Sound on Sound (June 2004)

In This Issue June 2004 In This Issue Click article title to open Reviews Behringer ADA8000 People Business End 8-c...

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In This Issue

June 2004 In This Issue Click article title to open Reviews

Behringer ADA8000

People

Business End

8-channel Preamp & A-D/D-A Converter Music Producers Guild recording assessments Eight channels of high-spec mic preamplification, A-D/D- More constructive comments from MPG (Music Producers A conversion, and ADAT lightpipe I/O, all at a remarkable Guild) members on SOS readers' submitted recordings. price. Is it too good to be true?

Bellari RP583 Valve Compressor This new dual-channel compressor combines valve circuitry with an opto-compressor design. But does it deliver the best or the worst of both worlds?

East West/Quantum Leap Symphonic Orchestra Orchestral Sample Library It's another multi-DVD orchestral library! Does it merit a thunderous introductory timp roll or a feeble tap on a vibraslap? Find out...

Emulator X Studio 1820M • 1820 • 1212M PC Soundcard & Software Sampler Emu's new range of soundcards offers an unprecedented level of flexibility, DSP power and sound quality for the price — with the added bonus of a very impressive software sampler.

Evolution MK461C Assignable USB MIDI Controller Keyboard Evolution's new class-compliant MIDI keyboards are a cinch to use with computer-based sequencers. But do they perform as well as their spec suggests?

Korg Legacy Collection (Part 1)

Kid 606, Cex & The Tigerbeat6 Label

Tiger Tales There's some amazing music being made in bedrooms these days. And bringing it to the wider public is the job of the Tigerbeat6 label, whose stars include label founder Kid 606 and Rjyan Kidwell, aka Cex.

Leader Loneliness and the Long-distance Programmer Sounding Off Paul Wiffen reflects on the evolving role of the programmer and the lonely life of the modern studio musician.

Recording Franz Ferdinand Tore Johansson Cardigans producer Tore Johansson was thrown into unfamiliar musical territory when asked to produce the debut album by Scottish guitar band Franz Ferdinand, but the result was a commercial and artistic triumph.

Studio SOS The Arcades This month, the SOS team help The Arcades to rock even harder than before!

Thermionic Culture Valve Designs Vic Keary SOS talks to a British designer who thinks audio in the 21st century is still best served by tube.

Virtual Instrument/Hardware Controller Now that Korg's Legacy Collection is properly complete, we follow up last month's preview with the first instalment Your Correspondence of our three-part review. This month, we focus on the Crosstalk Korg Wavestation plug-in... We respond to another batch of reader emails and letters.

Latest Sample CDs

Technique

Sample Shop Check out the hottest Sample CDs...

Apple news from NAB Show

Native Instruments Intakt

Apple Notes Apple had some interesting announcements to make at this

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In This Issue

NI have taken the technology behind Kontakt, and used it year's US National Association of Broadcasters show, including improvements to the eMac and portable line-up. to build a sampler that's all about loops.

Plug-in Folder

Avoiding grief when moving PT Projects

Latest Plug-ins reviewed We test and report on another crop of highly insertable software Plug-ins: PSP Master Q Formats: PC Direct X & VST; Mac OS X VST TLL Everyphase Formats: Mac & PC RTAS & TDM Fabfilter Fabfilter One Formats: PC VST & stand-alone Cranesong Phoenix Formats: Mac & PC TDM Audiorealism BassLine Formats: PC VST Sonic Charge µTonic Formats: PC VST

Pro Tools Notes A common occasion for grief in Pro Tools is when moving projects from one system to another. Here are some tips to help you keep your hair.

Red Submarine Dual Xeon PC Redmatica EXS Manager OS X Sample & Instrument Manager For EXS24 Multi-gigabyte sample libraries offer unprecedented realism and lots of potential for confusion. Redmatica's neat OS X utility allows Emagic EXS24 users to tame their unruly sample collections.

Roland VS2000CD Digital Multitracker This new recording workstation is the most affordable and portable of the VS series, but it still lets you record, mix, and master all in the one box.

SE Electronics Gemini Dual-valve Capacitor Microphone This impressive new mic from SE Electronics uses two valves instead of the usual one. But does it actually make any sonic difference?

Tannoy Ellipse 10 IDP & TS212 IDP Digital Loudspeaker System Tannoy's latest-generation Ellipse technology has been combined with DSP processing from TC Electronic, creating a versatile and powerful high-resolution monitoring system. Competition

Beat-slicing Masterclass What software you need and how best to do it! Beat-slicing can radically expand the creative potential in your loop library — you can match tempos and key signatures, rearrange loop events, and delve into inspirational sound design. SOS looks at all the leading beat-slicing software and shows you how to get the best from this powerful technique within your sequencer.

CLASSIC TRACKS: 'Wuthering Heights' Artist: Kate Bush; Producer: Andrew Powell; Engineer: Jon Kelly Kate Bush's 1978 smash hit debut single was also the first major project Jon Kelly had recorded. It proved to be a dream start for both artist and engineer, and a perfect illustration of the benefits of working with talented session musicians.

Creative Synthesis with Yamaha XG XG Masterclass: Part 3 In this final instalment of our series of XG programming tips, we take a look at how the advanced modulation parameters can bring your layered sounds to life.

Demo Doctor Readers' Recordings Assessed Listen to the tracks while you read what the Doctor thought of another batch of lucky SOS readers' demos.

FP Numbers How do they affect your music? Many of the articles in SOS, not to mention the specifications for audio hardware and software, use the terms fixed point numbers, floating point numbers and decibels. But what do terms like this actually mean? And what are the consequences for our music?

WIN Rode K2 Tube Mic or NT2000 MultiGigging Safely With A PC pattern Capacitor Mic Sound Advice

Q. Can I 're-amp' a line-level signal? Q. Can I use three different soundcards at the same time? Q. Do I really need some 'grot box' speakers? Q. Is analogue mixing superior to

PC Musician Increasing numbers of musicians want to gig with their computers — but home PCs are fragile and laptops may not always be powerful or adaptable enough. So what are your alternatives, and what measures can you take to protect the centrepiece of your live set?

Hyperthreading & Spring-cleaning PC Notes This month, we find out whether Hyperthreading is hyper-

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In This Issue

digital summing? Q. What are the clicks spoiling my digital recordings? Q. What should I use to play my backing tracks live? Q. What's the difference between ported and un-ported monitors?

helpful to the musician and discover some new freeware. First, though, it's time to spring-clean that Windows Registry...

More Creative Synthesis with Delays Synth Secrets In the penultimate instalment of this long-running series, we delve deeper into what can be achieved with just a few delays and some creative routing...

Real-time Jam Sessions in Logic Logic Notes Learn how to set up Logic's Environment so that you can jam with other musicians in real time on a single system.

Recording A Live Choral Performance From miking to mixdown The story of a multi-miked location recording session, from preconcert setup to post-recording, software-controlled mixdown.

Synchronisation in Cubase Cubase Notes If you get that sync'ing feeling when using Cubase in conjunction with external hardware devices, you may need to know more about its synchronisation options. This month's column explains what's what.

Using your MIDI Synth to control Plug-ins Digital Performer Notes Fancy turning your knob-laden MIDI synth into a control surface for tweaking plug-in synths? DP's Consoles make this kind of application possible.

Vocal Manipulation Sonar Notes This month we investigate a new DXi option that makes vocal manipulation as easy as editing MIDI data. Plus the usual haul of Sonar power tips...

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Behringer ADA8000

In this article:

Preamp Practicalities Pro Or No?

Behringer ADA8000 8-channel Preamp & A-D/D-A Converter Published in SOS June 2004

Behringer ADA8000 £187 pros Ridiculously inexpensive. Good-sounding mic preamps. Built in A-D/D-A conversion. Balanced line ins and outs.

cons No way to route the analogue inputs directly to the analogue outputs. Level metering is pretty basic.

Print article : Close window

Reviews : Software

Eight channels of mic preamplification, A-D/D-A conversion, and ADAT lightpipe I/O, all at a remarkable price. Is it too good to be true? Paul White

summary The ADA8000 is such a useful and nicely presented piece of kit that you have to do a double take when you see the price. A useful addition to any system with free ADAT sockets!

information £187 including VAT. Behringer UK +49 2154 9206 6441. +49 2154 9206 321. Click here to email www.behringer.co.uk www.behringer.de

Photos: Mark Ewing

Behringer's ADA8000, or the Ultragain Pro-8 Digital to give it its full name, is a conceptually simple product that should interest anyone who uses ADAT connectivity within their system. Its mains-powered 1U rack format houses eight mic/line preamps with phantom power, and these feed an oversampling deltasigma A-D converter, which in turn sends eight channels of 24-bit digitised signal out of the back panel via an ADAT lightpipe connector. These preamps use the same Invisible mic preamp circuitry found in the newer Behringer mixers, and my experience of them so far has been favourable. Phantom power is switchable globally, so the preamps may be used with active DI boxes or capacitor mics that need a 48V power source, and each of the channels has a rotary gain control (with up to 60dB of gain) plus a pair of LEDs to show signal present and clipping. The line input is on a balanced TRS jack and mic/line input selection is accomplished simply by deciding which socket to plug in to. The unit also features an ADAT lightpipe input that feeds a D-A converter, where the analogue signals emerge as electronically balanced XLR outputs on the rear panel. There's a word-clock BNC input to allow control from a master clock source, and a slide switch on the rear panel sets the unit to master or slave mode (with a choice of ADAT or word-clock sync) at a choice of 44.1kHz or 48kHz internal clock rates. I would have preferred this switch on the front panel, where it would be more accessible, and I'd have liked more digital status displays, though I don't know where they could have put them. Other than the mic/line preamp connectors and controls, the only other front-

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Behringer ADA8000

panel features are a mains power switch, the phantom-power switch, and LEDs to indicate master/slave status and sync lock. There's no sample-rate indicator on the front panel, though, which is mildly inconvenient. However, the only important practical feature missing is a means to route the eight preamps directly to the eight analogue outputs, as at this price I'm sure there are countless non-ADAT users who could find a use for eight decent mic preamps in a rack. No mention of this limitation is made in the manual, and no suggestion for a workaround is provided, but the obvious solution is to use an optical cable to connect the ADAT input and output ports together. This works fine, and though the signal is making an unnecessary trip through the converters, the signal quality is subjectively very good. If a MkII version is due any time soon, might I suggest some form of switching — a DIP switch on the rear panel for example — that could be used to route some or all of the preamps directly to the line outputs without going via the converters?

Preamp Practicalities I'm glad to say that the line inputs only need to see a signal level of +6dB with a gain setting of +10dB to register a digital full-scale reading, so you don't have to drive huge signal levels into the thing to get your digital record meters up to a healthy level. On the other hand, the maximum line input level is +26dB, so if you do happen to have signal sources that deliver very hot levels, it can handle these too. The dynamic range of the system is quoted as 103dB nominal, and the frequency response is 10Hz-21kHz at a 48kHz sample rate. The analogue To the left of the XLR analogue output circuitry of these preamps goes rather connections on the rear panel is a BNC wordclock input, and a choice of digital higher than this — it's simply the synchronisation options is available from its sample rate of the converters that adjacent switch. imposes this limit. There are also no noise figures in the spec, though I don't think the manufacturers are hiding anything, as there's no audible noise when working with capacitor mics in a typical close-to-medium-distance studio-miking situation. Subjectively, the mic amps compare well to those used in better project studio mixers, and though they may not qualify as esoteric, I have no qualms about their quality and would be happy to use them in any routine application. Even if you have one or two up-market stand-alone mic preamps to handle your main inputs, a unit such as this is still ideal for those situations where you have to mic a drum kit or record several musicians playing together. In my own studio, I plan to use an ADA8000 to utilise the ADAT input port on my MOTU 828 interface, so that I can have access to more mic and line inputs over and above the existing analogue inputs. The tests I've done so far confirm that this works perfectly, with no fuss, provided that you remember to set the 828 to file:///H|/SOS%2004-06/Behringer%20ADA8000.htm (2 of 3)9/22/2005 7:42:30 PM

Behringer ADA8000

its ADAT optical sync mode. The ADA8000 would also be useful in conjunction with an ADAT recorder for very simple live recording, though you'd still need a simple eight-channel line mixer to provide a meaningful headphone mix for monitoring.

Pro Or No? If price were no object, I'd say the ADA8000 should have more detailed metering, its rear-panel sync switch is difficult to access in most rackmount setups, and there really should be some way to send the preamp signals directly to the line outputs. I might also say that having XLR line outputs is overkill on a unit of this kind where most users would probably prefer to use TRS jack cables to connect with the back of their patchbay. Having said that, price clearly is an issue, as the unit costs less than most single mic preamps in the UK, yet it does what it sets out to do extremely well. I've already bought one to add to my system, and its subjective performance is as good as from units costing several times the price. It will also save me acres of rack space! Published in SOS June 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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Bellari RP583

In this article:

Getting To Grips With The Controls Golden Tone Beauty Or Beast?

Bellari RP583 £500

Bellari RP583 Valve Compressor Published in SOS June 2004 Print article : Close window

Reviews : Processor

pros Warm, musical sound. Doesn't adversely affect transients unless you deliberately set out to do so. Flexible enough for most normal applications.

This new dual-channel compressor combines valve circuitry with an opto-compressor design. But does it give the best or the worst of both worlds?

cons The moving-coil meters mean you're never quite sure how much gain reduction is being applied to signal peaks.

summary Bellari have succeeded in blending two 'classic' technologies (optical compression and tubes) to produce a great-sounding allrounder.

information £499.99 including VAT. Smart Sound Direct +44 (0)1883 346647. +44 (0)1883 340073. Click here to email www.smartsound direct.com www.rolls.com/bellari

Paul White Photos: Mike Cameron

Outwardly, the Bellari RP583 looks like a very standard two-channel compressor. Its brushed-gold, 2U front panel sports the familiar Output Level, Threshold, Ratio, Attack and Release controls for each channel, there are two moving-coil meters (that can be switched to read gain reduction or output level), and there's a Link button for compressing stereo signals. Each channel has its own bypass button and there's an illuminated rocker-style power switch. A glance around the back confirms that life is pretty ordinary there too, except that the mains comes in via a captive cable rather than an IEC connector. There's balanced I/O on XLRs plus unbalanced I/O on quarter-inch jacks, with two further jacks per channel providing an insert point for the side-chain and also allowing the compressor to be triggered from an outside source — to produce ducking effects, for example. The construction is tidy and proficient without being overly fancy, and the use of large knobs and clear legending will be welcomed in dimly lit studios. While the outside might suggest a very ordinary compressor, the circuitry reveals a number of 'classic' ingredients, specifically an optical gain-control element, that's used within a tube circuit with the aim of providing smooth and musical gain control rather than clinical accuracy. Each channel includes a 7025 dualtriode tube, where the opto circuit comes between the two tube stages. The quoted frequency response for the unit is a healthy 20Hz-40kHz, where the balanced input impedance is 10k(omega) and the unbalanced a rather high 1M

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Bellari RP583

(omega), which means you could plug guitars and basses directly into the jack input without getting an impedance mismatch. The signal-to-noise ratio is 90dB, and the value for total distortion plus noise is less than 0.1 percent under typical operating conditions. In use, no additional noise was evident when using this unit, other than where the act of compression brought up noise already present in the input signal.

Getting To Grips With The Controls There's no in-depth explanation of the characteristics of this compressor — indeed, the manual comprises little more than a couple of lines covering each of the controls — but given that the gain control circuit is optical it's probably also somewhat nonlinear, which is what gives optical compressors their unique character. Once the channel is active, Threshold sets the level above which compression will take place, while the Output Level control is used to make up any gain lost due to the compression process. Ratio works in the usual way by determining how many decibels the input needs to rise in order to achieve a 1dB rise in output, but if, as I suspect, the opto-compressor has a somewhat softknee characteristic, then this will only be approximate. The Attack and Release controls set the time the compressor takes to respond and the time the gain takes to return to normal after the signal has fallen back below the threshold level, so there's nothing out of the ordinary there. The Link switch combines the side-chain signals and puts both channels under the control of the lower channel's knobs (other than the output gains, which remain independent) so that the same amount of gain reduction will be applied to both channels even if the signals passing through them differ (as they do when the source is stereo). This prevents the stereo image shifts that occur when stereo signals are processed via independent compressors. Inserting an equaliser into the side-chain makes the compressor frequency selective, which can be useful for setting up simple de-essing or de-popping, or other more subtle frequency-dependent effects. The insert send may also be used as an output before the Output Level control, while the side-chain insert can be used to feed in an external signal to control the compressor. The most common application of a side-chain input is ducking, where the level of voice fed in via the side-chain reduces the level of any music passing through the unit. Ducking is commonly used by radio DJs to ensure their inane chatter is heard above the music at all times, but it is also useful in the studio for pulling down the levels of instruments in the presence of vocals or solo instruments. As a rule, setting the threshold and ratio so that the ducked material drops in level by only a decibel or two is enough, and prevents the effect from sounding too obvious.

Golden Tone


Bellari RP583

I have to admit that I was pleasantly surprised by this compressor, because not only could it deliver the 'opto-compressor with attitude' effect reminiscent of earlier Joe Meek designs, but it was also capable of very tasteful mix compression. The secret to setting how much attitude you hear is really a matter of how high a ratio you use (and of course how far the input rises above the threshold) combined with the attack- and release-time settings — higher ratios combined with a slowish attack and a fast release results in an audible, but still rather musical, pumping effect. Lengthening the release time tames the pumping, while shortening the attack brings transients under control more quickly. The only real way to evaluate how a compressor is affecting the sound is to set the output levels so that when you hit bypass there's no obvious level change. Having done this, I found the unit sounded rather better than expected for mix levelling — adjusting the gain so that the peak sections matched the bypass level resulted in the quieter sections becoming noticeably louder. There was virtually no tonal change when hitting bypass, which is also very important, and though compressing a mix hard does introduce some obvious compression artefacts, they are on the whole warm and musical. A little mild pumping can be useful to inject a sense of energy into an otherwise tame-sounding mix! Used to process individual voices and instruments, the RP583 injects an endearing warmth and solidity into the sound, but again without compromising the original tonal integrity. It works just as well on drums and bass as it does on vocals, and, though not as transparent sounding as some compressors I've used when you start to push it hard, it still manages to enhance musicality, and it does so without deadening transients or adding muddiness. At more sensible gainreduction levels, it sounds far more benign, but it's difficult to tell exactly how much gain reduction you are applying when the meters are moving-coil types rather than peak-reading LED meters.

Beauty Or Beast? On balance I really liked the RP583, because it delivered what I wanted to hear with very little effort on my behalf, and it was also versatile enough to be used both for overt compression effects and for more subtle gain management. It isn't completely transparent, of course, and probably isn't designed to be, but it has the kind of smooth, musical character that I tend to associate with more costly units, and it is far more versatile than its few controls might lead you to believe. It works well on mixes, submixes, and individual instruments, as well as on voices. Although it's hard for a unit such as this to be a Jack of all trades and master of them all, the RP583 tries very hard to please. A good all-rounder at a sensible UK price. Published in SOS June 2004


Bellari RP583

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



East West/Quantum Leap Symphonic Orchestra

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East West/Quantum Leap Symphonic Orchestra : June 2004

Sound On Sound

In this article:


The Full Package Orchestral Sample Library Installation & System Published in SOS June 2004 Requirements The Kompakt Player Printer friendly version EWQLSO Volume One: Reviews : Sound/Song Library Strings Symphonic Orchestra Instrument List EWQLSO Volume Two: Woodwinds It's another multi-DVD orchestral Behind You! — EWQLSO In thunderous introductory timp roll Surround vibraslap? EWQLSO Volume Three: Brass Versions & Pricing Dave Stewart & Mark Wherry EWQLSO Volume Four: Percussion Conclusions

East West/Quantum Leap Symphonic Orchestra pros Top-quality performances from the entire orchestra. Produces a real-life big concert hall sound straight out of the box, with no need for external reverbs. Offers a choice of three listening perspectives. 24-bit throughout, and 5.1 surround-sound compatible. The multi-platform Kompakt virtual instrument is easy to operate and quick to edit.

cons No solo viola, solo double bass, marimba, celeste or piccolo trumpet. No strings or brass runs, no brass trills, no harp chords or glissandi, no strings pitched col legno. The occasional gap in the implementation of performance styles reduce the library's flexibility.

summary Though designed with Hollywood film scoring in mind, EWQLSO's samples will work for many musical

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library! Does it merit a or a feeble tap on a

The goal behind the vast East West/Quantum Leap Symphonic Orchestra library (or EWQLSO to its friends) was to create a 24-bit orchestral sample library which could be reproduced in surround sound. This ambitious project was conceived by two American producers, Doug Rogers and Nick Phoenix (respective heads of the East West and Quantum Leap sample library empires) and brought to fruition by Grammy-winning recording engineer Professor Keith O Johnson. Making a library of this quality is not the sort of project you undertake on a spare weekend — in fact, making EWQLSO took a year of recording, editing and programming. For the library to live up to its creators' ideals, it was imperative to find a concert hall with great acoustics, and they seem to have succeeded, although for contractual reasons, the identity of the 2500-seater hall and its resident orchestra remain a mystery. Recording started in August 2002, the orchestra giving up their Summer break to the sampling sessions, and editing the resulting multi-channel recordings lasted nearly another year. The full version of the finished symphonic extravaganza weighs in at around 67GB (second in size only to the efforts of the Vienna Symphonic Library), and in terms of size and scope alone, must be considered a leading contender for any work requiring orchestral samples. Note, however, that smaller versions of the library are also available for those on a budget (see the final page of this review for more details).

The Full Package EWQLSO is divided into four volumes, Strings, Woodwinds, Brass and Percussion. Each comes in a large box covered in glossy artwork showing (for reasons of confidentiality) a different concert hall from the one used in the library! Inside are the library's 19 DVDs and an A5-sized, 126-page operation manual which covers the whole set (the manual is identical whichever volume you buy). The text is well written, and affords Professor Keith O Johnson the chance to give a spirited account of his multimiking methodology. In short, this is as follows. The orchestra's performances were

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genres. If you're interested in creating full orchestral arrangements, or just using top-quality orchestral sounds to enhance your productions, make a point of checking out this thoroughly impressive library.

recorded from three different positions in the concert hall, referred to as 'C' (close), 'F' (full mix, derived from clusters of stage mics) and 'S' (surround, from elevated mics near the back of the hall). All the recordings were made simultaneously from these three positions, creating three stereo versions of each sample. This enables users to adjust the mix of close, stage and hall 'surround' sound to their own taste, and allows the construction of 5.1 mixes.

information

During the recordings, the musicians took up their normal orchestral stage positions (violins and French horns stage right, violas centre stage, double basses and trombones stage left, and so on). The library's presets preserve these placements, so users can build up a full orchestral mix with ease. For further realism, the makers have attached www.arbitermt.co.uk release trails (aka 'release triggers') to the www.soundsonline.com library's presets to show off the sound of the concert hall. These reverb-only samples, GLOSSARY: technical terms which sound only when keys are released, are cleverly programmed to match the level explained of their corresponding samples, and you can edit their volume, decay and pan settings, or even turn them off altogether! See the 'Versions & Pricing' box above. Arbiter Music Technology +44 (0)20 8970 1909. +44 (0)20 8202 7076. Click here to email

Placing the stage mics at the Symphonic Orchestra sessions. Professor Keith O Johnson (centre, grey shirt and glasses) can be seen fine-tuning the mic positions.

£788,721 of Second-User Gear for sale today in SOS Readers Ads. Grab a bargain!

Screenshots too small? Click on photos, Professional composers chasing deadlines don't have time to wade through countless screenshots and diagrams Gigabytes of material to find the sound they need, so the producers' philosophy in articles (after August regarding performances has been to focus on what they consider the most useful and 2003 issue) to open a expressive articulations, with an unashamed bias towards Hollywood film scoring. As a Larger View window for result, there are no sampled licks, chords, mood pieces or full-orchestra tutti effects; detailed viewing/printing. instead, the library concentrates on providing users with a wide and expressive range of multisamples. The manual gives a clear and logical (though sometimes cryptic) list of all the variations in playing style, but says nothing about the instruments themselves. This is forgivable in a library geared to music production rather than musical education, but it would have been nice if somewhere amidst all the technical verbiage, someone had spent a few words explaining what some of the more obscure instruments are, or given some instrument pitch ranges for the benefit of less experienced composers.

Installation & System Requirements Rather than being supplied as a standard sample library, EWQLSO is supplied with its own sample player based on Native Instruments' Kompakt, which is itself based on their Kontakt software sampler. Installing the full library is a straightforward, if lengthy, process — you should allow at least a morning to copy the contents of all 19 DVDs to your computer. Simply run the provided installer for each volume of the library to install the basic application and plug-in files to your computer, and then copy the Kontaktformat NKS files to the appropriate location. Having the content files separate from the main installer is actually quite useful, since there are cases where you might want to remove these large files from your system and put them back at a later date. And in this situation, you won't have to uninstall and reinstall the actual application and plug-in. The copy-protection scheme used is the one found in most Native Instruments software these days. This means that after installation you'll be able to use the software for a five-day grace period, during which time you need to register and authorise your product with Native Instruments using the registration tool provided. The authorisation process is of the challenge-and-response type, and you'll get two licences for each part of the library, meaning that you can install the library on two separate machines. This is great for people working on large arrangements who need to spread their orchestra across multiple machines. Although the copy protection allows you to remove the authorisation from one computer to install on another, it's important to point out that once an authorisation is removed from a system it can never be installed on that particular system again, unless you reinstall your operating system.



NI have enjoyed great success licensing Kompakt to sample-library developers in the last year, and part of the reason for this success is to do with copyprotection, the theory being that it's easier to protect a sample library that's inseparable from the application in which it runs. However, there's an advantage for the user as well, in that it allows specific functionality to be added to the player to cater for the needs of a given library — each volume of EWQLSO, for example, is supplied with a different player to handle the different sections of the orchestra. It should come as no surprise to learn that you need A diagrammatic representation decent computers to get the most out of this library. showing how the mic arrays were The minimum requirements to run Kompakt (as placed in the hall, resulting in the recommended by NI) are for a 500MHz Pentium III, library's 'C' (close), 'F' (full, or stage) Athlon or G3 processor with 256MB of RAM, and 'S' (surround, or distant) samples. running Windows XP, ME, 98 or Mac OS 9.2, 10.2.6 or higher. The better system, again recommended by NI, is for a 700MHz Pentium III, Athlon or G4 processor with 1GB of RAM, although my personal opinion is that most users will need substantially more RAM to run EWQLSO to its full potential. In the US, East West (in conjunction with music PC company Vision DAW) are offering full computer systems designed to run EWQLSO, and these machines feature a 2.8GHz Pentium 4 processor, 2GB of DDR433 memory, and a 36GB Raptor drive for the sample data. These type of specs are certainly more commonplace these days for dedicated digital-audio or sampling workstations, and give a better idea of the specification East West feel is appropriate for their library. I'd also recommend at least two computers dedicated to the library to get the most out of the sounds. You could run Strings and Percussion on one, and Brass and Woodwinds on the other — a good idea given that the strings and brass are typically the most used sections in orchestral writing, especially in Hollywood! In an ideal world, though, the best performance would easily be obtained by having four computers, so you could dedicate each orchestral section to its own computer. This doesn't mean that single-computer users won't get much out of the library, it just means that you might be doing rather a lot of bouncing!

The Kompakt Player The Kompakt-based sample players are supplied in both stand-alone and plug-in versions for both Mac OS 9/X and Windows, with Audio Units, RTAS (OS X only) and VST Instrument support on the Mac, and VST and DirectX Instrument support on Windows, with RTAS support to follow. Using the Kompakt player is pretty straightforward, and anyone who's used a similar Kompakt-based product should feel at home straight away. Instruments are chosen from a neatly organised pop-up menu and loaded into one of eight 'slots' in the instrument section of the interface. Each slot enables you to configure the MIDI channel on which the instrument responds, and the audio output that should be used, along with key-range and transpose settings. These settings make it possible to layer multiple instruments; by setting slots to respond on the same MIDI channel, you can also make use of the multiple microphone positions available in EWQLSO (alternatively, there are multi-instruments that load all three mic-position instruments in one go, but the disadvantage is that a multi-instrument always loads all eight slots, so multi-instruments are great for auditioning, but not when you have existing instruments loaded in the slots). If you set the close, stage and surround mics to different outputs within Kompakt, you can then assign these channels in your host's mixer to surround busses, and position the close mics to the front, the stage mics a little further back, and the surround mics to the rear. More than any other orchestral library, this gives you great flexibility to alter 'mic placement' in your virtual orchestra (for more on this, see the 'Behind You!' box towards the end of this article).



Since the sample data takes up just under 70GB of hard disk space, it stands to reason that you won't get too far playing these samples directly from memory. To this end, Kompakt supports disk streaming via the DFD (Direct From Disk) extension so that only the first part of each sample is loaded into memory, with the remainder being streamed from disk. There are a couple of presets for suggested DFD usage to help you get the optimum performance by balancing between memory and disk space, along with the ability to configure these settings manually with an Expert mode. There is, however, one catch: the DFD extension isn't supplied with the library, and must be downloaded separately after you register the product on Native Instruments' web site. This isn't a problem, so long as you have immediate Internet access. One area where a sample-playback engine often falls down in comparison to loading a more conventional library into a fully fledged sampler is in the amount of editing the user can perform on the sample data. Kompakt offers a fairly flexible interface for performing the most typical modifications on instruments, such as setting the velocity curve, a glide mode, a filter and amplifier section with associated envelopes and an LFO. In addition to a note-based filter (where each note played is filtered differently, Symphonic Orchestra's supplied Kompakt player. allowing for different velocities to adjust filter settings on the fly), there's also a master filter section if you want to filter the overall output of all the notes played for a given instrument, which is quite useful for taking the edge off strings and brass instruments. On the downside, there's no way to really get into the nitty-gritty programming of the instruments in Kompakt, such as if you wanted to alter the velocity crossover points for sounds, and so on. And other more advanced elements of the programming, seen in the automatic up- and down-bow-switching, various key-switching instruments, or the modulation crossfading, are similarly hidden from your prying fingers. Most people probably won't miss not being able to delve this deep, but power-users do like to reprogram specific samples in this way on a regular basis. However, sample data can apparently be loaded into the latest version of Native Instruments fully-blown Kontakt sampler and tweaked as much as required. The only real possible criticism of the Kompakt player is that you can only load eight instruments at a time, so you can't use all the channels on a single MIDI port. This isn't a problem if you're running the plug-in versions of the player, since you can run multiple instances, but it is annoying if you want to run the stand-alone version. Those interested in doing this would probably be better off using a simple plug-in host like Steinberg's V-Stack (now available for both Mac OS X and Windows) to run more channels with multiple instances.

EWQLSO Volume One: Strings So what's on those 19 DVDs? EWQLSO has a healthy complement of sampled string sections — two violin ensembles comprising 18 and 11 players, 10 violas, 10 cellos and nine double basses. Grouped together with solo violin, solo cello and harp (but alas, no solo viola or solo double bass) these make up a single volume which, for many, will provide the yardstick by which the whole library is judged. All the string ensembles and solo strings play vibrato sustains, 'expressive' vibratos, martelé short notes and 'legato mf' samples whose initial bow attacks have been trimmed to give an instant note response, presumably with fast lines in mind. The ensembles play staccato short notes, and a large number of additional performance styles are implemented selectively. One very welcome feature is that the string sections' sustains (including tremolos and trills) are looped, enabling keyboard players to hold down notes for as long as they wish.



* VIOLIN SECTIONS Sound library reviewing is all about close listening, so the 18 violins' close samples seemed a logical place to start. Loading their 'sustained vibrato' patch took about 50 seconds, but once in place, the samples established their quality in no time at all. The sound is smooth, polished and well balanced, with a bright, clear timbre, plenty of sheen and depth, and an expressive but not over-the-top vibrato. Quieter samples are sweet and steady, while the high velocity range introduces a more vigorous bow attack. One vibrato sustain option offers automatic alternation of up and down-bows, which sound much more realistic and lively than a series of uni-directional bow movements. The 18 violins' 'expressive vibrato' performances start quietly and breathily, swell in volume while increasing their vibrato, then sink back to a lower level for their sustains. The effect is romantic and dramatic, but if you want less expression, there's a fast attack version which dispenses with the fade-in. The trimmed legato samples sound slightly unnatural when exposed, but would sound OK in a mix. Going against the Hollywood grain, no-vibrato sustains have an austere, slightly dispassionate atmosphere, but the muted con sordino performances produce a warm, inviting, very enjoyable timbre. This violin section plays three types of short note: marcato (three different lengths), staccato (played with a good sharp attack) and martelé. The short marcatos are the quickest of all, delivering an urgent, emphatic bowing which is very suitable for detaché fast lines. Although fairly forceful, the martelé samples' somewhat more lingering attack is better suited to slower passages. All three styles have built-in alternating up- and down-bows, and the staccatos even have an option that selects a down-bow for every third note played! The 18 violins bow out with controlled tremolo sustains (apparently one dynamic only, but very usable) and some fine pizzicatos — EWQLSO's version of the latter classic orchestral timbre is as good as they come. The included 'slurs' turn out to be fast semitone slides up to a sustained note. Effects come in two flavours: a collection of extremely creepy, slowly ascending atonal slides (reminiscent of the orchestral build-up in the Beatles' 'A Day In The Life'), and a distant gassy noise which sounds as though the players are hoovering out the insides of their violins. An 11-piece second-violin section covers a lot of the same ground as the 18 violins, but also supplies a few new performance categories. 'Expressive diminuendo' is a notable example: its samples start with a three-second crescendo, sustain for a couple of seconds, then fade down over about three seconds; a very lush effect. Short, cartoonsoundtrack-style glissandi (a semitone or minor third slide up to a short target note), well-played tone and semitone trills, and some neat spiccato short notes complete the new performance styles.

Keith O Johnson recording EWQLSO. The mic preamps used in the recording were the Professor's own design (the compact Mackie mixer at the edge of this shot was for monitoring only, and was not used in the recording path). Even the A-D converters, just visible at the extreme top left of this picture, were the Professor's own designs.

Why, you might ask, do we need two violin sections? One answer is that the two ensembles sound different, the smaller section making a purer, more transparent noise than the rich chorusing of the 18 violins. Another advantage is that orchestral repertoire written for first and second violins can be programmed without fear of sample duplication. It's worth noting that the 11 violins do not play con sordino, pizzicatos, tremolos or slurs, and that neither violin section plays harmonics. * VIOLAS & CELLO Early sound libraries' attempts at sampled viola sections yielded some traumatic results, but these 10 violas produce rich symphonic sustains. Some may find the



players' vibrato a little fruity, but a cunning preset which allows vibrato and non-vibrato samples to be crossfaded via the mod wheel enables users to reduce the amount of apparent vibrato by pushing the wheel halfway up! The violas replicate the violins' romantic 'expressive vibratos' and play marcato, staccato and martelé short notes, the last two featuring automated up- and down-bows. All sound very effective, and the staccato performances are very good indeed. But, in terms of basic styles, that's about it for the violas; there are no con sordinos, pizzicatos, tremolos, slurs or trills, which may prove a handicap for anyone trying to reproduce an orchestral score. The vibrato sustains on EWQLSO's 10 cellos sound great, and are an inspirational writing tool. The instruments have an unusually wide range (top note is C#6, two octaves above Middle C); with these soaring, singing samples spanning four full octaves, you could easily sketch out a full orchestral string arrangement. Arrangers will also be pleased with the large selection of performance styles. The 10 cellos are strong on dynamic mobility: their 'lyrical sustain' samples don't just sustain, but quickly fade in, pull back in volume slightly, sustain for a short period, then perform a crescendo building to a note that is abruptly cut off. The eight-second crescendos are a useful asset, and there are non-vibrato and con sordino sustains, portato samples which could be used for detaché passages, and staccato short notes (the latter are tightly played, but arguably a bit on the long side). Essential styles like pizzicato, tremolo and trills are all included, the latter sounding particularly exciting. The cellos perform their own brand of slithering slide effects, and top those with an unnerving noise which gives the impression that the auditorium is filling up with bees. * DOUBLE BASSES & ENSEMBLES The nine-piece double-bass section sound really committed and energised, turning in some fine bass notes which will really flap your woofers. A deep, powerful preset called 'big sustain' packs an aggressive bow attack — play this in octaves at 100 Watts, and your neighbours will soon be seeking alternative accommodation. The basses' vibrato sustains have a more lyrical style, but maintain a strong bottom end. There are also some 'expressive' double-bass performances containing two successive volume surges, which is a nice idea, but the fixed timing of the swells means that composers will have to write their music around the samples. The basses are the only string section to contribute fierce, confrontational forte piano and sfz (sforzando) samples, and overall they rack up an impressive tally of performance styles, including threesecond portatos, staccato up- and down-bows, pizzicatos, tremolos and long crescendos. In the effects department, the basses perform some frightening, groaning upward slides and unpitched col legno bow slaps. Perhaps hoping for a guest spot on the soundtrack of Jurassic Park 4, they also produce one amazing, guttural racket which resembles the growling of an enraged Tyrannosaurus. The EWQLSO team have supplied programmed combinations of their double basses, cellos, violas and violins. Billed with refreshing honesty as 'Fake Ensembles', these are mapped sensibly according to range and offer sustained, 'expressive' and pizzicato styles. The combined ensembles sound excellent, providing inspiring orchestral string patches for composers and keyboard players alike. * SOLO STRINGS As mentioned earlier, the EWQLSO strings library has no solo viola or solo double bass, but does contain a solo violin and a solo cello, both offering a decent (though not over-large) array of performance styles. The violin's 'sustained vibrato hard' preset has a biting attack and an intense, passionate vibrato. A 'smooth' version of this style offers some respite by introducing the vibrato gradually. There are some 'no vibrato' sustains, but the vibrato tends to be either off, or full on — a few samples featuring a more subtle vibrato would have been welcome. The violin's three-second crescendos are sensitively played, and one ambitious piece of programming sees the crescendo samples tacked on to the end of sustains in the form of release triggers — unpredictable to play, but quite entertaining! The martelé up-



and down-bows sound good and precise, but the marcatos' heavy vibrato is a bit overdone. 'Slurs' are back on the menu, but unfortunately there are no staccatos or pizzicatos. The solo cello more or less duplicates the solo violin's styles, but substitutes a double volume swell for straight crescendos. 'Expressive' samples also offer separate up- or down-bows, but to program a straightforward melody line, it's probably best to start with the vibrato sustains or trimmed legatos. This is a nice-sounding cello, and its short notes and slurs are played with conviction, sounding full, musical and engaging. No staccatos are supplied, but a preset called 'sustain accent' does a reasonable imitation if played in staccato style.

Symphonic Orchestra Instrument List STRINGS 18 violins. 11 violins. Ten violas. Ten cellos. Nine double basses. Solo violin. Solo cello.

The basic performance styles listed above provide the building blocks for useful musical combinations. Some presets combine different bowings, such as staccato upand down-bows which switch to short marcatos at top velocities. Others group together different, key-switchable performances — one 18-violin preset offers a choice of 10 different sustain styles, while another lets you switch instantly between legato, tremolo, staccato, marcato and martelé bowings. Sometimes velocity is used to control attack speed, but the most expressive, dynamic patches use the mod wheel to control the amount of attack accent, or crossfade between different performance styles (vibrato/ non-vibrato, sustain/tremolo). As mentioned earlier, you can't create velocity/key-switches or mod-wheel crossfades within Kompakt, but EWQLSO gives you plenty of choices — the 18 violins, for example, have 55 presets.

Harp.

WOODWINDS Three flutes. Three clarinets. Three oboes. Piccolo. Flute. Alto flute. Oboe. English horn. Clarinet. Bass clarinet. Bassoon.

* CONCERT HALL PERSPECTIVES

Contrabassoon.

BRASS

It's interesting to compare the string ensembles' close, stage and surround versions; the differences can appear fairly subtle on speakers, but careful listening on headphones reveals that the stage recordings have a wider stereo image and sound slightly more mellow, deep and spacious than their somewhat brighter close-miked counterparts. The surround samples are slightly delayed and have a more diffused attack than the other two sample sets, but although generally sounding more distant, they're less ambient and washy than you might expect.

Four trumpets. Four tenor & bass trombones. Six French horns. Three Wagner tubas. Trumpet. Trombone & bass trombone. French horn.

The Kompakt instrument doesn't display individual samples on screen, so it's hard to be sure how many dynamic performances are incorporated into a program. However, close listening to the 18 violins' vibrato sustains and pizzicatos revealed at least three distinct dynamic layers. The string ensembles' tuning sounds absolutely perfect, probably the result of some diligent post-production tweaking. * HARP Favourite instrument of the deranged, curly-haired Marx brother with the goggling eyes and cunningly concealed car horn, the harp adds a unique and indispensable colour to the orchestra's palette. This harp's basic plucked samples sound wonderful — six-and-a-half octaves of musical bliss, presented with short, medium and long release times. Supremely playable. The close-miked version sounds pristine and very detailed, while the 'full' stage mics add a file:///H|/SOS%2004-06/East%20West_Quantum%20Leap%20Symphonic%20Orchestra.htm (7 of 15)9/22/2005 7:42:42 PM

Tuba.

DRUMS & CYMBALS Bass drum. Snare drum. Snare drum ensemble. Field drum. Field drum ensemble. Tenor drum. Funeral drum. Tom-toms. Crash cymbals (piatti). Suspended cymbals.


lovely concert hall 'bloom'. The absence of harp glissandi, chords and arpeggios is partially compensated by the inclusion of harp harmonics, a beautiful, delicate timbre which spans two-and-a-half octaves rising from Middle C. Some special, lightly-played samples designed for playing arpeggios have been thoughtfully included, but they don't sound totally convincing when used for the quick sweep of a harp glissando.

TUNED PERCUSSION Timpani. Tubular bells. Glockenspiel. Crotales. Xylophone. Vibraphone.

EWQLSO Volume Two: Woodwinds Maintaining a convention established by Miroslav Vitous in 1992, the EWQLSO producers recorded woodwind ensembles of three players, providing unison samples of three flutes, three clarinets and three oboes. Solo woodwinds consist of the standard orchestral fare: piccolo, flute, alto flute, oboe, English horn, clarinet, bass clarinet, bassoon and contrabassoon. There are no exotic extras, but more crucially, no important instruments are missing. All the ensemble and solo woodwinds play sustained notes (most with a choice of vibrato or no vibrato), staccatos and trimmed legato samples like those supplied for the strings, and most have an 'expressive' sustain option. Long notes are generally looped, the only exceptions being the solo clarinet and contrabassoon. Beyond that, a variety of extra performance styles appear sporadically, among them tone and semitone trills, upward semitone grace notes, glissandi (the term here used to indicate an upward run of two or three semitones leading to a short note) and 'falls' (descending octave runs, some chromatic). These auxiliary performances are implemented differently (and rather unpredictably) from one instrument to the next.

UNPITCHED PERCUSSION Tam-tams. Anvils. Metal rail. Bell tree. Mark tree. Triangle. Castanets. Wood block. Claves. Ratchet. Vibraslap. Whistle. Slide whistle. Tambourine.

INDIVIDUAL VOLUME SIZES Strings: 27.9GB. Woodwinds: 16.5GB. Brass: 17.4GB.

* WOODWIND ENSEMBLES

Percussion: 5.41GB. Entire library:

67.21GB. The three unison flutes have a lovely breathy texture, and impart a great soothing atmosphere (use for your next new age Healing Moods album — do not use in kung fu film fight scenes). The straight sustains and legato samples work very well for pads, the latter producing a lovely mellow tone. The players' vibrato is quite subtle, but you certainly notice its absence in the 'no vib' preset. Cartoon soundtrack composers will be pleased with the perky grace notes, slurs and looped trills.

Unison notes on three clarinets can sound unpleasantly synth-like, but these samples are very easy on the ear, their success stemming from good ensemble tuning, fine timbral control and Professor Johnson's canny miking strategies. Although they play in only three basic styles, each one sounds great! Composers may prefer to avoid writing chords for the library's three oboes, but their naturally angular, slightly piercing timbre sounds very evocative in tandem with the subtle dynamic motion of the 'expressive vibrato' performances. Semitone grace notes played by the three oboes are also a colourful sound source. * THE FLUTE FAMILY The piccolo may be the smallest of the orchestral woodwinds, but its piercing falsetto shriek can cut through the densest orchestral chords. The instrument's bright, incisive quality is shown off well by glissandi and trills; its vibrato sustains are pretty cutting too, but the breathy, occasionally wispy delivery of the staccatos reveal the piccolo's fragile side.



The library's flute scores well, with a pure, lyrical singing quality which is particularly enjoyable in its middle and high register. The natural beauty of its vibrato sustains is somewhat obscured by the added release and reverb, but reducing those parameters restores the instrument's appeal. There's a nice little selection of handy performance styles: cartoony grace notes, 'falls', and flutter-tongue sustains. But this flute's strength lies in its evocative long notes — select the surround version of its 'slow expressive' performances, play a few sultry phrases, and it truly sounds like the Piper at the Gates of Dawn. Completing the family is a very nice alto flute, sounding comfortable and assured on its sustained and 'expressive' long notes and maintaining the same sumptuous, full tone in its staccato performances. The alto flute performs trills, and has one further trick up its sleeve — it plays up and down octave runs, the only instrument in the library to do so! The tempo and scale of the runs is undocumented, which probably means they were conceived primarily as an effect. * ORCHESTRAL REEDS EWQLSO's oboe sounds bright and piping, and its slow-growing, subtle vibrato underlines the credibility gap between fake 'mod-wheel' vibrato and the real thing. The instrument's straight sustains and legatos cope well with loud, assertive melodies, but for quieter passages, users of a delicate disposition may prefer the expressive sustains' more sensitive, evolving approach. The oboe clocks up the most performance variations of all the woodwinds, playing grace notes, falls, trills and glissandi (here sounding more like short, ascending chromatic phrases). It also serves up two fresh styles: some good, attacking sforzando crescendos, and a category called 'slide' in which, perplexingly, no slide is audible! The oboe's big brother, the English horn, shows solidarity by also playing non-slides (actually unlooped, four-second vibrato sustains). The English horn's vibrato is very restrained and develops slowly, and while that's preferable to an overstated 'pub singer' wobble, some might prefer a stronger vibrato. Putting such quibbles aside, the instrument's delightful, clear timbre and lovely breathy attack have an immediate emotional effect. Its grace notes are very appealing, and its falls and so-called glissando phrases have a nice languid flavour. Using the classical 'no-vibrato' delivery, the orchestra's clarinet player has very good sound and control, but the instrument suffers from a paucity of playing styles. The player's unlooped sustains dwindle in volume after five seconds and stop after eight, an unwelcome note length restriction. Fast melody lines pose a slight problem; the trimmed legato samples can actually handle them quite well, but you have to reduce their release setting first to stop the notes blurring together. There are two good crescendo options, loud portatos and excellent staccatos, but fancy stuff like trills and grace notes are off the menu. * LOW REEDS Bass clarinet, bassoon and contrabassoon provide a sound foundation to EWQLSO's higher-pitched woodwinds. Like the clarinet, the bass clarinet turns in steady, controlled performances, playing no-vibrato looped sustains in soft and hard varieties. There's a big jump in timbre between the soft and hard samples — a 'medium' option employs some crafty sample layering in an attempt to bridge the gap, but this introduces a few subtle chorusing artefacts you wouldn't expect to hear from a solo instrument! The bass clarinet's highlights are its liquid glissandi phrases and jocular staccatos. The basic requirements of a sampled bassoon are simple: it should have sustains capable of carrying a tune, and staccatos tight and energetic enough to handle quick rhythm patterns, like the little dancing arpeggios in Smokie Robinson's 'Tears Of A Clown'. Happily, this bassoon meets these specifications and more, offering sustains with and without vibrato (the former well-suited to melodic work), emphatic portatos, long and short crescendos, and glissandi phrases. All very jolly. Contrabassoon — now there's a good Scrabble word. The contrabassoon's samples also bag a good points score, showing off the instrument's wide timbral range. Its



vibrato sustains sound quite controlled and melodic, the 'expressive' volume swells introduce a menacing rasp, and the loud portatos make a big, bassy, buzzy racket like the glass-shattering two-note trump let off by the alien spacecraft in Close Encounters. The staccatos have a good dynamic response, and are a lot of fun to compose with. * FAKE WOODWIND ENSEMBLES Some bright spark of a programmer has lashed the individual woodwinds together to create a couple of virtual woodwind ensembles, which sound fabulous! Woodwind ensemble number one consists of (from the bottom up) contrabassoon, bassoon, three oboes, three flutes and piccolo. Number two is made up of bass clarinet, three clarinets, English horn and piccolo. Keyboard players will appreciate the exotically blended timbre and eight-octave span of the first; the second has a rounder, softer, more 'flutey' and supportive tone, with the English horn adding a delightful breathy attack. Overall, the library's woodwinds provide a wide range of timbres. There's no shortage of expressive performances, and, as with the strings, a great many presets feature keyswitching and mod-wheel crossfades, giving users even more expressive power. Occasionally, the omnipresent release trails can make solo instruments seem a little lacking in intimacy — if that's a problem, the trails can easily be reduced in volume or turned off, effectively placing the instrument closer to the listener. But if it's a concert hall sound you want, these woodwinds have it!

