, plus Mute, Mono, and Dim). These buttons all work in concert with those on the main unit, enabling source selections to be changed and monitoring parameters altered from either unit. However, the volume control can only be adjusted from the CSR1 when the remote is active — the control on the main unit is disabled and the balanced analogue signal is routed all the way out to the CSR1 and back! Talkback is also built into the remote controller. As on the main unit, there is an internal electret microphone with its own volume control and a 'push to talk'

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button. This can be used instead of, as well as, and even at the same time as, the talkback on the main unit.

Layout & Operation The front panel is very logically laid out and it's easy to see at a glance the status of everything. Like the rear panel, the various control sections are logically divided, starting on the left with the talkback functions, then the headphone outputs, the cue feed input selectors, the main input selectors, the bar-graph metering, the output selectors and controls, and the main volume knob. Although it might be an unusual approach, I'll describe the functionality of the Central Station by starting at the output end and working back towards the front. Trust me, it should hopefully make more sense... The unit can provide monitoring output signals for up to three stereo pairs of monitors (labelled A, B, and C) and illuminated buttons route the output to the required speakers, with an associated mechanical click from the box as the relays work. Outputs A and B are exclusive — meaning you can only have one or the other, not both — and these would normally be used for the main monitors and the nearfields (or nearfields and 'grot boxes') respectively. The switching logic is quite sophisticated, ensuring that if you select output B when A is active, the latter mutes, and vice versa. Pushing the button for the currently selected output again turns it off. Output C works in a different way, and can be switched on and off independently of the selection of Outputs A or B. It is intended for feeding an active subwoofer (with built-in low-pass filtering), and allows the subwoofer to be switched in and out as required. It can also be used as a third monitor selector, of course, but there is no interlocking logic to kill the A and B feeds when C is selected. It would have been more flexible had there been an option switch somewhere enabling interlocking or independent selection for the C output, but it remains a useful facility nonetheless. To the right of the speaker select buttons are six multi-turn trimmers which enable the outputs to be level balanced. As this is a passive signal path there is no gain available, only attenuation — but it is still very easy to match the various output levels to that of the least sensitive amp/speaker. The trimmers span a 90dB range. The main monitoring volume control is at the extreme right-hand side of the unit, and is a high-quality multi-element potentiometer operating directly on the balanced monitoring signal. The panel markings around the knob show file:///H|/SOS%2004-10/Presonus%20Central%20Station.htm (4 of 10)9/26/2005 12:36:07 AM


Pushing either of the headphone output numerous subdivisions, but only the knobs switches the monitoring source extremes are marked with level values allocated to it between the cue and main (0dB and -80dB). Some users may find mixes. this frustrating, as it makes it slightly harder to reset the volume knob to a precise level setting. However, the clear white index mark on the knob itself helps to identify its rotational angle.

Below the three output selector buttons are three more illuminated buttons for output-signal conditioning. These provide Mute, Dim, and Mono functions, all provided through relays again. The first function seems rather superfluous to me, given the fact that the outputs can be muted simply by pressing the currently active output button anyway. The Dim button provides a 30dB attenuation and is activated automatically when talkback is used to prevent howlrounds. The Mono button replaces the main stereo signal with derived mono created by a summing amplifier. This is obviously useful for checking mono compatibility and detecting any phase anomalies, as well as confirming the centre of the stereo image. However, a much better way to check mono compatibility is to listen to the derived mono on a single speaker, as mono on two speakers tends to give a misleading impression of the amount of bass. However, the Central Station cannot accommodate this function. To be fair, very few monitoring controllers are this fastidious, although this is a great shame, as the difference in the mono monitoring techniques is not subtle! While I'm being picky, there is no phase-reverse facility here either — something which I find an essential monitoring tool. Since the Mute button is superfluous, maybe Presonus would consider an updated version which replaced this with a phase-reverse facility — a relatively simple modification given the simplicity of the signal path. Being able to invert the polarity of one channel of the monitoring is very useful. For example, sometimes being able to put the speakers deliberately out of phase is useful to identify the presence of a phase error elsewhere in the signal chain. When trying to match the levels of two channels, such as when aligning a stereo pair, being able to invert one channel and then sum to mono (to produce a cancellation null) makes very fast and easy work of an otherwise fiddly process. Obviously, since the signal path is passive, the input impedance varies according to the setting of the various speaker output attenuators and the main volume control. The handbook suggests a range of 2-5k(omega), and while this is a little low in comparison to most line-level inputs, I can't think of any equipment that would struggle to drive it adequately. The passive nature of the signal path also provides some unusual specifications, such as a signal-to-noise ratio of 140dB; a frequency response of DC to over 1MHz, and distortion of less than 0.0005 percent.

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The input side of the Central Station is very simple and straightforward, with just four logic-interlocked illuminated buttons to select the required input source. There are two balanced line-level stereo inputs which are routed straight through to the outputs via relays. These are labelled TRS1 and TRS2, and provide the simplest and cleanest signal paths. There is also an Aux input, which is unbalanced on phono connectors, and a digital input which can be further selected between Toslink and S/PDIF sources. The Aux input incorporates a buffer amplifier to provide a balanced line-level signal for the input selectors to work with. A rotary gain control on the front panel spans the range -90 to +20dB, so pretty much any input level can be accommodated without too much trouble. Input impedance is still fairly low at 8k (omega)s, but frequency response extends between 10Hz and 50kHz (-0.5dB), distortion is quoted as better than 0.002 percent, and the signal-to-noise ratio is better than 115dB. The two digital inputs are very handy, and enables comparison of two separate sources through the same D-A converter — thereby providing a consistent reference sound quality. The rear-panel Toslink and coaxial S/PDIF connectors can be selected via two more illuminated buttons on the front panel, and the selected source is routed through a 24-bit D-A converter capable of accommodating all the standard sampling rates from 44.1kHz to 192kHz. The output from the D-A is presented as a balanced signal to the input selection matrix to match the other sources. There is no facility to match input signal levels at all (other than at the Aux input), but I don't think that will present a problem to most users, and it's a small price to pay for the advantages of the straight-wire signal path. The selected input source is not only routed to the appropriate speaker outputs, but also to a pair of TRS sockets on the rear panel as a balanced line-level recording feed. The signal is taken from a point before the mute/dim/mono switching and main volume control, and the feed is actively buffered to protect the monitoring signal path.

Between the main input-selection buttons and the monitoring controls is a large 30 segment bar-graph meter, scaled from -48dBu to +18dBu. Although this meter shows the analogue signal level at the main recording output, it is rather bizarrely also calibrated in dBFS (-66dBFS to 0dBFS) — the calibration conforming with the EBU specifications where 0dBu equals -18dBFS. Clearly, the meter should not be considered an accurate alternative to a true digital meter, despite its markings, but it is useful to be able to view the programme levels, all the same.



The meter features a peak-hold facility which refreshes automatically for peaks below +17dBu, but latches red LEDs at +18dBu until the Clear Peak button is pressed. A second button associated with the meter is labelled Calibrate, and this caters for those who like to operate with some meter calibration other than the EBU recommendation. The process is very simple: all you have to do is feed a constant tone into the Central Station at your nominal zero level, and then press and hold the Calibrate button for more than two seconds. The system then adjusts the internal calibration to accommodate the new level, showing this as 0dBu on the meter. The factory standard calibration can be recalled by simply turning off the Calibrate button, and the user calibration is remembered even after powering the unit off and on, until it is changed by holding the button down again for more than two seconds.

Cues & Talkback The Central Station provides more than just a monitoring controller; it also incorporates an independent cue monitoring system. The source selections are the same as those for the main input: TRS1, TRS2, Aux, and whichever digital input has already been selected. There is also a main volume control and the selected signal is available on a pair of TRS sockets on the rear panel as a buffered impedance-balanced output. When talkback is activated the cue source signal is attenuated by 30dB so that the talkback can be heard clearly over it. In addition to the line-level stereo Cue output on the rear panel, there are also two headphone outputs on the front panel, each with its own volume control and capable of a fairly generous 150mW into a 60(omega) load. The headphones would normally be fed with the cue/talkback signal (from after the cue level control), but they can be switched individually (by pressing their volume knobs) to output the main monitoring source selection instead. The current source (cue or main mix) is indicated by small blue LEDs beside each headphone volume control. This function is useful for control-room headphone monitoring, and extends the versatility of the Central Station significantly. Needless to say, the talkback signal is not present on the headphones if they are monitoring the main output. The final control section is for the talkback facility. The unit incorporates a built-in electret microphone which is activated via a momentary (non-latching) square button. The talkback volume can be set with a front-panel gain control spanning a 15-55dB range. The talkback can also be activated via a footswitch (or any other closing contact switch) connected to the rear-panel socket. An external microphone can be connected in place of the internal electret if required, but as there is no phantom power available a dynamic or self-powered mic is necessary. A rear-panel button disables the internal mic in favour of the external one.



Monitor Control The BBC Way The BBC has a standardised and comprehensive approach to the requirements of loudspeaker monitoring controls, and it's informative to consider the arguments for its traditional way of working in the light of the new breed of budget controllers now becoming available. The physical layout of controls may vary between consoles, but the common facilities are two rotary controls (Volume and Balance), with a trio of lever-key switches giving Dim and Polarity Reverse, Mono To Both and Mono To A, and Cut A and Cut B (BBC terminology for cutting the left and right speakers, respectively). The purpose of the Volume control is self-evident, but the need for a Balance control may not be so immediately obvious. It is vital, of course, that any stereo monitoring system is aligned such that equal signal levels in the two channels result in a central image from the monitors. In practice, though, the monitor amplifiers may have slightly different gains, and when listening to the 'illusion' of spatial sound over loudspeakers everyone has slightly different perceptions of where the centre actually is. A balance control allows small gain inaccuracies and personal hearing discrepancies to be corrected quickly and easily. Selecting Mono To Both sends identical signals to both loudspeakers, and then the Balance control can be adjusted so that the image is perceived to emanate from a point midway between the speakers. Changing the monitoring volume during a mixing session can result in level and balance changes within the mix which may only become noticeable when the material is played through continuously at a fixed volume. Hence, once a comfortable listening level is established it is advisable not to adjust the Volume control again, but to use the Dim or Cut switches instead as necessary. Note, however, that the BBC monitoring panel doesn't possess a single Cut switch to mute both speakers simultaneously. Instead, a full mute is achieved by selecting Mono To A in combination with Cut A. This might sound cumbersome, but it is very straightforward in practice, and this control arrangement provides several operational benefits. It is often very useful to be able to listen selectively to one channel or the other — for example to listen for crosstalk or tonal imbalances, both of which are very hard to spot when listening in stereo on loudspeakers. Being able to mute each monitor independently is therefore very helpful. Also, if the loudspeakers are not placed symmetrically in a room there can be a marked tonal imbalance, and again muting each speaker in turn is extremely revealing of such problems. Hopefully, readers will be aware of the need to check the mono compatibility of stereo material, and most do so by sending the same signal to both speakers, (Mono To Both). Unfortunately, though, this approach tends to give a false impression of the tonal balance of the mono signal. In particular, bass frequencies are artificially endowed when working with a phantom image. A far more accurate method is to monitor a derived mono signal on a single loudspeaker, using the Mono To A facility mentioned above. Finally, the polarity-reversal feature inverts the right (B) channel at a point before



the mono switching. Thus, by selecting the polarity reverse and a mono mode together, it is possible to listen to the difference signal (A-B). This is particularly useful for aligning a pair of desk channels for use with a stereo source, since when the channel gains are matched, identical signals in both channels will cancel out, producing a null in the sound level. This mode is also useful for judging the quantity and quality of any ambience or reverberation in a stereo signal. Flipping the monitoring out of phase is a very quick way to check that there isn't a phase problem somewhere in the mix, as well. If the sound doesn't become less focused with the polarity reverse switched in, something is wrong somewhere!

The Central Station In Use The Central Station is very easy and logical to configure, connect, and calibrate, especially with the well-written handbook, which includes several application examples. Once hooked up I couldn't fault the main balanced signal path at all — it really is a straight-wire connection from the TRS input sockets through to the speaker outlets, via the volume control. This unit doesn't add anything to or take anything away from the sound when using the balanced analogue inputs, which is all you can ask of a monitoring controller. The Aux input isn't quite as clean, although you need very good monitoring to be able to hear the difference. However, given that this input is designed for domestic-level unbalanced sources, the negligible quality loss is all but irrelevant, completely outweighed by its usefulness and convenience. The same can be said of the D-A converter, which is very useful and convenient, even if not bestowing class-leading transparency. In a monitoring unit of this price you aren't going to find a state-of-the-art D-A converter — it's just not a realistic proposition. However, the chip included in the Central Station is a very good one, and performs rather better than those you are likely to find in a budget CD player, for example. It is also capable of accommodating sample rates to 192kHz and boasts a 117dB dynamic range. While a decent dedicated D-A converter will perform significantly better (I compared the Central Station with my reference Apogee PSX100 and the improved resolution of the latter was clearly audible), having the facility built in is very convenient when you need to check a reference CD, for example. Also, as I said earlier, being able to route either one of two digital sources through the same converter enables reliable and accurate A/B comparison, and avoids the sonic differences attributed to the individual D-As of different devices. On the output side, I lament the absence of a phase-reverse facility, but the mono, dim, and speaker-switching functions all work very well indeed. The provision of separate switching for a subwoofer may please those who have adopted satellite/subwoofer monitoring systems, but it is a shame that those who have three separate monitoring systems can't configure the unit to operate with interlocking between all three outputs — especially as it would be so easy to do given the logic control already employed. file:///H|/SOS%2004-10/Presonus%20Central%20Station.htm (9 of 10)9/26/2005 12:36:07 AM


The Cue system is another very useful feature, especially as it can be set up independently of the main monitoring source selections, and the inclusion of such flexible talkback facilities makes working with a studio or remote recording room a doddle. I also liked the ability to configure the local headphone outputs to follow either the cue or main monitoring output feeds. Finally, the icing on the cake is the optional CSR1 remote controller. This is the ideal size for a remote controller, and provides the core functions in a very neat and stylish form. I was concerned at first when I realised that the main analogue monitoring signal was being routed all the way out to the volume control in the remote box and back again, but when I realised that the signal is high level and balanced I realised that this isn't likely to be a problem at all. To test the point, I listened extremely carefully to the output while switching between local and remote control, and could hear no significant differences at all. The ultimate purists may choose to spurn the remote control, but for the rest of us the ergonomic and practical advantages far outweigh any theoretical disadvantages. Having said all that, if you choose to wire unbalanced sources to the Central Station's TRS inputs and use the CSR1, the results may be rather disappointing. Overall, then, I liked the Central Station a lot. It is well designed, extremely well built, easy to use and, with the CSR1, very ergonomic too. Clearly, audio quality was at the top of the wish list and Presonus have achieved a commendable result without sacrificing versatility and real-world functionality — and at a very attractive UK price. It doesn't quite match the fastidious monitoring functionality of my bespoke BBC-style unit, but it does equal it in terms of signal quality, and it provides some additional functions to boot. This is definitely worth serious consideration by anyone who takes monitoring seriously and requires the functionality that the Central Station provides. Published in SOS October 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



Prosoniq Rayverb

In this article:

Raytracing Technology Under Control In Use

Prosoniq Rayverb £200

Prosoniq Rayverb Inverse Raytracing Convolution Reverb Plug-In (PC & Mac) Published in SOS October 2004 Print article : Close window

Reviews : Software

pros Excellent results possible. Good editing possibilities Available for Mac and PC.

cons No surround sound support as yet.

There are now plenty of convolution reverbs around, but Prosoniq's Rayverb offers something a little different. So just what is 'inverse raytracing' anyway?

summary Rayverb is another excellent addition to the convolution revolution — high-quality reverb is finally becoming available to the masses!

information £199.99 including VAT. Turnkey +44 (0)20 7419 9999. +44 (0)20 7379 0093. Click here to email www.turnkey.co.uk www.prosoniq.de

Test Spec Rayverb v1.0. PC with 2.4GHz Pentium 4 CPU and 1GB RAM running Windows XP Pro, with Echo Mia 24, Egosys Wami Rack 24 and Yamaha SW1000XG soundcards. Tested with Steinberg Cubase SX 2.2, Sony Sound Forge 7, Voxengo Pristine Space 1.1.

John Walden

Convolution reverb plug-ins are clearly in vogue at present. Recently, SOS have reviewed both Waves' IR1 (May 2004, available to subscribers on-line at www. soundonsound.com/sos/may04/articles/wavesir1.htm) and Voxengo's Pristine Space (in last month's issue), and the latest is Rayverb from Prosoniq. Priced between these two competitors, but close enough to Pristine Space to be affordable to those on a relatively limited budget, it's available for either PC or Mac, and provides a convolution-based reverb that can be used as a stereo insert or send/return effect. It is supplied with a small number of built-in room models and can also use impulse responses generated from other reverberant spaces — a selection is available for download via the Prosoniq web site. Aside from the convolution process, however, the unique selling point of Rayverb is something Prosoniq call 'inverse raytracing'. So what does a technology generally associated with computer graphics have to do with producing an authentic-sounding reverb?

Raytracing Technology Raytracing is used in computer graphics programming to trace beams of light from the viewer's perspective of a scene back to the objects within that scene, allowing the colour and lighting of each pixel representing the object to be calculated. Understandably, this is computationally serious stuff, as demonstrated by the cutting-edge hardware specs required to run many modern computer games. Similar mathematics can be used to model sound waves, and when an architect is designing a room, its size, shape and materials can be modelled to predict the acoustic properties of the space.

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Prosoniq Rayverb

As the name suggests, inverse raytracing is the reverse process: for a prerecorded impulse response, it tries to calculate backwards to identify the size and composition of the acoustic space the IR was recorded in and the position of a sound source within that space. Once these calculations are made, the room model created can be used to predict what the sound source might sound like if it was repositioned within the same reverberant space. While all sorts of simplifying assumptions have to be made along the way, this is essentially what Rayverb's inverse raytracing allows you to do — reposition a sound source within the reverberant space represented by the particular IR being used.

Under Control Rayverb's controls are grouped into a number of sections. Selection of the IR is done in the bottom centre of the display, from where one of the built-in room models can be selected; alternatively, any other IR file can be loaded via the Analyse buttons, whereupon Rayverb analyses the IR via its inverse raytracing process to generate a reverb model of that space. The built-in rooms range from very small to very large and are intended as fairly neutral spaces giving a starting point for further editing. When using other IR files, the left Analyse button is the primary slot, but for extra flexibility, a second IR can be loaded via the right-hand button and these two impulses can then be mixed in various ways via the Mix control (bottom right). At present, Rayverb can only load IRs in the AIFF file format, and any WAV files have to be converted prior to use. The surface type can also be selected from a drop-down menu situated in the bottom portion of the display, with the options being concrete, wood, glass, carpet and tin. These alter the tonal characteristics of the reverb in fairly obvious ways. The Separation control adjusts the stereo spread of the model, while the Dimension control can scale the room model down from its original size if a smaller space is required. Both these controls have a marked influence on the resulting reverb and offer considerable flexibility. Air Absorption provides high-frequency damping while, according to the manual, Silkiness does the opposite. The former certainly produces more natural results and is particularly useful for damping down rooms with long reverb tails. Further tonal control is provided by the 12-node equaliser that is positioned bottom left. Because of the real-time calculations required, the raytracing process does not deal with late reflections. Instead, a generic ambience model is used to create dense late reflections that are merged with those of the IR model to produce a natural result, and the four controls positioned top left make adjustments here. The Ambience Size and Reflection Amount control the size and reflection file:///H|/SOS%2004-10/Prosoniq%20Rayverb.htm (2 of 4)9/26/2005 12:36:12 AM

Prosoniq Rayverb

intensity used in this ambience model, while the Persistence and Iterations controls determine the length and degree of reflectivity (higher values produce a brighter sound). The '2D' space within the centre of the display puts a friendly face on some quite clever processing. The yellow dot can be dragged around within this space — moving the dot along the 'Z' axis will make the audio source seem to be more or less far back within the room. Left-right perspective is also adjustable. This is a neat trick and does provide some additional flexibility in terms of placing sounds within a mix — although, of course, if you're using Rayverb as a send effect, the virtual position of every sound being fed to the plug-in is altered at the same time. The group of controls on the right deal with the overall output from Rayverb and, with the exception of Jitter (used to break up repeating reflection patterns within the modelled room to make it sound smoother), are self-explanatory. Finally, if you create a reverb model that you particularly like, Rayverb can export an IR file which could be used in another convolution reverb plug-in if required.

In Use Given the amount of number-crunching being done, unsurprisingly, Rayverb does introduce a processing delay — the manual says 9417 samples. Of course, most VST hosts (including Cubase SX on the PC, with which I did my testing) automatically compensate for this on playback, but it is unsuitable for monitoring while recording. Although I experienced a few very occasional minor glitches, the plug-in certainly behaved very well. On the test system, a single instance of Rayverb produced a CPU load of about 12 percent. I also have no complaints about the sound quality. The generic models supplied cover a range of room sizes and, when combined with the different surfaces available, it is possible to conjure a wide range of reverb types, from very subtle, warm ambiences through to massive cathedral-like spaces. The other editing possibilities extend this flexibility, although a little trial and error is needed to get to grips with how the various controls interact. Combining two IRs to produce a composite reverb also takes some experimentation but, again, adds to the creative possibilities. I used Sound Forge 7 to convert a number of the WAV-format IRs I regularly use with Pristine Space in order to do a direct comparison with Rayverb. As should be expected, when tested side-by-side with the same IR file, Rayverb and Voxengo's Pristine Space produced very similar results, although the additional 'ambience' processing of the reverb tails in Rayverb did create some subtle differences. Both are capable of remarkably good results, even when used with solo instruments or vocals where the reverb is very exposed. For me, Pristine Space is perhaps easier to get to grips with but Rayverb probably offers a wider range of ways to customise the intrinsic sound of the IR. There are, of course, other differences: while Rayverb is available for both Mac and PC, Pristine Space file:///H|/SOS%2004-10/Prosoniq%20Rayverb.htm (3 of 4)9/26/2005 12:36:12 AM

Prosoniq Rayverb

is PC-only, but it does also offer surround support, which might be a consideration for some. Essentially, the decision for particular users may come down to how tight the budget is, whether surround support is required, the platform being used and the degree of fine-tuning you like to do on your reverb sound. Fortunately, Prosoniq hope to have a time-limited demo available for download soon so users can evaluate the possibilities on their own systems. What is in no doubt, however, is that convolution reverb plug-ins such as Rayverb and Pristine Space are now bringing high-quality reverb within the reach of almost any budget. Published in SOS October 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:


Overview 8-voice Digitally Controlled Analogue Synth Comparisons Published in SOS October 2004 The Oscillators The Filters Print article : Close window Thanks For The Memories Reviews : Sound Module The Envelopes and LFOs Arpeggiator & Multi Mode MIDI Processing External Signals Manual Override In this ever-changing world of virtual analogues, Glide, Pan & Unison modelled instruments, and software emulations, it's a Bugs & Omissions pleasant surprise to find that some companies are Pricing & Options making synths the old-fashioned way. But is nostalgia Sounds really what it used to be? Epilogue

Studio Electronics Omega 8 £3999 pros Well built and solid. Flexible voicing with or without add-on filter boards. Flexible modulation (for an analogue polysynth). Simple and accessible operating system and editing.

cons Limited multitimbrality. On the review model, voices had a tendency to drop out unpredictably. Many of the important controls are quantised at a low resolution. Poorly calibrated filters. Some bugs remain.

summary The Omega 8 promises much, and sounds good, but still suffers from bugs and limitations. It will nonetheless appeal to a wealthy analogue aficionados, for whom its very nature will overcome all objections.

Gordon Reid

Tempus does indeed fugit, as I realised when I looked back at the preview of Studio Electronics' Omega 8 analogue rackmount synth, which I wrote for the March 2000 issue of Sound On Sound (see www.soundonsound.com/sos/mar00/ articles/omega8.htm). I submitted the first draft of this in December 1999, which — as I write this — is over four and a half years ago. Has it really been that long? Since that time, Studio Electronics have produced a couple of revisions to the Omega's OS, bringing the long-awaited multitimbral mode to life in the process, and, more recently, they've switched UK distributors, resulting in a drop in UK price. Both of these factors justify us finally giving this full review to the synth. You can be forgiven for not remembering what I said last time I had the Omega 8 in my studio, so I'll recap; I suggested that it had the potential to be a superb, if expensive, addition to anybody's studio. Since then, though, the gap between DSPdriven 'virtual-analogue' synths (VAs) and genuine analogue synths has narrowed considerably, and the latest VAs, both hardware and software, are now producing sounds that are Photos: Mike Cameron indistinguishable from those generated by the real thing for most people. So, does the Omega 8 still have its place in the scheme of things? Let's find out...

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information See the 'Pricing & Options' box, left. Synergy Distribution +44 (0)1827 313134. +44 (0)1827 313737. Click here to email www.synergy distribution.co.uk www.studio electronics.com

Test Spec Omega 8 OS version reviewed: v2.2 (dated May 12th, 2002). Note: As mentioned in this review, Studio Electronics apparently have a version 1.3 OS update in the works, and its release is planned soon, although it had not yet appeared by the time SOS went to press (late August). We'll report on the changes in a future edition.

Overview I'll start with the basics. The Omega 8 is an eight-voice hybrid analogue/digital polysynth that offers two oscillators, a filter and an amplifier per voice. Sound shaping is provided by three contour generators, one of which you can assign to any three of 19 possible destinations. Modulation is supplied by two LFOs, again with a range of destinations. Niceties include oscillator sync, cross-mod, programmable glide, and an extensive MIDI implementation. There's also a Multi mode that offers a range of multitimbral facilities, plus individual signal outputs and inputs for each voice card that allow you to use the Omega 8 as a bank of individual monosynths, or as a set of limited signal processors. But the feature list is vintage synth-like in feel, by which I mean there are none of the bells and whistles we now associate with polyphonic synths: built-in multi-effects units, phrase recorders and the like. All of this arrives in a 4U rack that offers 33 knobs, 35 illuminated buttons, and a small 16 x 2 LCD (for a head-on look at the front panel, check out page 208). Editing is a two-stage affair. You can effect simple changes by turning a knob, but detailed editing requires you to dip into each section's menus. However, twiddling a knob does not take the on-screen editing system to the appropriate menu, so you must select this by pressing one of the numerous 'Edit' buttons located across the panel. Only one of the three knob-behaviour modes mentioned in the manual is currently implemented. This is 'Jump', which means that when you start to turn a knob, the associated parameter jumps to the value represented by the knob's current position. The manual suggests that Edit mode will turn the The Omega 8 offers full interfacing with each knobs into incremental controls, of its eight voices, providing inputs to and increasing or decreasing the parameter outputs from each of the voice boards, as well as the master stereo outputs. There's as you turn them clockwise or even a dedicted mono output. anticlockwise (respectively) irrespective of the current position, while Match mode will require you to move the knob 'through' the saved value before it affects the parameter. According to Studio Electronics, these modes will become active at the next OS revision, which is due soon. Once in an edit page, you move around using the three cursor keys, and change values using the so-called Q knob. However, with no numeric keypad and no 'fast' mode, you can spend inordinate amounts of time twisting, and twisting, and twisting. Parameters can pass directly from 127 to 0, or from 0 to 127, which makes it quicker to move between extremes, but it's still slow. What's more, there's no comparison of the parameter's control-panel value and the saved one.



Sure, there's a Compare function that compares the complete front-panel sound to whatever is in the current memory location, but there's nothing to allow you to compare individual parameters. Internally, there are eight voice cards. These have dedicated stereo outputs that allow you to direct each voice to a separate destination, and with the pan position of your choice. These outputs are in addition to the stereo pair to which all voices are directed (but from which they can be defeated), and the mono output. The filter on each voice card is a 12dB-per-octave Oberheim SEM-style device, but each card offers three expansion headers that accept piggyback filter boards. On a standard Omega 8, one of these headers (per board) is occupied by a 24dB-per-octave 'Moog-style' filter, while the other two are empty, awaiting the addition of further (optional) cards. As on their ATC1 monosynth, Studio Electronics offer two products to fill these slots; a filter based on the Roland TB303, and another based on the ARP 2600 (at the time of writing, however, it's unclear whether these options will be available in the UK — for more detail on the add-ons, see the 'Pricing & Options' box at the end of this review). There are two other boards in the machine: one that hosts a Motorola DSP, the operating system, the memory, and so forth; and the front-panel board that holds all the controls, the associated LEDs, and that diminutive display. According to Studio Electronics, the Omega 8 is "the world's first completely programmable discrete analogue synthesizer". In fact, there are plenty of integrated circuits inside: every voice card boasts multiplexors, demultiplexors, microprocessor supervisors, quad op-amps, flip-flops, transistor arrays, amplifiers, and more. However, as far as I can tell, these are all part of the voicecontrol architecture, not the sound-generation system. The Omega 8's signal path is indeed analogue.

Comparisons I have seen the voice structure of the Omega 8 compared to Studio Electronics' own SE1 and SE1X monosynths, but this is somewhat misleading. The SE1 (reviewed back in SOS January 1994 — see www.soundonsound.com/ sos/1994_articles/jan94/se1.html) offers three oscillators per voice rather than two, which is one of the reasons why it is sometimes considered equivalent to the Minimoog. With two oscillators per voice, the Omega 8 is actually much closer to Studio Electronics' ATC1, reviewed in SOS November 1996 (see www. soundonsound.com/sos/1996_articles/nov96/seatc1.html). Clearly, the similarities and differences between the ATC, SE, and Omega series (particularly the Omega 2) are the cause of a lot of queries, since Studio Electronics have a section in the FAQs on their company web site about this (see www.studioelectronics.com/support_faq.php).

The Oscillators file:///H|/SOS%2004-10/Studio%20Electronics%20Omega%A08.htm (3 of 14)9/26/2005 12:36:16 AM


Despite this, the tuning of the Omega 8's oscillators is controlled by a digital system that Studio Electronics call Accu-tune. The manual describes this as "a very accurate but not perfect software routine of computer-corrected oscillator tuning". Designed to retain some of the randomness of true analogue tuning and tracking, you can set Accu-tune to 'Off', 90-, 95-, 98- and 100-percent settings, but even at 100 percent there are inconsistencies from voice to voice. I have no problems with Accu-tune — after all, many vintage analogue synths have digital tuning mechanisms that hand pitch-control duties back to analogue circuitry once they have completed their routines. Each oscillator offers three waveforms — triangle, sawtooth and variable pulse (10 percent to 90 percent) with pulse-width modulation. You can select any combination of these simultaneously, as well as the square wave sub-oscillator derived from Osc 1, and noise, if desired. Other facilities include oscillator sync, pitch sweep of Osc 2 by Env 1, and five 'modes' for Osc 2. These are called Normal, Half (which causes the oscillator to track at half-a-volt per octave), 'No CV' (which disconnects the oscillator from the keyboard CV), Low 1 (which lowers the normal pitch by one octave), and Low 2 (which lowers Half by two octaves). Unfortunately, the accuracy of the oscillators' frequency knobs is too low, and it can sometimes be next to impossible to land on the pitch you want, which is frustrating. However, you can dial in the pitches accurately using the onscreen menus. This is slightly more long-winded, but more accurate.

The Filters The Omega 8 offers you one filter per voice, the nature of which you can select from a list of up to four voltage-controlled filters. As already noted, there are two types of filter available in the standard Omega 8: the multi-mode Oberheim SEMstyle filter with low-pass, high-pass, band-pass and band-reject options, and a Moog-style 24dB-per-octave low-pass filter. Three knobs are provided to control these, labelled Frequency, Resonance, and (keyboard) Tracking (from 0 to 100 percent). The Edit pages allow you to select the filter type used, with 'Aux 1' and 'Aux 2' options to access the optional filter cards. The only other parameter is 'Invt', which allows you to invert the effect on the filter of envelopes 1 and 3. However, this works on both contours simultaneously, so you cannot mix a positive contour with a negative one. Setting up a basic patch and playing through the eight voices demonstrates that they sound markedly different to one another. A little investigation reveals that these differences lie in the filter cutoff frequencies, which — because Accu-tune corrects only the oscillators — are not the same for a given note, and do not track each other precisely across the keyboard. Setting the filter to selfresonance and Tracking to 100 percent revealed a spread of just under a semitone between the eight voice cards. This improved as the Omega 8 warmed up, but the filters never moved into tune with one another. file:///H|/SOS%2004-10/Studio%20Electronics%20Omega%A08.htm (4 of 14)9/26/2005 12:36:16 AM


The manual states, "this is not wrong, or poor design", but I have many analogue synths whose filters self-oscillate and can be played in tune across the whole width of the keyboard. Whether this matters to you or not is a matter of taste, and of how you might wish to use the Omega 8. While investigating this, I discovered something else about the digital control system within the Omega 8. If you set the filter to self-oscillation and adjust the cutoff frequency, you'll find that it steps, with divisions of about a semitone. Furthermore, I found a bug here. If the Transpose Octave is set to Low as you play the self-oscillating filter up and down the keyboard, you'll find that it stops tracking below MIDI note C1, irrespective of the setting for the initial filter frequency. The stepping suggested to me that the control is quantised to seven bits, and Studio Electronics have confirmed that this is the case, but point out that control of the envelopes and LFO sweeps are 16-bit. Which brings us to...

Thanks For The Memories The Omega 8 offers 256 ROM patches, 256 RAM patch memories, and 128 RAM Multi memories. But some of the locations in the review machine initially produced nothing but silence. I quickly tracked down the reason for this; they were programmed to use the optional 'Aux 1' or 'Aux 2' filters, neither of which was installed in this machine. And, rather than default back to the SEM filter, the Omega 8 tried to push the audio signal through a piggyback board that wasn't there. Fortunately, downloading the latest set of factory patches for the Omega 8 (dated September 17th, 2002) cured this problem — the patches are correctly set up for an unexpanded Omega 8. Even without this fix, it's simple to edit patches in the RAM banks (C and D) and then store them in the ROM banks (A and B), although you can only do so in complete blocks of 128 sounds. This is worth bearing in mind if you're auditioning patches, though — if you're greeted with silences, it may be that the machine doesn't have the updated factory set in it.

The Envelopes and LFOs As the previous paragraph suggests, the Omega's envelopes and LFOs are also calculated by the DSP. Thankfully, the envelopes are fairly snappy, way ahead of the sluggish digital envelopes found on some of the vintage synths that used this architecture. Env 1 is hard-wired to the filter cutoff frequency, and Env 2 is hard-wired to the VCA gain. That leaves Env 3, which you can direct to any three of the destinations shown in the top table at the bottom of this page. There are dedicated knobs for Env 1 amount and Env 3 amount, but the control for Env 3 file:///H|/SOS%2004-10/Studio%20Electronics%20Omega%A08.htm (5 of 14)9/26/2005 12:36:16 AM


affects only the first destination. If you want to control the envelope depth for the second and third routings, you must do so via an edit page. As you would ENVELOPE 3 DESTINATIONS expect, all three Osc 1 frequency Osc 2 frequency Osc 1 & 2 frequency envelopes are Osc 1 level Osc 2 level velocity Osc 1 pulse width Osc 2 pulse width Osc 1 & 2 pulse width sensitive but, surprisingly, all Filter frequency Filter resonance three have an Noise level additional, knobless, but Cross Mod depth very important External input mix level facility, entitled LFO 1 rate LFO 2 rate LFO 1 & 2 rate 'Decay 2'. When invoked, LFO 1 depth LFO 2 depth LFO 1 & 2 depth this determines the time it takes for (what would otherwise be) the Sustain Level to decay to zero after the initial Decay is completed. I seem to remember that this was the architecture of the Prophet T8, and the combination of Decay 2 and velocity sensitivity allows you to recreate the dynamics of hammered and plucked sounds with a realism that eludes ADSR contours. The final option in this section is the ability to delay the onset of envelopes 1 and 3, with independent maximum delay times of 30 seconds or so. All in all, this means that the Omega 8 has two six-stage envelopes, and one five-stage envelope, which is pretty flexible. There are two LFOs in the modulation section. Each offers a rate knob and a depth knob that controls the amplitude of modulation applied to the first of three possible destinations per LFO. The ubiquitous Edit buttons provide access to the detailed control. Page 1 allows you to specify the destinations and depths of the LFOs (see the table to the left for a full list) while page 2 provides control over the six waveforms, allows you to choose whether the LFO is monophonic or polyphonic (in other words, whether the modulation of all voices is synchronised or not), and whether the modulations are synchronised to incoming MIDI Clock. Happily, each of the LFOs can have its own MIDI sync setting, so you can, in effect, lock the two together, but at different rates. It's just a pity that you can't select LFO waveforms directly from a front-panel control, although I understand that the designers have had to make some tough decisions about what gets a knob on the panel, and what is relegated to an on-screen menu. The third page is interesting, because this allows you to determine what Studio Electronics call the Key Trigger Modes and LFO Quantisation. The first of these is confusingly named... it allows you to choose whether the LFO is invoked in a random position within its cycle, or on the upward curve, or on the downward curve. The second quantises the LFO with a maximum spacing of two semitones



spread over a LFO DESTINATIONS sweep of a major ninth; Osc 1 frequency Osc 2 frequency Osc 1 & 2 frequency eight steps Osc 1 level Osc 2 level playing the soOsc 1 pulse width Osc 2 pulse width Osc 1 & 2 pulse width called 'strange' scale, plus an Filter frequency Filter resonance extra note. It Noise level would have been more Cross Mod depth musical had External input mix level the LFO range Volume been exactly 1V (ie. an octave), but there you go. The final element in the modulation section is Xmod, or cross-mod (audio frequency modulation) of Osc 1 and/or the filter cutoff frequency by Osc 2. There's a button to select the destination, and a knob to select the modulation depth, the latter of which is also affected by Env 3 and the LFOs, if desired. When I engaged Xmod and played a few notes, the voices had different timbres, and it was impossible to play them as a polyphonic patch. When one voice produced a pleasing FM timbre, the next might be a monstrous atonal blare. Whatever the cause, it means that FM is very much a monophonic tool. Studio Electronics claim that they've been unable to reproduce this degree of disparity from voice to voice when using Xmod except at very high mod-depth settings, but on the basis of the review unit's behaviour, I wouldn't personally recommend polyphonic use of Xmod.

Arpeggiator & Multi Mode The arpeggiator is one of the least developed elements within the Omega 8, being monophonic, and having just three modes: Up, Down, and Random. Actually, make that just two modes — Random doesn't work! With a tempo knob, Start/Stop button, one- to four- octave range, and MIDI sync, it's not what you would call fully featured. Multi mode wasn't working last time I played with the Omega 8, so I was looking forward to checking it out. Studio Electronics have changed the button presses you need to access it, which resulted in some delay (as of OS v2.2, the correct sequence is to press and hold Bank Part and the left cursor button; this is documented in the Read Me file accompanying OS 2.2 on the company's web site), but I got there in the end. In brief, the Omega 8 has four different multitimbral modes. 'Prepared' mode allows you to set up a different sound for every voice, allocating bank, patch, file:///H|/SOS%2004-10/Studio%20Electronics%20Omega%A08.htm (7 of 14)9/26/2005 12:36:16 AM


octave and volume, and then plays them sequentially on a single MIDI channel. I'm not sure how you would use this mode, but I suppose that it might find favour if you wanted to step through sound effects. Next comes 'Split 1+7' mode. Experimentation revealed that this allows you to set up two sounds, again allocating bank, patch, octave and volume, but played either side of a single split on a single MIDI channel. As the name suggests, you get one voice below the user-defined split point, and seven above.

The Omega 8 controls are clearly and attractively laid out, with well-delineated sections for the oscillators, filters, envelopes, modulation, patch programming, and multitimbral and MIDI operations. More options are accessible than is apparent from the front-panel controls, of course — plenty of other features are reached via the Edit buttons in each section and the two-line display.

The third option is 'Layer 4+4', and this also does what it says; it allows you to allocate one patch to voices 1 to 4 and another to voices 5 to 8. Then, when you press the keys, it plays voices 1&5 together, followed by 2&6 (and so on) in sequence. Again, the parameters offered are bank, patch, octave and volume. Of course, this reduces the polyphony to four notes, but it allows you to create some huge sounds. Finally, there's 'Multichan' mode. This differs from the others in that its associated parameters are bank, patch, number of voices and 'Type' (mono or poly). It is here that you configure the Omega 8 to act as anything from a single eight-voice polysynth to eight monosynths, or anything in between, subject of course to its limit of eight voices. However, you can't determine the MIDI channels completely freely. The Global channel 'N' controls the first part, 'N+1' drives the second, 'N +2' the third, and so on up to 'N+7', if such a number exists. The Split and Layer modes worked fine, but I couldn't get multichannel mode to work properly — voices kept dropping out, and whenever I set up a new Multi of eight voices, it wouldn't play. Eventually, I found the help I was looking for near the bottom of the v2.2 OS Read Me file — a note there explains that in order to hear any edits or changes made to a Multi, you have to save it, load another one, and then reload your saved one. The Read Me also mentions the need for all eight voices to be assigned — if one or two aren't, the whole Multi doesn't work. Even once I'd figured this out, I found some limitations while using Multi mode. In particular, while you can edit patches there, you can't save the changes. This is a shortcoming if you are setting up a complex sound, determining patch parameters while listening to the composite. Furthermore, if you use the arpeggiator in Multi mode, it cycles through the voices, rather than arpeggiating a single part, which would allow you to play on the other channels as it does so. But on a more positive note, each part in a Multi can use a different filter type, so



you can stack dissimilar sounds, or use the Omega 8 multitimbrally in ways that obscure the fact that a single synth is producing all the voices. This is a significant benefit.

MIDI You access most of the Omega 8's MIDI functions using the dedicated button in the Multi/MIDI section of the panel. Pressing the button repeatedly cycles through six edit pages that allow you to define the synth's response to modulation wheel, velocity, pitch-bend, pressure (aftertouch), and two additional dual-destination controllers that are factory assigned as either mod-wheel, pressure or breath (controller 1), and dynamics or key tracking (controller 2). Strangely, these override the dedicated mod-wheel, pressure, breath, dynamics and key-tracking parameters set in the previous pages. So what's the point? Well, to be fair, there are times when you want to direct a single controller to two destinations so, while it's confusing, it does makes sense. By and large, MIDI CONTROLLER DESTINATIONS this is all good stuff, but there Osc 1 frequency Osc 2 frequency Osc 1 & 2 frequency is a catch. If Osc 1 level Osc 2 level you send a Osc 1 pulse width Osc 2 pulse width Osc 1 & 2 pulse width non-zero velocity or nonFilter frequency Filter resonance zero Noise level modulation value to the Cross Mod depth synth and then Envelope 1 amount Envelope 3 amount change a External input mix level controller's destination, LFO 1 rate LFO 2 rate LFO 1 & 2 rate the existing LFO 1 depth LFO 2 depth LFO 1 & 2 depth value remains in effect until Pan depth Pan rate Pan position you eliminate it manually. This can be very confusing if, for example, you are scrolling through MIDI parameters and destinations while playing, or with the mod wheel slightly offset. For a full list of the controller destinations, see the table at the bottom of this page. The Omega 8 transmits the positions of the majority of its knobs and buttons as continuous MIDI controllers. There appear to be a handful of exceptions, such as oscillator sync on/off, but these are relatively minor omissions, so you can record and replay almost all of your tweaks if you wish. This means that you can sequence patch changes, editing and refining them in your computer as a stream of MIDI CCs. If you're into knob-tweaking as part of a performance, this can be a very powerful facility.



Processing External Signals The rear of every voice board features a socket that allows you to insert external signals into a voice's signal path. Processing the external audio is simple. From Filter Edit page 2, you set the input level, the balance between the external signal and any sound generated by that voice (known as 'Mix'), and the trigger level and trigger window. These parameters aren't explained in the version of the manual I was using, but experimentation revealed that the trigger window appears to set an amplitude range that will generate a Gate, whereas the trigger level seems to be a threshold above which all amplitudes output a Gate. The external signal passes through the voice's filter, and when it reaches the programmed trigger level (or range) it sends a Gate to that voice's envelope generators. Many owners will find uses for this, including the usual chopping up of existing audio using MIDI sequences and controllers.

Manual Override A word of warning on the manual — the version SOS received with the review Omega 8 has apparently been updated recently. This is probably just as well, as the original draft of this review criticised the documentation — a mere 13 pages long — for omitting a number of important details, including, most obviously, any mention of Multi mode (instead, the important information relating to the operation of this mode was, as I have explained above, documented in the Read Me file accompanying the OS v2.2 upgrade). If you're interested in an Omega 8, I would suggest that you check that the manual is the later version, and includes the Multi mode info.

Glide, Pan & Unison The final set of facilities include an extensive Glide section that offers linear and exponential curves (nice), normal and legato modes, extensive destination options (all combinations of Osc 1, Osc 2 and the filter cutoff frequency) glissando, and even 'Auto Glide' which is the old 'Bend' found on some vintage synths. This can be upwards or downward, with user-programmable distance and speed, and is velocity-sensitive. Very nice. There's also an extensive Pan page. This offers not only the pan position, but a panning LFO for each voice, with programmable rate and depth, a choice of panning waveforms, and optional MIDI Clock synchronisation. Additional keyboard facilities include Unison (two, four, six or eight voices), voice allocation mode (first available or cycle), note priority (last/low), and multitriggering on/off for any combination of the envelopes. Unison fattens the sound file:///H|/SOS%2004-10/Studio%20Electronics%20Omega%A08.htm (10 of 14)9/26/2005 12:36:16 AM


considerably, but it's worth noting that it is always monophonic, no matter how many voices are assigned to it.

Bugs & Omissions I think that it's fair to say that this revision of the Omega 8 OS is close to being complete. But I've already touched upon the issue of bugs, and I found others during my time with the Omega that I've not yet mentioned. For example, the manual admits that, when you change patches, the Omega 8 can lose a voice. This means that silence ensues when the lost voice is chosen by the voice-allocation system. The Multi mode also lost voices with alarming regularity, sometimes when I accessed the mode, and sometimes when I made an edit. In the early stages of my investigations, this may have been because I failed to reload the memories in the required manner, but I experienced problems throughout my couple of months with the Omega. There was also a volume bug in Multi mode; the output volume is always at maximum, even when the Master Volume control is at zero. Other bugs were more intermittent. For example, I selected a patch with the volume of Osc 2 set to zero, and then increased the oscillator level. The oscillator appeared on voice 1, but none of the others. Switching the Osc 2 sawtooth and square buttons off and on again cured this. Then there were the numerous occasions when Accu-tune tuned a voice a semitone flat. A second application usually corrected this, but why was it wrong in the first place? As far as omissions are concerned, I would have really liked a built-in means of backing up data, whether a cartridge slot, floppy drive, or ZIP. A headphone jack would also have been welcome — it seems odd to me that a synth of this price doesn't have one. However, there are two omissions that I am glad to report. Firstly, between the preview model and today's instrument, the Omega 8 seems to have lost the ability to crash. Secondly, it no longer demands a new EPROM each time you upgrade the operating system; you can upload new OS and sound files using SysEx. While not perfect, this is a huge improvement over the previous method.



Pricing & Options If, at £3999, the Omega 8 is beyond your budget, but your love of analogue means that you're still interested, there are several cheaper options, although these may only be available directly from Studio Electronics in the US (at the time of going to press, Synergy Distribution, who had just taken on the range in the UK, were uncertain as to whether they would be stocking the complete range of Omega-related products).

The Omega 2 is a more affordable way to obtain the Studio Electronics sound. Of course, there are fewer front-panel controls, but this is a fair compromise. You still get individual outputs from and inputs to each of the two voices in addition to the master ins and outs, too.

At the bottom end of the price range, there's the Omega 2. At $1949 in the US (about £1100), this is far from cheap, but you may well find the price sensible when compared to a pair of Moog Voyagers, or a pair of Minimoogs (two voices, you see). However, bear in mind that this price does not include any of the shipping or export duties you would need to pay to obtain the synth from the States. If you want to hold open your options to upgrade from a two-voice synth to the full eight-voice version, you can also buy abbreviated versions of the Omega 8 with two, four or six of the voice boards installed. In the USA, they will cost you $2295, $2995 and $3795 respectively (or around £1300, £1650, and £2100 respectively, again minus shipping and duty). Finally, there are the optional filter boards. These cost $119 for a single TB303 filter, $900 for a TB303 'eight-pack', $129 for a single ARP 2600 filter, and $975 for an eight-pack of these. These prices, again without shipping costs or export duty, are approximately equivalent to £65, £500, £70, and £550.

Sounds Studio Electronics describe the sound of the Omega 8 in different ways, usually variations on 'obese', 'phat', 'warm', and 'squelchy', the implication being that it has the depth and sound of classic American polysynths such as the Prophet 5, Oberheim OBX, and Memorymoog. But this was not what I experienced while using it. I programmed a range of voices during my couple of months with it, including impressive pads, brass, organs, leads, basses, and many of the other classic 'analogue' sounds that I enjoy. I found that there's also wide scope for experimentation and ample opportunity for extreme noises. I particularly liked some of the shimmering, other-worldly sounds that the Omega 8 can produce. But none of these patches sounded like a Prophet, Oberheim, or Moog to me. To my ears, the closest thing to the Omega 8 — in terms of look and feel, sound, features and, of course, being analogue — is the Roland MKS80, with a bit of the file:///H|/SOS%2004-10/Studio%20Electronics%20Omega%A08.htm (12 of 14)9/26/2005 12:36:16 AM


MKS70 thrown in for good measure. And another comparison occurred to me. The Omega 8 has a clean signal path, and a full-bandwidth frequency response, both of which are good things. In contrast, the Prophets, Oberheims and Moogs of previous years sound fat partly as a result of noisier signal paths and somewhat limited bandwidths that concentrate their sounds in the low and mid frequencies. So ask yourself; what sounds like a vintage-analogue synth, but also sounds clean, bright, and has full bandwidth? The answer is... a virtual analogue synth! If pushed for a comparison, I would say that the underlying character of the Omega 8 shares some of the qualities of the Roland MKS80 and some of a modern VA. Of course, we're talking about subtle distinctions here. Blindfold me and play a single note of (as near as possible) the same sound produced on an MKS80, a Supernova II and the Omega 8, and I would be hard-pressed to tell which was which. You may hate this comparison, and I fear that Studio Electronics may, but I intend it as no insult. Whereas my Prophets, Oberheims and Moogs tend to be wrapped in plastic sheets and put into storage at the back of my studio for lengthy periods of time, the flexibility and transparency of my MKS80 has ensured that it has remained plugged into my desk throughout the past two decades. But this leaves us with a problem. Modern virtual-analogue synths are much cheaper than the Omega 8, and are more flexible, especially in terms of voices, multitimbral options, and effects (56, for example, as opposed to none). So, perhaps the most obvious use for the Omega 8 is as a colony of discrete, dualoscillator monosynths. Given the facilities provided by each voice, and the flexibility of the MIDI control over each, it's not unreasonable to divide the price by eight, and think of it as eight instruments, each costing around £500 or thereabouts. If you have a need for so many simultaneous monosynths, and can afford the extra filter cards needed to give them diverse characters, the Omega 8 could then be a cost-effective solution. But not as a cost-effective polysynth — unless, of course, you lust exclusively after analogue instruments like the Omega 8. Consequently, I think that this synth is going to appeal most to wealthy musicians, as well as those who, although less likely to have the cash to buy one, don't like the sound of digital synths for whatever reason.

Epilogue To be honest, I have some sympathy for Studio Electronics. That they are enthusiasts is clear. They love their synths, and what they stand for. Furthermore, once you have made allowances for shipping, import duty, dealer



margins and VAT, £4000 is probably a fair price for a machine of this construction and complexity. But 2004 is not 1983, and although you once had to pay £4000 (or more!) to obtain an eight-voice polysynth with no effects and limited multitimbrality, this is no longer the case. I think it's fair to say that anyone deciding on the basis of price alone will pass up the Omega 8 and its ilk for a more modern synth. But that doesn't mean there isn't a potential market for it. Just as some people continue to champion vinyl over CDs, others still champion analogue synthesis over digital — although I don't necessarily count myself among them. On that basis, I say good luck to them, and good luck to Studio Electronics. The Omega 8 is a noble effort, and one which — when its current problems are resolved — we should respect. Published in SOS October 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



Virsyn Cantor

In this article:

Start Singing Finding The Balance Voice Working Middle Of The Load Pardon? Cantor In Context System Requirements

Virsyn Cantor Singing Synthesis Software ( XP/ Mac OS X) Published in SOS October 2004 Print article : Close window

Reviews : Software

Virsyn Cantor £200 pros Given the complexity of what Cantor is doing, its user interface is simple and intuitive. Automatic translation of English words into phonemes is flawless. All edits are implemented in real time. Vocal lines can be played from a MIDI keyboard. Lots of control over the sound of the voice, so plenty of unusual effects are possible. Has great potential as an educational tool.

cons The singing is much less intelligible than a human vocal or a typical speech synthesizer. It's not always easy to produce a pleasing timbre, or one that fits within a track. No keyboard shortcuts for Score editor, even in standalone version.

summary Cantor might turn out to have more theoretical interest than practical usefulness, but it's got enough of both to make it worth investigating.

information £199.99 including VAT. Turnkey +44 (0)20 7419

If your computer could sing, what would it sound like? With Virsyn's Cantor software synth, you can find out... Sam Inglis

To an engineer of 30 years ago, today's software tools would be the stuff of science fiction. Who could have predicted that a small grey box would be able to record hundreds of audio tracks, put a tone-deaf singer in tune, or place a virtual symphony orchestra at our fingertips? We're used to this sort of technological miracle by now, but even so, it's hard not to feel a touch of Tomorrow's World-style wonderment about the advent of singing synthesizers. Yamaha's Vocaloid technology, introduced last year, allows developers to sample the characteristic building blocks of a human voice and create a virtual vocalist, ready to sing anything you care to throw at him, her or it. At this year's Frankfurt Musikmesse, meanwhile, innovative German soft-synth designers Virsyn gave the world its first glimpse of their take on the singing computer concept. A mere four months later, the finished program is with us. Cantor bears a number of operational similarities to Vocaloid. It can operate either as a stand-alone application or a VST or Audio Units plug-in, with Rewire and RTAS support promised in a coming update; and even when used as a plugin within a host sequencer, it employs its own piano-roll-style grid for note and lyric entry. There are, however, three fundamental differences between Cantor and Vocaloid. First, Cantor is not based on samples. Instead, a morphing additive synthesis engine derived from Virsyn's Cube software synth is used to generate the 39 phonemes which Virsyn use to reproduce English speech or

file:///H|/SOS%2004-10/Virsyn%20Cantor.htm (1 of 8)9/26/2005 12:36:21 AM

Virsyn Cantor

9999. +44 (0)20 7379 0093. www.turnkey.uk.com www.virsyn.com

Test Spec 700MHz Apple iMac G4 with 256MB of RAM, running Mac OS 10.3.3. Tested with Raw Materials Tracktion v1.5.1.16.

singing. Each phoneme is created by passing an additive sound source through a formant filter, which morphs between a start and an end state. These filter responses are fully editable, and up to to six peaks and three troughs in the formant filter response can be specified as morph points. The second crucial difference, which is a consequence of the first, is that unlike Vocaloid, Cantor is a true virtual instrument, which can be 'played' in real time from a MIDI keyboard. What's more, all changes to notes, lyrics and vocal timbre are made instantaneously, with no need to wait whilst a file is rendered, and all parameters can be MIDI controlled and automated within your sequencer. The third key point of departure is a conceptual one. Whereas Vocaloid is intended to create realistic, human-sounding singing voices that could replace a real lead or backing vocalist without the listener being aware, Cantor belongs more to the realm of 'special effects' vocal tools such as vocoders, guitar talkers, ring modulators and Mellotron choirs. Virsyn don't claim that the results will be similar to real vocals, but have designed Cantor in such a way as to have qualities that no human vocalist could achieve, including a formant range that varies well beyond the usual male/female spectrum.

Start Singing The Cantor box contains a single CD-ROM with Mac and Windows installers, plus a slim but welcome perfect-bound manual in English and German. Installation is straightforward, and the copy protection consists of a serial number which you enter when you first run the program. Registering at Virsyn's web site is highly recommended, as it enables you to download the frequent updates, along with additional files such as example projects and new phoneme sets. Cantor's user interface is almost identical whether you use the stand-alone or plug-in version, apart from a few minor features which I'll point out as they crop up. Each instance of the program is eight-part multitimbral, but the parts are monophonic, so if you want Cantor to sing multi-part harmonies, you'll need to create each line separately. Editing duties are distributed between five screens, which are brought to the front using tabs at the left-hand side, with the bulk of your work being undertaken in the first Score window. This combines a fairly conventional piano-roll editor with control over the most important parameters used for adjusting the timbre of the synthesized voice. Getting started is pretty easy. You select the Pencil tool from the small array above the piano-roll and draw in a note of the required length. When you let go of the mouse button, the text field above the note will turn yellow, indicating that you should now enter a word or syllable. As you do so, Cantor automatically translates it into the appropriate phoneme or combination thereof, optionally displaying the results below the note bar. A hyphen is used to tell Cantor that the following note should be treated as part of the same word. The engine used to translate words into phonemes is licensed from Carnegie-Mellon University, and file:///H|/SOS%2004-10/Virsyn%20Cantor.htm (2 of 8)9/26/2005 12:36:21 AM

Virsyn Cantor

is remarkably good. In the entire time I used Cantor, the only cases I encountered where it didn't know the correct translation were with proper names, and even then it usually makes a good guess. In practice, you quickly forget it's even there. If it does fail you, it's possible to edit phoneme data for each note directly, but unlike Vocaloid, Cantor doesn't offer any way of adding your own words to its dictionary.

Cantor's Voice editing page. The partial display in the top half of the window allows you to edit the additive source used to generate voiced phonemes such as vowels, whilst the noise transfer function below determines the frequency content of the noise used in generating unvoiced phonemes such as 't'.

Other available editing tools include an Arrow tool for selecting and moving existing notes and an Eraser tool. Also in the toolbar are buttons to copy and paste selected notes, an Undo button allowing you to reverse up to 16 of your actions, and options to enable/disable quantisation and the phoneme display. Scrolling and zooming are achieved as in Cubase SX by clicking and dragging in the ruler view; this works well enough, but I missed having scroll bars and clickable zoom settings too. Overall, anyone familiar with the basic concept of a piano-roll editor will find Cantor's pretty straightforward to use, but one thing that gets annoying is that there are no keyboard shortcuts whatsoever. Editing would be a lot faster if you could use key input to switch between the different tools, but as it is, you can't even use the backspace key to delete selected notes; nor can the QWERTY keyboard be used to zoom or scroll the screen, or nudge notes. Some plug-in formats do impose restrictions on keyboard input to plug-ins, so this is perhaps understandable in the plug-in versions of Cantor, but the standalone version is no different. More superficially, it would be nice if selecting the Pencil or Eraser tools actually changed the shape of the cursor to a pencil or eraser; as it is, you just get an arrow, whichever tool is selected. I also encountered occasions when the Pencil tool didn't put the notes quite where I expected, but this wasn't a big problem in practice. Once you've entered some note and lyric data, you can use the transport buttons above the window to play it back; if you're using the plug-in version, you can also choose to slave Cantor to the host program's transport. However, it's more fun to play Cantor 'live' from a MIDI keyboard. In this mode, Cantor still cycles through the lyrics as you play, but the pitch and timing of the notes you've entered into the piano-roll editor is ignored in favour of incoming MIDI data, whether it's coming directly from your keyboard or a MIDI track in your sequencer. If you have the Legato button switched off, Cantor still uses the note lengths from the piano-roll editor, whilst engaging Legato mode means that each note is sustained until you let go of the key or play another note. When you're playing Cantor in this way you do have to be careful to leave gaps where a word ends in a consonant, as these consonants are triggered by the MIDI Note Off, and playing legato causes them to be missed out. With 'pedal legato' mode set in the Voice editing page, it is possible to have Cantor sustain a single syllable over multiple MIDI notes by holding down the sustain pedal and playing legato. file:///H|/SOS%2004-10/Virsyn%20Cantor.htm (3 of 8)9/26/2005 12:36:21 AM

Virsyn Cantor

Finding The Balance Having provided Cantor with some raw materials in the shape of lyrics and sequence data, you can begin to explore the voice-editing controls. Sensible user interface design means that you can do most of the editing you're likely to need using the dozen or so knobs at the left of the main Score editor. These include familiar parameters such as Volume, Glide (portamento), Pan, and Vibrato Rate and Depth, plus a selection of slightly more unusual controls. Of these, Ensemble provides a gentle chorus effect, whilst Bright allows you to adjust the amount of high-frequency content in the output and Humanise introduces elements of random variation in the pitching, volume and vibrato rate and depth, to offset the machine-like quality of the results. Four other controls have a deeper influence on the nature of the synthesized voice. Metallic, according to Virsyn, 'turns the vocal source from a harmonic, partial structure into an inharmonic, metallic one', with actual results not unlike a ring modulator. Balance adjusts the relative levels of voiced phonemes and unvoiced ones, which for most purposes means the balance between vowel sounds and The Phoneme editing window displays the soft consonants on the one hand, and start and finish states of a morphing formant sibilant and hard consonants such as filter. You can draw transfer functions for 's' and 't' on the other. Breath each with the mouse, and place morph introduces breath noise into the sound, points to tell Cantor how to map the filter response between the two. and Gender is perhaps the most fundamental control, covering the full spectrum from impossibly deep bass, through obviously male, female and child singer territory into areas previously inhabited only by the Smurfs and Kate Bush. All of these parameters can be automated if you're using Cantor within a suitable plug-in host, and most can also be automated within Cantor's Score editor itself, so there's plenty of scope for variation even within a single vocal phrase. A nice touch is that when you hover the mouse over a control, its name disappears to be replaced by a numeric readout for that parameter.

Voice Working If you want to get more deeply involved in shaping the vocal sound, you need to head for the Voice editing page. Here you'll find two graphical windows in which you can 'draw' by clicking and moving the mouse. The upper window defines the file:///H|/SOS%2004-10/Virsyn%20Cantor.htm (4 of 8)9/26/2005 12:36:21 AM

Virsyn Cantor

spectrum of the additive source that is to be passed through Cantor's formant filters, whilst the lower one specifies a transfer function (ie. EQ shape) for the breath noise. These use the same click-and-drag drawing method as Virsyn's Cube additive synth, which is about as easy to use as it gets. As on the Score editor, further parameters are available for control at the left of the screen. You can specify the number of partials to be generated by the additive source — using more partials increases the sound quality but also the CPU load — and the corner frequencies of a high- and a lowshelving filter. Less familiar parameters include Blur, which makes Cantor sing in a Mockney accent, and Noise Mod, Cantor's built-in effects include distortion, which determines the extent to which chorus, delay and global reverb. the noise is related in timbre to the pitch of the note; at low values, it's pure noise, but as you increase the dial, the noise acquires more of a pitched quality. This, again, is presumably derived from the noise-generating system in Cube, which works by modulating the partials to create a more or less anharmonic noise signal. Actually, I lied about Blur. What it does is allow you to speed up or slow down the rate at which Cantor morphs between phonemes. The results of varying this are described in the manual as sounding like a 'contrast enhancement' or 'motion blur' of the phoneme sequence, and are usually pretty subtle in practice. Although some of the more unusual parameters can be a little unpredictable until you get used to them, all of the editing tools on the Voice page are easy enough to use. The difficulty level takes a pretty vertical hike when you get to the Phoneme page, which enables dedicated users to create their own phoneme sets. That's no reflection on Virsyn's editing tools, which seem well designed for the job in hand — it's just that the job in hand is frighteningly complicated and requires a lot of specialised knowledge. In essence, you design the morphing filter response for each phoneme using two click-and-draw transfer-function graphs, one representing the filter state at the start of the phoneme and the other the end. You then mark the most prominent peaks and troughs in the transfer function as morph points, and specify whether the phoneme is voiced or unvoiced — voiced phonemes use the additive partial generator as a source, whereas unvoiced ones use the noise source. Other parameters include Transit (morph) times for the voiced and unvoiced parts of the sound, and Sustain, which specifies at what point during the morph Cantor holds the sound when you play a sustained note. Designing a phoneme set from scratch will be beyond the ken of most users, but it can be useful and interesting to play around with the default sets. For instance, the default 't' phoneme in one of the sets generates a burst of high-pitched noise


Virsyn Cantor

that I found distracting, and the phoneme editor makes it possible to tone this down. There are two further editing pages, FX and Mix. The former offers basic distortion, stereo delay, chorus and reverb effects, which might be useful in the stand-alone version of Cantor but will probably remain unused where there's access to dedicated effects plug-ins. Sensibly, the reverb is set up as a global effect, whilst the others are implemented on a per-voice basis. The Mix page allows you to assign MIDI input channels to each of Cantor's eight voices, and provides global control over their output level, pan and reverb send level.

Middle Of The Load CPU load is unlikely to be a huge problem with modern computers. On the modestly specified review machine, Cantor's CPU meter rarely exceeded 20 percent for a single-note part, and I imagine that a G5 or Pentium 4 box will cope without trouble — it's not as though you'd want to have multiple instances of Cantor playing 27-note chords in any case.

Pardon? When I first heard Cantor at this year's Frankfurt Musikmesse, I thought that it would probably become the voice of a thousand novelty hits. After I'd tried it out myself for the first time, though, I wasn't so sure. The results were certainly novel, and you'd have been unlikely to mistake them for a human vocal, even one that had been stuffed through Auto-Tune and vocoded to death. However, there's no way you'd have been able to use Cantor v1.0 for a lead vocal, because its singing was pretty much unintelligible: the output was recognisably vocal-like in character, but often impossible to decipher unless you had the lyrics in front of you. Fortunately, Virsyn are very responsive to user feedback, and they have worked hard to alleviate this problem. Whereas the initial release of Cantor came with just a single phoneme set, version 1.02 ships with eight different factory sets, most of which are clearly better than the original. The vowel sounds in the default set were always fine, as were some consonant sounds such as hard 'g' and 'k', but many of the other consonant sounds were very indistinct. The new phoneme sets go some way towards putting this right, with noticeable improvements to


A simple mixer allows you to adjust the level, pan and reverb send for each of the eight monophonic voices in an instance of Cantor, and to specify which MIDI channel they will respond to.

Virsyn Cantor

sounds such as the 'w' in 'we', the soft 'th' in 'this', the 'f' in 'fish' and the 'p' and 'd' in 'jumped'. The user can also aid intelligibility by playing in a sympathetic manner, leaving rests in the right places and not sustaining notes on unsuitable syllables, as well as keeping the Gender setting appropriate for the playing pitch; judicious twiddling of the Balance between voiced and unvoiced phonemes and the Noise Mod parameter can also help. Despite these improvements, though, making out what Cantor is singing still demands careful attention, and the clarity is substantially inferior to a typical sample-based speech synthesizer — it's about on the same level as a vocoded human voice. Some sounds are still problematic in most of the sets, including the 'n' in 'new', the 'y' in 'you', the 'sh' in 'shell' and the 'b' in 'bite'; with voiced consonants, as in the latter case, it's often impossible to emphasise an initial consonant sufficiently compared to the vowel that follows. However, the average intelligibility has been raised enough that you can now often decipher a dodgy word from the context, which definitely wasn't the case with version 1.0. What's more, the eight different factory sets do have noticeably different characters to them, although as you'd expect, these are all obviously machine-like.

Cantor In Context The improvements in v1.02 now make it possible to use Cantor in a role where it's required to convey a lyrical message, but it would be asking too much of it to replace a human lead singer. You still need to concentrate to understand the singing, and with its distinctive, synthetic character, it seems to work better for short, repeated phrases such as you might hear in a dance track. Creative abuse of parameters such as the vibrato rate and depth can generate some unique vocal timbres which it's easy to imagine the likes of Daft Punk exploiting, although there are plenty of dead ends to be encountered here, and it takes practice to achieve something you'd actually want to use. The sound that Cantor makes is undoubtedly distinctive, but whether it's actually pleasant or useful is of course a matter of taste. Personally, I found it most effective in small doses. It doesn't really sound lush or warm like a classic vocoder, and nor does it have the organic quality of a guitar talk box; on the other hand, if you're planning to bring A Brief History Of Time: The Musical to the West End, you might find it just the thing. There's often a pay-off to be made between intelligibility and musical usefulness, whereby increasing the prominence of the voiced consonants makes Cantor easier to understand, but also makes those consonants tend to become spitty or sibilant and stick out of the track in a way that's awkward to deal with.


System Requirements MAC G4 400MHz or faster, 256MB of RAM, Mac OS 10.2 or later.

PC Pentium III 600MHz or faster, or Pentium 4 or Athlon XP/MP, 256MB of RAM, Windows XP.

Virsyn Cantor

Cantor also has instrumental applications: you can get some very usable lead and bass sounds by forgetting that this is supposed to be a vocal synth, and strange vocoder-esque pads are easy enough to come by. In addition, there are lots of interesting effects to be had by forcing Cantor to gabble through a vocal sequence at ridiculously high speeds, or abusing parameter automation. The fact that each Cantor part is monophonic is quite a limitation when you're using it in a more textural role, though, as it's not possible to play chords in one pass from a keyboard. Even taking into account its potential as a pad or bass-line generator, you couldn't really describe Cantor as a versatile synth, but it is something you would turn to if you wanted to lift a track out of the ordinary. The best way to decide if Cantor is for you is to download the fully functional 4MB demo version from the Virsyn web site; if you do try it, make sure to listen to the different factory phoneme sets in order to give it a fair hearing. Finally, let's not forget that Cantor is just a really interesting piece of software, with obvious value as an educational tool. Although it doesn't sound realistic, it does makes it a lot easier to understand the nature of human speech and singing, and playing around with the phoneme sets can be fascinating. Virsyn deserve enormous credit for taking something as complex as vocal synthesis and implementing it without the aid of samples, in a format that's so intuitive and fast to edit. You won't want to use it on everything you record, but in a world full of 'me too' products and vintage emulations, Cantor stands out as an ambitious, intriguing and above all original software synth. Published in SOS October 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



Wharfedale Pro Diamond 8.1 Pro Active

In this article:

Round The Back Listening Sessions

Wharfedale Pro Diamond 8.1 Pro Active Active Monitors Published in SOS October 2004

Wharfedale Pro Diamond 8.1 Pro Active £234

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Reviews : Monitors

pros A neutral and slightly polite sound. Well built and specified. Good value for money.

cons Pleasant rather than revealing. Becomes hard sounding at higher levels.

Wharfedale's new active nearfields offer a polished sound for studio owners on a budget. Hugh Robjohns

Wharfedale's Diamond brand has been associated with a range of high-quality but Very much a beginner's affordable domestic and home-studio monitor, but one which is speakers, going back the best part of 20 workable and represents years. The most recent in this long line are good value for money. the 8.1 Pro and 8.2 Pro models, available in information both active and passive versions. The larger £233.82 per pair 8.2 Pro has a 6.5-inch bass/mid-range driver including VAT. and claims a few extra Hertz of useful bass Wharfedale Pro +44 (0) response, whereas the 8.1 Pro Active model 1480 447700. reviewed here is slightly smaller, employing +44 (0)1480 431767. a five-inch bass/mid-range driver along with Click here to email www.wharfedalepro.com the ubiquitous one-inch soft-dome tweeter. summary

The Diamond 8.1 Pro Active model reviewed Photos: Mike Cameron here is a compact two-way, active nearfield monitor, intended for small home studios and the like. The speakers can be used as a conventional stereo pair, or extended to form a nearfield 5.1 monitoring system complete with an optional matching subwoofer which includes bass management facilities. The active version is physically slightly deeper than the passive model to accommodate the amplifier chassis, and its very solid and welldamped cabinet measures 295 x 198 x 280mm (hwd), including the rear-panel heat sink. It is made from MDF and covered in an Iridium-coloured laminate which gives a shiny, mottled-grey finish. The front baffle appears to be a composite-loaded plastic material with a matt-black finish, and it is slightly convex — presumably to aid dispersion. The two drivers are mounted vertically above one another in the centre of the

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baffle, with a 1.5-inch circular port at the bottom extending some 4.5 inches into the cabinet. A pair of LEDs at the bottom of the front baffle indicate power and amplifier clipping. The bass/mid-range driver is a little unusual in that its cone is made from a 'bi-directional Kevlar weave', and it has a phase-plug cone at the centre which is intended to improve the off-axis performance and thus the stereo imaging. The voice coil is double wound and vented to ensure reliable highpower handling. The tweeter is a fairly conventional one-inch soft dome with ferrofluid cooling and a neodymium magnet, although it sits well inside a recessed chassis. The monitor is magnetically shielded to allow use close to TV and computer CRT screens. Although the 8.1 Pro Active is a fine looker au naturel, it is supplied with a removable cloth grille supported on a plastic frame. However, this doesn't appear to be very robust — especially the retaining clips — so best to either remove it and pack it away safely for when you come to sell the monitors, or leave it firmly in place and not fiddle!

Round The Back The rear panel carries the amplifier chassis, housed in a divided section at the back of the cabinet. At the top of the panel is a large finned heat sink, with the audio connectors and controls below. Both balanced (on combi jack/XLR) and unbalanced (on phono) inputs can be accommodated, the two effectively being wired in parallel, and there is a centre-detented volume knob and bass filter switch. The nominal input level is 0dBu, and the bass-cut switch introduces a 6dB/octave high-pass filter, reducing the output bass level gradually from 400Hz. This is intended to knock the level of low frequencies down in compensation for the natural bass boost that occurs when monitors are placed close to walls. At the bottom of the panel is the IEC mains inlet with integral fuse holder, and a power switch. There is no provision to change the mains input voltage. The amplifier electronics are very compact and relatively simple. The active crossover stage operates at 2kHz and is built around a couple of the ubiquitous TL072 op amps. This crossover is set to operate at a slightly lower frequency than the passive version (which is set at 2.2kHz), and amongst other things this is claimed to improve the stereo imaging slightly in comparison to the passive model. The power amplification is provided by a pair of TDA7294 chip amps, and the ratings are given as 60W for the bass/mid-range driver and 40W for the tweeter. A distortion figure of 0.1 percent is specified at 50W output, and the speaker can deliver a peak SPL of 106dB at one metre. The specifications also claim a usable bandwidth of 50Hz-24kHz.

Listening Sessions The Diamond 8.1 Pro Actives are very straightforward to set up, thanks to the file:///H|/SOS%2004-10/Wharfedale%20Pro%20Diamond%208.1%20Pro%20Active.htm (2 of 4)9/26/2005 12:36:24 AM


flexible connections and lack of EQ frills. I used the balanced inputs via TRS cables, as the rest of my monitoring chain is entirely balanced. The rear-panel volume control ranges from silence when fully counterclockwise, through to the nominal 0dBu sensitivity at the centre detent, and on to give additional gain as it is rotated further clockwise. The operating range is such that I can't foresee any situation that would present difficulties in finding a workable volume setting, regardless of the monitoring signal source. With my equipment, the centre detent position seemed to be the ideal setting for the kind of moderate listening level I tend to work with, and it made matching left and right gain controls very easy. I left the bass switches in their flat positions, as the monitors were supported on Atacama stands roughly 0.5m away from the rear wall and about 1m from the side walls in a nearfield environment. I tried a brief experiment with the speakers backed up to the rear wall, and found that the bass-cut facility was pretty well judged, restoring the level and extension of bass fairly well back to its freefield character. As I sat down to listen, my first impressions were very favourable indeed. There aren't many active monitors at this price that sound inherently 'right' from the off. As is my normal practice when reviewing monitors, I worked my way through my own test disc of familiar material, starting with some well-recorded male speech. This is actually a very demanding test for loudspeakers and often shows up subtle colorations and distortions that can be missed with musical tracks. The limited bass response of the 8.1 Pro Active was immediately evident, but the bottom end fell away fairly smoothly and gently, without the over-emphasised hump that so many ported speakers seem to suffer. I felt there was a bit of an upper-mid-range 'bloom' around 400Hz, which made the monitors sound slightly warm and 'likeable', and another small peak around 1.5 kHz which made the sound a little bit forward — a hint of a 'shouty' quality perhaps. The high end seemed to tail off very gently over the top octave or so, which produced a sound that was very easy to listen to. The speaker handles dynamics and transients pretty well, although it can start to sound noticeably hard and glassy when the listening levels are raised. It may be technically possible to produce the specified peak level of 106dB, but I don't think you would enjoy working at such levels with these monitors — not that many comparably sized monitors would cope significantly better, to be fair. However, working at moderate levels in a nearfield situation, the 8.1 Pro Active is capable of generating a usably clean output at levels to satisfy all but the most crazed of heavy rockers. Stereo imaging is good too, with stable spatial information over a reasonably file:///H|/SOS%2004-10/Wharfedale%20Pro%20Diamond%208.1%20Pro%20Active.htm (3 of 4)9/26/2005 12:36:24 AM


wide sweet spot. On good coincident-pair recordings made in nice acoustics I didn't feel the sense of perspective and depth was portrayed as well as I know it can be from more expensive monitors, but it was a fair performance given the price, and certainly up to the standards of the immediate competition. The overall sound quality is fairly neutral, but I'd say these speakers tend towards the pleasant rather than the revealing — although that's probably not surprising given their domestic hi-fi heritage. Technical faults don't scream out at you as they do on high-end professional monitors, and it is hard work trying to listen into the inner details of a complex mix. However, the trial mixes I produced with the 8.1 Pro Actives seemed to travel well to other systems without any surprises, and I was pleased to find that the bottom end, while not extensive, was sufficiently neutral and informative to enable the balance between basses and kick drums to be judged reliably. However, I did find it necessary to reference a lot of commercial material before I felt completely happy that I could interpret what I was hearing with confidence. At the recommended UK price, the Wharfedale Pro Diamond 8.1 Pro Actives represent good value for money, and would make a workable nearfield monitor for anyone on a limited budget, although I feel that more informative active designs are available for not much more outlay — it would be well worth auditioning these Wharfedales alongside classic favourites such as the Alesis M1 Active MkII, for example. Published in SOS October 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. Can dust affect microphone performance?

Q. Can dust affect microphone performance? Published in SOS October 2004 Print article : Close window

Sound Advice

This is probably a silly question, but I was just wondering: if I left my Neumann TLM103 mic out on its stand when not in use for long periods of time, would its performance be affected, by dust in particular? It does have a lovely wooden case to store it in when not in use, but I would prefer to leave it on the shockmount if possible. SOS Forum Post Technical Editor Hugh Robjohns replies: If anything, taking the mic on and off the stand and shockmount is likely to cause more damage in the long term, and certainly increases the risk of dropping it! I would suggest leaving the mic rigged on the stand, and placing a clean plastic bag (a large freezer bag, for example) over the mic when not in use to keep the dust off. Don't seal the bag at the bottom (to prevent condensation), but make sure the bottom of the bag hangs down well over the mic to minimise dust entering the bag. Many professional studios do exactly this, rather than putting the mics away in boxes or cupboards. Some even leave the mics connected to phantom power sources to ensure a constant internal temperature, but that might be going a bit far for a home studio. Microphones are delicate instruments capable of detecting minute changes in air pressure, so keep them away from any drafts and never slam shut the lid of a box with a mic inside. Ribbon mics are particularly prone to damage in this way. Condenser mics Protecting condenser mics like Neumann's are also prone to rapid changes in humidity, so avoid moving the mic from warm to cool TLM103 from dust places, and if you have to, allow the mic a chance to equalise in temperature before you will prolong their power it up and use it. They also don't like airborne contaminents like cigarette smoke lifespan. which will degrade performance quite significantly over time. Capacitor capsules can be cleaned and restored, but it is a very specialised job and only cost-effective for the higher-end mics. Dynamic mics, on the other hand, are pretty robust in most conditions — you have to try fairly hard to destroy a good dynamic! Published in SOS October 2004

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Q. Can dust affect microphone performance?



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Q. How do I compress a stereo source?

Q. How do I compress a stereo source? Published in SOS October 2004 Print article : Close window

Sound Advice

I am recording using a spaced-pair miking setup and I want to apply light compression to the signal. Where in the signal chain should I insert the compressor (pre- or post-fade), and will I have to buy a dedicated stereo compressor to do the job properly? SOS Forum Post Technical Editor Hugh Robjohns replies: You will have to use a stereo compressor, or two mono compressors that can be linked together for stereo operation. It is also wise to ensure that all the control parameters on both channels are set the same — that means attack and release time, threshold, ratio and make-up gain settings. In many cases, when a stereo or dual-channel compressor is operating in stereo mode, or when two compressors and stereolinked, one set of controls becomes redundant while the other controls both channels. However, this is far from standard practice, and any differences between the settings of each channel can produce some very odd and undesirable effects.

When the TLA 5052 dual valve recording channel is in stereo mode, one set of knobs controls both channels.

When applying compression to a stereo source, it's very important that both channels experience the same amount of gain reduction regardless of which channel signal exceeds the threshold — hence the need for both identical control settings and stereo linking between channels. If you use separate, unlinked compressors for the two channels, then if one compressor reacts to a peak that the other doesn't see, the stereo image will pull towards the uncompressed side, and your listeners will start to feel very sea-sick! Uniform compression on both channels is particularly important if you're recording using a coincident mic setup, as stereo imaging from a coincident pair is determined by tiny level differences between the two channels. In the case of a spaced-pair miking arrangement — the kind of setup you'll be using — the stereo image is dependent on phase differences as well as level differences, but uneven compression will disrupt the stereo image just the same. It doesn't matter whether you choose to insert the two compressor channels (one handling each mic of your stereo pair) as a pre-fade or post-fade insert in terms of the stereo linking requirements. However, if you insert it post-fader, then the channel faders effectively become threshold controls and will affect the onset and amount of compression.

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Q. How do I compress a stereo source?

Published in SOS October 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 you create a sense of height in a stereo soundscape?

Q. How do you create a sense of height in a stereo soundscape? Published in SOS October 2004 Print article : Close window

Sound Advice

I am working on the soundtrack for an independent movie, and I'm wondering how to create the sense of height in the (stereo) soundscape. I was thinking a bit of EQ applied to the higher frequencies might do the trick, since these are more directional. Or should I filter the high end out of the reverb and keep it dry?

SOS Forum Post Technical Editor Hugh Robjohns replies: Conventional stereo and surround systems have no intrinsic ability to convey height information. They are designed to produce only a horizontal soundfield. However, there are various 'tricks' that can create the impression of height. The most effective is to use complex and specific comb filtering to recreate the kind of natural comb filtering produced in the listener's ears when sounds are coming from above. Several pseudo-surround systems claim to be able to do this — Roland's RSS and Q Sound Labs' Q Sound are two. This type of 3D-from-stereo system has been discussed before in the Q&A pages, in July 2003 (www.soundonsound.com/sos/jul03/articles/qa0703.asp) and, more recently, in July 2004 (www. soundonsound.com/sos/jul04/articles/qa0704-5.htm). To recap briefly here, the problem is that the results are not all that reliable or consistent, as these systems are heavily dependent on both the nature and quality of the listening system and environment, and the perceptions of the listeners themselves! Fortunately, in your situation you are working with pictures, and since eyesight is a far more dominant sense in most people than hearing, if the audience can see a helicopter 'above' them, they will automatically associate the corresponding sound with a similar height. All you have to do is ensure that the perspective and relative panning movement of the sound in the horizontal soundfield doesn't break that illusion. Published in SOS October 2004

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Q. How do you create a sense of height in a stereo soundscape?



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Q. Is USB too slow for MIDI interfacing?

Q. Is USB too slow for MIDI interfacing? Published in SOS October 2004 Print article : Close window

Sound Advice

My question is about USB MIDI interfaces, which seem to be the only kind of MIDI interfaces people make now. I've just bought Tascam Gigastudio, and in the manual it says 'Note: Nemesys recommends ISA or PCI-based MIDI interfaces, as they are faster than USB or Parallel Port interfaces'. As well as being too slow, I've also read that USB is unsuitable for MIDI because USB MIDI has timing jitter that could smear the timing of dense MIDI passages. If software manufacturers think USB is so unsuitable that they discourage people from using USB MIDI interfaces, then why do hardware manufacturers make them, and to the exclusion of PCI MIDI interfaces ? Are there any multiple I/O PCI interfaces around any more? SOS Forum Post PC music specialist Martin Walker replies: I'm not surprised that you're confused, since there's lots of conflicting information around, and much of it is out of date. Although you've presumably just bought Gigastudio 2.5, that quote is actually from an FAQ document dated October 2001, which also says that Gigastudio is compatible with Creative Labs' SB Live soundcard using Direct Sound drivers (which is no longer true, since Gigastudio no longer supports Direct Sound under Windows 2000 and XP), and that laptops are not recommended due to the lack of GSIF-compatible PCMCIA soundcards (no longer true either, thanks to Echo's excellent Indigo range).

USB MIDI technology has improved enough to make PCI-based MIDI interfaces virtually a thing of the past.

However, the most obvious giveaway is the mention of ISA-based MIDI interfaces, since I don't know of any modern PC motherboard that still has any of the now extremely elderly ISA expansion slots that you'd need to plug one in — the last time I bought one was back in 1998! You could complain that manufacturers' support documents should be updated more often, but there is nevertheless still a grain of truth in the recommendation of PCI over USB, as the results of my two-part investigation into 'The Truth About Latency' in SOS September and October 2002 proved. My PCI-based MIDI interface gave me around 3.6ms input latency when capturing a keyboard performance, with a latency jitter of just 0.2ms; a serial-port interface gave around 4.2ms with jitter of 1.2ms; and a USB interface gave about 4.8ms with jitter of up to 1.9ms, all running under Windows XP. file:///H|/SOS%2004-10/Q.%20Is%20USB%20too%20slow%20for%20MIDI%20interfacing.htm (1 of 2)9/26/2005 12:36:51 AM

Q. Is USB too slow for MIDI interfacing?

This should hopefully prove to your satisfaction that USB isn't too slow, since an increase of just 1.2ms over PCI is simply not discernible while playing — a MIDI Note On message will itself last nearly 1ms, and a sixnote chord could therefore emerge spread over 6ms. I personally doubt that an increased jitter of up to 2ms would be noticeable in most situations either — many musicians can apparently detect timing jitter when it exceeds about 5ms, but below this it's likely to go unnoticed. It's possible that during dense MIDI passages the situation could get worse, but I don't think you should worry too much. Moreover, when playing software synths in 'real time' via MIDI, their timing jitter can be two to three times that of the MIDI interface. Sadly it's extremely rare to find a PCI-based multi-channel MIDI interface nowadays — nearly all are USB devices. But there are various things you can do to minimise timing jitter problems with a USB MIDI interface. It's important to plug it into a dedicated USB port rather than a USB hub (powered or otherwise) so that the interface isn't fighting for its share of the USB bandwidth with other devices. Always use the latest interface drivers, and try not to use too many USB devices simultaneously, even if they are plugged into separate ports. Also, I personally still think it's tempting providence to try to run separate USB audio and USB MIDI interfaces simultaneously, since their drivers may end up fighting for supremacy. The bottom line is that loads of musicians (including me) are now running multi-channel USB MIDI interfaces with no obvious timing problems, while some of the problems that others run into aren't necessarily due to the interface, but to other issues with their computers. When Nemesys wrote that FAQ, USB MIDI was still in its early stages, and things have improved since then. It's also important to remember that while the capturing of a MIDI performance may be subject to a couple of milliseconds of timing uncertainty, many of us are relying more and more on software synths, whose playback timing is always accurate. Published in SOS October 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 are the correct input impedances for Guitars and Mics?

Q. What are the correct input impedances for Guitars and Mics? Published in SOS October 2004 Print article : Close window

Sound Advice

I bought a new Boss guitar effects processor the other day, and noticed that, even without any effects, the guitar sounds a lot better through it than it does just straight through my mixer. I remembered reading something about electric guitars having a different input impedance to other kinds of equipment, and I'm wondering if this could be the cause. Also, could the sound from my mic be similarly improved by getting a stand-alone voice channel for it rather than just putting it through the mixer? Would the impedance match be much better? SOS Forum Post Technical Editor Hugh Robjohns replies: An electric guitar ideally needs to work into an impedance of around 1M(omega) or thereabouts. The line inputs of most sound consoles and mixers (I'm assuming that you had previously been plugging your guitar into a line input) generally have an impedance of about 10 to 50 k (omega) — way too low to allow a guitar's pickups to work properly. The Boss pedal will have a mich higher input impedance, and the guitar sounds a lot better because its pickups can work more effectively. Electric guitars always benefit from a decent 'DI' input, as you have now discovered for yourself. Modern microphones require an input impedance of between 1.2 and 5 k(omega), and most sound consoles, preamps and channel strips standardise on either 1.2 or 2.4 k(omega). A few preamps now offer variable impedance options that span the full range, but these are relatively rare. In impedance terms, there is unlikely to be any benefit at all in connecting your mics to channel strips rather than the mixer mic inputs. However, a dedicated channel strip may well have better-quality mic preamp circuitry and better EQ than a budget mixer, which may enable a better overall sound. Published in SOS October 2004

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Q. What are the correct input impedances for Guitars and Mics?



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Q. What do the different 'colours' of noise do?

Q. What do the different 'colours' of noise do? Published in SOS October 2004 Print article : Close window

Sound Advice

I'm familiar with white noise, but I've also seen mention of pink, red and even blue noise in your pages. What are these other kinds of noise, how are they produced and what are they used for? Derek Amesbury Technical Editor Hugh Robjohns replies: There are several 'colours' of noise, the most commonly encountered in audio circles being white and pink. White noise is a signal with the property of having constant energy per Hz bandwidth (an amplitude-frequency distribution of 1) and so has a flat frequency response. It is mainly used for testing audio equipment. Pink noise contains equal energy per octave (or per 1/3 octave). Of all the 'coloured' test tones, this one sounds most like naturally occurring noise (a waterfall, for instance) and the amplitude follows the function 1/f, which corresponds to the level falling by 3dB per octave, or 10dB per decade. Pink noise is mainly used for acoustic measurements. Brown noise, whose name is actually derived from Brownian motion, is similar to pink noise except that the frequency function is 1/(f squared). This produces a 6dB-per-octave attenuation (20dB per decade). Blue noise is essentially the inverse of pink noise, with its amplitude increasing by 3dB per octave (the amplitude is proportional to f); and violet noise is the inverse of brown noise, with a rising response of 6dB per octave (amplitude is proportional to f squared). There are several ways of creating braodband noise. The simplest is to use a special noisy diode or transistor and amplify the noise it produces naturally. However, a more modern practice is to use a microprocessor to generate random numbers which are then converted with a D-A coverter to produce the noise signal. The various colours of noise are produced either by simple analogue filtering, or by clever programming of the random number generator used to create the noise in the first place. Published in SOS October 2004

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Q. What do the different 'colours' of noise do?



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Q. What kind of stands are best for mounting monitors?

Q. What kind of stands are best for mounting monitors? Published in SOS October 2004 Print article : Close window

Sound Advice

Is it best to use table-top stands, floor stands or wall brackets to mount my speakers? They're Mackie HR624s, which aren't ported, so does it matter if they're positioned close to the wall? SOS Forum Post Technical Editor Hugh Robjohns replies: The HR624 is not ported in the conventional sense, but it does employ a 'passive radiator' which is mounted on the rear panel behind the amplifier chassis. Essentially, this is a port with a diaphragm stretched across it. Consequently, it is not a good idea to place the speaker hard against a rear wall, although, as the passive radiator is tucked in behind the amplifier chassis, it is impossible to place it hard against a wall anyway. Personally, I'd recommend wall-mounting your monitors using good sturdy hardware that holds the speaker about 4 to 6 inches away from the wall. That way, the speaker vibrations are completely isolated from everything else and the speaker is held firmly in position. Adjust the wall brackets or the speaker mounting so that the tweeters are aimed at the listening position. Many wall brackets are designed to be fixed to the speaker with bolts through its back panel. Some speakers intended for wall mounting in this way come with bolt holes already in place (the PMC DB1 and Genelec 1029, for example). However, if you plan to do the mounting yourself, take great care — the crossover of passive speakers is often mounted internally on the back panel behind the terminal plate, so it might be safer to bolt onto the base of the speaker instead. You can easily check by removing the connector panel and having a peek inside the back of the speaker. If wall-mounting is impractical, then floor stands are better than table stands, again because they're better at controlling vibrations. If you have to go down the table-mounting route, some form of high-mass damping is usually a good idea. Try either placing the monitors on high-density foam isolators, such as the Auralex Mo Pads, or putting some Spectra Dynamics Deflex anti-vibration sheeting on the table. Place a small concrete paving stone or heavy quarry tile on top of the foam, and Blu-tak the speaker on top of that. The extra mass of the slab will help control and damp vibration, and the foam will help prevent low-frequency vibrations from passing into the table structure. Published in SOS October 2004

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Q. What kind of stands are best for mounting monitors?



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Q. What mics should I use on a snare drum?

Q. What mics should I use on a snare drum? Published in SOS October 2004 Print article : Close window

Sound Advice

I am looking at buying a matched pair of SE Electronics SE1 mics for drum miking. I am prepared to pay more for the right mics, but would the SE1s be suitable for 'over and under' miking of the snare? If not, could you offer any alternatives for this kind of configuration? SOS Forum Post Technical Editor Hugh Robjohns replies: The SE1s are great as overheads, but I'd be wary of using them for close-miking a snare. In the case of jazz drumming you might get away with it, but for a heavy-handed rock drummer with a loud snare, you stand a very good chance of overloading the mic's internal head amp. Looking at the published specs, the SE1 is rated with a max SPL of 130dB for 0.5% total harmonic distortion (THD). If you really want to use smalldiaphragm condensers, I'd suggest something like the Rode NT5, which is rated at 143dB SPL (albeit at 1% distortion), and so should be able to cope with a close snare a lot better. It all depends on the kind of snare drum sound you are after, but condenser mics on snares can sound rather lightweight and thin. They deliver the transient 'thwack' of the hit very well, but often lack body. On the other hand, a good dynamic mic, like the venerable Shure SM57, inherently 'soft-limits' the transients and gives a much thicker, more full-bodied snare sound.

If you're going to 'over-andunder' mic a snare, remember to switch one of the mics out of phase.

The over-and-under technique can be useful, as long as you remember to phasereverse one of the mics. This is because when the stick hits the batter head, the head moves away from the top mic, producing lower air pressure (an initial rarefaction), while the snare head moves towards the bottom mic, producing a rise in air pressure (an initial compression). If you mix these two together without inverting the polarity of one of them, the two opposite pressure waves will tend to cancel each other out, and result in a very thin sound. Flip the phase of one mic (usually the bottom one, but it depends on the phasing of any other mics around the kit), and you should get a really big-sounding snare. It's a good trick to experiment with, but not essential, and in general if you position the right mic in the right place above a good, well-tuned snare drum with a decent batter head and a competent drummer you should still get excellent results.

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Q. What mics should I use on a snare drum?

Published in SOS October 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. Why does it take so long to freeze a VST Instrument track?

Q. Why does it take so long to freeze a VST Instrument track? Published in SOS October 2004 Print article : Close window

Sound Advice

I use Cubase SX 2 and find the Freeze function for VST Instrument tracks very useful. I was slightly disappointed to note that it can take up to six or seven minutes to freeze a VST track. Why is this the case when it only takes one to two minutes to export an entire composition as a WAV file? Graham Pettican PC music specialist Martin Walker replies: Steinberg introduced their Freeze function in Cubase SX version 2.0, and it's certainly handy to be able to turn CPU-hogging software synths into normal audio tracks and then deactivate them, especially as it's a reversible process, so you can still go back and make changes later on. Unfortunately the current version of Steinberg's Instrument Freeze function relies on the Project Length when performing its calculations (you can find this in the Project Setup window, launched from the The time it takes to freeze a track in Cubase SX 2 will vary depending on the length of the Project menu). By default, each time you create a new project, its Project. length gets set to a different default setting depending on which template you select. For instance, the 16-track MIDI Sequencer template defines a length of six minutes, the Empty and Default templates both have lengths of 10 minutes, while the 16-track Surround Mix and 24-track Audio Recorder are both defined as a huge 30 minutes. Each output of the various VST Instruments used in your songs is also scanned for a potential audio signal throughout the entire length of the project, which further complicates the calculations, making them take longer. Steinberg have admitted that this is the 'first generation' of their new Freeze function, and have already stated publicly that improved features and speed increases are planned for future releases. In the meantime, since most songs tend to be far shorter than the default settings, the easiest way to ensure that your freezes take less time is to open the Project Setup window and set the Project Length to just slightly longer than your song. Leaving this extra room should ensure that any long note-release times and reverb tails are properly captured in the frozen version. Published in SOS October 2004

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Q. Why does it take so long to freeze a VST Instrument track?



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All Things Must Pass Leader Published in SOS October 2004 Print article : Close window

People : Industry/Music Biz

The theme for this month's leader evolved after a discussion I had with Craig Anderton at a recent trade show where we were both bemoaning the ephemeral nature of modern software instruments. Craig pointed out that if you bought a violin you could play it today and expect to go on playing it for a long time to come, but if you were to buy a software instrument and then dedicate time to learning to make the most of it, there is no guarantee that it will even survive the next OS update, let alone last for 300 years. This may seem an odd criticism given that most of my editorial shows me to be a strong advocate of software instruments, but it's only because I like them so much that I worry about their premature extinction. It's evident that some people prefer hardware and some prefer software, but as a sequencer user whose outboard gear is now largely confined to the cupboard of eternal darkness, I have to say that I find soft-synth plug-ins are far easier to use than their hardware counterparts. They nearly always have better audio quality as they use your soundcard's converters and they can access much more waveform memory, to the extent that a single patch may have more sample memory than the whole of a competing high end hardware synth. Their Achilles heel is that, unlike hardware, they can be pirated, and if the company behind them stops bringing in revenue, they can't afford to rewrite the code when a new generation of OS comes along, or when a major player decides to switch from VST to Audio Units. What was a fantastic synth one day could become a worthless CD-ROM the next, though the majority of manufacturers have, to their credit, hung in there and come up with the necessary updates, albeit sometimes not as soon as we'd have liked. My own view is that the piracy situation is unlikely to get significantly better without draconian measures on behalf of the software companies, and the business of authorising and installing software is already a serious pain in the proverbial. Hardware has at least a measure of permanence — at least as long as MIDI lasts — and unlike software, it never runs out of resources or quits because you've added another plug-in to your mix. The obvious solution to all this is for the hardware synth manufacturers to pursue a greater degree of integration with the sequencing world to the extent that the control panels for their synths appear as plug-ins, the audio is routed into your sequencer's mixer in the same way as a plug-in and the instrument settings are saved in the sequencer song. Some external DSP boxes, such as TC's Powercore, prove this approach can be made to work effectively on a technical level and we're also now seeing external processing boxes that can run conventional VST plug-ins outside the computer. Perhaps the next logical step for companies who are worried about software piracy is to put their synth engines on hardware cards, then build an external unit to house multiple cards. For this to be successful, we'd have to file:///H|/SOS%2004-10/All%20Things%20Must%20Pass.htm (1 of 2)9/26/2005 12:37:28 AM


have a standard platform — there's absolutely no point in a box that only houses one manufacturer's synth cards. The manufacturers agreed on MIDI last century so maybe they can make this their good deed for this century? This external box might usefully include a hardware control surface and would connect to the computer via Firewire so that the reliability and security of hardware could be combined with the user interface and audio routing of a conventional plug-in. At least if the OS did change, only the host unit's interface code would need to be rewritten — there would be no need to redesign all the individual synths. Even so, I have a feeling that the Stradivarius will have the last word on this one! Paul White Editor In Chief Published in SOS October 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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Business End

In this article:

Business End

Spatialize Readers' Tracks evaluated Tim Published in SOS October 2004 Goalen & The Print article : Close window Merchant People : Miscellaneous Beat Quintet The Collectiv This Month's Business End enables you to have your MPG Panel producers, songwriters, musicians and

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

Spatialize Track 1 Nel Johnson (NJ): "He says in his letter that he started out in '93 and "soon 2.8Mb progressed into the serious world of sequencing" but I think this sounds like much older than that — the whole thing has a very early-'90s ambient feel to it. He also says that he performs his music live in chill-out tents at festivals, and I can imagine it working well in that sort of situation. It probably gets better as you get more stoned."