Behind You! — EWQLSO In Surround The producers recommend that users make their EWQLSO-based projects future-proof by using the 'C', 'F' and 'S' versions of the samples to create close, stage and ambient stereo mixes respectively. The idea is that these separate stereo files can later be combined to create six-channel surround mixes (a further use would be to do quick stereo remixes with different amounts of hall ambience in the sound). The diagram on the right shows the suggested speaker assignments for creating a 5.1-compatible mix. In this setup, the full (F) and surround (S) stereo mixes are sent to the front and Mapping the 'C', 'F', and 'S' samples to the rear speakers respectively, while one speakers in a 5.1 array to create a side of the close (C) mix (left or right, but convincing surround mix. not a sum of both) is sent to the centre speaker. The sub speaker carries a mono mix consisting of sub-bass frequencies derived from all the channels.

EWQLSO Volume Three: Brass There is something innately noble and impressive about orchestral brass letting rip in a big hall — EWQLSO aims to recreate the auditory thrill by supplying sampled brass ensembles of four trumpets, four tenor and bass trombones, six French horns and three Wagner tubas, supplemented by solo trumpet, solo trombone and bass trombone, French horn and tuba. All the ensembles perform looped sustains, and most instruments play staccatos, sforzando crescendos and accented, one-second portato notes. The ensembles (and solo trumpet) also perform a number of additional styles, as detailed below. * TENOR & BASS TROMBONES



The bass trombone's powerful, raunchy bass notes have been popular with composers and arrangers since the '60s. With this in mind, EWQLSO tacked a low octave of bass trombone ensemble samples on to the bottom of the tenor trombone ensemble's range, effectively extending the section's bottom note down to E1 (same pitch as a bass guitar's low E string). The composite super-ensemble of four tenor and four bass trombones makes a very flexible keyboard patch, giving players the best part of four octaves to compose with! The trombone ensemble has a very handy 'sus accent' preset which layers tight, ultrapowerful staccato attacks over rich, sonorous looped sustains. The mod wheel controls the amount of staccato, creating a versatile program which can switch instantly from short, punchy phrases to supportive pads. Weighty portatos and attention-grabbing fp samples sound suitably portentous, while very dynamic, one, two and three-second crescendos pile on the big screen drama. The players must have struggled to keep a straight face when playing their flutter-tongue samples (the only ones in the brass library) — depending on how cynical you're feeling, these convey a chilling sense of post-nuclear desolation, or sound like a family of hippos farting underwater. But whatever the performance style, this trombone ensemble sounds tremendous. A solo tenor/bass trombone combination stretches four octaves from E1 to E5, offering only three basic categories — no-vibrato sustain, staccato and sforzando crescendo, the latter confined to the bottom octave. Though the combined instrument delivers these performances with aplomb and also features a powerful 'sus accent' preset, the lack of more expressive, dynamic samples and the complete absence of trombone slides may occasionally be a restriction for users. * TRUMPETS/SOLO TRUMPET EWQLSO's ensemble of four trumpets is likely to prove a hit with samplists. Their fp and sfz crescendo samples are packed with power — both feature a fast, brilliant attack followed by a sudden dip in volume, but while the fp samples settle into a quiet, looped sustain, the sfz samples quickly flare up into a big, blasting volume swell which will have cinema audiences regurgitating their popcorn. The trumpet ensemble's threesecond crescendos are excellent, but their 'slurs' are a let-down; instead of the quick semitone slides performed by the strings, you get a weird, unnatural attack containing a trace of the lower octave — a pretty useless delivery, which the manual doesn't explain. But that's the only negative — the 'surround' samples are superb, making the trumpet ensemble's mighty sustains and staccatos sound absolutely regal. Nick Phoenix did a great job sampling solo trumpet in Quantum Leap Brass, and he and his colleagues have come up trumps (as it were) here too. The solo trumpet is the library's only brass instrument to play vibrato and non-vibrato sustains, both benefiting from high-quality note production. The vibrato is subtle but telling, and though unlooped, the sustains are long enough for melodic purposes. One unique preset provides staccatos with built-in alternating tongue attacks, giving the same advantages as the strings' up- and down-bows. Additional styles include a couple of the library's trademark 'expressive' volume swells, one subtle, the other histrionic! Dynamic and highly responsive, EWQLSO's solo trumpet is among the most playable and expressive instruments in the library. * FRENCH HORNS/SOLO FRENCH HORN The programmers have gone to town with their French-horn presets. The biggest, incorporating five dynamic layers, is a great showpiece, demonstrating the ensembles' wide timbral range from soft and soothing to bright, bold and blaring. Other presets focus on a particular timbral area by employing fewer dynamic layers. The horns' tight, fat staccatos are also very responsive to dynamic nuances and make a great symphonic keyboard patch, while the portato performances sound unhurried, stately and rich. Unlooped, hand-stopped long notes provide a marked tonal contrast, making a thin, metallic, attenuated section sound which evokes cinema's darker 'noir' side. The horn players let it all hang out with lusty octave rips and energetic 'shakes' (short, tremulous elephantine brays). Only one complaint — the so-called 'slides' are not slides at all, but the same kind of baffling racket as the trumpets' 'slurs'. Clearly



perceived by the makers as one of the jewels in their crown, the French horns are the most lavishly sampled by far of all the library's brass instruments, offering a grand total of 39 presets, and an impressive collection of big-sounding samples. There's not a duff moment amongst the solo French horn samples, either; the sustains and staccatos are spot on, all the way up to the improbably high A5 top note. One can overlook the limited selection of playing styles — this is a fine French horn that can really carry a melody, a worthy companion for the library's solo trumpet. * WAGNER TUBAS/SOLO TUBA A trio of Wagner tubas, noble-sounding, horn-like instruments, contribute some stirring sustains and octave rips, but for some unfathomable reason their samples are all played in octaves. Although this creates a very powerful and impressive brass sound, the built-in octave doubling (of which this is the only instance in the entire library) tends to muddy chord voicings, thereby reducing the instruments' flexibility. The solo tuba (the conventional instrument, rather than the higher-pitched Wagner contraption) sounds big and almost mournful in the empty hall, but its deep, round warm tones are equally comfortable playing a low bass line or a lyrical melody. The sfz crescendos reveal the tuba's fiercer side, and though the selection of playing styles is modest, the player puts in a big-hearted performance and delivers some high-quality samples. EWQLSO's brass has one more treat in store: a 'fake' brass ensemble made up of the four bass and tenor trombones, six French horns and four trumpets. The horns tend to dominate, but the overall sound has an epic, gargantuan quality — film composers will be on to it in a flash.

Versions & Pricing The full version of EWQLSO (reviewed here) is known as the Platinum Edition. Two scaled-down packages, both fully upwardcompatible with the Platinum Edition, are available at reduced prices. Both reduced editions can be imported into the Kontakt sampler (v1.5 and up), enabling users to build their own programs. GOLD EDITION (15GB) This includes the vast majority of instruments and articulations from the Platinum Edition in 16-bit stereo, but only has the output from the stage mics, plus release trails. SILVER EDITION (2.4GB) The most basic version of the library omits Wagner tubas and some percussion instruments, includes only basic articulations, and features fewer samples per preset. It's also only 16-bit stereo, and features the recordings from just the stage mics, with no release trails. However, there are some bonus sounds: an East West Steinway B grand piano, Post Musical Instruments pipe organ, and Quantum Leap Voices of the Apocalypse male and female choirs. PRICING Platinum Edition (19-DVD set) £1999.99. Gold Edition £649.99. Silver Edition £199.99.



Strings volume (seven-DVD set) £649.99. Woodwinds volume (five-DVD set) £649.99. Brass volume (five-DVD set) £649.99. Percussion volume (two-DVD set) £329.99. All prices include VAT.

EWQLSO Volume Four: Percussion At 5.41GB, EWQLSO's percussion is the smallest of its four volumes. According to Doug Rogers, power-users who run the library on multiple computers generally find they can install the percussion on the same system as their audio sequencer. This volume contains both tuned and unpitched percussion, covering all the main orchestral bases. * TUNED PERCUSSION The concert hall acoustic adds a fair bit of wallop to EWQLSO's timpani — if you're looking for a classic big orchestral sound, the stage mics produce impressively grand, powerful, clean-sounding hits, while the close mics add more definition if required. The drums are mapped chromatically over the best part of two octaves, with the same samples duplicated in two separate keyboard zones so players can use both hands to play fast repeated notes. Performances comprise single hits with hard and soft mallets, crescendo rolls in a choice of two and five-second lengths, and mp sustained rolls (conveniently looped, with release triggers adding a very nice, subtle final hit on noteoff). Moving to the other end of the frequency scale, the library has a very pretty glockenspiel. The close-miked incarnation of the instrument sounds pure and bright, but adding the 'surround' samples produces a halo of almost supernatural high-end energy. There's also a less high-octane variant called 'mellow glockenspiel' which sounds like the same glock played with softer beaters. Operating in the same rarefied pitch register are a two-octave chromatic set of crotales (small tuned cymbals). At the risk of sounding like a shampoo ad, sampled xylophones often suffer from dry, brittle tone, but this one maintains an appealingly round timbre through its whole range. The stage mics add a very attractive ambience, enhancing the xylophone's sweet, friendly sound. Once again, we mourn the absence of a marimba, but its jazzy pal, the vibraphone, does get a look-in. However, these vibes are not an unmitigated success: the vibrato mechanism was disabled during recording, the quiet samples sound slightly dull and the preset contains too much release. A little editing might be needed here before use. Orchestral chimes (or tubular bells, as we Brits know them) peal out their portentous, churchy clangs over two full octaves from G3 to G5. As with the glockenspiel, the stage mics add a fabulous extra dimension, transforming an attractive set of close-miked samples into a spectacular sonic event. * ORCHESTRAL DRUMS & CYMBALS As well as lending three of his custom-built tubas to the orchestra, Wagner kindly donated his gigantic bass drum. What a guy! The manual doesn't give dimensions for Wagner's drum, but judging from the selection of booming crescendo rolls, straight hits and one thunderous, almost sub-sonic sustained roll which rattled the floorboards, you'd have trouble fitting it into a Mini. A smaller-sounding concert bass drum performs a similar menu of rolls and hits — all these performances are fine, but a little more dynamic variation in the straight hits wouldn't have gone amiss. EWQLSO recorded three different orchestral snare drums, playing left- and right-hand single hits, sustained rolls and crescendo rolls. They also supply two excellent snare



drum ensemble programs, the larger one sounding slightly deeper than the smaller. Both ensembles have a small menu of very well-chosen samples which sound totally convincing when combined in performance. The library's field drum gives a lowerpitched, ominous snare sound, while a funeral drum and tenor drum darken the mood further with some agreeably doomy, resonant hits. Orchestral libraries usually make a dog's dinner of recording tom-toms, but the set of five included here are an unexpected treat, sounding nicely tuned, pure, melodic, dynamic, resonant and sustaining. The library offers six pairs of piatti (crash cymbals), ranging in size from 12 to 20 inches. Though the samples lack the explosive quality necessary to jolt the audience into full cardiac arrest, the recordings are super-clean and the hall ambience adds some weight and sheen. If you prefer the more subtle sound of a single cymbal played with mallets, there are four suspended cymbals of different sizes, playing a selection of single hits and lovely crescendo rolls. * UNPITCHED PERCUSSION The library's gongs are not tuned, but big, unpitched 'tam-tam' orchestral gongs that go 'BWOAARGH!!' There are four, ranging in size from 23 to 60 inches, performing straight hits, rim scrapes and a choice of short and long crescendo rolls. The latter performances really live up to their name, taking so long to, er, crescend that you begin to doubt whether the key has a sample assigned to it. The five-foot tam-tam turns in a couple of absolutely massive loud hits which megalomaniac producers will love. According to the manual, four anvils and some 'railroad tracks' were also used to create a bank of resounding metal clanks — hopefully someone remembered to put the rails back afterwards. EWQLSO's percussion volume concludes with two miscellaneous categories: 'various metals', containing tasty samples of bell tree, mark tree swishes and triangles, all beautifully recorded. Concealed under the heading 'various perc' are castanets, wood blocks, claves, ratchet, whistle, slide whistle, vibraslap and tambourine, plus a couple of unidentified percussion hits, which is a shame — busy composers would surely prefer to see an itemised list.

Conclusions Having spent some quality time with EWQLSO, it's clear that its pre-release hype was based on solid achievement rather than exaggerated claims. The library passes all technical and sonic tests with flying colours, leaving its musical stance the only area open for debate. Geographical, cultural and commercial connections with the West Coast film industry resulted (in Nick Phoenix's words) in 'a conscious decision to make a library aimed at Hollywood film scoring'. While many European composers will be delighted with that, some may be unhappy with the decision to omit certain performance styles (flute and clarinet trills, for example) on the grounds that they are not essential to Hollywood composers. But this is a minor concern compared to what's on offer here — a big, bountiful, powerful, expressive, sonically superior collection of top-quality orchestral sounds, recorded in a first-class concert hall, played, recorded and programmed by expert practitioners, waiting to burst into life in your compositions.

Published in SOS June 2004



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Emulator X Studio

In this article:

Emu 1820M Soundcard: Brief Specifications The Full Range Docking Procedure Drivers E-Wire Patchmix DSP Emulator X File Converter Formats Main Section Digital Details DSP Effects Emulator X Overview System Requirements Voice Processing In Use In The Balance

Emu Emulator X Studio £430 pros Excellent audio performance, with incredibly good 118dB measured dynamic range. Balanced analogue inputs and outputs — even on £149 1212M card. Wide range of good-quality DSP effects. Effects can also be accessed as VST plug-ins from within suitable host applications. Emulator X software features Emu's amazing Zplane filters. Good bundled sample library. Extremely good value for money.

cons DSP effects are disabled above 48kHz. Instrument inputs don't offer

Emulator X Studio 1820M • 1820 • 1212M PC Soundcard & Software Sampler Published in SOS June 2004 Print article : Close window

Reviews : Software

Emu's new range of soundcards offers an unprecedented level of flexibility, DSP power and sound quality for the price — with the added bonus of a very impressive software sampler. Martin Walker

As part of the Creative Labs empire, and designers of the core chips used in the Soundblaster Live and Audigy ranges, Emu have been largely responsible for the feature set of a vast number of consumer soundcards in general use around the globe today. They also, of course, have a huge amount of experience designing hardware samplers and synths. However, the only soundcard previously released under the Emu Emu's new hardware interfaces all use the name was the APS (Audio Production 1010 PCI card, with their proprietary E-DSP Studio), which I reviewed way back in processing chip. The 1820 and 1820M also SOS January 1999. This was feature the Audiodock breakout box. essentially a stereo sampler with builtin effects, launched without ASIO drivers, and with a fixed 48kHz audio 'engine', just like Creative's Soundblaster Live and subsequent Audigy series. This time around, Emu have launched a rather more ambitious range of five products based around their new 1010 PCI card. This not only supports 24bit/192kHz multi-channel recording, but also provides up to 16 simultaneous hardware-accelerated effects, courtesy of the new 100MIPS E-DSP chip. There are three audio interfaces in the range, and two of these are also available

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Emulator X Studio

high enough impedance for DI'ing passive guitar pickups. Emulator X File Converter utility doesn't read proprietary CD-ROM formats. No GSIF support, and no WDM support for 96kHz and 192kHz sample rates with current drivers. Changing sample rate can sometimes become a major undertaking. No readout of remaining DSP resources.

as bundles which include Emu's Emulator X software, heralded as 'the culmination of over 30 years of sampler development'. Emulating the designs of Emu's hardware samplers, with hard disk streaming or RAM playback, and using a complex engine with extensive modulation options, possibly its most intriguing feature is the inclusion of over 50 patented Z-plane morphing filters, as used in hardware such as Emu's Morpheus synth. With support for sample formats including Akai, Emu's own EOS and EIII, Emagic's EXS24, Gigastudio and Steinberg's Halion, as well as Creative's Soundfont 2.1, this hardware/software combination promises to be extremely powerful, and as long as there are no stings in the tail it looks set to take the market by storm.

summary With their new range Emu have managed to combine a soundcard with an amazing audio spec, balanced I/O, and a set of useful DSP effects, with a very sophisticated soft sampler. Only a few niggles along with a couple of current driver limitations slightly mar this impressive debut.

information 1212M £149.99; 1820 £299.99; 1820M £349.99; Emulator X Desktop Sampling System (1212M plus Emulator X) £219.99; Emulator X Studio (1820M plus Emulator X) £429.99. Prices include VAT. Emu Europe +353 1 433 3201. +353 1 806 6788. www.emu.com

Test Spec Emu 1820M Windows XP driver version 1.01, shown in Patchmix DSP utility as version 5.12.01.0488; Emulator X build 1.0.1.0557. Hardware: 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. Tested with Steinberg Cubase SX 2.0 and Wavelab

Emu 1820M Soundcard: Brief Specifications Supported sample rates: 44.1kHz, 48kHz, 96kHz and 192kHz from internal clock. Mic/line inputs: two, balanced XLR with switchable global +48V phantom power and 10 to +50 dB gain range, or unbalanced TS quarter-inch jack with -12 to +28 dB gain. Turntable input: twin phono, 47k(omega)/220pF input impedance, nominal 10mV RMS sensitivity. Analogue inputs: six, balanced or unbalanced line-level TRS jack at -10dBV or +4dBu sensitivity. Analogue outputs: eight balanced or unbalanced TRS quarter-inch jacks at -10dBV or +4dBu level, duplicated on four stereo 3.5mm jacks, plus headphone output with level control. Digital I/O: co-axial S/PDIF in and out at up to 24-bit/96kHz, ADAT optical up to 24bit/192kHz (switchable to S/PDIF format if required), Firewire port, all on PCI card, plus further S/PDIF optical output and two MIDI Ins and Outs on Audiodock. Sync: word clock in and out, SMPTE in and out, MTC out. DSP Core Effects: 1-Band Parametric, 1-Band Shelf, 3-Band EQ, 4-Band EQ, AutoWah, Chorus, Compressor, Distortion, Flanger, Frequency Shifter, Leveling Amp, Mono Delay 100 and 3000, Phase Shifter, Rotary, Speaker Simulator, Stereo Delay 100 and 1500, Stereo Reverb, Vocal Morpher. Frequency response: 20Hz to 20kHz, +0/-0.35dB. Dynamic range: 120dBA (analogue inputs and outputs). THD + noise: -105dB (0.0006%) with 1kHz signal at -1dBFS.

The Full Range All five products in the range include the 1010 PCI card featuring Emu's E-DSP chip. This provides the core processing, as well as co-axial S/PDIF in and out supporting up to 24-bit/96kHz and switchable to AES-EBU format if required, ADAT optical in and out at up to 24-bit/192kHz which is switchable to S/PDIF,


Emulator X Studio

4.01a, Cakewalk Sonar 3.1, Native Instruments Pro 53.

and a Firewire IEEE 1394 port. The analogue I/O varies from model to model. The £149.99 1212M model includes an 0202 daughterboard, which connects to the 1010 PCI card via an internal ribbon cable and provides a stereo pair of balanced/unbalanced inputs and outputs on quarter-inch jack sockets, plus a standard MIDI In and Out. This combination is also available with the Emulator X software for £219.99. The 1820 and 1820M packages If you want the Emulator X software but don't include a very smart Audiodock need the 1820M's multiple I/O, you could buy external breakout box featuring two the Desktop Sampling System bundle which includes the 1010 and 1212M cards plus flexible mic/guitar/line preamp inputs Emulator X. with optional phantom power, six balanced line-level inputs, eight balanced line level outputs, a hardware stereo RIAA preamp for direct connection of a turntable, complete with ground lug, two MIDI Ins and Outs, four stereo speaker outputs configurable from stereo to 7.1 surround, a further optical S/ PDIF output and a stereo headphone output. The Audiodock connects to the 1010 PCI card via a generous three-metre umbilical cable terminated in CAT5 (LAN) connectors. Emu say that this is a standard CAT5e network cable, and that you can use cables of up to 10 metres in length, but warn against plugging their specially RF shielded version into any other network connector. While the 1820 retails at a very reasonable £299.99 and provides audio performance up to 24-bit/192kHz on a par with various other soundcards, with a claimed dynamic range of 111dBA, the 1212M and 1820M both have 'mastering grade' converters (the A-D converters used are apparently the same as those of Digidesign's HD192 interface), which provide an even flatter frequency response, plus significantly greater dynamic range of a claimed 120dBA. The top-of-the-range £349.99 'M' version of the 1820 is also supplied with a Sync daughterboard that connects to the 1010 PCI card via an internal ribbon connector, and which provides word clock in and out, SMPTE in and out, and an MTC (MIDI Timecode) out. Finally, the 1820M is available with the Emulator X software as the Emulator X Studio package for £429.99.

Docking Procedure Like various other cards including the Audigy range, the 1010 PCI card requires more power than can be supplied via the PCI slot if you have an Audiodock unit plugged into it (it consumes 1.25 Amps at 12 Volts), so in this case you need to plug in the supplied Power Converter cable, which connects directly to a PSU spur. Other connectors on the card are for the 0202 daughterboard (only used in file:///H|/SOS%2004-06/Emulator%20X%20Studio.htm (3 of 15)9/22/2005 7:42:51 PM

Emulator X Studio

the 1212 system), the Sync card, and two mysterious ones labelled Xcard In and Out, which strongly suggest that multiple cards will at some stage be able to be sync'ed together internally, although Emu wouldn't be drawn on this when I queried it, and the drivers only currently support a single 1010 card. You can either use the supplied stickon rubber feet for desktop use, or four M3 6mm bolts and a standard 19-inch rack shelf, which will house two Audiodocks side by side. On the front panel are Neutrik combi sockets for mic (outer XLR) and guitar/line (inner TRS) inputs. Each one has red clip and green -12dB signal presence indicators, plus a rotary gain control calibrated from +20dB to +55dB when using the mic input and from -10dB to +25dB for line sensitivity. There's also a global +48 Volt phantom power switch with its own red LED indicator, and the preamps, which are apparently designed by Ted Fletcher of TF Pro, sounded good to me.

The Patchmix DSP utility lets you mix and route all the hardware input and output signals, plus the Wave and ASIO inputs and outputs, in virtually any way you please; you can also insert any combination of the DSP effects as aux sends, channel or main inserts.

Unlike on many other soundcards, these two inputs are independent of the backpanel offerings, and there are six more quarter-inch jack sockets that all support both balanced and unbalanced sources, along with eight identical sockets for the line-level outputs. The outputs are also duplicated by four alternate 3.5mm stereo jacks, intended for powered speakers in formats up to 7.1 surround. All six inputs and eight outputs can have their sensitivity switched in pairs to either -10dBV consumer or +4dBu professional levels, as can the 1212M's single stereo analogue inputs and outputs. Outputs 4L/R are designated as the stereo monitor outputs, although you can change this in the Patchmix DSP utility, while inputs 3L/R have a dedicated hardware RIAA-equalised preamp tied to them with its own pair of phono input sockets, plus a ground terminal (the low signals from deck cartridges are particularly susceptible to hum problems). This high-gain preamp can be disabled by plugging cables into the main 3L/R inputs, which will also mute its noise contribution. The remaining Audiodock controls comprise a pair of MIDI In/Out sockets on the front panel, a second pair round the back, the umbilical connector for the 1010 card, a front-panel optical S/PDIF output to connect up a DAT or Minidisc recorder, plus a headphone output and volume control. The latter can drive a higher current than the other outputs for lower-impedance headphones (the manual spec quotes figures down to 33(omega)). Finally, there's a set of understated front-panel indicators that remain invisible until lit — these show MIDI input activity on the two ports, correct clock Lock status, External clock,


Emulator X Studio

which of the four sample rates is being used, and SMPTE timecode transmission and reception. Eight analogue outputs are duplicated on

The manual misleadingly claims that quarter-inch stereo jacks for the easy the Audiodock 'has its own built-in connection of desktop speakers. power supply to isolate its 24-bit DACs and ADCs from the noisy computer environment', but like most other breakout boxes it derives the supply from its host computer. Emu also The 1820's two front-panel analogue inputs describe the inner TRS socket as a 'line-level/hi-Z input', but its impedance are independent of the rear-panel I/O, and is specified as 10k(omega), and guitars can be used to connect microphones, guitars or line-level sources. with passive pickups like to see at least 100k(omega) (and preferably 1M (omega)) to avoid high-frequency loss, so while the Emu preamps have plenty of gain, for best results you'll really need a DI box to avoid getting a dull sound. However, overall I was most impressed by the Audiodock's professional options, the number of extras, and the lack of any real compromises — the only one I spotted was that the Audiodock's rear-panel MIDI input only functions as a second discrete port if the Sync card's SMPTE input is disabled, and vice versa.

Drivers The Emu 1010 drivers and applications only run under Windows 2000 SP4 and XP (most developers are now abandoning the Windows 9x platform), and I had no problems installing them on my Windows XP PC. Emu do warn Audigy 2 and 2ZS owners that when you've installed the 1010 card and rebooted, Windows may attempt to install older Creative drivers, but as long as you cancel this you can install the proper 1010 drivers and software and run both products alongside each other. By the time I was ready to install the 1820M card, Emu had posted a version 1.01 driver update that fixed various phase inversion issues with the Audiodock and 1212M inputs, SysEx timing problems, headphone distortion at 96kHz, and incorrect headphone attenuation at 192kHz, but luckily it's only a 1.43MB Hotfix download that you run over the existing installation, not a full install. Currently, the ASIO drivers support sample rates of 96kHz and 192kHz, but not the MME-WDM ones. This will doubtless annoy Sonar users, although as always I suspect the majority of users will opt for 24-bit/44.1kHz format. Sadly, there are no GSIF drivers either, which means that you can't use this Emu range with Gigastudio.


Emulator X Studio

E-Wire If the already huge number of features wasn't enough for the boffins at Emu, they also managed to enable ASIO host applications to access the E-DSP effects as VST Inserts, using a specially designed VST/ASIO Bridge. This EWire plug-in lets you route audio to and from Emu's DSP effects, although the effect controls remain in the Patchmix DSP software. ASIO host applications that support latency compensation take care of all the timing, but Emu also thoughtfully provide a separate Delay Compensation plug-in to do this for those that don't. However, Emu's Patchmix DSP user interface is not under VST control, so unfortunately you can't automate the controls for the Emu effects within your sequencer.

Patchmix DSP Along with the drivers, Emu supply their Patchmix DSP software, which is a virtual mixing console that goes a long way beyond those of most other soundcards. All users also get Lite versions of Wavelab and SFX Machine, and in the 1820M package you also get a full version of Steinberg's Cubase VST 5.1, while 1212 users get Steinberg's Cubasis VST. Emu have stuck with the sculptedaluminium look of their previous APS EControl mixer for the Patchmix DSP mixer, but this time the whole thing's a lot more sophisticated. It's divided into four main sections, with a toolbar across the top (although the normal Windows title bar is missing), the lower right Main Section displaying the main levels along with their insert effects, Aux sends and returns, monitor and sync settings, plus a multi-function 'TV' Here you can see the Sync Card dialogue. info window at top right. Displayed on Notice also that this 192kHz Session has the left are as many 'input strips' as greatly reduced I/O, with the TV window displaying only two ADAT inputs and you've created for physical inputs or reduced Audiodock options. computer playback channels. Mic/line inputs have mono input strips, whilst the others have stereo ones. Below these are six visible insert slots, although if you need more you can access them by using the vertical scroll bar alongside the slots to drag them into view. Into these slots you can drag and drop effects from the Effects Palette (see box for details), or right-click and select from various other options. When you click on each insert its various controls appear in more detail in the multi-function TV window.


Emulator X Studio

The options include a peak-reading meter, trim control with up to ±30dB gain, variable-frequency sine wave, pink or white noise test tone, a send to an ASIO input or physical output, a send/return to a physical output and input (for adding external effects, for instance), or an ASIO direct monitor, in which case the mixer's strip signal is sent to an ASIO input during recording but monitored directly to avoid latency, and monitored via the ASIO host application during playback. This is an incredibly versatile selection. Below the inserts are pan controls, two aux sends, channel fader, mute and solo buttons, plus a handy 'scribble strip' for naming each channel. Patchmix DSP is 'dynamically configurable', which basically means you can add or delete mixer strips as you wish up to the number of available inputs and DSP resources, while a horizontal scroll bar and adjustable-width window let you change the way it looks. On creation you can also decide whether the aux sends are pre- or post-fader, and each new strip will appear to the right of the existing ones, although you can drag any strip to a different position at will, and reorder the inserts in the same way. If you click on the Inputs button near the top of the multi-function TV window, you get a display of the input strip assignments for each physical and host signal.

Emulator X File Converter Formats Emu supply a stand-alone File Converter utility that can convert many other sampler file formats into Emulator X's own EXB format. It's simplicity itself to use: you just drag and drop the source files into its upper window, select a destination folder, and then press the Convert button. It certainly worked well for me on files already on my PC hard drives in Gigastudio and Soundfont formats, but sadly it doesn't read non-native CD-ROM formats such as Akai S1000/3000 and Emu EIII/ESi (and will crash if you try). However, all these formats can apparently be written to DOS and accessed that way, and File Converter can apparently read Mac- as well as PC-formatted CD-ROMs. The PDF manual contains extensive details of the various conversions, and the converter even attempts to deal with Gigastudio, EXS24 and Halion features like keyswitching and controller switching by splitting them into separate Presets, although release triggering is ignored. The full list of supported file types is: Akai S1000/S3000, S5000 & Z series, MPC 3000/2000, Mesa. Emu E3/ESi, Emax II. Samplecell I & III. Tascam Gigastudio. Soundfont. Emagic EXS24 MkI & II. Steinberg Halion I & II. Creamware Pulsar/STS. Propellerhead Recycle I.


Emulator X Studio

Acidised WAV.

Main Section To the right of the input strips is the Main Section. The two aux sends each have two visible slots, but again a scroll bar lets you access more if required. You can drag effects into these slots from the Effects Palette or any of the other input strip options except for send/return — if you send to an ASIO, Wave or Physical output, the return signal will need a dedicated input strip. There are six slots displayed for the Main Inserts, plus a scroll bar if you need to add more. To the right of these are the main stereo output level meters with latching clip indicators, the main output level fader, a display of the clock source, current sample rate, and successful lock, plus volume, balance and mute buttons for the monitor section. The monitor mix itself is controlled from the Output Assignment section, accessed by clicking on the Outputs button above the info window. Like the Inputs display this shows both physical and host options in separate windows, but this time you can actively select whether each destination is connected to the Main or Monitor mix output by clicking in the appropriate box to link its patchbay graphic connection.

Digital Details Any changes of sample rate must be made in the Patchmix DSP software rather than the host application, since various functions change, and restrictions apply. First, at 96kHz and 192kHz sample rates the effects are disabled. This is not unexpected, given that their processing requirements would double and quadruple respectively compared with 48kHz. The ADAT I/O is reduced from eight to four channels at 96kHz, and two at 192kHz, while the S/PDIF I/O completely disappears at 192kHz. Meanwhile, while the 1212M still manages to run its stereo analogue I/O (and digital subject to the caveats above) at both 96kHz and 192kHz, with the two 1820 models you have to choose between the four-channel ADAT inputs or Line Inputs 2 and 3 at 96kHz, while at 192kHz there are four options: Mic and Line 2 inputs, Mic and ADAT inputs, Line Inputs 1 & ADAT, or Line Inputs 1 & 3. Given the complexity of this mixer, it's welcome that the toolbar contains a set of buttons to load, save, and create new 'Sessions' containing every Patchmix DSP setting, including the I/O sensitivities and digital options, sample rate and so on, plus the effects routings and settings for the input, aux and main inserts. Four further buttons launch windows for choosing Session Settings for the I/O, clock and sample rate options, Global Preferences, the Sync Card SMPTE settings, file:///H|/SOS%2004-06/Emulator%20X%20Studio.htm (8 of 15)9/22/2005 7:42:51 PM

Emulator X Studio

and showing/hiding the Effects Palette. Overall I found the Patchmix DSP software supremely configurable, although this can make it initially overwhelming, particularly for novice users. The only frustrating aspect for me was not being able to take an existing 44.1kHz Session, delete its ADAT and DSP aspects and change its sample rate to 96kHz or 192kHz — you can freely change from 44.1kHz to The Emulator X Multiset view lets you allocate Presets to each of the 32 MIDI 48kHz, but going any higher means channels, and tweak them in real time. starting from scratch. However, Emu have thoughtfully provided six Session starting points for both 96kHz and 192kHz, and 13 '44/48' sessions including an E-Wire Example, Percussion EQ, a setup for Rightmark's Audio Analyser, and even a Guitar Tuner using six channels of test tones! Small niggles include a difficult-to-read font for the insert slots and scribble strips, the inability to change your mind about pre/post aux send routing after creating an input strip, or any reminder of which you chose. I also found the red outline of the currently selected input strip quite subtle on my monitor, and a readout of available DSP resources would have been useful.

DSP Effects After clicking on Patchmix DSP's FX button to launch the floating Effects Palette window you'll see a list of folders, each containing 32-bit effects that can be loaded into the channel or main inserts or the two Aux slots. The uppermost folder is labeled Core Effects, and contains 20 basic algorithms that cannot be deleted — see the 'Basic Specification' box for a full list. Beneath this are folders containing presets created from one or more of the Core Effects in series, complete with all their settings — a list of the algorithms used appears in a small window below when you select it — and these are organised by default as Distortion Lo-Fi, Drums & Percussion, Environment, Equalisation, Guitar, Multi Effects, Reverb, Synth & Keys and Vocal. file:///H|/SOS%2004-06/Emulator%20X%20Studio.htm (9 of 15)9/22/2005 7:42:51 PM

The floating FX Palette lets you click and drag effects to the channel or main inserts or the two aux sends, shows any chained algorithms in its lower pane, and 'greys out' any that exceed the remaining DSP resources of the E-DSP chip. The effect parameters themselves appear in the right-

Emulator X Studio

hand 'TV' window. These presets can be renamed or deleted, and once dragged into an effect slot, can also be edited and subsequently saved as a new single or multi FX Preset.

Once you select the slot in question, its parameters all appear in Patchmix DSP's TV info window as a simple interface consisting of a set of horizontal 'faders' along with meters where appropriate, with dry/wet control and bypass and solo buttons at the top, and a drop-down list of Core Effect preset settings at the bottom, to which you can add your own user presets. Each Core Effect takes a different amount of DSP resources, and as these are used up in your mix you'll find that some effects in the palette become 'greyed out' as further options — the delays, reverbs and multi-effect chains are the hungriest and tend to go first! Unfortunately there's no readout for remaining DSP resources, but as a rough guide, you can only launch two Stereo Reverbs or two Stereo Delay 1500s, or one of each, before they get greyed out as further options. Even then you'll probably have enough DSP power left to run a couple of dozen EQs, chorus, compressors and the like. Saving a Session defragments the DSP resources, so this is a useful ploy if you're running low. Overall the DSP effects proved extremely versatile, and I particularly liked the simple yet versatile Speaker Simulator, the twin phoneme modulations of the Vocal Morpher, and the weirdness of the stereo Frequency Shifter. However, I must confess to being slightly disappointed with the Stereo Reverb, which is after all one of the prime candidates for DSP help. Although it's capable of reproducing plenty of useful ambiences, rooms and special effects, most of its longer tails were slightly metallic, and whilst it's significantly better than the majority of plug-ins bundled with MIDI + Audio sequencers, I've got several native plug-in reverbs in my collection that sound smoother.

Emulator X The Emulator X software is based on the design of Emu's very popular hardware samplers, and described as a 'Desktop Sampling System'. It certainly has an impressive specification, with stand-alone or VST Instrument operation, 24-bit sampling and playback at up to 192kHz with phase-locked stereo, RAM or streaming playback from hard disk, ultra-high-precision pitch interpolation, and support for a comprehensive range of sample formats (see box). Above all it's the advanced synth engine that will make people sit up and take notice, since this has 54 different filter types that not only include the familiar multi-pole resonant designs but also multi-phoneme vocal and morphing filters, as seen in Emu's hardware synths and modules such as the Morpheus. The two Emulator packages are effectively the 1212M and 1820M plus the Emulator X software for an extra £70 to £80. By the time you read this Emu will also be selling the software directly as an upgrade for existing users of these two cards for £109.99, with an initial special offer of just £69.99 until the end of May. Installation is easy, but due to an oversight the comprehensive 192-page PDF file:///H|/SOS%2004-06/Emulator%20X%20Studio.htm (10 of 15)9/22/2005 7:42:51 PM

Emulator X Studio

manual needs to be copied across by hand and renamed as 'Emulator X.PDF'. The software checks for the appropriate Emu hardware before it will run, although it doesn't specifically use its DSP capabilities, so Emu could theoretically release Emulator X as a stand-alone product to those who already have other soundcards. Emulator X is bundled with a 2GB library on four CD-ROMs. For me, the 401MB X Producer collection is the highlight, with over 1000 presets from the Proteus 2000 sound bank, but there's also the more aggressive Hip Hop Producer and the orchestral Saint Thomas Strings, plus a 20MB General MIDI sound set. The second and third CD-ROMs contain a Studio Grand Piano with eight dynamic layers, while Beat Shop One contains loads of kits, loops, and grooves.

Overview The Emulator X display is divided into two non-resizeable main panes, with a toolbar across the top of them, and a status bar beneath containing such details as the number of samples playing, RAM use and CPU overhead. At the heart of the Emulator X architecture is the Voice, which comprises one or more samples and a synth engine. A Preset maps one or more Voices across the keyboard, and may incorporate velocity layers or real-time crossfades, while a Multi Setup is a collection of Presets each mapped to one of up to 32 MIDI channels. Finally, Presets are collected together into a Bank. When you first load a Bank, its contents appear in the left-hand tree view as three folders for Presets, Samples and Multisets, and selecting one of these displays a list view of its contents in the right-hand pane, just like in Windows Explorer. Clicking on the Sampler Bank name box above the tree view switches the right-hand pane to the Multi Setup view, where you can choose Presets for each of Emulator X's 32 MIDI channels along with their Despite the huge number of parameters on level, pan, tuning and output routings, offer, Emulator X's Voice Processing editor is override the filter type, and alter any of really easy to use, and incredibly flexible. up to 16 continuous controllers that may be mapped to various preset parameters. If you have a MIDI controller you can also map its various knobs and sliders to control Emulator X. Multi Setups for each song can be saved and recalled via the main Multisets folder mentioned earlier. Selecting a specific Preset folder changes the view to the Preset Globals screen so you can permanently change the transposition and volume, controllers, choose from various scale tunings including such things as 19-tone and file:///H|/SOS%2004-06/Emulator%20X%20Studio.htm (11 of 15)9/22/2005 7:42:51 PM

Emulator X Studio

Gamelan, set initial positions for the 16 controllers, and modify various modulators and cords (more on these later on). This and the other editors can also be opened as separate windows if you prefer. Things become even more interesting if you open a Preset Folder: you can edit its sample allocation or easily assign single samples or multisamples to different note or velocity ranges using click-and-drag editing in the Voices and Zones screen, layer or split multiple presets in the Links display, or edit the synth engine itself in its Voice Processing screen. Opening the main Samples folder lets you see every sample used in the bank, and clicking on one opens the Sample Editor, where you get a large zoomable waveform display with loop points clearly marked, a transport bar for auditioning, and full control over the start and end points of the loop. A comprehensive set of DSP tools also becomes available on the main toolbar to edit the sample, including fades, pitch-shifting, time compression, sample-rate conversion, loop processing and so on. Most soft samplers rely on external editors for such functions, but since Emu use their own sample format an internal editor makes more sense. You can import WAV and AIFF files, and even record them using the Acquire function.

System Requirements Because of the hardware acceleration provided by the E-DSP chip, hardware requirements for the 1820M are modest, and Emu state a minimum of an Intel PIII 500MHz or AMD K6 processor, 128MB of RAM, and 500MB of free hard disk space, although personally I'd recommend 512MB or more of RAM to successfully run a modern MIDI + Audio sequencer as well. If you're also running the Emulator X software you'll require a considerably more powerful PC, as it relies totally on the host CPU (there's no hardware help from the E-DSP chip) — Emu recommend a 2.4GHz P4 or faster and 1GB or more of RAM. You'll also need to be running either Windows 2000 SP4 or XP, and have two spare PCI slot positions — one for the 1010 card and the other nearby to house the Sync card, although this doesn't actually plug into a slot, instead getting its power and other connections direct from the 1010 card.

Voice Processing The Voice processing is extremely impressive, with a dynamic filter, dynamic amplifier, up to three six-stage envelope generators, two multi-wave LFOs, and up to 32 modulation routings. Emu call the latter 'cords', and they are displayed in four groups of nine, each with a huge number of possible sources and destinations, plus a tiny rotary Amount control. Even the cords themselves can be modulated by other cords.


Emulator X Studio

The signal path is reasonably conventional, comprising an oscillator, filter and amplifier, but there are quite a few interesting twists en route. For instance, the oscillator lets you delay sample playback (useful when layering) and offset the start point (for missing out the transient at the start of a sample), and provides 13 different keyboard trigger modes to suit different mono and polyphonic instruments and playing styles. It also provides stereo chorus (although this doubles the number of voices used) and nine different glide (portamento) shapes.

With a large waveform display and an extensive range of editing, looping and DSP tools, Emulator X's Sample display is far more comprehensive than that of any other soft sampler I've reviewed to date.

The LFOs must be the most versatile I've seen, with 17 different waveforms including the usual suspects plus Octaves, Fifth+Octave, Sus4 trip and Pat: Neener (all of which provide simple arpeggios), Sine 1,2 and Sine 1,3,5 which combine several sine waves, Sine+noise, and the stepped Hemi-quaver. The six-stage envelope generators can be set up using rotary controls or by clicking and dragging in the graphic display window, and you can start from a range of templates, to which your own designs can be added. In fact, every graphic display has its own template options, making design work much more pleasurable. As you might expect from its VSTi capabilities, many aspects of the Emulator X engine can also be sync'ed to tempo, including envelope times, LFO frequency and delay time, and oscillator delay. For me, the highlight has to be the filter section, which goes beyond any other software synth/sampler I've ever used. Two thumbwheel controls for frequency and resonance, plus a graphic display, hide the massive 53 different filter types on offer. These cover all the classic low-pass, band-pass and high-pass types, but then move on to comb-filter responses of phasers and flangers, and complex responses with multiple peaks. In some models the frequency control is replaced by a control that morphs between two filter responses such as vocal cavities while the Q controls body size; in others, both controls may move multiple peaks in opposite directions, or change the frequency of some and the gain of others. Whether you want throaty 'talking' voices or ethereal movement in your sounds, I guarantee that if you've not come across Emu's filter set before you'll be blown away by it. I was also really impressed at the way Emu had managed to make so many Voice processing controls so accessible in a single display and yet so easy to use — far easier than the Gigastudio Editor for instance.

In Use


Emulator X Studio

Previous Emu/Creative designs such as the Soundblaster range have been notorious for their fixed-sample-rate engines and hidden sample-rate conversion, so it's reassuring to confirm that there are absolutely no compromises of this sort in the new Emu range — what you choose is what you get. After detailed comparison with my own Echo Mia using a wide range of classical, jazz, rock and dance music I found the Emu 1820M to sound quite similar on playback but noticeably more revealing in the details, particularly in the reverb tails and with stereo imaging, suggesting a lower-jitter clock. Rightmark's Audio Analyser provided some of the best results that I've measured to date. It reported a flat frequency response, with -0.5dB points at a low 7Hz and 20.5kHz at 24-bit/44.1kHz, a low THD of 0.0009 percent, and an exceptionally wide dynamic range of 118dBA — a good 10dB better than the Mia, which is one of the quieter soundcards around. Unfortunately, as I mentioned earlier, the MME-WDM drivers don't currently support 24-bit/96kHz or 192kHz sample rates, and since RMAA doesn't currently support ASIO drivers, I couldn't measure the audio performance at rates above 48kHz.

The PCI card itself features a CAT5 connector for the breakout box, co-axial S/ PDIF and optical ADAT digital I/O, and a Firewire port.

However, the ASIO drivers provided great performance, managing a 2ms latency with both Pro 53 and Cubase SX 2. I received exactly the same low latency using the ASIO drivers with Sonar 3.1, although I couldn't get the WDM ones to work much below 30ms on my PC. Like Halion and Kontakt, Emulator X can either stream its sample data from a hard drive, or store it in RAM if you have enough and want to work out the drive to full capacity playing back conventional audio tracks. There are various pre-roll and sample buffer settings in its Preference dialogue to optimise performance, as well as a useful CPU Cap setting to prevent it from hogging more than a specified amount of your processor's capability. I ran it with no problems as a VSTi within Cubase SX 2, and with the VST Adapter 4 within Sonar 3.1, and with typical Presets it took about 35 percent of my P4C 2.8GHz processor when running 32 voices. However, its advanced engine can be rather CPU-greedy — the Dynamic Grand in the Proteus Composer bank, for instance, uses three samples per voice, so even with 32-note polyphony you'll be accessing and streaming 96 samples, taking about 65 percent CPU capacity on my PC.

In The Balance During the last eight years or so I've reviewed over 60 different soundcards for SOS, and as you might expect, it's not often that they surprise or impress me any more. After all, many soundcards are permutations on existing designs, adding a mic/guitar preamp here and there, or providing the same set of features at ever file:///H|/SOS%2004-06/Emulator%20X%20Studio.htm (14 of 15)9/22/2005 7:42:51 PM

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lower prices. Many companies have tried to produce a soundcard with versatile I/ O and DSP effects, but few have succeeded. Lexicon's attempts were too expensive for most and finicky about their PC host, Yamaha's DSP Factory was launched with very little software support, and Creative's own Audigy cards were hampered by confusing software and engine limitations. In my opinion Emu are the first company to have got it right with their 1010 PCI card range, and have done so at prices that will result in some dropped jaws from their competitors. At an entry-level price of just £149.99 you can have extensive, freely configurable DSP effects as well as up to 32 virtual ASIO outputs with the basic 1212M stereo analogue in/out configuration, all at exactly the same high quality as the 1820M under review here. However, anyone considering the 1212M would be foolish not to pay the extra £70 for the incredibly versatile Emulator X software bundle with its 2GB library. Personally, I think Emu could cut also the hardware ties and sell loads more copies of the software by itself to those who've already bought soundcards. If you have more ambitious I/O requirements, the versatile 1820M under review here should provide enough for the majority of users, especially since its ADAT I/ O would let you add another eight analogue ins and outs by buying a third-party converter box. However, I do feel that the 1820 will become the poor relation, since for just £50 more the 1820M not only provides around 8dB more dynamic range and a flatter frequency response, but also the added features of the Sync daughterboard. Some potential users might consider a Firewire or USB 2.0 solution a better long-term purchase, but the basic PCI format is likely to be around for some years to come, so I don't personally think this is a worry. Overall, the 1820M and Emulator X bundle is the most impressive 'soundcard' that I've had the pleasure of reviewing for several years, and I've no doubts that it will sell and sell. Published in SOS June 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



Evolution MK461C

In this article:

Evolution MK461C

Instant Gratification Assignment & Reassignment Assignable USB MIDI Controller Published in SOS June 2004 MIDI Wrapped In An Enigma: The Free Editor/ Print article : Close window Librarian Reviews : MIDI Controller Any Omissions?

Evolution MK400Cseries Controller Keyboards pros Three models (425C, 449C, and 461C) to suit different needs and budget, all representing real value for money. Real plug and play under Mac OS X — no drivers thanks to USB class compliance. No power supply required (but a power input is available for stand-alone MIDI use anyway). Good user feedback from LCD display makes setup and use a breeze. Invertible faders for 'proper' drawbar imitation (not on MK425C). Polarity reverse on assignable footswitch input.

Keyboard

Evolution's new class-compliant MIDI keyboards are a cinch to use with computer-based sequencers. But do they perform as well as their spec suggests? Paul Wiffen Photos: Mark Ewing

Two years ago, I was staggered to find that when I plugged Evolution's MK249C controller into my laptop Mac for the first time, the keyboard powered up, installed itself under OS X and allowed me to trigger Ableton Live immediately without any drivers needing to be loaded. Having spent years doing battle with OMS, FreeMIDI and MIDI Manager within OS 9 on behalf of myself and others, I was an immediate convert to Core MIDI in OS X. Of course, I fondly thought that all USB MIDI keyboards were going to behave like this with OS X; I didn't realise that the manufacturers needed to go the extra mile to produce what is known as a class-compliant USB device. Numerous other keyboards have passed through my hands since, and none of them have done the same neat trick of simply showing up as a MIDI device in Mac OS X's Audio MIDI Setup screen. There are Enigma makes a great cross-platform Editor/Librarian now quite a few class-compliant USB Audio devices, but until I got my hands on the new MK461C, the MK249C was the only one I had come across which did — and it's free! the same thing with MIDI. cons No hard-wired volume control. No MIDI In for other controller devices/instruments. Not for Mac OS 9 or Windows 95/98/2000 users.

summary The MK461C is a marriage of my two favourite Evolution products in one box. I miss the volume control from the MK249C, but having sliders

With the MK249C, I managed to replicate everything in my gigging keyboard rig of 20 years earlier in a portable computer setup, except for drawbars. To my amazement, last year Evolution resolved that for me too when they released the UC33 fader controller, which had the ability to reverse the action of faders at the touch of a button, so that it could be used with virtual tonewheel organs like Native Instruments' B4 and Emagic's EVB3. I still needed a controller keyboard, though. So imagine how pleased I was when I heard about the MK461C, which includes the UC33's nine faders right in front of

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Evolution MK461C

and rotaries on one controller is great, especially if you play Hammonds. Price/ performance ratio is great across the whole range, though, and the Enigma editor is a very valuable addition.

the keys where you need them to be. I eventually discovered that it was just one in a new range, the MK400C series. Although the two-octave version, the MK425C, does not have enough room for the faders as well as the rotary knobs, the four-octave MK449C and five-octave MK461C do have them. I decided to ask for the latter for review.

information Evolution MK425C, £119.99; Evolution MK449C, £159.99; Evolution MK461C, £199.99. Prices include VAT. M Audio UK +44 (0)1442 416590. +44 (0)1442 246832. Click here to email www.maudio.co.uk www.evolution.co.uk

Instant Gratification When it arrived, I plugged it straight into my Mac Powerbook, and once again, it proved to be fully class-compliant under OS X. You will probably deduce from this that if you are using operating systems earlier than Mac OS X (and Windows XP, with which the MK400C series are also supposed to be compatible out of the box), then Evolution's products aren't for you. But if you are an up-to-date OS user, 'installation' is delightfully effortless — if you can call it that when all you have to do is connect a USB cable! The MK461C draws its power from its USB connection, so there is no need for a power supply, although there is a socket for a 'wall-wart' PSU, in case you are connecting the keyboard to the rest of your MIDI rig via the five-pin MIDI Out socket instead of via USB. The blue backlit LCD came on as soon as I connected the USB cable to my Mac and within five seconds, my copy of Emagic's Logic reported a new MIDI In and Out available. Playing the keyboard immediately caused new MIDI notes to be displayed in Logic's Arrange window. Similarly, moving the controller knobs or sliders had them showing up too. Noticing that 'Drawbar' mode was activated by a well-labelled simultaneous push of two of the front-panel Function Buttons, I tried this and found that the slider outputs were immediately reversed (ie. with the highest values when they were closest to me). This means that at any point you can reverse the faders with a single action, and don't need to spend half an hour reprogramming each fader to behave like this. You can also see at a glance whenever this feature is enabled, as a slider appears on the LCD. I launched NI's B4 and was about to reassign all the faders when it occurred to me that maybe someone had already done this donkey work for me. Looking in the manual, sure enough control preset 02 was designed for B4 use, so I typed this in on the MK's numeric pad and found that all nine faders were now correctly assigned to B4's drawbars. I was in tonewheel-emulation heaven! Having established that the MK461C did what I was particularly interested in, I recovered my dispassionate reviewer's composure and set about working through all the other features of the keyboard. After all, it does have 12 rotary knobs, pitch and mod wheels and eight Function buttons which work in conjunction with a numeric keypad, as well as the nine sliders. Not everyone is as obsessed with drawbars as I am...


Evolution MK461C

Assignment & Reassignment Another of my favourite things about the older MK249C is that its 10 rotary knobs are preassigned to and labelled with the most useful synth parameters like Filter Frequency and Resonance, and Envelope Attack and Release. On the MK461C, the 12 rotary knobs (eight on the MK449C and MK425C) are freely assignable. But when I opened up a virtual synth or two, I found that the first preset defaults to covering the same parameters as were hard-wired on the MK249C, and I was instantly able to change filter cutoff and resonance, attacks and decays, again without having to get into assigning the knobs to different continuous controller numbers. I find nothing dampens your enthusiasm for a new product so much as having to get the manual out and read it to achieve the smallest thing, but there was none of that here. The other factory presets with which the MK461C is supplied as default allow you to instantly recall setups for other popular virtual synths like Steinberg's Pro 53, Model E, PPG Wave 2.v and Gmedia's Oddity, as well as GM and XG/GS presets. I tried several of these out and they all seemed to address the parameters most of us turn to again and again. What's more, the same rotary controls are used repeatedly for standard parameters like cutoff, so that the main parameters are on the same controls, no matter what instrument you are addressing. Of course, sooner or later you are going to have to set up presets for instruments which Evolution haven't been able to cover, but I found even this simplicity itself. Whenever you The mains socket is not essential, as the move a rotary knob or slider, the blue MK461C can be powered via the adjacent backlit LCD display switches to tell you USB connection. It's a shame there's no fivethe controller number it is assigned to, pin MIDI In for interfacing as well, as on the after a few seconds of showing the last UC33, but at this price you can't have everything. value it has sent. If you then press the Function Button labelled Control Assign, you only have to type a MIDI controller number on the keypad, and the slider or knob is immediately remapped (just make sure you type three digits; ie. '019' for controller number 19). In this way, I was very quickly able to change the B4 preset into one which would control the Emagic EVB3 and then save this mapping as a control preset. The Control Assign button also allows you to change the assignments of controllers which are normally fixed, like the pitch-bend or mod wheels. This means that on an instrument like a Hammond, where you wouldn't normally use pitch-bend, you can assign the pitch wheel to do something more appropriate. You can also change the input for the MK461C's footswitch jack from its default (MIDI sustain messages) to something like MIDI Start/Stop, leaving your hands free when starting or stopping your sequencer. Incidentally, those of you who have ever got stuck with a wrong-polarity footswitch (ever tried pressing for no sustain and letting go for sustain?) will be pleased to find that Evolution have file:///H|/SOS%2004-06/Evolution%20MK461C.htm (3 of 5)9/22/2005 7:42:57 PM

Evolution MK461C

foreseen this problem, and allowed you to change and store the polarity of the switch. The other Function Buttons are equally easy to use. Obviously, Global Channel allows you to set the overall channel on which the MK461C is operating and Channel Assign lets you change the MIDI channel for each individual controller — very useful if you have two or three virtual synths open at the same time. It is also handy for devices like B4 which can respond on different MIDI channels for the upper and lower manuals. The Program button lets you send a MIDI program change message, while the Data LSB and Data MSB buttons let you switch Banks in your receiving instrument (although you may need to brush up on your arithmetic to work out how a simple bank change from 1 to 2 breaks down into MSB and LSB strings — best to look in the appendix of your instrument for this). The Store and Recall buttons are much easier to use, as you type the number of the preset you want to write the current settings to or recall. There are functions for all double button-presses (like the Drawbar mode I mentioned earlier) but you do not have to remember these, as they are clearly labelled below the two buttons you need. Control Mute is a useful 'double-button' function as it allows you to stop data being sent from all sliders and knobs while you get them in the right position. This is particularly useful in conjunction with another 'double-button' function, Snapshot, as it means you can set up everything in advance and then send all the settings in one go. Other 'doublebutton' functions allow you to set up different Velocity Curves (something I didn't feel was necessary, as the response felt fine to me from the start) or perform a complete dump of the memory to an external device. For greater control over your presets, Enigma, a cross-platform Editor/Librarian is available for free from Evolution's web site, www.evolution.co.uk (for more on this, see the box above).

MIDI Wrapped In An Enigma: The Free Editor/Librarian Enigma is easily downloaded and installed, although you do first have to fill in a registration document that would not shame an on-line mortgage application, and enter the serial number of your keyboard. The software acts as a librarian for the UC33 and XSession controllers as well as MKs 461C, 449C and 425C, so you can manage all the presets for your Evolution products in one program.

The cross-platform Enigma control preset Editor/Librarian works with all of the MK400C series keyboards, as well as the UC33 fader and X-Session controllers.

As soon as you hook your keyboard into the computer, it shows the contents of its memory as a new bank alongside the factory defaults and any other banks you may have created. A button on the bottom right allows you to toggle between the prettier Graphic and more functional List View displays. As well as being able to import, name, rearrange and re-export presets, you can niftily copy


Evolution MK461C

individual control parameters by dragging and dropping or cutting and pasting, which saves a lot of time when setting up new presets.

Any Omissions? So is there anything about the MK461C which I didn't like? Well, I did miss the Master Volume control which the MK249C has immediately below the LCD; I always like to be able to adjust volume quickly, especially in emergencies. If a synth is making the wrong sound, if a note is hanging or it is just way too loud, there is no substitute for a hard-wired volume knob, and the one on the MK249C has spared my blushes at many a demonstration or gig. On the MK461C, by the time you have worked out which (if any) of the controllers is set to overall volume, your audience may be deaf, or at the very least you have will have blown your cool 'in-control' image. Of course, if you are not using the keyboard in front of other people and are restricting your music-making activities to home, then this probably won't be a concern. The only thing I miss from the UC33 which hasn't made the transition to the MK461C is the MIDI In. I have found this terribly useful on several occasions, using it to connect my MIDI guitar (amongst other things) into the computer when playing live. Because the MK400C series don't have MIDI Ins, you have to use another USB device to connect another MIDI controller. However, these really are the only two omissions I would point out on a product that is extremely good value. What's more, I think the smaller products in the range have the same price/performance ratio. The MK449C has four fewer rotary controls than the 461C, and loses the extra octave, but the price has been adjusted accordingly. Whilst the MK425C would not appeal to me personally, as I have little use for a two-octave keyboard and would miss the nine reversible sliders, it's still well priced and, of course, it's class-compliant (isn't this where I came in?). For me, the MK461C is the best MIDI controller yet for use with software tonewheel emulations, and also works efficiently as a controller for other virtual synths, making programming and performance a breeze. Published in SOS June 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

The Legacy Hardware General Impressions Installation The Wavestation Family Legacy Wavestation Programming Legacy Wavestation How Do You Run Yours? Macros Effects Global Settings Plusses & Minuses Saving & Loading Legacy Wavestation Spec Conclusions

Korg Legacy Wavestation

Korg Legacy Collection (Part 1) Virtual Instrument/Hardware Controller Published in SOS June 2004 Print article : Close window

Reviews : Software

Now that Korg's Legacy Collection is properly complete, we follow up last month's preview with the first instalment of our three-part review. This month, we focus on the Wavestation plug-in... Gordon Reid

Occasionally, an expensive and unattainable technology will mature to the point that Like all Wavestations, it something previously esoteric, distant and sounds superb. impenetrable becomes warm, cuddly, and It makes Wavestation synthesis accessible... almost practical. Numerous examples exist if you look for them. Take the Korg MS20, which first simple. appeared in 1978. Its beguiling patchbay and Host CPU permitting (mine external signal processor may have promised didn't!), you can run multiple instances, for true multitimbral far more than they delivered, but the Wavestation-ism. instrument nonetheless signalled a new Version 1.0.0 was totally maturity and accessibility for modular crash-free during the review. synthesis, placing it in the grasping hands of Just one fifth of Legacy impoverished students, myself among them.

pros

Collection, it's remarkable value for money.

Another prime example appeared three years Original Photo: Mark Ewing later when Korg launched the 'Poor Man's It has no signal inputs, either physical or in software. Prophet 5', the Polysix. Sure, there was nothing particularly radical in this, but it It doesn't appear to respond was a 'real' polysynth as opposed to a paraphonic string synthesizer on steroids, to SysEx — which means that and it sounded superb. It showed that polyphonic synthesis had matured, and it has lost real-time parameter placed it for the first time in the grasping hands of impoverished musicians, control. myself among them.

cons

It's not light on processor usage — my G4 could only run one instance.

summary The Legacy Wavestation is, as far as I can determine, an

In 1988, Korg did it again, employing Yamaha's VLSI and surfacemount techniques to combine sample+synthesis technology with digital multi-effects units and 16-track MIDI sequencing. The result was the M1, the instrument that signalled the maturity of the digital synth, and established the form of the modern

file:///H|/SOS%2004-06/Korg%20Legacy%20Collection%20%28Part%201%29.htm (1 of 14)9/22/2005 7:43:07 PM


exact sonic recreation of the original Wavestations, but one that offers true multitimbrality and a user interface that has improved beyond recognition. As just one fifth of the Legacy Collection, it's possibly worth the £399 asking price all on its lonesome.

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

keyboard workstation. The company repeated the trick two years later when it released the Wavestation. With its greatly improved vector synthesis and wave sequencing, this gave everyone access to the expensive, 'produced' sounds that were once the preserve of million-dollar studios. Since the late '90s, everything has become smaller, lighter, faster, cheaper, and more powerful, and advances in computer technology have started to reap a tangible benefit, because the fledgling technology of software synthesizers has been improving rapidly of late. But, as yet, there has been no software synth that has made me want to jump up and down and shout "gimme! gimme! gimme!" as did the first MS20s, Polysixes, M1s and Wavestations, all those years ago. But today there's a product that threatens to signal the maturity of the software synth, both in terms of sexiness and capability. It's not a matter of pricing; software synths have never been particularly expensive. It's to do with the desirability and usability of the product. And, in a weird twist of fate, it's a recreation of the Korg MS20, a reincarnation of the Korg Polysix, and a reinvention of the Wavestation. There's even a bit of the M1 philosophy tucked away inside it. Is it a bird? Is it a plane? No... it's Korg's Legacy Collection.

The Legacy Hardware Don't you just love the English language? It has words such as 'cute' and 'unbearably' which, when combined correctly, let me tell you that the MS20 hardware controller is almost unbearably cute. Unfortunately, it's also very specific in its role as an MS20 hardware controller. Sure, you can plug it into the host computer using a USB cable, and use its velocity-sensitive mini-keyboard to play the Wavestation. But while the mod wheel will carry out its appropriate task, none of the knobs or patch-points are active, not even the master volume control. So we'll say no more about it until next month.

General Impressions Even before its release, the Legacy Collection generated a significant amount of interest. My expectations prior to trying it out were therefore very high. This is why I was so surprised to find that there appeared to be no single philosophy underlying the product. There are five components in the Legacy Collection: three independent software synths (the MS20, Polysix and the Wavestation), a 'Combination' module called Legacy Cell that links the MS20 and Polysix (but not the Wavestation) and a hardware controller modelled on the original MS20. Now, if Legacy Cell supported the Wavestation, everything would feel more integrated. Likewise, if the hardware controller were not so obviously designed to support the MS20



software, everything would feel more coherent. If Korg Native mode (which we'll come to next month) worked across the whole package, everything would feel more like a single product. But they don't, and it doesn't. If I had to guess — which I do, because I have no inside information The Legacy Wavestation's Performance one way or the other — I would say Select page. that the Legacy Collection might be two disparate development projects that happened to come to fruition simultaneously, and which the top brass at Korg decided to combine into a single product. I can think of no other explanation for the fact that the Wavestation software (a product of Korg USA) and the MS20/ Polysix/Legacy Cell/MS20 controller (products of Korg Japan) are so completely un-integrated. Nevertheless, in a clouds and linings sort of way, this has a benefit for me — I can treat the American and Japanese components as independent products. So in this, the first part of my Legacy review, I'm going to concentrate on the Wavestation alone, leaving the other components until later.

Installation The soft elements of the Legacy Collection come in a single box that contains a CD-ROM and four manuals. These further emphasise the independence of the products because, as well as the Installation and MS20 Controller manuals, there's one for the MS20/Polysix/Legacy Cell, and another for the Wavestation. Installation from CD-ROM is painless, but when it's complete, you must obtain a licence code over the Internet. This procedure also gives you access to the registered users' download area, which allows you re-register the software if you change computer. If you choose not to register right away, you'll be able to use the software for 10 days, but after that time the plug-ins will fail to load. I know, because I waited.

The Performance Edit page.

Once the software is installed, you'll need to set up the I/O parameters for each of the three synths. Obviously, many of the settings will be dependent upon the MIDI devices connected to your computer, your audio input and output options and so on, but this is also where you determine parameters such as clock file:///H|/SOS%2004-06/Korg%20Legacy%20Collection%20%28Part%201%29.htm (3 of 14)9/22/2005 7:43:07 PM


source, word length, buffer size and latency. As with any other software synth, get these wrong and things may go wobbly. I found that if I reduced the audio latency from its default setting of 5.8ms to the minimum of 0.73ms, I obtained a quiet but unpleasant buzz from my Mac itself (thankfully, this was not present on the audio outputs). Although operation appeared stable, I didn't like this, so I increased the latency until the buzz disappeared, which it did at... 5.8ms! This is an acceptable figure (about one hundredth of a beat at 120bpm) but means that the Legacy Collection will not be my studio's most responsive sound source.

The Wavestation Family Developed by the former Sequential Circuits engineering team and released in 1990, the 61-note Wavestation keyboard used PCM samples as the basis of its sounds. However, unlike conventional digital synths and workstations, it had the ability to layer PCMs within patches, to chain them together, and to 'morph' smoothly between them. You could even layer up to eight patches simultaneously. With a wave sequence, a pad, and a couple of lead sounds playing simultaneously, the Wavestation was capable of producing what sounded like complete tracks, even without a sequencer. However, the original Wavestation lacked piano and drum PCMs so, in 1991, Korg bowed to the inevitable and added them. With a handful of new effects thrown in for good measure, the expanded 'EX' version was otherwise identical to the original. Despite numerous calls to produce a 76-note version, Korg never did so. In contrast, their next offering was the Wavestation A/D, a 2U rackmount version of the EX with a larger memory and a pair of analogue inputs. These made the A/D a powerful vocoder, as well as an unusual signal processor. The final version appeared in 1992. This was the Wavestation SR, a 1U rackmount with a tiny screen. This meant that you had to use a computer-based editor to get inside it, and few players bothered. Korg balanced this with a huge increase in patch memory. The whole hardware family was discontinued in the mid-1990s.

Legacy Wavestation Like Paul White, who previewed the Collection in last month's SOS, I am a huge fan of Wavestations. There's a simple reason for this... the things sound glorious. From the lushest pads, to magical, evolving timbral sequences, to rich imitations of orchestral instruments, to screaming leads and tortured guitars, the Wavestation is king. This is why, like Paul, I'm well-placed to compare the Legacy Wavestation to the originals: within reach of my right hand as I type this, I have a Wavestation A/D rackmount module, while behind me is the Wavestation EX that I'm using as the controller for both the A/D and Legacy. However, despite our love for these beasties, there is one thing that nobody cherishes about them. This is the operating system, which hides the synth's sonic glories behind more file:///H|/SOS%2004-06/Korg%20Legacy%20Collection%20%28Part%201%29.htm (4 of 14)9/22/2005 7:43:07 PM


than 40 impenetrable and wholly inadequate screens. If you've never tried to program any of the hardware Wavestations, let me give you an idea of just how difficult this can be... When you switch on, the synth presents you with the Performance Select screen, which allows you to select a sound, or press one of five 'soft' buttons to enter the Bank Select, Performance Edit, MIDI, Global, and View Performance List sections, respectively. The sixth button makes the A/D play the current sound on middle 'C', just to check that everything is hooked up and working.

Patch editing on the Legacy Wavestation.

So far so good, but a Wavestation Performance comprises up to eight 'Parts' which each comprise a 'Patch' plus additional parameters that determined how the Patch responds within each Part in the Performance. Got that? To access the Parts on an original Wavestation, you press the Edit button, and the screen then displays the Patches used, and offers six further options; Detail, Patch, Solo, Name, Effect, and Write. Burrowing still deeper, the Patch button takes you to a page that lets you determine how many oscillators are used in that Patch, and which offers six further options... and so it goes on. Now, imagine trying to construct a Performance comprising, say, five Parts, each of which uses a different Patch comprising four oscillators, each with its own detailed set of parameters. In principle, it's horrendous. On a 40-character, sixline screen, it's 100 times worse. As for the SR, with its two-line screen... don't even think about it! But that's still far from the end of the story, because the next step down the hierarchical ladder enters the Wave Sequencing page and the Wave Mix Envelope (Vector Synthesis) pages, which themselves offered further tiers of options and pages beneath. As a result, Wavestations are not synths that many people program for fun, and few players have scratched much below the surface. So, given that the sound is superlative, how can the software synth improve upon the originals? The answer, obviously, lies in the user interface.

Programming Legacy Wavestation To invoke the stand-alone incarnation, you double-click on the program wherever it exists on your Mac/PC, and the single, non-scalable Wavestation screen appears. As with its hardware ancestors, this boots up into its Performance Select page (shown on the previous page) but this is much more informative than before, showing all 50 Performances available in the current Bank. It also offers



access to every other Bank (there are 11 in all) at the click of a button. This alone replaces many of the original Wavestations' pages and saves the endless scrolling through hundreds of Performances that you have to go through on the Wavestation SR. You select a new Performance by clicking on its name in the lower window, or by clicking on the Performance number in the upper window and dragging it up or down. Once selected, you can play a Performance immediately, or you can use the Preview button to play one of five preset phrases for auditioning purposes. A huge improvement, right from the start. Double-clicking on a Performance name in the lower window has the same effect as clicking on the Edit button in the upper window, and takes you to the Performance Edit page (shown above). As before, this replaces umpteen pages in the original Wavestations' operating systems, including Performance Edit, Part Detail (1 to 8), and the Key and Velocity Zones page. Nonetheless, the philosophy is the same and, as on the originals, this is where you insert Patches into Parts and determine how each contributes to the Performance. Clicking on the little 'speaker' icons to the left of the screen mutes and solos the Parts, which is vital when you come to edit individual sounds within the Performance.

How Do You Run Yours? You can run the Wavestation as a standalone program supporting Core Audio interfaces and the Mac's own sound I/O system, or as a VSTi/ Audio Units plug-in. You can even run each version of the program simultaneously, although this will place a high burden on any processor. I'm very impressed with this degree of flexibility, and the fact that all the options worked perfectly, straight out of the box.

Happily, editing the Parts is not just simple when compared to the original Wavestation, it's simple, full stop. For example, you can assign a Patch to a Part just by dragging the Patch number up or down or, if you prefer, by clicking on the 'List' button and selecting from a list that appears superimposed over the joystick on the left of the screen. Clicking on a Part brings up the relevant values for each of the parameters in the Details and Zones sections of the window, such as the Part level, its transposition and fine-tuning, its response to Note On messages, its temperament, and how it's routed to the effects busses. You edit these by clicking and dragging, by typing in new values, by clicking through a range of settings, or by selecting from drop-down menus, as appropriate for each parameter. Best of all, you can determine the key and velocity zones for each Part simply by dragging the appropriate end of the bar in each display. Suddenly, the Wavestation becomes quick, easy and intuitive to program. I love it! Clicking on the Patch button or double-clicking on an individual patch name takes you down a further level into the Patch Edit pages, which is where the real nuts and bolts of the Wavestation's synthesis lie. As on the originals, each Patch can comprise one, two or four oscillators, with or file:///H|/SOS%2004-06/Korg%20Legacy%20Collection%20%28Part%201%29.htm (6 of 14)9/22/2005 7:43:07 PM


without hard sync of oscillators B, C and D to oscillator A. Each oscillator has an independent pitch section with a dedicated pitch envelope, a filter, an amplifier with a dedicated four-stage envelope, a panning section, a freely assignable fourstage envelope, and two assignable LFOs. What's more, almost everything can be modulated (usually twice) by a wide range of sources including velocity, aftertouch, and two assignable MIDI sources. If you were to lay out all four oscillators in each of the eight Parts on a sensibly proportioned, physical control panel, I fear that it would approach the size of a squash court, but the Legacy Wavestation reduces this to something considerably smaller that nevertheless remains clear and remarkably intuitive (see above screenshot). Less intuitive, perhaps, are the Mix Envelope and Mix Envelope Modulation displays in the upper right of the Patch Edit window. These were all but incomprehensible on the original Wavestations, not least because you had to program each step of the four separate five-stage envelopes on individual pages. Here, the concepts remain obtuse, but at least you can see what's happening. It works something like this... The amount of sound contributed by each oscillator in a Patch is variable according to a two-dimensional Mix Envelope. This is much like dynamic automation of a mixer, except that the total contribution is always 100 percent, whether this is delivered by a single oscillator, by two, or by four. You control the relative mix using 'Mixenv' (as the envelope is called) to determine how much each oscillator contributes at each of five points in time. You can also decide whether the envelope will loop, and between which of these points.

Macros 'Macros' are another unusual facility in the Legacy Wavestation. These are predetermined sets of pitch, filter, amplifier and panning parameters that apply specific, appropriate characteristics to your sounds (such as piano, organ, brass, strings, and so on) without you having to program numerous values. The natures of the Macros differ on each of the pages, but offer configurations appropriate to each. Of course, you are not limited to using the factory Macros, and as soon as you start to edit these elements of the sound, the Macro name changes to 'User', and the parameters are handled in the normal fashion.

To illustrate this, I have created a four-oscillator patch and fashioned the simple Mix Envelope shown on the next page. As you can see, the sound starts at Point 0 with the whole contribution coming from oscillator 'D', which has the wave *WSTouch inserted. The sound then travels along the line (more properly called a 'vector'... hence the name 'Vector Synthesis') from Point 0 to Point 1, at which time all the sound is being generated by oscillator 'A', which is *DeepWav. Next, the sound passes to Points 2 and 3, with the sound morphing to *Quarks and then *ResXwav before, at Point 4, it moves to the centre of the diagram, at which time the output is an equal mix of all four oscillators. I have also set a Loop, so that — if I hold the key indefinitely — the sound loops repeatedly from Point 2 to Point 3. As it happens, this choice of waves and loop is rather poor and results in a



discontinuous sound. No matter, I could have made a better choice of waves, and there are other options that let you morph more smoothly from one sound to another. As you might expect from a synth of this complexity, this is isn't the end of the matter, because the Legacy Wavestation also allows you to modulate your position independently on both axes of the Mix Envelope using your choice of two modulators from a list of 13 sources. This means that, in addition to travelling around and looping within the mix path, the sound can be modulated between oscillators A and C and oscillators B and D in all manner of hideously complex ways. This is the source of the Wavestation's ability to create evolving pads that are, in my view, unequalled by any synth before or since. And so we reach the deepest level in the Wavestation, the Waves themselves. But lest you think that there's little to discuss here, I must remind you that a Wave need not be just a single waveform sucked from the Editing the Mix Envelope. ROM, it can be as many as 255 waveforms joined together, crossfaded, looped, and generally manipulated in an operation called Wave Sequencing. This is not unique to the Wavestation. However, by allowing you to determine the duration and level of each Wave in a wave sequence, where the playback starts, the direction, how the sequence loops, how each stage crossfades to the next, how the sequence responds to modulation and controllers, and more, the Wavestation developed the idea far beyond anything else available in its day. Strangely, creating a wave sequence on the original Wavestation was not as hard as you might imagine. A single page allowed you to select each Wave, its tuning, its level, its duration, the crossfade duration, the start point for a loop, the end point, and number of repeats. Nonetheless, the graphical display employed by the Legacy Wavestation (shown on the next page) is a huge step forward. You can select Waves within the sequence and determine their parameters numerically, just as on the original Wavestation, but it's much simpler to list all the waves available, clicking on each to hear what you might be entering into the sequence, and then inserting the Wave as a new stage, or overwriting an existing one. Next, each stage in the resulting graphic has 'handles' that let you adjust the duration, level and crossfade time of the Waves. If you want to delete a wave, you just hold down the Ctrl key and click on it, and then select the Delete option from the pop-up menu that appears. It's elegant, it's simple, and it works beautifully.

Effects



The final element in the Wavestation 'engine' is its effects system. This comprises two stereo multi-effects units, Effect 1 (FX1) and Effect 2 (FX2). You can configure these in two ways, in parallel and series (helpfully, the routings are displayed graphically, as you can see from the screenshots on the next page). You can use the 'FX Bus' settings in each of the Parts and the Pan settings in the Patches themselves to route the signals to the four effects busses. These busses are confusingly named A, B, C and D, but you must not confuse them with the A, B, C and D Waves within a Patch. The names are the same, but they are very different entities. It would take more room than I have available here to describe the operation of the effects busses in detail, let alone each of the 55 effects offered, but the system is flexible enough that I was able to set up test configurations where, for example, a stereo chorus was placed in series before a 'live stage' reverb across Creating wave sequences on the graphical busses A and B, while Busses C and D display. passed directly to the outputs. I also tried a parallel setup where busses A and B passed through a stereo multi-tap delay, while busses C and D passed through a rotary speaker effect (this configuration would be useful if you wanted to process two sounds — say, a guitar and a Hammond organ patch — with different treatments). While this proved straightforward, the two mixers provided in each configuration make matters considerably more elaborate. Called Mix3 and Mix4, these allow you to feed a proportion of C and D into the A and B busses, either before FX2 or after FX2, depending upon whether the Series or Parallel configuration is selected, and how the mix parameters are set. The amount and panning of each Mix can be modulated, and drop-down menus offer 14 mod sources, including the assignable MIDI controllers. One further complication here wasn't present on the hardware Wavestations. When running the Legacy Wavestation as a stand-alone module, outputs 3 and 4 do not exist and all four busses are routed to the 1/L and 2/R outputs. If you load the software as a plug-in, outputs 3 and 4 are available, and remain independent of 1/L and 2/R. Although they're part of the Effects section, I can't omit mention of the Wavestation's vocoders. These six algorithms were introduced on the Wavestation EX, but only achieved their true potential on the A/D, whose very name was defined by the dual analogue-to-digital converters that allowed you to process external audio signals and use the synth as a vocoder. Unfortunately, the Legacy Wavestation has no signal inputs, so you can only use the vocoding algorithms to affect internal sounds, as on the EX and SR hardware models. Given that you can use multiple Parts to construct a carrier, and multiple file:///H|/SOS%2004-06/Korg%20Legacy%20Collection%20%28Part%201%29.htm (9 of 14)9/22/2005 7:43:07 PM


Parts to construct a modulator, the possibilities are still enormous, but the loss of the external inputs means that the vocoders' usefulness has been curtailed, which is a great shame. To be honest, this seems wholly unnecessary. I can't imagine that it's beyond the ken of Korg to write a set of drivers that would route audio from a soundcard to 'virtual' signal inputs. This would turn the Legacy Wavestation into a soft-A/D, which would be an altogether desirable upgrade.

Global Settings Before starting to use the Legacy Wavestation, you'll want to set up the Global settings for your configuration. This is where you determine which MIDI controllers will appear as the MIDI1 and MIDI2 modulators mentioned elsewhere in this review, and the input/output controllers that relate to the joystick. You can also filter various MIDI messages, and set the Master Tune and an overall transposition for the instrument. Importantly, this is where you determine whether wave sequences will run in synchronisation with a host application, with clock received over MIDI, or whether they're driven by the Wavestation's internal clock. The Global page is also where you can define up to 12 User Scales. This was a sadly under-used feature on the original, and no doubt will remain so within the Legacy Collection!

Plusses & Minuses The original Wavestation offered 16 Multimode Setups, within each of which you could insert up to 16 Performances on separate MIDI channels. However, this did not mean that you had 32 effects units... you were still limited to just two. So, although you could decide which of the Performances in the 'Multiset' were routed to the effects (or not) and in what proportions, this was a far cry from having 16 Wavestations at your disposal. There was only one way to achieve this; you had to buy more Wavestations. You can see where I'm going, can't you? By turning the Wavestation into a plugin, Korg have been able to dispense with the Multi mode because — if your computer's processor can handle the load — you can launch multiple instances of the software. However, by dispensing with all four 'Modes' of the originals (Omni, Multi, Poly and Mono), the Legacy Wavestation is less suitable for use with a MIDI guitar. This won't affect many users, but it's worth noting nonetheless. More seriously, the Legacy Wavestation has lost SysEx. On the originals, you could transmit, receive, or transmit and receive every parameter change in real time. This meant that you could sequence changes in the voicing as you played,



and replay these performances. With two Wavestations hooked together, you could adjust both at the same time from one of the machines. This was a powerful capability that the Legacy Wavestation seems to have lost — certainly, firing SysEx from one of my hardware Wavestations to another produced the expected results on the hardware, but nothing when the SysEx was redirected to the Legacy Wavestation. The loss of SysEx might have been ameliorated had the voicing parameters of the Legacy Wavestation been controllable from within your host application, but I can find no way to do this. If you have existing sequences containing real-time SysEx control messages, and you want to use these again, it seems that you cannot use the Legacy Wavestation to wholly replace your hardware synth(s).

Saving & Loading The Write button in the upper window of the plugin is context-sensitive and, depending upon which page you are editing, allows you to write Performances, Patches and Wave Sequences into the Bank and location of your choice. Alongside this, the File button allows you to Save All and Load All as .fxb files, while the Import button allows you to load All Data, All Performances, Single Performances, All Patches, Single Patches, Wave Sequences and Micro Tune Scales created on the original hardware synths. Neatly done!

Another minor problem involves the rate of the internal clock within the Legacy Wavestation. My tests show that this runs at a slightly different rate to the clocks in the EX and A/D hardware. Given the very different hardware involved, this is not surprising, although it's worth noting that the two original synths remain in perfect sync over periods lasting many minutes. Happily, when the Wave Sequence Sync parameter in the Legacy Wavestation is set to 'On', the Legacy is synchronised to external MIDI Clock (well... provided that the Clock Master field in the Preferences is set correctly, that is — and this isn't mentioned in the Wavestation manual!). Having said that, if you change the MIDI Clock rate with a wave sequence running, the Legacy Wavestation and the originals lose sync with one another. Weird! Another curious bug appeared when I tried some Vector Synthesis using my hardware Wavestation EX as the control source. Rotating the joystick clockwise from the 'A' position, the response on the EX was A-B-C-D, as you would expect. On the Legacy Wavestation, it was A-D-C-B. I couldn't find a mis-set parameter that caused this, but it wasn't a problem... I just turned to the Global page and inverted the response of the Y-axis. Nonetheless, it seems to show that a couple of differences have crept in as the Wavestation code was rewritten for the Mac and PC. Finally, before concluding my list of concerns, I must mention the Legacy Wavestation's hunger for power; it peaks at around 50 percent of the CPU power available on my 1GHz G4, as measured using the OS X Process Viewer. This limited me to a maximum of, umm... one instance of the Wavestation. I accept that this will become less of a concern as computer power follows its inexorable file:///H|/SOS%2004-06/Korg%20Legacy%20Collection%20%28Part%201%29.htm (11 of 14)9/22/2005 7:43:07 PM


upward curve, but I'm a little concerned that one third of a software package consumes half the power of a reasonably specified, one-year-old Mac. Having noted the negatives, let's turn to the positives... My Wavestations are mainstays in almost everything that I record, not because I am limited in the number of sound sources at my disposal, but because they sound more appealing than almost anything else. Sure, they are not perfect: their screens can whine horribly, they can unexpectedly dump their memories, and they are susceptible to a bug transmitted on one of Korg's own voice cards, the only cure for which is reinitialisation. But despite their faults, their sounds are irreplaceable. Consequently, none of the previous pages will count for anything if the Legacy Wavestation loses the quality and character of the originals. Happily, it doesn't. In fact, as far as I could tell, it sounds identical. The only difference, as mentioned by Paul White last month, is Placing the effects processors in series that the original synths had a sample (above), and (below) in parallel. rate of 32kHz, whereas the Legacy Collection runs on my Mac at a minimum of 44.1kHz, making it a little brighter. We should not be surprised by the fact that the 'vintage' digital synths and their software recreation are identical, because Korg's programmers would have been able to rewrite the original Wavestation code for the G4 and Pentium processors. Nonetheless, I would like to complement them for resisting the urge to update and upgrade. Take, for example, the Wavestation's filter. While this appeared to be a conventional 24dB-per-octave digital low-pass filter, it was not, because it lacked resonance and substituted instead an 'exciter' that (Korg claimed) gave the sound greater clarity and definition. Some people used the lack of resonance as an excuse to criticise the Wavestations, and — to an extent — I have sympathy for this. What's more, the exciter was not what it seemed to be. While it added 'fizz' to many sounds, it was just as likely to reduce the impact of the top end as it was to enhance it. Nonetheless, I'm glad that Korg did not take the opportunity to upgrade the Legacy Wavestation's filter. While the improvement might have been interesting, it would not have been true to the Wavestation. My final point concerns the usability and playability of the new version. Unlike,



say, a Minimoog, for which the playing experience is as much a facet of the instrument as the sound, I've found that the Wavestation migrates into computer form without loss. In other words, the Legacy Wavestation plays and feels like a true Wavestation, with a hugely improved user interface. Who could resist that?

Legacy Wavestation Spec Performances: 11 Banks of 50 (three RAM Banks plus eight ROM Banks). Parts per Performance: Eight maximum. Patches per Part: One. Waves per Patch: One, two or four. Patches per Bank: 35 maximum. Wave Sequences per Bank: 32. Steps per Wave Sequence: 255 maximum, 500 maximum per Bank. Waveforms: 484 in total. Total number of oscillators: 32. Maximum polyphony: 32 notes. Effects: 55 in total. Formats: Audio Units (AU), VST, or stand-alone (no MAS).

Conclusions Unlike other classic synths, which have spawned software descendents sporting hitherto non-existent oscillator sync, effects where none previously existed, and a host of other enhancements, the Legacy Wavestation is a triumph because it remains exactly what it was before; a Wavestation. Sure, it has one significant deficiency compared to the originals (the loss of SysEx) but the improvements in the user interface vastly outweigh this. Like other classic synths, the Wavestation still does what it does better than anything else. Published in SOS June 2004






Latest Sample CDs

In this article:

Cimbalom **** Jazz/World *** Neo Soul **** Vienna Concert Guitar *****

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

Reviews : Sound/Song Library

Cimbalom **** KONTAKT The cimbalom is the largest instrument of the hammered dulcimer family, and the Hungarian national instrument. It spans 4.5 octaves, and is played by striking the strings with mallets. The instrument sampled here has 125 strings, removable legs, and two bars and a foot pedal to control damping. It has been sampled at 24-bit resolution by Dennis Burns of Bolder Sounds, and the programs created specifically to take advantage of the facilities within Native Instruments Kontakt. The library contains 28 programs, the largest 384MB versions providing two switched velocity layers of sustained samples and one of damped samples, crossfaded using the mod wheel. Versions are offered switched at either MIDI velocity 75 or 100 to suit your keyboard and technique, and other programs make use of Kontakt's reverb. Despite only having two layers, these programs are very expressive — the most effective way to play this cimbalom library is with the sustain pedal down most of the time, raising it occasionally to simulate the action of the dampers. A couple of samples in the mezzoforte layer do stick out slightly when soloed, but you don't notice this in a real performance. The remainder of the programs are smaller, using one or two samples per note, but with well-programmed use of the one-pole low-pass filter to provide velocity expression. They are offered with optional reverb, or with one of a wide range of creative treatments whose depth is mapped to the mod wheel, including attack speed (which creates a lovely rich swelling sound at slower settings); high-pass, low-pass, and multi-mode filters; and parametric EQ. There's also an extended version which covers six octaves.

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Latest Sample CDs

Although there are various sampled hammered dulcimers available (including a modest one in Bolder Sounds' own range), as far as I know this is the only cimbalom library in existence, and its richer and deeper resonances provide it with a unique sound. For about the same price as a pair of cimbalom mallets, this library has to be a bargain. Martin Walker Kontakt CD-ROM, $79.95 (around £45). Bolder Sounds +1 303 440 4297. +1 303 442 2025. Click here to email www.boldersounds.com

Jazz/World *** REX 2+WAV Jazz, perhaps the most promiscuous of all musical styles, seems to love getting into bed with other types of music, and clearly isn't picky. There's jazz rock and jazz funk, jazzy house and drum and bass, and, more pertinent to this review, latin jazz and afro jazz. On Loopmasters' Jazz/World sample CD, the emphasis is firmly on the jazzy side of South American and African music — Glenn Miller fans need not apply! The CD features percussion, double-bass, flute, guitar, piano and saxophone loops, plus a multi-sampled double bass (with associated Propellerhead Reason NNXT sampler file) and a range of individual percussion hits. The individual files are clearly labelled with name, tempo, and key where applicable, and are arranged in folders by tempo, although the library is not packed in Refill format as might be suggested by the Reason logo on the packaging. For me, the drum loops are the best thing here — they're interesting, well played, and nicely recorded. They cover 65-140bpm, and for each pattern the kit, the percussion (in most cases congas), and the full ensemble (plus fills for each) are available separately, which means that you can make the limited number of patterns go quite a long way. There are also 64 WAV files of individual hits, covering the kit drums, congas, and djembe, although they've not been arranged into kits for Redrum or patches for NNXT. The double-bass loops are much less polished than the drums. There's a constant high hiss and spill from other instruments is audible if you solo the samples, although the latter isn't really noticeable when combining the bass loop with a drum loop of the same tempo. The performances are also a little haphazard, with lots of finger and string noise, but, on the whole, they steer closer to 'characterful' than 'messy'.


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The four multi-sampled double-bass patches provided for NNXT suffer from similar problems which, in this case, are less forgivable. The samples are rather short, many with a pronounced 'clunk' at the end, and on one of the patches the samples are mysteriously all panned slightly to the left. Another patch suffers badly from hiss, while tuning goes awry in the lowest octave. The remainder of the library contains a smaller selection of flute, guitar, piano, and sax loops, all designed to sit happily with other loops in the same key and tempo. There are some nice breathy flute licks; single-note and chordal riffs covering Latin-American nylon-string and heavily chorused West African electricguitar styles; predominantly Latino piano chords and runs; and some tinny, African-style ensemble sax playing. There's also a folder of way-out solo sax noodling, probably the least World-related material on the CD. Jazz/World's strong suit is that it's very easy (and rather fun) to pick out a drum loop, add bass, guitar, and flute, and knock up a convincing bit of music in just a couple of minutes. But this is also the library's main weakness: while the loops are not overtly presented as construction kits, that's effectively what they are and therefore I fear much of the library will be of limited use elsewhere. David Greeves REX 2/WAV CD-ROM, £39.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.loopmasters.com

Neo Soul **** AUDIO+ACID This four-CD set comprises loops organised into nearly 50 construction kits. While the loops can obviously be time-stretched, the majority of the original tempos are below 95bpm. Each kit includes one or two (occasionally more) drum and bass loops that form the foundation. These are then generally supplemented by some combination of keys, synth, guitar, or string loops, with the occasional brass, percussion, or woodwind loop thrown in for good measure. The loops seem to be very well recorded and many have well-judged reverb or delay processing applied. The drum loops in particular have plenty of punch — and although some users may have preferred the snares to be a little drier, the sound is tight and tasteful. In many of the construction kits, it is Rhodes and wah-wah guitar that are the key elements. This combination, and the dominance of slower tempos, perhaps hints at the styles these loops are aimed at. 'New' or otherwise, the hip-hop and R&B reference in the subtitle is definitely right on the money, but the moods are generally very smooth — these loops are ideal for constructing tracks with file:///H|/SOS%2004-06/Latest%20Sample%20CDs.htm (3 of 6)9/22/2005 7:43:12 PM

Latest Sample CDs

bedroom-orientated lyrics! If you think along the lines of Beyoncé's 'Be With You' and 'Me, Myself And I' (both from Dangerously In Love) or The Black Eyed Peas 'The Apl Song' (from Elephunk), then you will be in the right ball-park. Some of the construction kits might also work in other musical contexts. For example, it is easy to imagine Sting's voice sitting over the kit named Deep Pockets, with its harp and acoustic-guitar loops. Equally, at really slow tempos, the combination of gentle Rhodes parts and the occasional wah guitar line might work for a chill-out track (think 'Passing By' from Zero 7's When It Falls). In use I found that it proved perfectly possible to mix and match between kits, as long as I took care when combining a chord sequence from one kit with a melody line from another. Perhaps my only criticism would be the relatively few drumloop variations within each kit, so you might quickly find yourself reaching for Recycle. The drum sounds used are pretty consistent across a number of the construction kits, however, so loops could also be combined in that way. Otherwise, if mellow, classy R&B or hip-hop is your thing, Neo Soul provides a fine collection of loops and represents good value for money. John Walden Audio CD and Acidised WAV CD-ROM 4-CD set, £56.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.bigfishaudio.com

Vienna Concert Guitar ***** GIGASTUDIO/EXS24 MKII Vienna Symphonic Library's Horizon series gives a taste of their fabulous samples at affordable prices. As well as presenting themed titles based on their megatastic 250GB orchestral Pro Edition, the Horizon disks branch out into nonorchestral territory, offering a brace of fine guitars, the first of these being the 11.9GB Vienna Concert Guitar. Played finger-style throughout, the nylon-stringed acoustic guitar has a rich, mature tone which is capable of both intimacy and bite. Its sustained notes (played with and without vibrato) were sampled at three dynamics, with an extra layer of metallic 'sul ponticello' (played near the bridge) samples kicking in at high velocities. For more lively rhythmic passages, there are staccato notes (ideal for jazzy chordal stabs), and muted étouffé (choked)


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samples, while softer, flautando performances give a subdued, almost harp-like timbre. Some strident Bartokstyle string snaps could be useful for Flamenco simulations! Other variations include percussive 'hammered' lefthand notes, tremolos, and a comprehensive, very pretty set of natural harmonics. Accompanying these Cinderella multisamples are the usual 'ugly sister' noises: woody knocks and thumps, string squeaks, behind-the-bridge plinks, and so on. There are no licks or runs, but semitone and whole-tone trills, along with some rather laborious semitone bends, offer some pitch variation. There is also a huge selection of consonant and dissonant chords, including some of the jazzier shapes. These are presented in a choice of six-, five-, or fourstring voicings, and a separate menu of three-note chords is also supplied. All chords are played in a variety of strumming styles (including Flamenco-style tremolo), open, muted, and arpeggiated, each available at the push of a keyswitch. The mapping system is very flexible: you can choose a single chord type mapped chromatically, or select all the chords based on a particular scale (say, 'E' major) — an impressive organisational feat. No VSL release would be complete without the mighty Performance Tool, a fantastic, expressive MIDI device whose legato mode produces convincingly joined-up melody lines. The sharp attack of a single, articulated classical guitar note militates against the sort of fluid legato produced by a violin or wind instrument, but when used in conjunction with carefully edited 'performance legato' samples of hammer-ons, pull-offs, and glissando slides, the effect can be breathtaking. The Tool also handles real-life performance repetitions of up to nine notes in a variety of styles, tempos and dynamics. For their Horizon series, VSL devised two new legato modes, which have been incorporated into version two of the Performance Tool — earlier versions do not function properly with the Horizon legatos, as they use a different note-mapping system. Needless to say, the recording quality, miking, and overall attention to detail is exemplary. This superior instrument's precise, classical delivery ensures that melody lines can be articulated with great clarity, and its distinct, slightly sharptoned attack works well for emphatic rhythm licks and arpeggios. Some might prefer a plummier, more rounded sound for use in quieter ballads, but if it's realism you seek, nothing on the market comes close to this classy title. Dave Stewart Gigastudio or EXS24 MkII DVD-ROM, £177 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400.