Pete Gordeno (PG): "I know what you mean — it doesn't really sound current. Some of it sounds like a Trevor Horn remix. It's quite simplistic structurally, you can kind of guess what's going to happen and when the changes are going to come." NJ: "There was a lot of this sort of thing in the early days of dance music. There was a lot of fallout stuff as people started using the same techniques to do slower more chilled-out type stuff but that was about 15 years ago now and this does sound like it could be 15 years old." Rennie Pilgrem (RP): "There's a real lack of clarity to this. There's no real oomph to the beats or the bass. I think if you were smoking and you were a bit stoned you could probably get quite into it though." Jamie Vaide (JV): "I can imagine this working at chill-out type things but it's not outstanding. It's very polite and I think it could be much better if it was a bit more dubby and in-your-face." PG: "I think the main problem with this as chill-out music is that there's too much going on. I agree with Jamie about it needing that punch in the low-end, there's far to much reverb on the mid-range stuff and that's really obscuring the rest of it." NJ: "There's some great programming in this, some really clever techniques. I get the the feeling that if this was stripped out and some of the more choice pieces were brought up in the mix and the bass was warmed up a bit, it could be a lot better."

file:///H|/SOS%2004-10/Business%20End.htm (1 of 5)9/26/2005 12:37:30 AM

Business End

Deirdre Melvin (DM): "I think some of this is a bit darker than you'd generally expect chill-out music to be. I think it could be interesting if he took that a bit further and got a bit more aggressive with it." JV: "I think if I was chilling out to this I'd get quite scared, I'd have to run away and chill out somewhere a bit quieter." RP: "It sounds like this guy is following things rather than doing his own take on a style. This could have come out at any point in the last 10 years. He's just he's just working within a genre — he needs to think more about how he can put his own spin on it."

Tim Goalen & The Merchant Beat Quintet DM: "There's clearly talent here and there's clearly a will. There are some good bits in this — the first track starts with a sample of people cheering, which I thought was funny and confident at the start but it goes on for too long and wears a bit thin.

Track 1 2.8Mb

"I wonder what's he trying to articulate with this? The blues is about expression and social commentary, it's about angst and passion — I'm not sure I can feel anything like that with this. I think it's competent but sometimes it's too close to degenerating into pub rock. You need much more passion to make this kind of music work well." PG: "I think this comes from a very muso sort of standpoint and he's trying to give it a trendy edge by putting in the samples and stuff. He's trying to give it a pop sensibility but what's hidden underneath that is basically a bad jazz quartet. "If he wants this to exist purely on the strength of the playing he needs to spend a lot more time playing gigs — he's a good guitar player, but so are tons of other people and he's not good enough to stand out for that alone. Also, it seems that virtually every one of these songs has the same chord sequence, they're all basic variations on a 12-bar. He needs to diversify a bit, musically, and show a few more strings to his bow." RP: "It's obviously the guitar player's band and the guitar seems to be the dominant instrument on all of these songs. He needs to give everything else a bit more space. I think if he had a proper blues singer and worked around that, then it could be really good. It sounds a bit like a backing track at the moment and I think concentrating on the singing and the other instruments rather than just focusing on the guitar would really help it." PG: "The guitar is the dominant thing and you can tell that he's been overdubbing it as well. I think it would be much cooler to record this as a band, so if you're playing lead there won't be any rhythm. The Red Hot Chili Peppers do that sort of thing and it always sounds like it's been recorded live. The White Stripes are doing that as well and whilst it might sound a bit thin sometimes I think it sounds a lot more honest without the overdubs — and that's important with this kind of music. "He says in his letter that he recorded this at home and went to an eight-track studio to record the drums. It's always going to be difficult to record a band like that and retain any sort of live feel.


Business End

"I think the whole thing could be refined — he's definitely got ability but he needs to think about when he's playing and whether what he's playing means anything." NJ: "I agree: he needs to think of the song as a whole instead of just the music within a song. On a positive note, I reckon, after hearing all the songs and seeing how he's presented himself with the disc and the artwork, there's definitely something to be built on here. I think it would have been really easy for him to take this in a Zero 7 direction and I admire the fact that he hasn't done that — I think it's good that there's some grit in this and it's not really polished." JV: "I really get the impression that this guy means it and believes in what he's doing. Maybe he hasn't got quite the terms of reference to draw on. I think the music is well played, maybe it's not outstanding but there's nothing wrong with it. I don't think the songs are breaking any new ground but they're quite pleasant, they're quite enjoyable. I think he's maybe got quite a way to go with this but good luck to him."

The Collectiv JV: "I really like the start of the first track but I was waiting for an epic chorus and it didn't happen. Then the next few tracks are just entirely unlike the first. I'm inclined to like them as a band because they've obviously got a good sense of humour. There just aren't really any hooks in any of the tracks though."

Track 1 2.8Mb

RP: "I think this is inbetween too many styles. The first track sounds like a live band, which was interesting but then the next two tracks were just fairly tame drum & bass. I can imagine the second two tracks working well live — at a festival or something — but I don't think they translate to CD very well. It sounds very programmed and just not very current. The female vocalist is very good, she's got a lot of power in her voice. Maybe it's a bit old-fashioned but I like to be able to understand what people are singing and the words are quite indistinct sometimes." JV: "Personally I'd be inclined to say settle for the style of the first track because I think that actually sounds pretty good. I think they just need to write a good chorus because I like the rest of it. I think they've got the makings of quite a good rock band and I think that that's more their strength. The presentation of the CD and the letter and biography which came with it are very nice and I was very predisposed to like them because of that. I love the thing on the sleeve that says 'The Future Sound of Exeter', I think that's great. I still could like them but I do think they should write more rock songs. "It's frustrating that they seem to shift between one style and the other — between the band and the electronics. If they could blend those styles into the same tracks then they could be a terrific band. I'd definitely like to hear more from them in the future, but not more of the same." NJ: "I think they need to give someone the lead. To me it sounds like a lot of people who aren't really listening to one another. They're all really good at what they're doing but it does seem a bit directionless sometimes. There's a lot of things all battling for prominence. It might be interesting if they got the drummer to play the drum & bass parts — that could make the two styles they're using work together." PG: "You can see where they're coming from with the name — the Collectiv. I'm sure there's a scene for this


Business End

sort of thing in Exeter and they've all come together through that. I'm not sure how well that translates this side of the M5 though. I wonder if there's a market for this sort of thing, I'm not sure there is beyond selling at festivals or something like that." DM: "I think they sound like two completely separate bands, and I don't really think that that's their intention. I think they need someone to take the lead and get them all going in the same direction musically. They could quite happily blend the rock and the drum & bass and create something that's much more interesting and creative. I think that could have quite a broad appeal if they did it right."

This Month's MPG Panel

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


Business End




Crosstalk

In this article:

It's A Matter Of Test... Bangers And Mash-ups

Crosstalk Reader Feedback Published in SOS October 2004 Print article : Close window

People

It's A Matter Of Test... The reason I'm contacting you is that I have been quite shaken by a soundcard review which appeared in the October 2004 edition of PC Pro magazine. You see, it is my intention to retire my beloved Studer Revox A700 half-track stereo tape machine and purchase a high-quality digital system. The glowing manufacturer's specification of the Emu 1820M soundcard, together with your own excellent review, meant that I was days away from purchase; that was until I saw the test specs in the PC Pro review. For the line out, they give figures of 1.3 percent THD (Total Harmonic Distorion) at 1kHz and 0dB and 79dB signal-tonoise ratio. For the line in, figures of 0.7 percent THD (1kHz, 0dB) and 82dB S/ N ratio are given. Apparently the Emu's analogue circuits are clipping at only 0db. I find this very disappointing. For a soundcard purporting to be of the highest quality, PC Pro's measured performance figures seem to me to be quite unacceptable. They do not state if the noise figures are weighted but even so — 1.3 percent distortion at 0dB? It is ironic, is it not, that the 1820 is beaten in these areas by Creative Labs' NX and ZS soundcards, the ZS being PC Pro's 'Labs Winner'. What I would like to know is, firstly, in your own review, were detailed distortion and S/N measurements taken, and, secondly, assuming that PC Pro's measurements are correct, how good would you consider the performance of the 1820 to be? Maurice Deacon PC music specialist Martin Walker replies: I don't think you need worry about these seemingly poor headroom and distortion figures. Although I haven't had file:///H|/SOS%2004-10/Crosstalk.htm (1 of 3)9/26/2005 12:37:33 AM

Crosstalk

the chance to read this PC Pro roundup yet myself, it's already the subject of much mirth on the SOS Forums, since at least some of its findings seem to be seriously flawed. For a start, no soundcard accepts signals above an input signal level that corresponds to 0dBFS — this is by definition the point at which digital clipping commences, and I can only think that the PC Pro tests went slightly beyond the input level needed to do this, which would certainly account for those high distortion figures. In practice, when recording with a digital soundcard it's very important that you allow at least a few dBs of headroom to avoid clipping. My own tests carried out for the SOS review of the 1820M gave a THD (Total Harmonic Distortion) measurement of an excellent 0.0009% under these conditions. I also measured the Emu 1820M's dynamic range as 117.8dBA, nearly 12dB better than Creative's Audigy 2 ZS Platinum Pro at 106dBA. Both of these figures are reasonably in line with the manufacturers' own figures, and given that both cards are essentially from the same manufacturer (Emu is owned by Creative, and the 1820M bears the logo 'Creative Professional' on its front panel), I think even they will have raised an eyebrow at the PC Pro figures.

Bangers And Mash-ups Just one thing regarding one of the Q&A entries in the September issue, about using phase cancellation to remove the vocals from a track — while Hugh Robjohns' reply states that it's impossible to isolate a vocal using the phase inversion trick, in my experience this is not the case. You can isolate a vocal if you have an instrumental version of the same track, where the instrumental is the same length and its frequency content is identical (minus the vox) to the vocal track. Simply line up the two tracks (vocal and instrumental) in an audio editor, invert the phase of one and mix the two together. The identical frequencies cancel each other out, leaving just the vocal. This is a common trick used by many people, especially on the dance scene, to create bootlegs and mash-ups. A recent example was Justin Timberlake's 'Like I Love You'. The CD single consisted of the full vocal mix and an instrumental, both exactly the same length, making it easy to extract the vocal. And more recently, a vinyl release of Brandy's 'Lets Talk About Our Love' also features an instrumental, hence the flood of bootlegs created using this track at the moment. However, I'm surprised that people have achieved such good results from vinyl, as once the tracks pass through the processes needed to get them into the digital domain, artifacts which would affect the crucial need for the frequencies to be identical would surely appear. Maybe they just got lucky?

file:///H|/SOS%2004-10/Crosstalk.htm (2 of 3)9/26/2005 12:37:33 AM

Crosstalk

Sara Farina News Editor David Greeves replies: On Hugh's behalf, let me first of all say that, in his reply, he was only talking about what could be removed using phase cancellation on a single stereo track, using one channel against the other. But you're absolutley right — combining a vocal mix with an instrumental mix while inverting the phase of one of them will cancel out everything but the vocal, so long as the two are identical in all other respects. If the vocal and instrumental versions have been mixed or mastered differently to even the smallest degree, the trick won't work. The material that you want to remove from the two waveforms must be identical in both frequency and amplitude if it is to cancel out completely. I can well believe that this can be done in the case of the recent releases you mention, though — while record companies will never openly condone the bootleggers, most now realise the value of the publicity they receive when a fashionable 'mash-up' of one of their releases causes a stir in the clubs. In addition to including instrumental versions on singles, the authorised release of a cappella (vocals only) versions is practically an open invitation to remixers — the a cappella version of Jay Z's The Black Album, which spawned DJ Dangermouse's controversial The Grey Album, is perhaps the most prominent recent example. Such is the cachet associated with bootlegs and white labels (in dance music, at least), that it's not uncommon for record companies to try and promote their records by putting out legitimate copies made to look like bootlegs! In regard to using vinyl, I would think that small variations in the speed of the turntable would be a bigger problem if transferring tracks to a digital system one at a time. Of course, if you're performing the phase cancellation trick on the left and right channels of a single track, you can do it all in the analogue domain (provided you have a reasonably well-specified mixer) and avoid these problems. Published in SOS October 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


file:///H|/SOS%2004-10/Crosstalk.htm (3 of 3)9/26/2005 12:37:33 AM


In this article:


Studio Layout Philip Taysom Being Boiled Published in SOS October 2004 Beginnings Are Friends Electric? Print article : Close window Making It Work People : ReaderZone Selected Gear List Problems Going Soft Museum Or Studio? Open to the Public? David Hughes

One Christmas, when I was just three or four years old, my father took me to Fenwick's department store in Newcastle-upon-Tyne to meet Santa Claus. Fenwick's toy department is legendary because every Christmas it's crammed to the rafters with more goodies than you can shake a djembe at. Standing in the middle of that crowded room, I'd never seen so many 'must have' toys in my life. Much to my father's astonishment, I found myself absolutely speechless. Forty years later I'm standing in Philip Taysom's studio and the very same emotions come flooding back.

Photos by David Hughes Philip Taysom (right) and studio technician Roy Harrison in Philip's converted loft studio.

Walk through the front door of Philip's studio and you're immediately confronted by two Fairlight CMIs, both recently restored. To the left is a stack of three PPG Wave synthesizers and an even rarer Wavecomputer 360. "These are here because we've run out of room upstairs," says Philip with a smile. Gosh.

Studio Layout Philip's studio is located on the first floor of a converted barn. It's an unusual design for a studio, around 35 feet in length, roughly 15 feet wide, and with steeply sloping sides. Anyone taller than five foot four is in for a stiff neck. At the far end of the studio is the main Pro Tools console, built around a Digidesign Control 24 desk, while the right-hand side of the studio is taken up with seven keyboard bays, each with four tiers. Similarly, the left-hand side of the studio houses a huge number of synth modules. It's hard to think of a single manufacturer, past or present, who isn't represented here. file:///H|/SOS%2004-10/Readerzone%20%20Synthfest%21.htm (1 of 9)9/26/2005 12:37:36 AM


Dominating the studio is the analogue modular synth area, a semi-enclosed space accommodating, amongst others, a complete Roland System 700, a 23-module Roland System 100M, and Polyfusion, Moog and Korg semi-modulars. Finishing the square is a complete PPG setup comprising a Waveterm B, two EVUs (Expansion Voice Units), Wave 2.3 and a matching PRK FD keyboard. The main floor space is shared between a Roland V-Drum kit (modified to trigger Simmons and Dynacord analogue drums), another recently restored Fairlight Series IIx and a Synclavier 9600 'master keyboard.'

Being Boiled With so much equipment in the studio, fan noise is obviously a problem. Some of the hardware has been enclosed in cabinets, which proved brilliant at deadening the noise but equally brilliant at frying the equipment, because there was virtually no airflow inside! Philip: "Without cooling, the cabinets can get to around 42 degrees centigrade in a matter of minutes, so a set of triple-insulated doors — basically conservatory doors — were installed. The internal frames were then mounted on three blocks of neoprene separated by hardwood, so that they were acoustically isolated from the rest of the studio. Finally, 7.5 kilowatts of air conditioning were mounted under each cabinet." That sounds like a lot of cooling. Does it work? "Yes, but in summer the Synclavier cabinets still get to about 24 or 25 degrees."

Beginnings "A mate and I had a bet that if I was able to give up smoking for a year I would buy a Minimoog — something I'd always lusted after," says Philip. During that year Philip began researching a 'wish list', because so much vintage equipment had become affordable as everyone rushed towards soft synths. Philip cites his Prophet T8 as a good example. He'd originally seen this instrument demo'd at Syco in the 1980s for around £8000, but the same machine could now be bought for around £1200. Philip continues. "In the '80s I had a lot of gear but I sold it all and thought I'd never return to owning a studio again — but in 1996 I bought a Korg 01/W Pro from a guy in Hull. It was his infectious enthusiasm that got me interested again. Then my friend Paul Gilby [brother of Ian Gilby and co-founder of SOS] taught me so much about recording and mixing, explaining the intricacies of EQ and how to construct the mix. Paul's a former lecturer with great skill in explaining complex theories — so I got free lessons from a master!"

The rear of the complex patchbays. Wiring of the studio took around 18 months in total.

The Korg synth was the start of something big, and now, around three years later, Philip has an astonishing array of kit. The first step in making it all accessible and usable was to solve some basic engineering problems. How, for instance, do you move something file:///H|/SOS%2004-10/Readerzone%20%20Synthfest%21.htm (2 of 9)9/26/2005 12:37:36 AM


as heavy as, say, a CS80 or a DX1? Philip found a manufacturer of shelf-runners so that moving these instruments would require no effort. The runners currently installed are rated at 400 pounds per side, so moving such a weighty keyboard is now very simple indeed and can be accomplished with little more than the push of a hand. The next step was to maximise the available space through careful design, so a local cabinet maker, Martin Hall, was contracted to fit out the studio area with custom shelving and racks in cherry wood.

Are Friends Electric? "The key thing I insisted on was that everything had to be cabled up," explains Philip. "I wasn't prepared to have wiring lying across the floor." The wiring solution is, indeed, very elegant and took an amazing 18 months to complete. Behind the equipment in the main rack there's another sub-rack which carries a set of plates. Everything is connected to multicore cables on these plates, which are then mounted sequentially so that, for example, slot three connects to bay three, and so on. Each sub-rack is capable of taking 12 instruments, four per keyboard bay, and there's complete flexibility as to where an instrument sits. Everything is connected — MIDI, CV, audio, digital audio — and all of the cabling goes back to three patchbay matrices in the desk: one for analogue audio, one for MIDI and one for digital audio. In addition, all of the wiring is balanced, and some provision has also been made for optical. Given the slightly unusual dimensions of the studio, the length of the cable runs would require some fairly novel engineering solutions. A good example is the recently installed Roland V-Synth. To drag and drop samples from a Mac over USB is fine over short distances, but in this instance, where the cable run is nearly 40 feet, USB doesn't work too well. To remedy this problem, a pair of USB extenders that utilise UTP wiring, made by Geffen in the USA, were installed. The USB connection will now work reliably over 300 feet. "An expensive but effective solution," says Philip.

Even heavy keyboards such as the Oberheim 8-voice slide out from their bays easily on heavy-duty shelf runners.

I asked if there had been any major problems with the cabling system. "At the modular end we made some mistakes," responds Philip. "We designed it with everything physically in a fixed location and totally unusable. What we've now done is to build a 48way patch-line system within the modular area so, rather than having a collection of modulars, there is now one very large modular made up of all the systems. If I want a patch of a Roland System 700 VCO filtered by the RSF Kobol, gated by a Polyfusion EG, with a final VCA stage from the Oakley modular, it's easy. The patch matrix makes that file:///H|/SOS%2004-10/Readerzone%20%20Synthfest%21.htm (3 of 9)9/26/2005 12:37:36 AM


possible. There are also localised Kenton Pro 2000 interfaces — using both primary channels of CV and gate and all six aux outs — underneath each bay." With everything up and running, the studio draws about 85 amps and a second 100-amp supply has just been installed by the local grid operator, specifically for the studio. However, they're still exceeding their limit, so negotiations for another 60- or 100-amp supply dedicated to the air-conditioning are currently under way, to provide some more headroom on the main supply. "That's going to cost between £10,000 and £20,000," explains Philip. "But it's a lot cheaper than moving to another building." With so many instruments installed in the studio, I was curious to know how they switched the instruments on and off. Philip explains. "I spent several years designing data centres, so we've applied some of the techniques I learned there to develop powerStudio diagram by Tom Flint. control systems inside the racks. Some really clever electricians from Spa Contracting in Harrogate came up with a brilliant solution using remote controllers. We can switch the whole lot on and off with just a key that has no current going through it, so it's completely safe. "The next thing we have to install is a fire-control system. That's going in over the next couple of weeks. We'll get both audible and visual alarms if the mains fails in either of the cabinets. In addition, if it detects smoke, it will kill the mains locally to the cabinets and then, ultimately, to the entire studio." Listening to a mix through the main desk, I noticed some mains hum — hardly surprising, given the number of instruments in the studio — and I wondered if there was an earthing problem somewhere. Philip explains: "This is a 450-year-old house, and we didn't realise that the whole studio was earthed off the earthing rod for the main house. An earthing rod should be, say between six feet and nine feet long. We went to investigate, pulled the earth out and only four inches came away because the rest had just rotted! Basically, we had no earth. We've had a massive job re-earthing. It's already making a difference but there's much more work to do."

Making It Work The studio's interfacing and composition facilities are centred around Emagic's Logic and Philip makes extensive use of the Emagic Unitor 8 and AMT8 MIDI boxes, which provide up to 60 ports of 16 MIDI channels each on the studio G5 Macintosh. "I'm currently using 48 ports," adds Philip. "And we're still expanding." All the MIDI sockets from each of the keyboard and rack bays are presented on a large patch matrix at the back of the desk, and short MIDI leads are used to route each instrument to a Logic MIDI port. However, many of the older instruments either have no

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MIDI Thru, or the Thru is highly inefficient (on some of the gear it can add a 200ms latency!). To open up some of those MIDI Thrus, a number of 8-way Thru boxes have been cascaded together. Philip admits that it's a slightly Heath-Robinson arrangement, but the cabling is all colour-coded — so, for instance, blue means a MIDI Out from the instrument, while red signifies a MIDI In. "I wish MIDI had been on a lockable connector instead of on a DIN," observes Philip. "It's so easy to come back here, brush past something and break a connection."

The modular synth area.

Philip's original choice of mixer was a Mackie CR1604, although he quickly upgraded to an Allen & Heath GS3000. "I had a look at the Mackie Digital 8-Buss but when configured with solely analogue ports that was too expensive for what it offered," says Philip. "I looked at some really silly options, such as the Amek Mozart or Big. I even toyed with the idea of a second-user SSL. Then I had a demo of a Sony Oxford and someone told me that they were developing the Oxford EQ for Pro Tools. I'd never considered Pro Tools because I'd always thought of it as a recording system and I very rarely record anything multitrack." A demo of Pro Tools HD with a Control 24 left Philip convinced: "I actually get very little time to come up here — sometimes as little as 30 minutes a day. To work on a different idea it can take more than an hour to set up an analogue desk, re-patch the matrix and get everything ready for a session. So Pro Tools is ideal. I simply recall the session file from disk, complete with configuration." The number of inputs available is obviously a major strength. At the moment, the installation uses 96, but the wiring is in place to extend this up to 128 inputs. Another strength is that all of the patching is in software. Philip: "For the Oberheim 8-voice synth there's a patch that adds just a tiny amount of reverb. I don't have to start patching cables to recall this, simply drag and drop it on screen. As I get more time up here we can start to deal with the really noisy stuff like the [Sequential] Pro One and the [Korg] MonoPoly and patch in a permanent gate on the relevant channel within Pro Tools for them, just to catch some of the low earth noise that you get because they're old machines." I noticed that there are no mixing facilities in the modular area and wondered if this caused any difficulties. Philip: "While we can't edit the mix when we're down there, we can at least build up a layer of audio that we can control from up here or via the Synclavier. I've been looking at the new Digidesign eight-channel control surface and plan on trying out one in the modular area, connected via the Geffen USB extenders we've already used with the Roland V-Synth." Monitoring duties are currently taken by Mackie HR824s and HR626s wired with a Mackie subwoofer for 5.1 monitoring, while a pair of Genelec 1029s provides localised monitoring in the modular-synth area. "I want more advanced monitoring," comments file:///H|/SOS%2004-10/Readerzone%20%20Synthfest%21.htm (5 of 9)9/26/2005 12:37:36 AM


Philip. "So I've been looking at Genelec 1038s, but I really favour the Dynaudio Air 25; with the built-in DSP I can compensate for the acoustics of the room. It's an idea I really like."

Selected Gear List Oberheim OBXa

MODULAR SYNTHESIZERS

Alesis Andromeda

Oberheim OB12

Formant Modular (43 modules)

Alesis Ion

Octave Cat

Polyfusion Modular

ARP Odyssey Mk I

PPG EVU (x3)

Roland System 100M

ARP Odyssey MK II

PPG PRK Keyboard

Roland System 700

Clavia Nord Lead

PPG Wave 2.2 (x2)

Elka Synthex

PPG Wave 2.3 (x2)

Access Virus C

Fairlight CMI Mark II (x3)

PPG Waveterm (x2)

Clavia Nord Lead 3

Fairlight CMI Mark III

Roland Jupiter 8

Clavia Nord Micro Modular

Gleeman Pentaphonic

Roland V-Synth

Dave Smith Evolver

Hartmann Neuron

SCI Prophet 5

Korg Triton

Korg Karma

SCI Prophet 10

Kurzweil K2000r

Korg Polysix

SCI Prophet T8

Oberheim OBMx

SCI Pro One

Novation Supernova

Synton Syrinx

Roland MKS30 Planet S

Waldorf Wave

Roland MKS80 Super Jupiter (x2)

Yamaha CS80

Roland JD990

Yamaha DX1

Roland JP8080

KEYBOARDS

Korg PS semi-modulars (x2) Korg MS20 semi-modular (x2) Korg Triton Moog Memorymoog Moog Minimoog Moog Prodigy

MODULES

Roland VariOS

MISCELLANEOUS

Moog Source

Apple Macintosh G5 computer (x3)

Moog Voyager

Digidesign Control 24 control surface

NED Synclavier (x2)

Digidesign Pro Tools HD system

Oberheim 8-voice Oberheim Matrix 12 Oberheim OB8


Symbolic Sound Capybara 320 DSP platform


Just some of the racked-up keyboards the studio is crammed with. The keyboard bays are set up so that all the gear is readily movable and will function when plugged elsewhere.

The unusual transparent Gleeman Pentaphonic synth.

Problems With a project of this magnitude, I wondered if there had been any major changes or redesigns along the way. For instance, I noticed that some of the equipment is now being bolted to the walls. Philip explains: "Yes, we've run out of space already and we're now bolting stuff to the bargeboards. For instance, with the Voyetra keyboard you can play it via MIDI but it's just so much nicer to play it via the VPK keyboard, the way it was intended. So that's why the keyboard is bolted above it. And with the Virus C it makes more sense to mount it on the wall instead of taking up another 4U of rack space. Rack space really is at a premium. I suppose we could lose a keyboard bay to make way for another rack, but I'm not keen to do that." Philip's studio technician, Roy Harrison, briefly joins the conversation. Roy: "This is certainly the biggest job I've ever done in all of my career. Every keyboard has every facility. You can pull one out and put another back in to the same space and it'll just work. And it's not just audio, it's power, MIDI In, Out and Thru and now digital audio. And all patchable too." Philip: "Out of the thousands and thousands of cables we've dragged through here, there's only been one minor problem. It's a real testament to Roy's ability. How much cable is there in here? Somewhere between four and seven kilometres?" "Something like that," responds Roy.

Going Soft Only one of the studio's Apple Mac computers (mostly G5s) is dedicated to running soft synths. "I'm not keen on them," says Philip. "In my personal opinion, the only real soft synth is something like the [Soundart] Chameleon or Symbolic Sound's Kyma, running on their Capybara hardware." Philip's Chameleon is currently running a Modulus



Electronics Monowave II skin, and he prefers this approach over a software package because it's a dedicated piece of hardware — unlike a PC, which is also trying to run an OS, get email, web browse, and so on — tasks which don't really mix with being a musical instrument. But while Philip doesn't like soft synths in general, he does have one or two favourite musical software applications, one of which is Ableton's Live. "As a groove tool it's superb! I can drag loops out of the Synclavier, drop them into Live and they're stretched to fit. Along the same lines, I've been playing with Melodyne recently and it's another fine tool for creativity."

Part of the rack synth area. Above the main racks, some gear is now bolted to the wall boards to make the most of the available space.

Looking around the studio, I notice that there isn't much outboard equipment, and I'm surprised by Philip's explanation, given that he's just told me he's not a huge fan of software synths. "We're moving over exclusively to plug-ins for processing. As I mentioned, I've bought the Sony Oxford EQ and I didn't see much point in going from digital to analogue to digital when a software solution is available. Most outboard these days uses a DSP anyway, so I might as well do it in Pro Tools!"

Museum Or Studio? The centrepiece of the studio is the Synclavier. Philip's machine came originally from the BBC film and TV studios in Bristol and so has all of their libraries installed. "It was used on many of the Natural History programmes, such as Life On Earth," says Philip. "This Synclavier also has the entire Lucasfilm library. In total, there's about 30GB of sound data." The Synclavier is used as a humble MIDI master keyboard most of the time, although recently it has also been used more and more as a sequencing tool, because its sequencer provides a really good way to rough out an idea first before fully arranging it via MIDI with the rest of the studio instruments. Despite the fact that there's so much vintage gear in here, Philip is keen to emphasise that "this isn't a collection — it's a studio. In a collection, you would probably aim to have one of everything. But in here, for example, there isn't an EDP Wasp. That's because, personally, I don't like the sound of the Wasp. I think you can make better sounds with other devices. Everything is here for its sonic merits, not its rarity. Also, we only install instruments that are fully working, because the studio isn't a sort of storage area or a museum. Everything in the studio must be working, more or less perfectly. I'd love to bring the old Polymoog up but it's just not very reliable." I asked if Philip's original wish list had changed significantly. "I went to the last Frankfurt Musikmesse, where I fell deeply for the Jomox SunSyn and the Vermona DRM drum file:///H|/SOS%2004-10/Readerzone%20%20Synthfest%21.htm (8 of 9)9/26/2005 12:37:36 AM


synth," says Philip. "Also, I've just found a PPG System 300 modular synthesizer in France, and you have to be very opportunistic. If I don't buy it now I probably won't get the chance to buy one for perhaps another 15 years — so the car's loaded up and I'm off tomorrow! There's another couple of instruments I'm frantically trying to find, such as a Movement Drum Computer — both the Mark I and the Mark II. The Mark II featured user sampling — which is amazing for its time — but I just cannot find one."

The venerable NED Synclavier, ex-BBC and stuffed with professional sound libraries.

Open to the Public? "I designed the studio for my own use and pleasure," says Philip. "I hadn't really thought that anyone would want to use it on a commercial basis." However, word has apparently spread and Philip has recently received a number of requests from musicians keen to use his studio. "I was blown away when people said it was pretty amazing and it should be a facility open to musicians looking for a wide sound palette," says Philip. "Now we're talking to musicians who want to come up and use the studio commercially. We're refining some guidelines about how it might be used but in general I'm really excited about the prospect." Published in SOS October 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



Sounding Off: Your CV

In this article:

About The Author

Sounding Off: Your CV Paul Bower Published in SOS October 2004 Print article : Close window

People : Sounding Off

Paul Bower explains why the CV may be more important than you think — even in the music business... Paul Bower

There's been a fair amount said about the qualities needed for a successful career in the studio business in the past months — but just how do you get on the first rung of this ladder? The answer is the not-so-humble CV. It's a subject that seems to be increasingly ignored by trainee sound engineers, or just omitted from their training. Recent CVs I've received include the perennial spelling of Cubase with a 'Q', electronic documents that spill out half a dozen blank pages, and the whole text of the CV jumping back and forth from the first to third person (read just about any CV and you'll see); yet these are job applications from degree graduates or final-year students. I've had calls from young hopefuls who've even been advised by their centre of learning that submitting a CV would be too formal in the music industry. Remember this: the phrase 'music business' is only 35 percent music and 65 percent business, so lesson one — submit that CV. While we're at it, lesson two — check your spelling (in Sound On Sound obviously). If you can't correctly spell the name of the application you're using, why should I have any confidence in your ability with it ? I know there must be form CVs out there, but if you're at college, and you copy the form CV handed out, as does everyone else, and then you all send your CVs to the same list of studios, what happens? That Stepford-like mantra "I am equally capable of working alone or as part of a team." As an employer, I want individuality, integrity, and attention to detail. At the bottom of the ladder, assistant engineers, tape-ops, runners, and so on are like Victorian children — they shouldn't speak unless they're spoken to, and they'll be expected to clean chimneys on a regular basis. You can teach the technical stuff to a monkey. The attention to detail, however — the natural urge to tidy away discarded cables, the ability to disappear when the artist is about to throw a wobbly — is something far harder to teach.

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And so we move onto showreels. Many would-be engineers think it necessary to send in a CD of tracks they've produced either at college or at home, or even for a commercial session. I'm personally against the use of showreels as they tend to have little relevance to a large studio, and really can only serve to undermine any prior good thoughts I'd had about that candidate (through reading their CV of course). In my experience, showreels are, on the whole, badly labelled, and in the age where the majority of masters are archived on anonymouslooking five-inch polycarbonate discs, what do you think I'll notice first about your showreel? About The Author

Back to the CVs. "I am currently studying at [insert name of college here] in London and will be finished my studies Paul Bower has this August." Our friend then goes on, as many do, to list worked in a well every single piece of equipment he's ever used, most of it respected studio for a few months duplicated in the 'tracked with' and 'mixed with' sections. I shy of a decade. also just got an email from Luke: "i wanna get into music By the time you an would like 2 work in ur studio, so if there is any jobs read this it will be available plz contact me!" OK, so maybe our friend is still closed. in school, just chancing his arm, but Luke — if you're serious, that's not a good start. Ben writes from Woking with a letter containing a little information about himself rather than submitting a formal CV — this is not a bad thing in itself, and does help to give the applicant an air of individuality. However, the opening gambit reads "I have recently quit university after completing my first year due to lack of interest and enthusiasm for my chosen course; Media Studies..." I won't go into Ben's haphazard career approach, but this is certainly not the first sentence I expected to leap off the page (and there was the minor problem of writing the wrong address on the envelope and just scribbling it out). I'm not trying to uphold the Queen's English — for instance I use the American spelling of 'rumor', and I write all my emails in lower case. And I start sentences with 'and'. My argument is this: you are spending years of your time and thousands of pounds in training for your career of choice, and you're just going to piss it all up against the wall because of one very tiny detail. OK, maybe I've concentrated a little too much on what's wrong with many CVs — so what would I like to see ? I've not seen a single CV that mentions the applicant has knowledge of the paperwork side of the industry. If I had and I was hiring, that CV would be at the top of the pile. Various sample usage forms, BPI Consent forms, the odd Joint Notification Of Works, and those brooding MCPS Studio Recording forms, even the simple tracksheet are just as essential as knowing which is the business end of a microphone. Get some practical experience 'moving air' in your educational establishment of choice. Learn different computer applications for both PC and Mac, not just the one your accredited music school tells you to, and if you can't get any practical studio experience, get some live music experience.



Finally, you have the training, the certificate, and now the killer CV, but where do you get the experience, and more importantly, are you sure you want a career in what is basically a declining industry? Oops, there goes the phone again — ah yes, it's the liquidator, but that, boys and girls is another story... Published in SOS October 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:

Studio SOS

Low-end Monitoring Dorian Kelly Problems Vocal Recording & MIDI Tips Published in SOS October 2004 Reverb Overhaul Print article : Close window Processing The Final Mix People : Studio SOS Dorian's Comments Optimising The PC All Of A Dither

The SOS team ride to the rescue of a budding media composer who's having trouble with his mixes. Paul White & Hugh Robjohns

Dorian Kelly has long-term plans to compose music for TV and film after successfully completing a media composing course (the home-study Music For The Media course run by Guy Michelmore from Atlantic Studios), but in order to gain experience he has set up his own Steinberg Cubase SXbased home studio where he works on every type of music possible, from classical to pop. This seems to be a Dorian's monitors were set up on shelving, which was not rigid enough for the purpose very sensible and practical approach, and which therefore compromised the bass but although Dorian's room has had response of his Spirit Absolute II monitors, the benefit of some acoustic treatment, speakers which already have limited low-end he was worried that he couldn't hear extension. enough low end in the studio, which left him with mixes that sounded bass heavy when played elsewhere. He was also keen to try to improve his mixing technique, so he called SOS.

Low-end Monitoring Problems When we arrived at Dorian's Reading home, where his studio is located in a converted garage, the kettle was boiling and Dorian had assembled a vast plate of chocolate digestive biscuits and croissants — so he was clearly aware of all the necessary Studio SOS protocols! While attempting to reduce the size of the cookie mountain, we played a CD of various test tracks over Dorian's system, file:///H|/SOS%2004-10/Studio%20SOS.htm (1 of 8)9/26/2005 12:37:41 AM

Studio SOS

which comprised a small Behringer mixer to handle monitoring sources, a Samson power amplifier, and a pair of passive Spirit Absolute II speakers. The recording system was based on a purpose-built PC running Cubase SX with another small Behringer mixer to manage the input signals. An Emu Proteus 2000 provided the only external MIDI sounds. As we have discovered on previous occasions, the Absolute II monitors don't have a great deal of low-end extension. This, combined with the room size and shape plus 'less than ideal' speaker mounting did indeed produce a bass-light result, though the mid-range and high end were fine. While Dorian's room is adequately long, it is only a couple of metres wide, and in a room this shape there's little alternative than to work across the width of the room, as there would be insufficient space to set up the equipment in an ergonomic fashion if working across the narrow axis of the room. The wall behind the mixing position was already treated with some acoustic foam tiles and a few small areas of thick carpet tile, while the ceiling was largely covered in fibrous office-style acoustic tiles — the overall result was reasonably well controlled, without any obvious boomy bass notes. The plasterboard walls, a cupboard, the doors, and two windows all contributed to some fortuitous lowfrequency trapping, so the main problem was the reduced level of bass from the monitors, rather than uneven bass.

Even when Paul switched the Absolute IIs for his own Mackie HR624s, which have a better bass response, some adjustment of the speakers' onboard frequency tailoring switches was required to give a representative low-end picture.

Dorian had his speakers set up on foam speaker pads that rested on the freestanding pine shelving he used to store all his studio accessories, both above and to either side of the mixing position. Using these foam pads is preferable to standing the speakers directly on non-rigid shelves, but there's no real substitute for solid stands or wall brackets. However, in this case these couldn't be accommodated for space reasons. To evaluate the room further, we set up our Mackie HR624 active monitors in place of the Absolute IIs and played the test CD again. The result was better, but we still weren't hearing as much low end as we are used to from these speakers, so the room was clearly a contributing factor. Setting the bass switches on the monitors from their half-space mode to full-space mode delivered a better balance, even though the speakers were being used close to a wall where the half space setting was technically correct. The full-space setting is normally only used when the speakers are set up away from walls, but as it worked we weren't going to knock it! After all, getting any kind of decent monitoring system working in a room this narrow is inevitably a compromise. After hearing the system with the Mackies in place and after playing some of his

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Studio SOS

own mixes through them, Dorian agreed that he should change his monitors as soon as possible, but because of budget constraints he felt he had to look for something rather cheaper than the Mackies. He wanted to go for active models, and after looking at the options he narrowed his choices down to Alesis, KRK, and Fostex units. On the subject of mounting the monitors, the existing position was just a little too high, as the tweeters should ideally be aimed at the engineer's head and not a few inches above it. This could be remedied by either lowering the shelves or by angling the speaker downwards and inwards, but it would also be beneficial to provide a more stable mounting platform. You could use a metal bracket to secure the speaker shelf to the wall behind it and then position a small paving slab (around 300mm square) as a base for the speaker — the additional mass of the slab helps to damp and control vibrations. You might also want to stand the speaker on the slab using blobs of Blu-Tac under each corner rather than relying on foam. In a fair and just world, the 'slab' approach should tighten up the bass end to a worthwhile extent.

Vocal Recording & MIDI Tips Dorian played us some vocals he'd recorded where the singer had stood at the opposite end of the room from the computer and sung with the back of the mic facing the mixing position to minimise noise pickup from the computer. The results were reasonable, but some room coloration was evident and, as vocals often need compressing to make them sit properly in a mix, the room ambience is usually emphasised further. Dorian had bought one of those Auralex foam gizmos that clip over the back of the mic to attenuate sound coming from that direction, but the wall directly behind the singer was fairly bare and had a mirror mounted on it, all of which conspired to bounce sound back into the microphone. A practical solution to this is to hang a double duvet from a rail fixed to the ceiling, so that it is directly behind the singer when recording. In this way, ambient sounds no longer reflect back into the front of the microphone, and the room coloration is greatly reduced. If the duvet can be hung from a curved rail so that it also comes around the sides of the singer slightly, then so much the better. Dorian liked this idea, as it was simple and inexpensive and the duvet could be pushed to one side when not needed provided that it was hung using curtain rings.

A combination of two reverbs in series was required to get the best out of Cubase SX's built-in plug-ins: the first (left) added a short brighter ambience to give a sense of roominess; and the second (right) contributed a subtle reverb tail without clogging up the mix.

Next Dorian played us some of his classical-style compositions, which had been file:///H|/SOS%2004-10/Studio%20SOS.htm (3 of 8)9/26/2005 12:37:41 AM

Studio SOS

created using Steinberg Halion samples combined with some tracks of him playing real trumpet and French horn. Overall, the sound was well balanced, though we did identify three specific problem areas. The most obvious problem was that some of the parts sounded quite stilted, and this turned out to be because Dorian composed his music using Sibelius before transferring it into Cubase SX to mix. As Sibelius creates MIDI files based on the bars and notes of the score, the data is inherently tightly quantised, but in addition Dorian wasn't making use of those instruments that supported legato mode to allow overlapping or consecutive notes to flow smoothly into each other. Selecting the desired MIDI notes in the Cubase SX grid edit page and then choosing Legato from the MIDI menu removed spaces from between the notes and allowed the sound to flow more naturally, without constant re-triggering of samples. This is particularly evident on string sounds (which 'suck' unnaturally if legato isn't used correctly), but solo wind instruments and bowed solo strings also tend to play legato a lot of the time, so trying to simulate a performance from distinctly separate notes will invariably sound mechanical. We tweaked a couple of parts to demonstrate the difference when parts were played legato, and Dorian immediately saw the advantage. Some instrument patches are created specifically with legato playing in mind, and some instruments have dedicated legato trigger modes which prevent the envelope from re-triggering when a new note is played, provided that it overlaps the previous one. These techniques are all worth exploring, as is the trick of using a MIDI volume pedal (transmitting MIDI Continuous Controller number seven) when recording strings to create more natural-sounding swells and crescendos.

Reverb Overhaul The other obvious shortcoming in Dorian's tracks was the rather limited reverb that comes with Cubase SX. Although the mix was obviously reverberant, there was no sense of space or acoustic — an obvious failing in the context of orchestral music. To improve on this, Dorian has the choice of buying a convolution-based reverb such as Altiverb (which will eat up a lot of processing power), adding a hardware DSP card such as the TC Electronic Powercore or Universal Audio UAD1, or plumbing in an external hardware reverb processor such as a TC Electronic M*One XL or a Lexicon MPX550. Because Dorian's M Audio Firewire Audiophile interface has two analogue inputs and outputs, plus S/ PDIF in and out, a hardware reverb with digital I/O could be plumbed into the system via S/PDIF to save unnecessary conversion. Any latency affecting the external reverb would be negligible compared to the usual pre-delay added to reverb — provided that the reverb was used in an aux send/return loop and set to 100 percent wet. However, my own preference would be for a card-based system, as you get the quality of hardware but the effect settings are stored as part of the Cubase SX song. So when you revisit old projects, you don't spend half the day trying to get


Studio SOS

back to where you started. You can also run multiple instances of the reverb plugin with a card-based system. As an experiment, however, we decided to see what could be achieved using the Cubase SX native Reverb A, and the strategy we tried was to use two reverbs in series, the first set to as short a decay time as possible to simulate an earlyreflections/ambience type of treatment. This was adjusted by ear to give a coloured roominess to the sound (we ended up with almost as much reverb as dry signal) and a short pre-delay of between 10-20ms seemed to help. The ambience reverb was followed by a longer concert hall effect with a 70-80ms predelay to provide the tail of the reverb. We set this at a much lower level so that it added a subtle reverb tail without choking the gaps in the music, and the result was surprisingly good. Obviously it was not as smooth as a real high-end reverb, but at least we managed to create a sense of space and depth without clogging up the mix with an excessively loud reverb tail. The use of an ambience-like reverb to create the basic sound character certainly helped the overall sound become more homogeneous, and the performance now sounded as though it had taken place in a real space.

Processing The Final Mix The final challenge was to process the final mix to create a more commercial, 'TV-friendly' sound. Dorian had tried some fairly radical EQ that combined shelving bass boost with a wide mid-range cut to generate something like the traditional smile curve (more of a manic leer really!). However, Hugh and I felt that using only the two parametric sections of the Cubase four-band EQ would give us more control, so we added a gentle 80Hz hump balanced by a broad 13kHz boost to add air and sizzle to the top end without making the mid-range sound harsh. This was teamed with a compressor setting using a low 1:1.2 ratio, a fairly fast attack, and the automatic release setting. The threshold was turned down until the gain-reduction meter showed around 5-6dB on the louder passages, which left us with a threshold setting of around -35dBFS. This strategy reduces the whole dynamic range of the material in a very gentle way, and differs from the more usual tracking approach where a higher threshold is set and then everything above the threshold is squashed much more aggressively. file:///H|/SOS%2004-10/Studio%20SOS.htm (5 of 8)9/26/2005 12:37:41 AM

With the Mackie monitors in place, Paul had a look at the processing Dorian had used on his final mixes. His EQ settings in particular were felt to be a little extreme, so a gentler 'smile curve' EQ was used instead.

Studio SOS

Dorian then played us a pop song he'd recorded with a female vocalist. Her performance lacked a little conviction, but was basically sound, and we tried to treat it with a more robust 8:1 compression ratio, the fastest attack, and the automatic release mode. Again, the threshold was adjusted to show the desired amount of gain reduction, in this case around 8dB on peaks. This worked fine, making the vocal sit better in the mix and also making the sound appear more confident. Dorian was a little unsure about the best reverb treatment to use so we again tried the approach of layering a short pre-delayed bright setting to create an almost slapback ambience together with a longer reverb tail at a lower level and with more pre-delay. After a little tweaking we arrived at something more contemporary sounding, where the vocal took command of the mix, rather than apologising for being there as it had before! A little high EQ to sharpen it up and we were done. When discussing how the vocal was recorded, Dorian also admitted that he hadn't added any reverb to the monitor mix, though Cubase SX does allow this and it can help make a singer feel more confident. However, you do need to check with the singer how much reverb they like to hear in their cans, as everyone seems to have their own preferences in this area. Too much is just as bad as too little. It can also make a difference to reverse the polarity of the headphone feed while recording, which can be achieved by using the phase-reverse button in the dynamics section of the vocal channel in the case of Cubase SX. This affects the way the singer hears the sound from the headphones when it combines with what they hear of themselves directly, and usually one polarity sounds more comfortable than the other.

Dorian's Comments "Despite feeling a little nervous of what Paul and Hugh would make of my setup and my music, it was great to have the SOS team here. Being a musician first and an engineer second, I do not always find it easy to resolve technical problems I encounter. Paul's Mackie monitors made a big difference, and after hearing them in action I will definitely invest in a pair of active monitors in the near future (probably the Alesis M1 Active MkIIs). It's very reassuring to know that there is not too much wrong with the room itself. "Paul's advice on the MIDI files will be very useful. Most of the time I am trying to recreate a live music sound by combining some audio tracks (such as trumpet, French horn, and violin) with samples triggered over MIDI, and this is often more difficult when going for an orchestral file:///H|/SOS%2004-10/Studio%20SOS.htm (6 of 8)9/26/2005 12:37:41 AM

Studio SOS

sound. Using Paul's tips to make these sound more realistic will be very useful. If I had more space and some kind of budget I would use live musicians all the time, but at the moment that just isn't practical. "Out of all the great information that Paul and Hugh passed on, I think the most useful for me will be the compression, EQ, and reverb settings. They managed to improve my mixes enormously in a very short space of time, just using the bundled plug-ins within Cubase SX. I was particularly impressed with the reverb settings for the orchestral mix — layering two different reverbs was a great idea, and one I would not have come up with myself. Combined with subtler compression and EQ, it made the mix sound much more natural — almost like it had been processed 'invisibly', which is very important when creating convincing orchestral tracks. I am currently in the process of creating show reels to send out to every man and his dog in an attempt to actually get some work, and after the SOS visit I feel a lot more confident — I'm sure my CDs will now sound more professional as a result."

Optimising The PC Towards the end of our visit, Dorian asked a few questions from a checklist he'd been compiling before our visit. The first was about latency and buffer sizes. He wasn't quite sure what to adjust or what the best setting was, but he was aware that it was important. Latency is usually defined as the time delay between a signal being piped into a computer workstation and that same signal appearing at the monitor outputs. It is most important in the context of software instruments, as the monitoring workarounds that can be tried when recording (monitoring the vocal mic at the input to the system rather than at the output, for example) can't be used with software instruments. If the latency is too long, the time lag between pressing a key and hearing a sound will put you off playing, but on the other hand if you try to set up your system for a very low latency, the essential data buffers in RAM may be too small to allow a continuous flow of audio data, and that's when you get clicks and pops. On a modern computer like Dorian's (which has 3GB of memory), it is usually OK to set the buffer sizes to 128 or 256 samples, which at a 44.1kHz sample rate will give a low enough latency that most users will be unaware of it, yet the buffers are still large enough for reliable operation. Of course the stress on the system gets worse as you add more plug-in effects and instruments, so when the computer is pushed close to its limit, it may be necessary to increase the buffer size to avoid glitching. To minimise the CPU load, a handy tip is to scale the song display so you can see it all on one screen, which avoids the computer having to work on screen redraws during playback and when mixing. You can also increase the buffer size when you're mixing, as latency isn't an issue unless you're recording. The other useful strategy is to render instrument tracks as audio files, which saves on CPU load at the expense of a little more hard-drive activity, but in my experience most modern systems are capable of handling far more audio tracks than any sane composer would ever need unless they're scoring a Hollywood movie and creating all their ensembles by layering individual monophonic tracks. There are file:///H|/SOS%2004-10/Studio%20SOS.htm (7 of 8)9/26/2005 12:37:41 AM

Studio SOS

many articles discussing the causes of latency and its workarounds on the SOS web site, so it isn't necessary to go into more detail here.

All Of A Dither Dorian's final question involved dither. Again, he had heard that it was a good thing and was beneficial when reducing the resolution from 24 to 16 bits, but he wasn't sure where or when to apply it. In this context the reality is that dither is only of benefit if it is the very last thing you do to a piece of audio, so it's no good dithering your mixes into 16-bit files, putting them into a CD playlist, and then changing their relative levels — the act of adjusting the levels will destroy the dithering benefits. My own strategy is to record and mix everything at 24-bit resolution, then use a CD playlist compiling program (Roxio Jam for Mac in my case) that automatically applies dither during word-length reduction as the very last stage after any other adjustments you may have made before burning the CD-R. At the end of the afternoon, we felt we'd made a worthwhile amount of progress. We'd demonstrated that the room acoustics were reasonably good but that the current choice of monitoring wasn't ideal for the room. We'd also come up with some cheap and cheerful ways of improving the recorded vocal sound, and our impromptu dual-reverb treatments had worked out better than expected. We had also covered some useful compression and EQ basics, as well as made some suggestions for tweaking MIDI files to make the playing sound more realistic — specifically by using legato. The job was done, the plate of biscuits was looking decidedly well hammered, so we packed up our toys and made for home. Published in SOS October 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 Engineers Who Changed Recording

In this article:

Meek By Name, Not By Nature Tom Dowd: Change The Machine, Not The Man Bill Putnam: Mixing Technology & Business Tom Scholz: The Rockman George Massenberg

The Engineers Who Changed Recording Fathers Of Invention Published in SOS October 2004 Print article : Close window

People : Artists/Engineers/Producers/Programmers

Some of the legendary names in engineering and production didn't just make great records — they also invented equipment and techniques we take for granted today. Dan Daley

The confluence between musicians and inventors goes back hundreds, if not thousands of years. Even the words 'invent' and 'improvise' are often interchangeable. However, the ability to think outside the box, which we expect musicians to do, and yet do so in a methodical, calculated manner, is a requirement very few are expected to fulfil. Thomas Edison may have invented the recording process, but I can't recall any of his songs ('Mary Had Photo courtesy of Universal Audio A Little Lamb' does not count). Les Bench-testing new equipment in the Paul, on the other hand, figured out Universal Audio laboratory. how to record an entire band with one person in the room and did it while composing 'Vaya Con Dios'. It's worth looking at how some synthesized their creativity and inventiveness, in the process impacting our own lives and careers. And just keep in mind as you read, every time you have to navigate an ingenious patch around a problem in the studio, you're emulating what these folks did.

Meek By Name, Not By Nature Joe Meek's father returned from WWI after having a horse shot out from under

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him and seeing his friends blown to bits on either side of him. Joe would find himself in the line of fire himself later in his life, which ultimately proved just as violent and lethal. He was also tone-deaf and had no sense of rhythm, none of which stopped him from composing and recording timeless songs, including the Cold War classic 'Telstar', and creating a compendium of studio techniques, including flanging and close-miking, and hardware such as one of the first spring reverb units, that today are so commonplace that they seemed to have always been there. "Joe Meek was a scientist to the extent that he was the 'Wacky Professor', though not nearly as benevolent," observes Barry Cleveland, who wrote the definitive technical biography of Meek, Creative Music Production: Joe Meek's Bold Techniques (available at www.artistpro.com and www.barrycleveland.com). "His temper was legendary — he regularly and literally threw musicians and their equipment down the three flights of stairs from his apartment," at 304 Holloway Road. The most famous of Meek's 'black boxes' was the spring reverb unit he fashioned from a broken HMV-made fan heater in 1958. While working at the IBC-owned Lansdowne House studio in Holland Park, London, he also developed a compressor/limiter based on Langevin designs and an EQ based on a Pultec, described by its current owner, Nigel Woodward, as "probably the warmest, smoothest, most transparent equaliser ever made." "Joe made a lot of things based on existing designs, but he took them to another level," says Cleveland. As a scientist, Meek was assertively autodidactic — his only formal technical training came during his National Service stint as a radar technician. But as a teenager he built a television set from scratch, despite the fact that TV signals hadn't reached his part of the country yet. "As a technology developer, Joe was largely intuitive," says Cleveland. "He considered himself as very technical, but he had very little formal technical education. What he could do very well, though, was conceptualise something — a sound, a way of recording — and then find a way to achieve it."

Photo: AOK Ware, with thanks to Denis Blackham Joe Meek in the cutting room at IBC

Studios. Meek's fellow engineers at Lansdowne and other studios were reportedly both disdainful and jealous of his capacity to try new techniques, and in doing so, he upset the highly conservative and hierarchial order that was the atmosphere in the British music recording industry at the time. "The orthodoxy then was to use few microphones, placed away from the sound source, and use a lot of natural room ambience," says Cleveland. "Exactly this microphone was placed exactly here for each instrument. It was all done by rote. Meek ignored the rules and put



whatever microphones he wanted to wherever he wanted to, usually a lot of them placed close to the sources, and then used artificial ambience like reverbs to create a sense of space. This approach to recording is what gave engineers the kind of control over sound that today we take for granted in the studio. He taught us to create sounds rather than just attempt to control dynamics." According to Cleveland, Meek was the first engineer in the UK to use compressors to create pumping and breathing effects rather than to merely to control dynamic range. He also pushed limiters to the max to get the hottest possible levels on tape and took advantage of analogue tape's natural compression characteristics. It is also likely that Meek was one of the first engineers to direct inject the electric bass by plugging it straight in to the mixer. If it's true that there is a fine line between genius and madness, then Meek straddled that boundary. His pathologies rivalled his accomplishments, and included depression, pharmaceutical abuse and paranoid schizophrenia, causing him to cover many of his inventions with tape lest they be seen and copied. "He thought everyone was out to steal his secrets," Cleveland says. "Still, the innovative techniques he devised were done in studios for all to see. He was not a note-taker, so that's how his innovations became part of the way records came to be made — because people did see what he did and copied it." Meek would disregard regulations and never 'zero' any piece of gear, leaving parameters with his settings and often leaving entire pieces of equipment modified with their chassis lying open. Joe Meek was troubled, in the way many artists and innovators are. The way he tried to deal with his demons ultimately failed — he committed suicide in 1967 — but left the industry with a legacy that changed the course of sound recording.

Tom Dowd: Change The Machine, Not The Man In many ways Meek's opposite, Tom Dowd was equally instrumental in changing the way records were made as chief engineer and later producer for Atlantic Records, where he was involved in records for the Clovers, Ruth Brown, Joe Turner, Clyde McPhatter, LaVern Baker, the Drifters and Ray Charles, with whom he illustrated his unique ability to turn out records that sold equally well on both sides of the racial divide — which in and of itself was a remarkable accomplishment for the time. Dowd's scientific and musical background was as formal as Meek's was ad hoc. Dowd attended the prestigious science and mathematics-oriented Stuyvesant High School in Manhattan, graduating at the age of 16 in 1942. He began taking classes at City University and Columbia University, also playing trombone and drums in Columbia's band, adding to the violin and piano that his mother, a classically trained opera singer, and father, who designed sets at the Roxy Theater on Broadway, had encouraged as a child. He registered with his draft board in 1943 upon turning 18 in October of that year, and his scientific background did not go unnoticed: he was assigned to the Special Engineer file:///H|/SOS%2004-10/The%20Engineers%20Who%20Changed%20Recording.htm (3 of 9)9/26/2005 12:37:45 AM


Detachment, Manhattan District, one of a web of university-based scientific cells working on the myriad components of what would come to be called the Manhattan Project — the effort to build the atomic bomb that ended the war. During his stint on the project, Dowd operated a cyclotron, performed density tests of various elements, and recorded statistics as part of the neutron beam spectography division.

Tom Dowd mixing Derek & The Dominos' 'Layla'.

"When the war was over, Tom came out of the service and wanted to finish his degree at Columbia," Dana Dowd, Tom's daughter and now the manager of his estate, told me. "He found that he needed just a few physics and math classes to graduate, so he asked the school to give him the credits because of the work he had done during the war. He didn't want to go to a classroom where a professor would tell him there were this many elements in the universe after he had spent two years helping develop technologies that discovered new elements. It would have driven him crazy." Unfortunately, the nature of Dowd's wartime work was top secret, so the school declined to award credit and Dowd went off to work for a radio station, engineering music sessions on the side. Physics' loss was pro audio's gain. Teaming with Atlantic Records' vice-president and producer Jerry Wexler, Dowd worked at the major independent facilities of the day in New York and later designed a series of studios there for Atlantic. As Dowd became more entrenched in music engineering, Wexler found himself relaxing a bit. "With Tom around, I never had to touch a fader," he told me from his home in Florida, and he credits Dowd as the engineering genius behind the many hits they made together. In fact, those faders Wexler refers to were Dowd's invention, as well. Dana Dowd asserts that her father was the first to use faders instead of the huge Bakelite knobs then common on music consoles. "He was a piano player, and he wanted to be able to play the console the same way he could the piano: with several keys in each hand," she says. "With the knobs, you couldn't have more than one in each had at a time. So he found a company that was making wire slider [potentiometers] and installed them into the console at the first Atlantic Studios. Too bad he didn't patent the idea."

Bill Putnam at one of the mixing consoles he designed.



Where Joe Meek was secretive to the point of being furtive, Dowd — who probably had enough of secrecy during the war — was good at project management and identifying people to whom to delegate tasks. "He hooked up with Mac Emerman [founder of Criteria Studios in Miami] and Jeep Harned [founder of console and tape-deck maker MCI] and would say 'This is what we need to do, this is what we need to do it with. You guys build it,'" recalls Trevor Fletcher, longtime general manager at Criteria Studios. "He had a very technical mind but he was also good with people and had the common sense to know when someone else could do something better than he could. He created the amalgam of needs that became the specs for Criteria and had Mac put it together. But he didn't wait for people to act. He told me the story about when he went to Muscle Shoals the first time to do records there with Jerry Wexler and Rick Hall. The only tape machine in town was broken. Tom called around, found the part and fixed it." Tom Dowd remained deeply engaged with both music and technology until his death in 2002.

Bill Putnam: Mixing Technology & Business Key components of the heyday of analogue recording can be traced back to one man. Bill Putnam not only devised classic equipment like the 1176N and Urei Time Align monitors, but he also designed and built equally classic recording facilities to use them in, such as Universal Audio in Chicago and United Recording in Los Angeles. Putnam also made records: the Harmonicats' 'Peg-OMy-Heart', which Putnam recorded in 1947, was not only a million-seller but is also widely acknowledged as the first pop record to use artificial reverberation, which came in the form of a tiled toilet at Universal to which the vocal signal was sent. It should be noted that Putnam was as comfortable with the business of audio as he was with the technology of it. The Harmonicats record underscores that: he financed the recording in return for a piece of the profits. He would pull a similarly profitable coup years later when, in the late 1950s, record labels were still unconvinced of stereo's sales potential. Putnam, however, felt otherwise, and began mixing everything he did in stereo as Photos courtesy of Universal Audio well as mono. When stereo finally did Another of Bill Putnam's mixing desks. take off in the early 1960s, labels scrambled for content to fill their catalogues and Putnam was waiting for them, collecting a handsome premium for his anticipation of the situation. file:///H|/SOS%2004-10/The%20Engineers%20Who%20Changed%20Recording.htm (5 of 9)9/26/2005 12:37:45 AM


Putnam's list of accomplishments is huge. He developed the first multi-band equalisers, was a pioneer in studio acoustics and design, designed the 1108 FET preamp, and was a leader in half-speed mastering techniques — usually on mastering equipment he built himself. Putnam was born in 1920, at the beginning of the radio age, and like Meek and others was fascinated by sound to the point that he built his first radio at the age of 15. Putnam's father seems to have been a template for much of his son's future development: he was a businessman who also produced radio programmes. In high school, he assembled, rented and repaired PA systems, while also singing on weekends with dance bands, for five dollars a night — which included the PA rental. Bill Putnam had much of Meek's intensity, but was able to channel and harness it, thanks largely to an innate ability to see the fiscal as well as the sonic potential of new ideas. Like the other inventors talked about here, Putnam's inventiveness stemmed mainly from the need to create solutions to difficulties he encountered while working as a producer and engineer. His son, Bill Putnam Jr, puts it succinctly. "He was a guy who built equipment to solve problems in the studio." That approach, which truly did put the music before the moolah, kept him in demand by artists including Nat 'King' Cole, Ella Fitzgerald, Duke Ellington (Putnam was reputed to be the Duke's favourite engineer) and Count Basie. That, in turn, pushed Putnam's own limits and compelled him to create new solutions to audio issues as they arose, many of which resulted in equipment still used regularly today. Bruce Swedien, who went to work for Putnam at Universal in Chicago as a teenager, relates on the Universal Audio web site: "Bill Putnam was the father of recording as we know it today. The processes and designs which we take for granted — the design of modern recording desks, the way components are laid out and the way they function, console design, cue sends, echo returns, multitrack switching — they all originated in Bill's imagination."

Tom Scholz: The Rockman Painfully thin and angular in the consumptive manner of Romantic poets, Tom Scholz is what all musical geeks aspire to be. He graduated at the head of his classes in Mechanical Engineering degrees (bachelors and masters) at MIT in Boston, the city he would name his band after. For someone legendary in the music industry for procrastinative perfectionism — Epic Records sued Scholz and Boston for $20 million to get the band to release their third record, one of only three they would release over 11 years on the label — Scholz generated a prodigious output when it comes to invention. Scholz Research & Development (SRD) is built upon Scholz's inherent scientific nature and the experience he gained working as a design engineer at Polaroid in the early 1970s. In 1980, in between the second and third Boston albums, SRD released the Power Soak, which, using a series of resistors between the output of a 100 Watt tube amp and a speaker cabinet, allowed for big sounds at low volumes and was an instant hit



with guitarists (and SPL-stunned engineers). In 1982 came the Rockman, a beltworn solid-state headphone amp which reproduced the guitar sounds he astounded the world with on the debut Boston album. The company went on to release numerous related products, and Scholz personally was awarded over two dozen design patents before he sold SRD to Dunlop in 1995. Scholz's musical background was more classical than pop — he played piano as a child and didn't pick up the guitar, which would become his trademark, till he was 21. The same ability to synthesize science and music that made Boston's records so uniquesounding also helped Scholz create products that helped feed the niche being carved by Tascam's Portastudio: he was taking what previously had required large amounts of space, technical adroitness and volume (not to mention money) and putting it into a simple, affordable box. Like Bill Putnam, Scholz recognised the need to acknowledge a market, not just solve a studio challenge.

Photo courtesy of Universal Audio As well as designing much of the equipment that went into his studios, Bill Putnam also planned and built the studios themselves.

"I was a fixer, a builder — an inventor — ever since I can remember," Scholz once told writer Larry Lange in an interview. John Boylan, who produced Boston's debut record, which has sold over 16 million units, recalls that Scholz's engineering foundation was critical to the music, and provides insight into how Scholz created the Rockman. "When the first album was a huge success, and he had some money, [Tom] bought a special oscilloscope which would freeze-frame the waveform of an audio signal," Boylan explains. "He would play the guitar sound into the 'scope, freeze the waveform, then take a picture of it with a special Polaroid camera that he had acquired when he worked there. He used this method to be sure that he was always getting the same guitar sound. "To me, Tom Scholz is interesting because he got his start with engineering that had nothing to do with audio. He helped [Polaroid chief] Ed Land perfect the illfated instant movie camera. The two technical achievements that he had worked out on his own that impressed me were his use of analogue, bucket-brigade delay on guitars, and his use of a variable resistor between the Marshall 100 Watt head and the cabinet, which he later marketed as the Power Soak." Still Boylan contends, "I'd venture a guess that the two domains of inventor and engineer are in separate compartments of his thinking process." Business was definitely compartmentalised. "I hated it," Scholz told Larry Lange of his brief career as an entrepreneur, despite selling tens of thousands of products. He continues to use analogue recording techniques and technologies — he still has a stash of Scotch 226 tape — and live Boston performances



continue to haul with them a ton of studio-level gear connected in complex ways. One can wear more than one hat, but not all of them always look good on you.

George Massenberg Like other audio inventors, George Massenberg showed a proclivity for music and technology at an early age Photo: Ron Pownall, courtesy Boston — at 15 he was working part-time both Tom Scholz, with his wall of Rockman gear. in a recording studio and in an electronics laboratory. In his '20s, he was chief engineer of Europa Sonar Studios in Paris, France in 1973 and 1974, and also did freelance engineering and equipment design in Europe during those years. He founded his technology company, GML, in 1982 to make devices he wanted for his own projects, most notably the first parametric equaliser, as well as refinements to console automation systems, which ultimately earned him a Grammy award for Technical Achievement. He has been equally fecund on the other side of the console, earning two Grammy awards for his work with artists including Billy Joel, Kenny Loggins, Journey, Lyle Lovett, Toto, the Dixie Chicks, Mary Chapin Carpenter and Linda Ronstadt. He has designed, built and managed several recording studios, including the Complex in Los Angeles. Like Dowd, Massenberg had issues with the rigidity of academic science. While taking Electrical Engineering at Johns Hopkins University — a school he describes as "medieval, apathetic and oppressive as schooling in the '60s could get" — he got into a row with a teacher who looked at a schematic for a gyrator that he had built and declared it 'of theoretical interest only' and 'impractical' to implement. "Seeing this as a sign," he says, "I dropped out of college." John Boylan, who has worked with Massenberg for years and owns one of Massenberg's early ITI parametric units, reminds us that "George was also a significant innovator in the field of limiting — his compressor/limiter has certain variable parameters that no other unit has. With George, the two domains [of science and art] interact in music. Unlike many tech-heads who are either unaware or not interested in what it is they are recording, George has a true feeling for the music. He's really a renaissance man." George Massenburg.



"I don't believe anything is totally original or creative," Massenberg explains, in response to a question about how his own creativity applies to music and technology. "Nothing is created in a vacuum. Things are created and invented in response to awareness of a need. You're sitting in the studio and something will piss you off and after the nth time it bothers you, an idea will begin to emerge." It's what one does with the idea that sets the entrepreneurial apart from the creative, Massenberg believes. If the process of invention is non-linear and draws from an array of sources, as he says it does, then turning an inspiration into a practical and (hopefully) bankable reality requires other skills. He makes a trenchant, almost startling statement: "Those who are motivated by entrepreneurship and profit create products that are distinctly different from those created by people motivated by need." The comment prompts Massenberg to observe further that there seems to be a parallel between the state of the music industry and that of the pro audio equipment sector. "The market has taken over the creation and development of boxes, just as conglomerates have taken over record labels and radio," he says. "The loss of creativity in both domains is palpable." That said, though, Massenberg also believes that new technologies and platforms can still be created by individuals searching for tools that the market hasn't yet discovered in focus groups. "The way for that to happen, though, is for people to rearrange their priorities," he says. "Don't calibrate your thinking to what common wisdom says is what you should be working on or thinking about. Guard against arrogance and insularity. And leave yourself open to serendipity. That's where invention flows from." Published in SOS October 2004




The Prodigy

In this article:

Getting Away From It All Liam Vs Noel Escape To The City Soundbites & Samples Mixing It All Up A Cure For Equipment Overload

The Prodigy Liam Howlett: Recording Always Outnumbered, Never Outgunned Published in SOS October 2004 Print article : Close window

People : Artists/Engineers/Producers/Programmers

Nowadays, plenty of hit albums are recorded in bedroom studios — but Liam Howlett of the Prodigy has gone one better, by recording his latest in bed. Sue Sillitoe

It's been seven years since the Prodigy had a worldwide hit album with Fat Of The Land. And that's a long time in the rock business. Sitting in a London pub with an afternoon of press interviews ahead of him, Liam Howlett, the band's main man, anticipates the questions he is likely to face about this lengthy absence. "Everyone's bound to ask what we've been up to and why it's taken so long to get our new album together," he says. "But actually it's not that long — not when you consider we were on tour for over two years and then I recorded half an album and ditched it before I started on this one." The Prodigy's new album, Always Outnumbered, Never Outgunned, was released in August through XL Records. Described by the label as 'a trashy, adrenalised, sleaze-funk masterpiece', it has a very different feel to Fat Of The Land. It is more fluid and less formulaic, thanks to the way Liam Howlett has treated the vocals almost like samples, using them to create an overall sound rather than focusing on up-front performances. "The most important thing for this album was to concentrate on the music and bring that to the fore," he says. "I wanted the vocals to be less important and for the album to be about energy and aggression, so that we pushed the music forward without a vocalist carrying the weight. I wanted this album to be about getting back to the beats, because this is what the Prodigy has always been about."

file:///H|/SOS%2004-10/The%20Prodigy.htm (1 of 9)9/26/2005 12:37:48 AM

The Prodigy

Getting Away From It All In order to do get back to the beats, Liam had to rethink the way he worked, although it took him a while to realise this. He obviously enjoys the recording process and despite professing to be no expert in the studio, he has actually written, produced, engineered and recorded the whole of the Prodigy's output. But for this album he ended up shutting the door on his studio in a bid to free himself from his own preconceptions about how music should be created. "People always go on about bedroom studios, and when I first had a studio it was in my bedroom," he says. "But this album was the first time I'd taken the concept a stage further and actually written songs in bed." Understanding how that came about is convoluted but fascinating, as it shows how the creative juices can really dry up if the environment and mood don't work. Liam is the first to admit that he struggled for a while, messing around with tracks that he ended up dumping. It was only when he went right back to basics — and back to bed — that the new album began to work. "Touring Fat Of The Land kept us out of the studio for quite a long time because we were always on the road," he says. "We'd be away for three weeks, back for two days, then off again, so there was never enough time to get into the studio and do anything new. By the beginning of 2000, I'd had enough and I decided I didn't want to do it any more. I wanted to take time off. I didn't want to be in the studio. I didn't want to be Prodigy. I just wanted to be Liam and go out with my mates and get drunk." So Liam took time out to relax, eventually returning to his home studio in Essex in 2001 with the aim of making a start on a new album. But after writing five or six tracks, he realised his head was still in Fat Of The Land mode and that he wasn't inspired by anything he'd created. Even though the sessions resulted in a hit single, 'Baby's Got A Temper', which was released in 2002 and reached number eight in the UK charts, it felt like a dead end. "I wasn't happy with that record," he says. "It felt like a step backwards, both sonically and musically. It was quite slow and had no energy — it certainly didn't sum up what the Prodigy is all about." When a band is known for its distinctive sound, there's always a danger of becoming formulaic, and Liam believes this is exactly what happened with those five tracks. "I had become lazy," he admits. "The single came out but I decided to bin all those tracks — half an album. I knew 'Baby' wasn't right and all those tracks were grouped in with that sound. The songs were OK — bits of them, anyway — but I wasn't particularly excited by any of them. I felt it was better to start from scratch and be brave, because it's only when you're brave that exciting things happen."


The Prodigy

What followed was a period of four months that Liam describes as "the most frustrating four months of my life". Working with producer Neil McClellan, he sat in front of a computer screen in his studio in Essex, surrounded by every bit of kit imaginable, and achieved absolutely nothing. "My studio is crammed with equipment, but I ended up feeling I was being The control room at The Mews recording overcome by it all — it was just too studios. much," he says. "I used to go to bed every night thinking 'Tomorrow, I'm going to write the tune, tomorrow is going to be the day,' but nothing ever happened. Eventually Neil pointed out that we'd been in the studio for four months without having anything to show for it. When you're in the middle of something mad you don't really realise what's going on. You just go round in circles. There were a lot of distractions at home. Nat [his wife, All Saints' Natalie Appleton] was doing her own record so we weren't spending a lot of time together, but there were always dogs to stroke and videos to watch and gardens to walk round, so I didn't ever feel like I was at work — I was too laid-back. And, of course, my house is very peaceful; it's in the country so it doesn't have an edge and it wasn't giving me the right inspiration. Neil said we had to get out, get back to London. I knew I physically couldn't sit in my room any more, and for the first time in my life I listened to someone else and realised I actually needed help. It wasn't that I needed help with the writing, just that I needed help finding the right headspace to get into the right frame of mind. I wanted to write a good album — one I was happy with — but to do that I knew I'd have to jerk myself out of situation I was in and start again."

Liam Vs Noel Always Outnumbered, Never Outgunned features some interesting collaborations with vocalists, notably Oasis's Liam Gallagher and actress Juliette Lewis. Given that Liam Howlett is married to one Appleton sister and Liam Gallagher lives with the other, it would be easy to assume that this particular collaboration came about through family connections. Not so, says Liam — at least not initially. "I met Liam Gallagher six years ago at a festival, soon after Noel had released his collaboration with the Chemical Brothers. He said he wanted to do a track with the Prodigy so he could blow the Chemicals out of the water. Liam Gallagher carries the attitude and the spirit of punk rock, which is why I had to work with him. I was curious to see if we could capture that." Years later, the family connections did come into play. "My house is a party house — we always have people over at the weekend and there's always stuff going on," Liam says. "It was during one of those weekends that Liam and I finally made it into the studio. We were both really pissed and it was three in the morning but we thought what the hell — let's go up and do that track now. So we went into the studio and recorded the vocals. I had various tracks lying around and I played him file:///H|/SOS%2004-10/The%20Prodigy.htm (3 of 9)9/26/2005 12:37:48 AM

The Prodigy

through some stuff and let him choose the one he liked. It was just a loop with a guitar and he spat loads of lyrics down. We listened to them the next day, picked what we liked and laid them down properly. The funny thing was, because I was really pissed, I managed to set the mic up and get levels on the mic but I was so fucked I couldn't work out how to get his voice on to separate tracks on the tape machine. So I ended up recording his vocal onto the same track as the music. The next day I had loads of wicked delays and mad effects set up and was all ready to use them when I realised what I'd done. Liam was saying 'Let's use that bit,' and I was going 'I can't — it's all on two-track.' Somehow we managed to piece it together by re-recording the bits of the vocals that were really screwed, and eventually it worked." The track, 'Shootdown', is the album's most garage guitar-driven tune. The original version was even rockier, and at first Liam wasn't sure if it fitted in. "There were a couple of different versions to start with, so I messed around with them until I got a track that sounded how I wanted it," he explains. "It was almost the last track to be finished because it was quite different from the others. I wanted to retain its raw, punk-rock spirit without making it too electronic. It sounds weird because we're known for really crunching up the speakers, but I didn't want it to be a dance record. I wanted it more of an open rock production, more loose and organic than heavily programmed and with lots of unquantised guitars." The track 'Hot Ride', featuring Juliette Lewis, was always destined for the album. "I'm always up for doing the unexpected," he says, "and I like challenging people's perceptions about who should be on a Prodigy album. It's exciting when two different things clash together. Anyway, I think Juliette is a rock star trapped in an actress's body. She's completely insane — more rock & roll than most people in bands that I know." The collaboration was inspired by a friend who saw Lewis playing a gig at the Viper Rooms in Los Angeles and was blown away by her voice. At that time few people knew she was into music, but Liam got in touch with her management company and discovered that she had a demo she was shipping around. "When I got the demo I was really excited because her voice was menacing but also melodic. We messed around for a couple of weeks, sending each other stuff through the post. I sent her a track called 'Good Morning' and she sent the vocal back, but being so far away made communication difficult and I eventually decided we needed to be in the same room. Luckily, she was working in France so we got her over to England and that's when I discovered just how well she fitted into the Prodigy vibe. We recorded 'Good Morning', which is a straight punkrock track, then I played her the instrumental to 'Hot Ride' and she started telling me about this other vocal idea she had, a lyric about escaping in an air balloon. That got me thinking about the Fifth Dimension track, 'Up Up And Away In My Beautiful Balloon', and I decided it would be really bizarre to steal it. I'm seriously into the idea of blatant beat thievery. The whole bootleg scene excites me because it allows you to do mad things by mashing up all kinds of different genres. Juliette wasn't sure at first, but when we put it down we both thought it was just fuckin' cool. I can't pinpoint why that track works, it just does. It's not a dance track, nor punk — just weird — but it rocks."

Escape To The City Turning his back on most of the equipment in his studio, Liam bought a laptop and, after hearing about it through a friend, a copy of Propellerhead's Reason file:///H|/SOS%2004-10/The%20Prodigy.htm (4 of 9)9/26/2005 12:37:48 AM

The Prodigy

software. He then selected just five pieces of kit from the studio — a Thermionic Culture Phoenix valve compressor and Culture Vulture distortion unit, a Korg Micro Keyboard, a Manley valve EQ and a 1980s Korg MS20 analogue keyboard. "Since 1993 I'd written everything in Cubase, with hardware synths and Akai samplers as my main setup," he says. "But when I bought the laptop I realised it didn't have to be that way. The laptop and Reason gave me the freedom to work anywhere I wanted. It was very liberating." Liam adds that the laptop-Reason combo reminded him of the early days One of Liam Howlett's main reasons for of Prodigy when he was a big fan of choosing to work at The Mews was its the Roland W30 workstation. Indeed Mackie analogue desk. he stayed a big fan for a long time, despite its limitations (only 16 seconds of sampling time being one of them!) because he loved the idea of having everything in one box. "Part of the W30's appeal was having a simple machine that did everything. It was refreshing to go back to that. Reason allowed me to work my ideas through using just one machine. Initially I just wrote beats on it, but before long I was bringing in more and more sounds and building up a library of samples on my hard drive. Once I'd got the beat, I'd gradually introduce a bass line. It just progressed from there and eventually I was writing entire tracks on it." Liam is now a firm fan of Reason, which he describes as "like a computer game to use". He also credits it with getting him back on track when the creative juices had temporarily dried up, and for making the process of writing music fun again. Armed with his laptop and his few chosen pieces of equipment, Liam decamped to London with McClellan, hooked up with Pro Tools operator Damian Taylor and hired a studio called The Mews in Stoke Newington. "The Mews is a very small studio owned by some friends of ours. Its main advantage is that it has a Mackie desk, which I was keen to use for pre-production and recording," Howlett explains. "The Prodigy sound owes a lot to Mackie because I've been using their stuff for years. The first mixer I ever bought was a Mackie CR1604, which I got when I was a teenager, and I've now got a 32:8 analogue eight-buss console that I love because it gives us our distinctive Prodigy sound. It's a desk that lends itself to being driven very hard. It has a naturally warm sound that allows you to


The Thermionic Culture Phoenix compressor (second from top) was part of the small selection of

The Prodigy

push the channels as far as they'll go. Part of the appeal of The Mews was that it had a 32:8, so I immediately felt at home."

gear that Liam brought to The Mews.

Working at The Mews, often for 11 hours at a time, then driving home listening to the sounds they'd created that day helped Liam capture the sense of excitement he'd been looking for. When he got home he was so fired up that he couldn't stop, which is where his bed comes in. "I'd get into bed, put Moonraker on my DVD and create this idyllic cocoon of happiness that allowed me to think really clearly. I'd know exactly what I wanted to do. Reason really helped because it's so quick to get ideas down and that was when I really started writing. I wrote 'Spitfire', 'Wake Up' and 'Girls', which is the single, in bed. I'd get the demos and ideas down on Reason — the beats, the programming, the basic layouts and the initial vibe — then I'd go into the studio the next day and play the boys the results. When they knew I was writing stuff in bed they joked about getting a replica of my bedroom set up in the control room because it worked so well." At The Mews, the sounds were transferred into Pro Tools so that vocals and overdubs could be added. "Reason is purely for sequencing," Liam says. "It has its uses but moving to Pro Tools allowed us to up a gear and get the tracks sounding pretty good. Damian Taylor is without doubt the best Pro Tools operator I've ever met — he's incredible. He's very precise with his fades and really likes to get deep into the system to make sure everything is accurate. I wanted a more cut-and-paste approach to the sound of this album, so I really had to fight his natural instincts and hold him back. But even so I couldn't have done without him. He was with us throughout the entire time we were at The Mews and then came to Whitfield Street for the mix. At The Mews, we recorded stems — drums, keyboard parts, vocals and so on — on to stereo pairs in Tools so we could finish everything off. That worked well because Reason integrates really easily with Pro Tools."

Soundbites & Samples Liam's mobile approach to writing has injected a new spontaneity into this latest collection of Prodigy tracks. They are still dirty and punky, but they do feel fresher and more exciting than anything he has done for a long time. Part of this is definitely down to the vocals, provided by an interesting range of artists including Kool Keith, Twista, actress Juliette Lewis and Liam Gallagher (see box on previous page).


The Prodigy

Liam explains: "I didn't want this album to turn into a collaboration album. The full vocal collaborations were with Liam, Juliette and Twister. All of the other vocalists were used as soundbite lyrics that I incorporated into the track in the same way that I'd use a sample. They back the attitude of the song without taking the focus away from the music. Even with a track like 'Spitfire', where the lyrics are very much 'in your face', there's still the sense that the music is the main thing. "The first single, 'Girls', was written in an 'Old Skool' sense. Usually I write a track and then fit samples into it, but with this one I thought about the vocal sample first and wrote the track around it. When it was done I realised it lacked sexiness and trashiness, so I brought in the Ping Pong Bitches, who are friends of mine, and used them to give it the vocal the edge it was lacking. That's when it really came together." The live area at The Mews.

The majority of the vocals were recorded at The Mews using a valve Neumann microphone and heavy compression. Liam explains: "There isn't a specific formula we use to record vocals. Usually it's just a case of trying out whatever outboard equipment I've got in the studio and seeing what works. The Culture Vulture distortion unit was crucial to this album — we used that a lot. We also compressed Paul Jackson's vocals two or three times on 'Action RADAR' to get them sounding really crunchy, and we used the [Logic plug-in] clip distortion on the Ping Pongs' vocals because it added a definite trashy close sound, which was sexy." The vocal contributions from Liam Gallagher and Juliette Lewis, however, were not recorded at The Mews. Gallagher was recorded in Liam's Essex studio (before he locked the door on it) and Juliette Lewis did her sessions at Whitfield Street — in the large orchestral room where one of the booths was turned into a cosy faux living room complete with nice rug!

Mixing It All Up The writing and recording process went on for 10 months, from March 2003 to January 2004. Then the project moved to Whitfield Street, which is where Liam and McClellan mixed the album. Liam explains: "I assumed we'd use Whitfield Street to just balance the mixes on the big speakers. I'd mixed the tracks at The Mews on the Mackie using Yamaha NS10s and I knew we needed to hear them through big speakers to check everything was OK. I thought we'd just push the stems up through Whitfield Street's big Neve console and it would all sound great. But when we got there I asked Neil to humour me and let me spend another day mixing one track, 'Spitfire' — and bang! It sounded so good I just file:///H|/SOS%2004-10/The%20Prodigy.htm (7 of 9)9/26/2005 12:37:48 AM

The Prodigy

had to mix the whole album again." Despite having a Neve at his disposal Liam still wanted access to a Mackie desk, so he borrowed a 32:8 and used it as a submixer because it was able to create that signature Prodigy sound. He also raided Whitfield Street's extensive equipment store and found loads of old valve stuff, including five Fairchild compressors, which he carted up to the studio. "We had all the tops of the studio units covered in retro gear, which was great," he says. "The new mixes gave us the sound we wanted. They still retained the trashiness that is Prodigy, but somehow they were fresher without losing any of their edge."

For the mixing sessions, Liam Howlett augmented Whitfield Street Studios' own impressive selection of equipment with another Mackie desk, and also brought some vintage valve gear out of their store room (as can be seen on top of the right-hand rack).

Always Outnumbered, Never Outgunned was eventually mastered in the Lodge Studios, New York, by Emily Lazar. "She's brilliant — she does a lot of hip-hop and she totally understands bass," Liam says. "And she gets volumes that other people can't get. I went to New York for the mastering and the sound was monstrous. It just shifted up a few more gears. We had Pro Tools set up and were still tweaking the tracks in another room linked to the studio, adding more bass until we were happy with it. It was a good idea to take Pro Tools to the mastering room and I'd certainly do it again."

A Cure For Equipment Overload So, having written this album in bed, what about the next one? Will Liam Howlett be returning to the Essex home studio where he was so overwhelmed by equipment overload? His answer is simple: "No. The whole lot's up for sale." Qualifying this, he adds: "Now I'm back home with the baby [five-month-old Ace] I need somewhere to sit and be comfortable. I'm going to take all that stuff out, put in a couch and get rid of the boxes and racks. I'm going back to basics with two or three favorite analogue vintage keyboards and a nice big Manley EQ. I'll keep the Akai Z8 and a few compressors but I'm getting rid of the digital stuff, including the Mackie digital eight-buss and the big speakers, which I'm replacing with Genelec 1030As because that's what I used on this new album. All that other stuff has served its purpose and earned its money, but I want to go back to how I worked in the beginning — purely in my head and not swayed by all this equipment."


The Prodigy

What else he plans to keep is easily listed — his laptop, Reason, a sample library, his own imagination and his DJ mates Nobby and James, whose 'amazing' record collection is always at his disposal. He'll also keep his Oberheim Four-voice synth, Korg MS20, Culture Vulture distortion unit, Phoenix compressor and Korg Micro Keyboard. But apart from that lot, the rest is going up in the loft. "You have to be happy with your surroundings when you are in a studio, regardless of what type of music you play," he says. "I still needed to be happy, even if our sound is aggressive. It's more to do with having a creative frame of mind. The reason I write music is because I'm constantly battling with myself. I push myself hard. I'm always hungry to write something better or tougher, so finding the best time and situation to write a song is really important to me. For kids who are new to it, inspiration comes easy but when you're this far down the line you have to search for the inspiration — and search for the right headspace to slip into." He's a nice bloke is Liam Howlett, and this new album is really good, so let's hope that by simplifying his working setup and by going back to basics he'll keep finding his headspace. Published in SOS October 2004




The Resurrection Of Brian Wilson's SMiLE

In this article:


Raising A SMiLE Mark Linett & Darian Sahanaja The Assembly Line Published in SOS October 2004 The Dynamic Duo SMiLE Or Smiley? Print article : Close window A Digital Keyboard? People : Artists/Engineers/Producers/Programmers Brian Wilson's '60s Productions From Stage To Studio The Original SMiLE Recordings The Beach Boys' SMiLE album was to have been the Picking Up The Pieces pinnacle of Brian Wilson's groundbreaking recording Mix Me A SMiLE and production achievements — but it was never Further Reading & Listening completed. This year, in an extraordinary tale of 37 Years In The Making

emotional drama, tape-vault archaeology, and recording technology, Brian finished it, debuted it live, and then re-recorded it in the studio. SOS brings you the full story... Matt Bell

In 1966, at the age of 24, Brian Wilson hit a creative peak in his musical career that most of us can only imagine. His enormous success with the Beach Boys had already afforded him the artistic freedom to work however he chose. In 1963, he was one of the first artists to wrest control of his career from the Beach Boys' record label Capitol. He began to produce the group's records himself, and spurned Capitol's own inferior facilities in favour of the cream of Los Angeles' independent recording SMiLE logo courtesy of Mark London studios, such as Western Recorders (now Cello), Goldstar and Sunset Sound. Over the following three years, his records grew ever more instrumentally elaborate, to the point where the other Beach Boys could no longer provide him with the musical backing he needed to realise the advanced arrangements in his head. Instead, he came to rely on a crack team of the best session musicians in Los Angeles, known loosely as the Wrecking Crew, and increasingly, he restricted the rest of his group to exclusively vocal duties. At the same time, his production skills matured as recording file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (1 of 20)9/26/2005 12:37:54 AM


technology improved; from 1962 to 1966 he moved from using basic three-track recording to the latest eight-track machines, constantly pushing at the boundaries of what was technically possible. And the results of these artistic and technical endeavours continued to be hugely successful (for more technical details on Brian's '60s productions, see the box on page 122). By late 1966, the Beach Boys had eclipsed the Beatles as readers' favourite group in the end-of-year poll conducted by UK music paper The New Musical Express — the first time this had happened since the Liverpool foursome's rise to fame. Paul McCartney was already admitting to friends that the Beach Boys' Pet Sounds was the best thing he'd ever heard, and was wondering what the Beatles could do next to top it. And in December 1966, Brian's most complex ever record production, the single 'Good Vibrations', went to number one in the USA. By then, he was already back in the studio working on an album that was to employ in its assembly the same advanced recording, multitracking and tape-splicing techniques that had resulted in 'Good Vibrations'. It was to be called SMiLE, and promised to meld advanced production, Brian's intricate instrumental arrangements, as realised by the skilled musicianship of the Wrecking Crew, the dazzling and esoteric wordplay of his chosen lyrical collaborator Van Dyke Parks, and the heavenly vocal harmonies of the Beach Boys. But hauntingly, SMiLE remained unfinished, and most of it was never released. Paul McCartney came up with the title of Sgt. Pepper's Lonely Hearts Club Band for the next Beatles project, and the Beach Boys' moment passed; never again did they reach the pinnacle of artistic achievement and commercial success they had occupied at the end of 1966. Crushed by the expectations placed upon him and his inability to deliver, over the next few years Brian gradually withdrew from the recording business and his group, and lived a hermit-like existence for many years, racked with insecurity and self-doubt. By the late 1990s, although he had made an admirable recovery and had begun releasing solo records and touring, nobody seriously believed SMiLE would ever see the light of day. For Brian Wilson, it seemed that the music that should have been regarded as his finest work had become forever associated with catastrophic failure. This year, however, has seen the unthinkable happen. On February 20th, Brian, now in his early '60s, debuted a completed version of SMiLE live at the Royal Festival Hall in London. And on September 27th, a newly recorded version of the album will be released to the world. Everyone loves a happy ending — but how did it come about?

Raising A SMiLE By 2002, Brian had regained sufficient confidence in his music that he was able to play Pet Sounds, his best completed work, on a live world tour. Discussions on how he could top this triumph all led in the direction of SMiLE; after all, what was to have followed Pet Sounds and bettered it back in the '60s? Since the music was still such a source of discomfort to its creator, one man in his live band was file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (2 of 20)9/26/2005 12:37:54 AM


charged with the responsibility of helping Brian bring SMiLE back to life in some form. Multi-instrumentalist Darian Sahanaja now describes this past year and a half as a blur of "eating, breathing, and sleeping SMiLE", but in truth few musicians could have been better suited to the task — he has been a huge fan of the unfinished SMiLE music since the 1980s. A decade later, Darian's band the Wondermints impressed Brian so much live that he took several members of the group into his touring outfit. But even with such a knowledgable ally, Brian still found the sense of expectation associated with Photo: Melinda Wilson the music almost overwhelming at first. Brian Wilson during the final writing sessions "SMiLE was the probably the last time for SMiLE, Los Angeles, late 2003. he had the confidence to follow through on a vision for many years," explains Darian. "I had to get him comfortable with the music again. So I suggested to him that it was all about performing it live. Believe me, if the assignment had been to go straight into the studio and re-record SMiLE, it would never have happened!" The first step towards creating a coherent live SMiLE presentation was to find all the recordings associated with the album, to which end Brian and Darian turned to Brian's long-time engineer Mark Linett, who first worked with Brian in 1987 on his first solo album. However, many tapes from the SMiLE period are missing, and no surviving demos have been found (for more on the '60s SMiLE recordings, see the box on page 126). Mark partially attributes this to the fact that Brian recorded the Beach Boys in several different studios during the period, but as with many other aspects of SMiLE, that's not the end of the story. "We've pretty well searched all the legitimate places," explains Mark. "I've been through the library from CBS, where Brian used to do vocals, for example; all the places that we assume would bring up tapes if they still existed. An awful lot of SMiLE has simply gone missing over the years. What's tragic about it is that I think some of it's not inadvertent; I think people have deliberately stolen stuff. So some tapes couldn't be referenced to finish the record." However, even where tapes do exist, they remain frustratingly incomplete — and Mark believes that this was always the case. "We can't rule out that there wasn't more done which is now missing, but there was no finished album, that's for sure. There's an awful lot of stuff where tracks were recorded, but no vocals, and some tracks were never recorded."

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The Assembly Line To help Darian, Mark ran off copies of the original tapes. "I gave him everything we could find as multitrack Pro Tools files, so that he and Brian could listen to the tracks as isolated as possible, learn and teach parts to the band, and work on sounds. With Brian's stuff of that period, if you try to dissect it from the finished product, especially where you have vocals on top of the instruments, you're never going to get it all." From this early stage, Pro Tools proved invaluable to the project, as Darian and Brian worked on possible presentations of the SMiLE material using the Pro Tools Session files on Darian's Apple G3 iBook. One of the problems was the interchangeable nature of much of the surviving music. Darian: "So much of SMiLE is variations on a musical theme. It was all about making those variations work Photo courtesy of Mark Linett as one piece." Mark Linett adds: "The idea of recurring themes was very Engineer Mark Linett at Sunset Sound during the tracking sessions for the completed much on Brian's mind during SMiLE. SMiLE album, April 2004. The 'Heroes And Villains' theme, for example, recurs in several songs. One of the problems I think Brian had in 1966 was that there were so many different ways to put it all together. But with Pro Tools, Darian and Brian could instantly try out different arrangements". A triumph occurred in this way when Darian and Brian noticed the similarity between the incomplete ending of the song 'Wonderful' and the musically related intro to an unfinished arrangement entitled 'Look'. Darian: "I was moving things around in Pro Tools, putting things together to show Brian. I dropped 'Wonderful' next to 'Look', and we listened to it. Brian's eyes lit up, and he said 'That's it! That's how we'll do it!'" As well as assembling existing material in a coherent order, Darian and Brian did their best to use all the available sources from the period. A poor-quality 1966 tape of Brian playing an early version of 'Heroes And Villains' supplied lyrics for two numbers, 'I'm In Great Shape' and 'Barnyard'. And while listening to the multitracks for the song 'Child Is Father Of The Man', Darian soloed a chorus lead vocal by Brian's late brother Carl and made discoveries which were later incorporated into the finished arrangements. Mark Linett explains: "When he's not singing, you can hear faint background vocal parts that no longer exist on the multitrack. They must have been in his headphones, and were picked up by the vocal mic. It could be that Brian decided he didn't need them, or that he was going to re-record them, but never did. You hear this sort of stuff throughout the tapes."



1966-vintage lyric sheets were also salvaged from the archives, proving that some songs that exist on tape with only partial vocals, such as 'Do You Like Worms?', once had more words. Once confronted with the rediscovered verse lyrics, Brian was able to remember the vocal melody without difficulty, but he and Darian had trouble reading lyricist Van Dyke Parks' handwriting. Brian's solution was obvious — call in Van Dyke! Another piece of the original jigsaw fell into place.

The Dynamic Duo Van Dyke's arrival changed everything, as Darian explains. "Another dynamic kicked in. Brian is very impressed with Van Dyke, and Van Dyke is equally impressed with Brian, so they played off each other, trying to top each other, turn each other on." Brian's confidence grew, and Van Dyke proved invaluable in boosting it when it wavered. Darian: "Part of Brian's insecurity was the fact that some of SMiLE was risky, you know... uncommercial. Take the 'Workshop' section, with all the power tools playing. Brian wasn't sure about including that, but Van Dyke said, 'We must have courage, my friend,' and so it went in." In what Darian has subsequently described as this positive, collaborative "think tank", it wasn't long before Brian and Van Dyke were coming up with musical and lyrical ideas that would finish the incomplete parts of SMiLE. Darian describes his role during this part of the process as a 'secretary' and 'facilitator', responsible for keeping track of Brian and Van Dyke's new ideas, and ensuring that they could still be realised on stage.

Darian Sahanaja of band the Wondermints, a SMiLE fan for years, has been a member of Brian's touring group for the past six years, and was charged with the responsibility of helping Brian to ready the music for live perfomance.

With unbelievable speed, SMiLE was pulled together in Los Angeles at the end of last year. New lyrics were written for former instrumentals, and some tracks, such as 'Do You Like Worms?', were retitled. Best of all, the completed songs were strung together as three continuous pieces of music, with 'Good Vibrations' acting as a coda, an idea of Brian's that apparently harks back to a structure he had in mind for SMiLE in mid-1966. The music was kept in its original recorded key at all times, so to effect transitions between pieces in different keys, a handful of short orchestral segues were constructed in a collaborative effort between Brian, Darian, Van Dyke, and Wilson live band member Paul Mertens. Mertens also helped to score the orchestration for the second part of the song 'Surf's Up'. A 1966 version featuring Brian playing the song alone at the piano has survived, and a more produced instrumental track for the first section exists featuring percussion, basses and horns, but no similarly



developed recording of the second section has ever been found. "I asked Brian what he remembered of it," says Darian, "and he said there were some strings, so we worked on that a little bit, and Van Dyke and Paul Mertens did some orchestrating."