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Click here to email www.timespace.com www.viennasymphoniclibrary.com

Star LEGO Follies ***** Rev Smith's Brick Testament **** Luke Skinner's David Blaine In A Box *** Henry Lim's Harpsichord ** Erich Harshbarger's Grandfather Clock * Dave's Bricktannia Published in SOS June 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

Getting Started System Requirements Beat Machine Intakt Live? Time Machine Loop Pool Modulation And Effects Intakt Instruments Overall

Native Instruments Intakt £150

Native Instruments Intakt Sample Looping Instrument: Mac & PC Published in SOS June 2004 Print article : Close window

Reviews : Software

NI have taken the technology behind Kontakt, and used it to build a sampler that's all about loops.

pros Beat slicing, granular time/ pitch warping, and standard sample playback in one plugin. Quickly syncs samples to the tempo of your song, no matter which 'engine' you use. Can read just about any format. Real-time sync'ing of loops played from keyboard. A lot of technology for your money.

cons Can't resize the small waveform window so editing is fiddly. Limited manual. Can't undo. Modulation options and effects could be better.

summary Probably the most sophisticated dedicated looping tool available, and the only one that combines Recycle-style beat slicing with granular time-stretching, Intakt has plenty to add to any of the hosts it can run in, and quickly becomes the first thing you reach for when bringing in loops of any format.

information

Simon Price

Intakt is the third member of Native Instruments' Holy Trinity of samplers, the other two being Kontakt and Battery. Kontakt is the 'daddy' full-featured sampler package, while Battery is a sophisticated sample-based drum machine. Intakt lays claim to different territory, being a dedicated loop tool. There's been quite a resurgence of loop and beat-slicing software of late, probably spurred on by Reason's Dr. Rex loop player. Plug-ins like Phatmatik Pro and Beatburner are appealing to a whole new wave of desktop music producers who want to create something new from loops, rather than just slot together library material. So what does Intakt have to offer to make it stand out from the crowd? Well, for a start, as far as I'm aware it's the first of its kind to combine Recycle-style beat slicing with granular time-stretching of the kind available in Live and Acid. Before we get stuck into specifics, what exactly is Intakt for? Well, it has as number of uses, but at the simplest level it loads sampled loops and plays them back in time with your sequencer. Intakt has three distinct methods of achieving this, all of which are possible in various other samplers, but are packaged together here in a unit dedicated to the task. Two of the available methods (or sampling 'engines') can alter the tempo of the audio loop without affecting its pitch, and vice versa. As well as simply playing back samples, Intakt can manipulate, modulate, rearrange and otherwise mess with the loops loaded into it. As its raw material Intakt can read standard audio files in most formats as well as Recycle (REX/RX2), Kontakt, Battery, Akai, EXS, LM4 and Soundfont 2 files. As such it makes a very good access point to the majority of loop libraries, and has in fact been adopted by some sample CD manufacturers as a supplied front end (see 'Intakt Instruments' box). Intakt itself comes with a 1.2GB library of loops picked from Zero-G's and East West's catalogues.

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£149.99; Intakt for Kontakt users 99 Euros; upgrade from Intakt Instruments 129 Euros. Prices include VAT. Arbiter Music Technology +44 (0)20 8970 1909. Click here to email www.arbitermt.co.uk www.native instruments.com

Like all NI software, Intakt supports a wide range of formats. It can run as an Audio Unit, VSTi, DXi and RTAS plug-in, and can also run stand-alone using ASIO, Core Audio or Direct Sound. The installer lets you choose which plug-in formats you want to use, or you can install all of them. You then have a 30-day grace period to handle the registration and copy protection procedure. A separate registration tool application gets installed with Intakt, which generates a machine-unique challenge code to be sent to NI. If your studio computer is online this happens instantaneously, otherwise the utility generates a file that you can email from another machine. When you get the response code, you run it through the registration tool again and your machine is authorised.

Test Spec Intakt v1.03. Apple iBook 800 running Mac OS 10.3.2, with Cubase SE v1.06. Apple G4 dual 1.2GHz running Mac OS 10.2.8, with Digidesign Mix system and Pro Tools v6.1.

Getting Started After having a good play with the factory sounds (see the 'Loop Pool' box) using Intakt in stand-alone mode, I decided to get stuck in with one my own loops. As I've already mentioned Intakt gives you a choice of three options for handling each loop: Sampler, Beat Machine and Time Machine. I thought I'd try them all with the same loop. I wanted to use the RTAS plug-in version with Pro Tools but this wasn't working at first, so I fired off an email to NI's support and switched to Cubase SE. As you'd expect with a plug-in, everything is presented in one window, divided into modules as in Kontakt. Native Instruments have standardised much of their line, Intakt included, with a proprietary file browser that slides out to the left of the window. The top pane displays the directory hierarchy of your drives, while the bottom lists the readable files in the highlighted folder. While this is a little fiddly at first, it does ensure consistency between Mac and PC, and will also be familiar to Ableton If you tell Intakt the length of your loop in Live users. The browser also features bars and beats, it will calculate the tempo. a format filter, user favourites directories, and an audition system. Double-clicking my loop loaded it in, displayed the waveform in the Source Editor module (the top section of the window) and assigned it to the first MIDI key for playback. Alternatively, samples can be dragged from the browser onto the keyboard display to choose the key mapping. When doing this you can position the cursor higher or lower above the keys, and a coloured range indicator lets you spread the sample over several keys. When first loaded, a loop defaults to being handled by the Sampler engine, indicated by the orange button at the left of the Source Editor. This operates like any traditional sampler, pitching the sample up and down to adjust the playback



speed and vice versa. The sample can be pitched over several keys by adjusting the red range marker above the keyboard, with the root key (original speed) being set by the white triangle beneath. So far, this is pretty normal, and of little interest to us as far as looping is concerned! Much more important are the tempo and sync'ing controls located to the left of the waveform display. The larger of these two dials indicates the master tempo, and is locked when you are running as a plug-in because it's sync'ed to the song's tempo. The smaller dial is the playback speed of the selected sample, measured as a ratio against the loop's original tempo. Critically, this can be locked to the master tempo, ensuring that the loop is automatically in sync with the song. However, before this can work you need to tell the software how long the loop is in bars and beats. Pressing the small clock icon between the dials opens up a window for setting sync options. From here you enter the loop length, from which Intakt will derive the original tempo. A retrigger length, combined with the Retr button on the main panel, will start the sample from the beginning at each loop length, avoiding any tiny length error accumulating over time. From now on, the assigned MIDI key will run the loop in time with the song. Anyone who's sync'ed up a loop sample on a traditional sampler will appreciate how much time's just been saved, and we haven't even got to the really clever stuff yet! As you can probably see from the display, you can set in and out locators, and also specify a separate loop zone for looping a second portion within the main sample. If there a loop is written in the original file by a separate editor like BIAS Peak, this will transfer to Intakt.

System Requirements Mac: Mac OS 10.2.6 or higher, 500MHz G3 or better CPU, 256MB RAM. PC: Windows XP, 500MHz Pentium III/Athlon or better, 256MB RAM.

Beat Machine While the Sampler engine has the advantage of automatically pitching the sample to sync with the song, it still suffers from the fact that pitch is inextricably tied to speed. That's where the Beat Machine and Time Machine engines come in. Beat Machine uses the trick of 'beat slicing', which will be familiar to users of Recycle and REX loops. This method chops the sample up into slices, and achieves tempo adjustment by sliding them apart, or closer together. The technique only works transparently on rhythmic drum and percussion loops, or music broken into staccato transient hits or notes. Note that an Intakt patch (or 'instrument' in NI-speak) can have lots of different loop samples spread across the keyboard, using a mixture of engines. The keys are colour-coded, with Sampler-based loops in yellow, Beat Machines in blue and Time Machines in red. As you can see from the screen shot (right), the sample is analysed and white



lines appear in the display, marking where Intakt thinks the individual hits of the drum loop occur. The Sen dial to the right of the waveform display adjusts detection sensitivity, generating more or fewer beat markers. You can play back each slice individually by clicking it in the display, revealing where you need more slices. With my loop, pushing up the Sen a little got everything except for one quiet hit in the last slice. Pushing any further Beat Machine is designed to detect individual resulted in lots of 'false alarm' markers hits within a drum loop and slice it up appearing all over the place. At this accordingly. point you have to bring the sensitivity back down a bit and manually add what's missing. To do this you zoom in and find the peak in the waveform, then right-click (or Ctrl-click on the Mac) to add the slice. Slices can also be nudged earlier or later. This brings me to my biggest gripe about Intakt: you can't adjust the size of the waveform display. The display is way too small for comfortable editing and really slows you down, especially if you're used to using Recycle to do the same task. However, it gets the job done in the end. As with the Sampler, the loop and retrigger lengths must be entered in the sync options panel. Now when you play back, the loop is in sync, but this time the original pitch is preserved. At first, the whole loop is assigned to one MIDI key, and can be played back by a held note in the sequencer. This feature makes using beat-sliced loops particularly quick and simple in Intakt. By widening the range of keys assigned to the loop, you can adjust the pitch of the loop, without changing the tempo. Traditionally, beat-slicers assign each slice to a different key, with the loop triggered by a MIDI sequence that steps through the keys. This gives the extra flexibility to re-sequence the loop (or strip parts out) by editing the MIDI notes, and is possible in Intakt by switching from Global to Sliced edit mode. As soon as you enter Sliced mode a window pops up with options for mapping the slices to individual keys (left). As well as key mapping, the window provides a few other options, including adding short start and end fades to smooth playback, and artificially stretching the end of each slice. The 'end stretch' smooths playback when you reduce the tempo, filling the gaps that are left as the file:///H|/SOS%2004-06/Native%20Instruments%20Intakt.htm (4 of 10)9/22/2005 7:43:16 PM


slices move apart in time. In Intakt, stretching is achieved by introducing back-and-forth looping over the percentage of the loop specified in the option box. I found this technique very successful for slices containing smooth audio at the end. Inevitably it doesn't work so well for erratic audio. One Various options are available for mapping the slices of a loop to MIDI notes. shortcoming of Intakt is that to change the stretch percentage you have to go through the slice mapping process again, losing any subsequent editing from the previous time. The last option in the mapping box is to export a Standard MIDI file that will play back the slices in their original order (and can be edited to change the loop). The resulting file is imported into a sequencer track. This works fine, although I'd have liked a quicker method, such as that in Reason, where you can dump the MIDI directly into a track. It's nearly always worth mapping the slices, even if you don't use a MIDI file, as it lets you alter each slice individually in the modulation and effects sections. Additionally it lets you sequence the slices in any order via the keyboard. However, Intakt has another method of rearranging slices, while still playing back the loop from a single key. The top right of the window has a slice order pop-up, where you can switch the position of each slice in Intakt's internal playback sequencer. There is also a slice order randomise function in the Command menu, which is lots of fun and can quickly transform your loops into something unexpected.

Intakt Live? One thing that kept occurring to me was that Intakt would make a good live looping station. The fact that you can run it stand-alone, with multiple loops spread across the keyboard, and using different looping methods for each, sets it apart from the competition. Also, within a patch you can split off each sample to different outputs. MIDI Controller mapping is The keyboard quantise options allow you to implemented: Command-clicking (or Ctrl-clicking on PC) opens up a trigger loops exactly in time from the keyboard. controller page, and all you have to do is wiggle the desired input device and it gets learned. Intakt's best live tricks, though, are the keyboard quantise and latching functions. The quantise function lets you start your singlekey loops and samples exactly in time, based on your choice of time division. For example you can set it so that loops will only start or stop at the nearest bar to your key press, while smaller divisions let you offset loops but stay in sync. The Latch function keeps a key held down until you press it again, so you can file:///H|/SOS%2004-06/Native%20Instruments%20Intakt.htm (5 of 10)9/22/2005 7:43:16 PM


quickly switch loops on or off. Unfortunately, on my system Latch only worked from the on-screen keyboard, whereas you really want to use it from your MIDI keyboard. However, Intakt does use the standard Native Instruments trick of turning your computer keyboard into a playback device, and Latch does work with this, so all you need is your laptop!

Time Machine Chances are that if you mainly use drum and percussion loops, most of your work will be in the Beat Machine. However, there are limits inherent in the method, as many loops will not sound right when chopped up and tempoadjusted. For example, in my example loop the last few slices have hits that overlap, and the sound breaks up when slowed down. The final trick up Intakt's sleeve, Time Machine, can overcome these problems, and works with any kind of material, including vocals. This engine uses granulation, or granular resynthesis time-stretching, to give it its full name, to fit samples to the tempo. This technology also gives you separate control of pitch with respect to speed. It's the same technique used by Acid and Live to 'warp' audio. As before, Intakt needs to know the length of the loop, but after that no further preparation is needed for temposync'ed playback. The sample's playback speed is automatically adjusted to stay in tempo. As with the Sampler, the key range can be extended so that you can play back at different pitches, but this time the tempo is unaffected. Although this is bound to be the most common way of using the Time Machine, you are free The Time Machine sampling engine uses to switch off tempo sync, and play with granular resynthesis time-stretching to the speed controller to create some manipulate the tempo and pitch of a loop mental-sounding granular synth-style independently. effects. In fact, loops aside, I found Intakt really useful for playing back long pad and atmosphere samples at different speeds. The Legato option means all keys will lock to playing the same point in the sample at any time: nice! Obviously, Time Machine is not a miracle worker, or else you would use it for everything. The tempo cannot be changed too drastically or the sound begins to suffer, sounding 'grainy' or 'burbly'. The usable limit varies depending on the type of sound. I found that most material, including drums, could be adjusted by up to 10 percent without too much trouble. Some sounds are more amenable to the scheme: for example, I found that some analogue synth loops that sounded perfect right down to half their original speed. The Source Editor contains a grain size setting, which optimises the algorithm for different material types, although file:///H|/SOS%2004-06/Native%20Instruments%20Intakt.htm (6 of 10)9/22/2005 7:43:16 PM


it's not labelled as such and the manual makes no mention of it or how to use it! There is also NI's ingenious Transient Copy function available (from the TRC button and TRS knob), which seriously improves the quality of drum and percussion loops when stretched. Like the Beat Machine, TRC detects transients and tries to preserve their integrity during playback. This stops drum hits from being 'granulated' and sounding crumbly, as isoften the case with this kind of technology. I tried it on a number of things and it really works: I reckon it doubles the usable tempo range for drum loops.

Loop Pool The 1.2GB sample library included with Intakt is packed into two files containing the audio data and a folder of programmed patches ('instruments'). If you use this folder for your own instruments you can load them from the patch selector, without having to wade through the main file browser. By my reckoning there are over 1000 patches split into 10 categories. Most of the further subcategories have a master patch called, for example, 'All Breakbeats', which has all the main loops mapped out across the keyboard on one key each. This is incredibly useful when browsing through the samples. Once you think you've found what you want you can load the main instrument for that loop, which will have more sophisticated, key-mapped patch programming. As for the content, there is of course a broad variety, mostly of pretty good quality. Strangely, the first set of drum loops has the least successful beat-slicing — you'd think that they would want to start off the list with something really good. Luckily, it gets better quickly. On the whole, sensible choices have been made for which engine to use for each sample. As expected, the majority of the drum-based loops use the Beat Machine option, switching to Time Machine where slicing is inappropriate. It doesn't sound as though many of the programmers got into the end-stretching function, because most sliced stuff gets gappy when slowed down. Many of the 'keyboard' loops really need sorting out, having been saved with strange bar lengths and tempos that don't work at normal speeds. However, there are some outstanding highlights, such as the orchestral loops section. These are all short ensemble phrases from East West's Symphonic Adventures library, and could be described as 'movie scary bits'. Most of the percussion stuff will be useful, with the African percussion patches having been particularly well programmed. The sound-effects and ambient material is also quite inspiring. While the library is a welcome addition, and useful for seeing what's possible, I suspect most people are going to want to start mashing up their own loops straight away.

Modulation And Effects While you can mess around with samples a fair bit in the source editor, more sound-shaping possibilities can be found in the modulation and effects sections of the plug-in. As you can see from the screen above, you get an AHDSR envelope generator, an envelope follower, and two LFOs, all of which can be assigned to different destinations. The pitch envelope is a highlight, especially



with Time Machine, as you can mess up the whole loop without affecting the timing. The best fun with loops is being able to trigger envelopes at each hit, which is what happens with Beat Machine. In fact you can have different settings for each slice, so there's scope for much weirdness there. However, with the Sampler and Time Machine there are no slices, making the envelope follower essential for doing anything other than having envelopes triggered at the start of the loop. To be honest, though, it feels a bit restrictive: you really want two normal envelopes for beat-sliced zones, and two followers for the Time Machine. A compromise would be if you could send the envelope and follower to more than one destination at a time, but you can't do this either. I though I might be able to sync up the LFOs to use as pseudo envelope generators, but when Intakt is in Sync mode you can't set the LFOs to any subdivisions of time other than the loop length. Effects-wise you get multi-mode filter, lo-fi, distortion and delay modules. To be honest the filter is good, but the rest is nothing to shout about. Lo-fi is just a bitdepth and sample-rate reducer, which let's face it has very little scope. Rather than being fat and beefy, on most material the distortion is quite scratchy and thin. Overall, despite the nice pitch mod, and the thoughtful inclusion of the envelope follower, Intakt is a little underdeveloped on the sound-shaping front. There is certainly more fun to be had in close rival Phatmatik Pro, which has a versatile mod matrix, separate envelopes for everything and cool slice-looping options. Maybe NI will feel generous and include the Step Mod module from Kontakt in Intakt 2!

Intakt Instruments Given Intakt's ability to load just about every type of file format (including REX) it's not surprising that it's been picked up by East West, Zero-G and Best Service to act as the front end for several of their sample loop libraries. These libraries ship with a version of the sampler called Intakt Instruments, which has all the functionality of the full version but can only load the loops that make up the library. The built-in file browser is absent and you simply load samples from the bundle from the main patch selector. This kind of symbiosis makes a lot of sense, as the sample library is no longer limited to one format. What's more, the bundled player can alter and rearrange the loops so they can be reused, extending the use of the library. As a bonus, if you buy one of these libraries you get a discounted upgrade price if you later choose to buy the full version of Intakt.

Overall Intakt was very stable on my system running stand-alone and as a VSTi,



although I never got to the bottom of why it wouldn't run on my Pro Tools LE system, despite quick responses from NI tech support. I did, however, use it with no trouble on another Pro Tools system so I know it can be done. Performance was surprisingly good, even on my humble G3/800 Mac. Screen redraw dropped to a crawl in Cubase, but not in the plug-in itself, and came right back up when the plug-in window was closed. Sampler and Beat Machine are inherently lowimpact processes, but even Time Machine didn't impose as much CPU load as I'd expected. The only major gripe I have other than the size of the edit window is the manual. It's very brief, misses a fair bit out, and on occasion alludes to things that don't exist! I have to admit that when NI started releasing different samplers I was a bit doubtful, thinking there would be too much overlap. However, I was proved wrong with Battery, and this is reaffirmed by Intakt. Intakt is not The sampler functions are complemented by various modulation and effects options. Kontakt Lite: it stands up on its own as a streamlined tool for getting a particular set of jobs done quickly and efficiently. Although Kontakt has the same sampling engines, and also has Intakt's essential beat-slicing functionality, they are part of a more cumbersome whole, much of which is superfluous for running loops and certain other sample-playback jobs. There's also a certain financial flexibility: if Intakt does everything you need, or you already have a sampler, you can get your hands on NI's powerful looping tools at half the price of Kontakt. Plus, if you already have Kontakt you can get Intakt for half the normal price (just 99 Euros). But do you need it if you already have Kontakt? Well it does have advantages. Setting up the relationship between tempo and pitch, and how they sync to the master tempo, is really easy in Intakt, and the global and slice-based modulation options are all ready to go. Intakt extras also include slice order rearrangement, and quantised keyboard triggering for manually playing back loops in sync. As for comparisons with other products, other beat-slicing options come to mind, such as Phatmatik, or Recycle teamed up with a compatible sampler, but Intakt wins out for me with all its extra functionality, not least being the Time Machine's real-time tempo warping. However, the fixed waveform display means that you're still better off using Recycle to prepare your own loops before importing them into Intakt. On the whole, I can see Intakt becoming a core part of my set up, and if you're looking for something that has all the latest looping tools, I doubt you'll do better than this. Published in SOS June 2004






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In this article:

PSP Master Q TLL Everyphase Fabfilter Fabfilter One Cranesong Phoenix Audiorealism Bass Line Sonic Charge µTonic

Plug-in Folder Latest Plug-ins reviewed Published in SOS June 2004 Print article : Close window

Reviews : Software

PSP Master Q Formats: PC Direct X & VST; Mac OS X VST Master Q is PSP's first release in the EQ arena. At its heart are seven filters, comprising low cut and high cut at 12 or 24 dB/octave for trimming unwanted frequencies at the extremes of the audio spectrum, low-shelf and high-shelf filters for general-purpose warming and cooling over a wide band, and three overlapping peaking filters. All seven are fully parametric, with frequency adjustable over a wide range, plus variable Q, again over a range that's significantly wider than most competing products in my collection — this doesn't necessarily make Master Q better, but it certainly makes it versatile! High Q values create a characteristic hump at the corner frequency and a double resonant peak at shelf transitions, just like those of Waves' Renaissance and Linear Phase EQ designs. All seven filters offer up to 24dB boost or cut, and Master Q's combined frequency response is displayed graphically, with a black line for the overall response, and different coloured lines showing the contribution of each interacting band. You can switch the resolution of the graph, while a Range knob provides an overall boost or cut for all seven gain controls — negative values invert the entire response. This is already an extremely versatile EQ, but like most PSP products, it doesn't stop there. Next to the Proc (bypass) button is one labeled FAT, which stands for Frequency Authentication Technique. This activates a double-sampling option to move any non-linearities and phase errors beyond the normal audio band to ensure sweeter and more 'analogue' results at high frequencies — a similar technique is offered by Spectral Design's QMetric. FAT's only down side is an extra 64-sample latency and an increase in CPU overhead. Master Q already consumes more power than many EQ plug-ins, taking about 3 percent of my file:///H|/SOS%2004-06/Plug-in%20Folder.htm (1 of 10)9/22/2005 7:43:21 PM

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Pentium 4 2.8GHz processor with all seven bands active without FAT, and about 8 percent with it on. Two small arrow buttons hide another of Master Q's strongest features: Lim-Sat. They allow you to select one of seven soft-clipping algorithms. VintLim is based on Vintage Warmer's single-band limiter for the most transparent sound, DynSoft and DynHard offer dynamic saturation and tighten attack and release times close to 0dBFS and are ideal for drums, LimSoft and LimHard are similar but with longer release times for a smoother sound, and SatSoft and SatHard are derived from PSP's Mix Pack and provide more obviously saturated and distorted results. I auditioned Master Q against quite a few other EQ plug-ins, including Waves' Renaissance and Linear Phase, TL Audio's EQ1 and TC Works' Native EQ, and they all sounded different. To my ears, Linear Phase was always the sweetest and most transparent, while Master Q sounded closer to the Renaissance and Native Parametric EQ, although of the three I often preferred Master Q depending on the programme material. My old favourite EQ1 brought up the rear in most cases, sounding completely different again. Master Q's VintLim algorithm proved really useful in avoiding the harsh sound of digital clipping without adding any obvious character of its own, although it can do so if you drive it hard. PSP's SatSoft option was most similar to the TC Works SoftSat algorithm; the EQ1 'tube' saturation was altogether different from any of the others and rather more subtle. I was most impressed by Master Q, for both its sound and PSP's trademark versatility. Although EQ preferences are a matter of taste, I found Master Q to sound very good, with extremely flexible curves, and it's far more versatile than the competition. With the transparent VintLim in circuit it's far more tolerant than Waves' Renaissance EQ when pushed, but if you want an EQ with character you can add plenty of creative attitude. I only wish PSP could perform such feats with lower CPU consumption, which while low in absolute terms, is still four or five times that of the majority of the mid-range competition. Perhaps we can't have it all. Martin Walker $149 (EU and Polish customers are also liable for VAT). Click here to email www.pspaudioware.com

TLL Everyphase Formats: Mac & PC RTAS & TDM When conversation turns to the inferiority of plug-in processors, it's usually compressors and EQs that are done down. In my experience, though, some of the worst offenders are delay-based effects such as phasers, flangers and choruses. They may be designed to add thickness or richness, but all too often

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the result is the exact opposite — a thin, watery sound with no body to it — whilst where you expect subtle motion and expansive stereo presence, you get the annoying, repetitive cycling of a single LFO. With this in mind, I was interested to try out Trillium Lane Labs' new Everyphase, which as the name suggests, is a flexible plug-in devoted to conventional and not-so-conventional phasing effects. It runs as a TDM or RTAS plug-in on Pro Tools systems and is available for Mac and PC, with authorisation through iLok. To get you started, there's a healthy array of presets, divided into categories such as Bass, Drums, Guitar and so on, and a prowl through these is enough to show that Everyphase is both versatile and goodsounding. At one end of the spectrum, it delivers subtle thickening and widening effects that don't sound overtly treated; at the other, there are jarring bursts of modulation, submarine squelches, broken Leslies and alien voices. Inbetween, there are hundreds of flavours of hazy warmth, offering all the good features of stomp-box phasers without the noise or the appetite for batteries. These impressive results are generated by a phaser algorithm which offers up to 18 stages, with a Resonance control that allows you to specify which stage is tapped for the feedback circuit, and Depth variable from -100 to +100. Modulation can be derived either from a built-in LFO or from an envelope follower, and it's the wide range of options in this department that helps to make Everyphase so versatile. The envelope follower can be fed either by the signal being processed, or via a side-chain, and its output is available as a trigger for the LFO, as well as as a modulation source in its own right. The LFO can also be triggered manually or via MIDI Beat Clock, with a wide range of LFO shapes available. When it's linked to host tempo, you can specify how many times it should cycle per beat or bar. A neat animated display shows the modulation in progress. As you can probably tell, I was quite impressed by Everyphase. In terms of richness and analogue flavour, it doesn't always match Eventide's Instant Flanger and Instant Phaser, but its superiority over run-of-the-mill modulation effects is immediately obvious. And it's much more versatile than Eventide's plugins, which are faithful recreations of ancient hardware devices; Everyphase, for instance, makes full use of possibilities such as tempo sync. Other bonuses include support for the Multishell architecture, allowing it to share a DSP chip with other compatible plug-ins, and the ability to work as a true 5.1 effect in HD Accel systems. Whether your interest lies in vintage stomp-box sounds or in more experimental areas, it's definitely worth investigating. Sam Inglis $ UK price tbc. Digidesign UK +44 (0)1753 655999. +44 (0)1753 658501. Click here to email


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www.digidesign.com www.tllabs.com

Fabfilter Fabfilter One Formats: PC VST & stand-alone Fabfilter One is the first product from new Dutch company Fabfilter, and is a small monosynth that comes in Windows VST and stand-alone formats, with Mac OS X support under development. It's sold as a 1.7MB download that requires the entering of a licence code to activate from its trial version. Fabfilter have obviously taken a lot of care to get the synth's single oscillator sounding just right; the sawtooth and square waves are very harmonically rich with just the tiniest hint of noise sometimes apparent, which gives them a nice analogue edge. This is particularly evident in some of the bassier presets, which sound much more full-bodied than you'd expect from a single-oscillator soft synth. Adding in some white or pink noise from the noise generator can produce some very warm sounds. As you'd expect given the name, the filter plays a big part in the little synth's appeal. Whilst it lacks the deliberate grunginess that tends to mark filters out from the crowd these days, Fabfilter have certainly produced a very satisfying low-pass design. It gives nice chunky sweeps with just the right amount of resonant shrillness available at the top end, and seems entirely free of any undesirable aliasing artefacts. In keeping with the rest of the synth, the modulation options are pretty basic, with just the main envelope and a single LFO available for modulating the oscillator frequency, filter cutoff, and square-wave pulse width. You can, however, invert the envelope separately for each of the three destinations, whilst the LFO provides continuous degrees of 'slope' for the triangle wave and variable pulse width for the square. All round, whilst Fabfilter One is unlikely to be the most versatile instrument in your collection, it does what it sets out to do with style and polish. This is most noticeable in the nice preset system and way the knobs respond to both linear and rotary movements depending on whether you click on the centre or the 'grabable' bit at the edge. In a world abounding with freely downloadable VST instruments, stumping up for file:///H|/SOS%2004-06/Plug-in%20Folder.htm (4 of 10)9/22/2005 7:43:21 PM

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a simple monosynth might not come naturally, even if you consider the bonus of a stand-alone version. Those who do, however, will find Fabfilter One a product that sounds great and which shows a lot of love invested in its design. Mike Bryant $79 Click here to email www.fabfilter.com

Cranesong Phoenix Formats: Mac & PC TDM Dave Hill is a very highly respected designer of analogue equipment: some producers and engineers, such as Elliot Mazer, consider his Aria-modified ATR reel-to-reel recorders the best in existence, and his own company Cranesong are renowned for their high-end analogue processors. So when he announces that he's cracked the perennial problem of introducing analogue warmth to digital recording, it's clear that his efforts should be taken rather more seriously than some. The product Cranesong have come up with is Phoenix, a suite of plug-ins for Pro Tools TDM and HD systems on Mac OS and Windows XP which is designed to reproduce the audible effects not only of tape saturation, but of the other elements in a typical tape machine signal path. Rather than create a single plug-in with numerous parameters and a potentially high DSP load, they've produced five separate versions, each with slightly different sonic qualities. Presumably the various versions are designed to reflect the effects of different tape formulations and recorder settings, but the very brief documentation doesn't explain. Each of the five plug-ins — Phoenix Iridescent, Dark Essence, Luster, Luminescent and Radiant — has the same interface, and it's clear from the controls that this is intended to be one of those 'Make it sound better, and don't bother me with the details' products. In each case there are three operational modes, selected using buttons labelled Gold, Sapphire and Opal, plus a large dial labelled Process Level. You can also adjust the input trim, but to do this you have to click in a box and enter a trim level manually. A simple slider control would have been nice. So what does Phoenix actually do? Well, the process seems to be a subtle file:///H|/SOS%2004-06/Plug-in%20Folder.htm (5 of 10)9/22/2005 7:43:21 PM

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combination of 'tape compression' (ie. frequency-dependent soft limiting with zero attack and release times), equalisation and harmonic enhancement. The operative word here is 'subtle': if you want to turn your drum loops into a pumping mess, or recreate the Strokes' vocal sound, this is not the product for you. But don't let its relative mildness fool you into thinking that it can't make an important contribution to your recordings. In each of the five plug-ins, Gold mode seems to be the most neutral with respect to the frequency content of the sound, whereas Sapphire adds presence in the upper mids and Opal thickens the lower mids. As you turn the Process Level knob towards 0dB you'll also hear a hike in the overall loudness of perhaps 4 or 5 dB. This makes Phoenix an ideal way to subtly boost the level a signal without running into clipping, though it also makes it hard to A/B the processed and unprocessed sounds. I tried Phoenix on a variety of instrumental and vocal sources, and it was rare to find a track that wasn't improved by at least one of the five plug-ins — more often I found myself struggling to decide which of them I liked the best. The dynamic element of the process is especially good for adding body to sounds that might otherwise be a little thin, whilst the Sapphire and Opal settings emphasise useful areas of the frequency spectrum in ways that usually sound very natural. The differences between the five plug-ins are more noticeable on some sources than others; again, the frequency response seems to vary subtly, whilst transients are also affected in different ways. Of the five, Dark Essence, Radiant and Luster tend to have the most obvious effects; I found them ideal for sources such as distorted guitars and drums, whilst the more gentle Iridescent and Luminescent provide sympathetic treatment for acoustic guitars and the like. You can also try using Phoenix across the mix buss, although this is one area where it's easy to overdo things. On a purely practical level, Phoenix scores for its low DSP load (about 13 percent of a Mix DSP chip), simple interface and implementation of the Multishell architecture, which allows it to share DSP chips with other plug-ins, but I'd knock a couple of points off for the lack of an output gain control, which would have made it easier to A/B the dry and processed sounds. And on a sonic level, I can't think of anything else quite like it. It's a well-thought-out attempt to introduce analogue flavour to digital recording, with results a million miles from the overdone, muddy distortion that often passes for 'warmth' in similar products. There will always be engineers for whom only a fully analogue recording path will do, but I suspect that Phoenix might just help to persuade a few diehards to convert. Sam Inglis £305.50 including VAT. KMR Audio +44 (0)20 8445 2446. +44 (0)20 8369 5529. Click here to email www.kmraudio.com www.cranesong.com


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Audiorealism Bass Line Formats: PC VST If you are a fan of the original Roland TB303 but can't find (or afford!) a good second-hand example, Audiorealism's Bass Line — the company's first commercial plug-in — might just do the trick. I was supplied with version 1.1 of the PC VSTi (a Mac OS X version is in development) and, as can be seen from the screen shot, the user interface of Bass Line contains all the essential elements that are found in the original hardware TB303. The basic structure of the synth is monophonic and features one VCO and one VCA, both of which are controlled by an envelope generator with decay and modulation controls. The waveform can be toggled between a sawtooth or square wave. A low-pass filter can be switched between a 'classic' 18dB slope modelled on the original TB303, a somewhat brighter-sounding 'pure' 18dB, or a less bright 24dB mode. The central strip of virtual knobs includes tuning, filter cutoff, resonance, envelope modulation, decay and accent, while a distortion circuit (with drive and distortion amount controls) is positioned at the top alongside the overall volume control. Bass Line can operate in two modes, PTN and Note. The former uses the built-in pattern sequencer, while the latter receives the note sequence from a MIDI track in the host application. Patterns can have up to 64 steps and the Pattern Interface (the grey area that makes up the bottom half of the window) can be used to enter note information, rests, slides or accents. In PTN mode, each of the 127 patterns can be triggered via a MIDI note, with C0 triggering pattern 0, C#0 pattern 1, and so on). Audiorealism have included some useful additional features. For example, from the Options button, the tuning can be switched between a stable 'digital' mode and a non-exact mode that tries to simulate the way old analogue synths go out of tune as they warm up. A Reason-like MIDI Learn mode is also included, allowing the Continuous Controllers on your master keyboard to be linked to one of Bass Line's virtual controls. While it is possible to coax a range of bass or lead sounds from Bass Line, the synth's architecture means this is never going to be the most versatile of sound sources (but then, neither was the original TB303!). This said, in testing within Cubase SX, Bass Line performed in a very stable fashion and could provide that typical acid vibe produced by the hardware unit it is attempting to emulate but with the convenience (if not the cool!) that only software can supply. You can, of course, run multiple instances to overcome the monophonic limitation and, with just the addition of a suitable set of drum samples loaded into Halion, it was possible to produce a basic acid-style track using just Bass Line. Bass Line is available from the Audiorealism web site, and is about a 0.75MB file:///H|/SOS%2004-06/Plug-in%20Folder.htm (7 of 10)9/22/2005 7:43:21 PM

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download. While it might not have the tangible magic of the original hardware TB303, it certainly comes with a lower price tag and the convenience of a plug-in. Although there is a range of alternative ways of getting close to the TB303 sound via software, Bass Line is certainly worthy of a listen. A number of WAV and MP3 samples of Bass Line in action can be auditioned via the web, and these do give a good feel for what the VSTi can do. Those that prefer to 'try before they buy' will have to wait a little while, however, as at the time of writing, a demo version was still in development. John Walden 95 Euros. www.audiorealism.se

Sonic Charge µTonic Formats: PC VST Sonic Charge µTonic (also known as Microtonic, which I'll use from now on to save me reaching for the symbol menu each time) is an eight-channel drum and percussion synthesizer with its own built-in, pattern-based sequencer. The man behind its design is Magnus Lidström. Users of Propellerhead's Reason software studio will recognise the name, as Lidström also created its Malström synth. Microtonic is his first step into the world of VST Instruments. Perhaps the first thing to mention about Microtonic is that all the sounds created by it are 100 percent synthetic and rendered in real time. There are no samples used at all, which means that those looking for realistic, multisampled hits are going to be disappointed. However, if new, interesting and diverse drum and percussion sounds are your thing, read on, as this plug-in might well fit the bill. Aesthetically, Microtonic looks uncluttered and relatively unimposing with its array of large controls and fetching grey/blue colour scheme. Each drum part is accessed (and auditioned) by one of the eight buttons marked in Roman numerals at the top of the interface. Underneath these buttons, a good chunk of available screen space is given up to the nuts and bolts of the program: the actual sound engine. This is split into four sections: oscillator, noise, velocity and mixer. The oscillator section features three basic shapes (sine, triangle and sawtooth) with a slider controlling the chosen oscillator's frequency from 20Hz to 20kHz. Underneath this you have three types of pitch modulation (decaying, sine and random), with variable Amount, Rate and Decay controls. The noise section is equally well equipped, with a choice of three filters with file:///H|/SOS%2004-06/Plug-in%20Folder.htm (8 of 10)9/22/2005 7:43:21 PM

Plug-in Folder

adjustable frequency and Q adjustment, and three envelope shapes, with Attack and Decay controls. The noise generator also features a stereo mode that, when activated, uses two separate sources, one for the left channel and one for the right. This creates a dispersed stereo effect that can be used to create an odd reverb-like quality. Enough techie stuff — how does all this sound? In a word, great! Its user friendly interface (I like the mix of dials and sliders) invites knob-twiddling, and it's easy to get some really strange-sounding percussive noises just with some erratic mouse work. There is also a 'randomise all' option in the pull-down menu for those who enjoy pot-luck programming. Microtonic comes stuffed with over 60 example patterns that range from blip-hop stutters through to pounding hard techno, and a large batch of single hits. Each is well programmed and gives a good indication of the breadth of synthetic drum sounds that this plug-in can create. From deep subs to clicky, glitch-like kicks, sharp fizzes or cutting snares, Microtonic handles them all. Also worth mentioning is the range of one-off effects and odd noises that can be coaxed out, all good for adding some character to those standard 808/909 tones. The other major trick up Microtonic's sleeve is the built-in sequencer. This follows the familiar, pattern-based 'light the buttons' route used by other applications like FL Studio. I personally like using step sequencers to program beats, and Microtonic's is no exception; it's easy to understand and syncs into your host sequencer's tempo. The addition of a fill button (with an adjustable rate, from two to eight times per step) alongside the usual accent and swing controls makes putting a useable beat together quickly a doddle. Whole songs can then be built up in blocks by chaining patterns together in true drum-machine style, with pattern length and step rate all adjustable. It's quick and easy to use, although I'm slightly confused as to why Microtonic's on-board sequencer seems to stay paused until Play is activated in the host sequencer. This is a slight pain when working with stacks of other MIDI or audio tracks, but hardly the end of the world. Overall, I really like Microtonic. Its friendly interface, twinned with some real power under the bonnet, means you can get some truly interesting hits and beats out of it with very little effort — a real plus in my (slightly lazy) book. As I said at the beginning of this review, Microtonic is not going to conjure up the layered sound of a vintage Gretsch kit, but for pure, synthetic drum sounds I think Lidström might be on to a winner. Oli Bell $69 www.soniccharge.com Published in SOS June 2004


Plug-in Folder




Red Submarine Dual Xeon PC

In this article:


Under The Hood Published in SOS June 2004 Dual Processor Options Print article : Close window Powering Up Windows Performance Reviews : Computer Specifications Of Review PC Audio Performance Final Thoughts

Red Submarine Dual Xeon PC £2031 pros A huge performance improvement over the fastest available Pentium 4C model, particularly with Cubase SX. Very low acoustic noise considering the two processor fans and 520 Watt PSU. Rechargeable wireless keyboard and mouse.

Most of the major Windows sequencers now support dual processing, and if you want the ultimate performance, machines like Red Submarine's dual Xeon PC could be the way to go. Martin Walker

Despite huge leaps in processor clock speed and performance over the last few years, some musicians are still cons finding that they can't run all the plugSonar users won't see as ins and soft synths that they want, much benefit, particularly even with the fastest processors when running at low latencies. currently available such as AMD's summary Athlon XP3200+, Athlon 64 3400+ and If you want to run yet more Athlon FX51, or Intel's hugely plug-ins and soft synths expensive P4C 3.4GHz Extreme without networking multiple Edition. Adding a TC Powercore or Photos: Mike Cameron PCs, the dual Xeon 3.06GHz Universal Audio UAD1 DSP card can processors in this Red Submarine PC take its help, but only a small range of plug-ins is available in these two proprietary performance way beyond any formats. The other solution is to install a motherboard that supports two other machine I've measured processors to share the load, running under an operating system that also to date, yet despite this power supports multi-processing, such as Microsoft's XP Professional or Linux 2.4x. it still remains very quiet. information Basic system as reviewed without monitor, music hardware or software £2031.09; including 17-inch TFT monitor as reviewed £2388.59; including M Audio Delta 1010LT soundcard and Steinberg Cubase SX 2.0 £3135.48. Prices include VAT. Red Submarine +44 (0) 870 740 4787.

Red Submarine already have a wide range of systems, including machines running AMD Athlon XP and Athlon 64 processors, and have now added one featuring two Intel Xeon 3.06GHz processors. With 1GB of RAM and twin 80GB Seagate Barracuda 7200.7 hard drives, it's the most powerful system they have ever assembled.

Under The Hood

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+44 (0)870 740 4788. Click here to email www.sub.co.uk

The Lian-Li PC7 MIDI Tower case is a favourite with system builders, and it's not hard to see why. It's well engineered with attractive hard anodised aluminium panels, nearly every part bolting rather than being clipped together, and provides plenty of expansion potential with its four 5.25-inch drive bays, three 3.5-inch bays, and three more internal 3.5-inch bays. Red Submarine had also fitted the CD-RW and floppy drives with aluminium bezels to complete the all-metal effect. The inside of this PC proved to be a model of neatness, with all wiring formed into tidy looms. All exposed panels had been carefully lined with Acoustipack Deluxe acoustic material, a composite of thin and heavy damping sheet joined to acoustic foam, while all the unused 5.25inch and 3.5-inch drive bays had been filled with blocks of acoustic foam to minimise the chances of internal noise reaching the outside world. To complete the cooling arrangements, a 520 Watt SilenX power supply unit had also been fitted — these truly live up to their name, with incredibly quiet cooling fans. The beefier-than-usual PSU is needed for the Xeon processors, each of which can consume a hefty 100W. The Asus PC-DL motherboard runs Intel's 875P chip set The two Xeon with an FSB (front side buss) of 400 or 533 MHz, here processors can each running at 533MHz. Each of the Intel Xeon 3.06GHz draw 100 Watts of power, necessitating processors is cooled by a Zalman Ultra Quiet Xeon CPU the use of a hefty cooler fitted with a 60mm fan and Fan Mate speed 520W supply, but the controller. Dual Xeon machines traditionally use thoughtful cooling motherboards with Intel's E7505, which only supports the arrangements mean much slower dual DDR266 RAM, and are more that this computer is quiet enough for most expensive. Intel's 875P chip set officially supports only studio environments. the single-processor Pentium 4C series partnered with dual DDR400 RAM, but on the PC-DL Deluxe motherboard Asus have managed to couple the 875P chip set with dual Xeon processors, so they can use up to 4GB of the faster dual-channel DDR333 RAM, as well as providing SATA (Serial ATA) support. Red Sub had filled RAM slots one and three with 512MB sticks in dual-channel configuration. They had used DDR400 RAM, although the motherboard only supports speeds up to 333MHz, so the DDR400 sticks appeared in the BIOS readout as DDR333 Dual Channel Mode. Up to six drives are supported by the PC-DL Deluxe motherboard: the ICH5R South Bridge chip provides two UDMA100 connectors, plus two Serial ATA connectors, while an additional Promise PDC20378 controller (disabled by Red Sub) provides a further UDMA133 and two Serial ATA connectors. Each SATA pair can support drives in a RAID 0 (extra performance) or RAID 1 (extra security) configuration, although in this system the two 80GB drives were left separate and devoted to System and Audio duties respectively, showing up inside the BIOS as Primary Master and Primary Slave, while the LG CD-RW drive was Master on the Secondary IDE connection.



There are five 32-bit PCI slots and one AGP Pro slot; PCI slots one and five share an IRQ, as do slots slot two and three, while slot four shares with the onboard 1394 Firewire controller. Red Sub had wisely selected slot two for the M Audio 1010LT soundcard they supplied with the review system, since slots two and three, while sharing among themselves, are the only ones that don't seem to share with any other onboard device. The ATI Radeon 9200SE graphics card with 128MB RAM is a popular choice with DAW builders at the moment. It's not hard to see why, since it's got dualhead capability, doesn't have a noisy cooling fan, provides perfectly adequate graphic performance for a music PC, and is fairly cheap. However, Red Sub currently offer a choice of 12 graphics cards with this system, so there are alternatives. Around the back the motherboard provides the usual clutch of PS/2 mouse and keyboard ports, two serial and one parallel port, RJ45 LAN port, one IEEE 1394 Firewire port, and four USB 2.0 ports. A further two USB 2.0 ports had been wired to the Lian-Li's drop-down front-panel access. However, the motherboard supports eight ports in total, and since you can't enable/disable them individually in the BIOS, they all show up in Windows — perhaps Red Sub should fit a backplate with the missing two USB ports so you can take advantage of them. This particular review system was supplied with a 17-inch TFT monitor from Relisys with a maximum resolution of 1280 x 1024 pixels, although Red Sub had set it to 1024 x 768, which I felt resulted in a less sharp image. Resetting it to its native 1280 x 1024 sharpened this up noticeably, as well as providing significantly more screen real estate in Cubase, so unless you anticipate working a long way away from the screen I suggest you do the same. Completing the setup was an A4Tech wireless keyboard and mouse with 'A-type' ergonomic key layout. The latest models have a big advantage over their predecessors: the RF receiver also acts as a twin AA-cell recharger, and you get six AA cells in total, so two spare ones are always fully charged and available.