SMiLE Or Smiley? More discoveries were made as SMiLE was being assembled. When Brian's confidence was judged high enough to listen to 'Mrs O'Leary's Cow', a dischordant, frightening piece of music that had severely unnerved him even back in 1966, Darian was amazed to hear him humming along to it. And it sounded familiar... Also known as 'Fire', 'Mrs O'Leary's Cow' (named after the farmyard beast that supposedly caused the Great Chicago Fire of 1871 by kicking over a lantern) shares chordal similarities with 'Fall Breaks And Back To Winter', a recording that made it onto the later, much simplified Beach Boys album Smiley Smile. Unlike the incomplete 'Fire', though, 'Fall Breaks...' has Photos: Melinda Wilson vocals, and it was the melody line of Brian Wilson (left), his collaborator and these that Brian was singing. "It just original SMiLE lyricist Van Dyke Parks (right) made so much sense," says Darian. and self-styled musical 'facilitator' Darian Sahanaja working to complete SMiLE in LA, "'Fall Breaks...' is a reworking of 'Mrs late 2003. O'Leary's Cow'. It's the same chords, just a different arrangement." The vocal harmonies were duly restored to the live arrangement for 'Fire'. Several other Smiley Smile pieces with rumoured connections to the original SMiLE were considered for the concert, as Darian explains. "There's a piece called 'He Gives Speeches', another SMiLE-era out-take, but Brian didn't want to do that," says Darian, "and one called 'With Me Tonight'... it's just another one of those tracks that sounds like a part of 'Heroes And Villains', or also part of 'VegaTables'. I played it for Brian, but... I imagine that it's just like when you're making a movie — you film a lot of scenes, and then it's impossible to fit them all in. You're most likely going to leave footage on the cutting-room floor." Darian feels that the finished results are in keeping with the music recorded 37 years ago. "I don't think there's a single piece in there that doesn't incorporate something that Brian or Van Dyke wrote. Even the little segues and transitions are all little phrases from 'Surf's Up', or the 'cantina' section of 'Heroes And Villains'... It was really important to me that we kept the integrity of what SMiLE was about; the feeling you get from those original recordings and arrangements. We didn't want there to be anything jarring, like a DX7, or a some kind of drum file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (6 of 20)9/26/2005 12:37:54 AM


machine beat." And of course, Darian wasn't the only one concerned in this way — he feels that a splendid 'acid test' is Van Dyke's stated happiness with the results of the revived project: "Van Dyke is a man of total integrity. If he felt that this was going in the wrong direction, he would have stepped out right away, like he did in the '60s."

A Digital Keyboard? In the interests of staying true to the original arrangements, string and horn players, the Stockholm Strings And Horns, were hired for the live performances. But when it came to reproducing the keyboard sounds live, Darian hit a problem. The original recordings used many delicate keyboards, such as electroacoustic organs, harpsichords, and multiple grand and tack pianos, as well as subtly detuned prepared piano. The difficulties of taking all of this on the road proved insurmountable, and he opted instead for a pared-down rig consisting of a digital Hammond organ, a real glockenspiel for bell and chime parts, and the synth with the best samples of the other keyboards he could find — Kurzweil's top-of-therange K2600 sampling workstation. Even so, the logistics of zoning the samples so that he could play different instruments simultaneously from different parts of the keyboard were formidable. And the detuned piano parts required some serious synth reprogramming in the depths of Kurzweil's VAST synth engine by Dave Weiser, an engineer from Kurzweil R&D.

Brian Wilson's '60s Productions Despite being almost deaf in one ear, Brian Wilson was responsible for some of the most imaginative record production techniques of the '60s. His musical achievements are reasonably well documented, for example in the detailed notes accompanying the Pet Sounds Sessions box set, which explain how he arrived at some of his arrangements with studio session musicians. But his technical achievements have received less attention. When Brian began producing in 1963, most LA recording facilities offered three-track recording at best, the idea being that this would accommodate a stereo backing track and leave a track free for a lead vocal. But Brian had no interest in stereo. Together with his favourite engineer Chuck Britz, he rapidly began stretching what could be done with those three tracks. "Brian became very adept at getting his tracks live in mono, or maybe with one overdub," explains Mark Linett. "And then he'd have a track, or maybe two, for the vocals, which was

Photos: Jasper Dailey, courtesy of David



Leaf. usually enough, because in the early days, the Beach Boys would do the Brian at Western Recorders, LA, during the vocals live round a couple of mics. '60s SMiLE Sessions. So he could double the lead and backing vocals with the remaining spare track, which was an effect he liked."

From 1963 onwards, Brian began planning his arrangements carefully, frequently leaving carefully timed instrumental silences in the backing tracks, over which he would later record a cappella harmony breaks with the Beach Boys. He also began to use tape-cutting techniques around 1964, initially to splice short, hard-tosing a cappella intros or outros on to the beginnings or endings of previously recorded songs. He soon outgrew the tracks that were available to him. Mark Linett: "If he needed to keep overdubbing after he'd filled up all his tracks, he'd make a reduction mix to one mono track of a new tape, and then he'd have some more spare — although this wasn't unusual for the time. "In some cases, like [1965 US number one] 'Help Me Rhonda', a whole bunch of overdubs were done as the tape was being mixed to mono, to save another layer of tape copying, and the generation loss that goes with it." This means, though, that those mono mixes are the only places you find those overdubs, and true stereo remixes from the multitrack that precisely replicate the mono single mixes are now impossible. From 1965, the record label Columbia acquired an eight-track recorder at their LA studio. This changed everything. "By now, Brian was recording the instrumental backings on three tracks. I think this was just something the engineers did with his consent, so that he would have a little more control when he mixed it to mono, which he would do almost immediately. And then he'd either put that mono backing onto one track of an eight-track tape at CBS, or one track of a new threeor four-track tape. And then he'd overdub vocals onto the remaining spare tracks. Half of the Summer Days (And Summer Nights!) album was made using the eighttrack at CBS like that, and so was half of Pet Sounds." The Summer Days album also saw Brian becoming bolder with tape splicing. The a cappella harmony piece 'And Your Dreams Come True' was recorded in sections of a few bars long, and then carefully edited together. "I've never really asked him where this concept of doing things in pieces came from," admits Mark. "He certainly wasn't the first person to do edits, but it was unusual to record a song in four or five sections, and then cut it together. The vocal double-tracking was usually done over the top of the edited version." 'Good Vibrations', begun during the Pet Sounds sessions, took tape-splicing techniques to new heights. Darian Sahanaja comments: "Before that, by and large, they'd go in and cut an entire track, and then he'd lay vocals over it. But with 'Good Vibrations', he'd just hire the top session guys, go into the studio, and spend three or four hours working on a groove, trying different tempos and instrumentation... It wasn't a finished song, it was just a riff. He'd do it again the next day in a different way, and again the next week, and after a few months, he'd have all these different variations on a theme. Eventually, he edited them together in the form of a song, and we got 'Good Vibrations'. For every section of the finished piece, there's hours of just the verse, just the chorus, each with slightly different instrumentation, different rhythmic feels, and different percussion... That was tremendously satisfying to him artistically, and of course, it also did very well on the charts. So then he decides he's going to do a whole album that way. That was the idea of SMiLE... a whole album of these 'feels' that he was going to file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (8 of 20)9/26/2005 12:37:54 AM


weave together. And that's why you've got all these recorded sections for 'Heroes And Villains'. It's a bunch of different experiments, and some of them branched off and became separate songs." 'Good Vibrations' was also groundbreaking for being the first song in which Brian began to realise that if he was constructing the song from sections anyway, he might as well reuse some of them. The instrumental backing for the choruses is the same recording repeated several times, as are the first two verses. "In a way," says Mark Linett, "Brian invented the method of modular recording that we take for granted today. Initially, he would still construct it as Brian listening at one of Western's a song, you know, copy the verse twice, monitors during a SMiLE tracking copy the chorus twice, and then splice it all session. together. Eventually, he didn't do that either; he would just put two different lead vocals on different tracks on one section, mix with one, then mix with the other, and he'd have two verses from one piece of tape." This is how the SMiLE-era version of 'Vega-Tables' was recorded, which provides a masterful example of using extensive overdubbing to build up a track from very basic beginnings. The track began with Brian playing solo piano backing for two sections, with a brief pause in the middle. He then added a barrage of overdubs to the track, including bass, percussion, interlocking vocals and several leads on each section, seemingly intending to mix most of the sections he needed for the song from these two parts. Unfortunately, this method makes it very difficult to work out which leads and overdubs he wanted to include on each section, and in what order he intended to place them. To confuse matters further, Brian also recorded further sections for 'Vega-Tables' later on. Arguments about the correct sequence of the SMiLE version, and the choice of sections, continue among fans to this day. Following his abandonment of SMiLE, Brian moved to a home-based studio, but continued to use tape-splicing and section-repeating techniques throughout the next three albums released by the Beach Boys. Many of the tracks on 1967's Smiley Smile have their roots in SMiLE material — 'Good Vibrations' is the 1966 single version, and 'Heroes And Villains' and the mostly re-recorded version of 'Vegetables' contain some spliced-in sections recorded during the SMiLE sessions. Other SMiLE titles, such as 'Wonderful' and 'Wind Chimes', were completely re-recorded for Smiley Smile in much simpler arrangements. However, Brian still put a lot of work in to build up the tracks from basic foundations, using a piano much as producers began using click tracks over a decade later, to lay down a tempo map over which the rest of the song



could then be overdubbed. The piano was frequently mixed out in the final version of the song, as in the Smiley Smile version of 'Wind Chimes'. Overall, although the tape edits carry some of the spliced-together feel of the original SMiLE project over to Smiley Smile, this is a very different album, with very little of SMiLE's grand orchestration, except on the SMiLE-era singles. Instead, a laid-back feeling permeates the record; the organ-heavy backing tracks are sparse and have an almost a lo-fi, bedroomrecorded feel to them, perhaps reflecting where they were made. Coupled with the cuttogether nature of the recordings, this was a very uncommercial album to release to the world in September 1967, which was still expecting something to top the lushly produced Pet Sounds. Fans of its quirky charms rate it not least because the pared-back arrangements allow Brian's extraordinary vocal harmonies to shine through, especially on tracks like 'Little Pad', 'With Me Tonight', and the gorgeous choral tag to the re-recorded 'Wind Chimes'. After Smiley Smile, Wild Honey was also constructed largely from pieces. Even on songs like the single 'Darlin', where a complete backing track was cut, only one of the choruses was overdubbed with vocals, and then this was copied and spliced between the rest of the verses. By the release of Friends in 1968, Brian was beginning his retreat from active involvement in the group's recordings. Ironically, the other Beach Boys began to use old tapes of his performances, and his overdubbing and splicing tricks, to maintain the illusion that he was still creatively involved. Recordings with their origins in the '60s SMiLE sessions concluded the Beach Boys' three albums after Friends. The final such assembly was 'Surf's Up' on the 1971 album of the same name. Carl Wilson took Brian's instrumental recording of the first part of the song and overdubbed vocals to complete it. The backing for the second part of the song had to be supplied by splicing in the appropriate part of the surviving 1966 Brian solo piano performance. According to Mark Linett, an abandoned 1971 tape exists in the Beach Boys' archive upon which an attempt has been made to fly in Brian's double-tracked 1966 vocals from the solo piano version over the January 1967 orchestral backing track for the first part of the song. However, with the technology of the day, it was impossible to marry the variant tempos of the two recordings, and the tape contains no vocals after the first few lines. We can only speculate what they might have done with a software sampler and a laptop!

From Stage To Studio The concert opened in London on February 20th, 2004 to rave reviews all over the world from fans and press alike; Q magazine placed it in their Top Five Gigs



Of All Time the following month. All six nights were recorded for posterity by Mark Linett on a 48-track Genex hard disk recorder acquired especially for the occasion, and some of those performances will appear in a TV documentary on SMiLE that will air in late 2004. "It was an amazing moment for me at the end of 'Blue Hawaii' on the first night", comments Mark. "Sitting in the remote truck mixing the show, I got emotional as I realised that Brian had finally presented SMiLE to the world after 37 years." Darian was delighted too, particularly at the end of the second night, when Brian received a lengthy standing ovation after the SMiLE portion of the concert. "Before the first show, my biggest fear was 'Is Brian going to make it all the way through this?' The second night, after the SMiLE performance... it was the longest standing ovation that I've ever witnessed. I felt that was the night that Brian really acknowledged SMiLE as his work. He actually feels proud of it now." Two weeks later, the fate of the studio-based recording that eventually followed was still uncertain. Darian: "The idea evolved. It was only after the success of the concerts that Brian warmed up to the idea of doing the studio version." The 10-piece touring band, plus the nine members of the Stockholm Strings And Horns, eventually reconvened at Sunset Sound Studios, Los Angeles, on April 13th, 2004 following some intense preparation and a day setting up at the studio. With Brian supervising and honing the performances via the talkback, as of old, they finished recording the required instrumental backing tracks in a mere four days, thanks to Darian's efficient organisation before the sessions and the group's fine musicianship. "When Brian cut this originally, he started with just an idea, and had to work out the sounds and arrangements. We came in knowing the stuff; the road was already paved! But it was still a lot of work. I had all these sheets printed out showing the sections, all colour-coded, to make sure we got everything down." The recording was made using a sensible mixture of old and new techniques and technology where appropriate. It was decided to cut the instrumental tracks in a way that, for the most part, would have been very familiar to Brian in the late '60s. Mark Linett takes up the story. "We put everybody in the same room, apart from the strings and horns, and decided to go for the sound of good, creative bleed. Sunset's Studio 1 was the original site of sessions for both 'Good Vibrations' and 'Heroes And Villains' in 1966, so we knew this method would work. "Before eight-track came along,

Photo: Richard Ecclestone Engineer Mark Linett and Genex operator Simon Burgess in the mobile recording truck outside the Royal Festival Hall, London, February 21st 2004, preparing to record the second night of sold-out live SMiLE performances. The Genex GX9048 hard disk recorder on the right was used to make all the 48-track live recordings, which are destined for use on a SMiLE documentary, to be screened in late 2004.



everything was still recorded pretty much live in one room; all the players could hear each other, and to record them and balance them off against each other, you'd move them physically around the room, or move the mics. And most importantly, you looked for good bleed. What people don't understand about those records of Brian's or Phil Spector's, their big sound, and the reverb, is that it was all recorded in a very small room. Studio 3 at Western Recorders, where Brian did many sessions in the '60s, is only something like 30 feet by 12 by 20. The reason it sounds so big is that you can use the reflected information — it sounds good. "Nowadays, if you're making a rock record, you start with the drums and build it up. If we go right back to the days when there would have been a live vocalist in the room, you would have started with the vocal mic, with all the bleed that was coming into it, and then add enough of the other instrumental mics to fill in. And if that had too much bleed, then you'd have to move things around and make everyone play differently. You have to think about dynamics and arrangements. [Drummer] Jimmy Hines commented that his cymbals and hi-hats were virtually untouched, which was typical of recordings of the period — the high-end stuff was almost always done with percussion, because there was more control. Splashy cymbals would have obliterated everything on the three-track! "We used the isolation booth at Sunset for the live strings and horns, because we had the same small section we did on stage: two violins, a viola, two cellos, and four horns. Originally, you'd put everybody in one room, and so you'd have to have a certain number of strings and horns just to be loud enough to compete onmic with everyone else. When amplified rock music came along, you really had to have more isolation. Sunset was the first studio in town to build an iso room big enough to put a small orchestra in. "It was funny that even though we made it very clear that we were going to be recording everybody live in the same room, I still had musicians coming up to me after takes saying, 'I wanna fix that on a drop-in'. And of course we couldn't. While it's nice when you can do that, and sometimes it's an advantage, it means that you're not capturing entirely real performances. Back in the '60s, you played it right first time, and if you didn't, they got someone else who could! The result of that was music with much more human interaction present, and I think we achieved the same thing." "For miking, we tried to stay pretty close to what would have been done back then... so there wasn't much in the way of condenser mics, because you tend to get extremely intense bleed with those. For the most part, it was dynamics and ribbon mics [see diagram on page 128]. The drums all had dynamics on them. We had a full percussion rig miked up with dynamics and condensers, and decided which to use depending on the sound of the track we were recording. Sometimes we cut both and decided which to use later. I had two setups on the grand piano that I could call on: a close-miked arrangement and a Mid + Side setup for the bigger-sounding numbers, like 'Surf's Up', where we wanted something more 'roomy'. We also used the sound of Sunset's live echo chamber on the drums and percussion, and printed the reverb returns, just like Brian used



to."

The Original SMiLE Recordings In 1965, Brian was the acknowledged master of the twominute pop song, but it was not enough for him. By 1966's Pet Sounds, he was reaching for larger, more adult lyrical themes and coupling them to longer tracks, and after the realisation of production epic 'Good Vibrations', the new album, SMiLE, was to take these developments even further. Photo: Guy Webster, courtesy of Brian and Enlisting poetic wordsmith Van Melinda Wilson. Dyke Parks to help him with lyrics, The Beach Boys learning parts from Brian Brian began to work on a bewildering variety of themes for the Wilson around the piano at a SMiLE recording session, November 1966. new record: spiritual development, emotional growth from child to man, physical health, the environment, and the Old West, the last of these perhaps best encapsulated in 'Heroes And Villains'. And the new music was awe-inspiring and evocative, too, the arrangements painting in sound what Van Dyke's lyrics achieved in rhyme. The calm verses of 'Cabin Essence' conjured up rural life on the farm, before the thundering 'iron horse' of progress, the railroad engine in the chorus, swept everything before it — and it really sounded like it, too. 'Fire', written to evoke an enormously destructive conflagration, sounded so alarming that even Brian was unsettled by it. Journalist Jules Siegal watched the recording of what he would later describe as "sequences of great power and beauty" for 'Heroes And Villains', and even Leonard Bernstein was full of praise for 'Surf's Up'. But as 1967 dawned, Brian proved incapable of completing 'Heroes And Villains', reworking it several times and constantly adding new sections. The other Beach Boys, alarmed at the avant-garde turn the music was taking, and concerned that it would be virtually impossible to reproduce on stage, rounded on Brian. In the Spring, Van Dyke Parks, disturbed by these developments, left the project. Photo: Jasper Dailey, courtesy of David Leaf. Eventually, perhaps feeling that his Brian Wilson and Van Dyke Parks (right) creative vision was now fatally working on SMiLE, 1966. compromised, Brian scrapped SMiLE, dramatically scaling down his ambitions at the same time. He stopped recording with the Wrecking Crew at external studios and had one built instead in his home. In early Autumn 1967, the homespun, much simplified Smiley Smile was released, and failed to crack the US Top 40. A partially re-recorded, shortened single version of 'Heroes And Villains' had already dropped off the charts. file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (13 of 20)9/26/2005 12:37:54 AM


At the start of the SMiLE sessions, Brian bestrode the world; after Smiley Smile, he rarely left his house. For years afterwards, whenever the occasional curious journalist enquired about SMiLE, Brian would always play it down, maintaining that he had destroyed the master tapes, or writing off the music as unfinished, drug-crazed self-indulgence. By the late '80s, when Mark Linett began to catalogue the recordings from the period, the library was in a terrible state, with many missing tapes. The most famous of these is the eight-track master tape for 'Good Vibrations' with all the vocal overdubs, but there are countless other examples. The session tape for the SMiLE-era minor-key cover of 'You Are My Sunshine' exists, but lacks the master take. According to Mark Linett, it was physically snipped off and added to a separate compilation reel (which also included the multitrack master of the track 'Wonderful') so that Brian's brother Dennis could add a vocal. Mono mixdowns of 'Wonderful' and 'You Are My Sunshine' performed from this reel (the latter in 1968) exist as evidence of this, but the comp reel itself has disappeared. Similarly, a tape exists which includes a safety copy of the track 'Do You Like Worms?', featuring backing vocals that are no longer present on the main surviving archive tape of the song. Documentation shows that the safety was made from a fourtrack tape — but that tape, of course, is no longer in the archive. "I hope some day we can get all this stuff back", says Mark. "I don't care how somebody got something — if they have it, I'd love to hear from them." Perhaps the greatest mystery of all the original SMiLE recordings concerns the second part of the instrumental backing track for 'Surf's Up'; it's completely unclear whether it has been stolen, or whether it was never recorded in the first place. Darian Sahanaja says "The talk is that it exists, somewhere," but Mark Linett's reading of the situation is more prosaic. "If it does exist, we haven't found it."

Picking Up The Pieces The backing tracks were also recorded in a way that was very familiar to Brian. Although the 'three-movement' structure and order of the songs was not altered from the live version, the songs were cut out of sequence in sections and strung together later, and similar instrumental sections, such as the verses and choruses of 'Child Is The Father Of The Man' or 'Cabin Essence', were recorded only once and copied, as on the original sessions. Not all of the songs were recorded in exactly the same way — ironically, on 'Good Vibrations', the song on which Brian had originally fully developed the method of recording in sections, the first two verses and choruses were played straight through as one piece — but on most of the songs, the sections were divided up as on the '60s recordings, even the opening wordless chorale, 'Our Prayer'. Darian: "The only way to do it was in pieces. Listen to the '60s versions of 'Good Vibrations' or 'Heroes And Villains', which were both released; you can hear the file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (14 of 20)9/26/2005 12:37:54 AM


cuts as they go by, and that has so much to do with the feel of the original recordings. Van Dyke is a huge fan of those hard tape-style edits, he thinks that's so 'Brian'." For all that, the recording itself was not made to tape, but to Mark Linett's Pro Tools HD system. But given the nature of the record, Mark thinks this too was only sensible. "Randomaccess editing has been a part of this from the very conception. Brian was making a record that was tailor-made for something like Pro Tools, but trying to do it 40 years ago! If he had had these tools, who knows what he might have been able to accomplish. Any concerns about the sonic quality — and I don't have any of those now we're recording at high resolutions, with better converters — are more than outweighed by the benefits. At heart, I'm an analogue guy — I still mix in analogue — but I have no problem with the sound of Pro Tools. I don't think recording to tape would have benefited us sonically, and we would have had to transfer into Pro Tools pretty quickly for the editing anyway." Mark's system comprised a custom Pro Tools HD rig, recording at 24-bit, 88.2kHz. An Apogee Big Ben was used as the master clock ("Having an external clock is a must, if you can afford it," he avers), and he expanded the I/O from its usual 32-input, 48-output configuration (on an Apogee AD16 and Rosetta, and DA16s respectively) to 40 inputs by adding another Rosetta. "We didn't record that many tracks at once — it was more that we had that much stuff out in the studio, and I wanted to be able to say "Right, we're going to add those inputs,' without having to think about it."

Diagram: Sam Inglis, courtesy of Mark Linett A plan view of the rough layout of the 2004 SMiLE tracking sessions at Sunset Sound. The control room is shown at the bottom, and the isolation booth for the strings and horns is at the top, with the main studio area in the middle.

Pragmatically, a few instruments were added after the basic tracks were recorded, such as on 'Surf's Up', where the bass line is composed of several bass parts. To save having to teach parts to new bassists so they could all be recorded live, the electric bass was recorded first, and then acoustic bass was overdubbed. On a few songs, Mark double-tracked the strings and/or horns to more closely reflect the number of players on the original sessions. When recording 'Fire', which originally featured a large string section, the Stockholm Strings were triple-tracked. "Overdubbing is so easy in Pro Tools... we'd just arm the tracks and go. In the old days, you needed to repatch, make sure you had enough tracks, repatch the monitoring...". Although a grand piano was used live, an upright piano, which was not available at Sunset, was overdubbed at Mark Linett's Your Place Or Mine studio later. An early plan to replace the sampled Kurzweil harpsichord with the real thing later was abandoned, as everyone professed themselves happy with the sounds from the K2600 on hearing rough mixes.



Aside from these tweaks, a few alterations were made by Brian and Darian to instrumental arrangements between the concerts and the recordings. For example, care was taken to replicate the delicate rhythm arrangement of the original 'Heroes And Villains'. Although the track has a strong 16th-note feel, it derives this not from hi-hat parts, but from a combination of snare and tom-toms, with the high-end feel supplied by strummed acoustic guitar and banjo. Likewise, some of the heavy tom-tom fills used to power along the live rendition of 'Good Vibrations' were pared back to reflect the more restrained studio arrangements. In contrast, some instruments that had been omitted to allow the music to be performed by the 10-strong group live were restored, such as the timpani and parade drum used on the verses of 'Roll Plymouth Rock', the former 'Do You Like Worms?'. Vocal sessions began towards the end of April at Your Place Or Mine studios. The background vocals and beds for the clutch of a cappella-only sections were recorded first, with Neumann U47 and U67s, in classic Beach Boys fashion (three to five vocalists live at a time, with a bass vocalist on a separate mic). Work continued on Brian's leads through May and into June, recording for the most part on a Neumann U67 or an old Shure 545 dynamic, running through Universal Audio 610 and 610a valve mic preamps (just as on the '60s recordings). This signal was then passed straight into Pro Tools via a Fairchild 670 compressor and the Apogee AD converters. As with the instrumental arrangements, Brian and Darian tweaked the vocal arrangements for the studio performances. Mark: "The tracks that got vocals back in the '60s were heavily multitracked, and we tried to do the same thing, doubling vocals and using whoever was most appropriate for certain parts, rather than being forced to do everything at the same time, as on stage." One set of 'vocals' remained unaltered; the percussive sounds of vegetables being chewed and chomped on the quirky paean to good health, 'Vega-Tables'. As the band battled over who would get to play 'lead celery', Darian and Mark began trying to work out how best to close-mike a variety of market produce. Darian: "That was hilarious, finding out which vegetables would give you really monster crunches down a mic..." And the result? "Carrots were OK, but celery is the one. You get that big tearing sound from all the fibres!"

Mix Me A SMiLE



Mixing began in June and ran into early July, with Brian and Darian present. As is his preference, Mark mixed in the analogue domain, using Your Place Or Mine's custom API 2488 console, and both Pro Tools-based and Flying Fader level automation. Similarly, processing was carried out both by plug-ins such as Waves' Renaissance Compressor and also analogue outboard, including a Fairman TMEQ six-band valve equaliser, and a Universal Audio 175A compressor. Universal Audio also supplied a UAD1 card, which ran several hardware emulations as plug-ins, including a Pultec EQ plug-in. The mix proceeded from Pro Tools via Mark's Apogee D-As, was mixed and laid back to Pro Tools in stereo at 24-bit, 88.2kHz resolution, passing back in via a highquality DCS 904 stereo A-D converter. Tannoy SGM10s (with Mastering Labs crossovers), handled the monitoring, fed from a DCS 954 D-A attached to the stereo output from Pro Tools. The mixing nonetheless posed a problem which had not been resolved even after the basic backing tracks had been recorded. On the original SMiLE recordings, Brian recorded most of the songs in sections, but cut them together to make a complete backing track before adding vocals over the top. The exception was 'Heroes And Villains', where vocals were added to sections prior to assembly. But what was the best approach for the 2004 recordings? In the end, a 'Heroes And Villains'-type approach was adopted, whereby vocals were recorded separately on each section, each section was mixed into stereo complete with its vocals, and then the parts were assembled into songs by slightly overlapping them on two sets of stereo pairs in Pro Tools. Mark: "Everything comes in 'hard' on each section after the count-ins — which the editing process removed — so it was just a question of how we got out of the previous section. Usually I let the natural reverb tail off over the start of the next section, or did very quick crossfades.

Photo courtesy of Mark Linett The Pro Tools Session for the third SMiLE movement during mixing at Mark Linett's Your Place Or Mine studios. On screen, the orchestral intro to the third movement has already been assembled with the next section, 'I'm In Great Shape'.

"If we had a song with multiple verses and multiple choruses, we would mix all of those first, and then put them together. So there was never an assembled multitrack from which we mixed whole songs. If I'd known we were going to do it this way, I would have done it differently. In some of the repeat sections, we did wind up reusing a lot of the same backing tracks, and in some cases the same background vocals, just like in 1966, but I created a separate copied section each time the lead vocal was different. And that meant mixing each section separately; so in order to ensure that they still sounded alike, anything we did in Pro Tools' automation, and also on my console automation, had to be duplicated in both sections. In retrospect, it would have been easier and faster if we had done it all on one copy of the section, with the different leads recorded on two separate sets of tracks, and then just brought up the lead vocal we needed for file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (17 of 20)9/26/2005 12:37:54 AM


that particular verse or whatever. Just like Brian did on the original version of 'Vega-Tables', actually." Some effects were added during the mix, including the transfer functions of several sampled echo chambers and spring reverbs using Audio Ease's Altiverb, and the profiles in Mark's hardware Sony DRE S777 convolving reverb. "I have a large collection of analogue reverbs, including springs. I love them, but they tend to be very noisy. When you use the sampled versions, you retain all the nice qualities, but you get rid of the mechanical noise," explains Mark. Altiverb was also used for limited-bandwidth vocal effects, lending an 'old gramophone'-style ambience to the snatches of old cover versions dotted around the completed album. But by far the most obvious Altiverb moment was the emulation of Brian's original 'tape delay plus feedback' explosion effect. Brian experimented with this then-innovative effect on the track 'I'm In Great Shape', recorded in late October 1966. Back then, he used tape delay, but fed the delayed feed back into itself at high gain levels so that the delay built cumulatively until the tape he was recording on saturated completely, creating the effect of a delay-based explosion. Later, early in 1967, he recreated the effect on an early version of 'Heroes And Villains'. For the re-recorded SMiLE, the effect was restored to 'I'm In Great Shape', and was created using a delay plug-in running on Mark's Universal Audio UAD1 card. "We could have done it the way Brian did; I still have the tape machines. But the UAD plug-in just sounded great. It overloads in a very analogue-sounding way, and gives you full control over the regeneration level, the feedback, the delay time, and so on. And the great thing is, because it's a plug-in, you can automate it! So we mixed that section without the effect, and then I automated the plug-in to create the effect when we were assembling the sections. We experimented with the settings several times to create the best effect."

Further Reading & Listening If your curiosity has been piqued, and you want to know more about SMiLE, seek out Domenic Priore's book Look! Listen! Vibrate! SMiLE! (ISBN 0-86719-417-0), which collects most of the contemporary writings about the original sessions in one handy scrapbook-type volume, together with some exploratory essays and much theorising from Domenic. Some of this is based on information that is now out of date in the light of discoveries made since the book's last revision, but Priore's burning enthusiasm for the music shines through, and it's worth bearing in mind that it was first written at a time when perhaps barely 50 people worldwide knew anything about the subject. An updated version is apparently in the works. file:///H|/SOS%2004-10/The%20Resurrection%20Of%20Brian%20Wilson%27s%20SMiLE.htm (18 of 20)9/26/2005 12:37:54 AM


There are plenty of sites devoted to the Beach Boys on the 'Net, but the best one devoted to SMiLE is unquestionably The SMiLE Shop (www.thesmileshop.net). Founded in 1998 by SMiLE fans Jon Hunt and John Lane, the Shop collects essays, information and session data for the original recordings in one place. Even better is the discussion board attached to the Shop, which is one of the bestbehaved and most thought-provoking forums around. Actively presided over by the two Jo(h)ns, who post there most days, there's no finer place to discuss SMiLE, or indeed most other pop and rock music. There's no way to hear a finished version of SMiLE other than by buying this September's release. However, some of the recordings from the '60s are commercially available, mostly on the 1993 five-disc Good Vibrations boxed set. Even in their unfinished form, the 30-odd minutes of SMiLE tracks compiled and mixed for Disc 2 of the set by Mark Linett convey a powerful, compelling sense of what might have been. Sadly, most of the '60s tracks and recording sessions are only available via the shadowy realm of the bootleg collector, and some are rumoured to have been lost forever. Dedicated fans of the original recordings continue to hold out hope that Capitol Records, with Brian's agreement, will one day officially release a multi-disc boxed set of all of the unfinished '60s sessions. Given Brian Wilson's enthusiasm for the newly completed version of the album, however, this seems unlikely to happen in the short term.

37 Years In The Making Mixing and assembly were complete by mid-July, only just before Brian and the band had to travel to Europe for another round of SMiLE concerts. A few tweaks were deemed necessary before mastering, and Mark sought the approval of Brian and Darian by putting Pro Tools Sessions of the slightly remixed sections up on a secure FTP server so that they could download them and hear them in their London hotel rooms. The same method was used to transfer the final Brianapproved mix to Bob Ludwig of Gateway Mastering on July 16th. By late July, the record was done. The completed album stands up as Wilson's masterpiece, presenting its music in high-fidelity stereo, with vibrant bass and silky high end. The cut-together arrangements accentuate the strong dynamic contrasts throughout, from the shimmering, glittering beauty of the gentle harpsichord and vocals on 'Wonderful' to the pounding timpani on the verses of 'Roll Plymouth Rock', and the sawing, driving cellos in the choruses of 'Cabin Essence' and 'Good Vibrations'. And Brian Wilson's arrangement skills are all over the record, from the muted trumpet of 'Child Is Father Of The Man' to the rasping, dixie-like trombones of 'On A Holiday', from the mournful horns of 'Surf's Up' to the soaring strings of 'Blue Hawaii', and from the absurdity of the celery accompaniment in 'Vega-Tables' to the harmony acrobatics of 'Heroes And Villains'. You'll be hard-pressed, too, to find another album in 2004 that includes something as beautiful as the sublime 'Our Prayer' on the one hand, and the downright terrifying 'Mrs O'Leary's Cow' on the other. In short, this is a SMiLE that finally fulfils the promise of the sadly incomplete '60s recordings — no mean achievement.



As my final interview with Mark Linett wraps up, he is poring over a test acetate of the fourth side of the double vinyl gatefold album. With three sides taken up by SMiLE, the fourth will include instrumental-only mixes of four songs which allow the complexity of the backing tracks to shine through. And there is talk of a 5.1 version for next year. Funding has yet to be confirmed, but Mark, in his enthusiasm, has already begun to create rough surround mixes of 'Surf's Up' and 'Wind Chimes'. "We're all very proud of this," he says — and rightly so. Darian Sahanaja certainly is. And Brian? Darian sums up: "In theory, it was probably the worst thing you could present to Brian: finish SMiLE! And yet we did it. It's a miracle that everything was in the right place at the right time, but Brian is so proud of it now. It's all been worth it, just for that." SMiLE will be released on Nonesuch Records on September 27th. 'Wonderful' is coming out as a limited-edition UK vinyl single on September 20th. Published in SOS October 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:


Hard Sync & FM Synthesis Ever Heard A 48dB/octave Programming Guide Published in SOS October 2004 Filter? Drive Effects Print article : Close window Tracking Generator Technique : Synthesis Are You Up To Date? Using The Ion As A Beat Box Quick Tips The Ion As Signal Processor Recording Performances Although the Ion is already well-respected for its Into A Sequencer sound quality, its synthesis architecture is also The Most Important Tip Of surprisingly flexible. We show you how to get the All!

from this little beast in your studio.

best

Craig Anderton

Despite its low cost, the Ion is well endowed with features, and it's remarkably well built: for example, the rubbery knobs are mounted to the front panel with nuts, so you can lean on them without fear of stressing the underlying circuit board. Plus this synth has the advantage that the filter cutoff and other controls have an analogue 'feel' because of how they divide their rotation into thousands, rather than hundreds, of steps — the encoders have about 8000 steps of resolution.

Original photo: Mark Ewing

However, what immediately strikes most musicians is the sound — it certainly impressed reviewer Paul Nagle back in SOS September 2003. While few would say it's a clone of classic analogue synths, the Ion stands on its own thanks to a full bass and brash, but not arrogant, high end — apologies if I'm sounding like I'm doing a wine-tasting! Although the factory presets do a good job of showing off the synth, there's a lot that's hidden 'under the hood', so I'm going to show you some ways to squeeze additional options and sounds out of what is already a very capable synthesizer.

Hard Sync & FM Synthesis file:///H|/SOS%2004-10/Alesis%20Ion%20User%20Tips.htm (1 of 10)9/26/2005 12:38:10 AM


The traditional sync sound results from using the modulation matrix to feed an envelope with a moderate decay into the Osc 2 Pitch Full modulation matrix destination (and into Osc 3 Pitch Full if you're using both as sync oscillators). You also need to turn down the Osc 1 Level parameter from the appropriate Pre Filter Mix screen. Then enable oscillator sync using the dedicated Osc Sync button. However, I must admit to being somewhat disappointed with this sound; it didn't immediately have the smoothness I wanted. The way around this was to turn up the Ring Mod Level parameter to the same value as Osc 1 Level, again in the Pre Filter Mix screens. This was the aural equivalent of 'filling in the cracks', and resulted in a smooth, yet defined, hard or soft sync sound. Although the Ion won't replace a Yamaha TX802 or Native Instruments' FM7, it can do simple FM (frequency Routing the output of the pre-filter mixer modulation) effects where one or more through the Ion's filters in series is the key to creating the steepest response slopes. oscillators serve as modulators for another oscillator. There are three basic FM algorithms: Oscillator 3 modulates Oscillator 2 which modulates Oscillator 1; Oscillator 3 and Oscillator 2 modulate Oscillator 1; and Oscillator 2 modulates Oscillator 1. You select the algorithm under the Oscillator 1 Edit button. We'll choose the last of the three algorithms for a very simple example of how you can use FM on the Ion. In Pre Filter Mix, make sure that all audio sources are turned down except for Oscillator 1. One classic FM sound is a marimba/thumb-piano percussive attack. For this, set both Oscillator 1 and Oscillator 2 to sine waves, and transpose Oscillator 2 up one or two octaves compared to Oscillator 1. Now set the pitch/mod envelope for a fast decay (for example: Attack Time 0.56, Decay Time 15.78, Sustain Level zero percent) and, in the modulation matrix, assign it as a modulation source with FM Level as its destination. The Pitch/mod Env Level parameter should be around 30 percent, with zero percent Offset. This injects the Oscillator 2 signal into Oscillator 1 for a brief transient, adding the complex, interesting attack associated with FM synthesis.

Ever Heard A 48dB/octave Filter? If you want a really fat low-pass filter, you can gang two 24dB/octave filters in series and have them track each other. It was extremely difficult to get this kind

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of response with analogue circuitry, but using the Ion's Moog-emulation filters gives a good combination of vintage sound and digital accuracy. To do this, hit the Pre Filter Mix button and set parameters as follows. The object is to send all audio into Filter 1 only, and not into Filter 2, and then to route the Filter 1 output to Filter 2 so they're in series. Osc 1 Level and Osc 2 Level: 100 percent. All four other levels: zero percent. (I'm just doing a two-oscillator patch for now.) Osc 1 Balance and Osc 2 Balance: zero percent to Oscillator 2 and 100 percent to Oscillator 1. F1 -> F2 Level: 100 percent. (This effectively puts the filters in series. The other Pre Filter Mix parameters don't matter. Now hit the Post Filter Mix button and make the following adjustments:

If you latch all the arpeggiators in a four-part multi-timbral setup, it's possible to create a complete rhythm groove. Once this is on the go, you can easily drop the parts in and out using the Part Enable buttons.

Filter 1 Level: zero percent. Filter 2 Level: 100 percent. Pre Filter Level: zero percent. Filter 2 Pan: centre. Filter 2 Polarity: positive. The other parameters aren't crucial; the point is to listen only to the output of Filter 2, not Filter 1. Assign the response for both filters to MG4 LP, a 24dB/octave analogueemulation response. You probably want the two filters to track; the quickest way to evaluate this patch is to assign one of the mod wheels to the cutoff frequency of both Filter 1 and Filter 2, setting identical Level and Offset amounts (start with 100 percent and zero percent respectively). And now, the payoff. Play some notes and move the mod wheel — you'll hear a mondo low-pass filter. To compare it to a 24dB/octave response, hit the Pre Filter



Mix button, then change the Osc 1 Balance and Osc 2 Balance parameters so that 100 percent of both signals goes into Filter 2. All of this helps illustrate the flexibility of filter mixing within the Ion. You can send audio sources (oscillator, external input, ring modulator, and so forth) to either or both filters in any percentage, pan the filter outputs in the stereo field, include prefiltered sound, and more. It's worth spending some time in the Pre Filter Mix and Post Filter Mix screens to become acquainted with the possible routing options.

Drive Effects Some people have complained that the effects selection in the Ion isn't extensive enough ("dude, where's my reverb?"). Hey, who cares? You can always patch the Ion into a multi-effects box! Still, let's not overlook the effects that are built in, particularly the Drive effects. Available distortion devices are Tube Overdrive, Tube Amp, Distortion, and Fuzz Pedal, all of which are related, but which have somewhat different overall sounds. Distortion can add an 'oomph' to kick and bass sounds that goes somewhere beyond macho, as well as making polyphonic pads more gritty. The Drive Level parameter is a crucial adjustment for the drive effects. As this will change the overall output, trim it with the Program Level value to match other patches. The RMS Limiter is excellent for holding down resonant peaks from filters, while the Compressor can really 'harden' a sound if you feed the effect with high drive levels.

Tracking Generator The Tracking Generator feature recalls the glory days of the Oberheim Xpander, and allows considerable flexibility in altering controls. Basically, it re-maps a control input (shown on the 'X' axis of the on-screen graph) to create a different output (the 'Y' axis). The Tracking Generator appears as a modulation source within the modulation matrix. Each column represents a point on the Tracking Generator curve; the point's value is adjustable. The Tracking Point parameter selects a particular column, while Level chooses the value. Another parameter chooses the control signal that appears at the Tracking Generator input (for example, mod wheel, note velocity, or keyboard tracking). One situation where I've used the



The Ion's three different FM algorithms give Tracking Generator is where I want to mimic the way guitarists will often bend a lot of scope for experimentation, but if you're new to the concept of synthesis by a note up and, at the highest pitch, add frequency modulation then it's probably best vibrato. The Tracking Generator uses to choose the simplest to start with. pitch-bend as an input, and applies that signal to the LFO depth. For most of the pitch-bend input value, the output equals zero. But with the highest input values, the pitch-bend wheel produces an output, thus increasing LFO depth.

I've also used the Tracking Generator to alter the 'feel' of an expression pedal and, oddly enough, to program a stepped modulation response. This was because I wanted to demonstrate a quantised control response during a seminar, but quantisation is not obvious with the Ion. So, I had to create a quantised control curve so people could hear what I was talking about! The process of programming the Tracking Generator can be tedious, but there are a couple of shortcuts. A fourth Tracking Generator parameter determines how many columns there are, so if you don't need too much resolution go for fewer; furthermore, if you double-click the Level parameter, it goes back to the default value for that column.

Are You Up To Date? Here's how to check the boot code revision (v1.00 at time of writing) and operating system code. With the synthesizer's power off, hold down the Panel Active A and D buttons together, and then turn on the power. If the OS code is earlier than v1.05, then go to the support pages at www.alesis.com and select Software Updates from the dropdown menu. Click on Ion Software and download the 1.05 firmware. After unzipping, you'll have a SysEx MIDI file you can load into a sequencer. After connecting the Ion's MIDI In to the sequencer's MIDI Out, play back the sequence to send the SysEx file. After a minute or two, all the data will have been transferred, and the Ion will reboot. If the transfer isn't successful, don't panic — redo the procedure, but slow down the MIDI transmission rate. You can also download the factory patch banks from the web site. If you want to transfer these to the Ion, first go to the Ion's fifth page of Global parameters, and turn off Program Protect.

Using The Ion As A Beat Box The fact that you can split and layer the keyboard may seem pretty useless, given that there are only eight voices — at least, that's what I thought at first. But file:///H|/SOS%2004-10/Alesis%20Ion%20User%20Tips.htm (5 of 10)9/26/2005 12:38:10 AM


actually, the multitimbral Setup patches can do some great beat-box effects (particularly because there are a lot of cool percussive/analogue drum sounds programmed into the Ion — look towards the top of the factory default Red bank). To get acquainted with this feature, hit the Recall Setup button and, assuming you have the factory Setup patches loaded, select Setup 08: Groove Soccer. Make sure there's no master transposition on the keyboard (both Octave button LEDs should be off). Select the Panel Active A button, play the lowest 'C' key, and you'll hear a nifty little techno groove with a tough kick sound (part A) and a percussive pattern (part B). These both result from arpeggiation. While you hold the low 'C', go to the Arp section in the upper left-hand corner and hit the Latch button. Now release your finger from the key; you'll hear just the techno kick, because it's been latched on. Next, select the Panel Active B button. The panel controls now affect the sound in Part B, which is the percussive rhythm. Play low 'C' again; latch this pattern, and you'll hear them both play together. Release your finger from the key. Select the Panel Active C button and play the next higher C key. You'll add a bass pattern to the mix. Latch it, and now try playing with Filter 1's cutoff control for some expressive variations. Tasty! Note that Mod Wheel 1 affects the percussive sound, while Mod Wheel 2 affects the bass. Now that a groove is going, play a melody line in the top two octaves. Go wild with the pitch bend, and if you want to modify the melody sound in real time hit the Panel Active D button. For the coup de grâce, how about a little breakbeat action? Use the Enable buttons to bring the latched patterns in and out, effectively adding solo and mute capabilities. By now you should be getting the point that an Ion setup can be a great groove generator. But there's even more beneath the surface; each part can have its own program, respond to its own MIDI channel, and respond to different MIDI note ranges. The pitch-bender, both mod wheels, the sustain pedal, and the expression pedal can be enabled individually for each part, and each part has adjustable output, balance (between main and aux outs), pan, and effects buss send. Parts can also be transposed in semitones and octaves.



Quick Tips The Ion's Held pitch-bend mode is great for patches like sitar, where you have notes with long release times that overlap other notes. When the P Wheel Mode parameter is set to Held, the pitch-bend wheel will only affect any notes actually being held down — any slow-release notes still sounding retain their pitch. You can find the P Wheel Mode parameter by hitting the Voice Edit button; it's on page one. Several envelope parameters have adjustable curves: +Exp (convex shape), -Exp (concave shape), or Linear. Having these three choices lets you model a variety of different envelope types. A +Exp attack and -Exp decay mimics the type of curve that happens when you charge a capacitor, the basis of analogue envelope generators. A -Exp attack produces a more dramatic rise, such as is often associated with backwards tape. A Linear slope, on the other hand, is more characteristic of early digital synths that had rate/level envelopes. The Ion's keyboard doesn't respond to channel pressure, which is an unfortunate by-product of trying to hit a price point. But all is not lost. Not only will the Ion respond to channel pressure data coming in from an external keyboard, it will respond to polyphonic aftertouch. If you're one of the lucky few with a keyboard that generates polyphonic aftertouch, try it with the Ion. Just go to the modulation matrix, and select Poly Aftertouch as a modulation source for the parameters you want to control. In the modulation matrix you can control the same parameter with two instances of the same modulator to increase its effect. This technique can also change the control law curve into more of a switching action. Take velocity, for example: assigning velocity control twice to the same synthesis parameter causes the whole parameter range to be mapped over the lower velocities — anything above a mid-range value will kick the parameter up to its full value. One of the elements that give analogue synths their character is imprecision — component values drift over time, thus causing small tuning changes that fatten the sound. The Ion lets you add an adjustable amount of drift, and the effect ranges from digitally drift-free to sounding like you just brought an analogue synth in from a cold car to a hot club! To access the drift percentage, hit the Voice Edit button and go to the third screen of parameters. This effect is not global, so different patches can have different amounts of drift.

The Ion As Signal Processor I doubt anyone bought their Ion to be a signal processor, but the fact that it can process external inputs is quite cool. You can plug a mic or electric guitar directly into the external inputs, but I'd recommend adding a preamp, as the input impedance is sufficiently low to load down pickups. One of my favourite effects is using the vocoder with guitar, triggered by a drum machine. To do this, first plug the guitar into the left-hand external input, and drums into the right-hand one. Now hit the Ext In button, go to page two, and set the External Input Level parameter to 100 percent.



Next, push the Effect Edit button, select the 40-band vocoder effect type, and set the following parameter values: Effect Mix: 100 percent wet. Analysis Sensitivity: 100 percent. Sibilance, Decay, and Band Shift: adjust to taste once you get the patch working. Synthesis Signal: Ext In Left. Analysis Signal: Ext In Right. Analysis Mix: zero percent. Finally, start the drum machine and play the guitar to get a rhythmic chopping effect. For more options, try assigning Decay and Band Shift control to your expression pedal. Another reason to use the Ion as a processor is that its Phaser effect is If you set the Sustain Time and Release Time of the Ion's amplifier envelope to Hold outstanding; I prefer it to any other (left), any note you press will continue to phase-shifter effect in my studio, and sound indefinitely — very useful if you want use it a lot with guitar. However, you to use the Ion as a processor for external need to keep the Amp envelope open audio. However, if you then wish to mute the or you won't hear any sound. Although envelope, you have to double-click the Home button (above). you could set sustain to its maximum value and put a weight on a key, there's actually a better way: hit the Env Edit button, and set the parameters as follows: Sustain Level: 100 percent. Sustain Time: Hold. Release Time: Hold. Keyboard Velocity Track: zero percent. Now if you hit any key, the amplitude envelope will open and stay open. Because the velocity tracking is set at zero percent, the level will be at maximum file:///H|/SOS%2004-10/Alesis%20Ion%20User%20Tips.htm (8 of 10)9/26/2005 12:38:10 AM


regardless of how hard you hit the key. If you want to control level with velocity, then increase this parameter to 100 percent. Each time you hit another key it adds another 'voice' with the effected sound, so be careful to hit the key once and leave it at that, because hitting more keys will add level, which could lead to distortion. To mute the envelope, double-click on the Home button. If you call up a new preset that uses effects with the external inputs, double-click Home as well to 'clear' the sound, then hit a keyboard key. A final trick is that you can use the Ion to create tempo-synced tremolo effects with guitar and voice. In the modulation matrix, assign an LFO to Program Level. Select the LFO you want to edit, hit its Edit button, then set its Sync parameter to Tempo, adjusting the rate as desired. Either feed in MIDI Clock messages to control the LFO frequency, or use the Ion's Tap button to set a tempo. (You may also need to edit the Global MIDI Sync parameter to Ext Sync.)

Recording Performances Into A Sequencer One of the great things about the Ion is that you can tweak controls in real time for more expressive playing. But what's even better is that you can record all these motions (not just those from continuous controllers) into a sequencer, which then controls the Ion on playback. Because the Ion sends out MIDI NRPN messages for recording, the resolution is better than what you would get if you were controlling the Ion solely with MIDI controller messages. Just remember that you need to enable the 'live front panel' mode to transmit NRPNs. To do this, hit the Global button and scroll over to page four, where you can set the Panel MIDI parameter to On. Also note that this also lets you control a second Ion from the front panel of a 'master' Ion.

The Most Important Tip Of All! One of the most significant aspects of the Ion is that it was made to be played. It's not a sample playback synth where you program a sound and just hit keys; those rubbery knobs beg to be tweaked. A few people have found it illogical that the Filter 2 controls are higher on the panel than the Filter 1 knobs, but the Filter 1 knobs are therefore closer to the keyboard and easier to grab in the heat of performance. Similarly, having the envelope controls right above the keyboard is great — many times I've played the decay parameter in real time to change a bass line into something more or less staccato. Those two mod wheels are also there for a reason, and so is the expression pedal input. In fact, if you have an Ion and don't have an expression pedal, you're doing yourself a disservice. Almost all the patches I've programmed have the pedal doing something, from changing envelope times to editing waveform symmetry. And, of course, when using the Ion as an effects device with guitar, the expression pedal makes a fine wah-wah controller for all those wonderful filter types.



The Ion does great things straight out of the box, but I hope these tips will help you get more out of this cost-effective piece of gear. For me, the Ion is one of those rare machines which has enough depth to keep you interested, but also a simple enough interface to let you access that depth without frying your brain. Now go twist some knobs! Published in SOS October 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



Bilocation: Binaural Recording & 5.1 Surround

In this article:


'This Is A Journey...' New Surroundings Portable Recording Published in SOS October 2004 Binaural & Transaural Surround Soundscape Print article : Close window Putting It Together Technique : Recording/Mixing Multi-channel Monitoring On The Cheap Musical Sounds DTS Coding New Developments Binaural recordings can

sound amazing on headphones, but don't work very well on conventional stereo speakers. Can they be adapted more successfully for surround systems? Steve Marshall

When 5.1 surround sound started to become a viable consumer format, I got very excited about the possibilities. I expected lots of amazing albums to appear that really explored this new medium. Instead, look what's happened: surround remixes of Led Zeppelin, Fleetwood Mac, the Beatles... Just as when CD first appeared, the record companies are The Saxon church in Wiltshire that provided concentrating on their back catalogues. a natural ambience for many of the I really wanted to hear something that recordings on Bilocation. wasn't just a remix of a standard music album, so I set about making the surround album that I wanted to hear. I wanted sound that came from different directions around me, but I also wanted to hear sounds coming from above. This was a feature of the Ambisonics system invented 30 years ago, and I eventually found that it's possible in 5.1 surround — without hanging speakers from the ceiling! I also discovered that it's possible to make recordings with only two microphones and convert them into 3D surround sound, and I used both techniques in the creation of my album Bilocation.

'This Is A Journey...' Bilocation is an album in 5.1 surround. If anything, it's 'about' the magic of

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acoustic spaces. It's made up of many different recordings that I made over a long period of time. Some of the recordings are musical, and some are records of places I've been. I'm very fond of the stereo test records of the '50s and '60s, with their steam trains and fireworks, and wanted my album to have more than a hint of that. When I bought my first portable DAT recorder over 10 years ago, I set about recording everything around me. I lived in East London at the time and intended to eventually move, so I recorded the traffic, the trains, the helicopters. I wanted to eventually listen to them in a quiet place in the country. Then when I was commissioned to compose music for Channel 4's Wild India I took a trip to India with my DAT machine, to research Indian music. I made an epic five-week trip around India by train and came back with hundreds of hours of recording. Some recordings were of music, but most were of the amazing sounds of everyday life in India — parrots in the street, chanting inside the temples, the noisiest traffic I've ever heard. I also recorded lots of Indian radio, which can be very bizarre! As soon as I got back from India I went to the huge annual fair in Hull and made lots more recordings — some of them actually on the rides. Eventually I did move to the country, and discovered a Saxon church with an amazing acoustic. The church is in the middle of a field and is hardly used, so it's practically empty and has a reverb better than anything electronic. It doesn't have electricity, though, and I had to work with batteries. I began playing back music and sounds in the church and rerecording them with my portable DAT. Snippets of Indian radio were played back in the church, under canal bridges and in empty farm buildings. The collection of sounds kept on growing.

Portable Recording My setup for recording whilst travelling has been the same for over 10 years, and it's never let me down yet. My portable DAT is a Casio DAR100. I chose this machine because it's reasonably small and light, and can be set to record at 44.1kHz. Replacement rechargeable batteries can still be found for it (direct from Casio). Its only disadvantage is that it's not Steve Marshall's portable recording setup, possible to use dry batteries in an based around a Casio DAT recorder. emergency. The mic amps, though, are very noisy, and not really usable for quiet recordings, so I use a custom-built high-quality preamp. This cost me about £80 and was well worth it. This means I can use the line inputs, which are very quiet. Minidisc recorders are smaller and lighter than DATs, but I took a brand new Sharp Minidisc to the Pyramids a couple of years ago, and it totally file:///H|/SOS%2004-10/Bilocation%20%20Binaural%20Recording%20&%205.1%20Surround.htm (2 of 10)9/26/2005 12:38:14 AM


failed to work when I got there, so I'll be sticking to DAT in future! I still use a pair of modified tie-clip electret mics bought from the now sadly defunct Tandy. I think they were only £30 for the pair. Electret mics like this are usually powered at 1.5 Volts, but this is just for the manufacturer's convenience — it's easy to fit a 1.5 Volt battery in a small box — and most electret mics can actually be powered at 9V, which dramatically improves their performance. They're usually omni-directional as well, which means a flat frequency response of at least 30Hz-20kHz. The tie-clip fitting is also useful when travelling. I always carry a tatty old canvas shoulder bag with my gear in it — it's as well to not attract thieves! When I want to record something, I just leave the DAT and preamp inside the bag and clip the mics onto the front of it, spaced at about six inches to a foot apart. This means I don't attract attention, and don't end up with recordings of people asking what I'm doing. It also damps the mechanical sound of the DAT recorder working. Sometimes I've rolled a towel up to something like a head shape, and clipped the mics where the ears should be. It's crude, but it works... My Tandy PZMs are also converted to run at 9 Volts. They are good, but a bit heavy when travelling and noisier. For a time I used them fixed to both faces of a ping-pong bat, with reasonable results.

Binaural & Transaural Many of my recordings are binaural — recorded with mics either mounted either where the ears should be on a dummy head, or on a human head. (Some purists would argue that this is 'pseudo-binaural' as they put the mics inside their ears!) However it's done, the results can be startlingly real when played back on headphones. Sounds can be located around, above and below the head. But when played on speakers, the effect is not so great. The problem is crosstalk from the speakers. With headphones, each ear can only hear the sound recorded on that side of the head, but speakers spread the sound, and spill the sound meant for the left ear to the right ear, and vice versa. This seriously muddies the three-dimensional imaging. One solution that's been around a long time is transaural processing. Take a standard pair of speakers, placed at the correct stereo angle of 30 degrees each side of centre. The output of each channel is fed out of phase to the other to cancel out the crosstalk. A small delay is also added to the out-of-phase component, to allow for the length of time it takes for the sound to pass from one side of the head to the other; if separate bands of frequencies are delayed by differing amounts, the effect is even more realistic. Much research is being done in this area, and it is widely reported on the Internet. The transaural technique has one serious disadvantage: the 3D imaging is only really effective in a tiny 'sweet spot', which can sometimes be only a few inches wide! What I realised (and it's pretty obvious) is that with a 5.1 system the 'sweet spot' can be made much bigger, by using two speakers on each side. If the front and rear speakers are fed the same signals on each side, the sweet spot can be several feet across. I set up a 5.1 mixer on my Soundscape R*Ed system, and bought a fairly cheap surround decoder and speakers. I made sure I got a decoder with six external inputs for the surround channels, and started file:///H|/SOS%2004-10/Bilocation%20%20Binaural%20Recording%20&%205.1%20Surround.htm (3 of 10)9/26/2005 12:38:14 AM


experimenting. My most spectacular binaural recording is of a late-night police car and helicopter chase in London. On headphones, the effect is amazing — the helicopter really appears to be above your head. I played the DAT recording into Soundscape, and copied the take. On the copy I swapped the left and right channels, and reversed the phase. I then experimented with delaying the out-of-phase, channelswapped copy. First I dragged the takes around on screen by small amounts, but I could hear an echo. Then I did some sums. Sound travels at about a foot per millisecond, so the delay across a human head should be around half a millisecond at the most. Soundscape has a Sample Delay module that can be dropped into the mixer channels. It allows a track to be delayed by any amount from 1 to 225 samples. At a sampling frequency of 44.1kHz, one sample is one 44100th of a second, so a millisecond is 44.1 samples. I found that delays of up to 20 samples were the most effective, but knew that in the higher frequencies there is more complex stuff going on. What our brains decode as directional information is derived from the varying audio spectra and delays produced when sounds come from different directions. The complexity of it increases with frequency. Many acoustics labs have developed intricate algorithms for dealing with this, but I didn't have access to them. So on the out-of-phase copy, I just EQ'ed the highest frequencies away with a low-pass filter. The result was like magic — a helicopter flying around my studio! I tried other recordings of crowds, spinning round on fairground rides, and so on, and they all produced spectacular results. The spinning sounds actually appeared to pan around the speakers.

Surround Soundscape I've used Soundscape for the last 10 years or so, but had never tried mixing for surround before. My system is a Soundscape R*Ed, with 32 audio tracks and 16 analogue inputs and outputs. I used six of the analogue outputs, and connected them to the phono inputs of my Sherwood decoder. I then made a mixer for surround. This is very easy in Soundscape: modules are simply selected and dropped into place. I built the mixer by starting with stereo input channels. Each stereo channel strip used aux sends for the different surround channels: a stereo send to the front pair (L and R), a mono send to the centre (C), another stereo send to the rears (SL and SR) and a mono send to the subwoofer channel (LFE).

Part of the Soundscape arrangement, showing one of the transaural sequences.

With some of the tracks, I wanted to have a little stereo image that file:///H|/SOS%2004-10/Bilocation%20%20Binaural%20Recording%20&%205.1%20Surround.htm (4 of 10)9/26/2005 12:38:14 AM


appears to sit a few feet in front of the centre speaker — a bit like a small stage. I used this for some of the material that was recorded from Indian radio and rerecorded in the church. To do this, I put the stereo source recording into two mono strips, each with only the 5.1 module in it. This appears on the channel strip as a small square that can be made bigger when clicked on. A dot representing the pan For some of the stereo sources a surround position can be moved around the panner was used, whilst aux sends were square, and it outputs to all used to route the transaurally processed channels apart from the subwoofer. stereo tracks to the various surround The module can be automated, as outputs. can every element of the mixer. One of my setups needed a 'V' formation between the rear channels and centre, so I dropped in four more mono strips that each had mono sends to all of the six channels. This way I could spread stereo tracks across any positions I wanted. The transaural processing was done by making two stereo mixer strips. The first was for the original binaural recording, and was initially set up to send equally to the front and rear channels. A copy was made of the binaural audio in the arrange page, and assigned to the next pair of tracks. The copy was then phase-inverted and channel-reversed. The mixer strip for the copied channel was also fed equally to the front and rear channels, and contained a Sample Delay module and a lowpass filter. The delay was adjusted according to what worked best for each piece of audio. Delay times varied between 9 and 224 samples (at 44.1 samples per millisecond). I made sure that six tracks of Soundscape were left available to mix the six busses down onto, and the last few strips of the mixer were for monitoring those channels. Each channel strip also ended in a fader that fed another pair of busses for making a stereo mix. The mixing was done mostly by setting levels within the arrange page — I never move faders. The only thing that needed adjustment throughout the mix was in the transaural channels. Some sounds needed a different delay, and the front-to-back balance varied too. So I used the Soundscape automation in snapshot mode at the start of each of these sections. When I was happy with the surround mix, I digitally bounced it onto the six empty tracks. This produced six Soundscape takes of exactly the same length, which were converted to WAVs for encoding.

Putting It Together I had come up with the idea for Bilocation some time ago, as a stereo project, and had experimented with putting my sounds together in different combinations. I had to be able to make sense of the vast amount of material I'd recorded. I started by editing my best sounds, compiling them onto 10 CDs, and I made detailed track sheets. This all took ages, but was worth the effort. One thing in particular had put the stereo version of Bilocation on hold: mixing different recordings made in different acoustics together sometimes produced a muddy and confused result. In surround, by contrast, many different sources can be



combined and still be heard separately. I found that the recordings made in the old church would fit with virtually anything, as they have such a distinctive acoustic character. Other sounds were difficult to combine, and some just sounded best on their own. The stereo version of Bilocation had started with a recording of surf in the Indian Ocean. I'd EQ'ed the top off it, making pink noise that still sounded like surf. The idea was to 'wash out' the listeners' ears, so they could then hear more clearly. Some recording engineers use this trick, and listen to pink noise on headphones before a mix. This didn't really work in the 5.1 version — I needed something more readily identifiable. So I tried rain A low-pass filter was used to remove the instead. Across the front left and right highest frequencies in the out-of-phase speakers I put a recording of rain in a signals. London back yard, with water gurgling down the drainpipes. Then I put a recording of rain hitting a window across the rear left and right speakers, and a different bit of rain into the fronts. The result was very filmic, and firmly established England as a starting location! I then took a binaural recording I'd made at dawn in a banana grove in South India, and did the transaural treatment on it. This filled the room with tropical sound — parrots, insects, and so on. When the rain was very slowly crossfaded into the banana grove the effect was magical; rather like in the children's book Where The Wild Things Are, a forest appears to grow in the room. It took me about three weeks of working every day to compile the 40-minute final version of Bilocation. I loved every minute of it! It was rather like editing a film. I'd find some sounds I liked from the 10 CDs, and just play with them until I found interesting combinations. I built up sections of a couple of minutes long, then found ways of joining them together. I really wanted 'magical' transitions between scenes, and this was achieved by a lot of finicky work with cross-fades.



Multi-channel Monitoring On The Cheap I bought my surround system very cheaply, in the Richer Sounds January sale. For £80 I got a Sherwood tuner/decoder, with six RCA phono inputs for the individual channels. The Pioneer speakers are five satellites that go down to 100Hz, and a large subwoofer. All are solidly built from metal, and cost me £70. My studio is very large, so I set up my speakers in a 14ft circle, positioning them using string. I then mounted them all at ear-height on mic stands. I put a hefty chair exactly on the sweet spot, and marked its position on the floor. When the mix was finished, I wanted some 'grot boxes' to check that the same effects could be obtained on a smaller, cheaper system such as a computer. I found a set of five speakers and a subwoofer in Safeway for £30! I set these up in a six-foot circle around my PC, and found that although they are over-bright and 'tinny', the surround and overhead effects are just as powerful. The main difference I found was that the angle of the rear speakers could be much greater. With this smaller circle an angle of 120 degrees was preferable. I suspect this has something to do with the angle of dispersion of the two different sets of speakers.

Musical Sounds In Bilocation the location sounds are often combined with musical sounds made in different acoustic spaces. I'd sometimes set up stereo ping-pong echoes on a synth sound or an E-bowed guitar, and place the speakers wide apart in a live space. This could be a concrete garage, an old church, under a brick arch — anything interesting. Then I'd play live and record it to DAT. One section of Bilocation has a beautiful pedal steel guitar played by a friend in my studio. I set up speakers with a ping-pong echo at opposite ends of the empty church and rerecorded it binaurally. I had also made a set of recordings in the church of myself 'overtone singing'. This is a way of singing two notes at once, by changing the shape of the mouth. It produces a drone with changing 'whistle' tones that sweep through the harmonics of the voice. I'd recorded many different notes in the church, each sung in different positions, and combined them in Soundscape into slowly evolving chords. As the voices are actually recorded inside the reverb, the effect is quite eerie. The overtone singing is used on two sections. One features a violent electrical storm in England, where I almost got killed! When the storm started, I'd grabbed my DAT and mics, and an umbrella, and set off to a quiet field to record. The thunder got louder and closer, and ended in a lightning strike in the field — about 20 feet from me! When transaurally processed, the thunder can be heard moving overhead, and the final strike is pretty scary. Another section is my homage to the stereo test records, where a huge steam train materialises in your living room! I had been on a overnight train in South India, when it stopped at a tiny country station. I started to record night insects out of the open carriage window. Just as I started to record, another steam train pulled slowly into the station, on the next track to mine, and stopped right in front of my mics. The passengers are chattering - one keeps spitting out of the window. Then the whistles blow, and the train slowly pulls away and dissolves



again. My favourite recording of all is of a wonderful building in India, the Golgumbaz in Bijapur. This is a huge square building topped by a 38-metre dome, the second biggest in the world. There is a 'whispering gallery' at the top, similar to St Paul's, but whispering is futile, as most people come in and The Golgumbaz in Bijapur, India, is claimed shout! The bottom of the building has to have the second largest dome in the an open arch on all four sides, and world, with a 'whispering gallery' that thousands of swallows go whizzing provided some amazing binaural recordings. though, twittering loudly. The sound at the top of the dome is amazing! The dome produces a clear echo of about half a second, which repeats — allegedly 12 times. The echoing shouts and screams of children combine with the swallow calls, and produce something very musical and beautiful. The recordings I made were binaural, and when transaurally processed the effect is just like being in the dome. This was then combined with some musical loops I made many years ago, binaurally recorded onto analogue tape. I had made a large cross by clamping two aluminium ladders together at 90 degrees, and suspended it from the ceiling in a large concrete room, like a carousel. I hung four speakers onto the ends of the ladders, and played back tape loops into each of them. I had several sets of loops that, when played together, formed chords. A dummy head was placed inside the circle of moving speakers. When the arrangement was spun around, the loops played arpeggios that had a Doppler shift and very strong spatial imaging. This works beautifully when combined with the echoing dome. Another favourite is the sound of a galloping horse — recorded from on the horse! In London I used to share a horse with a woman who became terminally ill and couldn't ride any more. I wanted to record the horse going round her favourite route in Epping Forest, so she could play it on a Walkman in hospital. This is tricky, as a galloping horse goes at about 25 miles an hour, and wind noise is a big problem. I took my Tandy PZM mics, and drilled a hole in each corner of the base plate. I bent some pieces of thick wire into arcs, and hot-glued them into the holes, so they spanned opposite corners and formed something like the frame of a mountain tent. Then I stretched two layers of nylon stocking over the top. I tested the mics with a car, and the wind noise didn't appear until the car got to 35 mph. Then I fixed the mics to my riding boots and set off with the DAT. The results were excellent, but as the mics were nearly a metre apart, hardly binaural. This was corrected by converting the recording to M&S and narrowing the stereo image. When transaurally processed, this sounds disturbingly like the real thing! The finale of Bilocation is the spectacular flying helicopter. Psychoacoustics is a very powerful thing. When people listen to the helicopter recording cold, they can all hear it flying above their head. But after listening to 40 minutes of surround sound with differing acoustics, the ears really tune in to the spatial information,



which becomes more focussed. When the helicopter is heard at the end of the album, you can practically see it moving around above you! Bilocation costs £13.99 (plus £1 p&p for orders from outside the UK) and is available from my web site at www.bilocation.co.uk, or by post from: Bilocation, PO Box 2700, Devizes, Wilts, SN10 3ZU, UK. The site also includes further information on transaural processing, binaural recording, setting up a 5.1 system and surround sound in general.

DTS Coding Making a satisfactory 5.1 mix in the studio could be seen as merely an intellectual exercise — rather pointless unless it can be made available for others to hear. So it must be coded somehow. I had read about a piece of software called Soft Encode for combining six mono WAVs into one Dolby Digital bitstream, but it is no longer available. I eventually found someone who had the software, and made a 5.1 coded file. It sounded terrible! The top frequencies suffered very badly, and lots of the spatial information that depended on short delays had simply vanished. I was gutted. The fact is that coding involves some hefty data compression (around 10:1) and Soft Encode just wasn't sophisticated enough for what I needed. After ringing around, I was advised to go the DTS route. Not all surround systems in the world are DTScapable, but apparently 200 million of them are... DTS coding uses a very complex algorithm to compress the data and interleave the six channels. Luckily, a friend had just bought the latest software from the Minnetonka company, and coded it for me. I supplied the six WAVs on two CD-Rs, and the process simply involved loading them into his machine and pressing go. I was amazed at the result — it sounded exactly as I'd monitored it originally, yet it now fitted onto one CD-R. I have since been told that equally good results can be had with Dolby Digital, but more tweaking is needed to get the best from it. www.dtsonline.com www.minnetonkaaudio.com

New Developments Binaural recording with microphones has something of Heath Robinson about it, but it can be a powerful tool. Even more powerful is synthesized binaural sound. The Head Related Transfer Function is what is behind transaural processing, and it's complex. It takes about half a millisecond for sound to pass from one side of a human head to the other — but not all sounds come from the same direction. Sounds reaching the head from different angles and elevations have complex peaks and troughs in their audio spectra, and these again are time-related. This mass of information is decoded by the brain to construct a 3D sound picture. If this process can be synthesized, then false 3D sound images can be created. This has huge implications for virtual reality, gaming, and so on. A great deal of research is being done in calculating HRTFs, but although I found lots of information on the Internet about them, none of it was comprehensive. The fact is, HRTFs are potentially big business. The data is very difficult to collect, and too file:///H|/SOS%2004-10/Bilocation%20%20Binaural%20Recording%20&%205.1%20Surround.htm (9 of 10)9/26/2005 12:38:14 AM


valuable to give away. I emailed several universities who had posted HRTF information, asking for more. Not one replied... There is some information available though, and it's worth looking. The HRTF data can of course be programmed into a DSP. So would anyone out there like to have a crack at making a transaural plug-in, in VST, and release it as freeware? Please. An excellent page of links for 3D audio sites can be found at www.wareing.dircon. co.uk/3daudio.htm. Published in SOS October 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:


A New Life In A New Town Artist: David Bowie; Producers: A Social Event Making Do Studio: Hansa Ton, Berlin Cracking The Combination Published in SOS October 2004 No Going Back Print article : Close window Searching For Inspiration The Fabric Of Time Remains Technique : Recording/Mixing Intact Kick And Drag Accidental Muses

David Bowie, Tony Visconti;

With 'Heroes', David Bowie pulled off the rare feat of having a major hit with a highly experimental piece of art-rock, which featured among other highlights live synth treatments from Brian Eno, pitched feedback from Robert Fripp and a lead vocal with leveltriggered ambience. Richard Buskin

It was after working with producer Denny Cordell on a Georgie Fame session that engineer, producer and guitarist Tony Visconti relocated to London in 1968. Soon after arriving, he produced Tyrannosaurus Rex's My People Were Fair... and Prophets, Seers And Sages albums, followed by David Bowie's 'Space Oddity' in 1969. The work with Marc Bolan continued through classic records such as Electric Warrior and The Slider, while that with Bowie took in the likes of The Man Who Sold The World, Diamond Dogs, Young Americans, and the 'Berlin trilogy' of Low, Heroes and Lodger (although Low was actually recorded in France). For a short time early on, Bowie and Visconti had shared a flat in Beckenham, Kent, with girlfriends and band members, yet by the late '70s their friendship had become, to quote the producer, "more pragmatic than idealistic", while their working relationship continued to flourish. "David's way of thinking is extremely different from most human beings, while mine is very practical," Visconti remarks. "So, we got on really great in the studio."

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Nevertheless, after 1980's Scary Monsters (And Super Creeps), Bowie and Visconti didn't collaborate on another major new album project for more than two decades. They ended their hiatus with Heathen in 2002, and joined forces again for Bowie's most recent album Reality, as described in SOS October 2003 (read the article on-line at www.soundonsound.com/sos/oct03/articles/reality.htm).

A New Life In A New Town 1977 was a busy year for David Bowie. He produced Iggy Pop's records The Idiot and Lust For Life, toured anonymously as Iggy's keyboard player, made a film appearance alongside Marlene Dietrich and Kim Novak in Just A Gigolo, and narrated Eugene Ormandy's version of Peter And The Wolf. After relocating to Berlin following the release of his avant-pop opus Station To Station the previous year, he began straightening out from a coke addiction, studying art and immersing himself in Euro-expressionist, synth-based music. Accordingly, early 1977 saw the release of Low, Bowie's experimental and highly influential electronic album which, courtesy of his collaboration with Brian Eno — an acquaintance from Bowie's Ziggy Stardust days, when Roxy Music were his opening act — fused mainstream pop with the avant garde, and he quickly followed this up with the equally groundbreaking Heroes. Again utilising the talents of Eno, this record echoed Low's semi-vocal/semi-instrumental structure while boasting a more heavily layered, harder-edged sound, thanks in part to the guitar contributions of Robert Fripp. In turn, the hauntingly atmospheric title track became an international hit.

A Social Event "Working with Bowie is much more than going to a studio," asserts Tony Visconti. "It's a social event, too. We would eat together, go to shows together, go to clubs together, and really soak in the local culture. That was always his way of working, and Berlin was perfect for him in terms of what he wanted at that time. It was a stark, scary place, yet it had a very exciting nightlife, with exotic locales such as the Turkish quarter, and it was swarming with artists like Tangerine Dream, who were friends of ours. David was writing with Brian Eno back then, and the three of us got on really great. "Since Station To Station, David had been working with a rhythm section consisting of Carlos Alomar on guitar, George Murray on bass and Dennis Davis on drums. All three were amazing musicians. You'd just throw a few chord changes at them and they'd run with it. Carlos, especially, could whip up these little instant licks that would gel the whole thing together — he's a rhythm guitarist extraordinaire, and his lead playing ain't so bad either. Bowie and he would bounce off each other brilliantly — Carlos might come up with the germ of a part, and then Bowie would help him elaborate, but once the two of them began exchanging ideas back and forth, you'd get amazing stuff.



"For instance, the underlying riff on 'Heroes' was Carlos's idea, as was the pre-chorus part, which is like a viola and cello section, whereas Fripp overdubbed the high, wailing parts. The point is, David's modus operandi would be to throw a bunch of chord changes and a bunch of ideas in a very loose structure at the band, and he knew he could rely on those guys to immediately do something. They were jam experts, and so within half an hour they would jam the few chords that David threw at them into a wonderful structure. The lyrics would often come about a month or two later. We would work on the musical content, David would have some idea as to what the song was about, and we would use that idea — like if it was going to be a happy song or a depressing song — to make the instruments come out with an emphatic arrangement or sound in order to invoke the desired emotion. Then the stage would be set and David would just throw his lyrics on at the very last minute. He would write his lyrics in a morning, it would take him an hour or two, but beforehand he'd also need a month or two to let the ideas really germinate." Such was the case with 'Heroes'. Before recording commenced on the album, Bowie and Eno spent a couple of weeks working out some basic song structures, but again there were no lyrics and no melodies. One of the stronger structures was that which would evolve into 'Heroes', yet whereas the finished version clocks in at just over six minutes, the track was actually about eight minutes long when it was recorded.

Photos courtesy of Tony Visconti Tony Visconti "in a house painter's hat decorated with Sharpies by my 11-year-old daughter, Lara. It was taken in studio B of Looking Glass during the making of the Bowie album Reality. David refused to wear the hat."

"We just kept going round and round with these very long cycles of verse, two verses and then pre-chorus and chorus," Visconti recalls. "Then, about four minutes down the line, a bridge kicks in. When you record in this fashion it's inevitable that you bring out the razor blade, and if you looked at a lot of the Heroes multitrack tapes you'd see loads of edits. Most of the tracks on that album had to be cut down as we would just over-record, but sometimes it would be useful to do that because we'd find, for instance, that there was a better chorus at the end. I would make a 24-track to 24-track copy and edit that chorus earlier because it was played better or whatever. This was before Pro Tools! And it was dangerous living, because you couldn't do too many edits on the same point without the tape starting to curl up or the backing coming off. You had a maximum of, say, two edits that you could do and undo in the same area, but we firmly believed that if you didn't do it, it wouldn't be worth keeping the track anyway. So, living dangerously wasn't that file:///H|/SOS%2004-10/CLASSIC%20TRACKS%20%20Heroes.htm (3 of 10)9/26/2005 12:38:18 AM


dangerous really."

Making Do Having completed his contribution after about two and a half weeks into recording of the Heroes album (representing the first third of the project), Brian Eno departed, and at that point it was a case of 'What shall we do next?' for the coproducers. Among the answers was a sound resembling a Stax horn part that surfaces at the start of the title track's second verse. As it happens, this was a brass patch out of a Chamberlin, successor to the fabled Mellotron, which Bowie had first employed on the Diamond Dogs album and which now provided a cheap and instant — if less desirable — alternative to real-life sax and trumpet players. "It was definitely written as a trumpet part, but it sounds more like a weedy little violin patch," Visconti admits. "Still, we liked it in the end. We just said 'Oh, that'll do,' because it sounded weird." In line with this way of thinking, when the two men wanted to add a cowbell and didn't have one immediately to hand, they sufficed with an empty tape reel of the German variety; a metal plate on which the tape basically sits. The echoey result of David Bowie and Tony Visconti alternately bending it out of shape with a drumstick was achieved not using artificial reverb, but by miking it in the large room at Hansa. "It was really painful to hold it, because it's not ergonomic like a cowbell," Visconti recalls. "When I have a big room, I use that room's sound, and we had an 87 in there that picked up all the ambience — the 87 is my Swiss Army Knife microphone, I use it for everything. Sometimes, I'd try to use two mics, but again it was a sheer luxury to have two tracks for ambience. If I mixed that record in 5.1 today I would certainly have that ambient mic in the back speakers, conveying a sense of the room's size which stereo can't reveal. Anyway, we played the drum reel, as well as a tambourine that was the other percussion instrument, about 15 feet from the mic to ensure we got all that room sound. It's like a gift to be in a room like that, and I would not only use the ambience for percussion but also throw a guitar out there and have a close mic on the amp. I suspect I may have used that sound during the mix, too, because the room served as a real echo chamber. You just throw the sound out there via the board — you take an auxiliary send and send it through a speaker in the room, and then have a microphone at the other end of the room picking up the sound which is brought back into the board as an effect."

Cracking The Combination One of Bowie's most hypnotic recordings, 'Heroes' draws in the listener by way of a multi-layered rhythm, while the vocalist builds from a low croon to nearhysteria. The aforementioned live trio of Alomar, Murray and Davis was augmented by Bowie on piano, the band remaining in Berlin for about 10 days as per the norm for his projects. "David likes to do these backing tracks, he gets very enthusiastic about them," Visconti explains, "but we send the band away very quickly and maybe keep a file:///H|/SOS%2004-10/CLASSIC%20TRACKS%20%20Heroes.htm (4 of 10)9/26/2005 12:38:18 AM


person like Carlos for an extra day or two so that we can double-track some of his parts. With 'Heroes', on the other hand, we built the track over the course of an entire week of careful overdubbing. For instance, Brian brought his EMS Synthi with him, which is a synthesizer built in a briefcase, and it has no real keyboard — it's got a kind of flat, plastic keyboard which Brian very rarely used. He used the joystick a lot, and the oscillator banks, and he would do live dialling — they look like combination-safe rotary knobs on the three oscillator banks. Brian goes down on record as saying that he's a non-musician — he even tried unsuccessfully to have that listed as his occupation on his British passport — and, like David, he thinks very radically and from a completely different space. "So, after recording the live rhythm section, everyone went home and a week later we came back to this track that was tentatively called 'Heroes', and Brian took out the EMS Synthi and got this shuddering, chattering effect by using oscillator 1 at a very, very low frequency rate — probably five cycles per second — and working the noise filter. He would slowly change the speed or change the intensity with Bowie, Visconti and assistant engineer, Edu other knobs, and he did that in a Meyer, taken in the control room of Hansa Studios. Photo by Edu Meyer's wife, couple of passes of the tape, which by Barbara. now had been edited down to just over six minutes. If you listen to the track now, this shuddering, chattering effect slowly builds up and gets more and more obvious towards the end, and that kind of set the mood. "Then I'd say the next thing that really moved the track along was Fripp's contribution. We already had Carlos's beautiful lines, like the bass line that was doubled on the guitar as well as the melodic part on the pre-chorus, and when Fripp came along about a week later he added a whole other dimension. He and Eno had already enjoyed a long partnership where Fripp would plug his guitar into the EMS Synthi and Brian would just play around with it, so Fripp did exactly that and he came up with that beautiful line which everyone thinks is an E-bow sound, but which is actually just Fripp standing in the right place with his volume up at the right level and getting feedback. "Everyone who's played the song with Bowie since then has had to use an E-bow to duplicate it, but Fripp had a technique in those days where he measured the distance between the guitar and the speaker where each note would feed back. For instance, an 'A' would feed back maybe at about four feet from the speaker, whereas a 'G' would feed back maybe three and a half feet from it. He had a strip that they would place on the floor, and when he was playing the note 'F' sharp he would stand on the strip's 'F' sharp point and 'F' sharp would feed back better. He really worked this out to a fine science, and we were playing this at a terrific level in the studio, too. It was very, very loud, and all the while he was playing these notes — that beautiful overhead line — Eno was turning the dials and creating a



new envelope and just playing with the filter bank. We did three takes of that, and although one take would sound very patchy, three takes had all of these filter changes and feedback blending into that very smooth, haunting, overlaying melody which you hear."

No Going Back Positioned behind the Neve console in Hansa's Studio Two, Tony Visconti was taking care of the recording while also throwing in his two cents' worth of production advice regarding the evolving arrangement and whether or not things were in tune. With the levels fluctuating wildly, he had to employ heavy compression as part of his chain of effects, while the 24-track medium necessitated a considerable amount of bouncing down. "When you're doing filter sweeps on an oscillator, especially in the mid-range, The layout of Hansa Studios for the Heroes sessions. In Visconti's own you'll suddenly have a bump of maybe words: "The studio was large, symphony 15dB," he explains. "Then, when you're size. The choir riser was about a metre boosting the lower frequencies, depending and a half higher than the studio floor, on what your EQ is, the sound might go so George Murray could hear and feel away, so all the while I was putting the kick drum in his face, without headphones. The bass and guitar amps everything through my processing gear were covered with gobos, as was the just to get it on tape. At the same time, grand piano. No gobos were used for given what was on that track, I had to do the drums and an ambience mic was some bouncing. For instance, even though placed at the far end of the studio, there were three Fripp tracks, I had to opposite the drums." bounce them to two because in those days you always knew that down the line you were going to need more tracks. The same went for drums. Back then I'd record them over eight or nine tracks, but then I would do a submix down to six, putting all of the toms to two tracks. So, that's what I was doing the whole time, knowing that this song was going to get bigger and bigger, and trying to figure out how I'd record this stuff while imagining how it would sound in the mix a couple of weeks later. I'd be sitting there and thinking 'I hope I've got these levels right, and I hope this guitar is balanced correctly to this guitar,' because once I'd bounced them down they were locked." In line with these concerns, it was natural for Visconti to basically mix as he went along, making decisions on the spot with regard to level changes that would theareafter be irrevocable. "It was fun, and Bowie loved to work that way," he remarks. "He'd say 'Once I've made up my mind, I don't want to change it.'"



Searching For Inspiration Between Bowie, Visconti, Eno and Fripp, everything was done by committee, with each person throwing in suggestions that might contribute to the final product. It was the producer/engineer's idea, for example, to triple-track the guitars in order to smooth out a sound that was patchy on the first pass. "When the song was first recorded, we had no idea what a verse and chorus were," says Visconti. "If you listened to it, the verse could have easily been a chorus, because it had a hook to it, but that was slowly evolving as we were overdubbing these instruments. Those guys are real artists. They're truly imaginative throughout the whole process, and nothing is done for the sake of it. You know, with Bowie you don't double-track a guitar because you have to. It has to have a meaning for him in order to do that, and quite often he likes things to remain single-tracked. I mean, the Beatles didn't double-track everything — sometimes they had one guitar, but it would be at the forefront of the mix, providing a very important, very solid sound, and double-tracking or triple-tracking could have weakened that. However, in the case of Fripp, we wanted this dreamy, floaty effect, so triple-tracking had a purpose. And of course it was inconvenient, because I only had 24 tracks to play with. "Sometimes in those days, if we had something ridiculous like 16 voices and we only had two tracks left, we would do a slave mix on another piece of tape, mix the 16 voices to a two-track tape, and then fly that in by chance to the spot where it belonged. I'd achieve this by having my assistant — in this case an invaluable German engineer by the name of Edu Meyer — start the multitrack while I'd make little marks on the two-track tape and just hit the Play button, hoping this might be it. And if this was not it, then I'd move my mark on the tape and try again. No one had yet devised the system of sync'ing up two multitrack machines.

Visconti and Bowie during the making of the Heathen album at Allaire Studios in Shokan, NY. Photo taken by assistant engineer, Brandon Mason.

"Tape was often rolling all the time. We would have a two-track tape running at 7.5ips, and it came in handy many times when searching for that moment of inspiration. You'd go back an hour and there it was! Beforehand, we'd be talking about it, saying 'Gosh, I'm sure you were onto something earlier,' because we'd get lost. We'd start with an idea and then we might go in the wrong direction, and after an hour we'd say 'How did this start out again? It sounded good an hour ago. Why are we not getting that?' So, we'd often have that tape running, and if we went back and heard something we really liked, it could be duplicated. Of course, we couldn't lift it off the 7.5ips tape — it was in the mix, but at least the musician could replay it."



The Fabric Of Time Remains Intact For all the sonic experimentation on display, very few effects were employed during the recording, not even the Eventide Harmonizer which Visconti had applied to live drums on Low after memorably informing Bowie and Eno that "it fucks with the fabric of time". Visconti only used it when mixing some of the Heroes album, and not the title track itself. "It was enough to hear the sound of that large room on the drum kit," he explains. "The studio, which had been used to record symphonies during World War II, could accommodate about 150 musicians, and there was a stage which was probably a riser for the choir. It was at one end of the room, and we used that for Dennis Davis's drum kit. In those days, we'd usually put a drummer in a booth, but Dennis took advantage of all the space, and besides his tom-toms he sometimes had an extra snare, a set of conga drums and a single timpani. When that was the case, he'd often go around his kit, and as he was doing a tom fill he'd also play a few beats on the conga drum and a few beats on the timpani. So, although those fills may sound like overdubs, that's actually Dennis Davis playing live. He's a wacko guy, one of the best drummers I've ever worked with. 'Heroes' wasn't played to a click track, but its tempo is virtually the same throughout the entire six minutes, and that's thanks to Dennis. He's not only an innovative drummer but a human metronome, and he's also a jazz guy who never plays the same thing twice. Some of his fills were priceless — on the song 'Blackout', you'll hear a lot of those fills going right around the kit, from the toms to the conga drums, whereas on 'Heroes' he was a little more sedate."

Kick And Drag One of the unusual features of the final mix on 'Heroes' is that the kick drum is almost inaudible. This, as Tony Visconti explains, was entirely deliberate: "You see, the track seemed to plod, and at the mix, I realised that when we played it with the kick drum lower it had more of an energetic feel to it. Usually, the kick drum is to the fore in the mix and equally loud as the snare, but we discovered that, by pulling back the kick drum and emphasising the bass guitar, the song had more of a flow and a better forward energy, while the shuddering effect [created by Eno's synth playing] was another tempo element that helped it move along. So, this was one of the very tracks that I've ever mixed with almost no kick drum in there, and that was entirely intentional because it made all the difference in the world."

Accidental Muses Less than sedate, when it came time to write the lyrics, was Bowie. "He gets very, very tense," Visconti confirms, "because he's now got to commit. So, I could feel it in the air."



And the producer/engineer soon got a breath of fresh air when the main man asked him to take a break in the middle of the day. Happy to oblige, Visconti went for a walk by the adjacent Berlin Wall with backing singer Antonia Maass, and this couple then unwittingly aided the songwriting process by indulging in what they thought was a spot of covert smooching. "David could see us, and we quickly got written into the lyrics as the lovers who kissed by the wall," Visconti admits. "He wrote the entire lyrics looking out through the windows of Hansa Studios, and when I returned after a couple of hours and asked him how it was going, he said 'Oh, I've finished.' His assistant, Coco Schwab, then took me aside and said 'I think you and Antonia are in the song.' I was married at the time, so this story was never allowed to be made public, but I don't mind now. "Anyway, David then told me he was ready to record his vocal, and after we did a couple of run-throughs during which he wasn't sure where to place the octave, we eventually came to the conclusion to sing the first two verses down an octave and the rest of the song up an octave. That was another good way of building up the track, and it prompted the break in his voice which he himself calls 'Bowie histrionics', where he has to put everything into it in order to hit those high notes. It's right at the end of his range.

Photo: Ian Dickson/Redferns David Bowie on tour in 1977, playing keyboards in Iggy Pop's backing band.

"My input at this point was to suggest using the room on his voice and drop the conventional method of just singing into one microphone. He agreed, and so I set up three microphones. We only had two or three tracks left, and I needed one of these for backing vocals. I couldn't even bounce down, and so we'd snookered ourselves. Therefore, even though I would have ideally loved to put each mic on a separate track — enabling us to capture the whole room when he sang loud, and just that one mic right in front of his face when he sang quietly — I put gates on mics two and three. Mic number one was in front of him with fairly heavy compression, because I knew beforehand that he was really going to shout, and it all went down to one track. This was recording by the seat of your pants, and Bowie was thrilled with the idea that I wanted to do something unique. He thrives on anything that's different and someone else hasn't thought of yet, and I just thought 'Let's do this live,' because he's a great singer and he could always sing it again if I made a mistake. That's the luxury of working with him: he's consistently good when he sings. He's in tune, he's passionate, and he delivers an arena-type performance every time. "Mic number one was a valve U47, and with the other two on gates I made sure that number two, an 87 placed about 15 feet away from him, would go on at a certain level, while the third mic, another 87 that was all the way at the other end of the room, didn't open up until he really sang loud. That reverb on his voice is file:///H|/SOS%2004-10/CLASSIC%20TRACKS%20%20Heroes.htm (9 of 10)9/26/2005 12:38:18 AM


therefore the room itself, none of it is artificial, and it's his voice triggering the gates. What is really great is that the sound of the opening two verses is really intimate. It doesn't sound like a big room yet, it sounds like somebody just singing about a foot away from your ear. The whole idea worked, and what you hear on the record is probably take three. We wouldn't go beyond that. He was really worked up by then and I can tell you he was feeling it. It was quite an emotional song for him to sing, he deliberated long and hard over these lyrics, and he was ready to go, there was no holding him back. We probably punched in a few things, but it's pretty much a complete take save for a couple of notes that he redid. "This was immediately after he'd written the lyrics, and immediately after this he said 'Come on in, let's do backing vocals.' You see, I'm his utility person — if there's a guitar part that needs to be played and there's no guitarist in the studio, I'll play it, and the same goes for bass guitar, keyboards and singing. So, Bowie and I performed the two tracks of backing vocals on that song, meaning that writing the lyrics, singing the lead vocal and then the backing vocals was all done within the space of about five hours. That doesn't always happen, and since then I've regretted telling this story to other groups I've worked with who think they can do the same thing. Very few people can write the lyrics on the spot in the studio and then perform a great vocal in just a few takes. Bowie's one of the few people on this planet who can actually pull that off." Published in SOS October 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Demo Doctor

In this article:

Demo Doctor

Andy Gunn Doctor's Advice: Feel Factor Analysis of Reader Recordings Published in SOS October 2004 Kevlar QUICKIES Print article : Close window How To Submit Your Demo

Technique : Recording/Mixing

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

Andy Gunn Venue: Home. Equipment: PC running Steinberg Nuendo DAW, Universal Audio UAD1 DSP card, Fostex VC8 A-D converter, RME 9636 soundcard, TC Electronic Finalizer mastering processor, Behringer 24-channel desk, Avalon 737SP mono channel strip, AKG Solidtube microphone. Track 1 Although this is Andy's first self-produced effort he is 1.4Mb no stranger to recording and has had record deals in Track 2 the past, notably with Virgin subsidiary Point Blank. 1.4mb This label deals primarily with roots and blues music Track 3 and it seems Andy didn't quite fit the mould so he's 1.4Mb had to strike out on his own. Since he's a guitarist, you'd expect the guitar sounds to be pretty good and they do not disappoint. From the classic blues sound of the second track, 'Singing My Song', to the more subtle electric guitar textures and acoustic guitar picking elsewhere, Andy demonstrates that he can handle recording both as a producer and performer. However, he is modest enough to acknowledge help from a couple of engineering friends who gave technical support with mixing and drum recording, so it's not quite a solo effort.

Even so, there are some slip-ups on the mix, betraying a touch of inexperience. One example is the use of overdrive on the vocals on 'Singing My Song' — it's a thickish sound that has taken much of the clarity away. I think less distortion would be better but it's also worth considering using a bit of creative EQ to make the vocal clearer. To maintain a sense aggression without losing clarity, use some lower mid-range cut and even a bit of upper-mid boost, and back off the overdrive a touch. I also noticed that on the same track the organ pad sound is mixed very low. Although this is probably a conscious mix decision designed to

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Demo Doctor

highlight the guitar, the organ is the perfect instrument to fill the hole in the mix between guitar and bass and could be louder without obliterating the guitar. Andy has splashed out on an Avalon 737SP channel strip which he uses with an AKG Solidtube microphone. I'm a big fan of using a quality front end for any recording system so I'd say it was money well spent. Most of the sounds (excluding the drum kit) were recorded using this setup and the vocals and acoustic guitar have benefitted mo. Listening to track four, 'Mermaids', I was impressed with the guitar arrangement, especially the use of a double-tracked picked guitar with some sparse chords hovering in the background. The vocal of track three, 'Holdin' The Storm', has also been double-tracked, but here I find the decision to lift the double-track in level for verse two a bit odd as it makes the vocal sound a bit loose. Apart from the occasional small oddity in the mix this is a very good album, accessible without being too commercial. As well as good sounds, this recording also has that indefinable 'feel' factor without which good sounds mean nothing. www.andygunn.com

Doctor's Advice: Feel Factor How do you make a recording that has a great feel? Creating the right environment and atmosphere for the performers to do their stuff is a big part of it, and is usually considered to be part of the job of the producer. But what happens when you are producing yourself? For bands, it's about getting everyone in the right mood at the same time and keeping them there — no wonder so many split up! Solo artists with no-one to bounce ideas off have a much harder job and have to think themselves into the right frame of mind. Technical factors like a good foldback setup and monitor mix will obviously help inspire a good performance — if you don't like what you're hearing you're not going to play well. But in the end, you have to be prepared to work for it. If things aren't going your way, be patient, take a deep breath and try again.

Kevlar Venue: Home. Equipment: Mac running Digidesign Pro Tools Free and Felt Tip Sound Studio audio editors, Seck mixer, Zoom RFX1000 multieffects, Genexxa 10-band stereo graphic EQ, Meridian active monitors.


Demo Doctor

Phil Melin specialised in music technology for his Track 1 degree in Contemporary Music and since finishing 1.4Mb college he has been pushing his music via his own independent record label. He describes it as dark, Track 2 atmospheric electronica and it tends to focus on the 1.4mb seamy underbelly of society for its lyrical inspiration. Track 3 'CCTV', for example, is more voyeuristic than 1.4Mb political, while 'Overdrive' tackles another contemporary theme, that of addictive behaviour, inevitably ending on a depressing note. In fact there is little solace to be found on the CD, which ends so abruptly I can almost hear the sound of a needle being dragged sharply across a vinyl record! The music keeps up the aural bombardment for the most part and is relentless and busy. For example, the forceful pounding of the bass drum on the first track is cleverly layered with a bouncing synthesized throb which takes no prisoners. The stereo percussion tracks, using classic electronic hi-hat and snare sounds, continually pester you for attention and are embellished by synth noises and effects. You'd think that there wouldn't be any room for a vocal, but Phil's almost spoken delivery is given its own space in the mix by the use of a heavy, retrostyle reverb. A vocal EQ which softens the sound also helps it to sit well against the harsher tones of synthesizer and percussion. Thankfully, he steers clear of using overdrive on the voice on the track called 'Overdrive'! 'CCTV' however, does use such an effect, which could actually be improved with a mid-range cut to give it more clarity and even a telephonic tone. Some light and shade is introduced to the arrangement as the mix drops to the vocal alone for the line 'Shut me down', and I also found the short breakdown to a string pad very effective. This runs into an electronic drill sound created using a highly resonant filter sweep. In combination with other noises like gasps and halfaudible vocals in the background, it creates a disturbing sound collage over which the sneering vocal slithers. Phil doesn't say what microphone he's using but with such heavy effects on the voice (overdrive and delay with perhaps a touch of modulation) he could use a budget dynamic and get away with it. The mixes are well handled considering how busy they are and the bass end of the mix is powerful without being overbearing. Personally I'd be tempted to allow a little light into this dark sonic landscape occasionally — the use of retro string pads is a good idea which should be built on. The steely sound of a Mellotron sample is another option which might be worth investigating. The bass lines and sounds seem to be continually pulling in a drum & bass direction, so a few breaks into that genre wouldn't be that out of place either.


Demo Doctor

QUICKIES

Trevor Jones Trevor has overcompressed the opening track, causing the whole mix to dip audibly every time the kick drum is hit. The drum pattern used is a looped sample and doesn't really need much in the way of compression to stay consistent and punchy in this mix. If the rest of the backing requires compression, I'd suggest submixing everything which needs heavy dynamic treatment to a stereo buss and applying compression just to that. The drum loop and vocals are then free to be given a different treatment, or none at all. Remember to route the submixed channel signals to the main left/right buss and not the source channel signals, otherwise you'll get phase problems and the punchy effect of the compression will be weakened. Trevor has got a good, traditional rock voice but needs to elimate the popping on the vocal track at source by using a good pop shield or better microphone technique. The vocal could also be mixed louder with a little less reverb too. www.trevorjones.com

Da Capo The picture of Da Capo's studio supplied with the CD shows a room even more cluttered than my place! I don't think they're going to get any problems with highfrequency reflections, given the number of books in there acting as diffusors. Despite the homely, relaxed-looking studio, Da Capo acknowledge that they are having trouble with their recordings. It seems that people like the songs when they are played live, but find the recorded versions less satisfactory. One of the problems is that the performances aren't tight enough and this is a direct result of the recording method: if the drums must be recorded after the acoustic guitar and vocals, then you have to make damn sure that the guitar take is rhythmically very solid. Using a click track is one possibility, but if this is difficult for the guitarist to play to then I would suggest trying to record acoustic guitar, bass and drums together. Since, as Da Capo say, the drums are recorded in another room anyway and the bass can be DI'd, it would only require two or three different foldback mixes. This approach might yield a recording closer to the band's live sound.