Dual Processor Options At first, the only real option for multi-processing was Intel, because there were no dual-CPU AMD motherboards. Then AMD introduced their 760MP chip set in mid-2001, partnering it with the Athlon MP processors, whose architecture is very closely based on AMD's Athlon XP 'Palomino' range, and this gave Intel some much-needed competition for running servers. AMD's subsequent introduction of the Opteron processor early in 2003 must have sounded alarm bells at Intel HQ. Again based on an enhanced Athlon XP (this time with the 'Thoroughbred' core), and with 1MB L2 cache and an integrated memory controller, a dual Opteron 244 system with each processor clocking at 1.8GHz proved to run many benchmarks nearly as fast as Intel's own dual Xeon 3.06GHz with 512MB L2 cache.



Only a few months elapsed before Intel fought back with a new Xeon, fitted with an additional 1MB L3 cache as well as the existing 512k L2 cache, but this made only a few percent difference to the majority of benchmark tests, and most experts ended up recommending that users buy the far cheaper Xeon model without the extra cache (as used in this Red Submarine review system). Moreover, the dual Xeon only runs with a 133MHz FSB (with Dual DDR266 RAM), compared with the 200MHz (and Dual DDR400 RAM) of Intel's latest Pentium 4C range, so in some memory-intensive tests a dual Xeon 3.06GHz lags behind PCs equipped with a single P4C 3.2GHz or Athlon XP3200+. Games benchmarks also consistently show that dual Xeon machines lag behind Opteron ones. So far I've painted a pretty damning picture of dual Xeon machines, but don't write them off just yet. With multimedia encoding benchmarks (which, after all, are probably the closest mainstream test to running music applications), the dual Xeon 3.06GHz processors and 875P chip set used in the Asus PC-DL Deluxe motherboard in the PC under review consistently perform much better, beating off P4C 3.2GHz and XP3200+ machines with ease (some published tests show a doubling or even trebling of performance). My own tests show that when running the most popular PC music applications, the dual Xeon 3.06GHz would outperform a P4C 3.2GHz by some 43 percent running Cubase SX 2.0 at 23ms latency, and 37 percent at 3ms latency, and by about 36 percent when running Sonar 3.1 with a high latency of 46.4ms, or about 19 percent at 3ms.

Powering Up It didn't take more than a couple of seconds after pressing the power switch to answer what is always the most important practical question about any PC built for musicians: is it going to be quiet enough? Thankfully, Red Sub's good reputation in this area remained intact, since the combined noise of the cooling fans was still a soft purr, and you ought to be able to record through mics in the same room as this PC, as long as they stay a couple of metres away. The motherboard provides a readout of three different temperatures, plus one for each of the CPUs. After several hours of general activity the motherboard ended up at about 35 degrees Centigrade, while both processors stabilised at around 50 degrees. This proves that the cooling system was working well, despite its low noise. I was particularly interested to see how the BIOS had been set up, since Xeon processors, being based on the Pentium 4 range, support Hyperthreading, so if this was enabled, Windows XP would find a total of four processors (two physical and two virtual). However, as I expected, Red Sub had disabled the HT technology in the BIOS, since it would work against the true dual-processing in this configuration, and apparently many plug-ins, soft synths and applications have problems when more than two processors are used. Also disabled were the onboard AC97 sound chip, MIDI and Game ports, and the Promise controller. However, if you fancied using the MIDI interface and/or two further USB 2.0 ports, it would only cost a few pounds to buy a Game/MIDI file:///H|/SOS%2004-06/Red%20Submarine%20Dual%20Xeon%20PC.htm (4 of 8)9/22/2005 7:43:26 PM


backplate and Gameport cable and plug it into the appropriate motherboard headers. I got on well with the keyboard and mouse, both of which worked reliably up to a couple of metres away from their base unit, although I did find it difficult to adapt to the way the cursor keys merged into the numeric keypad.

Windows Performance

Compared with a single Pentium 4C processor, the dual Xeon provides a huge leap in performance at both high and low latency values.

Although Linux 2.4x also supports multiple processors, the most obvious choice of operating system for musicians is still Windows XP Professional, which is what Red Sub had installed. All the usual XP tweaks had been implemented to disable cosmetic slowdown and prevent interruptions from untimely tasks, and a fixed 2GB page file had been set up. Both drives had been formatted as NTFS, and Dskbench showed that the Seagate Barracuda 7220.7 drives provided exactly the same speedy performance as they did when I tested them in INTA Audio's 3GHz P4 PC in SOS November 2003, namely a 56MB/second sustained read and write speed, and over 250 potential 44.1kHz/16-bit audio tracks using a 128k block buffer size. As expected given that the RAM was running at 333 rather than 400 MHz, memory bandwidth as measured by Sisoftware's Sandra 2004 was significantly lower than with Pentium 4C PCs, at 3044MB/second integer and 3041MB/second floating-point. However, the CPU Arithmetic benchmarks for the dual Xeon When running Sonar 3.1 the dual Xeon still 3.06GHz were a huge improvement shows considerable improvement over singleover my own single Pentium 4C processor machines, and seems to match 2.8GHz machine. Dhrystone ALU more expensive dual Opteron PCs at high measured 15839 MIPS compared with latencies, but loses this lead at lower latency values. 8496 MIPS for the P4C, Whetstone FPU measured 4320 MFLOPS (3548 for the P4C), and iSSE2 managed 8306 MFLOPS (6210). Probably more important still for a musician, Sandra's CPU Multimedia benchmarks measured 38256 it/s and 48204 it/s for Integer and Floating Point respectively, compared with 21573 it/s and 30782 it/s for the 2.8GHz P4C. If we accept that Multimedia floating-point performance is probably the single most important parameter for the best performance of modern music applications, this is a good result for the dual Xeon machine, providing 1.43 times more processing power than a single processor of similar clock speed — in other words you'd



need a 4.4GHz P4C to match it, if one existed. However, benchmarks never tell the whole story, so I was just as interested in real-world figures measured with music applications.

Specifications Of Review PC Case: Lian-Li PC7 Aluminium ATX MIDI Tower, fitted with aluminium CD and floppy doors, and 520 Watt SilenX PSU. Motherboard: Asus PC-DL Deluxe, with Intel 875P 'Canterwood' chip set running 533MHz system buss, with four DDR DIMM sockets supporting up to 4GB of PC2700/2100 DDR SDRAM. Processor: two Intel Socket 603 Xeon 3.06GHz 512kb L2 cache, 4 times 133MHz front side buss. CPU heatsink and fan: two Zalman Ultra Quiet Xeon with Fan Mate controller. Case cooling & silencing: Lian-Li 80 rear 80mm case fan, twin Zalman 80mm front case fans with Fan Mate controllers, case lined and all unused drive bays filled with Acoustipack Deluxe acoustic material kit. System RAM: 1GB of PC3200 (DDR400) CAS2.5 SDRAM, running as DDR333 Dual Channel. System drive: Seagate Barracuda ST380013AS, 80GB, 7200rpm, 8MB buffer, Serial ATA. Audio drive: Seagate Barracuda ST3120026AS, 80GB, 7200rpm, 8MB buffer, Serial ATA. Graphics card: ATI Radeon 9200SE dual head with 128MB RAM. Floppy drive: 1.4MB 3.5-inch. CD-RW Drive: LG GCE-8542B, E-IDE interface, 52x read, 52x write, 32x rewrite speed, 2MB buffer with buffer under-run protection. Monitor: Relisys TL775 silver/black TFT colour, with 17-inch diagonal, 1280 x 1024 maximum resolution, 16ms response time, DVI (Digital Video Interface) connector, with built-in multimedia speakers. Keyboard & mouse: A4Tech rechargeable wireless desktop Ergo A-shape keyboard & optical mouse. Installed operating system: Windows XP Professional Edition plus Service Pack 1. Installed soundcard: M Audio Delta 1010LT with version 5.10.00.0036 drivers.

Audio Performance Red Submarine had installed Steinberg's Cubase SX 2.0.1 and an M Audio 1010LT in the review system, but of course you can choose whichever combination of music hardware and software you want, or buy the PC without either. The 1010LT drivers had been set to a 6ms latency at 44.1kHz, which is normally the most suitable compromise on modern PCs between responsiveness and CPU overhead, but in line with Steinberg's own suggestions, I measured file:///H|/SOS%2004-06/Red%20Submarine%20Dual%20Xeon%20PC.htm (6 of 8)9/22/2005 7:43:26 PM


performance with both 23ms and 3ms latency when running the Fivetowers 2.0 test songs. You can see my results along with some comparative systems I've measured in the graph. By extrapolation, you'd need a Pentium 4C processor of a hypothetical 5.6GHz to match the results for Red Sub's dual Xeon 3.06GHz machine, or a 3.5GHz Centrino laptop. This is an excellent result. I also installed Cakewalk's Sonar Producer 3.0 and then the version 3.1 update. This adds a redesigned multi-threaded engine to maximise the benefits of multiple processors, which shares the load between the two processors far more effectively than before, a result easily proven by a quick look in the twin CPU Usage windows of the Windows Task Manager, making it another ideal candidate for testing the benefits of a dual Xeon machine. Even better, Scott Reams of California-based LiquidDAW (www.liquiddaw.com) has created a new benchmark test specifically for Sonar 3.1, which attempts to test the CPU, front side buss, This case provides four 5.25-inch and memory in isolation by running a wide drive bays and a further three 3.5inch bays, with five PCI slots variety of plug-ins and soft synths, but without offering extensive expansion disk access. This is achieved by using input potential. monitoring rather than playback of audio tracks, and the CPU overhead is far easier and more consistent to measure than any Cubase test, since Cakewalk thoughtfully provide a numeric rather than a bargraph readout. Cakewalk have endorsed this test, and what's more, lots of users of Sonar Producer 3.1 have already posted their results, so I can compare the performance of Red Sub's dual Xeon with various other PCs using both single and dual processors. Using ASIO drivers with a range of buffer sizes providing latencies of between 46.4ms right down to 1.5ms, I can reveal that with Cakewalk's Multiprocessing Engine enabled, the dual Xeon 3.06GHz CPU overhead was just 16 percent at 46ms and 18 percent at 23ms, making it neck and neck with a dual Opteron 248 running 2.2GHz processors. However, although Cubase SX overheads increase by between 10 and 25 percent as you lower latency from 23 to 3 ms, Sonar overheads can more than double over the same range, and dual Opteron systems pull ahead once you lower the latency to more realistic real-world values. At 12ms and 6ms, the dual Xeon measured 23 and 31 percent respectively, slightly faster than a dual Opteron 242 at 1.6GHz. Sonar 3.1 overheads rise alarmingly with nearly all processor types below 6ms latency, and this time the Opteron 242 pulled slightly ahead, at 44 percent to the Xeon's 46.

Final Thoughts



This dual Xeon PC is certainly a very capable performer, and Red Sub are to be congratulated on building their fastest yet still extremely quiet machine to date. Comparing its performance with other processor families isn't a trivial exercise, but Cubase SX 2.0 users should find this dual Xeon machine a huge improvement over even the current fastest 3.4GHz Pentium 4C, despite the lower memory bandwidth. Users of DIY dual Opteron systems have reported excellent figures when running Cubase SX, but some vary so wildly that I'm loath to publish them as a comparison. When it comes to Sonar 3.1 there are some more established comparative figures available. The dual Xeon still beats a 3.4GHz Pentium 4C by a comfortable margin, but the difference is not as great as with Cubase SX 2, particularly at lower latency values. At high latency this dual Xeon 3.06GHz PC seems to equal the performance of a dual Opteron 248 when running Sonar, and considering the latter processor is currently almost twice the price of the Xeon, this is an excellent result. Unfortunately, at lower latencies the dual Xeon/Sonar combination measures closer to an Opteron 242 system, which would cost about £200 less overall. Both Xeon and Opteron systems still seem to suffer from occasional incompatibility problems with music software and hardware, but buying from a specialist music retailer like Red Submarine means you get a guarantee that your dual-processor machine will work perfectly with whatever combination of soundcard and music software you purchase, and for many musicians this makes a dual Xeon the way to go. Millennium Music also have two dual Xeon PCs in their range, both using the same Asus PC-DL Deluxe motherboard, for the same reasons of performance and cost. Overall, as a Cubase SX user, I was most impressed with the performance of Red Sub's dual Xeon PC. Published in SOS June 2004




Redmatica EXS Manager

In this article:

Speed Dating Friends Reunited Your Place Or Mine? Samples & Instruments In EXS24 Know Your Limits RTFM

Redmatica EXS Manager 40 Euros/80 Euros pros Offers a painless way of doing what you probably ought to do, but haven't! Very powerful organisational tool, and much faster than using Project Manager. Helps EXS instruments load fast.

cons Still feels a bit 'young', although it certainly does the job. No progress meter. Could be more intuitive and interactive.

summary EXS Manager is an essential piece of 'back office' kit if you have a large EXS24 library. There's quite a learning curve, but it isn't going to get any easier than this! You'll soon save the purchase price in working time and disk space.

Redmatica EXS Manager OS X Sample & Instrument Manager For EXS24 Published in SOS June 2004 Print article : Close window

Reviews : Software

Multi-gigabyte sample libraries offer unprecedented realism — and lots of potential for confusion. Redmatica's neat OS X utility allows EXS24 users to tame their unruly sample collections. Roger Jackson

Software samplers such as EXS24 have all but replaced their hardware counterparts, thanks to the increasing power of computers and close integration with host sequencing software. Many of us now have huge sample libraries which can resemble a rarely-visited attic or garden shed — all sorts of things thrown in with the intention of tidying up later!

If you recognise this scenario, you probably have several instruments where some or all of the samples are missing or damaged, and very likely a few gigabytes of duplicated samples. This obviously isn't too serious in these days of cheap disk space, but Logic has to hunt through this sample jungle every time you want to hear a new instrument, and with some of the orchestral collections now available it can have up to information 250GB to tramp through. Similarly, if you take your project to someone else's Standard Edition (10,000 Mac to work on it, you could be in for a long wait while Logic tracks down the instruments) 40 Euros; Pro samples you have used in your track. Redmatica's EXS Manager is a utility for Edition (1,000,000 matching up your instruments with their samples so that Logic can load them instruments) 80 Euros; Standard to Pro upgrade 45 almost instantly. It can also help you organise and optimise your sample Euros. collection. www.redmatica.com

Test Spec

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EXS24 Manager v2.5.3. Apple G4 dual 1GHz with 768MB RAM running Mac OS 10.3.3. Tested with Emagic Logic Pro 6.4.

Speed Dating Now, those well-organised folks among you will be smugly comfortable in the knowledge that you have already used Logic's very own excellent Project Manager to compile a database of all the files you need, including matching up EXS instruments and their samples. Others may not have got around to letting Project Manager do its thing — which, with a large library, can mean several days. EXS Manager is a very efficiently coded utility which offers you the choice of just linking instruments with their samples or going further and tidying up samples in a variety of ways, to optimise disk space by eliminating repetition, and also weed out damaged or incomplete instruments, all in a two-click process. You can do all of this in Project Manager, but you have to do it one function at a time. And, what's more, function for function EXS Manager's algorithms leave Project Manager standing. Some speed test results are posted on their web site: Redmatica claim operation speeds 100 times faster than Project Manager, performing a full re-link of about half of the vast Vienna Symphonic library (260,000 samples in 5000 instruments) in under five minutes. My own more humble comparison on a much smaller selection gave nine seconds versus 12 minutes — a mere 80 times faster. Suffice to say that the main limiting factor is going to be the speed of your hard disk.

Friends Reunited There are two modes of operation: simplified and advanced. In simplified mode you only re-link your instruments and samples. EM asks for the folder where your samples are and the folder where your instruments are. The latter will be the Sampler Instruments folder in your Logic folder if you are doing a whole library. I chose the EXS factory sounds as a test, copying them first to another disk so that they would all need updating. Pressing the button marked 'Re-link' does what it says on the tin. Depending on the size of library and the speed of your computer and disks, it may take a few minutes, but my small test selection sorted itself in seconds. Going back to Logic and EXS24, I found new instruments loading gratifyingly fast — almost instantly. Advanced mode gives you the choice of reorganising your library as well as relinking. You select three directories (folders): the folder where your samples are to be found, the folder where your instruments are to be found (this may be the Sampler Instruments folder in your Logic folder), and the folder where you want your samples to be re-filed. The third is optional.

Your Place Or Mine?



Instruments stay as they are, while your samples can be moved or copied to a different location and stored in new folders as you desire. The possibilities are too numerous and esoteric to cover here, but you are sure to find a structure to suit you from the pop-up menus on the samples page. You can choose whether to delete or carefully catalogue unused ('orphan') and clone (duplicated) samples. There are two stages. First EM analyses your selection, which is done very quickly, whereupon you can have a look at the comprehensive report on Simplified mode provides a quick way of rethe status of instruments and samples, linking your EXS24 samples. which is clearly presented in tabbed pages. You can proceed immediately to process your selection, and your files will be linked and re-organised. As with Project Manager, it is wise to choose to delete emptied folders as you go (an automatic option), or you can end up with your disk looking mighty crowded. Apart from this, a lot of data gets shifted at a keystroke. A 'snapshot' option allows you to return to the original state if you change your mind. If your instrument can't find its samples at the first pass, you can use different criteria to match them up. These will probably be slower than the default of 'name and length', due to the number-crunching involved, and it would be reassuring to have some feedback from a progress meter. You have the choice either to conserve disk space and share samples as much as possible, or ensure that every instrument has a complete set of samples to itself, even if this involves duplication. I felt that this 'belt and braces' approach was not necessary with such a powerful search engine, but it's nice to have the choice. A batch process option allows you to set the creator code of your samples, so that when you double-click a sample, it will open in the editor you have chosen for that file type. You can 'spring clean' your disk of empty folders too.

Samples & Instruments In EXS24 Your EXS24 library is composed of instruments and samples. While the instruments are 'definition' files, the samples are the actual sounds, and are therefore much larger. Logic can only read instruments from the Sampler Instruments folder in the main Logic program folder. Some people keep their samples here too, and when EXS24 translates Akai discs for you, it puts the samples by default into the Akai Samples folder, although you can (and probably should) move your samples to elsewhere on your system. Each instrument contains a reference to its own samples, so if you move the samples you need to tell the instruments where to find them. It is good practice not to store data (songs, instruments, samples) alongside your applications, and many users now have their whole sample library on an external drive which can be carried around and harnessed to whatever computer they happen to be working on. If you choose to keep your library on an external disk, you need to have a way of updating the Sampler Instruments folder in the file:///H|/SOS%2004-06/Redmatica%20EXS%20Manager.htm (3 of 5)9/22/2005 7:43:33 PM


main Logic program folder. A good way to achieve this is to put all your instruments into a folder on your library disk and create an alias of that, which you then rename 'Sampler Instruments' and put into your main Logic program folder. Incidentally, if your samples happen to be in the original too, it doesn't matter, since Logic will only display the instrument files it finds in there.

Know Your Limits Core Audio in OS X currently has a limit of 8192 sample files open at a time, which also limits your use of EXS instruments. Redmatica have cleverly realised that as EXS can use regions contained in whole files even for multisamples, they can combine whole instruments sets into a single file up to 2GB in size, effectively doing away with this restriction. These new combined 'Samplemerge' files, which are given an extension of '.EX2', have to be made from AIFF or WAV files and not SDIIs. You can edit these files, but it is recommended that you perform edits on the originals and re-Samplemerge them, so I would keep them for instruments you have already fine-tuned. The EX2 format is not as yet supported by Emagic, so use it with care as its future may not be guaranteed.

RTFM Organisation and backing up are not what most of us would choose to do on a Saturday night, but it can save a lot of anguish later on, and having your instruments' links up-to-date can save hours of valuable session time. It's possible that a more interactive interface to the software could help us through this data-handling minefield — not all of us are used to this sort of IT management problem. Still, EXS Manager makes all this relatively painless while still offering a lot of flexibility. I found that printing out and reading the PDF manual is to be recommended — it certainly helped me get my head around the concepts involved. Redmatica have very helpfully included half a dozen 'how to' examples of typical operations at the back. You can download a demo version to experiment with up to 40 instruments. When you are ready to buy, there's the choice of a 40 Euro version to handle up to 10,000 instruments and an 80 Euro version for 1,000,000 instruments. Any PC owners still using Logic can download the older version 1.3 of the Manager for free. The performance of EXS Manager reminds me of the big boost that Gallery software's Samplesearch brings to Pro Tools users, and I suspect that Andrea Gozzi at Redmatica has a few more interesting ideas up his sleeve. I fully expect this application to mature into an essential part of the Logic samplist's armoury, possibly with a more interactive and graphic interface, but even now it is a powerful tool to have if you are interested in optimising your EXS24 use.



Published in SOS June 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



Roland VS2000CD

In this article:

Overview Teething Troubles? Getting Around The Front Panel Recording Resolution Options The Rhythm Track Tuning Up In Harmony Sizing Up The Competition

Roland VS2000CD Digital Multitracker Published in SOS June 2004 Print article : Close window

Reviews : Multitrack Recorder

This new recording workstation is the most affordable and portable of the VS series, but it still lets you Roland VS2000CD £1599 record, mix, and master all in the one box. pros

Compact and robust. Lots of features. The Harmony processor is a useful creative tool. USB 2 file transfer and fileconversion software.

Tom Flint Photos: Mark Ewing

cons

The VS2000CD is the most affordable product in Roland's current range of three No channel or sample delay. VS workstations, which otherwise includes the flagship VS2480CD and the midNo I/O expansion options. range VS2400CD. In terms of its I/O and mixer specifications, the VS2000CD is more basic than the VS2400CD, but, to complicate the product hierarchy slightly, Some may miss a 96kHz sample rate. it offers a few facilities not found on its forbears; namely a Harmony processor, a Rhythm Track steals two tuner, USB 2 file transfer, Mac/PC file-conversion software, and a stereo Rhythm playback tracks. Track. It also includes the RSS 3D spatial emulation and V-Link video interface, No disk defragmentation both of which were introduced by the VS2400CD, but were not present on the option. original VS2480. (See the VS2400CD review back in SOS May 2003 for details Effects, Harmony on these functions.) generation, RSS panning, and mastering processing all rely on the internal effects boards, so you're likely to feel cramped without a full complement of three.

summary A pretty good attempt by Roland to provide all the tools you might need to take a recording project from its inception through to its mastered conclusion, including facilities for archiving your work along the way.

information

Overview At the heart of the VS2000CD is a 40-channel mixer, comprising 10 input channels, 18 track playback channels, and 12 mono (or six stereo) effects return channels. Apart from the effects returns, all of the above offer four bands of EQ and the choice of either compression or expansion, and each of the analogue inputs has its own preamp and volume pot. Routing options include two auxiliary busses, plus eight direct paths that can be used to carry mono signals to more or less any output or internal buss of your choice. The mixer also provides the usual monitoring, mute, phase-reverse, and channel-link facilities you'd expect, although there is no channel sample delay control.

file:///H|/SOS%2004-06/Roland%20VS2000CD.htm (1 of 10)9/22/2005 7:43:39 PM

Roland VS2000CD

VS2000CD, £1599; VS8F2 optional effects board, £272; VS8F3 optional effects board, £299; VS20VGA optional card for connecting a monitor, mouse, and keyboard, £199. Prices include VAT. Roland UK +44 (0)1792 515020. +44 (0)1792 799644. www.roland.co.uk www.roland.co.jp

Test Spec Roland VS2000CD OS v1.0. 2.66MHz Pentium IV PC with 256MB RAM running Windows XP Home.

The machine's other raw components are a 40GB hard drive, a CD-RW drive, and a VS8F2 effects board. Expansion is possible by adding a further two VS8F2s, although the VS2000CD will also accept a VS8F3 board in the same slots. This alternative board is designed to support third-party effects. The VS2000CD is capable of recording up to eight tracks simultaneously, and suitably provides eight analogue input sources. There is also a stereo digital S/ PDIF input routed, by default, to tracks nine and ten, but unfortunately it can't be used to increase the number of tracks recorded simultaneously. Both balanced XLRs and quarter-inch jack connectors are provided for each of the analogue input channels. Four back-panel switches provide +48V phantom power to four sets of two XLR inputs, so inputs one and two are served by the first switch, three and four by the second switch, and so on. One further alternative input is provided on quarter-inch jack for the DI'ing of high-impedance sources, and is accordingly labelled Guitar/Bass. An adjacent switch toggles between the highimpedance input and analogue input number eight. A VGA monitor, ASCII keyboard, and mouse can be connected to the VS2000CD once an optional VS20VGA board is installed in the rear-panel slot, but sadly this board was not available at the time of the review. It looks set to cost an extra £199 in the UK, but for a realistic budget you also have to consider the cost of a monitor, keyboard, and mouse to go with it. Word-clock synchronisation has not been included, as was the case with the VS2400CD, but clock information from another device can be received via the digital S/PDIF connection. The VS2000CD provides a variety of outputs which include quarter-inch jacks for line-level master signals, phono connectors for the monitor outs, a co-axial digital out, and a pair of phono connectors labelled Aux, which are intended for sending signals to external effects units. There is also a single headphone output with its own volume pot, while another pot serves monitoring output levels. The remaining I/O comprises a footswitch input for remote triggering, a standard pair of MIDI sockets (In and Out/Thru), and the USB 2 connector. Unfortunately, the VS2000CD does not allow the installation of additional I/O boards (other than the aforementioned VGA option). Although simultaneous recording is limited to eight tracks, extra input options can provide alternative interfacing, such as ADAT I/O. It would have been nice to have the option of installing more outputs so that recorded and processed audio tracks could be ported out into another mixer, or into a computer's multi-channel interface. However, the USB port does allow files to be moved between the VS and any computer running the appropriate software. A CD-ROM supplied with the machine contains the relevant software for converting VS-format audio files into AIFF or WAV formats for editing or storage in a Mac/PC. Similarly, WAV and AIFF files residing in the computer can be converted to VS format for importing back into the multitracker. This can be done on a song-by-song basis or for individual tracks or phrases. The software is compatible with Windows ME, 2000, and XP, as well as with


Roland VS2000CD

Apple Mac OS 10.2 or later. Even if a computer is not available, it is still possible to save individual tracks out of the VS by burning them as WAV files to CD. Similarly, WAV files can be imported into the VS from CDs. Not only can the CD drive be used to back up project data and burn audio CDs in all the usual ways, but it's a player of both commercial and non-finalised CDs as well.

Teething Troubles? These days multitrackers rarely reach their first birthday without several software updates to fix minor bugs. The first bug I experienced with the VS2000CD was that, after recording the first eight tracks in one 24-bit project, I was unable to record to any further tracks. This problem did not occur on other 16- or 24-bit projects, and was eventually cured by resetting the mixer and utility settings. Quite whether this was a bug or not, I can't be sure, but even if it was to do with the machine setup, it only goes to show how a hard-to-find incorrect setting can cause problems. The second problem I found was a loud clicking in the headphone and monitor outputs when recording a final stereo master. Reducing the audio levels significantly had no effect on the glitches, which sounded like very regular and consistent level overloads. However, the clicking was not recorded with the master track, proving to be an irritation rather than a serious problem. And this problem also didn't appear to afflict other projects. I was quite disturbed at coming across two problems like this so quickly, and it made me wonder what else I might find wrong with the VS in the long term. However, I never managed to crash the machine, and I found that switching between different modes was swift and smooth. This went some way to reassuring me that the VS2000CD is in essence a stable bit of hardware, and that the glitches are unlikely to be symptomatic of any major design faults.

Getting Around The Front Panel Like all pieces of complicated hardware, the VS2000CD's screen is vitally important, and is used to display a great deal of information at any one time. Navigation through pages is done using the six function keys which correspond with tabs at the base of the screen. On some views there are more than six tab choices available, so a nearby Page button toggles through all the available tabs as necessary. Selecting elements on the screen is done by using a combination of the four cursor keys and the data wheel, together with the Enter/Yes and Exit/ No buttons, which tend to light up and flash when they require pressing. Some of the VS2000CD's more important functions, like the routing pages for example, have their own dedicated buttons, whereas most of the others perform different tasks when in different modes, which can be a little confusing at times. file:///H|/SOS%2004-06/Roland%20VS2000CD.htm (3 of 10)9/22/2005 7:43:39 PM

Roland VS2000CD

However, having multi-function buttons means that Roland have been able to keep the VS2000CD small. In fact, the machine is exactly the same width as the keyboard I am using to type these words, making it eminently well suited for occupancy in a bedroom studio, or for use as a mobile recording device. The VS2000CD also manages to find space for 17 faders, including two stereo track faders, and a master fader for the stereo buss. On the left-hand side of the unit is one of the machine's most user-friendly sections, called Ch Parameters, which includes six continuous rotary encoders alongside buttons labelled Dynamics and EQ. Pressing either button opens a screen showing dynamics and EQ parameters together with the currently selected mixer channel's routing details. From here there is easy access to more detailed screens specific to the dynamics and EQ functions. With the appropriate button lit, the top four CH Parameter knobs provide Threshold, Attack, Release and Level control in the Dynamics screen or gain controls for the four bands of EQ.

The internal CD burner allows WAV-file import and export, project backup, and audio CD authoring.

Navigation around the various screens is still achieved by using the cursor keys, and some values are still adjusted by the data wheel, all of which can be controlled with the right hand while the left deals with the CH Parameter knobs. For example, the scroll wheel can be used to sweep the selected EQ band up and down the frequency range while the CH Parameter knob tweaks the gain. The VS2000CD has a comprehensive transport section providing various ways of looping, punching, scrubbing, and marking audio, all of which are now pretty familiar to the VS range. On this particular machine everything I tested worked as it should and gave me no cause for concern. The digital editing options provided by the VS2000CD are quite comprehensive, which is not surprising given Roland's experience of sampler and multitracker design. Both Region and Phrase editing are available, the former working across multiple tracks according to user-specified time selections, and the latter allowing quick manipulation of sections of recorded tracks. The list of options in Phrase mode includes Copy, Move, Trim In, Trim Out, Delete, Split, New, Normalise Divide, Name, and Take Manager, whereas the Region mode offers Copy, Move, Insert, Cut, Erase, Comp/Exp, Import, Exchange, Arrange and Name. Whichever editing menu is on screen, a complete Virtual track matrix is always visible in the bottom right-hand side of the LCD, which has the effect of removing any divisions between the 'real' and 'virtual'. In fact, as far as editing is concerned, the machine functions as a 320 tracker! Editing audio on the small file:///H|/SOS%2004-06/Roland%20VS2000CD.htm (4 of 10)9/22/2005 7:43:39 PM

Roland VS2000CD

screen using the various hardware controls can be a little fiddly at times, but it's worth noting that installation of the VGA card should make it possible to use a mouse to quickly select items from drop-down menus, while other processes, like selecting areas of audio for editing, can be done by clicking and dragging instead of by keying in edit points. Although I couldn't try these facilities, I can imagine the usability of the machine improving significantly in some instances. Although the VS doesn't have motorised faders, it does have a pretty comprehensive set of automation features. Individual tracks are armed simply by changing them from Manual mode to Write, and recorded automation is played back when a track is set to Read. Options for controlling Level, Pan/Balance, EQ, Mute, Aux Send and Ins FX/Level can be clicked on and off as appropriate. Editing options are Copy, Move, Insert, Cut, Erase, Comp/Exp and Gradation, and editing can also be performed in an event list. One the whole, the Automation is easy to apply and works just as it should.

The VS2000CD's screen often carries a lot of information, such as on the channel settings screen shown here. The numbered function keys below the LCD select functions displayed as buttons at the bottom of the screen, and a Page button toggles between sets of these functions when there are more than six.

USB 2 is a great addition to the VS2000CD's arsenal. Hooking the VS up to a computer is easy enough, and a dedicated button provides a way of instantly initiating the USB mode. Once engaged, individual files from each track become viewable on the computer and can be copied from location to location within the computer, just like any other file. Using a PC running Windows XP, I managed to locate and convert a recorded track into a WAV file, and had it open and running in Steinberg's Wavelab in a matter of seconds. The file transfer implications of this feature are obvious, and I, for one, wish my own multitrack machine were similarly equipped. The VS2000CD has six effects busses, all of which have dedicated buttons labelled FX1 through to FX6. The first two busses are served by the pre-installed board, but FX3 to FX6 require the installation of the other effects expansion boards before they become active. As standard, the VS8F2 board offers 250 effects presets derived from 36 algorithms — the manual details exactly how the various effects and processors have been designed. It's worth finding out which of the stereo algorithms crossmodulate the left channel with the right, and which ones simply offer the same effect independently on left and right channels, because the routing scheme of the VS does allow the use of the left and right channels of an effect independently.


Roland VS2000CD

Inserting an effect into an input channel for recording, or into a track channel to affect pre-recorded material, is easy to set up. Sending to an effect from a channel is straightforward enough too. There are plenty of other possible routing paradigms: effects returns can be routed to tracks or output via one of the mixer's direct paths to achieve a particular monitor or headphone mix. And of course effects can be used on the stereo buss just as they can on channels.

Recording Resolution Options The VS2000CD has done away with the data compression options seen on older VS multitrackers, and now offers just the M16 and M24 recording options, relating to 16- and 24-bit resolutions respectively. Sampling frequency is fixed at 44.1kHz for both modes. While it no longer seems necessary to offer compression options, it's still a little surprising not to see a 96kHz mode, considering the market's obsession with impressive specifications. Nevertheless, in real terms this is not a serious omission, given that a great recording made at 44.1kHz will still sound like a great recording. I certainly thought the VS2000CD sounded good, and felt that it was comparable with other similarly specified machines operating at the same resolutions. Both M16 and M24 modes allow eight tracks of simultaneous recording, but the 24-bit mode does unfortunately reduce the number of playback tracks from 18 to just 12.

The Rhythm Track The VS2000CD has its own Rhythm Track, which is accessed by pressing the dedicated Rhythm Track button. This feature is permanently assigned and routed to track channels 17 and 18, so any audio recorded on those tracks is muted when the Rhythm Track is activated. What you get is a selection of drum and percussion sounds derived from a set of PCM waveforms. These are divided into nine separate kits, usefully named Standard 1, Standard 2, Heavy, House, Jazz, Reggae, Room, Hip-hop and 808. Individual kit elements are played via the 16 Track/Status buttons, which automatically adopt their new 'drum pad' role in the Rhythm Track mode. The buttons retain the same role from kit to kit so that their labelling remains true at all times. For example, button one is always a kick drum, and button three is always a snare. Each Rhythm Track part is constructed using pre-programmed patterns, which can be created from up to eight measures, and there is control of both time signature and swing. To get things started, the VS offers 295 preset patterns which can be edited and stored in any of the 999 user slots or 999 projectspecific pattern locations. An arrange facility provides a structure for linking patterns into a chain. The 49 presets are given explanatory names like Intro, Verse 1, Bridge, Chorus, and Verse 2, and these can be edited and stored in the 10 user or 10 project slots. Patterns are pieced together using the Rhythm Track's own sequencer, which can also be used to control the sounds of an


Roland VS2000CD

external drum machine or sound module. Even the sequencer's metronome can be output if the three internal options are not enough. Drum patterns can be created in either real-time or step-time recording modes, although a Micro Edit facility is also provided for note-by-note adjustment. In realtime mode, beats are simply played in from the Track/Status keys, while the steptime mode lets you create patterns more precisely by entering notes into a grid. Changes can be made to velocity and gate time from the grid editor, and you can also access extra information in the event list — you can even change which drum sound an event applies to. Unfortunately, there seems to be no way to adjust the pan positions of the individual drums.

Tuning Up The built-in tuner is simple and quick to use, and is accessed via its own dedicated front-panel button. Once pressed, the button brings up a display on the screen showing a scale ranging from -50 to +50 cents. A virtual needle moves back and forth according to pitch, and settles in the centre when correct tuning is achieved. The tuner accepts signals from any of the 10 input or eighteen track channels, and it can take its tuning reference from a non-concert pitch instrument. The addition of a tuner really makes sense, considering that guitars and basses can be used via the high-impedance input and treated to Roland's COSM amp and speaker simulations. This means that guitar or bass recording requires no outboard gear at all.

In Harmony The first thing to understand about the harmony feature is that it doesn't require the use of any audio tracks — it's actually an effect. In truth, the name is a little misleading, because the VS Harmony does not actually work out or create harmonies. Instead, harmonising notes have to be played in by hand and are recorded into a dedicated sequencer. After that, the Harmony algorithm cleverly analyses the source material, such as a lead vocal track, to determine the character of the harmony performance, and it applies that character to the sequenced notes. The harmony algorithm can be applied to any track, and each track is given its own sequencer to independently play its own harmony arrangement. General performance settings are provided for vibrato rate, depth, detune, and delay, and you can also adjust the portamento glide time, bender range, and overall tuning. Harmony is available in Note or Chord mode, but at least one optional effects board needs to be installed to create chords. Separate from the Chord and Note modes are the Poly and Solo options. The former, as its name suggests, allows you to play more than one note at once, and the review machine, with its single file:///H|/SOS%2004-06/Roland%20VS2000CD.htm (7 of 10)9/22/2005 7:43:39 PM

Roland VS2000CD

effects board, managed two at a time. What you sacrifice in Poly mode, though, is portamento, which is only available in Mono mode. Inputting notes can be done from either an external MIDI device, or from the Track/Status buttons. The buttons provide hands-on control of upwards and downwards octave shifting, portamento and legato, and note data entry. Nevertheless, without a bend wheel or touch-sensitivity, the Track/ Status buttons are not very expressive. By turning MIDI on, notes can be played and recorded into the Harmony sequencer from an external keyboard. The sequencer recognises any note velocity, gate time, and pitch bend information input from the keyboard, and I found that, even with my dodgy keyboard skills, getting a competent result was much easier that way. I suspect that the Harmony processor's sound is too artificial to be of much use in a sparse rock or soul vocal arrangement, but I can imagine it finding a home in pop production, where processed vocal sounds are common. It could also be a useful fix tool to have available, especially if, for example, the vocalist has gone home without recording backing parts. In such a scenario, the process offers a way to create backing parts and affect them with the missing vocalist's articulations. For me the Harmony feature really shone when it was applied to non-vocal tracks. Harmonising a busy keyboard mix, for example, immediately generated some new synth sounds which still had the essential timing and character of the original track. On drums, the effect was just as interesting, and seemed worthy of experimentation. Unfortunately the feature can't really fulfil its potential until further effects boards are added, and even with a full complement of effects horsepower it will still probably be necessary to mix down your harmony arrangements to virtual tracks in order to free up the processors for other jobs.

Sizing Up The Competition The VS2000CD is a nicely designed and well-specified machine which looks to have found its own spot in the market between competing products from other manufacturers. At this price it's just a little cheaper than the Korg D16XD in the UK, although this brings I/O expansion and a touchscreen to the table. Both the Yamaha's AW16G and the Zoom MRS1266 offer slightly fewer features than the VS2000CD, but they both also cost substantially less, and the Zoom even includes rhythm and bass sequencing. Tascam's recently launched 2488 digital Portastudio is also sure to tempt some buyers, but its 24-track capability perhaps file:///H|/SOS%2004-06/Roland%20VS2000CD.htm (8 of 10)9/22/2005 7:43:39 PM

Roland VS2000CD

makes it more likely to be compared to the substantially more powerful VS2400CD. Some potential buyers will hanker after Korg's touchscreen, while others will be tempted by Yamaha's O2R-derived mixing facilities, but Roland have some unique selling points of their own, in particular the Rhythm Track facilities and Harmony sequencer. Roland's effects and COSM modelling algorithms are as useable as ever, and are still an important asset for the VS. What's more, being able to hook the VS up to a computer via USB 2 is also a great selling point. But the VS2000CD does have a few shortcomings. Only being able to use the Rhythm Track at the expense of two ordinary tracks is a bit of a shame. It's also a shame that the Harmony feature requires the full set of effects boards before it can be exploited fully. Similarly, the RSS panning and Mastering Tool Kit processes also eat into the processing offered by these same effects boards, and so are subject to the same limitations. Another small gripe is that the screen seems at times a little small for the amount of information it needs to contain (the status of the pan knobs is really hard to see, for example), and in some instances I longed for the VGA board so I could hook up a large colour monitor. Nevertheless, familiarity with the machine lessened my frustration with the screen size. It's also a shame that recording at 24-bit resolution limits the number of playback tracks to just 12. Furthermore, track count is compromised over time by disk defragmentation, and Roland's only solution to this is to recommend that you reformat the drive regularly — I'd have liked to have seen the inclusion of a dedicated defragmentation utility. Although I didn't find the machine sluggish, I did experience a few random operational difficulties (see the 'Teething Troubles?' box), which seemed like classic first-version software eccentricities. In terms of hardware features, I'd like to see a second headphone output — the VS is well suited to mobile recording jobs, and in such situations a second headphone feed would be a useful thing to have. In its SOS review, the VS2400CD was criticised for its noise level, and this machine too makes a bit of a commotion. However, when recording I deliberately left my microphone near the machine to see what noise was picked up, and found that the kind of drone created by the VS2000CD was quite benign, and not really a significant problem. Despite these gripes, I'm confident that the VS2000CD will be a success for Roland. The 441-page manual gives some indication of the sheer number of features that are packed into the one box, yet despite its complexity the unit always feels like one machine and one design, rather than several bolted together. All in all, the VS2000CD is a pretty good effort by Roland to nail the allin-one hardware studio workstation concept.


Roland VS2000CD




SE Electronics Gemini

In this article:

Twin-tube Design The Technical Specs Studio Performance Double The Quality?

SE Electronics Gemini Dual-valve Capacitor Microphone Published in SOS June 2004 Print article : Close window

SE Electronics Gemini £799

Reviews : Microphone

pros Lively, warm sound. Very sensitive. Less constricted-sounding than many cardioid mics.

cons The weight of this mic means you need a good stand! No low-cut or pad switches.

This impressive new mic from SE Electronics uses two valves instead of the usual one. But does it actually make any difference? Paul White

summary The Gemini is a serious mic and is priced accordingly, but I was soon won over by its larger-than-life, upfront sound. If you need a tube mic that offers something a little different from the ordinary, check this one out.

The SE Electronics Gemini is the first of the company's redesigned largediaphragm mics I have tested since SE Electronics set up independent design and manufacturing facilities in Shanghai last year with a mission statement to move out of the high-volume market and to focus on more innovative designs. Given that there are so many 'me too' microphones of Chinese origin flooding the market these days, this would seem to be an astute move, and from what I've seen so far the new SE Electronics designs seem to be in a different league to information most of the other Chinese-built microphones I've looked at, both in terms of £799 including VAT. Sonic Distribution +44 (0) design and build quality. Although this approach has taken SE away from the 'cheaper than thou' sector of the market, their microphones are still very 1525 840400. affordable and enjoy tighter manufacturing tolerances and more extensive testing +44 (0)1582 843901. than is typical in this market sector. They also employ Japanese-made Click here to email Panasonic electronic components to reduce noise and improve reliability. www.sonicdistribution.com www.seelectronics.com

Twin-tube Design The Gemini represents the top of the current SE range and is nothing if not distinctive, both inside and out. This is a big microphone by any standards, and if you dug one up in your garden, you'd probably evacuate the street and call in the MOD to blow it up! It weighs well over 1kg (the limit of my kitchen scales!) and even the included shockmount weighs more than many microphones, at 480g. Dismantling the 80mm x 230mm microphone reveals the reason for both its size and its weight.

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At the heart of the Gemini is a 1.07-inch, centre-terminated cardioid capsule mounted inside a generously large, dual-layer mesh basket. This features a goldsputtered mylar diaphragm and is mounted on a shock-absorbing support. However, what makes this mic very different is that it features a pair of dualtriode tubes (ECC83s) in a transformerless design. The CAD VX2 mic is the only other dual-tube mic I know of, and it costs significantly more than the Gemini. When I visited the SE factory last year, the Gemini microphone was still in the design stage, and I understood that the transformerless balanced output would be based on bipolar transistors, but having looked inside this production model I can see no active devices at all, other than the two tubes, and there's no transformer. The twin tubes are mounted in porcelain bases, and the extremely substantial machined brass body sleeve has ventilation holes to allow the heat to escape. I thought Photo: Mike Cameron the output transistors might have been hidden inside the power-supply box, but I couldn't find any there either, so either they've hidden them where I can't find them or they've found some way to do without them altogether. One possibility is that one of the tubes has been configured as a dual cathode follower (or similar) to provide a relatively low output impedance without loading the preceding stage — that would certainly explain the need for two dual-triode tubes. At the bottom of the mic is a massively machined base housing the militarygrade, locking eight-pin connector for the cable feeding the power supply unit, and once the sleeve has been fitted it is held in place by an equally solid, machined end cap that screws onto the base. The outside of the protruding connector is threaded so that it can screw directly into the included shockmount, where an integral locking ring allows the mic to be secured pointing in any direction. The single circuit board is populated with good-quality capacitors and resistors, and the inside of the microphone follows a monocoque construction, making it rigid while allowing easy access to the components. There are no retaining clips for the tubes as there are, for example, in Rode's tube models, but as the mic made the trip from China to here without the tubes working loose, this appears not to be a problem. Cosmetically, the SE range has been redesigned with a satin nickel finish on the upper half and a tough textured grey finish on the main body. The mic model name is engraved into the brass body sleeve, and the new SE embossed logo denotes the front of the microphone.



The Technical Specs The Gemini has a 20Hz-20kHz frequency range, but with a little presence lift to add air and detail to the high end. Its sensitivity is a fairly typical 20mV/Pa, and the output impedance is an equally typical 200(omega). Its equivalent input noise level is 16dBA, which is on the good side for a tube mic, and though the mic has no pad switch fitted it can accommodate SPLs are high as 130dB (for 0.5 percent THD at 1000Hz). This isn't as high as some models I've reviewed, but is still more than adequate for most situations in which this type of microphone is likely to be used. It is certainly more than enough to deal with even the loudest vocalist. There's also no low-frequency roll-off switch on this mic, so if this facility is needed it would have to be provided by the mic preamp. As with all tube microphones, the Gemini comes with its own external power supply, and as is invariably the case, this is an unassuming steel box with a multi-pin input for the mic cable, a balanced XLR output, and a mains switch. Power comes in via the usual IEC socket, and there's a voltage selector that switches between 220V and 110V operation.