Simon Robinson Simon's classically orientated prog rock could almost be a contemporary version of ELP. The standard of musicianship is staggeringly high, yet Simon occasionally has a tendency to place the organ to the fore of the mixes, sometimes at the expense of the drums. These have been sympathetically programmed and sound excellent but I can understand why Simon would have preferred to record with a real drummer and bass player. However, finding musicians good enough to tackle such technically demanding material would be difficult! While a real bass player in particular would perhaps have a more fluid style of playing, Simon's programming is more than good enough to get his musical ideas across. The organ sounds, provided by a Korg CX3, are terrific, though the textured synth tracks do sound a little old fashioned. I was particularly impressed by the compositional skill shown in the fourth track, 'Truth And Reconciliation', which made effective use of the Roland JP8000's sounds.

Urge


Demo Doctor

This band have a well equipped studio in North Wales which has obviously taken some effort to put together. The songs are for the most part well recorded, with some particularly good snare, vocal and electric guitar sounds, although the acoustic guitar has been DI'd and sounds rather artificial. This works for the opening number but not the slower tracks. It's a shame that the acoustic guitar wasn't miked up using one of those lovely Rode or Neumann microphones mentioned in the band equipment list — even a combination of DI and microphone can be effective. The drummer is a little inconsistent with the volume of his snare hits and occasionally drops one out or disappears completely when the backing goes up in level (on the chorus of the first song, for example). To make mixing easier, Urge could bounce the snare from the one-inch reel-reel machine used to record it onto their PC for editing in Cubase. I see that a synchroniser is included in the kit list, and it's a vital bit of equipment for this kind of job. Finally, the bass end of the mix doesn't seem to have the depth it should do, especially since it's coming off tape, so the band need to make sure that they're not over-EQ'ing at source or rolling too much of the low end off at the final mix.

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



First Look: Cubase SX3

In this article:

Warp Factor 9 Editing The Global Project Timeline Making Your Selection It's All In The Wrist

First Look: Cubase SX3 Cubase Notes Published in SOS October 2004 Print article : Close window

Technique : Cubase Notes

This month we provide an exclusive preview of the new version of Cubase, SX 3, and offer advice on how to get the most out of Cubase's Range Selection tool. Mark Wherry

The big news in the Cubase community by the time you're reading this month's column will be Steinberg's announcement of Cubase SX 3, the first major revision to Cubase SX since, well, Cubase SX 2 last summer. But seriously, SX 3 is a very significant update to the application, and, in many ways, is a bigger step forward for Cubase users than last year's update to SX, mainly due to the fact that the emphasis in the new feature set is on musically creative functions. There are also a great many bug fixes and workflow improvements that are especially gratifying to see, as a few stem from a year's worth of campaigning the good people at Steinberg on behalf of a film composer of my acquaintance who uses Cubase and Nuendo.

Version 3 of Cubase SX maintains the visual style of its predecessor but includes a number of welcome changes, such as the ability to see the colour assigned to a track, viewing of the Sections for a VST Instrument Channel below the Sections in the Inspector for the corresponding MIDI track, and a smaller 'smallest' height for tracks. Notice also the Play Order Editor, which allows you to specify an order for the defined Play Order Parts when a Project is played back.

Warp Factor 9 If you were exhausted by the bad jokes that could be associated with the Time

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Warp tool introduced in SX 2 (and covered a couple of months back in Cubase Notes), you'll be pleased at the prospect of more warp-related puns in SX 3, courtesy of the new Audio Warping feature. Audio Warping essentially brings Acid-, Live- or Soundtrack-like functionality into Cubase, for real-time timestretching and pitch-shifting of audio, which includes the ability to import Acid loops, opening up a vast library of content for SX users. The feature works in a similar way to the current method of creating hitpoints for slicing up audio, except that once you have identified the rhythm of an Audio Event with so-called Warp Tabs, you can either drag these to timestretch the audio, or have Cubase use them to adapt the tempo of the audio to match the Project during playback. As I mentioned earlier, this all happens in real time and enables the audio to follow even the most eccentric of tempo changes — including ramps. For those who enjoyed the days of pattern-based sequencers, before the current linear graphical arrange windows came into vogue, Play Order Tracks will be especially welcome, and certainly rank among my favourite new features. A Play Order Track is a new track type that allows you to define regions within your arrangement (called Play Order Parts), which you can then specify to play in a certain order, rather than have your Project play back linearly from bar one to the end. This is a very powerful arranging feature that's great for trying out ideas, and once you have a Play Order you like, it's possible to 'flatten' playback of the Play Order back into a linear arrangement. One neat feature of the old Cubase was Mixer Maps: the ability to create onscreen panels to control external MIDI devices, so that if you wanted an onscreen representation of your Access Virus synth, for example, you could define various rotary controls, faders and switches to send messages and remotely alter the physical controls on the device, creating a Mixer Map. This functionality has returned in SX 3, as an extension to MIDI Device Maps, called MIDI Device Panels. Once you've created a custom Panel, it can appear in its own window, much like a VST Instrument editor, or in a dedicated Inspection Section or Mixer view. Last, but not least, in my very brief highlights of SX 3 is Inplace Editing, which enables you to expand a MIDI Track so that it displays an almost complete Key Editor, including Controller Lanes, in the space that the Track occupies in the Project window. Although that's all we've got space for this month as a preview, look out for the review of SX 3 in next month's issue. There we'll be making a more in-depth examination of what Steinberg have done with the latest version of Cubase.



Editing The Global Project Timeline As you may know, it's possible to manipulate the global timeline of a Project based on the positions of the left and right locators, moving all objects forward or backwards in time on the Event Display, or cutting and pasting sections of Parts across all tracks in the Project in one pass. These time-based commands are particularly useful because they also affect the Tempo Track, moving tempo changes forwards and backwards as needed. That can normally only be done via the Tempo Track editor. To cut all Events between the locators to the clipboard, select Edit / Range / Cut Time or press Shift+Control/Apple+X. This selection can later be pasted to a new part of the Project at the current position of the Project Cursor by selecting Edit / Range / Paste Time or pressing Shift+Control/Apple+P. Alternatively, you can paste the cut section back to its original position by selecting Edit / Range / Paste Time at Origin. If you wanted to delete the section of the Project (without it being copied to the clipboard), you'd simply choose Edit / Range / Delete Time. When you're working on a Project, it's quite common to reach a stage where you need to insert an extra couple of bars between objects, and it can be a real pain to manually split all the objects (where necessary) and move them forward. Instead, to insert a couple of bars at a point in the Project, use the left locator to indicate where you want the extra time to be inserted, and the right locator to specify the length of time you want to add. So, if you wanted to add four bars at bar 41, for example, you'd drag out a selection from bars 41 to 45 and select Edit / Range / Insert Silence or press Shift+Control/Apple+E. The reason the above techniques are worth knowing about in the context of Range Selection, as covered in the main text, is that all of these commands work within the scope of a Range Selection if you don't want to affect the whole Project. For example, if you wanted to insert two bars in the middle of a Project on only two tracks, you could drag out a Region Selection across these two tracks (where the start represents the point to add the extra time and the end denotes the length of time to add), and then use the same Edit / Range / Insert Silence Command.

Making Your Selection When you make selections on the Project window with the Select tool, it's only possible to fully select one or more objects, which is great if you want to copy the eight-bar piano Part from bar one to bar nine, for example. However, what if you wanted to copy only bars three to six of the piano to bar nine, instead of the whole thing? While this isn't exactly a cumbersome process, you'd have to perform a little resizing after copying the piano Part, to trim the start and end points, and also make sure the Part started correctly at bar nine. To make life easier in these situations, especially when working with multiple objects across multiple tracks, Steinberg implemented the Range Selection tool, which allows you to select just a region of the Project window. Taking the previous example, the Range Selection tool would allow us to select just bars three to six, rather than the whole Part, and copy only that section to bar nine.



The Range Selection tool's icon appears next to the main Select 'arrow' tool on the Project window's toolbar and can be selected using the same methods as for any other tool — either by clicking on this icon, pressing the relevant Key Command (2 on the main part of the keyboard, by default), or from the Quick Menu. With the Range Selection tool selected, you can make a selection by simply dragging the mouse in any direction to cover the area you want. The selection can be made across a single object and track, or over multiple objects and tracks; if you've used a word-processing application such as Microsoft's Word, it might be helpful to think of Cubase's Range Selection procedure like making a text selection in Word.

The Region Selection tool makes accomplishing certain operations, such as the one illustrated in these two screens, incredibly quick and easy. Here, a selection that includes certain sections of specific Parts is moved to another area of the Project in one drag of the mouse: no need for the Scissors tool!

While you're making a selection, you might notice that it automatically snaps to the Event Display, if Snap Mode is enabled. Once a selection has been made, you can modify its start and end points by placing the mouse pointer over the start or end of the selection, until you see the resizing tool, and dragging the start or end point to the new position. Alternatively, make more precise numerical alterations to the selection by using the Event Infoline, which automatically displays the settings of the current selection.

It's All In The Wrist Once you've selected a range with the Range Selection tool, there are many tasks you can carry out on the selection while the tool is still active — but once you choose another tool, the selection made with the Range Selection tool disappears. Two of the most useful transport commands also work when you've made a selection, allowing you to set the position of the left and right locators based on the selection (by selecting Transport / Locators to Selection or pressing P), or set the position of the Project Cursor to the start of the selection (by selecting Transport / Locate Selection or pressing L). You can move the data from whatever objects are encapsulated within the selection by simply dragging the selection to another area of the Event Display. As you drag, an indicator will appear to show you what position it will be moved to when you release the mouse, just as when moving objects. The fundamental difference between using the Range Selection tool for tasks like moving, as file:///H|/SOS%2004-10/First%20Look%20%20Cubase%20SX3.htm (4 of 5)9/26/2005 12:38:27 AM


opposed to using the Object Selection tool, is that only the selected area is moved when you use the Range Selection tool, regardless of whether the selection covers the entire length of an object. The Range Selection tool thus provides a quick way of moving parts of an object, without having to use the Split tool first before moving the relevant section. If you hold down the Alt/Option key while dragging a selection with the Range Selection tool, the selected area is copied to another part of the Event Display, leaving the original objects unharmed. To delete an area marked by a selection, select Edit / Delete or press Backspace. You can split objects covered by the selection at the start and end points of the selection by choosing Edit / Range / Split or pressing Shift+X. A useful way of resizing an object or objects is to drag out a selection representing the new length of the object, with the Range Selection tool, and select Edit / Range / Crop. The objects covered by the selection are cropped to the start and end points of the selection, although no data is actually lost and you can easily resize the object to restore it. Published in SOS October 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



How To Remote Control Sonar Effects

In this article:


More Track View Sonar Notes Customisation Published in SOS October 2004 Wipeout! Remote Effects Control Print article : Close window Another Track View Tip Technique : Sonar Notes Easy Linear Tempo Changes Quick Tips

We offer our customary haul of hints and tips. This month, how to use the handy Scrub tool, how to customise your track view, and an explanation of the procedure for setting up remote control of Sonar effects parameters. Craig Anderton

I always enjoy getting email from SOS readers, because you give me ideas for subjects to write about. For example, a reader mentioned that he had switched to Sonar from another sequencer, and that the only feature he really missed was the ability to click on a MIDI note and hear it. That's when I realized that we've never paid proper attention to the Scrub tool. When the FX button in the Console View's Click on the Scrub icon in the Track toolbar glows blue, you can see up to four View (or type 'B'), and as you drag the effects parameters associated with the cursor over audio data you'll hear it. selected effect in the FX slot (see 'Remote But in the MIDI Piano Roll or Staff Effects Control'). These are easy to control View, you can scrub or click at any remotely and automate. arbitrary location (including on a note) and you'll hear any MIDI notes that lie within the vertical line where you clicked. For example, if there's a chord, you'll hear the entire chord.

More Track View Customisation

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Even though Sonar 3 added a shiny new Console View, I still do most of my work in the Track View. As a result, I'm always seeking ways to make the interface as efficient as possible. One way is to customise it by reordering the various control strips (aka 'Widgets' — volume, pan, trim, channel in/out, and so on) so that the ones you use the most are easily accessible. Ideally, you want the parameters you use the most toward the top, so you can have a relatively short track height (thus letting you view the greatest number of tracks at a time), yet still see the parameters you need. The secret to re-ordering Widgets lies within the CAKEWALK.INI file, found within the Sonar folder (typically the path is C:\Program Files\Cakewalk\Sonar 3). This file can be edited with a text editor such as Notepad. Add the following anywhere within the file: [Audio Widgets] W0=Volume W1=Pan W2=Interleave W3=Trim W4=FX Widget No.

W5=Aux W6=Output W7=Input The lower the Widget number, the higher the Widget is in the area below the Track Header bar (for example, W0 is just below the Track Header). The above is the default setting. I prefer the following changes: W2=Trim

[MIDI Widgets]

W0=

Volume

InputGain

W1=

Pan

InputPan

W2=

Trim

Volume

W3=

FX

W4=

Reverb

Output

W5=

Chorus

Aux

W6=

Channel

FX

W7=

Bank

W8=

Patch

W9=

Output

W10=

Input

W11=

Key

W12=

Time

Pan

Interleave

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[Buss Widgets]


W5=Output W6=Input W7=Interleave Note that although entries in the CAKEWALK.INI file are not case-sensitive, the syntax (ie. no space before or after the '=' sign) and number of Widgets must be correct for re-ordering to work. Also, each parameter must be on its own line. Don't worry, this isn't like messing around with the registry: if something's wrong, Sonar will simply ignore it and load the defaults. The table on the right shows the default settings for the MIDI and Buss Widgets. If you like the defaults, you don't need to enter anything, as Sonar will just load the defaults. But if you want to make changes, enter them into CAKEWALK.INI using the same format as for the Audio Widgets (ie. bracketed heading, followed by each parameter on its own line).

Wipeout! The Wipe function can save quite a bit of time when you want to dispose of data. It removes all data for a selected track, but unlike the Delete command — which removes a track and everything in it — Wipe removes only the Clip data within the track, leaving all other parameters and inserted effects intact. Wipe applies to all selected tracks, so you can wipe multiple tracks at once when you go Track / Wipe. Of course, you can accomplish the same thing by drawing a marquee around all the Clips in a track and then hitting Delete, but that usually means you'll need to resize the Track View so that you can see the song's entire duration.

Remote Effects Control One of the welcome features in Sonar Producer Edition is the ability to choose four parameters from any effect in the effects slot and control them from the Console View. We talked about changing control assignments in the December 2003 Sonar Notes; these four controls normally default to the first four parameters in whatever effect you've selected, but you can change them to any four available effect parameters. However, we didn't cover remote control, which is becoming increasingly relevant in these days of external hardware control devices such as the M-Audio Oxygen 8 and the various Evolution and Kenton fader boxes. Fortunately, setting up this kind of control is easy. file:///H|/SOS%2004-10/How%20To%20Remote%20Control%20Sonar%20Effects.htm (3 of 6)9/26/2005 12:38:31 AM


Remote control has two main uses. One is so that you can vary parameters much more conveniently than with a mouse, when setting up an effect. Once you have the sound exactly as desired, right-click on the parameters you adjusted and select 'Disable Remote Control' so that further control movement doesn't disturb the The Series Of Tempos option is the quickest settings. You can also use remote way to produce a linear tempo change over a control for dynamic automation of the specific number of measures. effects parameters. This works in the same basic way as automating any other parameter, such as pan or volume: right-click on the parameter and choose 'Arm for Automation', then click on the Transport's Record Automation button. To set up the controls, make sure that the four parameters are visible. To do this, click on the FX button in the console view's lower left toolbar. When the effects are hidden, the button is grey. Click again, and it glows green; this shows the FX slot, and which effects are inserted. Click again so that it glows blue, and you'll see the four parameters show up under the FX slot. To control a parameter remotely, right-click on the parameter name and select Remote Control. This brings up the Remote Control menu, where you have multiple options for control devices: notes, controllers, wheel, RPN, SysEx, and so on. We'll assume you want to use a controller, so click on its button. You can enter a controller number and MIDI channel, but it's easier to just click on 'Learn' (or type Alt-L) while turning the control. As you turn the control, the parameter's virtual slider should move in tandem. Note that you can assign multiple parameters to the same controller — not just in the same effect, but in different effects. So, for example, you could increase the delay feedback in a delay effect while simultaneously changing LFO depth in a phase shifter. Incidentally, if you're not sure whether an effect has automatable parameters, it's easy to check. Right-click on the effect name in the FX slot, and choose 'Arm Parameter'. A dialogue box will appear showing the parameters that are available for automation (if any). If you've already enabled a parameter for automation, it will have a tick mark next to its name.



Another Track View Tip Normally, the Track Header bar contains only the track name and other 'fixed' parameters (mute, solo, minimise, and so on). As you narrow the track height, the upper control strips (typically Volume, Pan, etc) slide into the header to conserve space. In Sonar 3, it's now possible to force the top control strips into the Track Header bar regardless of the track height, as long as there's sufficient room to display them. I set the Track Header width so there's room for the Level and Pan controls, then choose horizontal meters and, finally, reduce the track height so that only the Track Header Bar and meter are visible. This allows me to fit a large number of tracks into the Track View, which is excellent for mixing. To do this tweak, add the following anywhere under the [Wincake] heading in CAKEWALK.INI: TVWidgetsStickInHeader=1 To return to normal operation, set the variable to '0'.

Easy Linear Tempo Changes Sonar's Tempo View provides a detailed way to change tempo, as you can draw in any kind of arbitrary changes and Sonar will follow along. However, when you simply want to speed up or slow down over a certain period of time, there's a much simpler option. Go Insert / Series of Tempos, and you'll see a dialogue box where you can specify a start and end tempo, and the range over which the tempo change between the two occurs. The 'Step' parameter is particularly useful, as it sets the increments by which the tempo changes. 0.10 BPM seems plenty smooth enough, but you can go to an even smaller step size if desired. If you open up the Tempo View, you'll see the results of any tempo changes you created with the Series of Tempos option. You can tweak them further in this view if needed.



Quick Tips If you find it annoying to alternate among the various editing tools via mousing, learn the keyboard equivalents: [S]election, [E]rase, [D]raw, and Scru[B]. To change the order of effects in the FX Slot, click on an effect name and drag it into the desired place. You can choose different 'woods' for the virtual guitar neck that appears in the Staff View. Right-click on the neck and select rosewood, ebony, or maple. To copy an effect and its current settings to another track, Ctrl-drag the effect to the track. Published in SOS October 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:

Hogging The Buss PC Snippets PCI Dawg The Results Are In Space: The Final Frontier Tweaking Timings

Incurable Soundcard Stuttering? PC Notes Published in SOS October 2004 Print article : Close window

Technique : PC Notes

We investigate an obscure PC parameter that may be responsible for some otherwise 'incurable' soundcard stuttering problems. Martin Walker

Back in SOS December 2002, as part of my 'Bionic PC' collection of BIOS tweaks for better music performance, I mentioned the PCI Latency Timer setting. When a device such as a PCI expansion card gets hold of the PCI buss, this Timer starts counting down, and when it reaches zero it releases the buss to let other devices have their turn. If there are none waiting, it can grab the buss again and continue. The PCI Latency Timer can range in value from zero to 255 clock cycles. When it's set to zero, a device will give up the buss immediately if another device needs it, but as the timer value increases the device will continue using the buss for progressively longer before 'letting go', while other devices wait their turn.

Mark Knutson's freeware Double Dawg utility lets you view and change PCI latency values, which may cure some soundcard stuttering problems. Here my laptop's ATI graphics display an extremely high value of 255. Reducing this to 128 may prove beneficial to audio performance.

If all devices have high settings they may each have to wait longer before they get a chance to send their own data over the PCI buss, but once they get hold of it they will themselves be able to hang on to it for a longer period. However, if all devices have low settings they tend to swap control more often, sometimes preventing large data bursts, resulting in increased overheads and therefore reduced performance (just as using very small soundcard buffers results in increased CPU overhead). However, system stability may improve because no file:///H|/SOS%2004-10/Incurable%20Soundcard%20Stuttering.htm (1 of 6)9/26/2005 12:38:35 AM


device has to wait for the buss.

Hogging The Buss Most of the time the default BIOS Latency Timer setting for PCI devices (usually 32 or 64 clocks) is a compromise that works well, and some motherboards even hardwire a default value rather than let you change it in the BIOS. However, occasionally some PCs exhibit problems that just don't respond to the normal array of procedures, and sometimes they can be traced back to the Latency Timer. Top of the list for musicians are audio and video glitching and stuttering problems, occasional lock-ups when playing sounds, audio distortion, and sometimes even continuous white noise instead of music coming from the output socket. Some people have also suffered inexplicable non-audio problems, such as corruption when copying files between hard drives. The common factor in all these problems tends to be that there's loads of data travelling to and fro on the PCI buss, and some such problems have eventually been attributed to system devices that ignore the default PCI Latency Timer setting. Instead they use a value specified by their device driver software, a value that may even change from one driver revision to another — which may explain why driver updates can cure some problems but make others worse. The most common culprits tend to be AGP graphics cards (although AGP is a different type of expansion slot, it still shows up on a scan of PCI devices, and interacts with the PCI buss). Some cards are forced by their manufacturers to a very high setting, and can therefore hog the buss. The setting can be as high as 255 clocks, although some systems round it down to the nearest multiple of eight — in this case, 248. With many general-purpose applications this won't be noticed, since graphics performance is often of the essence, but when you're trying to record and play back lots of simultaneous audio channels you may run into conflicts. In such cases, reducing the PCI Latency Timer setting of an AGP card to 128 (or less) can sometimes completely cure occasional stuttering problems with other PCI devices, and rarely results in any degradation in video performance. Many Matrox graphic cards have the default setting of a modest 64, whereas several different ATI Radeon and Nvidia cards found in both desktop and laptop PCs have been found to be set at 255. Such trends among manufacturers may explain why some ranges of graphic cards end up being preferred by musicians as being more 'compatible' with audio hardware and software, when a simple change to the PCI Latency Timer might level the playing field a bit.

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PC Snippets I mentioned, in my recent review of Emu's 1820M (see SOS June 2004), that I found the font used for its PatchMix DSP inserts and scribble strips rather difficult to read, and others obviously agree. Emu user 'freideuter' has created a new replacement font that's a tiny 15kb zipped download containing four BMP files. You just copy these into the existing 'Creative Professional\E-MU PatchMix DSP\Skins\Default\images' folder (saving the original ones first, just in case), and then restart PatchMix DSP. In my opinion the new font is a great improvement. www.gehirntanz.de/NewFontPatchMixDSP.zip The handy Miditest 2.2 utility that I mentioned in the PC Notes colums of April 2004 is now up to version 2.6, and since I last looked at it, it has benefited from some useful tweaks.

As well as a few bug fixes, Miditest now offers a set of tick boxes to allow you to select what types of MIDI data to include in the test. A brand-new test report has also been implemented, which reveals yet more facts about your MIDI interface. The report even includes details of your MIDI interface's latency and jitter performance, which are extremely welcome. http://earthvegaconnection.com

PCI Dawg Various PCI Latency Timer utilities have been written to solve specific Timerrelated problems. However, until now I haven't come across an easy-to-use Windows utility that lets you examine and change PCI Latency timings for individual devices. Enter Double Dawg, from SOS Forum contributor Mark Knutson (www.mark-knutson.com/technique_to_target_inc). This free utility runs under Windows 2000 and XP and provides a simple readout of all devices found on the PCI buss, along with their current latency timings. Non buss-mastering devices and those that are hardwired show up with a zero setting (see the screenshot on page 276), but you can override the setting of any 'non-zero' device by typing a new value into the adjacent box. Clicking on the 'Set Latency' button makes these new values 'take' immediately, so you can see what effect they have on your soundcard's performance. The settings revert to their original default each time you boot up, so you'll need to run Double Dawg each time to change them, although Mark can also supply Double Dawg Boot-O-Magic to do this automatically, for a modest $10 shareware payment.

The Results Are In I examined the Timer settings for a few soundcards, to get a feel for typical figures. The majority of soundcards seem to adopt the BIOS or motherboard default setting, including Emu's 1820M and Yamaha's SW1000XG, but Echo's Mia and Indigo both have a high forced value of 192, presumably to give them more of a chance against various graphic cards. file:///H|/SOS%2004-10/Incurable%20Soundcard%20Stuttering.htm (3 of 6)9/26/2005 12:38:35 AM


My Matrox G450 graphics card also adopts the default setting, unless I set this to 32 or 248, in which case it reverts to 64, presumably to prevent graphic performance degradation at one end and PCI buss hogging at the other. Sometimes values change from one driver update to another — like, for example, Universal Audio's UAD1 DSP card, which temporarily jumped to a forced value of 255 in version 2.0, to compete with a few other soundcards that do the same, but dropped back to the default BIOS value in version 2.1. One RME HDSP Multiface user reported a value of 255, and one MOTU interface user claimed that his interface had a PCI Latency Timer value of zero. If true, that might explain some of the system-specific problems that some PC MOTU users have been having. I've been collating a list of devices that either use the default timer value or force their own value, and Mark Knutson has kindly agreed to host this on his web site. If you're using a DSP card, such as Universal Audio's UAD1 or TC's Powercore, alongside your soundcard, you may have to be even more careful about relative timings, since you have two audio devices sharing PCI bandwidth. Depending on what relative proportion of DSP plug-ins and audio tracks you typically have in your songs, you may get more 'balanced' performance by tweaking your soundcard or DSP card PCI Latency Timer.

Space: The Final Frontier Have you ever wondered what proportion of your hard drives are devoted to Windows, its applications, your audio files, samples, and so on? I've used utilities in the past that display drive contents as tiny pie charts, but a program called Spacemonger is far more informative. It provides a window showing file contents as rectangles whose size on screen is proportional to their file-size, so you With the free Spacemonger utility you can can instantly see what's taking up instantly see how your hard drive space is most of the space. You can make divided up, and what's taking up the most this window any size you like, up to room. In my case it's a 241MB file belonging full screen, zoom in and out to view to Steinberg's Groove Agent. the contents of folders, choose how much detail to display, open or delete files, and change various preferences. It's a single 212Kb file that runs on WIndows 95, 98, and NT (it worked fine for me on XP as well). Best of all, it's freeware. www.werkema.com/software/spacemonger.html

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If you're not having any problems with your audio, don't be tempted to start radically altering every PCI Latency Timer in sight. You may simply make everything worse! However, if you have unexplained glitching that doesn't respond to increasing soundcard buffer sizes, disabling background tasks and setting Windows XP's processor scheduling to 'Background services', Double Dawg may just help you solve your problems. Some people suggest simply giving most devices a high setting such as 176, to 'open up' the PCI buss for larger data bursts, maximising its bandwidth, and giving the soundcard an even higher setting, of 255. Personally, I prefer to tread far more carefully, since we're trying to balance the PCI buss requirements of the various devices so that none causes drop-outs for another. Some may require a higher priority than others to achieve the best results, so forcing your soundcard to the highest value may prove counterproductive, and may even reduce the stability of your PC. It's difficult to predict the outcome of lots of changes at once, particularly if you have video or gaming requirements, and it's possible to cause data corruption during hard-drive reads and writes with some extreme settings — as well as crashes, allegedly. The most obvious approach is to identify any device with a latency setting much higher than the rest and try reducing this if it seems sensible. For instance, graphics cards could be reduced from 255 down to 128 or 96 to see if audio stuttering can be cured without affecting graphic performance, while a network device currently set at 128 might be reduced to 32. Running benchmark tests before and after the changes may help to identify performance gains (particularly in disk transfer rates), although in many cases I doubt that you'll measure any difference. One musician did find that after increasing the overall BIOS setting from 32 to 96 he could reduce his ASIO buffer latency from about 40ms at 44.1kHz right down to 12ms, although I suspect this was probably because another device forced to a higher setting provided less competition. However, I came across more conclusive evidence on a UAD1 forum, where one user who only managed to run two instances of the 1176 LN compressor plug-in without crackles managed to run six after reducing a few PCI Latency timings (soundcard, USB card, Firewire card, and mostly the graphics card, from 248 to 80). To sum up, this can be a hugely confusing area, even for the expert, since you'll find a different combination of PCI and AGP devices in each PC, and they and their particular driver versions may interact in various ways. Don't assume that tweaking a few Timer values will solve every soundcard problem — it won't. However, if you're having glitches that won't go away, it's another area to investigate that might just pay dividends. Published in SOS October 2004






Linux Jack Sound Server Ported to OS X

In this article:

Soundflower The Jack Tools Team Web Links Reliability & Performance Glossary Routing With Plug-ins

Linux Jack Sound Server Ported to OS X Apple Notes Published in SOS October 2004 Print article : Close window

Technique : Apple Notes

The Jack low-latency sound server has been ported to OS X from Linux. We discuss its background and implications. Daniel James

The UNIX core of Apple's OS X means that, with appropriate modifications, it can make use of software originally created on other UNIX-family platforms. Ready availability of source code and licenses which allow derivative versions of programs make the process all the more likely, with most Linux and BSD (Berkeley Software Distribution) distributions featuring thousands of packages which are anything but exclusive, since a program found on one UNIX platform is generally available to any other. Apple quietly acknowledged this aspect of UNIX when they used parts of the Konqueror web browser, developed on Linux, as the basis of their Safari. Perhaps the single most significant pro audio project on Linux is the Jack audio server (see the feature on Mirror Image Studios, SOS February 2004), which has now been ported to OS X, with good results, and thanks to some handy features in the Mach microkernel underlying Apple's UNIX, may even outperform the Linux original. A free download, Jack has been tested on OS 10.2.6 and later versions, including Panther. Unlike Propellerheads' Rewire interconnection protocol, it doesn't require specific application support, being capable of connecting any application that is Core Audio-compatible, and it can allow a third application to act as an effects insert with the use of a VST or Audio Units plugin. At the same time, a growing number of Linux audio programs are adding native Jack support.

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Soundflower Jack Tools isn't the only software to bring the Rewire concept up to date. Cycling '74 have released Soundflower, which is similar in concept, if not in detail. It's also available as a free download, from the Cycling '74 website. Dan Nigrin (see main text) isn't dismissive of the rival effort: "Soundflower is a great piece of software, and in some ways it's a shame that the Jack OS X team and Matt Ingalls, who developed Soundflower for Cycling '74, didn't know about each other's work until after we each released our respective applications. Soundflower was originally introduced in beta form to the Max/MSP mailing list as MSP Patcher, literally on the day we announced Jack OS X — go figure!" Nigrin argues that these two alternatives to Rewire both have their merits, and that there's no reason why OS X users shouldn't download and try both. "Soundflower's advantage is that it's simple — just select the driver and go. Also, there are a few applications that Jack currently has trouble interacting with — specifically some of the Native Instruments ones — and Soundflower seems to do better with those. On the other hand, Jack OS X is more flexible than Soundflower; it's not limited to either two channels or 16. Applications that don't allow you to choose separate input and output drivers will have their options limited when using Soundflower. Jack's extension into the AU and VST plug-in space allows for more complex audio patching configurations between (and within) applications. And the fact that Jack is open source, and — at least in the Linux world — has a large developer base, ensures that it will continue to develop and improve." Letz adds: "Soundflower is a Core Audio kernel driver that uses a buss approach; this is a simple solution, but it's less flexible. Jack allows for centralised control of the running applications' connection state, giving the ability to save and restore setups. The architecture of Jack allows you to describe any kind of connection graph between audio clients, as long as there aren't loops."

The Jack Tools Team I talked to Dan Nigrin, a member of the Jack Tools project, who also runs the Defective Records label in Baltimore, USA. "Although Rewire works beautifully on applications that have Rewire functionality built into them, either as client or host, it unfortunately does nothing for those that don't," comments Dan. "Where Rewire still has advantages is that it allows for MIDI and transport information to flow between connected applications. We hope to have this functionality in Jack Tools in upcoming versions; the Jack architecture definitely supports it." Working on the project at GRAME, a national musical institute in France, Stéphane Letz explained why the Jack Tools team felt the need to go beyond Rewire. "Our first goal in porting Jack to OS X was to get a free, flexible and lowlatency audio server that had already proved its utility on Linux, thus facilitating the port of Linux audio programs to the Mac. The Jack Audio Router, a driver that allows any Core Audio application to access the Jack server, was developed as a second step, giving all Core Audio applications the ability to take advantage of Jack's capabilities." file:///H|/SOS%2004-10/Linux%20Jack%20Sound%20Server%20Ported%20to%20OS%20X.htm (2 of 6)9/26/2005 12:38:40 AM


Jack isn't a simple clone of Propellerheads' existing system. "The Rewire and Jack architectures are quite different," says Stéphane Letz. "Rewire allows an audio master The Jack Pilot main window and connections application to run and synchronise window, showing iTunes sending audio to slave applications — all audio code Max/MSP and in turn to the built-in runs in the context of the master soundcard. application, and the GUIs of the client applications run in the client context. By contrast, Jack is based on a client/server model, with the Jack server at the centre of the system. It interacts with the soundcard driver and communicates with all registered clients. Each node in the Jack 'client graph' has its own process functions, which have to be called in a specific order. The whole graph is executed synchronously by the hardware driver, typically waking the server at regular intervals determined by its buffer size. The Jack server then distributes this audio interrupt to all running clients." Architectural merits aren't the only advantage of Jack, as Letz explains: "Developing a Rewire-enabled application requires its developer to adapt the program's source code. With the Jack Audio Router driver for Core Audio, there is no need. The Jack approach allows for any kind of connection topology, and is therefore very flexible. Extensions can be developed very easily in the future." In addition, the proprietary nature of the Rewire protocol creates licensing problems for third-party developers porting free software from other UNIX-family platforms. Jack Tools development team member Johnny Petrantoni adds: "Consider that Rewire is written using the old binary Macintosh style CFM [Code Fragment Manager format, used in the classic Mac OS]. There is a lot of wrapping required to make it work in a Mach-O [Mach object file format] application, and of course this creates processing overhead."

Web Links Jack Tools main site: http://www.jackosx.com Development site: http://sourceforge.net/projects/jackosx Jack on OS X discussion group: http://groups.yahoo.com/group/jackosx/ Soundflower: http://www.cycling74.com

Reliability & Performance The crucial factor in acceptance of either Jack Tools or Cycling 74's Soundflower (see 'Soundflower' box, previous page) is likely to be real-world reliability. After all, neat routing features aren't much use if the audio infrastructure starts to throw up problems halfway through a critical mixing session. The Jack Tools team file:///H|/SOS%2004-10/Linux%20Jack%20Sound%20Server%20Ported%20to%20OS%20X.htm (3 of 6)9/26/2005 12:38:40 AM


found that OS X machines can suffer from clock drift in certain configurations, as Letz relates: "The clock drift problem is a tricky one as soon as several audio drivers are used at the same time in an application, and there is no generic solution provided by Apple yet." Fortunately, Jack Tools doesn't suffer from this particular problem. "By design the Jack server uses the real soundcard driver it is connected to, and all Jack clients share the same timing source. We may add multi-driver support in a future version so that we can use several soundcards at the same time, but then we'll need to solve the clock drift problem at the Jack server level. One real driver would be the master and all other drivers would have to be synchronised to that."

Here's what the desktop corresponding to the screen overleaf looks like, with iTunes playing an Internet radio station into Max/ MSP, which is pitch-shifting the audio and displaying three sonograms.

A few workarounds exist already which enable multiple soundcards to be used with Jack, such as in the scenario where a laptop musician or DJ needs a cue mix on headphones but only has a single stereo output on their external audio interface. For example, the prelistening outputs in Ableton's Live can be routed via a virtual four-channel Jack device to an application which uses the internal soundcard on the laptop, while the main outputs are routed to the external interface. Putting audio routing under control of the CPU rather than a fistful of cables means that a split-second glitch could cause audio output to fail completely, should the sound server stop. However, Nigrin is reasonably confident that Jack is ready for serious work: "Purely from anecdotal personal use, it's incredibly stable if your CPU can manage the various applications you're connecting. There is a theoretical limit over which Jack would begin to drop clients, but I don't think we've encountered this ourselves, or heard about it on our Jack OS X forum." Letz explains that it's the audio applications that need to keep up with Jack, rather than the other way around. "The fundamental requirement for audio applications is that they complete their work during the audio cycle. If three apps are running in Jack at 64 frames per buffer, the sum of their audio callback duration must not exceed the duration of one 64-frame buffer; that's about 1.45 milliseconds. The Jack server contains profiling code to measure each application's audio callback duration at every cycle. The profiling code may decide to drop an application that cannot meet its deadline, and remove it from the Jack client graph." Direct comparisons of performance between the three systems — Rewire, Jack and Soundflower — are difficult to make, due to the differences in architecture. "We have not yet made extensive formal performance comparisons," says Letz. file:///H|/SOS%2004-10/Linux%20Jack%20Sound%20Server%20Ported%20to%20OS%20X.htm (4 of 6)9/26/2005 12:38:40 AM


"The original Jack Tools version had some room for improvement, and we are currently working on a new simplified design for the Jack server that will be even more optimised, and will take advantage of multi-processor machines." This forthcoming version of Jack Tools, currently in beta, will also include the NetJack application, enabling audio to be streamed between different studio machines on a network, using UDP packets (see glossary).

Glossary Distribution: The traditional way that a free UNIX is packaged. Operating system, drivers, utilities and applications all in one bundle. BSD: The Berkeley Software Distribution, a free UNIX originally produced at the University of California and mostly used on servers. A lot of BSD technology was incorporated by Apple into OS X. Linux: Usually taken to mean GNU/Linux, an operating system designed from scratch as a free implementation of the UNIX concept. Runs on just about any hardware, including Apple and PC. UDP: User Datagram Protocol, a method for sending data over an Internet Protocol (IP) network. Web and email servers use Transmission Control Protocol (TCP) instead of UDP.

Routing With Plug-ins As complex as it might seem, Nigrin claims that using a VST or Audio Units plugin to send and return Jack audio streams directly into an application is no less stable than connecting to Jack with main inputs or outputs. "It's really no different than using it via a Core Audio driver. When using a Jack VST or AU plug-in, you create an audio window into an application via its plug-in interfaces. So, for instance, you could allow an application to send audio to the Jack server via a stereo-track effect-send channel using the plug-in; from there, you could have the Jack server send the audio on to another application for processing, and then return that audio back into the first application, via another VST or AU plug-in instance on another channel." It could be that the limitation on routing between audio applications in future will be our ability to comprehend and keep track of the connections, rather than any technological constraint. "It's a bit of a challenge to enable users to visualise all of these connections," says Nigrin. "This kind of flexibility is really quite new for everyone. One of our goals for the future is to be able to come up with a graphical connections manager, instead of the tabular one we have now, as we think that this will allow people to better understand the signal flows that are possible." Published in SOS October 2004






Logic Notes

In this article:

Emulating The TC2290 Delay In Logic The Klopfgeist Challenge Dynamic Reverb Logic Tips De-essing Your Reverb Have Your Say!

Logic Notes More hints and tips Published in SOS October 2004 Print article : Close window

Technique : Logic Notes

Current Versions Mac OS X: Logic Pro v6.4.2 Mac OS 9: Logic Pro v6.4.2

Learn how to combine dynamics processing with delay and reverb to breathe life into your mixes.

PC: Logic Audio Platinum v5.5.1

Mike Senior

One of my favourite interviews in SOS was back in December 1999, when we interviewed the producer Al Stone about his production work with Jamiroquai. When the topic moved to mixing, he mentioned that he's a fan of using TC Electronic's old TC2290 delay unit on vocals, one of the reasons being that it allows you to "suppress the delay until the vocal line is finished, so it doesn't clutter". In other words, the delay effect is kept low in level while the vocalist is singing, helping the words to remain intelligible, but is then allowed to bubble up in the gaps between phrases.

Here you can see how to set up Logic to emulate the TC Electronic TC2290 duckingdelay effect.

Although you could reproduce the action of the TC2290 in Logic using the automation facilities, this would quickly become a tedious job on all but the most simple of vocal lines. However, there is another way to recreate TC Electronic's dynamic delay effect by using Logic's flexible side-chaining facilities.

Emulating The TC2290 Delay In Logic The first thing to do is set up a standard delay effect. For this example, let's say

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Logic Notes

that your vocal line is playing back through the first Track Audio object, and that you're using the first Buss Audio object to return the delay effect. Create a send to the buss from the vocal Track object by selecting from its drop-down Sends menu. This creates the audio routing from the Track object to the Buss object, into which you should now insert a delay plug-in of your choice. I find that the Tape Delay plug-in is a good one for understated vocal treatments, so that's what I'd tend to use. Just make sure that the mix control of the delay is set to 100 percent wet, so that no dry signal comes through the Buss object. (Although Logic's plug-in delay compensation should prevent phasing problems, any dry signal coming through the Buss object will still mess with your mix's vocal balance.) With the routing set up, you should be able to fade up the send level on the vocal Track object and hear the delay effect appear. You'll need to decide on the delay plug-in's settings at this point, but don't worry about setting the return level properly for the moment. The next stage is to set up the dynamic element of the effect. Insert a Compressor plug-in into the Buss object after whatever delay plug-in you've chosen. In the plug-in window's pull-down Side Chain menu select Track 1, so that the level of vocal signal will control the compressor, not the level of the Buss object's delay effect. For the moment, set the Ratio to 2:1 and the Release control to about 150ms, leaving the Attack control at its default zero setting. Now switch of the plug-in's Auto Gain function and adjust the Threshold parameter until you get the desired modulation of the delay effect — you'll need to tweak the Buss object's fader level in tandem with this to get the delay sitting correctly in the mix. As a rough guide, I got the kind of effect I was after with the gainreduction meter showing 8-10dB of reduction while the vocalist was singing.

The Klopfgeist Challenge Perhaps as a reaction to the increasing complexity of their sequencer, a group of Logic users from the Sonikmatter forum (http:// community.sonikmatter.com) decided to go back to basics by launching a competition to create an entire track using only Logic's built-in metronome instrument Klopfgeist as a sound source. Entrants were allowed to use up to 16 instances of the metronome, in addition to around 30 basic plug-ins. Despite these restrictions, a short list of six tracks was eventually posted for voting purposes, and you can have a listen to them for yourself by surfing over to http://ttsh.co.uk/klopfgeist/tracks. If you fancy taking the Klopfgeist challenge for yourself, we at SOS would love to hear the results — send them in to [email protected] and we'll post a link to your file so that other Logic Notes readers can see what's possible.


Logic Notes

Dynamic Reverb Here's another nice stunt you can pull by controlling a reverb dynamically, but this one works in the opposite way. It's not uncommon to have your vocal mixed pretty dry during its quieter moments, but to then add more reverb as the performance hots up, and you can set Logic up to adjust reverb levels like this automatically. The first thing to do is to create a new track in the Arrange window, and to put an Alias of your vocal region into it. Once you've assigned a new Track Audio object as the Track Instrument, you should get the identical vocal signal appearing on both channels. Unassign the output of the duplicate track, so that it doesn't directly influence your vocal mix level. Your aim now is to use plug-ins to make the dynamics of the new track more extreme than those of the original, and then to feed your chosen reverb from this track. If you're already using compression on the main vocal, then it's easy — just leave out the compression on the duplicate track. However, if you're not after a compressed lead-vocal sound, or if the audio file you're working from is already compressed, then you need to expand the dynamics of the duplicate.

By placing an Expander plug-in before your vocal reverb, you can keep the sound drier in the quieter passages than in the louder passages.

As a starting point for doing this, to insert an Expander plug-in into the duplicate track, setting the Ratio to 0.5:1, the Attack and Release times to 20ms, and the Peak/RMS setting to RMS. Now play the track and compare the track level meters while you adjust the Threshold control. If you can get the meters of the duplicate track to be higher than those of the original during the loudest moments, and otherwise below them, then your settings will be in the right ball park. Now you can send from the Duplicate track to your reverb and set its return level to taste. If you find that the modulation is too dramatic, then head for the Expander plug-in's Ratio control — as you move this towards 1:1, the dynamic changes will get more subtle.


Logic Notes

Logic Tips When you want to transfer the audio setup from one Song to another, simply importing the Environment's Audio Layer is not enough. That will get you all the objects, but not their settings and plug-ins. For that you will also need to transfer the Audio Configuration. Use the main Audio menu to open the Audio Configuration for the source Song, then copy its contents to the clipboard (Apple +C), open the Audio Configuration window for the destination song, and paste (Apple+V). Len Sasso While editing a live recording of a church service, I noticed that edit points placed during gaps in the proceedings caused the background noise to dip noticeably. This turned out to be because I was using the default audio crossfade curve, as I would normally do when attempting tracking tasks such as vocal comping. Switching instead to the EqP setting (by selecting it from the drop-down menu in the Parameters box) solved the problem and made the edit points inaudible. Mike Senior There are Global Preferences for limiting dragging to one direction in the Arrange, Matrix Edit, and Score windows. Whichever mode you've enabled, however, holding Shift while dragging with the cursor will invoke the alternate method. Len Sasso If you've assigned the number keys for opening various windows (as Logic does by default), you can quickly open those same windows as floating windows by adding the Alt key to the Key Command. For example, Apple+Alt+1 is the default Key Command to open the Arrange window as a float. Len Sasso

De-essing Your Reverb Finally, here's a little tip for when you're using bright reverbs with vocals. You know the problem — those 'S' sounds hit the reverb like Indiana Jones' worst nightmare! The accepted wisdom on how to get around this problem is to use a de-esser on the feed to the reverb, but the De-esser plug-in in Logic is a little too tame for this purpose — even if you crank it as far as it will go, a lot still gets through. So I use a little-known facility within the Noise Gate plug-in to duck the offending 'S' sounds more severely. Here's how you do it. First, insert a Noise Gate plug-in before the reverb plug-in you're using. Adjust the Attack and Release times to 1ms, the Hold time to 50ms, and Low Cut to around 7kHz. With the Range parameter at about -20dB (enough to hear the action of the gate), adjust the threshold so that the gate only opens on the 'S' sounds. Once you get the Threshold pretty much right, fine-tune the effect using the Low Cut and Hold parameters. Finally, change the range setting to +20dB, and the Gate will now act in reverse, ducking all the sibilant consonants by 20dB. The sound of the reverb feed itself will be pretty awful and lispy, but you'll not notice anything amiss once the signal's been through the reverb and is combined with the dry signal again.


Logic Notes

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

In this article:

Sources & Destinations Essential MIDI Tips Recording & Editing MIDI Operations

Current Versions

MIDI Sequencing In Pro Tools Pro Tools Notes Published in SOS October 2004 Print article : Close window

Technique : Pro Tools Notes

6.4cs7: HD, Accel and LE systems on Mac OS X, LE systems on Windows XP. 6.4cs6: HD & Accel systems on Windows XP. 6.4cs1: Older TDM systems on Mac OS X and Windows XP.

While Pro Tools is widely known for its dominance in the field of computer-based audio recording, little is made of the fact that it also features fully fledged MIDI sequencing facilities. Simon Price

Pro Tools' MIDI features are not as extensive as those of Logic, Cubase, DP or Sonar, but they certainly don't deserve the derision they often receive. You often hear sentiments like "Yeah, but MIDI in Pro Tools is crap isn't it?", or similar, from people who've probably never actually used it. So where does this particular piece of discussion-board lore come from? Well, it's probably fair to say that MIDI in Pro Tools was 'pretty crap' a few years ago. However, Digi have tried to make sure that Pro Tools now does everything it needs to do for the majority of MIDI users. It doesn't do score editing or anything fancy like audio-to-MIDI conversion, but otherwise it has everything covered. Many MIDI packages have several environments for displaying and working with MIDI, but Pro Tools has just one. Recording and editing sections of MIDI is handled in a very similar way to audio. MIDI tracks are created from File menu's New Track command, and each track always exists as both a record/edit lane in the Edit window, and a fader strip in the Mix window. Like audio tracks, MIDI tracks have input and output selectors in the Mix window, except that this time they are use to choose source and destination MIDI devices. Just like audio, MIDI is recorded directly into the Edit window, with all further editing and arrangement happening here. This is consistent with Pro Tools' philosophy of not having special edit windows for different tasks. If you need to fine-tune something, you can zoom right in and adjust individual elements. The theory goes that it's good to view whatever you're editing alongside other

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elements, and see MIDI notes displayed with sample accuracy against audio tracks.

Sources & Destinations Although it's possible to draw MIDI data into tracks using the pencil tool, chances are you will be connecting a MIDI keyboard to record with. This means that even if you aren't actually using any hardware MIDI sound sources, you'll need to get to grips with setting up MIDI I/O for the computer, if you haven't already. Pro Tools doesn't communicate directly with MIDI devices, instead relying on the computer's OS to handle the low-level I/O. On Mac OS X, MIDI interface and instrument communication is handled by the built-in Core MIDI services, using the Audio MIDI Setup utility to add and configure devices. On the Windows platform the Multimedia Control Panel performs the same task as AMS. If you're still on Mac OS 9, you have to use the add-on OMS (Open Music System) extension, with its OMS Studio Setup utility. Once a MIDI instrument is connected and configured under any of these platforms, it will be available as an input and/or output in Pro Tools. If you connect a MIDI keyboard for input, you need to open the Input Devices page from the MIDI menu, and tick the relevant box to enable your device. You will then be able to choose it as the input for a MIDI track. If you then record arm the track, incoming MIDI data should light up the track's meter. The list of devices accessed from a MIDI track's output selector (see screen, right) contains three types of destination. The first category covers the traditional hardware devices connected to the computer, such as your main keyboard if it has builtin sounds, or any other sound module, drum machine or sampler. Then you have software instrument plug-ins running in Pro Tools. Whenever you insert an audio instrument plug-in in a track, a virtual MIDI input port is created for it, and this shows up as a destination in MIDI tracks. Finally, MIDI tracks can be routed to modules running in any slaved Rewire applications. In the screenshot you can see that devices in Reason's rack appear with their correct names, due to the wonders of MIDI tracks can be routed to Rewire 2. Note that unlike some other software, hardware devices, soft synths such as Logic, there are no special tracks for audio or modules in slaved Rewire instruments. Instead, these run like normal plug-ins applications. on any standard audio or Aux Input track, so you end up with two tracks in the mixer relating to an audio instrument: one audio track for the plug-in, and one MIDI track to control it. This can make things a little confusing, with the faders and panners on both channels having similar effects. Obviously, if you route hardware sound sources through Pro Tools this same situation applies.