Studio Performance I have to admit that when I first tried the Gemini I didn't hear quite the result I was expecting. Instead of a thickened low end combined with slightly restrained highs — the typical tube-mic sound — the Gemini sounded full-on right across the audio spectrum. The high end seemed to go on forever and, interestingly, the mic appeared to be a few decibels more sensitive than the other mics I put it up against. The Because of the way the retaining screw is assembled on the shockmount bracket, the ability of this mic to capture high-end detail puts vocals right in your face and substantial weight of the mic tends to loosen it, causing the mic to droop. However, it is captures every nuance of stringed the work of only a few seconds to remove instruments, but it still doesn't sound at the retaining screw and reassemble it all harsh or over-assertive. backwards, thereby remedying the problem. Furthermore, if this level of transient detail is too honest for some applications, moving off axis a little way, or deliberately singing or playing over the mic rather than directly into it, gives a warmer, more forgiving tone reminiscent of those valve mics that deliberately deliver the expected 'tube' sound. Whereas some valve mics can sound constricted, albeit in an artistically pleasing way, the Gemini retains a very open character with a lot of air around the sound, and I soon grew to like this aspect of the mic's character very much. Because of this ability reproduce transients so effectively, the Gemini also makes a nice percussion mic, but it would be wrong to try to typecast it, because it's really a very good all-round performer. file:///H|/SOS%2004-06/SE%20Electronics%20Gemini.htm (3 of 4)9/22/2005 7:43:46 PM


In combination with its shockmount, the Gemini is reasonably resistant to lowfrequency vibrations, and the large basket makes it less susceptible to popping than most, though a pop shield should still be used for close-miked vocal work. The shockmount is reassuringly solid, though I found it worked better if I removed the swivel screw and then turned the fitting around the other way before reassembly — by doing this, the weight of the mic tended to tighten the restraining screw, whereas if left as shipped the weight of the mic tried to loosen the screw.

Double The Quality? SE Electronics' Gemini is pleasantly unusual, and while it may not be the obvious choice for smoothing away rough edges, it's fantastic for placing your vocals right at the front of a mix or for delivering an intimate sound. It has warmth and it has depth, but that's balanced by its airy high end, and I'm sure the transformerless output stage has a lot to do with this. Most people buy tube mics because they want something that flatters the sound and, though it doesn't flatter in quite the same way as most other tube mics, the Gemini certainly paints a big picture, adding gloss and sparkle as well as weight. For such a big mic, it is capable of extreme delicacy, but it also sounds great with powerful vocals and can double up for just about any other studio task. If you already have a 'traditional'-sounding tube mic, the Gemini makes a useful alternative, but if you don't yet have a tube mic, you should as always evaluate it using the type of voice you're going to be recording — like any mic, it will favour some voices more than others. This isn't a cheap cardioid microphone in the UK by any standards, but it does deliver a uniquely detailed and spacious sound and it's sheer size should impress the hell out of your clients! Published in SOS June 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



Tannoy Ellipse 10 IDP & TS212 IDP

In this article:

System Overview The Hardware Tannoy TS212 IDP Subwoofer Masters & Slaves Ellipse 10 IDP In Action Options & Pricing

Tannoy Ellipse 10 IDP £4324 pros Probably the best-sounding Tannoy system yet. Integral remote-control facilities. IDP Soft and Ellipse PC-IP alignment software provided as standard in the UK. Sophisticated room-tuning facilities available via Ellipse PC-IP. Ultra-wide bandwidth stretching to 50kHz. Analogue and digital inputs as standard.

Tannoy Ellipse 10 IDP & TS212 IDP Digital Loudspeaker System Published in SOS June 2004 Print article : Close window

Reviews : Monitors

Tannoy's latest-generation Ellipse technology has been combined with DSP processing from TC Electronic, creating a versatile and powerful highresolution monitoring system. Hugh Robjohns

A couple of years ago the Danish TC Group bought the British TGI Group which owns Tannoy, so technology is now being shared between these two companies. The first fruits of this collaboration are Tannoy's new Ellipse IDP monitors, an update of the allcons analogue units I reviewed back in SOS None. March 2003, but incorporating summary Interactive Digital Programming The sophisticated IDP system technology from TC Electronic, similar extracts the best possible to that first seen in the Dynaudio Airperformance and resolution Photos: Mark Ewing from Tannoy's dual concentric series speakers I reviewed in SOS drivers. A clever data network September 2002. The new product links all speakers in a stereo represents a step forward for both technologies, so although much of this or surround system for remote monitoring system will seem familiar if you have read the earlier reviews, there control and system setup. are also some distinct enhancements. information See 'Options & Pricing' box. Tannoy +44 (0)1236 420199. +44 (0)1236 428230. Click here to email www.tannoy.com

Test Spec

System Overview The Ellipse IDP 10 is an active monitor for nearfield to midfield use, incorporating DSP facilities for the crossover, general equalisation, room correction, and bass management. The drivers are magnetically shielded and there is an optional matching TS212 subwoofer. The speakers are supplied in master and slave

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Tannoy Ellipse 10 IDP OS v1.16. IDP Soft configuration software v2.00. Ellipse PC-IP advanced installation software v1.15.

versions, where the audio signals are connected to the master and passed on to one or more slaves via simple Cat-5 cables and TC Link sockets — this networking interface also carries control and configuration data. Although a complete system can be set up from the master's small LCD display, it's a lot easier to connect up a PC to the system and use the supplied IDP Soft or PC-IP software. The latter program provides more advanced features, including sophisticated alignment facilities, room-correction EQs, and channel delays. Multiple speakers can be set up in a variety of stereo and surround-sound configurations, with or without subwoofers. The master units accept a dualchannel analogue or digital input via three XLRs. The digital input expects AESEBU data, while the analogue inputs can be configured for full-scale levels between +9dBu and +27dBu. The dynamic range is claimed to exceed 113dB, and all the digital conversion, processing, and transmission operates at a 96kHz sample rate. Optionally, 192kHz sample rates can be supported, but by default the digital inputs accommodate standard sample rates between 32kHz and 96kHz. There is also an option to replace the two analogue input connectors with a dual AES-EBU input module, thereby allowing up to six digital audio channels to be connected to the master speaker for routing around a 5.1 surround system over the TC Link network.

The Hardware There are two Ellipse IDP monitors currently available: the review model was the Ellipse 10 IDP, with a 254mm bass driver for which the quoted frequency response is 33Hz-50kHz (±1.5dB) and the maximum continuous SPL at the mix position (in a nearfield configuration) is 122dB. This model is intended for rooms of 80-130 cubic metres in size, with a typical listening distance of 1.5-3m. The main dual-concentric driver represents the latest generation of this intriguing design, with an injection-moulded polypropylene cone and an improved highfrequency Tulip waveguide to couple the output of the low-compression HF driver. The main driver is supplemented by a one-inch Supertweeter dealing with extreme high frequencies, and this is mounted in a pod firmly attached to the top of the enclosure. The Ellipse 10 IDP is a true three-way active design, employing three amplifier modules, with crossovers performed in the digital domain at 1.4kHz and 21kHz. The more affordable Ellipse 8 IDP has a 200mm bass driver, and a 40Hz-50kHz frequency response (±1.5dB) with 118dB maximum SPL. Naturally, this is intended for smaller rooms, typically 50-100 cubic metres, with listening distances of 1.2-2m. This monitor is described as 'semi-active', because the bass and mid-range/high-frequency units are driven actively, while the Supertweeter is coupled passively, transitioning at 16kHz. A pair of amplifiers power this monitor and, as in the Ellipse 10 IDP, these are Class-D designs, providing the equivalent of 200W to each element of the dual-concentric driver. Both versions of the Ellipse IDP share the same style of elliptical reflex cabinet, file:///H|/SOS%2004-06/Tannoy%20Ellipse%2010%20IDP%20&%20TS212%20IDP.htm (2 of 6)9/22/2005 7:43:52 PM


with two ports on the front baffle. The unusual shape is claimed to minimise edge diffraction effects thanks to the continuously changing dimensions and angles of the cabinet edges from the drivers, and it also reduces internal standing waves and panel resonances. The smaller version of the monitor encloses a volume of 18 litres, while the larger one encloses 30 litres. The units measure 373 x 460 x 361mm and 423 x 540 x 301mm respectively, and weigh 14kg and 18kg. Both versions are finished in a grey suede paint effect, and a neoprene rubber base stops the monitor from rolling around — this can be removed for bracket mounting if required. The internal mains power supply is a switched-mode type, accepting 100-240V mains at 50Hz or 60Hz. The mains switch is adjacent to the IEC inlet socket, and the LCD control panel carries a simple yellow status LED when the unit is powered. (The slave units have a dummy LCD panel, but with a functional status LED.) The system is designed to go into standby mode or full power-down mode automatically after preset times.

Tannoy TS212 IDP Subwoofer The matching subwoofer is a seriously heavy box, weighing in at a monster 51kg. It has two 12-inch aluminium-cone drivers with 1500W Class-D amplifiers, and the quoted frequency range is 25-150Hz (±3dB) with a maximum SPL of 128dB. Measuring 520 x 471 x 497mm, the cabinet encloses a volume of 45 litres and is constructed from MDF, with removable black fabric sides. Again, the internal PSU can accept any mains voltage from 100V to 240V, at 50Hz or 60Hz, and the unit has a single RJ45 connector for the TC Link network.

Masters & Slaves In most applications, the speakers are supplied in master-slave pairs (the subwoofer is always a slave). The master version incorporates the input circuitry and control panel, while the slaves have neither, being fully dependent on the associated master. In the more elaborate surround configurations there are usually more masters than slaves: one of the masters is designated the system controller, while the others are used for their additional audio inputs. The only exception to this is when you're using a single master unit fitted with the optional digital interface card — in this case the single master distributes all six input signals. In addition to the analogue and digital audio inputs, each master speaker is also fitted with a BNC word-clock input and three RJ45 sockets (ie. standard Ethernet ports). The first two are intended as outputs for the slave speakers, while the third can be switched to function either as input or output. The slave speakers



have just two RJ45 sockets permanently configured as one input and one output. Chaining any number of speakers together is pretty intuitive, but the handbook includes hook-up diagrams for all the usual stereo and surround configurations. The master unit's small backlit LCD panel displays system information and the various configuration menus — you navigate using two rocker switches, the left one providing Exit and Enter functions and the right one scrolling and adjusting parameters.

Here you can see some of the advanced setup parameters available with the bundled Ellipse PC-IP software utility.

The dedicated Ellipse IDP Remote controller unit is also connected via TC Link, using any spare socket in the system. The unit provides a system volume control, as well as buttons to access three user-definable preset listening levels and four system presets. There are also mute/solo buttons for each speaker in a 5.1 configuration. Although this is all probably a little over the top for a simple stereo system, the controller becomes essential in a 5.1 system, and solves many of the practical problems experienced when monitoring in surround from a stereo mixing desk.

Ellipse 10 IDP In Action I auditioned the Ellipse 10 IDP in two separate situations: firstly in a 5.1 studio environment, carefully aligned by Tannoy; and secondly in my own listening room as a 2.0 and 2.1 system. Hooking up for stereo was fast and easy, with just two TC Link cables to be plugged in. Positioning and aiming the speakers was easy too, because you can only see the gold dome at the back of the Tulip waveguide when directly on the correct axis. I initially configured the setup from the controls on the side of the master speaker, and this involved allocating the appropriate role (left channel, right channel, or subwoofer) to each speaker based on its serial number. I then configured the bass-management mode and crossover frequency, and tweaked the tonality of each speaker to suit its positioning — there are presets for positions near walls, in corners, on the console, and so forth. Each speaker could also be calibrated for levels, with simple bass and treble curves as well. The subwoofer controls included parameters for polarity, phase, level, and low-pass filter frequency. I also ran the supplied IDP Soft software on my laptop to configure the 2.1 system, although I didn't need to change any parameters from my initial setup. The software is certainly a lot easier to navigate than the control panel, and although there are no extra parameters available it does display everything in a file:///H|/SOS%2004-06/Tannoy%20Ellipse%2010%20IDP%20&%20TS212%20IDP.htm (4 of 6)9/22/2005 7:43:52 PM


much clearer way. With the system set up I sat back to audition a variety of material (commercial CDs and DVD-As, as well as some of my own 24-bit/96kHz location recordings) and found the system performed extremely well. It compared very favourably with my reference passive three-way PMC IB1s (which cost roughly the same in the UK including the Bryston 4B amplifier) in The rear panel of the standard Ellipse 10 IDP terms of bass extension, clarity and carries two channels of analogue and digital stereo imaging. In fact, the resolution input. These audio signals are distributed and ability to convey subtle detail were around the system via the TC Link impressive, as the precise stereo networking connections. imaging was enhanced with a realistic impression of depth. The low-frequency extension of the Ellipse 10 IDP is pretty good on its own, but the extra weight and solidity brought about by the massive subwoofer was welcome. The 5.1 studio system was based on five Ellipse 10 IDPs and the TS212 IDP subwoofer. The particular room we used had a room mode problem which caused a hole at 100Hz, and although this could be compensated for to some extent by the system's EQ facilities, it was impossible to correct it completely — some additional bass trapping was needed. Nevertheless, when properly set up, the system delivered an excellent sound stage, with precise and stable imaging, and a consistent sound character from all quarters. Given the room's acoustic problems, the system was configured using the advanced PC-IP program, in conjunction with a measurement microphone and some spectral analysis software running on a separate laptop. The facilities presented through this optional package are mind-boggling — in addition to much more flexible bass management and crossover facilities for the subwoofer, there is also a powerful four-band equaliser which can be used for correcting room anomalies or for voicing the speakers. Overall I was extremely impressed with the Ellipse IDPs. The integrated control system works well and makes using and aligning a complex setup extremely quick and easy. The DSP signal processing clearly optimises the quality of the drivers and ensures tight tolerances are maintained, while also allowing some room anomalies to be compensated for with relative ease. Anyone considering an upgrade of their monitoring for serious surround-sound work should add the Ellipse IDP to their 'must audition' list.



Options & Pricing Ellipse 8 IDP, £3812.87 per pair: comprises one master at £2068 and one slave at £1744.87. Ellipse 10 IDP, £4324 per pair: comprises one master at £2326.50 and one slave at £1997.50. TS212 IDP subwoofer, £2203.12. Ellipse IDP Remote controller, £146.87. Ellipse 8 IDP 2.1 system, £6162.87. Ellipse 8 IDP 5.1 system, £11744.12. Ellipse 10 IDP 2.1 system, £6674. Ellipse 10 IDP 5.1 system, £12919.12.

All Ellipse IDP systems include Ellipse IDP Remote, and ship with IDP Soft and Ellipse PC-IP software utilities. Six-channel digital interfacing option can be specified for new 5.1 systems at no extra charge. Prices include VAT. Published in SOS June 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q. Can I 're-amp' a line-level signal?

Q. Can I 're-amp' a line-level signal? Published in SOS June 2004 Print article : Close window

Sound Advice

If I record the DI'd signal from a guitar so I can put it through a guitar amp later, what are the concerns, if any? I'm wondering whether the linelevel signal coming out of my mixer will be too high or otherwise inappropriate for the amp. SOS Forum Post Editor In Chief Paul White replies: This is a common technique and shouldn't present you with too many problems, provided that you watch your levels. The standard line out from the desk is likely to be rather high for a guitar amp input, so either use an attenuator to lower the signal level or keep the aux send master level control low and be very careful when you turn it up. Guitar amps sound different depending on how much level you feed in, so a good test is to directly compare the level you're getting from a guitar plugged directly into the amp and the level you're getting from your line feed. Adjust the line level feeding the amp until the sound is right and you're pretty much ready to go. The other issue that you need to be aware of is impedance. Guitar amps are designed to accept a high-impedance input, and plugging in a low-impedance, line-level signal will result in an increase in amp hiss. This will be more of problem if you like to turn your amp up loud. There are a number of possible solutions.

The Little Labs Multi Z PIP adjusts level and impedance and is a handy tool for re-amping.

The first is to buy a ready-made device to correct the signal's impedance, such as the Reamp (www.reamp. com), a small box which provides a balanced, line-level (+4dB) input, a high-impedance output and a level-trim knob. Similar devices are also available from the likes of Little Labs (www.littlelabs.com/www. audioagencyeurope.com) and Millennia Media (www.mil-media.com/www.hhb.co.uk). This kind of device is also handy if you want to use guitar effects pedals on line-level signals (as a send effect from your mixer, for example), though some pedals are more picky about impedance than others. Another possibility is to run the signal from the mixer through a guitar pedal or preamp that has a buffered bypass. Passing the signal through the bypassed pedal will correct the signal's impedance. Obviously, a pedal

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Q. Can I 're-amp' a line-level signal?

with a hard-wired or 'true' bypass circuit won't be any use for this. I have one final suggestion, which is that you could consider buying an amp modeller device with a line input, such as the Line 6 Pod XP Pro or Behringer V Amp, or even perhaps an amp-modelling plug-in. You may initially think that this solution would be unsatisfactory, but the latest generation of amp-modelling preamps and plug-ins, if carefully set up, can get pretty close to the sound of a real cab miked up. They're also very convenient if you're working in a small space with poor acoustics or unsympathetic neighbours! Published in SOS June 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 I use three different soundcards at the same time?

Q. Can I use three different soundcards at the same time? Published in SOS June 2004 Print article : Close window

Sound Advice

I use a large analogue Soundtracs desk with wonderful EQ, coupled to various bits of outboard gear. I want to output my audio from my PC into the desk for processing. Using all three of my soundcards simultaneously will give me 24 balanced outs. However, these soundcards are different models from different manufacturers. Is there a workaround or do I need three identical cards? David Fleming PC music specialist Martin Walker replies: The answer all depends on which MIDI + Audio application you want to run, and which type of soundcard drivers it uses. Although Steinberg's Cubase can run with multiple soundcards from different manufacturers, it can only do so on the PC when running its ASIO Multimedia or ASIO DirectX drivers, neither of which provide low latency. Only true ASIO drivers on either Mac or PC will give you the responsiveness of latencies lower than about 20ms, but unfortunately you can only choose a single ASIO driver from within Cubase, so you could only use one of the three cards at a time. If, on the other hand, you're using Cakewalk Sonar on the PC, you'll probably be taking advantage of its support for WDM drivers, which can be run in tandem across multiple soundcards of differing makes and models, as well as providing fairly low latency. In most cases you'll be able to run several different soundcards side by side without problems, although there are no guarantees, and some rare combinations may suffer from audio clicks and pops, or cause your computer to crash occasionally or even refuse to boot up at all. MOTU's 24I/O interface provides 24 simultaneous inputs and outputs from a single PCI card.

However, if your cards are different, and even if you do manage to run them all simultaneously from a suitable application, you will have to lock their timing together externally. Most 8-in/8-out cards offer S/ PDIF I/O, and sometimes word clock, and either of these can be used for sync. Designate one card as Master (therefore running from its Internal clock signal), wire its S/PDIF or word clock output to the S/PDIF or Word clock input of the second card and set the second to expect an external clock. Make the same connection between the second and third cards, with the third also relying on external sync. If you don't do this, the three cards will 'freewheel', and while they may start in perfect sync they will gradually file:///H|/SOS%2004-06/Q.%20Can%20I%20use%20three%20different%20soundcards%20at%20the%20same%20time.htm (1 of 2)9/22/2005 7:44:13 PM

Q. Can I use three different soundcards at the same time?

drift apart during the course of a song, giving rise to possible flanging between tracks running on the different cards, and eventually (on long songs) more obvious inter-track timing problems. Some manufacturers write soundcard drivers that support multiple cards of the same family, and which can also be internally synchronised to sample accuracy using proprietary sync cables. This is by far the easiest way to approach your problem, since with three identical soundcards of this type you simply end up with an assembly that acts as one huge soundcard with 24 ins and outs, but which appears to ASIO audio applications as one device that can therefore be used with Cubase. Personally, if I owned a large Soundtracs mixing desk I'd find out which of the three existing cards has drivers that support expansion, and then buy two more cards of the same type — this is the only real way to get professional results when transferring 24 simultaneous tracks from a computer. Alternatively, MOTU's 24I/O interface provides 24 ins and outs from a single PCI card. You could also combine a 24-channel digital audio card with external D-A converters. Published in SOS June 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. Do I really need some 'grot box' speakers?

Q. Do I really need some 'grot box' speakers? Published in SOS June 2004 Print article : Close window

Sound Advice

Having read your review of the Triple P Pyramid monitors [SOS March 2004], I can understand the benefit of having lower-quality, or, at least, 'limited-range' monitors alongside studio-quality, full-range monitors so that you can hear what your mixes will sound like on small domestic 'hi-fi' systems. What I don't understand is what advantage there is in buying these monitors for £250, when a cheap pair of hi-fi speakers can cost as little as £30. Surely that would give a similar sound for a cheaper price? SOS Forum Post Technical Editor Hugh Robjohns replies: I'm afraid it's not quite that simple. The Pyramid monitors, like the Auratone 5C and Yamaha NS10 monitors they take their lead from, aren't just 'limited-range' speakers. They have a very specific set of characteristics which are fundamental to their usefulness. For a start, they are all infinite baffle designs — meaning that the speaker cabinet is a sealed box — whereas most cheap hi-fi speakers use ported (or reflex) cabinets [see the Q&A on page 28 for more information on this distinction — Ed]. Porting is used to elevate the low-frequency response at the expense of mid-range clarity and transient response (often referred to as 'overhang'). So if you want to find a cheap substitute for these classic studio references, you have to look for a speaker that is a sealed-box design. Next, the frequency response has a distinctive inverted 'V' shape, peaking in the mid-range and falling away gently above and below. This presents the most critical mid-range region in the best light and reduces the distraction caused by other, less critical, parts of the spectrum. Most cheap domestic speakers have an overly bright high end (to make them sound exciting), with a slightly reduced mid-range (to make them sound more pleasant and larger than they really are), and a lumpy, resonant bass end. The other thing to mention is that many cheap hi-fi speakers might not be able to cope with the often continuous high levels required in a professional mixing situation.

The Triple P Pyramid monitors are designed to reveal problematic areas in the mix.

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Q. Do I really need some 'grot box' speakers?

Trying to recreate the way these kind of monitors sound using EQ simply won't work either — the issue is far more complex than that, and is as much about time-domain response as frequency response. One of the key attributes of speakers like the Auratones and NS10s is that they have remarkably tidy impulse responses. If you input a sound with a fast transient — a kick drum, say — the sound starts very quickly and stops almost as quickly. Many reflex speakers tend to resonate or 'ring' for a significant period after the input signal has stopped, and that inherent ringing tends to mask subtle low-level detail, as well as giving a false impression of level, dynamics and even spectral balance. These aspects are absolutely critical when judging the right balance between instruments, especially at the bass end. I agree that paying £250 for what are clearly very simple speakers that don't even sound particularly pleasant seems a foolish enterprise. However, these things are designed to serve as an accurate and reliable mixing tool, and a lot of engineers have found Auratones and NS10s indispensable in helping them to craft the mixes that keep them in employment. The Pyramids follow in the same genre and should serve as well in the future. That said, high-end monitoring has improved considerably in the last twenty years, so the role of monitors like the Auratones, NS10s and now the Pyramids is arguably not nearly as crucial now as it was then. Published in SOS June 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. Is analogue mixing superior to digital summing?

Q. Is analogue mixing superior to digital summing? Published in SOS June 2004 Print article : Close window

Sound Advice

I understand that mixes from DAWs can be improved significantly if, instead of using the digital mix buss within the computer, individual tracks are converted to analogue and then summed/mixed externally. Could you explain the difference between digital and analogue mixing, and whether this analogue approach really does offer significant benefits? Paul Cooper Technical Editor Hugh Robjohns: There is nothing necessarily wrong with digital summing (the digital equivalent of 'mixing' in analogue systems), and when performed correctly it is technically equal or superior to analogue mixing. However, many engineers and musicians enjoy the inherent imperfections associated with analogue signal processing, and that may be enough justification to warrant the use of analogue mixing systems in some circumstances. In an analogue console the channel signals are mixed together at the group or main stereo busses. This operation is simply performed by adding the instantaneous signal voltages together. The electronic circuit design has to be optimised to provide sufficient headroom to accommodate the signals from a large number of channels, as well as to maintain a sufficiently low noise floor — mix-buss noise is a perennial problem of analogue mixers.

SPL's brand new Mix Dream is designed as an external analogue mix buss for digital systems.

In a digital system the same thing is achieved by adding corresponding sample values together — hence 'digital summing' — but the same requirements remain, namely providing sufficient headroom with a suitably low noise floor. Careful attention must be paid to DSP software to achieve this. I think it is true to say that there were a number of cases in the early days of DAWs and digital consoles where flawed programming resulted in less than perfect results when mixing large numbers of tracks. The problems concern the proper handling of the large binary numbers that result when adding lots of signal samples together, and the necessary rounding and dithering required to output a fixed-length sample representing the mixed signal. The notion that analogue mixing is in some way better than digital summing really dates back to a few bad experiences caused by poor programming in those early days, combined with some dubious engineering file:///H|/SOS%2004-06/Q.%20Is%20analogue%20mixing%20superior%20to%20digital%20summing.htm (1 of 2)9/22/2005 7:44:23 PM

Q. Is analogue mixing superior to digital summing?

practices which were not appropriate to the digital domain. Unfortunately, once some 'name' engineer is reported in the press as preferring analogue mixing because it 'sounds better' (in a very particular set of circumstances), the urban myth is established and it is very hard to dislodge! At the moment, it appears to be fashionable in some quarters to mix in the analogue domain and several analogue manufacturers, spotting a golden opportunity, have produced dedicated outboard analogue mixing systems to cash in on this vogue. However, most of these systems can only handle eight source channels, which is not particularly challenging for either analogue or digital systems. The problems of poor mix-buss engineering (in either domain) only tend to materialise when mixing more than about 30 channels together. That being the case, even using an outboard analogue mix-buss unit, the chances are that you will have to premix a lot of tracks internally in the DAW before being able to output just eight channels to the analogue box, which really defeats the logic of the argument for using an external analogue mix buss! From personal experience I've not noticed any kind of mixing-buss issues with any modern, professional console, whether analogue or digital, except in the case of some of the budget analogue mixers which simply don't have the necessary supply rails to provide a lot of headroom. I have, in the past, noticed occasional problems with a couple of early DAWs, and these could be attributed to either poor gain structuring (user error, in other words), poor implementation of plug-in processing (third-party software problems) or, less commonly, to the core DSP of the system itself. However, there are now more than enough superb-sounding digital consoles and DAWs on the market to give the lie to the argument that analogue summing is inherently better. The most demanding area of the music industry as far as mixing is concerned is probably the film industry. Most big film dubbing theatres use huge digital consoles these days, usually with three engineers working together to mix literally hundreds of source tracks, often with several generations of pre-mixing — and all in the digital domain. The film industry has also been producing release material with a far greater dynamic range than most music-only formats for a very long time. So any inherent problems with digital mix busses would have been revealed a long time ago, and these engineers would be using analogue equipment! With properly engineered systems running carefully designed software and being operated sensibly, there shouldn't be any problems at all with digital summing. Sure, digital mixing lacks the 'rich sounding' harmonic distortion and benign transient clipping that you can achieve by slightly overdriving an analogue mixer, but if you want to deliberately distort the material in some musical way there are plenty of ways to do that in a more controlled, predictable and less expensive fashion! Published in SOS June 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 clicks spoiling my digital recordings?

Q. What are the clicks spoiling my digital recordings? Published in SOS June 2004 Print article : Close window

Sound Advice

I record guitar using a Line 6 Pod Pro going into a Roland VMC7200 mixing desk via S/PDIF. While the guitar is plugged in, every so often a little audio spike comes through the monitors from the Pod. It's not particularly loud but it is loud enough to ruin a perfectly good take. I have tried several different guitars so that is not the problem. Can you tell me what this noise could be and how to stop it? SOS Forum Post Technical Editor Hugh Robjohns replies: The clicks you're hearing are most likely the result of an asynchronous clock problem. The timing of the individual samples in a digital audio signal is governed by what is called the word clock. S/PDIF and AES-EBU signals both carry an embedded word clock, and the idea is that the device receiving the signal uses the embedded clock data to accurately synchronise itself to the source, and thereby receive and decode the digital signal correctly. However, if the receiving device is configured to use an entirely different clock source as its reference (perhaps its internal clock, or the clock from a different source) then the two devices will effectively be working independently of one another and will drift in and out of sync with each other. If you are lucky, most of the time the two clocks will be close enough that when the receiver goes looking for a sample from the source, it will find one. However, sooner or later the receiving device will look for a sample and not find one, and that's when you'll hear the click! The rate at which the clicks occur is an indication of the rate at which the two clocks are drifting relative to one another. I came across one professional installation with asynchronous clocks where the clicks only happened once every several days! What you need to do is make sure that the Pod and VMC7200 are reading off the same page, as it were — you need to make sure that both devices are governed by a single word clock. There are several possible solutions. The most appropriate one will depend on your precise setup, but any and all of these solutions will be effective.

The Line 6 Pod XT Pro has S/PDIF and AESEBU inputs and outputs for interfacing with other digital gear.

The simplest option is to configure the VMC7200 to take its clock reference from the S/PDIF output of the Pod: set the Pod to run on its internal clock, and in the VMC's Digital I/O menu, set the Digital Clock Source to 'Digital In' and select the appropriate sample rate. This will stop the clicks from the Pod, but if you are trying to mix the Pod's signal with other sources at the same time they will all start clicking instead, unless they themselves are synchronised to

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Q. What are the clicks spoiling my digital recordings?

the Pod's word clock! The downside to this solution is that it places the clock stability (and therefore jitter) of the entire system in the hands of the Pod's crystal clock circuit, which may not be a particularly wise idea! If you're using the VMC7200 as the clock master (or if it is being clocked from something other than the Pod) the next possibility is to take a word clock signal from the VMC (or the master clock source) and use this to clock the Pod. Simply switch the Pod to accept this new external clock, connect the digital output from the VMC to the Pod's digital input (either AES-EBU or S/PDIF) and away you go. This is probably the best longterm solution, but it may be impractical from the point of view of cables and connections. Finally, the last option would be to simply ditch the S/PDIF connection completely and run a good old analogue signal to the desk instead. There are no sync problems with analogue — it's the ultimate clock-free interface — and the quality of modern converters is such that there won't be any significant difference in quality anyway.



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Q. What should I use to play my backing tracks live?

Q. What should I use to play my backing tracks live? Published in SOS June 2004 Print article : Close window

Sound Advice

I need some way to play a stereo backing track when playing live with my band. There's a keyboard part on the left channel, which would go to the PA, and a click track on the right channel, fed to our drummer's headphones. I don't want to use a CD player, as they skip, but I'm wondering what other options there are. Something with a start/stop footswitch or remote control would be good. SOS Forum Post Editor In Chief Paul White: I too use backing tracks with my band, and I've just recently stopped using a CD player in favour of a Mac laptop, although standing the CD player on a foam cushion fixed the skipping. If CD is the most convenient format for you, and your only concern is the danger of skipping, you might consider making a foam mount for a portable CD player (good quality ones will feature extra buffering to prevent skipping) or, even better, buying a professional, rackmounted CD player. There are numerous models to choose from, and most of them will provide separate left and right outputs as well as a headphone out, often with its own level and balance controls, which could be handy for your drummer, and can be triggered by a remote control. You could also consider a portable or rackmounted Minidisc player. While MDs are compact, durable, less susceptable to skipping and more tolerant of knocks and vibrations, you'll lose a little sound quality to MD data compression. I've also found that many portable MD players are far less hard-wearing than the discs they play and, with all their small moving parts, ageing machines have a tendency to misbehave. Another option is to use a portable MP3 player — they're small, light and affordable, and skipping is practically unheard of. Better models will play WAV files as well as MP3s, and some will even play AIFFs. You'll need to use a signal splitter to send separate feeds to the PA and to your drummer, and some gaffer tape to ensure that the mini-jack doesn't get pulled out of the player by mistake. When I play live with backing tracks, I find using a laptop to be the most convenient option. I have all the backing parts arranged in my sequencer, where I can use the Solo function to select which ones will play. You can configure various controls, including Return-to-Zero and Play, to be triggered from a small MIDI keyboard or, indeed, a footswitch. For me, the other benefit of using a computer, and one I'm sure you're too professional to need, is that I can file:///H|/SOS%2004-06/Q.%20What%20should%20I%20use%20to%20play%20my%20backing%20tracks%20live.htm (1 of 2)9/22/2005 7:44:33 PM

Q. What should I use to play my backing tracks live?

create dummy MIDI tracks under my audio backing mixes and arrange them as coloured boxes with text in them to tell me where we are in the songs! I'm told Peter Gabriel does something very similar. If you use a USB audio interface with multiple outputs, you can play your backings in stereo as well as providing a click track, though my own motivation was to allow me to experiment with four-channel surround backing parts live.



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Q. What's the difference between ported and un-ported monitors?

Q. What's the difference between ported and un-ported monitors? Published in SOS June 2004 Print article : Close window

Sound Advice

I'd like to know more about the difference between monitor designs that feature ports and those that do not. How do they differ in terms of sound and performance? Also, why is the term 'infinite baffle' used to refer to un-ported designs? Surely 'sealed cabinet' or 'enclosed speaker' is a more accurate description. Martin Lambert Technical Editor Hugh Robjohns replies: Strictly speaking, an infinite baffle is exactly that: a baffle panel of infinite size in all directions, so that the front of the loudspeaker drives sound into our world, while the rear of the speaker is loaded by an equally infinite volume of air behind it in someone else's world, and none of the rearward sound can ever reach the front. This is obviously impossible to achieve in practice, and the simplest solution is to fold the edges of the baffle around to form an enclosed space. This kind of cabinet construction is often referred to as an infinite-baffle or IB design, but it is actually rather different at a technical level. One of the most significant aspects is that the volume of trapped air is obviously finite, and in smaller cabinets acts as an appreciable 'spring' against which the rear of the driver has to work. This affects the linearity of the system, as the driver has to work against different resistive loads when responding to input signals of each polarity, and at differing levels. So, yes, calling it a sealed cabinet would be more accurate and informative! Reflex, or ported, monitors like the Fostex The most common form of loudspeaker cabinet is the reflex or PM2 boost low frequencies at the expense of ported design, with one or more holes in the cabinet — either on the definition. front baffle or sometimes on the rear panel. The idea of the port is to make the cabinet resonate at a carefully chosen low frequency — not unlike the effect of blowing across the top of an empty bottle. The effect of the resonance is to bolster the low-frequency response, and thus generate a greater low-frequency output from a cabinet of a given size than is possible with an infinite-baffle — sorry, sealed cabinet — design.

There's no free lunch though. In an infinite-baffle cabinet the bass rolls off gently at 6dB-per-octave, which file:///H|/SOS%2004-06/Q.%20What%27s%20the%20differ...20between%20ported%20and%20un-ported%20monitors.htm (1 of 2)9/22/2005 7:44:45 PM

Q. What's the difference between ported and un-ported monitors?

means that even small cabinets can produce an audible output at surprisingly low frequencies. The reflex design has a 12dB-per-octave slope, which means that although the upper end of the low-frequency region is louder than it is in an infinite-baffle cabinet of equivalent size, the lower end of the spectrum falls away much more quickly. This can give rise to the 'one note' bass effect — where the monitor's resonant frequency is emphasised at the expense of all others — which is common in many reflex designs. Another problem is that the port resonance causes 'time smearing' — its inherent 'ringing' at a low frequency when triggered by a transient signal can cloud low-level detail and give a false impression of the transient dynamics and balance of instruments. There are several other less common variations on these cabinet themes. Some designs place a 'passive radiator' or diaphragm over the port, so that the cabinet is a compromise between a sealed enclosure and a vented design. The Mackie HR series monitors take this approach, for example. Another technique is the transmission line favoured by PMC. This involves an altogether more complex and expensive cabinet design, and requires drivers with unusual characteristics, but it offers some significant advantages. These include a large reduction in low-frequency distortion, a more consistent balance at different listening levels, and a vastly improved low-frequency extension over equivalent-sized reflex or infinite baffle cabinets. Published in SOS June 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

Brand Violet Distribution Music Producers Guild Published in SOS June 2004 Tips For Unsigned Acts Print article : Close window Tom Cullen People : Miscellaneous This Month's MPG Panel

recording assessments

Business End enables you to have your demo reviewed by a panel of producers, songwriters, musicians and managers. If you want your demo to be heard by them, please mark it 'Business End'. This month's industry panel is drawn from the MPG (Music Producer's Guild).

Brand Violet Barry Sage (BS): "I like this. It seems very '60s and trashy, it makes me think of cheap horror movies and that sort of thing. I think if I was going to compare them to anyone else I might suggest the Cramps or the B52s, something like that.

Track 1 2.2Mb

"I really like tracks one and two, I thought they were great but I'm not so sure about the third track. It really surprised me when they suddenly started doing really good, polished vocals, I'm not sure that that fits in with the impression that I got from the rest of it. The strings on track eleven sound quite over produced and cinematic, and I don't think that fits with the trailer-trash sort of feel that they've got going on. They sound great in an off-the-wall, quirky sort of way but I think they could be in danger of losing that feel by going too far towards the cinematic string sound. In general I think this is fantastic, the first two sound great." Sam Stubbings (SS): "I really liked it when they went into that chorus with the cinematic strings, I thought it lifted it and gave the whole thing an extra dimension. I think these are very well crafted songs. I especially like the chorus on the second track; I think that's my favourite passage on any of the songs. It's really unexpected and just seems to come out of nowhere. I think that it's quite an unusual thing to do but it works really well. They've got some really strong melodies but it at the same time it still feels quite punk. I think they could do really well, I could see them signing to Rough Trade or someone like that." Karen Murphy (KM): "I don't think they have enough direction in their songs. I get the impression that when they start writing a song they don't know how it's going to end, and that lets it down. It seems like all the songs were trying to go somewhere but never quite arrived, and that makes it feel like the songs never resolve themselves. "The other thing that bothers me about this is the incongruity between the two styles they seem to be going for. They come across as being quite an edgy and punky sort of band but at the same time they're going for this big, polished, cinematic sound, and I don't think the two go together very well. It seems a bit conflicting and forced." SS: "I think a good producer could sort out a lot of those problems. I think it could work if it was done a bit more more subtly and the different elements were blended together a little better. A good producer could make these into some really great records."

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Business End

Andy Rogers (AR): "This is really good, it's easily the best thing I've heard tonight. I like the idea of having very harsh, punk music with such cute female vocals; I think that really gets your attention. I've actually seen this band play live and they're very exciting to watch. They've got quite a distinctive image — the band all wear suits and then the singer wears some sort of rubber or latex dress. "I agree that it's a bit stop-start in places and it needs a bit more direction, but they've got some really strong ideas. I think it would be worthwhile for any record company to stick this lot in a studio for a week with a decent producer and see what they come out with. I think the music has definitely got potential and, like I say, they're interesting to look at, which is always important for a band." Nikolaj Bloch (NB): "I really like the first track, I can imagine them being very good live. I like the singer's voice, I think it would be good if she took more of a lead though. It seems like they've got a lot of good ideas and they want to take it further than the usual punk-pop band. I like the way they surprise you and do things that you wouldn't necessarily expect from this sort of act. If they had the chance to work with someone who could help them enhance the good bits and polish the overall production, then they could be a really great band."

Distribution Tips For Unsigned Acts These days there are an endless amount of web sites offering exposure for unsigned bands and artists. A lot of these are based around the MP3. com format and simply provide web hosting for tracks, but there are alternatives. Music retailer Fopp are offering unsigned bands the chance to get their CDs distributed throughout their chain of shops via their Unsigned Network scheme (www.fopp.co.uk/unsigned_network/intro. htm). This scheme not only gives bands the chance to get their selfreleased CDs into actual record shops but also provides them with a larger cut of the retail price than normal, as there are no third-party distributors involved. Fopp are keen to promote the Unsigned Network and have recently teamed up with the PRS Foundation (www.prsf.co.uk), XFM (www.xfm.co.uk) and CD manufacturers Clear Sound And Vision (www.clearsound-vision.co.uk) to create the Unsigned Awards. This will take the form of an annual competition and is expected to run for the next four years. Winners will be given the chance to record at a professional studio, have their recordings mastered, manufactured and then distributed throughout the UK in Fopp stores.

Tom Cullen AR: "I like this, it's interesting. I'm not sure I want to liken it to anything but I think you can really see the quirkiness of Royksopp's stuff in this, maybe a bit of a Daft Punk and an Air influence as well. It's good, it's very clever stuff, it's original enough to do well and grab people's attention. I can imagine these tracks being used on adverts or TV programmes.

Track 1 2.7Mb

"He sounds like he knows exactly what he wants to do — they're very well crafted songs. He's obviously got confidence in this and it's well-placed confidence because it's good. In his letter he says that his CD's available in Fopp record shops and mentions how he's been contacted about licensing his music through being featured on the Indie Connections web site. It seems like he's really getting himself out there and pushing his music — there's so much oppurtunity for this sort of promotion these days, and there's no reason not to take advantage. "This is definitely interesting enough to make you want to investigate it further but at the same time a couple of potential singles couldn't do any harm. I'm also impressed by the packaging of this demo. It's obviously been pressed professionally but it seems like he's put a lot of thought into the design of the sleeve. The presentation suits the style and quality of the music as well; things like that can really help a demo stand out." KM: "I really like the last song on this, I like the way the mood of it changes very gradually — it draws you in and


Business End

holds your attention. The production is interesting in the way that it keeps moving and evolving. With some of the other songs it seems like he's got the basis of the melodies and a basic idea of what he wants to do, but I'm not sure that there are enough ideas in some of these tracks to make them work. Maybe he could try combining some of the basic ideas from a couple of these tracks into one. There aren't really enough surprises in the early songs. But this last one is really good, it's got all these little touches that drag you into it and involve you. I think maybe this guy should listen to some classical music. I think a lot of dance music is based on very basic classical tunes where it's moving melodically from one note to the other. I think he could really learn a lot from listening to Bach or Mozart or anything like that." SS: "This is obviously someone who takes a lot of care over the production and that's really nice to hear. I like this, but at the same time I'm not sure where I'd play it; it's a bit too low-key for a club. Maybe I'd listen to some of the slower songs around the house or something. I'd really like to hear this guy do a really full-out disco dance track; I think he would do a great job with that sort of thing. I think he needs at least one track like that; something that DJs would play and that people would play at parties. I think we've all talked about how important it is to have songs that could work as singles tonight and this is no exception. He just needs that one track to lead you into the rest of the songs on the album. "I think the trouble with mid-paced electronic songs like these is where to put them. They're not really laid-back enough to work as chill-out tracks and they're not big enough to work in clubs. I think if he had these songs on an album with a few more in-your-face tracks, then it would be a fantastic album." BS: "This guy's obviously a good producer. You really can't fault the engineering on this. It's very well mixed and very clear. He's done some fairly creative and imaginative programming on these tracks. I think it would be really interesting to let him loose with some dance tracks just to see what he does with them. I think another thing he should maybe consider is doing production work for other bands and artists, because he's obviously very good at that side of it. I like these songs but I do think it would be interesting to hear him working with a songwriter; a situation where he's not doing all the writing and production himself might just give it the extra element it needs." NB: "I agree with Barry when he says that this guy needs to work with someone else, I think that collaborating with a good songwriter would make this so much more interesting. The production and programming on this is very good but I think the songs themselves could be better. This is something we keep coming back to, but I think that it's so important to have some sort of hook. This is so well produced, it's easily good enough to release, but the difference between this and someone like Daft Punk is that Daft Punk have the catchy chorus that draws you into the song and makes you remember it afterwards."

This Month's MPG Panel

Barry Sage is a freelance producer and engineer. He specialises in Latin music and is well established in Spain and South America for his work with pop acts La Oreja de Van Gogh and Melon Diesel. As an engineer he has worked with a wide variety of artists, including New Order and the Rolling Stones. Recently he has been involved in the creation of a sample library of Cuban percussion (Beats Working — In Cuba) for


Business End

Zero-G and Native Instruments. A full list of Barry's work can be seen on his web site at www.barrysage.co.uk. Sam Stubbings is the Senior Producer for the DVD division of Metropolis. He began his career five years ago at Abbey Road and has since worked with artists ranging from Paul McCartney to Muse. More recently he has produced both the first DVD single (Bjork's 'All Is Full Of Love') and the first commercial DVD-Audio disc (Holst's The Planets). He also has his own act, Redstar, who are currently recording an album and gigging in London. Nikolaj Bloch is a freelance engineer, writer and programmer with almost 20 years' experience in the music business. As the guitarist in the band Subcircus, he played all over the world until their split in 2000. Since then he has worked as a programmer and soloist on several major Hollywood films. He has also written for a range of artists varied enough to include American country singers and Jimmy Somerville. He enjoys spending time writing and collaborating in Nashville throughout the year. Andy Rogers is a producer of production music for BMG/ Zomba. After graduating with a degree in music from Kingston University he spent a year as an engineer in a Hammersmith studio. Following a six-month stint at Abbey Road he joined BMG as an office junior. A year later he was producing albums of production music for TV, film and radio. Having left BMG to join Zomba, he was reunited with his former colleagues two years later when the companies merged. Karen Murphy began her musical career by training to become an opera singer. She later worked as a professional rock, pop and jazz singer in Australia, Japan and the UK, performing with original bands, cover bands, and in commercials. Since moving to the UK she has been employed as a Post Production Co-ordinator at Abbey Road Studios and currently works as a Project Co-ordinator for film and TV specialists Videosonics. Many thanks to The Firebird Suite who hosted the session. Their web site is at www.thefirebirdsuite.com. The MPG's web site is at www.mpg.org.uk. Published in SOS June 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:


Avoiding The Traps Tiger Tales Practice Makes Perfect Getting The Best From Plug- Published in SOS June 2004 ins Print article : Close window Killing Sounds People : Industry/Music Biz Do It Yourself (Unless It's Mastering)

There's some amazing music being made in bedrooms these days. And bringing it to the wider public is the job of the Tigerbeat6 label, whose stars include label founder Kid 606 and Rjyan Kidwell, aka Cex. Mark Pytlik

As the founder and lead star of Tigerbeat6, one of bedroom electronic music's most prominent independent labels, Miguel Depedro has witnessed the genre's sudden boom and subsequent identity crisis at first hand. In that time, his notoriously bratty alter ego Kid 606 has dabbled in everything from abrasive noisecore (Down With The Scene) and prettified glitch (PS I Love You) to heavily DSP'ed landscapes (Soccer Girl) and pop bootlegs (Action Packed Mentallist Brings You The F**king Jams). Meanwhile, Depedro's friend, Miguel Depedro, aka Kid 606. labelmate and occasional right-hand man Rjyan Kidwell has kept his musical CV just as varied. As Cex (pronounced with a soft 'c'), Kidwell has leapt from Aphex Twin-inspired knob twiddling (Role Model) and jokey bedsit electronica (Oops I Did It Again) to geeky hip-hop (Tall, Dark & Handcuffed) and earnest acoustic claustrophobia (Being Ridden). The sheer variety of the pair's output is testament to what can be achieved with a basic recording setup, in Cex's case consisting of little more than a laptop and a few pieces of software.

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Avoiding The Traps Born in Venezuela, Miguel Depedro was raised in San Diego, California. By his teenage years, he'd developed a keen interest in recording and sound synthesis, and so he enrolled himself in studio courses at a local community college. "I had an awesome professor who played me stuff like Steve Reich and really influenced me and was a great guy," Depedro recalls. "Studio-wise, it was just an analogue quarter-inch eight-track, a Mackie board and eventually an ADAT, with some MIDI sequencing off to Mac Classics running Digital Performer. I had better stuff at my house, but the whole process of being in a studio and miking stuff and getting strange samples was really cool and influential. I quickly became an intern and would spend the night doing lots of crazy feedback and tape-loop experiments." Like so many others from his generation, by the time Depedro was 18, he'd seized upon the mouse and keyboard as his primary instruments. "It wasn't till I got a computer that things really started to come together for me musically and I could catalogue and collate ideas and sounds and basically 'design' music like a designer would create something, rather then Rjyan Kidwell in full-on performance mode. simply capture and record it," he says. "I learned tons about digital editing within the first couple of years of having a computer, which is what allows me to do things quickly now." With an impressive back catalogue and a new full-length album (Kill Sound Before Sound Kills You) to promote, it's obvious that Depedro isn't at a loss for material these days. He wasn't always this prolific, though, a failing he puts down to a fascination with equipment for its own sake. "I was a serious f**king gear whore as a kid," he admits. "I lived off of buying and trading gear for years, searching pawn shops, thrift stores, newspaper ads, you name it. And then I'd just play around for the fun of it and never finish much." Because he was more impatient and far less technically inclined than Depedro, Kidwell never fell into that trap, but he's seen others falter creatively because of gear lust, or — more accurately — because of the fear of failure that gear lust handily conceals. "There wasn't an electronic music scene in Baltimore, but pretty much everyone I knew who made music was stuck in this thing of like 'Aw, next week I'm getting this, and then I'm gonna make these tracks that'll blow your mind!' and the next week you'd see this person again and they'd be like 'Hey, do you wanna buy this thing I bought, cause I gotta get this other thing,' and they'd never make a f**king track," he laughs.



"So immediately, from the outset, once I got into software I said: I am not gonna be one of those techie gear nerd dudes. I write songs. I didn't get into electronic music because I wanted to innovate some new crazy s**t that no-one had ever heard before, but because I wanted to be the whole band myself. I didn't want to deal with some idiot with a bass guitar in his hand going 'Why am I playing this part like this?'" It's a problem, he contends, that has been exacerbated by the wild proliferation of cracked software floating around on the Internet. Kidwell remembers touring with Depedro for the first time: "We were running into guys all over the place who wanted to talk about crazy software, and the software talk was even worse than the gear. At least the gear dudes had to work a job and physically buy something and physically have room for it. A software addiction, though — you could just get that shit pirated and then you've gotta spend six weeks learning it and then you get something else and you gotta spend six weeks learning that. Those people are in even more of a weird limbo purgatory of never producing anything listenable."

Practice Makes Perfect In defiance, Kidwell recorded his first album (2000's Role Model, released on Tigerbeat6) with nothing but the barest essentials. It might not have sounded shiny or new, but it marked a crucial first step, and at least it was out there for people to hear, which was more than his friends could claim. "The first record was done on a Dell computer with the Dell speakers and the Dell sound input," he laughs. "It was all so ghetto. And you can hear it." Kidwell credits his gradual improvement over the years to simple experience. Practice, he says, is the only thing that can really hone an ear. "Do it a lot," he suggests. "In secret if you can, where you don't expect anyone to hear it. You sort of need to make a couple of records that no-one hears before you make anything that's not 'ahh!' a year or two later. You can learn to hear things yourself, but I don't know if there's any way to have them pointed out to you. I used to have people point out sound issues in my records and it was like they were talking Cantonese to me. I didn't have that in my ear's vocabulary. The only way to do it is to listen to a lot of music and to pay close attention if you can. Eventually, shit starts to fall into place. You learn that a little bit of reverb on a drum beat can go a long way. Or 'Ah, compression — that's what it does!' "I guess the other thing would be to get really nice monitors. If you're just starting, you're probably not going to hear such a big difference, but it'll give you room to grow. If you're slumming it like I was, you really won't learn that much about sound, because your speakers just won't talk to you. It sounds like such a nerdy thing, to have nice monitors when you're just starting out, but it gives you room to grow right off the bat. I didn't get real speakers until the third record! That was the first time I used speakers that didn't come with the computer."



Kidwell's ears and recording chops have improved exponentially in the four years since, but his desire for no-frills simplicity remains. After recently relocating from his native Baltimore, Maryland to Oakland, California, he liquidated most of his gear. Among the jettisoned items include an Akai S2000 sampler, a Clavia Nord Lead 2 synth, an Alesis Nanopiano module and a Seasound Soloist soundcard — his main interface for years. "I was trying Kid 606's studio is based around Apple Macs to make a point about not being running Logic with a Logic Control fader surface, MOTU 2408 interface and Yamaha attached to things," he jokes, before 01v digital desk. Synths include (left) a Korg admitting that he put the money MS20, Yamaha CS30 and Access Virus kb, towards a much simpler and more and (right) Sequential Pro-One, Clavia Nord portable studio. "I got a 17-inch G4 Lead 2 and Nord Modular. Powerbook and I bought an RME Hammerfall audio card. I've heard it's the shit, but I haven't gotten a chance to get it out of the box because I'm still on the road. I also bought an [Access Virus] Indigo 2, because it was the most powerful two-octave controller, and something I could fit into a backpack... my priority right now is to be able to travel as much as I want and write songs on the road. "I have a regular old Boss compression pedal and I used to run beats through that for real quick and dirty compression. I'd run one channel of the drums through that, run the other channel of the drums through an Ibanez pedal and it'd make a regular drum beat into something that was really exciting to listen to. Having two different compressions on two hard-panned channels of a drum beat is one of my favourite things to do in the game. "The only other thing besides my compressor pedals that I didn't sell was my Yamaha Portasound keyboard. I got it for Christmas when I was 10, and it's in no way a classic retro keyboard of any type. It's a 20-dollar Radio Shack keyboard. Battery-operated, no MIDI, but it does have a quarter-inch keyboard output, which is nice. A lot of the stuff on Being Ridden and Maryland Mansions is a Yamaha Portasound keyboard played live through a combination of guitar pedals. You could get a Korg Triton but the patches sound so nice that even if you're tweaking them out, someone'll still go, 'Aha, Korg Triton!' The thing about this Yamaha is that no one is ever gonna go, 'A-ha, Yamaha Portasound PSS140 run through an Ibanez bass fuzz with an MXR distortion pedal!' To me, that's part of making it transparent. I want you to hear the guy playing on the keyboard, not the keyboard." As he learns more about recording techniques, Kidwell has become increasingly willing to spend extra time fussing over his mixes, and his relentless pursuit of the perfect hip-hop beat has him crossing over into the territory of the contemplative knob-twiddler. "I'm getting more patient and more interested in



making a kick and a snare sound like the best kick and the best snare you've ever heard, and I never used to do that," he marvels. "I used to have no patience for that — 'It's a boom and it's a bap, let's go!' And as a result, you can hear all the issues on my records as far as EQ and distortion and all that bullshit goes. But I don't regret it. I'd definitely much rather be that impatient kid who wants to write a song than be the chin-stroking old man who wants to tweak the frequency on some hi-hat for eight days. If I didn't have ADD, it'd be easy to do that. But for me personally, it's not. And it's definitely not a noble thing — 'I would love to sit here and adjust this phaser, but I'm going to move on for the good of humanity.'"

Getting The Best From Plug-ins "There was a time in electronic music when you could pick out exactly what plugins were being used," says Rjyan Kidwell. "It was like, all right, here comes the Sonic Decimator again! That was the big one. In graphic design, it'd be like using a font that makes everybody who does graphic design groan. So go into a plug-in, and as the first thing you do, make the most wrong, awful noise you can with it and work backwards. "I use the [Arboretum] Hyperprism Hyperverb on everything — it's superversatile and I like it. I've used that [Steinberg] Quadrafuzz VST plug-in a lot for distortion too. And another big one would've been that whole Waves Native Power Pack; I made some really f**ked up things by doing wrong settings on the three-voice and six-voice octaver. You can make some awful noises with it, shit that doesn't even sound like a keyboard any more. I hear the Waves Enigma in a lot of people's work, and it's cool, but go into the octaver and you can make some really wild shit."

Killing Sounds Meanwhile, over at Depedro's home studio, nothing is simple. Over the years, Kid 606's setup has ballooned, and now centres around a hub of Mac laptops and desktops, an MOTU 2408 interface, a Yamaha 01v mixer, enough synths to fill a small pawn shop district and a bucketload of software, of which Emagic's Logic Audio 6 is his current favourite toy. "The majority of stuff [for new album Kill Sound Before Sound Kills You] was started on one of my laptops by sketching out ideas in Logic, then bringing it to my desktop where I run separate outs of the MOTU 2408 and run individual channels through external filters and delays. This new album is real sample and DSP-based — more so than my other upcoming releases, which have a lot more vocals and instruments and MIDI production. "I use tons of soft synths for little things but if I'm actually trying to get some serious synth or bass line stuff going, I don't think anything beats hardware or MIDI-CV'ing analogue synths, especially if you can then process them really cleanly inside the computer. Ever since I got Logic 6, I've been obsessed with creatively using plug-in automation to get interesting sounds. I've always wanted file:///H|/SOS%2004-06/Kid%20606,%20Cex%20&%20The%20Tigerbeat6%20Label.htm (5 of 8)9/22/2005 7:45:31 PM


to do that a lot: because I do so much knob-twiddling with my hardware, I want to get the similar sound and style from the computer. It was just such a pain in the arse to do before Logic 6 and pre-mapped VST Continuous Controller plug-ins." Kid 606's prodigious output schedule is that much more impressive when you Rjyan Kidwell relies on a minimalist setup consider how frequently he changes centred around an Apple Powerbook with an his working methodology. Some laptop RME interface. musicians base entire careers on a single major software package; Depedro is lucky to ever make two albums the same way. "The difference between Soccer Girl and Action Packed Mentallist is as drastic as can be," he explains. "Soccer Girl was me going through hours of analogue synth noodlings done in my mum's garage when I had no place to live and no computer. I clocked everything on a TR909 or 808 or 606 drum machine and then grabbed my favourite ones and edited them and added some DSP processing. Action Packed Mentallist was me being completely sick of geeking out on electronic music and processing and just wanting to make the Kid 606 equivalent of a DJ mix CD and just downloading a gazillion MP3s, converting everything into AIFFs, chopping up everything and just having a blast playing the stuff live and eventually just bouncing everything down in Logic and putting it out just so I could stop doing it." His working methods may change frequently, but Depedro tries not to rely on his gear or his software for ideas. "That's like going to a dead person for advice," he jokes. "If I don't have my own ideas stirring around in my head, I just don't make music, go do something else, save everyone from some noise, and generally make the world a better place by not contributing more music to an already overflooded sonic society! I don't even sit in front of studio monitors or put on headphones to work on something unless I have a specific idea of what I wanna do. Or else I'll just be noodling around or geeking out, which is great, but I haven't had any time to do that these days and only have time to work on the ideas I am 110 percent into finishing and getting out there. That being said, I still have a gazillion sounds from old [Ensoniq] EPS16+ or [Kurzweil] K200 disks and scribbled patch sheets for analogue synths that I occasionally go back to and one day want to truly exploit in new music. I'm really into combining new and old things, which is why I like having tons of time to finish stuff, and why making tracks lasts a long time for me. It almost ends up that I'm remixing myself by the time I sit down to finish something." Has anything grabbed Depedro's attention on the software front lately? "[Green Oak's soft synth] Crystal is real awesome, [NI] FM7 shreds, and the internal Logic stuff is good," he offers. "Absynth is super-awesome. When it comes to actual synth stuff though, I can't sing the praises of analogue enough. Soft synths are the bomb for sound design and more basic stuff, but most of the actual synth stuff you'll hear from me for the next couple years is analogue, or Virus or FM



synths. That said, I'm also really into the simple act of just sampling a note or sound and playing it up the keyboard, adding some glide and envelope." Not surprisingly, Kidwell is much less likely to indulge in the latest software algorithms, preferring instead to pick his spots. "The problem with a lot of soft synths like Generator and Muon is that they sound exactly like Miguel Depedro's Tigerbeat6 is one of the themselves," he complains. "I've gotten best established specialist labels in the field. a little more use out of the software samplers. I really like Battery. It's super-intuitive; you've got these banks where you can load samples in from your computer, and you've got a ton of different effects you can put on each individual track. It's primarily for percussion, so when you have a drum kit, you can tweak out every single piece on the kit and control it really well. To me, it was so much more convenient to go right into programming with that, rather than having to sit and map all the keys out with the Akai. "I like Reason, but I'm scared to get too much into it because it'd be something that'd be easy to rely on if I knew it really well. It's kinda like Rebirth on steroids. 606 is really into Reaktor. It's been a year and a half since I tried to mess with it, and I really didn't know what I was doing at all. It didn't seem intuitive, although it probably is — I mean, I'm a blockhead. Maybe if I looked at it now, I'd get it more. I guess I'm one of those dudes who'd rather have a bunch of little things that each do one specific thing instead of one program that's the be-all and end-all."

Do It Yourself (Unless It's Mastering) In keeping with the self-sufficient thread running through both of their careers, both Kid 606 and Cex have always mixed their own records by themselves at home. While Depedro acknowledges that it's not exactly an ideal setup, he's come up with his own workable routine. "I generally mix in Logic and only run stuff externally if I'm gonna process it," he says. "I'd love to mix everything in analogue, but I don't have room for a board capable of doing it correctly, and I don't see the point in doing external mixing on a digital board like an 01v. I'd want something like a [Soundcraft] Ghost or a fancy Allen & Heath, and then I'd need a ton of good audio interfaces to give everything its own channel. The main 'trick' to all my mixing is tons of automation, and if I'm really anal about things, I just take all the individual tracks into a two-track editor like Peak and just draw envelopes or process little bits here and there for it to be exact." The job of mastering, on the other hand, is always entrusted to a professional. "I was always wary of letting people master stuff because I didn't trust them and because I've had such miserable experiences with bad or lazy mastering, but I file:///H|/SOS%2004-06/Kid%20606,%20Cex%20&%20The%20Tigerbeat6%20Label.htm (7 of 8)9/22/2005 7:45:31 PM


truly feel that if you can find someone who you trust and can let do all your stuff you should totally hand it over to someone else and just oversee it. It's something you don't realise how badly you need until after you get it done." Meanwhile, Kidwell's aspirations for his next record are tellingly modest: "Whenever the next full-length comes out, I hope to write it in my room on the G4 and then take the entire setup into a studio where I can mix and master it there. But really, nobody on Tigerbeat goes into studios. Even Numbers, who are a rock band, record themselves. That's sort of been a Tigerbeat thing — it's been a real do-it-yourself label. But now that the Cex-man is pulling some sales figures, they've realised, 'Hey, we can push this pro if we want to; we can afford to make it sound like it could be on the radio. Instead of just telling people it could be on the radio...'" Published in SOS June 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



Leader

Leader Published in SOS June 2004 Print article : Close window

People : Industry/Music Biz

I spend a lot of my time at music trade shows, and one of the most frequently discussed subjects is the future of our industry. This is understandable enough in any business, because the participants have to know which way the wind is blowing in order to survive, but it's doubly interesting to members of the music business community because most of them are also musicians or studio owners in their spare time, and they're as curious as anyone else as to what the next big thing will be. We've seen the meteoric rise of software synths over the past few years, and while there are some great new models on the market, the concept is no longer new, though there are still those who refuse to take them as seriously as hardware synths. For live performance, the hardware synth still has a lot going for it, but in the studio, a soft synth that never wears out and that comes with three or more Gigabytes of wave ROM seems awfully attractive when compared with a hardware unit that offers maybe 256MB of ROM. Even hardware studios are getting smaller, not to mention more affordable, with powerful all-in-one DAWs being built by just about every major manufacturer, and one of my previous predictions for the future was that users would finally start spending a little money on nice-looking studio furniture and effective acoustic treatment. After all, if you can no longer impress your friends with an old-fashioned analogue mixer that you could land a Sea Harrier on, you might feel inclined to make your setup look classy in some other way. And acoustic treatment makes perfect sense in any system, because if you can't hear what's really going on in your mixes, the end result probably isn't going to sound good when played back on other systems. It all seemed so logical to me that I was quite surprised to learn from our recent reader survey that almost nobody said they were going to commit much in the way of budget to either studio furniture or acoustic treatment! OK, furniture could be considered a bit of a vanity issue (though a good studio chair is essential given the amount of time you spend in it), but ignoring acoustic treatment seems very unwise, and we continually cite its importance. As our Studio SOS visits continue to confirm, most home studio monitoring environments are far from accurate, yet just a small amount of acoustic treatment will bring about a significant improvement, and if you're prepared to spend a little more still, most domestic rooms can be turned into pretty good mixing rooms providing you choose your monitors wisely. What's more, today's acoustic treatment kits are visually attractive and they also suggest to your clients and friends that your recording setup is a studio, and not just some recording gear piled up in a bedroom. And before anyone else mentions it, you can buy some cheap duvets to hang on your walls — but while they can be useful in some situations, they don't look as good as proper acoustic foam or panel absorbers, and they are only effective for mid-range and high frequencies. I still maintain they make great vocal screens though! Ultimately, I guess we all have different priorities, which in my own case means buying better mics and

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Leader

preamps, better monitors and sorting out the room acoustics. In my opinion, now that studio equipment is getting cheaper and smaller, it makes sense to spend some of the money saved on making our studios nicer and, more importantly, better-sounding places to work in! Paul White Editor In Chief Published in SOS June 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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Loneliness and the Long-distance Programmer

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About The Author

Loneliness and the Long-distance Programmer Sounding Off Published in SOS June 2004 Print article : Close window

People : Sounding Off

Paul Wiffen reflects on the evolving role of the programmer and the lonely life of the modern studio musician... Paul Wiffen

About The Author Paul Wiffen made his name as a synth programmer for Paul McCartney and Stevie Wonder, but after adventures in digital audio, orchestral composition and digital filmmaking, now likes to think of himself as a renaissance man.

At the first proper studio session I worked on 20 years ago, there were six different people in the studio: the producer, the engineer, the keyboard player, the guitarist, the programmer (me) and the tape op (who set up the tape machine and made the tea, though not necessarily in that order). In the old days, these were clearly defined roles and, as a programmer, my future career path was all mapped out. All I had to do was set up the synth sounds, then the keyboard player would take over. If things went well and I kept up the piano practice, then one day I might get to do the playing as well. But at that first session, there were already two traditional members of the band missing! The keyboard player was covering bass with his left hand and a drum machine was used instead of a real drummer. The frightening thing was that, as the producer had brought me along as his programmer, I was expected to program said drum machine. Not having used one before, I ended up making it up as I went along and, with a bit of input from the producer, we got there in the end. With the advent of sequencers, I found myself responsible for tidying up performances as well as getting the sounds. As producers obsessed with getting as tight a sound as possible insisted on more and more severe quantisation, I would be recording a live performance into a sequencer and, by the time it had file:///H|/SOS%2004-06/Loneliness%20and%20the%20Long-distance%20Programmer.htm (1 of 3)9/22/2005 7:45:38 PM


been edited and quantised, there would be hardly anything of the performance left. I remember Peter Vitesse (whose playing I had admired for years on records by Jethro Tull, Go West and Simple Minds) walking off a session because he felt his skills weren't being used. After hours of me taking all the life out of his great playing at the producer's request, he asked if he could do a solo on the track. The producer agreed and he did an amazing solo, but after Peter had gone back to the hotel, the producer had me program a much simpler part. This was the last straw for Pete! By the end of the '80s, with companies like Roland, Ensoniq and Korg adding effects into synths and workstations, I was suddenly responsible for programming reverbs, chorus and other DSP effects. As workstations developed, this expanded to cover EQ, compressors and the rest of the signal path, all previously the responsibility of the engineer. Then people started hiring programmers to do remixes with DJs, who often had little studio experience. I remember one remix where the DJ in charge of the session discovered the SMPTE timecode on track 24 and insisted on putting bursts of it in the middle of the percussion breakdown on the 12-inch version! As hard disk recording became the norm, I found myself looking after the entire recording process on the computer, not just sequencing all the MIDI gear. It wasn't long before people were asking me which microphone they should be using to record things. From that moment on, at every session I worked on with a real engineer, I started watching him like a hawk to see which mics he used and where he placed them. By the end of the '90s, I realised that I was doing all of the jobs which had been divided between six people on that first session at the beginning of the '80s, including making the tea! The main reason for this is probably that all the technology had disappeared inside the computer, but where along the line did I suddenly become qualified to program drum parts, let alone mic and compress things? For that matter, how come the engineer on that first session now plays and programs all the synths on the sessions he works on? And so does the tape op, who had a number one record on which he did everything but sing? At least this is evidence of an exchange of skills: I end up showing the engineer how to set up a Mac for music, and he explains to me how a compressor works, while the tape op listens and picks both our brains. But surely something has been lost along the way. Even if you now know how all this gear works, what you really need is someone to turn to and say, "What do you think?". Nine times out of 10, there is no one there to do that with. I've found myself really enjoying some of the film sessions which I have been working on recently because the traditional job roles are still more or less adhered to, so the old sense of teamwork is back. For the first time in years I have experienced the joy of collaboration on a piece of music. As a result, wherever possible, I try to do the same on all the music projects I am involved with now. In the last year, I have taken several projects back into a real studio to record a real drummer or into a concert hall to record real strings. Suddenly, I am starting to remember why I got file:///H|/SOS%2004-06/Loneliness%20and%20the%20Long-distance%20Programmer.htm (2 of 3)9/22/2005 7:45:38 PM


into this industry in the first place: to make music with other people, not all by myself! Published in SOS June 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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Recording Franz Ferdinand

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Natural Light, Natural Sound Tore Johansson Into The Mothership Published in SOS June 2004 Luxury Spill Farming The Mix Print article : Close window Out Of The Control Room People : Artists/Engineers/Producers/Programmers Hard Vocals More Thrash Praise Indeed

Cardigans producer Tore Johansson was thrown into unfamiliar musical territory when asked to produce the debut album by Scottish guitar band Franz Ferdinand, but the result was a commercial and artistic triumph. Tom Doyle

With UK sales of 300,000 and rising, Franz Ferdinand's debut album is the surprise indie-to-mainstream hit of the year, thanks in no small part to their angular guitar-funk single 'Take Me Out' making the top 10 back in February. But when Malmo-based producer Tore Johansson was first approached to work with the Scottish quartet, he might have easily been forgiven for thinking their record company had the wrong man. He admits to having only a sketchy knowledge of the scratchy early-'80s guitar bands like Orange Juice and Josef K that the group list as their main influences. "I didn't get into the post-punk stuff at the time," he says. "But I thought that was what was fresh about Franz Ferdinand, the fact that they made reference to these early '80s records. Normally I reach for '60s and '70s records when I'm in the studio, so it was a change to bring out those old synthesizers instead of the Hammond organ." In other ways, of course, Johansson was the perfect man for the job, having guided the Cardigans through five albums, from the jangling guitars of 1996's First Band On The Moon, through the pristine electronic pop of Gran Turismo in 2002 to the polished country-rock of last year's Long Gone Before Daylight. Most file:///H|/SOS%2004-06/Recording%20Franz%20Ferdinand.htm (1 of 9)9/22/2005 7:45:41 PM


of these albums were recorded at his now-defunct Tambourine studio in Malmo, and Johansson admits that having to negotiate the group's stylistic diversions down the years really helped to expand his repertoire as a producer. "When we started out, I was very inexperienced and we kind of grew up together. Every time we started on a new album, we sat down and listened to the old album and we decided what we liked about it and what we didn't like about it and what direction the new album should go in. So my first five years actually working as a producer, that was very much developing and growing up together with the Cardigans."

Natural Light, Natural Sound Johansson then relocated to England for a year, writing and producing pop tracks on a Pro Tools setup from his home in Sussex. But despite moderate successes with Mel C and Sophie-Ellis Bextor, he says he didn't much enjoy the experience. "I was actually fed up of working in studios with bands. But then when I came to England and started to write, I hadn't realised that the competition was so tough. I had no idea how many people were sitting out there with Pro Tools doing fantastic demos of fantastic songs with fantastic singers." On his return to Malmo, Johansson began working in a new sister facility of Tambourine, Gula Studion, which had been recently built in an old printing factory. Gula — Swedish for 'yellow' — is bright and airy with plenty of natural light. More than anything else, the producer credits this working environment for helping him to regain his enthusiasm for studio recording. "When you go into studios for whole albums, 30 or 40 days," he points out, "natural light is so important. Natural light, oxygen, table tennis... all the important stuff." The producer was first sent the Franz Ferdinand demos in the spring of 2003 and says he was immediately impressed by the artful energy of the songs, no matter how makeshift the actual recordings were. "The demos had been recorded very simply in rehearsal rooms and kitchens," Johansson explains, "so they were very rough, but they worked well. The album is just kind of a luxury version of that. "My first feeling was 'I have to make sure that I don't overproduce this.' But I did! It took some time and actually it took some help from them to get into thinking indie again, to get thinking that the important thing is to get this band sounding right as a band and not to go wild in the studio and start recording a lot of vibraphones and backing vocals and things. But I didn't realise that until we started to mix and I had to mute quite a few tracks because it felt overproduced.

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"Working with the Cardigans had been different because they're very much a studio band, so they come in without having played the songs at all and develop their music in the studio. But with a band like Franz Ferdinand, they come to the studio with the songs and they can play them and they have their sound. So it's more about trying to find some working model of how to capture it." The main live area in Gula's Mothership.

Into The Mothership The main room at Gula, the Mothership, is built around a Neve 8014 with 16 x 1066 modules. "It's really good for basic recordings. It's hard to fail when you're recording with that kind of stuff. If I want something extra, some more expressive EQ or compression, then I use other stuff. I really like the Neve 33122. There are a lot of those late '70s preamps and EQs around, but the Neve is the most aggressive one I've used. If I want something more edgy, I use that. The midrange is fantastic, you can bring out the guitars on their own shelf." No matter what the production, Johansson always turns to a 16-track two-inch MCI JH16 reel-to-reel recorder for initial tracking. "I think we've had three of those over the years, we really like them. We were really lucky because when we started Tambourine we bought a whole studio that went down and we didn't know anything about equipment. So it just happened to be that tape machine and a couple of good Dbx compressors and Franz Ferdinand's material was initially this old Amek desk from the mid '70s tracked to an MCI JH16 reel-to-reel recorder. that had belonged to Manfred Mann's Earth Band. Desks always have a history. Once we went to the UK to buy a desk and the guy told us that it'd been involved in the late Beatles recordings — and it'd actually been made in 1972! With all of those EMI desks and compressors it's always 'Yeah, this was at Abbey Road...'" Naturally enough, in sticking with the JH16, the producer swears by the benefits of tape compression and admits he isn't afraid of pushing the recording levels. "I record very high and keep on listening back to the tape. I have the drummer play and I go up, up, up until the bass drum flattens out and then I back down a couple of dBs. Because there's something extra at that point, some kind of mushy effect that's very musical. I haven't tried all the plug-ins that you get for



Pro Tools, so I can't really say if there's anything else can emulate that. But I have the tape recorder available, so I don't need it. I've also tried to get that tape compression effect on 24-track and it's quite a bit different." For monitoring in the Mothership, there are KRK 6000s — which Johansson says he never uses — and Altec 19s. The Altecs are an old favourite. "They're a bit harsh when it comes to listening to modern music, but if you're into doing '60s and '70ssounding productions, they're great. They have that kind of presence that modern speakers don't really have. "Otherwise, for a couple of years I've had these American speakers from the '60s/'70s called EPI or Epicure. I have a small model called the EPI 100 with one eight-inch woofer and a concave tweeter and they're Franz Ferdinand, taking more than their so good. I'd have to go up to $10,000 to get musical influences from the early '80s... something that have a similar sound and these cost me almost nothing, maybe $200. I bought them from a company in America called Human Speakers. Normally when I switch between nearfields and those big Altecs, the difference is too big and I can't make up my mind. But when I switch between the EPIs and the Altecs, it's fine. They're so much more like a proper studio monitor. Everyone who passes the studio comments on how great they sound and they can't believe how cheap they are."

Luxury Spill Sessions for the Franz Ferdinand album began in June 2003 when the band arrived in Malmo for a month of recording. Since they were so well-rehearsed, Johansson decided to try to record as much of the basic tracks as possible live, setting the band up in the live room at Gula, with their amps cranked as high as they normally are in rehearsals and on stage. He says that limiting potential problems with spill wasn't high on his list of priorities. "Of course there's spill, but in Gula it's a luxury kind of spill, it doesn't have a rehearsal-room sound to it. Sometimes they played with headphones and sometimes they didn't, depending on the song. Sometimes we wanted to separate the drum sound more and so the amplifiers were too low in the mix for the drummer to hear, and we had to use headphones." Rather than ship in a dozen different guitar/amp configurations, Tore and the band decided to stick with their tried and trusted setups. In the end, nearly all of singer/guitarist Alex Kapranos and guitarist Nick McCarthy's parts were recorded file:///H|/SOS%2004-06/Recording%20Franz%20Ferdinand.htm (4 of 9)9/22/2005 7:45:41 PM


using the former's battered old Selmer amp and bassist Robert Hardy's Fender Bassman 100. "Both Nick and Alex swap around playing rhythm and lead guitars. Nick has a tendency to play those kind of funky Talking Heads/ Devo parts and Alex plays the more orthodox rock guitar parts." Less orthodox is Johansson's approach to recording drums. For a Tore Johansson's favoured nearfield start, he often gaffer-tapes two bass monitors are these battered Epicure EPI drums together lengthwise and places 100s. a 57 inside, which he says never fails to give him a great kick sound. Elsewhere on the kit, he maintains a minimal mic setup: another 57 on the snare, 'anything' on the hi-hats and a pair of Sennheiser MD421s doubling up as tom mics and overheads. "I also love the BBC AM6/3 compressor we have, which is part of a lot of stuff that we bought from BBC TV that we butchered and turned into modules. It's very aggressive, really fast, you can use it on a snare drum, no problem. It distorts and it's very noisy, but it's good in a musical way." The producer is also very keen to sing the praises of Franz Ferdinand drummer Paul Thomson. "He's brilliant, one of the best indie drummers I've ever recorded. I mean, he could hardly walk... he was one of those guys who was always stumbling round the studio, spilling milk into the desk and stuff! But when he got behind the drums he was brilliant. So it was easy to do those driving songs because he was driving them. The rockier the song, the easier it was to record. A lot of them sounded kind of finished even from the basic take." Even the distinctive intro of 'Take Me Out' — in which the band usher the song in by way of a punky introduction before slowing the tempo down into the main groove of the track — was actually recorded live, rather than created through editing. "There was no editing other than that we compiled the best take. We didn't do any tricks of, like, doing that on a separate take and then doing the rest of the track. They actually did the whole track with the tempo changes because we were in the situation where they'd played it live quite a few times, so they could actually all slow down in the same way."



Farming The Mix Two tracks on the Franz Ferdinand album, 'Tell Her Tonight' and 'This Fire', were recorded by the band themselves and then mixed by Johansson. "Actually we recorded those songs here, but we weren't happy with the results. We were trying different tempos and stuff and we'd experimented quite a lot with them in the studio. But then we didn't have time to continue to record them because the band had to go to England to do some touring. So they had a couple of days off and then they did the recording of the songs themselves in Scotland. It ended up that they wanted to play them more like they play them live. "So then I mixed them in Pro Tools. At first I had problems with them because when I just mixed them normally, they tended to sound a bit wimpy and clean, '80s kind of sounding. Then once I realised they'd been recorded with a bit too little compression, I started to experiment putting the whole mix through the Neve compressor. And that was quite good, until I actually did something that you're not supposed to do. I actually put the whole mixes through [Line 6's plugin] Amp Farm — no speaker though, just the amp. So I used the cleanest amp, the Fender Twin, and then I had to fiddle around a lot with the equalisers because it was way too bright. "If you don't use the speakers in Amp Farm, it has this tendency of making everything sound bigger, almost like those Aural Exciters in the '80s. It's simulating some kind of tube distortion and you get that extra twinkling brightness on top of it and a bit of extra bass. When I did that, those two songs sounded more like the things we'd recorded here. If you've got recordings that sound good, but wimpy, that's a good trick."

Out Of The Control Room Once the basic live passes had been committed to 16-track, Johansson threw them over into Pro Tools on his Apple G4 and began editing together the takes until he and the band had one comped performance that they were all happy with. From there on in, all the overdubbing was done in Pro Tools, which the producer first began using when making the Cardigans' Gran Turismo and hasn't really bothered to update since. "I'm not a computer buff," he insists, "so I don't have the energy to use new programs and stuff. I started off with Pro Tools, learned that and so I stay with that. It will have to come to a situation where everyone is saying that the latest thing is so much better before I decide to change. I think I'll wait for the next generation because it's always a couple of generations before you see a big change. "My favourite development has been Autosave, because it saved a lot of pain looking for audio files and lost sessions. I've hundreds and hundreds of old sessions, so I can go back any time I want. You can think maybe it was a better



groove last Friday when you worked on the song, so you can go back. Also if I want to upgrade now, I'd have to start using OS X on the G4 and that's just a nightmare. I've checked out Logic, but I feel that Pro Tools looks and feels more like an analogue studio. It's easier to find your way around." Once an album reaches the overdub stage, Johansson likes to get out of the Most of Alex Kapranos's guitar parts were control room, setting up his G4 and recorded using this ancient Selmer valve EPIs in the live area. "Normally I sit in combo. the corner of the big room with my system. It just means that I can have assistants editing vocals or whatever in the control room. Normally I'm quite efficient and I like to get a lot of things done, so they can be compiling a vocal and I'll sit in the live room doing a rough mix or something. Also I like to sit in a room where there are instruments, so I can get a guitar amp up or whatever. I can't sit in control rooms for too long. "When I'm working in the big room, if I'm not using the EPI 100s, I use AKG 1000s, those weird headphones that don't connect to your head — they're speakers that hang at either side of your ears. They're really good because they're inbetween listening to speakers and headphones, and because they're not actually on your ears, you can have them on forever, your ears don't get tired."

Hard Vocals In keeping with the live nature of the sessions, singer Alex Kapranos would usually manage to nail each of his vocals within two or three takes, in an attempt to capture the 'attitude' of the performance. "We were certain that we wanted a very, very hard, compressed, distorted sound," says Tore Johansson, "so we actually recorded it like that. I'm a fan of rolling off a lot of bass before the compressors because I The Neve compressors were used in tandem think they work better when they with an 1176 to create Alex Kapranos's gritty don't get a lot of bass. Quite often vocal sound. we used the 421 and rolled off a lot of bass using the filter on the microphone. So we had this very, very thin sound going into first the Neve compressor — very heavy, almost distorting — and then the Universal Audio



1176, so we had double compression. The sound you hear is the sound we recorded. Then sometimes we'd spend a couple of hours adding backing vocals and stuff. But normally we ended up with quite a simple version of Nick shouting in the background, which is what he does live."

More Thrash Once recording was complete, Franz Ferdinand returned to Scotland and let Johansson get on with the preliminary mixes. As mentioned earlier, some of the slower tracks on the album had seen good use being made of Gula's array of synths, celestes and vibraphones during the overdubbing process, but very few of these parts actually made the final mixes. An unusual but effective approach to getting "I did some basic mixing first and then Alex and Paul came over for the proper a great kick-drum sound, using a Shure SM57 inside a double-length drum. mixes. Normally they listened to my mix and then said 'No, it's too good! It has to be more thrashy!' Then I found myself having to mute most of the keyboard parts. Normally they'd played everything themselves — they all swapped around on the keyboard parts. Paul is a very musical guy. He actually tried to play bass on a couple of tracks, but I don't think we used his parts. Overall, they wanted to scale it down and have it sound more like a band playing, not too produced. On 'Take Me Out', I put almost all the instruments through a couple of echoes to get that marching, machiney, industrial feel to it. It's very organic, but we wanted it to sound like you're in a big workshop or something."

As opposed to the recording process, where the rockier tracks had been relatively easy to commit to tape, during mixing Johansson found that the more produced tracks were easier to balance, while the rock songs proved trickier. "Tracks like 'Jacqueline' were hard. We ended up spending a lot of time going half a dB up and down on different guitar parts and it was quite hard because, you know, when you have three guitars playing kind of the same thing, it's hard to get the perfect mix. But on tracks when you have an acoustic guitar and an electric guitar and a piano, it's easier to get the levels."

Praise Indeed Impressive sales aside, Johansson is clearly very happy with his work on the Franz Ferdinand album. So much so, he affords it perhaps the highest



compliment any producer can pay an album that they've worked on, day in day out, for weeks on end. "It's one of those albums that you actually want to listen to when you've done it!" he laughs. "I think that's because I actually didn't overproduce it and just captured the band. I think I did a good job of recording it — it sounds good — but it's very much the band sounding like that. So it's all thanks to them really." Published in SOS June 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:

Monitoring Assessments Setting Up For A Big Drum Sound Adding The Guitars & Bass Kill That Spill Mastering Tweaks Ready To Rock The Band's Comments

Studio SOS The Arcades Published in SOS June 2004 Print article : Close window

People : Studio SOS

This month, the SOS team help The Arcades to rock even harder than before! Paul White

We were lured to The Arcades' studio by the promise of chocolate Hobnobs and French Fancies, the mission brief being to improve the way they recorded their four-piece rock band. Their influences, at least sound-wise, were Led Zeppelin, Deep Purple, and AC/DC, so at least we knew what we were aiming for. When we arrived, we discovered that the studio actually comprised two wooden outbuildings which had been built by brothers Greg (drums) and Graham (lead guitar) who both live where the studio is located. One room was for use as a control room and the larger one was the live room. Greg had completely wired the studio with balanced wiring for the best signal path from the live room to the control room — they were separated by a couple of metres and, as the band do their own engineering, it was pretty much a matter of 'last one out hit record'! The live room was almost entirely lined with mattresses, foam, and reclaimed sound tiles, which in combination with the bass-trapping nature of the wooden building gave a reasonably dead but fairly well-balanced sound, not unlike that of many professional studios during the '70s. There was some sound leakage to the outside, but as the studio is out in the countryside this wasn't a problem. The control room measured about 2.5 x 3.5m and was lined out with batten and plasterboard. Some acoustic foam tiles were fixed to the rear wall, ceiling, and side wall, but we noticed straightaway that there was a patch of bare wall to the right of the mixing position that was likely to throw back reflections from the monitors, so we suggested adding another foam tile or moving one of the existing ones. Similarly, although there were two foam squares on the ceiling, the part most likely to produce reflections hadn't been covered, so we suggested doing the same there. file:///H|/SOS%2004-06/Studio%20SOS.htm (1 of 10)9/22/2005 7:45:45 PM

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Monitoring Assessments As has now become routine on our visits, the first thing we looked at was the monitoring system, which comprised a pair of Genelec 2029As and the matching 7050A subwoofer. The speakers were set up at the correct heights and angles on rigid shelves either side of the band's two computer monitor screens. On playing back some commercial material, we felt that the sub was turned up a little too high, and set about rebalancing it by ear. Because of the natural bass trapping of the building and the fortuitous location of the subwoofer under the mixer table, the bass end was reasonably even (no noticeable level differences between different bass notes) so no further tweaking was necessary once the level had been turned down by about 2-3dB. However, it is difficult to set up subs accurately without specialist test equipment, so it is recommended that mixes be played on as many different systems as possible and that then, depending on the results, the subwoofer level be adjusted again accordingly. If the mixes are generally perceived as being bass light, then the subwoofer level is probably too high, which causes the engineer to reign in the bass end to compensate. Naturally, the exact opposite is the case if the subwoofer level is too low, as that fools the engineer into adding more bass.

The hole in the front of the kick drum's front head was too small to allow a suitable mic position, so it was removed. The sound was then damped to taste using a pillow resting in the drum shell, and some extra beater definition was added by taping Paul's RAC card to the point where the beater hit the drum head. Finally, the drum was miked using an AKG D112. This was placed centrally to start with, but auditioning showed that a position slightly to one side produced a better sound.

The Arcades' recording system is based around a PC running the latest version of Steinberg's Cubase SX, which they're still getting used to. The audio interface is a MOTU 828 MkII and the mixing console is a recently acquired second-hand Spirit Studio model, though it seems to have a couple of temperamental channels. We didn't have time to remedy this, but Hugh suggested wiring up a TRS jack plug with the tip and sleeve connected together to push into the insert points of the suspect channels, since it appeared that dirty or damaged insert sockets were causing some of the problem. If the insert socket was at fault, leaving the jack in place should fix the problem. file:///H|/SOS%2004-06/Studio%20SOS.htm (2 of 10)9/22/2005 7:45:45 PM

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There was also a small amount of outboard equipment, though most processing tended to be done in software. By utilising all the 828's analogue inputs, the band could record up to 10 channels simultaneously, which was more than enough, as only nine feeds were needed to record the two guitars, the bass, and the six drum mics. The band's own Lexicon MPX100 reverb was broken, so I brought along my own Lexicon MPX550, which includes some useful ambience and stone-hallway algorithms that I thought might help us in getting a big drum sound. This was patched into one of the aux sends on the Spirit desk and brought back via a pair of return channels before we started work. Before trying anything new, we listened to the multitrack playback of a song the band had been working on. It didn't sound at all bad, but it lacked the sense of space and power they were after, and the boys were particularly keen to improve the drum sound if possible. The band were also experiencing spill problems when they all played together, but as is so often the case with this type of music, the recording loses its energy if you try to layer it up as individual overdubs.

Setting Up For A Big Drum Sound We decided to tackle the drum kit first, and though the band had a decent selection of mics, they had already told me they didn't have two identical capacitor mics to use as overheads (other than a pair of AKG C1000s, which didn't sound particularly good as overheads), so I brought along my own SE Electronics SE1s, as I thought they'd do the job and they were in a price range the band could afford. These are small-diaphragm, cardioid-pattern capacitor models and are surprisingly good all-round instrument mics for anyone on a tight budget, though pretty much any of the current crop of budget cardioid capacitor mics (small or large diaphragm) from the Far East would do the job well enough. Before setting up the mics, I checked over the drum kit and found that the kick's front head had only a small hole cut into it, which meant I couldn't position the mic correctly. Greg removed the head of the drum while I checked the rest of his DW kit. He'd tuned the toms using a specialist skintension tuning device, which I'd never come across before, so the tension was pretty even and the tone was OK. However, to get more of a rock sound I tried slackening just one lug on each tom by around half a turn to get a hint of pitch drop after the drum was struck. This worked nicely, but we had too much ring on the toms, so I taped a small wad of tissue, around one inch square, to both the top and bottom heads. This dried up the sound without killing all the natural ring.

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The snare was a perfectly tuned, wooden-shelled job around four inches deep, so it had a bright lively tone. It was never going to sound deep and fat, but it sounded good and wasn't too far away from the AC/DC snare sound the band liked. That left the kick, which was again already perfectly tuned, so once the front head had been removed, we adjusted the integral damping pillow and I taped my plastic RAC membership card to the head where the beater hits to get a bit more of a snap into the sound. I did remember to retrieve it before we drove home, through! For miking, we used the band's own Shure SM57s on the snare and toms, all in the traditional positions around two inches up and two inches in from the rim. Where possible, these were aimed away from the cymbals to minimise spill. The SE1s went up as spaced overheads about eighteen inches above the cymbals, and an AKG D112 was set up on a short boom stand so that it sat roughly in the Although the toms were well tuned, they were ringing a little too much, but it was centre of the kick drum shell, aimed at nothing that a bit of masking tape and tissue the beater. The ceiling above the drum paper couldn't deal with! kit had been covered with proper, reclaimed acoustic absorber units which soaked up a lot of the cymbal noise and prevented it from splashing all around the room. A test recording showed that the kick drum still sounded a little tubby, so I moved the mic slightly to one side and added EQ boost around those old standby frequencies of 70Hz for depth and 4kHz for click. This sounded much more solid, and when a software gate was applied in Cubase to cut out the spill from the other drums it sounded very sweet indeed. The overheads were picking up a good overall kit sound and bringing up the cymbals, but I rolled some low end off these mics, as I find that often gives a clearer sound overall by avoiding conflict with the close mics. Where the individual drums are going to be panned to create a stereo image, care must be taken to get the close-mic pan positions to match the image captured by the overheads, but in most cases you can get a very solid sound by leaving the close mics panned centre and then relying on the overheads and stereo reverb to add a sense of width and space. I've always had problems when miking double-headed toms, in that they ring almost continuously, because they invariably resonate in sympathy with the kick drum. There's little you can do about this other than damping all the life out of them, so my preferred option is to gate them after recording, which is exactly what we did. To avoid producing an unnatural and completely dead sound, we set the gate range to about 12dB, so that it never actually closed completely, just reduced the spill by the 12dB. The attack was set very fast, with a short hold time and a release time set to follow the natural decay time of each tom. This tidied up the sound immeasurably, but if you have the patience it's even better to go


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through the tom tracks manually and silence all the sections between tom hits using your software's waveform editor. This doesn't take as long as you might imagine, and it ensures that nothing comes through on the tom tracks except toms. It also gives you the opportunity to do a destructive gain change on any hits that are too loud or too soft. The other advantage is that this way you can ensure the initial Paul experimented with adding different transients are retained in all their glory, patches from his Lexicon MPX550 reverb unit, and this yielded a bigger, more whereas some analogue gates tend to spacious drum sound. remove or significantly reduce that initial transient 'thwack'. If you're using a software gate, engaging the lookahead facility can avoid the transients being damaged, as the gate will open slightly before the transient arrives. Other than the kick-drum EQ and the low-end roll-off on the overheads, very little EQ was needed to get a nice punchy drum sound, after which I tried adding a Lexicon ambience program to the whole kit to try to approximate that 'recorded in a rock star's mansion' sound. This did actually work quite well, and while adding reverb to kick drums isn't usually a good idea, short ambience programs work fine, adding depth and space without muddying the sound. As an alternative, I also demonstrated the Marble Foyer patch in the MPX550, which gave a bigger, more live sound that was still not too busy. Of course nothing sounds quite like a real live room, apart from perhaps a convolution reverb with a suitable impulse response, but we were all pretty happy with the overall effect.

Adding The Guitars & Bass Having got the drums sorted, it was time to turn our attention to the guitars and bass. The bass had been DI'd from a socket on the amp, and though this doesn't always sound great it was a good starting point. However, the bass and guitars had been tuned down a semitone for artistic reasons (apparently it sounds heavier!), which had left the bass with a noticeable amount of fret rattle that was plainly audible on our recordings. Rolling off some of the high end helped, but the only true solution was to have the bass set up professionally to get rid of the rattle at source. For the sake of our tests, we plugged a jack into the bass file:///H|/SOS%2004-06/Studio%20SOS.htm (5 of 10)9/22/2005 7:45:45 PM

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amp's extension speaker socket to kill the speaker feed (this is OK on solidstate amps but not wise on tube amps!) and everyone monitored on headphones while playing. A liberal helping of the Waves Renaissance Compressor plug-in helped even out and thicken the bass sound, and, other than the odd rattle, it Turning the guitar cabinet round to face the sounded quite acceptable with very little absorbent acoustic treatment on the studio wall significantly reduced the acoustic spill in the way of EQ. However, I do feel that you get a much punchier rock bass into the drum mics, and a small gap left between the speaker and the wall allowed sound from modelling preamps such as an AKG C1000 to be positioned for a very the Line 6 Bass Pod or Behringer Bass respectable recorded sound. V-Amp than from a straight DI, as DI'ing loses the important coloration of the speaker cab. Even the guitar versions of these modelling processors produce great bass sounds if you team up something like a Fender Bassman amp model with a 15-inch speaker model. We still had to deal with the guitar spill problem, and I also felt the guitar sounds weren't focused enough, so I spent some time adjusting the amp controls while the guys played. Fortunately, they both used power soaks (THD Hot Plates from America) so that they could play with their big Marshall amps (a 1959 SLP and a JCM800) set to '11' without deafening each other — though I did notice a few packets of disposable ear plugs in the live room! By turning off the Bright switch on the power soaks and restoring the bite using the amps' Presence controls, I found that we got a sweeter, less gritty tone that better suited the musical style they were aiming for.