The 'p' button on MIDI tracks provides a patch selection facility, meaning you won't have to set up the sounds on your MIDI instruments each time you come back to work on a session, as patch change messages will be sent automatically. This is more useful for hardware instruments, though, as not all software destinations will respond to Bank and Patch Select messages. The patch list contains a default list of patch names, but it's possible to tailor this to reflect the actual patches in your instruments. The method of doing this varies by OS.

Essential MIDI Tips A couple of Pro Tools tips are particularly relevant when working with MIDI. The first is to get to grips with Relative Grid mode. Pro Tools' Grid Edit mode defaults to Absolute Grid mode, which means that when you pick up and move an object it snaps to the nearest grid boundary. By clicking and holding on the Grid button you can switch to Relative mode. Regions/notes will now be moved in steps corresponding to the current grid value, keeping their initial relationship to the grid lines instead of snapping to them. The second thing you can do is to make Pro Tools' playback and looping behaviour act more like those of other sequencers. The key to this is turn off Link Edit and Timeline Selections in the Operations menu (also available as a button in the Edit Window), and also turn off Timeline Insertion Follows Playback in Preferences. You can now set permanent playback and loop points in the time ruler, and these will remain independent of any selections or operations you make in the tracks.

Recording & Editing Recording MIDI is basically the same as recording audio. Record arm the track (s) you wish to record on to, and chuck Pro Tools into Record. There are some extra options over audio, however. The first is that you can punch into Record on the fly at any time, without needing to engage Quick Punch mode beforehand. The second is the ability to choose between Merge and Replace when overdubbing. In Replace mode, recording over previous regions in a track behaves the same way as it does on audio tracks (with some variation in the way that the resulting regions are handled). In Merge mode, any new notes that are recorded are added to those that were already there. In addition to the normal view modes you'd get in an audio track, such as different automation graphs, MIDI data can be displayed in a number of different ways. The two main views are Notes and Regions. Notes view simply displays a traditional 'piano roll'-style representation of all the data in the track. Regions view behaves more like an audio track, with data grouped together in chunks. Initially, these regions of MIDI are based on the original recording times, but they file:///H|/SOS%2004-10/MIDI%20Sequencing%20In%20Pro%20Tools.htm (3 of 5)9/26/2005 12:38:49 AM


can be chopped up and manipulated using exactly the same methods as with audio regions. The right-hand side of the Edit window has a MIDI regions list below the familiar audio regions list. As you can see from the Editing note velocity in Velocity view. screenshots, the view setting affects the way that edit operations work. In Regions view the edit tools operate on chunks of the track in the same way as MIDI data can be viewed with audio in numerous different tracks. In formats within a Pro Notes view, Tools track. however, the Grabber tool can be used to pick up and move individual MIDI notes. The Grabber can select several notes by clicking and dragging to lasso around them. The Trimmer also behaves differently in Notes view, adjusting the lengths of specifically selected notes instead of regions. Again, unlike many other sequencers there are no separate windows for editing MIDI regions, so you have to work slightly differently. You are probably used to double-clicking a MIDI section in your sequencer's arrange page and swapping to a matrix/piano-roll or score page. In Pro Tools the equivalent would be to select the MIDI region and press the 'E' key, which will zoom the selection to fill the screen. You can then use the '-' (minus) key to toggle the track between Region and Note displays. After editing, you'd then hit the 'E' key again to zoom back to the original view. As well as notes, any other type of MIDI data can be displayed and edited in the tracks. In Velocity view, note velocity is displayed as a vertical 'stalk' superimposed over the start of each note (see screenshot above). Indivdual stalks can be adjusted with the Grabber, or you can draw an overall velocity graph with the Pencil tool, which the stalks will all snap to. When you click on an individual note, the velocity stalk that belongs to it is highlighted, which is useful when notes are very close together and you can't tell the stalks apart.

MIDI Operations



So what about quantising and all that? Pro Tools provides a host of automated MIDI data manipulation tools, all listed in the MIDI menu, and grouped together in the floating MIDI Operations window. Obviously Quantize and Groove Quantize are the two that will see the most action. Groove Quantize templates are provided that emulate several other MIDI packages and devices, and you can also use groove templates that you have extracted from audio recordings using Beat Detective. Quantising can be controlled in some sophisticated ways, such as being able to exclude notes within a specified distance from the grid, as well as the strength (amount) of correction. A persistently irritating and mysterious omission is Pro Tools' inability to create groove templates from MIDI tracks. The ability to do this from audio tracks is little consolation to LE users, who don't get Beat Detective. One feature worth knowing about, though, is Restore Performance. This is a safety command which can revert a MIDI region to its original recorded state, reversing the effects of other MIDI Operations commands. Published in SOS October 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



PC Music Shareware Roundup

In this article:

Shareware Sources Audiomulch Fractal Tune Smithy Voxengo Deconvolver Voxengo Impulse Modeler Exact Audio Copy

PC Music Shareware Roundup PC Musician Published in SOS October 2004 Print article : Close window

Technique : PC Musician

Shareware might be cheap, but it can also make you very cheerful. We round up some of the best on the current PC music shareware scene. Martin Walker

Back in SOS July 2004 I rounded up some of the best freeware available for the PC musician, but mentioned that if you were prepared to indulge your credit card a little there was lots more low-cost software available as shareware. The difference between freeware and shareware is fairly obvious — developers want you to pay for the latter — but the difference between shareware and full commercial software releases is becoming increasingly blurred.

You can't always judge a book by its cover, and this is definitely true of Audiomulch, whose 'interactive musician's environment' is incredibly flexible and user-friendly for anyone who wants to create new sounds.

Strictly speaking, shareware is copyrighted software that you can try before you part with any money and, unlike the demo versions available from commercial developers, you can normally download the full version to try out. Some shareware authors let you download the full version with no restrictions and trust you to be honest enough to make a small donation if you carry on using the software. However, most let it run for 30 days before timing out, or provide a slightly reduced feature set, occasional bursts of low-level noise, 8-bit rather than 16-bit or 24-bit support, or add increasingly frustrating nag screens to encourage you to do the right thing. Whatever the approach, once any trial period is over the author expects you to make a donation or pay a registration fee to carry on using the software, although

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this fee is generally a lot lower than the cost of commercial releases because shareware authors' overheads are much lower — generally no commercial premises need to be rented or teams of staff paid, for instance. In return for your registration, you will normally get valuable technical support by email, extra documentation, some free updates in the future, and often a registration number that uniquely unlocks extra features in the software (or removes any trial limitations such as those just mentioned). There's a lot of PC music shareware out there and this month I'm going to round up some that I've found to be most useful.

Shareware Sources Most of the advice I gave in SOS July 2004 about where to look for freeware applies equally well to shareware. Indeed, there seem to be very few web sites that specifically cater only for the latter. Of course, once you open out your search from music-related software to PC shareware in general there are lots of new sites worth exploring. CNet's Download.com (www.download.com) is one of the most obvious, with 12 download categories, while WindowsPC (www.windowspc.com) provides details of PC shareware and freeware in 26 categories, including Multimedia Tools, which itself includes 191 items spread over 92 pages. SourceForge.net (http:// sourceforge.net) is the world's largest Open Source software development web site, and if you click on their software map link you'll find tens of thousands of projects in 18 categories. However, they are well organised into sub-folders so it's easier to find items of interest than you might expect. The main problem with such huge sites is tracking down the most appropriate shareware for your purpose, especially when there may potentially be dozens of software items with a similar goal. I do my best to alert SOS readers to any of particular merit and some sites, such as Download.com, do also provide occasional reviews, but for my money the most helpful site is ZD Net (www.zdnet. com), whose downloads area not only hosts 14 categories (each with further subdivisions), but also features lists of the most popular downloads and the most talked-about downloads, and offers product spotlights, as well as letting you sort site contents by Name, Date Added (if you want to rule out old Windows 95 utilities, for instance), User Rating (to discover what other people think), and number of Downloads (their ultimate vote).

Audiomulch Audiomulch already has a cult following, includes among its users Kieran Hebden (aka Four Tet — you can read our interview with him in SOS July 2003), and is one of the only PC-only applications I've come across, apart from Gigastudio, that has Mac users drooling. So what's all the fuss about? Well, Audiomulch is described as an 'interactive musician's environment', will run on nearly all versions of Windows including 96,



98, ME, NT 4.0, 2000 and XP, and supports soundcards with ASIO, MME-WDM and DirectSound drivers. The version I downloaded was 0.9b16 (it's still officially a Beta release), although Australian Ross Bencina has been developing Audiomulch for a total of five years. Each released version is a free download and not restricted in any way, but times out on a certain date, encouraging you to download a more recent one. However, if you register, for $50, the shareware warning messages disappear, versions no longer expire, and you are eligible to download all future versions up to (but not including) 2.0. With Audiomulch you can either treat sounds coming through the input of your soundcard, treat an existing audio file, or generate audio files from within the application. You then pass them through a network of 'Contraptions' (what a lovely choice of term) that constitute a 'Patch', strung together using virtual cables, before sending them to the output of your soundcard or exporting them as a WAV or AIFF file with a bit depth of up to 24-bit and sample rate up to 192kHz. Audiomulch is so simple to use that you almost don't need a manual, although the detailed help file is wellwritten and informative. Essentially, the user interface is divided into three 'panes'. At top left is the Patcher pane where you use a right-click menu to select your chosen Contraptions, and a left-click to select them, drag them about, and attach patch cords to string them together. Most Contraptions have a 'Property Editor' (essentially a front panel featuring knobs and sliders) that Despite its toy-like appearance, Fractal Tune Smithy is a surprisingly versatile MIDI utility can be opened in the Properties pane to suit both the dabbler and the academic. It at top right. The third and final pane is can help you incorporate more exotic for Automation, where you can alter instruments and tunings into your music, as any front-panel parameter over time. well as generating plenty of new ideas. Each parameter you decide to automate appears in this pane as a graphic vector lane. You can click anywhere on its timeline to add further points to it and drag them backwards and forwards, up or down. Apart from the automation, we could still, so far, be talking about DIY synth/effect designers such as NI's Reaktor or Tassman from AAS, but the huge difference is that Audiomulch is optimised for live performance. You can get flexible and rewarding results with just a handful of Contraptions, but, best of all, you can repatch them or introduce new ones and connect them to the existing setup during playback without skipping a beat or disturbing the flow of audio in any way. For this reason, Audiomulch is also perfect for sound designers, since it makes it possible for them to generate and capture continuous, evolving soundscapes and snip the best bits out afterwards — just the thing when inspiration strikes!



The Contraptions themselves fall into seven categories: Input/Output, Signal Generators, Effects, Filters, Busses, Mixers, and Beta (new and experimental ones that may not yet be completely bug-free). Input/Output: This section is for patching in your soundcard's in and out, and caters for multi-channel devices via its auxiliary ins and outs. Signal Generators: This is where things start to get more interesting. There's the '10 Harmonics' additive oscillator, a comprehensive dual-oscilllator arpeggiator, a monophonic Bassline synth, Bubbleblower sample granular synth, a sample-based drum machine, looping Fileplayer for replaying existing audio files, Loop Player for doing the same thing but synchronised to MIDI clock, the weird, infinitely-sliding Risset Tones generator, and the simple sine/ noise Test Gen. Effects currently comprise the bit-depth and decimation options of Digigrunge, the DL Granulator delay-line granulator, a clutch of more traditional Flanger, Nasty Reverb, Phaser, harmonic Shaper, Ring AM and stereo delay options, plus the more intriguing Pulse Comb (a sort of delay line where each repeat has its own envelope), and my favourite — SSpat, a stereo spatialiser that allows you to set the path and trajectory of a moving input signal. Filters: There are only five of these, including traditional mono and stereo parametric EQs, but on offer are some interesting options, including 5 Combs, which adds resonating chordal drones; Risset Filters, which provides a bank of up to 50 moving band-pass filters for an intriguing range of possibilities; and Nebuliser, another granular synthesis variant that passes each grain through a band-pass filter. Busses: This section contains Contraptions that conveniently combine up to 12 mono or stereo signals, although I also found I could patch multiple signals directly into any Contraption's input or output, which makes it easy to patch as you go without interrupting an existing signal path. Mixers: The selection here offers various permutations of mono and stereo mixer with added level and pan controls, plus a selection of Matrix Contraptions for dynamically switching routing between various input sources and output destinations, crossfaders to move smoothly between them, and Inverters and Gain controls. Beta: Amongst the current offerings here is a useful Frequency Shifter, the currently semi-impenetrable 16-voice Cannon Looper, the 16-track Live Looper, and the many permutations and complexities of South Pole, a four-pole resonant filter with multiple envelope followers and LFOs. VST plug-ins and instruments are also supported, so you can patch any of your existing collection into an Audiomulch design. A new Contraption will appear with the appropriate number of inputs and outputs, and you can launch the plug-in or file:///H|/SOS%2004-10/PC%20Music%20Shareware%20Roundup.htm (4 of 11)9/26/2005 12:38:54 AM


instrument's normal interface in Audiomulch's Property Editor pane. You can also sync Audiomulch to an external MIDI sequencer, and allocate Contraption parameters to MIDI controllers. So you really do get the best of all worlds — the free-form 'patchability' of the Audiomulch interface, the ability to use within it any combination of its own Contraptions and your existing VST collection, and the added flexibility of running it alongside your existing MIDI + Audio sequencer. The graphic interface won't win any awards for its beauty, but is extremely functional and easy-to-use. The program also has never crashed on me. Compared with a standard MIDI + Audio sequencer or pre-mastering application such as Wavelab, some of the features I found invaluable were the ability to set up several parallel chains of effects on one source, for more complex treatments, and the ability to add several generators at different points in the chain, introducing subtle background sounds to complement the main one. With sufficient thought and some lateral thinking you could probably use your favourite sequencer to set up some of the routings I tried, and automate some of their parameters in similar ways, but with Audiomulch it's just so much easier and such a lot more fun — plus you never have to stop playback while you patch in further Contraptions and alter routings. The word that kept popping into my head all the time I was using Audiomulch was 'freedom'. I found this program great for creating evolving soundscapes, background drones and ambiences, and it's also one of the cheapest ways to explore granular synthesis. Anyone who creates new sounds destined for sample CDs should definitely have this in their software collection. Audiomulch is also being used for avant-garde and experimental composition, and live performances of loop-based techno and trance, either as a self-contained environment or in conjunction with other software. No wonder it's gained a cult following. I'm taking out full membership immediately!

Fractal Tune Smithy Ever stuck for inspiration? There have been various MIDI-based musicgenerating applications launched over the years, and many MIDI + Audio sequencers also have integral tools for modifying MIDI data, such as the old Interactive Phrase Synthesiser of Cubase VST. Fractal Tune Smithy (www. tunesmithy.co.uk) is a music generating program that runs both as a stand-alone utility on any version of Windows from 95/NT onwards (including XP), and as an MFX plug-in. According to its advertising material, it "plays tunes as intricate as snowflakes", which is certainly an intriguing prospect — but FTS goes a lot further than generating random sequences of MIDI notes. The most important additional feature is its ability to retune any MIDI synth to a completely new scale tuning. It does this using standard pitch-bend data to create notes that sit



'between' those of normal equal-tempered tuning. There's a huge number of scales to choose from. The default list contains 32 options, including standard 12-tone equal temperament, plus 15-tone to 31-tone equal tempered, just temperament, and various ethnic and folk tunings, such as Gamelan from Indonesia and Java, Japanese Koto, West African Xylophone and Indian Raga. Further drop-down lists cover historical and modern twelve-tone scales, bagpipes, Idiophones, and even wind chimes. So if you want to play a keyboard in the authentic tuning that Bach or Mozart would have used, you can. The handy Play button above the dropdown Scale menu auditions an octave's worth of your selection, and you can also play in real time from your keyboard in both monophonic and polyphonic modes (this is termed MIDI Relaying). The MFX version lets Sonar and Cubase SX users patch FTS in as a real-time effect; you could use MIDI Yoke or a similar utility to do the same in other sequencers. Various note mappings are used to Anyone with a convolution reverb that make it easier to play unusual scales accepts WAV impulse responses should investigate Voxengo's Impulse Modeler if from a normal keyboard. For example, they want to design their own acoustic the Japanese koto has just six notes spaces. per octave, and these have been mapped onto the white notes, with the black notes generally identical in tuning to the next highest white note. Within a few seconds I was playing what sounded (to me!) like authentic Japanese music. Of course, it helps if your MIDI synth is playing a suitable sound, and while FTS can choose these for you using General MIDI, you'll get much better results by skipping these automatic choices and allocating higher quality voices by hand at the synth or sampler end. Developer Robert Inventor (aka Robert Walker) has obviously used FTS with more up-market gear, as his web site provides details on how to use it in conjunction with Gigastudio. MIDI merge from multiple input devices is also supported, as are multiple output devices so that you can route different FTS channels to different synths or samplers. Since music makes more sense if you stick to recognised scales, arpeggios or modes, along with the desired tuning, the drop-down Arpeggio box lets you choose from shedloads of options (a couple of hundred, at least). Along with the better-known options, such as Major, Minor, Diminished, Whole Tone and Diminished Seventh, there are many more exotic modes containing two, four, five, six, seven, or eight or more notes per octave. To give you an idea of the available scope, a few examples selected at random are Messiaen Truncated Mode 5, Bi Yu (China), a set of Raga options, and 'Half-diminished Bebop'!



The next stage in the FTS process is to choose a Musical Seed — a sequence of intervals, plus a note duration, which provide the starting point for generating tunes. There are plenty of examples available and you can even create your own in a variety of ways, from typing in a set of numbers with spaces between them, to playing them in as notes from a MIDI keyboard, or using the mouse and PC keyboard. The final stage in the journey is the Fractal Tune — a combination of Scale, Arpeggio and Musical Seed, plus a choice of one or more instruments across multiple MIDI channels. Using the Play button now generates continuous, everchanging tunes based on your settings. If you want impressionistic flurries and cascades of notes, try 'rushes blown in a storm' or 'echo effects in rests'. More extreme examples include 'Fibonacci rain shower', the unsettling 'Paleolithic field recording', and 'bird calls with Afro-Caribbean percussion'. Don't go away thinking FTS can only generate avant-garde meanderings for classical and jazz buffs. Although many of the offerings are 'off the beaten track', they may still inspire new songs, while others, such as 'string quintet' and 'shakuhachi and koto' are gently melodic, and still others (such as 'resting in the shade') create floating backdrops. You can also explore the more rhythmicallybased offerings, such as the improvised 'percussion medley' and 'non-repeating bongos'. The best way for any new user to get creative is to take advantage of the 'Randomise Now' key, which randomly chooses a new set of parameters each time you press it. Once you hear something that takes your fancy you can start fine-tuning it in every sense, and save the honed and polished version as a Tune Smithy file. You can also capture the FTS MIDI output to incorporate in your work, or record its audio output as a WAV or other audio-format file, It's even possible to make a musical 'e-card' with embedded MIDI data, to send as an email. Even after quite a few hours delving into this Tardis of a utility, I felt there was still lots more to explore, and while this does make Fractal Tune Smithy rather overwhelming at first sight, you're unlikely to get bored with it. The HTML help-file is a mine of information about the various features, scales and arpeggios, and there are also 75 'Tips of the Day' that point out things you may not have discovered for yourself. Whether you want to while away an afternoon generating new ideas, modify existing MIDI files, attempt to play exotic instruments with unusual tunings and scales, or explore the more academic and mathematically-based composition techniques, Fractal Tune Smithy can help. While its colourful graphics do make it look initially like a toy, this is a serious tool that's capable of a wide range of musically useful results if you're prepared to spend some time with it. It does support skins for those who prefer a more subdued appearance, but once I'd heard what it could do I didn't really mind what it looked like. Just download the 3MB demo file, complete with 1MB of help, tutorials, and FAQs, and decide for yourself. Depending on what features you want to use, there are various file:///H|/SOS%2004-10/PC%20Music%20Shareware%20Roundup.htm (7 of 11)9/26/2005 12:38:54 AM


payment levels, ranging from $14 to $45.

Voxengo Deconvolver If virtual acoustic spaces don't float your boat, you may be more interested in capturing the sound of a real one, or that of some existing hardware gear or software plug-in such as a reverb (including, dare I mention it, other convolution reverbs), or indeed any effect, preamp, guitar amp, amp simulator or other device that you can pass a test signal through. Voxengo's Deconvolver is a simple utility with a tiny $23 price tag. It can generate suitable sweep-tone source signals that you send to the input of your test device (or to a It may not look very exciting, but Voxengo's loudspeaker in the environment you deceptively simple Deconvolver utility lets want to capture). You then record you capture the sounds of real acoustic spaces (as well as of audio hardware and the output from the device (or the software) for use with playback-only sound of the space via a convolution reverb plug-ins. microphone) using any software application or hardware recorder. Those involved with film or other location work could use these tones to quickly capture live acoustics for post-production use. Deconvolver then provides an easy way to recover the impulse response from these field or device recordings, and accepts them in a wide range of input sample formats. After calculating the impulse responses, it normalises them to a peak level of -0.4dB and can then write them in a wide range of formats, from 8-bit to 32-bit and any sample rates, ready for use in any convolution reverb that accepts WAV files. A Minimum Phase transform option is provided that sometimes creates bettersounding impulses of devices such as loudspeakers and amplifiers (you may otherwise get some pre-echoing), and there's also a Reversed test tone option that can work better with noisy acoustic recordings and hardware with a limited bandwidth. Since most experimenters end up with lots of recordings made in the field, Deconvolver also provides batch processing to convert them all in one go. There are a few caveats to getting good impulse responses — such as making sure there is some silence at the beginning and end of the original recordings — and you'll probably end up creating a few before you get the hang of things, but I found that Deconvolver worked very well indeed. For the best results you should top and tail any silences from the final responses, although Christian Andersch of Noisetime has even developed a handy stand-alone batch-file utility that will do this for you automatically (www.noisetime.com/silrem.html), as well as normalising files to any peak value. By the way, anyone interested in buying Voxengo's Pristine Space would do well



to consider the Convolution Suite, which bundles Impulse Modeler and Deconvolver for a total price of $159, a saving of about $43.

Voxengo Impulse Modeler I make no apologies for including a convolution reverb section in this shareware roundup as well as in my recent freeware one. A huge number of PC musicians are investigating or already using these plug-ins to provide realistic acoustic spaces. On the shareware scene the most popular convolution reverb playback plug-in must be Voxengo's Pristine Space (www.voxengo.com). John Walden gave this a thumbs up in last month's SOS review, but a few days after that issue went to press a newer version (1.2) was released. New features include support for multichannel and AIFF impulse files, faster loading and recalculations, automatic resampling if any loaded impulse differs in sample rate from that of the host application, new search path settings to locate impulse files that have been moved on your hard drive, and some tidying up of channel and slot options. Like John, I'm impressed with Pristine Space, even when compared with the Waves IR1 running the same impulse responses. However, unlike IR1, which comes with a world-class two-CD IR library, Pristine Space doesn't have a bundled library of its own, although Voxengo do have two other plug-ins in their range that help considerably in this respect. One of their very first releases was Impulse Modeler, which I briefly mentioned way back in PC Notes October 2002. At the time, its ability to design acoustic spaces using a simple graphic plan view and then generate an impulse response from them was useful for users of Sony's (then Sonic Foundry's) Acoustic Mirror and Samplitude's Room Simulator. Now its potential user base has also expanded to PC users of SIR, Pristine Space, Waves IR1, and indeed any convolution reverb plug-in that accepts impulse responses in WAV format. The latest version of Impulse Modeler (1.7) is also quite a bit more sophisticated, although the principles remain the same. First you define the end of a solid wall by dropping a 'vertex', then you drop another elsewhere to define the other end, and finally you choose a material — a pre-defined selection of these is already available and includes carpet, concrete, drapery, glass, gypsum board, plaster on lath, and plywood panelling, or you can create your own. You can drag the vertices around to extend or move your walls. By default, the workspace is 5x5 metres, but a Scale slider lets you immediately make the whole design bigger by up to a factor of fifteen, or smaller right down to a hundreth of its nominal size. The surrounding air or other medium can also be defined — so you could, for example, simulate underwater acoustics. Once your floor design is complete, you then add one or more 'emitters' with variable direction and dispersion characteristics to act as sound sources (you can file:///H|/SOS%2004-10/PC%20Music%20Shareware%20Roundup.htm (9 of 11)9/26/2005 12:38:54 AM


think of them as loudspeakers), and 'Recepting' walls that act like microphones to pick up the sound of your 'room'. Finally, you can generate your impulse file in WAV format at a sample rate of anywhere between 44.1 and 192kHz, and a bit depth of between 8 and 32 bits. Impulse Modeler calculates all the sound reflections until they decay to a very low level, which, depending on the complexity of your design, may take quite a few seconds, or even minutes. Admittedly, Impulse Modeler only works from a two-dimensional ground plan, but even so the results can be as dense and believable as a real three-dimensional acoustic space. You can stick to traditional structures such as rooms, halls, theatres, and churches, but the beauty of Impulse Modeler is that you can also investigate other-worldly structures, so if you want to hear the reverb characteristics of rooms shaped like an egg, seashell, triangle, circle, coiled labyrinth or irregular cave, this is the tool for you. Despite a simple and workmanlike interface, Impulse Modeler is a serious artistic tool capable of a wide range of usable reverbs, echos and delay effects that really do sound good with any compatible convolution plug-in. It should not only appeal to game and film sound designers who need to realise the sound of an unusual structure, but also to anyone who's interested in space — the final frontier. For just $39.95, it's also extremely good value for money.

Exact Audio Copy Strictly speaking, Exact Audio Copy (www.ExactAudioCopy.de) is Cardware, since although its German author, Andre Wiethoff, does welcome any donations you care to send via PayPal, he only specifically requests registration from his users in the form of a nice picture postcard showing your home town or its surrounding countryside. However, his utility performs an extremely handy function for musicians: ripping data accurately from Red Book Audio CDs. Before you skip this section because your audio editor already provides a DAE (Digital Audio Extraction) function, hang on a minute, because Exact Audio Copy was written out of frustration with many such functions that sometimes leave your 'rips' with If you want to extract CD audio tracks rhythmic clicks and pops, due to accurately, Exact Audio Copy will live up to scratches or other CD surface its name where many others fall by the blemishes. You probably won't hear wayside. these when playing the same CD in your hi-fi, because the hi-fi employs error correction to mask all but the most severe read errors (which mute the player's output momentarily). CD-ROM drives don't implement such error correction during DAE.



Audio data is digitally extracted in chunks, and the start and end points of these are matched up to create a continuous file. Most DAE functions in audio applications extract extra data so that the chunks overlap. They can then be exactly positioned end to end so that the overlapping portions are identical. This is termed re-syncing or jitter correction, but it still doesn't guarantee that the extracted file is totally a bit-for-bit copy of the original. To achieve this you ideally need a CD player and PC application that both understand 'C2 error flagging'. Your software is then aware of any read errors and can re-read the sector in question. Exact Audio Copy offers a speedy 'Burst' extraction mode with no error correction, and a 'Fast' mode with jitter correction similar to those provided by most other applications. However, its main strength is 'Secure' mode, which can use C2 error flagging if your drive supports it (although many don't), but more often relies on its own multiple extractions of the same sector, re-reading and comparing until it's sure that the data has been read perfectly. While the aforementioned process can make it much slower than other copiers, especially when reading a scratched or otherwise damaged CD, the digitally extracted data will be as near 'perfect' as possible. You also won't have to listen through each track to make sure it's fault-free, since EAC also displays an informative 'Status and Error Messages' window after extraction has finished. This provides details of track quality (anything less than a 100 percent report means that bad sectors were found but EAC's secure mode corrected them after a re-read), plus any uncorrectable read errors despite multiple reads. During my tests with EAC I became increasingly aware that many CD tracks (even those with a reported 99.9 percent track quality) would have slipped through most DAE functions with read errors, while other extractions of tracks with bad scratches might not have been usable at all with lesser utilities. Although EAC supports CD Text and database file naming, saving of tracks with a wide variety of audio compression formats and a host of more advanced functions, it's the 'secure copy' function that will be most prized by the majority of musicians. If you want an exact audio copy, you now know how to do it. Published in SOS October 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



Running Autotune within Digital Performer

In this article:


Quick Tips Digital Performer Notes Axe Attack On DP! New Multi-band Compressor Published in SOS October 2004 Autotune Tips Print article : Close window Nautilus Bundle Floated

Technique : Digital Performer Notes

A bundle of tips helps you get the most from Autotune running inside DP, and there's plug-in news aplenty... Robin Bigwood

There have been some important developments in the DP world this last month, including new and newly-updated plug-ins that could prove extremely useful to DP users. Perhaps the most widely anticipated of these is MOTU's own MX4 'multisynth', which has been shown in pre-release versions since the start of 2004. Whilst its $295 price tag puts it amongst the most expensive soft synths on the market, stability and compatibility with DP 4 should be about as good as it gets, and it certainly looks to have some interesting features. Possibly the best of these are the multi-architecture design, with options for standard subtractive synthesis alongside FM and AM, and wavetable Guitar Rig: Be careful with oscillators. MX4 also continues the tradition of that axe Eugene. American synths of yore with its mind-boggling modulation options. Not only, it seems, are single modulation sources capable of controlling multiple parameters, but multiple sources can control single parameters — a programmer's delight. Less great, perhaps, is the processor hit, described recently by one of MOTU's head honchos as similar to synths such as Camel Audio's Cameleon 5000 — which, in my experience, puts it easily into the 'hog' category. Also, as is the way with Mach Five, copy protection is via an iLok USB dongle. While MOTU state that MX4's authorisation can be transferred to a Mach Five iLok, this approach doesn't make it easy for legitimate owners to shop out processor hit to multiple Macs as and when necessary — and, as far as I'm concerned, a single iLok is already one too many. The sound quality of the MP3 demos posted up at www.

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motu.com put MX4 on the 'digital' side of warm, but as soon as I can get my hands on a copy I'll confirm whether this is really the case or not, and report on ease of integration with DP 4.

Quick Tips DP is, and always has been, notorious for its potential window clutter. Having dozens of windows open not only makes your work environment confusing but also saps processor power. Two simple settings can help with this. First, consider choosing 'Apply Closes Window' from the mini-menus of the Transpose and Quantise windows, so that they don't outstay their welcome. Also try the 'Open one Graphic Editor for each sequence' option in DP's preferences if you work with MIDI much — the resulting single Graphic Editor, with a track list selector, is much easier to work with, I find. I mentioned Audio Hijack Pro in last month's column as a possible candidate for Bounce To Disk alternatives in DP. No sooner had I submitted my copy than Audio Hijack Pro 2 was announced, with some important new features, such as built-in CD burning and support for AAC and Apple Lossless file formats. It costs $32 from www.rogueamoeba.com, or a measly $10 for owners of AHP version 1.

Axe Attack On DP! If there's been one type of plug-in that DP 4 users have missed out on until now, it's undoubtedly the guitar effects and amp/speaker modelling combo. MOTU's own Preamp1 is useful, but it's no Pod, and despite great initial promise, IK Multimedia's Amplitube has never been a reliable match with DP 4. Enter, then, Native Instruments' Guitar Rig (reviewed in the September issue of SOS), which offers a complete guitar effects and amp modelling environment in the form of an Audio Unit plug-in. My initial reaction, when I heard about Guitar Rig, was to wonder why anyone would choose it over a hardware device such as a Pod or JStation, but having used it for a couple of weeks now I can unreservedly recommend it. Compatibility with and reliability in DP 4 is excellent, and both the huge range and sheer quality of sounds it produces makes the 449 Euro price tag (which includes a hardware controller) look nothing short of a bargain. The plug-in is useful for more than just guitar too — the huge flexibility of Guitar Rig's 'rack' architecture means that vocals, drums, piano, synths, and almost anything else, can benefit from its capabilities. The cabinet modelling, in particular, can impart anything from subtle to massive tonal changes, on a wide range of sources, and all with a distinctly 'musical' quality.

New Multi-band Compressor Another new plug-in, not quite arriving in time for my round-up of DP 4-based mastering plug-ins in the August edition of SOS, is Wave Arts' Multidynamics.



Where their Finalplug takes its cue very obviously from Waves' L1 and L2 limiters, Multidynamics looks uncannily like a Waves C4 or Linear Multiband, but with up to six independent bands of compression. Multidynamics sounds different to the Waves offerings, though, not least because it doesn't pretend to preserve relative phase between the bands, and it's definitely a case of 'different', not 'worse' — this is a seriously nice processor. Wave Arts' MAS plug-ins are amongst the most bomb-proof out there, represent good value for money, and are extremely useful on a wide range of projects. Multidynamics is no exception. Wave Arts also have excellent customer support and a sensible copy-protection policy. Multidynamics costs $150 by itself, or comes as part of the bread-andbutter Power Suite bundle for $500, from www.wavearts.com.

Autotune Tips Autotune, the automatic pitch-correction plug-in by Antares, occupies a very special position in the world of plug-ins. It can call attention to itself like almost no other, and has been heard of well outside of the world of music technology, often thanks to some ethical debate or other about modern production practices. Indeed, the mere mention of it can cause some people to get very hot under the collar, but it's a safe bet that we're all hearing it all the time, even when we don't realise we are. Autotune has always been a DP-friendly plug-in, available in MAS format and respectful of DP's conventions, and there's much to recommend it. Getting the most out of it in DP, though, isn't always obvious, so here are a few pointers which I hope both new and experienced users might find helpful. Some like it hot: While Autotune's Auto correction mode is virtually a 'fit and forget' solution for some vocal lines, it's hard to find settings that will work throughout an entire song, especially if you're trying to avoid re-tuning artifacts. Selecting the appropriate Input Type is, of course, important in helping Autotune get things right, and bypassing or removing degrees of the selected scale is vital, but you should also consider placing a Trim plug-in, or even a compressor, ahead of it in the channel's signal path, to ensure that it's always being fed a good, hot signal, particularly from tracks that are a bit weedy. Clean it up: Be aware that extraneous vocal slavers, slaps and pops can throw Autotune off the scent — knock them out using a gate or some judicious soundbite editing. Consider placing a high-pass-type EQ ahead of Autotune, too, to zap any low-frequency rumble that might have crept in to the recording. Go Graphical: If real-time correction is just not doing it, Autotune's potent Graphical mode can nearly always do the trick, as well as being the key to extreme vocal treatments such as Madonna's 'Impressive Instant'. The basic operation of this mode is well covered in Antares' documentation, but again always make sure that the signal entering Autotune is nice and hot, both during the detection and correction stages. Sometimes it's necessary to apply the same graphical corrections to a soundbite that you've used repeatedly during a song, and having to go through multiple learn/draw/correct cycles is clearly a drag. The way to fix this would be to correct just one soundbite, somehow 'render' the result, and copy it to the remaining locations, but the obvious method of doing this — applying a graphical-mode Autotune to the soundbite 'offline' via DP's audio menu — doesn't work. The simplest workaround is to first place Autotune on the relevant track, 'learn' the duration of the soundbite, and draw the necessary corrections. Then, assuming the soundbite is on a mono track, temporarily select a mono output for the track, select just the soundbite, and choose Freeze Selected Tracks from the Audio menu. This produces a new track in your sequence with the corrected soundbite in exactly the same location as the original. It can then be dragged and copied to replace the other segments that need file:///H|/SOS%2004-10/Running%20Autotune%20within%20Digital%20Performer.htm (3 of 5)9/26/2005 12:39:00 AM


correcting. Target MIDI: One of my favourite features of Autotune is Target Notes Via MIDI, in Automatic mode. To activate this you also need to record-enable a MIDI track in DP with Autotune as its output. When this is set up, Autotune 'removes' all notes of the selected scale, except, on a momentary basis, those being played via MIDI. While you can hold down chords to select multiple pitches, I find a monophonic line most useful, and this certainly opens the door to some of the most extreme Autotune treatments possible. It's also a good way to quickly correct a single note of an entire performance that is otherwise perfect, as when no MIDI notes are received by Autotune the original audio is passed without processing.

Nautilus Bundle Floated Finally, it's a very welcome hello to an old friend, Audioease's Nautilus bundle, which has just been released in a variety of formats, including MAS, for OS X. Only Audioease could offer such a disparate set of plug-ins as a bundle, though between them they cover a lot of audio treatment possibilities, and the water-themed unifying concept is nothing short of inspired. Possibly the most straightforward of the three plug-ins The River Run granular synthesis plug-in is that make up Nautilus is the delightfully one of the more unusual parts of nerdy Deep Phase Nine which, as a Audioease's Nautilus bundle. MAS-based phaser, will surely never be surpassed. The key here is flexibility, with control over the number of frequency notches (or boosts), frequency 'spread' between the highest and lowest notch, depth and speed of sweep, and convergence between left and right channels. All these parameters are presented in a very intuitive fashion, using a colour-coded animated display. Next up is Periscope, which, for me, is an indispensable plug-in. This is a real-time frequency analyser and phase-accurate equaliser in equal parts, and it excels in both areas. With Periscope, the structure of the input signal is displayed in real time on what is effectively a frequency versus amplitude 'graph', and a series of vertical sliders is overlaid on top of this. Moving a slider cuts or boosts the range of frequencies which lies beneath it, and the total range of frequencies the sliders affect can be adjusted using an additional pair of 'flags', superimposed on their own 'graph' beneath the main display which always shows the entire bandwidth of the input signal. What makes Periscope special is the surgical accuracy of its cuts and boosts, and the way in which, with the help of the visual display, problem areas of a signal can be sorted out in no time at all. I also love Periscope's ability to strip away the harmonics of a monophonic input one by one: removing and then replacing the lower harmonics of a sustained note never fails to produce a "wow." Finally, if you're interested in granular synthesis, River Run will give you plenty to work with. Audio can be streamed into this plug-in, frozen, and then smaller portions of it used as a basis for 'grain' production. Pitch can be randomised completely or 'semi-randomised' to conform to a a series of musical scales or chords. In short, file:///H|/SOS%2004-10/Running%20Autotune%20within%20Digital%20Performer.htm (4 of 5)9/26/2005 12:39:00 AM


brand-new, often unexpected and very beautiful musical structures can emerge from River Run, even with the most unpromising audio as a starting point. The Nautilus bundle costs $299 from www.audioease.com, and it's no exaggeration to say that Periscope or River Run are worth that amount by themselves. Published in SOS October 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 Mackie Tracktion Sequencer As A Rewire Host

In this article:

On The Right Tracktion M-Audio Avidly Acquired Ever-expanding Patch Quick Tips Total Control

Using Mackie Tracktion Sequencer As A Rewire Host Reason Notes Published in SOS October 2004 Print article : Close window

Technique : Reason Notes

Mackie's Tracktion sequencer makes an ideal budget Rewire host for adding audio recording and plug-in capabilities to your Reason setup, as we explain... Derek Johnson

What if I told you there was a way to add multitrack audio recording and editing, plus VST plug-in capabilities, to your Mac or PC Reason setup for the less than £50? Yes, for a lot less than the price of many commercial plug-in processors or instruments, these two most desirable facilities can be retrofitted to Propellerhead's electronic music studio. Admittedly, I'm cheating a bit by Tracktion shown in the background, hosting Reason while playing back some freshlyusing a Rewire host to do the job, but recorded audio. You can just see, to the that host must be the cheapest right, how plug-ins can be chained after currently available that offers all the Reason audio channels. basic facilities needed for the job — and it's cross-platform. My search for budget Rewire lead me to Raw Material Software's Tracktion, now being marketed exclusively by Mackie and costing a mere US$80 as a download from Mackie's web site. British readers may be aware of the US dollar's current poor performance against Sterling, resulting, as I write, in a £45 bargain. Mackie provide full support for registered users, and there's even a handy — and excellent — on-line user forum (www.kvr-vst.com/forum/viewforum.php?f=22).

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On The Right Tracktion Tracktion was reviewed in SOS back in April 2003, in the software's pre-Mackie days. Its Rewire capabilities seem to have been a little uncertain back then, but no more. This efficient alternative to the high-priced MIDI + Audio big boys hosts any Rewire-capable application with ease, and is compatible, on both platforms, with VST plug-ins and instruments. Feel free to import WAVs and MIDI files too. And if that's not enough, Tracktion actually comes with a suite of its own processors, plus a collection from shareware specialists MDA. A handful of virtual instruments are also included, and free, shareware and commercial instruments can be added at will. Tracktion's designer, Julian Storer, wrote the software deliberately to be unlike the competition. He felt other sequencers didn't work as they should to enable him to record audio (and MIDI) to his computer with as little trouble as possible. The result is a sequencer with an approach that will feel different to experienced users of standard MIDI + Audio sequencers, though it's completely logical and friendly. MIDI and audio tracks 'feel' similar in an operational way, but without confusing the two forms of data as some 'entry-level' music programs do. Think of how Ableton's Live is different from competing products, and you'll have an idea, remembering that Tracktion and Live are in no way equivalent products. As a Rewire host, Tracktion is ideal. It's efficiently programmed — no wizzy graphics or bloated code — so will run well alongside Reason on a modest modern computer. It works fine on my ageing G4 and brilliantly on my more up-to-date P4 laptop. I'm a long-time user of the Cubase family, but (especially on my G4) there are times when attempting to run it alongside Reason at the same time as trying to record anything like a busy audio session results in frustration.

It's getting pretty busy back there, and I've tidied away some cables for clarity! Here, some of the parts of last month's one-finger NNXT patch are treated by ECF42s, followed by COMP01s to keep filter-induced level problems at bay.

Of course, this solution doesn't work from within Reason, which remains as closed as it ever was, but at least it's cheap. Because of the way Rewire works, you can control the host or slave app from the transport of the other, thus you can create a skeleton arrangement in Reason and go to Tracktion to record any audio — vocals, guitars, whatever — for the length of the song, returning to Reason to flesh out the arrangement. When it comes to reorganising and fine-tuning the tandem result, edits and chops will have to be duplicated in both pieces of software. But Tracktion is so easy to use (once you've got the hang of how it does things) that this shouldn't be too much trouble. Treating Reason audio via VST plug-ins hosted by Tracktion is straightforward.



Just create more tracks and insert more instances of the Rewire 'filter' (Tracktion's generic term for plug-ins, whether effect or instrument), choosing different audio streams for each one, passing audio out of Reason via the Hardware Interface device. A final mix can be rendered to disk from within Tracktion. This is a real-time procedure, though mixes without Rewire slaves can be done in faster than real time. Tracktion saves all its audio and settings in a self-contained file (audio is datacompressed), but obviously it won't save Reason's state. So unless you're just using Reason as a sound module (this is possible via Rewire's MIDI streaming), make sure you save the last version of an edited song before closing down. So, while Tracktion may not be as cheap (ie. potentially free) as the Mac-only Garageband option I mentioned a couple of issues ago, it is cross-platform and much more of a serious composition and recording tool than Apple's offering. If you've come to computer music via Reason and want to add audio recording and plug-in capabilities cheaply, check it out.

M-Audio Avidly Acquired As revealed in SOS's main news pages this month, digital video giant Avid have effectively taken over M-Audio, the company which distributes Reason in most markets outside Sweden. Of course, Avid also own Digidesign, the high-end audio sequencing company, so the move isn't, perhaps, all that strange: Reason Adapted, a cut-down version of the virtual studio, has even been a part of Digidesign bundles. So far, the move appears to be having no impact on Reason or other M-Audio-distributed products.

Ever-expanding Patch Sometimes, a couple of pages just doesn't seem enough for a column about a program like Reason! This became particularly apparent last month when, for space reasons, I had to lose a few extra bits that took the NNXT one-finger performance patch I explained even further. As you may already know, NNXT is one of the more powerful devices in the Reason rack, and can accommodate a huge number of samples, each of which has the potential to become its own 'group' and have its own collection of synth and sample playback parameters. This aspect of the device means that it would require a MIDI spec the size of the known universe to make it possible to assign a unique MIDI controller to each NNXT parameter. For the same reason, NNXT only has a small collection of gate and CV control sockets on its rear panel. But if, when you're creating a layer such as we discussed last month, or something similar, you'd really like to be able to add MIDI, gate or CV control to a parameter or two of samples or groups in an NNXT patch — perhaps for triggered gate effects or to add some movement to a filter cutoff frequency — there's still hope.



Start by making use of the individual outputs on the back of NNXT. There are 16 stereo pairs (or 32 mono outputs if you use hard panning between pairs of samples or groups), and this facility means that you can treat almost any NNXT samples with Reason effects. To get the desired effect, why not start with the ECF42 envelope-controlled filter? You could, of course, use the filters of a Malström, as we did last month, but ECF42 uses less space and DSP resources! It's not quite as dynamic a filter as those on the synths or NNXT itself, but it is still quite capable, and two can be chained in series, or chained with other effects (such as the PEQ2 parametric EQ) to build on the sound. Patch different NNXT layers to their own ECF42, set up the filter parameters to suit, and assign triggers or variable CVs (and even MIDI controllers) to them. You'll be able to automate front-panel knobs in the main sequencer: assign each ECF42 to its own rack, then make sure each is highlighted and its track shows the MIDI icon, as you make the automation moves. And don't forget that spare modulation sources — LFOs or EG curves — produced by other devices can be routed to ECF42 CV inputs. As ever with Reason, experiment.

Quick Tips Last month, I mentioned in passing that any key on an attached MIDI keyboard could be set to start Reason playback. Of course, you can also assign the transport's 'Stop' function to a key (as well as its Record, Fast Forward and Rewind functions). Choose notes at the highest range of your keyboard if you'd like to do this, as those notes will tend not to be played too often. In any case, as long as there's no MIDI icon assigned to any sequencer track during playback, your 'trigger' notes should not end up playing devices you don't want to play. Here's an easy way to add a sophisticated contrast between low-velocity and highvelocity playing on the Reason synths and sample players. Simply take your mouse to the device's velocity routing section, and turn the A(mplitude) Attack knob fully left. Now set the Amp Envelope Attack parameter to mid-travel (a value of about 64). Now, when you play softly, not only does the synth play quietly, but you'll get a gentle bowed effect (depending on the rest of the patch's parameters, of course). Play harder and the volume rises as the attack becomes more definite. Combine with other velocity-controlled parameters for a more dynamic-sounding patch. This isn't so much a quick tip as an observation: it's possible to play Reason on a Mac, under Mac OS X at least, when the computer is asleep. The software feels even less like software if you can ignore the computer! If there's one thing I'd recommend any Mac user — let alone the Mac-based Reason user — buy to improve their overall computing experience, it would be a dual- or multi-button mouse. Simply being able to do the equivalent of a Windows right-click when adding devices to, or customising devices in, Reason's rack is a revelation when it comes to your average workflow. Use Reason Songs as patch libraries for the older effects devices that have no saving and loading facilities. If you have favourite effect chains or configurations, with useful parameter settings, simply highlight the whole chain, copy, paste it into a 'library' rack, and save it. Then just highlight, copy and paste the chain back to any Song in future.



Under both flavours of MacOS, devices that can save patches give you the option to save with or without a file extender. Choose to save with, if you think your patches might be imported in some way to Reason running under Windows, since the files won't be recognised otherwise.

Total Control Another use of NNXT's individual outputs in last month's patch could help keep distortion at bay — and this technique can be used with other devices. Yes, I know that Reason is technically not capable of distortion, but overloads do occur! When designing a heavily-filtered sound with any of the sound-generating devices, you might find that unpredictable changes in level play havoc with the Audio Out Clipping meter, and the quality of your mix. It will probably occur with the textures produced by the NNXT technique described last month. Simply place a COMP01 compressor in line with any instrument or signal path that's causing problems. Even if you don't alter its parameters, the level will be tamed somewhat and you'll be able to more easily set up a balanced mix that doesn't blow out. Doing this has the additional benefit, with NNXT individual outs, of naming them sensibly on Remix mixer channel scribble strips, via the scribble strip of the compressor to which they're connected. Otherwise, individual output pairs turn up as something like 'NNXT 15-16', which is not always useful during a big mixing session. Published in SOS October 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Sound on Sound (October 2004)

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Sound on Sound (September 2004)

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Sound on Sound (December 2005)

Sound on Sound (June 2005)

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Sound on Sound (October 2004)

Sound on Sound (November 2004)

Sound on Sound (March 2004)

Sound on Sound (January 2004)

Sound on Sound (June 2004)

Sound on Sound (February 2004)

Sound on Sound (September 2004)

Sound on Sound (July 2004)

Sound on Sound (May 2004)

Sound on Sound (September 2005)

Sound on Sound (December 2005)

Sound on Sound (June 2005)

Sound on Sound (January 2005)

Sound on Sound (May 2005)

Sound on Sound (November 2005)

Sound on Sound (January 2006)

Sound on Sound (February 2005)

Sound on Sound (July 2005)

Sound on Sound (March 2005)

Sound

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Sound

Sound Advice on Mixing

Sound Symbolism

Sound Insulation

Sound Doctrine

meyer sound design reference (for sound reinforcement)

Sound Advice on Microphone Techniques

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Secondary Sound

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