Kill That Spill Sorting out the spill problem was easy — we simply turned the sealed-back 4x12 cabs around to face the mattress-lined walls, leaving just enough space to get a mic stand's boom arm in. Not only did the wall treatment massively reduce the sound from the guitar amp bouncing back into the room, but it also minimised the amount of drum sound reaching the guitar mics. While SM57s are an obvious choice for recording guitar cabinets, its worth trying every mic you have, as some combinations can produce surprising results. We decided to try the AKG C1000s, as they have quite a smooth top end not unlike that of a dynamic mic, and these were placed up close and personal, aimed directly at the centre of one of the speaker cones. This is just a starting position, and if the tone is too hard you can warm it up by moving the mic slightly towards the edge of the speaker cone. You can't get a great guitar sound just by following textbooks — you really do have to experiment with mic types and positions to get the best results in any given situation.


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At this point we made a test recording and found that the amount of spill between guitars and drums was minimal. The guitar spill to the drum overhead mics was around 25dB down, and likewise the drums reaching the guitar mics. This separation provided far more clarity to the sound overall, and allowed greater scope for individual processing of tracks without spill colouring the effects. It also made it practical to compress the guitars without pulling up loads of drum spill, and to put ambience reverb on the drums without making the guitars sound like they were in a bathroom. The sound we got was already an improvement on what the band had Because both guitarists were using power recorded earlier, though I felt the soaks (inset), monitoring levels in the guitars needed more bite, but without recording room could be kept fairly low. It making them sound gritty or edgy. also meant that Paul could comfortably stand Even though the two guitarists had by the cabs tweaking the amp controls while very different guitars (a Fender the guitarists played, even though the amps were cranked up all the way, so he could Stratocaster and a Gibson Les Paul), quickly optimise the recorded sound at the solution was the same: 2-3dB boost at around 700Hz, which gave the source. Les Paul more of that AC/DC Gibson SG honk and made the Strat sound less brittle. When I first tried these EQ settings on the soloed guitar tracks, the band were a little unsure, but once the rest of the mix was up and running, it was agreed that the guitars sounded more punchy and sat better in the mix. Another small tweak we made was to reduce the amount of overdrive used on the lead guitar part and then to insert another Renaissance Compressor to maintain the sustain. Again the result was more clarity and punch. If you record using too much overdrive, you run the risk of ending up with a very indistinct 'snails in a blender' sound that eats up all the space in the mix but still doesn't cut through. As an experiment, we also inserted the band's stereo SPL Vitalizer into the stereo drum submix and used it to add more weight to the low end. The highend controls were then tweaked to add a little more shimmer to the cymbals. Provided that these devices are used with care, they can add weight, clarity, and depth to a track, submix, or full mix, but the secret is not to overuse them.

Mastering Tweaks By the end of the session, we had recorded a complete backing track that only file:///H|/SOS%2004-06/Studio%20SOS.htm (7 of 10)9/22/2005 7:45:45 PM

Studio SOS

needed the vocals to be overdubbed, and the band were all surprised by the difference some simple adjustments and techniques had made to their sound. The band then asked about mastering processes that could be applied to polish up their finished mixes. The difficulty about advising on this subject is that every track is different and, more importantly, that you can only make accurate With the bass guitar DI'd and the guitar cabs turned towards the walls, there was ample judgements if you can rely on your separation between the different instruments monitoring system to be accurate. for the whole band to record together in the Some people also believe it's same room. impossible to make a good-sounding master using plug-in processing, but I dispute this — great tools obviously help, but if you have good ears and good monitors, you can do wonders with relatively simple plug-ins or pieces of outboard equipment. The reason material is mastered at all is largely to do with making all the tracks on an album match each other tonally and in relative level (which must be judged subjectively rather than by meters alone), but these days there's also a demand for more loudness and also for making the music sound even better than the final mix. The main tools are EQ, compression, and limiting, where careful use of a nice-sounding equaliser can be used to tame hot spots, scoop out a boxy midrange, and add a high-frequency gloss. Most times this ends up being a variation on the classic smile curve, where the mid-range is dipped slightly and the high end (12-14kHz) is boosted by a couple of decibels using a very low-Q setting. However, you can't be too formulaic about this — you have to understand what the mix needs and then use modest amounts of EQ to achieve that. As the band have a Vitalizer, they could use that to do much the same thing. Compression is also important, but unlike individual track compression, where you typically set a high ratio and a high threshold so that the compressor only stamps on signal peaks, the settings used for mastering tend to involve very low ratios of between 1.1:1 and 1.2:1 using threshold settings of -35dB to -40dB. This means that the whole dynamic range of the music is gently squeezed, Getting the right mix of guitars and drums and gain-reduction readings of 3-4dB initially proved tricky, but once the guitars are typical. I usually put the had been EQ'd a little, Hugh was able to find compressor before the equaliser, but a good balance. others prefer to compress afterwards, so try both to see if you can hear a difference. The final stage is to apply limiting so that you can push up the file:///H|/SOS%2004-06/Studio%20SOS.htm (8 of 10)9/22/2005 7:45:45 PM

Studio SOS

average signal level, and as before a gain reduction of around 4dB on signal peaks is usually enough to achieve the desired result without over-processing the sound. As the band had a good selection of Waves plug-ins, my approach would be to try the Renaissance Compressor first, followed by the Q4 parametric equaliser, though a version with more bands (up to 10) is available if there are any awkward areas that need individual attention. The last stage would be the L1 limiter, which allows the user to set the peak level (usually around 0.5dB from clipping) after which the input gain is increased until the desired amount of gain reduction shows on the signal peaks. Getting all this just right requires practice, but if you err on the side of under-processing rather than over-processing, you should be OK. Of course it also helps to compare every stage of your mastering with a commercial record in the same musical style played back over the same monitor system at the same level. In this case the band were after a Led Zeppelin kind of sound, but some of Led Zeppelin's early records sound lacking in punch compared with modern productions, so choosing a good variety of material would be safest.

Ready To Rock Despite us only being on site for a few hours, we all worked together effectively to improve both the guitar and drum sounds while also solving the spill problem. Once we'd turned the guitar speaker around, soloing the individual guitar and drum tracks showed virtually no spill other than on the drum overheads, where a very low level of guitar was audible. Listening to the new final mix side by side with their previous efforts, the band could already hear a big difference. However, comparisons to commercial releases showed that some careful mastering processing would still prove beneficial.

The band had commented that they found it hard to record the vocal overdubs, which they had been doing facing directly into a heavily foam-lined corner of the room. This arrangement meant that the foam was soaking up the voice, making it hard for the singer to hear himself — he consequently ended up straining to sing louder. Furthermore, this setup wasn't preventing the ambient sound and reflections from reaching the mic, which was facing out into the room behind the singer. Hugh suggested that a better way of working would be to place the singer a little out from the corner, looking out into the room, with the microphone facing into the corner. That way, the vocalist wouldn't feel like a dunce sent to the corner, and would find it easier to sing at a more comfortable level, while the foam absorbers behind him would prevent spill and ambience from being reflected back into the file:///H|/SOS%2004-06/Studio%20SOS.htm (9 of 10)9/22/2005 7:45:45 PM

Studio SOS

microphone. The only issue that needed outside help was the setup of the bass guitar to avoid the string/fret rattle. As mentioned in the main text, I think a bass recording preamp would give a more suitable sound, but the band also need to set up a better monitoring system with a headphone distribution unit that allows them to control their individual monitoring levels. As these are now so cheap, that is easily resolved.

The Band's Comments Recording as a band gives us the chance to keep the magic that happens in the room, so reducing the spill was essential and Paul and Hugh were able to give us hints and ideas on how to achieve this. We've since bought a headphone amp which is now enabling us to record live without ruining our ears — a perfect solution. The drum sound we got on the day was also good, and the reverb unit certainly gave the kit a sense of power. And what a relief to have someone else in the other room to hit record! www.thearcades.co.uk Published in SOS June 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



Thermionic Culture Valve Designs

In this article:

Engineering Beginnings Product Designs The Future

Thermionic Culture Valve Designs Vic Keary Published in SOS June 2004 Print article : Close window

People : Industry/Music Biz

We talk to a British designer who still thinks audio is best served by tube. Hugh Robjohns

Vic Keary in his workshop. A Culture Vulture can be seen on test on the workbench.

Among audio engineers the arguments over the virtues and demerits of valves (or tubes) versus solid-state electronics are almost as ceaseless and polarised as those over analogue and digital recorders. However, for those who favour glowing glass bottles, one of the best-kept secrets of the British Isles has to be a quiet little company called Thermionic Culture. Although they currently only manufacture three products, all three are unusual, distinctive all-valve designs. The designer behind these products is recording-industry veteran Vic Keary, who argues very coherently that every stage of the audio signal chain — amplification, compression, mixing and equalisation — all took a step backwards when solidstate technology became commonplace in the late 1960s. Consequently, none of his products have any transistors or ICs in their signal paths at all, and the few solid-state components that are used are restricted to serving in the power supplies — where they have distinct practical advantages.

Engineering Beginnings Vic has a long and colourful history in the British recording scene dating back to the 1960s. He built his very first semi-professional recording studio in 1957 above a cow-shed — and 'built' is the right word. In those days you more or less had to build everything yourself, as there was very little in the way of commercially produced equipment. From this novel beginning, Vic's professional career began in 1960 as a maintenance engineer at Lansdowne Studios in file:///H|/SOS%2004-06/Thermionic%20Culture%20Valve%20Designs.htm (1 of 6)9/22/2005 7:45:50 PM


London. Among his engineering successes at the studio was a significant modification to the EQ of the studio's EMI console — a design which has been carried over to Thermionic Culture's forthcoming Merlin EQ. In the early '60s, Vic progressed up the ranks at Lansdowne to become a mixing engineer in his own right, crafting several hits by the Terry Lightfoot Jazz Band, among others. Possibly his best-known chart success from that era, though, was Acker Bilk's 'Stranger on the Shore', which he mastered. After leaving Lansdowne in 1964, Vic worked with a company called Rush Electronics, and during his brief time there he designed some valve compressors derived — but very heavily modified — from an early Altec design. Vic's design had the unique The Phoenix compressor, as reviewed in feature of employing both a neon and a SOS April 2000. bulb illuminating a light-dependent resistor (LDR) as the gain-controlling element. Most opto-compressors have fairly slow attack times because of the relatively slow light build-up of an incandescent bulb, but in Vic's design, the neon bestowed the compressor with a fast attack time, while the bulb controlled the release time. One of these unique compressors was sold to Pete Townsend, and he apparently still uses it today (allegedly with the original valves!). Elements of that revised design later went on to form the heart of the Chiswick Reach compressor, which Vic designed, and more recently Thermionic Culture's Phoenix. Vic couldn't stay away from the recording business for long, and soon left Rush Electronics to set up a London-based three-track recording studio, Maximum Sound. Vic built the studio's 10-channel console himself, based on the design of the EMI console at Lansdowne, and expanded the studio to four-track over the years. Eventually, the studio acquired quite a reputation for recording successful ska and reggae music. Vic continued this success at his next venture, Chalk Farm Studios, which he founded in 1968 following the sale of Maximum Sound to Manfred Mann. He reinstalled his valve desk along with an eight-track Leevers Rich recorder, and some of his compressors. Reggae and soul success soon followed — at one time, eight of the top 50 singles had been recorded at Chalk Farm, Vic having mixed five or six of them! Dandy Livingston (then Trojan's chief producer) made many of his records at the studios, and Harry J produced Bob and Marcia's single 'Young, Gifted and Black' — the studio's first hit — even though the equipment was pretty basic by today's standards. Vic recalls today that the outboard comprised a single spring reverb, and that the unique delay on the strings on that record was achieved by mixing outputs from both the sync and replay heads on the eight-track! The studio gradually expanded to a 16-track facility (and later to 24-track), file:///H|/SOS%2004-06/Thermionic%20Culture%20Valve%20Designs.htm (2 of 6)9/22/2005 7:45:50 PM


outgrowing the original console, so Vic built a larger 20-channel version with a separate monitor mixer, which was better suited to multitrack working. Sadly though, as the popularity of reggae diminished at the turn of the '80s, so too did the studio bookings, and Vic was forced to close the studio in 1982. For the rest of the decade he scratched a living from freelance studio maintenance, vinyl disc mastering and occasional location recordings. Vic was drawn back into the recording studio world at the beginning of 1990 after the private studio of an acquaintance was burgled and stripped of all its equipment. He agreed to install some of his own gear to get the place up and running again, and the large proportion of valve outboard quickly became the highlight of the facility. Many customers suggested setting up an all-valve studio, so Vic did just that in 1992 at a converted brewery near the river Thames in Chiswick, West London, naming it Chiswick Reach, and was joined there the following year by Nick Terry, a valve-obsessed recording engineer. The studio used Vic's (by then) 28-input, eight-buss, 16-monitor channel valve console from the Chalk Farm days, together with his impressive Thermionic Culture's cocollection of valve outboard equipment. This founder Jon Bailes. included Leevers Rich graphic EQs, an EMT plate, and a Sean Davies valve limiter (made in the 1960s for IBC Studios, where it was apparently used on The Who's 'My Generation'). Tape recorders included a 3M M79 24-track and a Brenell Mini 8 (both dating from the 1970s), as well as a Leevers Rich E242 two-track valve recorder, which Vic still owns! Vic left Chiswick Reach in 1998, but continued to design and build valve audio equipment for friends and colleagues, and this led to the idea of manufacturing his designs commercially. To this end, Thermionic Culture was founded in 1999 by Vic and Jon Bailes, a designer in the electronic manufacturing industry. While Vic develops the circuit designs, Jon is responsible for the mechanical design, visual appearance and production engineering of the products. Nick Terry stayed at Chiswick Reach for a few more years honing his skills, before leaving to work as a freelance engineer/producer on projects such as McAlmont & Butler and The Libertines, and has now become the third partner in Thermionic Culture. Nick is responsible for the testing and quality-control side of things, and also provides a lot of the ideas for new or improved products. He often uses Thermionic Culture products on his recording projects — including some prototypes — and this helps him to evaluate the designs and provide realworld feedback for their development.

Product Designs file:///H|/SOS%2004-06/Thermionic%20Culture%20Valve%20Designs.htm (3 of 6)9/22/2005 7:45:50 PM


Some of the Thermionic Culture circuit topologies and concepts can be traced back to the classic valve designs from the 1940s, but a lot of the detailed designs involve Vic's own innovative work. Regardless of the source of inspiration, all of the valve circuitry has been revised and optimised to achieve the performance, low noise levels and minimal distortion expected of modern recording equipment. High-spec components are used throughout, including one-percent metal-oxide resistors and polypropylene capacitors where they influence the sound quality and reliability. Valves are selected by hand (and matched where necessary), usually from military or industrial types to ensure the longest possible life and accuracy. Many are rather unusual models not normally seen in audio applications, but are employed where they bring worthwhile benefits to noise, distortion or reliability. Some of the company's products retain the point-to-point wiring which is traditional in valve equipment, but while this approach can have certain technical advantages, it also makes production slow and expensive. The intention for the newer designs is to start using Jon Bailes' expertise in designing printed circuit boards, and although these can take a long time to optimise, production is quicker and more cost-effective — which should translate into more affordable products. The Phoenix was the first of Thermionic Culture's products, and its design derives in part from the Altec 436 'vari-mu' compressor. This used 6BC8 triodes in a balanced (push-pull) configuration, but produced what Vic felt was unacceptably high distortion, so he improved the design during his time at Chiswick Reach, and it became known as the Chiswick Reach compressor (reviewed in SOS in February 2000; see www.soundonsound.com/sos/feb00/ articles/chiswick.htm). However, the noise floor was still not as low as Vic wanted, and further development eventually resulted in Thermionic Culture's Phoenix compressor (see www.soundonsound.com/sos/apr00/articles/phoenix. htm). Some of the noise-floor improvement came from changes to the power supply, but some also came from a redesigned front-end which used a PCC85 valve instead of the Chiswick Reach's 6BQ7A. The PCC85 tube was designed for FM radio applications and hasn't been used in an audio application before; it requires an unusual 9V heater supply. However, all these cumulative changes brought the overall noise figure below -100dB, improved distortion further to 0.1 percent, and enhanced the general reliability of the compressor. Vic also improved the machine's flexibility and performance over the Chiswick Reach design with faster attack and release times. As this history shows, Vic is an inveterate tweaker, and still feels there are advances to be made with valve technology. He is currently experimenting with a different output valve for the Phoenix. The Phoenix's 'vari-mu' label means it has a very gentle 'knee' (in other words, the compression ratio increases gradually around the threshold), producing a very subtle, transparent compression effect. This makes it ideal for enhancing vocals and solo instruments, adding both warmth and body as well as making the source more powerful — all in an unobtrusive way. It also serves well as an

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overall mix compressor. Apparently, over a hundred have now been sold. The Culture Vulture was Nick Terry's idea, and is essentially a distortion processor for adding creative dirt to individual instruments or complete mixes. The large rackmount unit, which is available in mono or stereo versions, provides triode or pentode distortion (mainly even and odd harmonics respectively), with the option of Engineer Nick Terry testing an Early Bird 2 including some first-stage feedback to mic preamp in the quality-control department. create a distortion character somewhere between the two. In this way, the Culture Vulture can generate a wide range of different and distinctive distortion effects, recreating the recognisable characteristics of different valve amp topologies (for more information, see the SOS review at www. soundonsound.com/sos/aug03/articles/culturevulture.htm). Given that the raison d'être of this unit is to generate distortion, the specs make entertaining reading! Base distortion is quoted as 0.2 percent, but at full stretch the unit produces a staggering 99.9-percent distortion. The company's third product is the Early Bird mic preamp. This was originally introduced three years ago, but it has now been redesigned by Vic. The new version — the Early Bird 2 — includes simple EQ facilities, with a high-pass filter, bottom lift, a broad mid-range control, and a top lift. The EQ is within a feedback circuit and the way it has been designed means that the filter slopes vary with gain settings. In particular, the high-pass filter has a 12dB-per-octave slope at low-frequency settings, but it softens considerably at higher settings, providing a gentle mid-cut when combined with the bass lift control. The circuitry is again based on a unique push-pull balanced design, but the basic topology is similar to an amplifier design employed in a Pultec equaliser. Unlike the other products, the Early Bird 2 uses fairly conventional valves throughout — ECC83s and ECC82s. Sowter transformers are used throughout Thermionic Culture's products — both for audio and mains — and in the case of the Early Bird 2, the input transformer offers dual impedance settings of 300 and 1200 Ohms, allowing both vintage and modern mics to be matched to the input. It requires a good solid-state preamp to deliver a signal-to-noise ratio of more than 100dB, but that's also what this all-valve design can achieve — combined with distortion below 0.005 percent, and an extended bandwidth to 80kHz (all at 44dB gain). These specs are certainly impressive, regardless of the fact that it is an all-valve design, and the sound is even more so. It has a very clean, slightly warm but open and airy sound quality — and the extended frequency response and very low phase shift make it a popular choice for both high-quality classical



and rock recording duties.

The Future Several new Thermionic Culture products are currently at the planning stages. The Merlin, for example, is a versatile equaliser with an interesting 'passive lift' feature. Also in the wings, with a launch provisionally scheduled for the end of the year, is the Nightingale — a valve recording channel with a mic preamp, simple EQ and a scaled-down version of the Phoenix compressor. But with Vic's history of console design — and the enormous commercial success of the studios equipped with his early consoles — an all-valve console seems an obvious gap in the product line-up. In response to this, Vic has told me that he and his partners are considering producing a Thermionic Culture console, but they are still deciding the exact form it should take. Clearly the market for such a desk would be relatively small and the cost relatively high, so getting the right facilities in the package is paramount. In my view, though, an all-valve console incorporating Vic's mic preamp, compressor and EQ circuits is worth waiting for. Watch this space... Published in SOS June 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



Your Correspondence

In this article:

A Moscar Or An Ooog? Dabbling With DAB

Your Correspondence Crosstalk Published in SOS June 2004 Print article : Close window

People

A Moscar Or An Ooog? It's always a wonderful pleasure to find the SOS April Fools jokes — very clever stuff. But what was that on page 41 of the Roger Lyons interview — a picture of a Moog Source with knobs on? Can we have the full story? Nigel Rushbrook SOS contributor Tom Flint replies: Roger Lyons admits that the synth is a bit of a mystery, and, since its appearance in the April edition of SOS, he has received a number of queries from interested parties. Not only does the synth have 'OSCar Programmable Music Synthesiser', printed to the left of the top panel, but it also bears the unmistakable Moog logo on its right-hand edge, so it's not surprising that it's caused some confusion. The synth came into Roger's possession a long time ago, shortly after a much-loved standard OSCar disappeared from his studio. Being an avid user of the instrument, Roger wanted to replace it, and contacted his local supplier of OSC synths to find out if they could help. Eventually Roger received a call from the shop, at about the time OSCar manufacturers the Oxford Synthesizer Company went bust, telling him that they had bought up the remaining OSCar stock and could therefore sell him a replacement.

Roger Lyons' unusual OSCar synth.

When Roger visited the shop he noticed the OSCar/Moog-branded synth amongst the normal OSCars, and asked about its origins. According to the shop's staff, it had arrived together with the other OSCar units, and was thought to be some kind of prototype.

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Your Correspondence

Having tried it in the shop, and established that it seemed to be more or less the same as an OSCar, Roger bought the synth and has had it ever since. Roger describes the synth as being identical to an OSCar in terms of its sounds and features, but with a Moog Source box and hand-wired innards. In fact, the case is the only Moog component. Roger explains: "If you look at a proper OSCar and compare it to this synth you will see that it is all laid out and labelled exactly the same. A Moog Source has just one knob, and the rest of the controls are membrane switches, and they are all in completely different places to an OSCar. So it's a Moog Source case and keyboard, but with totally different controls and front panel." To complicate matters slightly, Roger also reckons that the synth's pitch-bend and modulation controls are the same as those on a Roland SH101. Roger has come to the conclusion that the synth must have been a prototype, put together in the OSC factory using spare hardware taken from other synths, and eventually sold together with the rest of the stock. However, even that scenario has been called into question by a synth fanatic who contacted Roger, saying that there was only ever one prototype made of the OSCar, and that was built into a Wasp case! Roger himself is sceptical about that particular story, as it would be difficult to fit all the guts of an OSCar into a much smaller Wasp case. Possible evidence that the synth was an on-going prototype is the addition of MIDI, which was not available on the original OSCar when it was released in 1983. Curiously, Roger describes the MIDI parts as being "hastily fitted", and the machine was running what Roger believes to be the latest versions of available software when he bought it, suggesting that it was a work-in-progress synth, rather than a discarded experiment. Quite where the truth lies remains uncertain. Nevertheless the synth has now been bought from Roger by a collector in LA who took a fancy to the strange hybrid.

Dabbling With DAB I was stunned and staggered to read Dave Shapton's comparison of the digital version of Radio 1 with the FM transmission and see him conclude that 'the FM station won hands down' [Cutting Edge, SOS April 2004]. Now I'm seriously worried for Dave's ears, as, for the past five years or so, the FM version has been ravaged by Optimod multi-band compression. This is why DJs on Radio 1 sound like they are terminally asthmatic with sibilant, splattery speech. Having determined that the BBC supplied DAB with an un-Optimod-ed signal, I splashed out on a Videologic receiver to connect to my stereo system (note for Dave: some of us do listen to DAB through a high-quality speaker system!).


Your Correspondence

It was wonderful! At 192kb-per-second, and with no unnecessary processing, I was able to hear records as they would have sounded in my own CD player. Then one fateful day, the asthma and splattering was back. The Optimods had invaded DAB. I was in tears. Investigation showed that bit rates, with the exception of Radio 3, were down to 128kb-persecond as the Beeb prepared for a plethora of new narrow-casting stations. But why Optimod DAB anyway? After all, the DAB system allows the use of DRC (the Dynamic Reduction System) so the listener can choose whether to compress his audio or not. But only Radio 3 and Classic FM have enabled this feature; for the rest of the stations, it's take it or leave it. When I contacted the BBC, I was told that, "the near-high-fidelity output you seek from terrestrial sources has been consigned to broadcasting history", and, as if that wasn't enough, "the signal is considered adequate for broadcasting", an expression I thought was reserved to describe AM radio. Terry Bracey SOS contributor Dave Shapton replies: Thanks for your comments. It's good to get the debate about quality going and I think we probably agree on most points, including the state of my ears, by the way! That said, I definitely thought the FM versions of the radio stations I listened to sounded brighter than they did on DAB at 128kb-per-second. This doesn't necessarily mean they sounded 'better', although that was my first reaction. I spoke to technical people at the manufacturers of both the Bush and Videologic brand radios mentioned in the article and they both expressed their frustration at the way the low bit rates used by the BBC and commercial stations made their radios sound nowhere near as good as they potentially could. I think the issue of Optimod compression is a separate one, because the way Optimod mangles audio is quite different to the MPEG data-compression process. And, yes, it is completely inappropriate to apply dynamic range compression to DAB signals, since you can apply DRC on the fly at the receiving end according to metadata transmitted on a data sub-channel. This, surely, is the best way to do things: leave the settings up to the user. Unfortunately, I suspect that the Beeb is constrained by bandwidth restrictions. Without an additional national multiplex, they've simply got too many stations to cram into an already small space.


Your Correspondence





In this article:

I Left My Files In Xsan Francisco Apple Bump It Up: Spec Enhancements For eMacs & Powerbooks Mac OS 10.3.3 & mLAN

Apple news from NAB Show Apple Notes Published in SOS June 2004 Print article : Close window

Technique : Apple Notes

Apple had some interesting announcements to make at this year's US National Association of Broadcasters show, including improvements to the eMac and portable line-up. Mark Wherry

As Apple are continuing to focus on the desktop video market, it comes as no surprise that the company had a fairly substantial presence at this year's National Association of Broadcasters (NAB) show in Las Vegas. Following last year's announcements of Final Cut Pro 4, DVD Studio 2 and Shake 3, Apple again revised their professional video software line-up with Final Cut Pro HD, DVD Studio Pro 3, Shake 3.5 and the announcement of two new products: Motion and Xsan. Final Cut HD is the latest release of Apple's popular video-editing application, which now features support for HD video. While the new HD features look very interesting, it's worth pointing out that phrases such as 'HD over Firewire' are a bit misleading because Firewire doesn't actually have the bandwidth for HD video. For the most part, Apple are talking about the Anyone who wants to add real-time graphics consumer-orientated HD format, with effects to their Final Cut productions DVCPRO HD (or sometimes just could find that Apple's Motion, announced HDV), and it's this format that can be recently at NAB, is just what they're looking for — and something of a bargain at £199. transported over Firewire and handled natively (ie. with no extra hardware) by Final Cut Pro. To handle professional HD still requires extra hardware fitted to your Power Mac's PCI buss, such as AJA's Kona (www.aja.com), Pinnacle's Cinewave (www.pinnaclesys.com), or Black Magic Design's DeckLink (www.

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blackmagic-design.com). Final Cut Pro HD is available now for £699. Users of the first three versions can upgrade for £279, while users of Final Cut Pro 4 can download a free update from the Apple web site (www.apple.com/finalcutpro/ download). Motion is a brand-new application in Apple's professional video software line-up, providing an easy way to create real-time graphics for video productions. Whether it's animated text or special effects using particles such as smoke or fire you're hoping to add to your video masterpieces, Motion could be just the job. There's also support for Photoshop layers and After Effects plug-ins, and Motion will, of course, integrate with other Apple professional video products, such as Final Cut Pro and DVD Studio Pro. While Apple are slightly less specific about when Motion will be available, stating only "sometime this summer", the application will be aggressively priced at £199, presumably to compete with the likes of Discreet's Combustion and Adobe's After Effects — even though Motion doesn't support 3D animation. One of the new features in DVD Studio Pro 3 is the ability to encode audio in the DTS 5.1 format, in addition to the support for Dolby Digital and uncompressed stereo found in previous versions. While DVD Studio Pro won't produce DVDAudio disks, the application is still useful for users who want to produce a DVDVideo disk containing their surround mixes to share with others. DVD Studio 3 will arrive during May for £349; existing users of the first two versions can upgrade for £139.

I Left My Files In Xsan Francisco Xsan is also potentially interesting to the audio community — despite Apple's initial focus on the video market — especially in larger studios where there's a need for substantial shared storage between multiple audio workstations. For those unfamiliar with the technology, a SAN (Storage Area Network) can basically be thought of as a large, fast hard drive that's shared by multiple computers. A SAN differs from a regular LAN (Local Area Network) in two important ways: firstly, the available bandwidth is far greater, since SANs are based on a Fibre Channel infrastructure, which those who work in larger IT departments should be familiar with. This interface supports bandwidths of up to 200MB/s, and you can use multiple Fibre Channel connections to increase the theoretical amount of bandwidth available to each machine on the network. Secondly, a SAN uses protocols that are, in many ways, a development of SCSI, rather than being based on network filing systems and protocols, so attaching storage to your computer logically by way of a SAN is pretty much like having a real drive attached to your system. For audio and video work, SANs make a lot of sense. In the video world, if you



have multiple editors all working on the same video footage, it can be kept in one centralised location, which makes backing up easier, and it's easier to share data between users than it is to carry around Firewire ATA drives. In the audio world, connecting an audio workstation to a SAN is useful if you have people comp-ing sessions in an editing room after finishing recording in the main studio. In short, SANs make it easy to share data without creating unnecessary duplication, which makes for a better workflow where large audio and video files are concerned. Xsan makes use of an Xserve RAID for storage, which needs to connect to an approved Fibre Channel switch. That, in turn, connects to all the other systems on the SAN. Each of these systems needs to have Fibre Channel connectivity (Mac users can use Apple's own PCI Fibre Channel card), and Xsan supports multiple client platforms in addition to Mac OS X, such as Windows and Linux. Users will need to dedicate one Mac system on their SAN to be the so-called metadata controller, and this Mac needs to be either an Xserve or Xserve G5, a Power Mac G4 (with at least dual-800MHz G4 processors) or a PowerMac G5. It's not a system that's going to appeal to the modest home or project user, but that's not really its purpose. Where Xsan will find its home is medium-to-large audio and video production facilities where there's a big need for centralised storage.

Apple Bump It Up: Spec Enhancements For eMacs & Powerbooks In the absence of faster G5s, the iMac turned out to be the only G4based product line Apple didn't speed-bump this month, with revised eMacs being first out of the door during the middle of April. The eMac includes a 1.25GHz G4 Photos courtesy of Apple processor, 256MB DDR333 Apple's new Powerbook line-up includes 12SDRAM and ATI Radeon 9200 and 15-inch models with 1.33GHz G4 graphics with 32MB video processors, plus high-end 15- and 17-inch memory, supporting a 17-inch builtmodels offering 1.5GHz G4 processors. Now in CRT display with a resolution of that all models retail for less than £2000, the 1280x960 pixels. Also featured are Powerbook is better value than ever for Firewire 400 and five USB ports musicians on the move. (three of which support USB 2.0). A low-end eMac is available for £549 with a 40GB Ultra ATA/100 drive and a combo drive, and for £699 you can get an eMac with an 80GB drive and a Super Drive. Since these specifications are not too dissimilar to the previous range of Powerbooks, it seemed likely that Apple would upgrade this line as well — it would be a little strange to have a consumer/education-oriented product that was as or more capable than a professional portable, after all! And so it was: the new range of Powerbooks features a broad range of general improvements, including faster G4 processors, Airport Extreme as standard on every model, improved video hardware and larger hard drives. The 12-inch Powerbook now features a 1.33GHz G4 processor with 512k Level



2 cache, 256MB DDR333 SDRAM, a 60GB 4200rpm Ultra ATA/100 hard drive, Nvidia GeForceFX Go5200 graphics with 64MB video RAM, and the same connectivity as in previous models. This includes Firewire 400, two USB 2.0 ports, a Mini-DVI connector, 10/100 Ethernet, headphone and microphone ports, and a 56k modem. A 12-inch Powerbook with a combo drive is available for £1149 or with a Super Drive for £1299. The £1399 15-inch Powerbook features the same basic specifications as the 12inch in terms of processor, memory and storage, and a combo drive, but includes an ATI Mobility Radeon 9700 with 64MB video memory for its larger 1280x854 screen, plus a Firewire 800 port, a full-sized DVI connection, an SVideo output and Gigabit Ethernet. The £1749 model adds a 1.5GHz G4 processor with a 512k Level 2 cache, 512MB DDR333 SDRAM, an 80GB Ultra ATA/100 drive, and the now infamous backlit keyboard. And heading up the revamped Powerbook line is the venerable 17-inch model, which features the same specification as the high-end 15-inch model, but with a 1440x900 display instead for £1949. I have to confess that I'm tempted by the new 12-inch Powerbook specifications, for a cute mobile Logic rig, but the new iBook G4 specifications are equally interesting considering the price point. There is now just one 12-inch model, featuring a 1GHz G4 processor with a 512k Level 2 cache, 256MB DDR266 SDRAM, a 30GB Ultra ATA/100 drive, and ATI Mobility Radeon 9200 graphics with 32MB video memory, for £799. A 14-inch model is available for £899 with the same specifications as the 12-inch, save for a 40GB drive, while the £1049 14-inch iBook features a 1.2GHz G4 processor with a 512k Level 2 cache and a 60GB drive. All models feature a combo drive, with a Super Drive being available as a buildto-order option on the 14-inch iBook. In terms of connectivity, all iBooks are Airport Extreme-ready (with the high-end 14-inch model offering Airport Extreme built in) and feature a headphone port, Firewire 400, two USB 2.0 ports, a 56k modem, 10/100 Ethernet and a video output supporting VGA or S-Video.

Mac OS 10.3.3 & mLAN Just as I was finishing last month's column, Apple released Mac OS 10.3.3. Probably the biggest news to those reading this article is the improved mLAN support for the latest generation of devices supporting this interface. Of course, a version of Mac OS X that claims to really support mLAN is nothing new, and the new independent mlancentral.com web site claiming that "2004 will be the year mLAN breaks loose" seems much like previous claims Yamaha made for 2003, and 2002. You get the idea. Still, improved mLAN support in OS X is great news if you already own an mLAN-compatible product or are planning on purchasing an 01X. Firmware updates for first-generation mLAN products are available at mlancentral.com, to bring devices such as Presonus' now discontinued Firestation, Apogee's AD8000 and Kurzweil's KSP8 up to speed. Published in SOS June 2004







In this article:

Relinking Track Counts Quick Tips DSP, Plug-ins & Plug-in Settings Formats And Versions

Avoiding grief when moving PT Projects Pro Tools Notes Published in SOS June 2004 Print article : Close window

Technique : Pro Tools Notes

Current Versions 6.2.3 for HD, Accel, 002 and M Box on Mac OS 10.3.2 'Panther' only. 6.2r2 for HD on G5 Macs running OS 10.2.8 'Jaguar': if you're running HD on a G5 you are recommended to use this as the Panther release currently results in reduced track counts. 6.2 for Windows XP HD and Accel systems. 6.1.2 for Mac OS X LE systems on non-G5 machines. 6.1.1 for Windows XP LE systems. 6.1 for OS X & Windows XP Mix systems. All version 5 and Mac OS 9 releases remain unchanged.

A common occasion for grief in Pro Tools is when moving projects from one system to another. Here are some tips to help you keep your hair. Simon Price

One of Pro Tools's strong points is its scalability, and the fact that you can move projects around between different Pro Tools-based studios. It's appealing to know that you can take your PT LE/M Box track and open it up in a pro studio's DSP-packed super rig. Conversely, engineers can take stuff out of the studio on a laptop system. With a few bits of knowledge, moving around needn't be a hassle. Obviously, the degree of success when porting projects depends on the differences between the two locations, and in particular whether you are moving to a system with fewer capabilities. But let's start with the simplest situation: moving to another identical system. The things you need to make sure you take with you are the Session file itself, and all the associated media files. The only The most straightforward way to prepare a potential problem in this case is that Pro Tools project for transfer to another your project may use audio (and video) system is to use the Save Session Copy In... files from several different locations dialogue. and drives. If you are planning to move the Session on a single removable drive (such as a Firewire drive) you need to gather everything together. The simplest way achieve this is to use the File / Save Session Copy In command. If you tick the 'Copy audio files' option, a new Session folder will be created in your chosen location, containing a copy of the Session document and an Audio Files folder with all the audio used by the project. If you have a high track count, you

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may need to split the files off to different drives at the destination. In Pro Tools 6 this can be achieved using the Browser, as detailed in the next example. You may be working on removable drives from the outset, with portability in mind. In this case you don't really want to use Save Session Copy In, because this doubles up your audio files unnecessarily. However, you should check very carefully that you have all the media you need on the drives that you take with you. It's unbelievably common to take a project to a new destination, only to discover that some imported audio or 'Audiosuited' files have been left on the internal drive of the original system. There are two ways to check where all your files are located. The first is to select Show Disk Names in the Region List's submenu (at the right of the Edit Window). If you also choose Sorting by Disk Names you will very quickly be able to see if there are any regions attached to the project that aren't on the drives you'd expected. In PT6 you can do the same thing from the Project Browser: open the Audio Files folder and you'll get a list of files and their file paths. You can then use the full Workspace Browser to copy files to one of your removable drives by dragging and dropping.

Relinking After you've taken your drive(s) to the new location you may find that Pro Tools has lost track of where everything is (although this will never happen if you've used Save Session Copy In). Depending on whether you use version 5 or 6 of Pro Tools there are different procedures for re-linking the session with its media. In PT5 you are asked 'where is...' when the Session is opening. Choose to search Sorting by Disk Names in the Pro Tools by name on all drives and matching region list provides an easy way to see files will be listed. Pro Tools will use its whether your Session incorporates any stray system of unique file IDs to point out files that are on the 'wrong' disk. the correct file with a small arrow. Double-click this file to accept. In Pro Tools 6 you can choose automated re-linking, or opt to do it manually. Selecting the manual route opens the re-link window, which lists all lost files, and provides similar options to PT5 for finding them.

Track Counts Not all Pro Tools system are made equal, and the first hurdle you may encounter is a difference in the maximum number of tracks available from one system to the next. In particular, you may be moving a project to LE and have more than the



permitted 32 tracks, but there are also differences between TDM systems (for instance, HD systems provide 128 tracks, and Mix setups 'only' 64). However, you may still be able to work, as the destination system will open the tracks, even though it can't play them all back. You may not need all the tracks if you are, say, just taking the Session out to do some recording. You may also be able to reduce your track count by copying audio and automation from one track to another, so that parts that don't overlap 'share' a track. Another option is to bounce some groups of finished tracks down to stereo stems, and reduce track count that way.

Quick Tips One way to transfer data between computers if at least one of them is a Mac is to connect them via a Firewire cable and boot the Mac in Target disk mode. If you are going between a Mac and PC, the PC will need Mac Opener installed. To boot a Mac in Target mode, restart it with the 'T' key held down. The Mac will boot with just a big Firewire symbol on the screen, and its drives will be available to the other computer. If you send a Mac Pro Tools project or audio files via email, or otherwise via a PCbased Internet host, you will need to archive the files or folders first using Stuffit or similar. Otherwise the files will get stripped of their 'resource fork' and be rendered useless.

DSP, Plug-ins & Plug-in Settings Plug-ins present problems on at least two fronts: the destination system may not have the same plug-ins, and/or it might not have enough DSP power to handle the Session. Depending on the situation, you can approach this in different ways. If you are only moving location temporarily, you can try to live without the plug-ins for a while. The inserts and settings will not be lost, they will just be greyed-out and The Project Browser is a new addition in Pro inactive until you return to the original Tools 6. system. If it's a permanent move, you will either need to bounce the tracks (record the processed audio), or switch to using a different plug-in that is available. When moving in either direction between LE and TDM systems that have the same plug-ins, the Session will automatically switch between RTAS and TDM versions. DSP resource inequality also causes tricky problems. Even when your destination studio has the same plug-ins, it may have fewer DSP cards, or in the case of LE, a slower computer. Again, bouncing may be necessary to print the



effects, or you could switch to less intensive plug-ins. There's a neat trick when moving between TDM systems, which is to use both TDM and RTAS formats. For example, when moving to a TDM system with less DSP, you may well encounter the dreaded 'could not create plug-in x due to insufficient DSP...' message. You can often get around this by switching some plug-ins to RTAS versions, squeezing some extra power from the host computer. To switch a plugin's format, open it and click where it says 'TDM' (above the target icon). A small pop-up window will appear with the option to switch to RTAS. Any time you free up DSP, you can reactivate greyed-out plug-ins by command-option-clicking them (Control-Start-click on Windows versions). Plug-in settings are not really a big issue. For a start, all current settings for plug-ins will be maintained, as they are stored as part of the Session data. However, you may have created some library settings during the project (or previously) that you want to take with you. Usually, when you save plug-in When you first open a Pro Tools Session settings they are kept in the 'root' Settings folder, although it's possible to that has been transferred to a different machine, you may need to re-link its files. choose to save them in the Session folder. The easiest way to take these with you is to use the good old Save Session Copy In command and tick the plugin settings options (Session and 'root'). This is the quickest method even if you don't really need another copy of the Session, as it avoids ferreting around in the OS to find the Settings folder!

Formats And Versions Save Session Copy In... is also the key to handling most data format issues. The first example of this is different versions of the Pro Tools application. Pro Tools version 5.1+ and v6 files are interchangeable, which has made this much easier. Earlier versions of PT5 cannot handle mixed audio file types, so you may need to use Save Session Copy In to convert the audio files to one type. Projects that use high sample rates (greater than 48kHz) will need converting for compatibility with pre-HD systems. Should you need to drop to 16-bit audio for some reason, Save Session Copy In can handle this too, and will apply dither automatically. The final issue to look at is compatibility between Pro Tools on Mac and PC computers. You've probably noticed that when you create a new Session there is a tick box for 'Enforce Mac/PC compatibility', and if you use this you're already ahead on this score. If you've not used this it's possible to use the good old Save Session Copy In... command and tick the box at this stage. Compatible Sessions use WAV or AIF-format audio files, and all files have three-character file extensions, including '.PTS' for the Pro Tools Session document. You also have to make sure there are no 'weird' characters in the file names. Aside from file



compatibility, there is the issue of transferring the data between the two platforms, which arises because Mac and PC drives use different filing system formats. If the Session isn't too huge, sometimes the quickest and easiest method of transferring is to burn the data onto CD-ROMs, selecting the PC/Mac hybrid CD format. Otherwise, you can mount Mac hard drives on a PC using the Mac Opener utility. There is a demo of Mac Opener on the Windows Pro Tools installer CD that will get you going.

You can switch an open plug-in between RTAS and TDM formats, without losing settings, by clicking above the target icon.




Beat-slicing Masterclass

In this article:


Getting In The Loop What software you need and how best to do it! Slicing The Beat Integrating REX 2 Files With Published in SOS June 2004 Your Sequencer Print article : Close window Beat Slicing Within Your Technique : Sequencing & MIDI Controllers Sequencer Beyond The Basics Beat Slicing Versus Warping It's Not All About Tempo Adventures In Sound Design Beat slicing can radically expand the creative Useful Software potential in your loop library — you can match

tempos and key signatures, rearrange loop events, and delve into inspirational sound design. We look at all the leading beat-slicing software and show you how to get the best from this powerful technique within your sequencer. Simon Price

Sampled loops are part of the fabric of modern music production. For a long time they were associated with hardware samplers, but now they're everywhere: in REX players such as Propellerhead Reason, Emagic Logic, and Steinberg Cubase; in specialised plug-ins such as Bitshift Audio Phatmatik Pro and Native Instruments Intakt; and in audio sequencers such as Ableton Live, Sonic Foundry Acid, and Apple Garage Band. While loops were once predominantly dance and hip-hop tools, they are now used routinely in the production of most genres of music. And loops can also make things rather easy. This can be cool: you sit down with Garage Band or Reason and throw together a bunch of library loops until you have a song. This is fantastic for kids, or if you enjoy putting together songs as a hobby without having the spare time to learn all the dorky synthesis stuff. It can also be a starting point to inspire something else: you set a few loops going and you hear a new groove that you can start working against. Sometimes, after a while, you can take out the original loops altogether and the construction stands up on its own, like removing the framework from an arch when all the bricks are in place. But another option is to take loops and transform them into something file:///H|/SOS%2004-06/Beat-slicing%20Masterclass.htm (1 of 12)9/22/2005 7:46:17 PM


that's your own, and that's what we're interested in here.

Getting In The Loop Obviously, nearly all music is cyclical or loop based, but the idea of looping short pieces of recorded audio to create a new song really took hold with turntablism, hip-hop and electro. OK, actually a big nod has to go to pioneers like American composer Steve Reich, who were constructing pieces from tape loops in the '60s. But really the pioneers of modern looping were skilled DJs. Samplers and sequencers made triggering a short loop much easier, and meant that you could start layering them up by pitching samples up and down in order to match tempos. Samplers later began to get time-stretching abilities, so you could alter the tempo without changing the pitch of the sample. But time-stretching could only be used over small tempo ranges without causing serious degradation of the audio quality. Digidesign's early Mac-based audio workstation Sound Designer moved things along by enabling you to manipulate loops on screen and dump them back into the sampler. Of course with Sound Designer came the hint that the future of looping lay inside the computer with your MIDI tracks, rather than out there in a beige box. Anyway (drum roll), 1994 saw the launch of a product that began the new era of looping technique: Recycle. Although originally packaged as a Steinberg product, this was in fact the first outing for our favourite Swedes: Propellerhead Software. Recycle is a beat slicer, originally designed to work with hardware samplers. Its main purpose is to take a drum/percussion loop sample, and automatically chop it into sections using transient detection. Each 'slice' is loaded into a sample playback device of some kind and assigned a MIDI note number. A MIDI file is generated that plays through the loop one slice at a time with the original order and timing. Now your drum loop will run at any If you've just got a simple drum loop to slice tempo you set in the sequencer. No up, then Propellerhead Recycle's Sens control may be all you need to know about. audio has to be stretched because all Just drag it to the right until slice marks you're doing is changing the intervals appear at each individual drum hit. between individual hits being triggered. In addition to freeing the loop up from the tempo it was recorded at, you can also mess with the sequence, triggering the slices in any order you like, or removing parts altogether. Having individual slices opens up many other possibilities. Today, Recycle is still going strong but no longer supports communication with hardware samplers. Instead, Recycle v2 files (REX 2 format, using the 'rx2' PC file extension) can be played back directly by many software studios, including Cubase, Logic and

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Reason. The idea of beat slicing has also been adopted by a number of other software products. Native Instruments Intakt, Bitshift Audio Phatmatik Pro, and GForce Beat Burner (among others) all have Recycle as their common ancestor. Most of the big sequencers also have functions for quickly slicing up loops, without relying on Recycle. Plus, beat slicing has been joined by a newer breed of looping tools based on granulation or granular resynthesis technology, which is outside the scope of this article — but see the 'Beat Slicing Versus Warping' box for the main pros and cons of each approach.

Slicing The Beat Because beat slicing involves cutting a continuous audio file up and re-spacing the chunks to sync to a new tempo, it follows that only certain types of material are suitable. Drums, percussion, and other inherently transient and 'gappy' audio sources are the main targets. However, many monophonic musical loops can be treated with a high degree of success. There are ways to get around the limitations, especially if you don't mind altering the loop a little, but we'll come back to those. Before we get carried away, let's have a look at how you go about the process of beat slicing, which should help make the whole idea a little clearer. For this example I'll use Recycle to chop up a one-bar drum loop and create a REX 2 file that will then be opened up in Reason — Reason, Cubase and Logic all support REX 2 loops without any additional software. Alternative products like Phatmatik and Intakt, are generally operated as plugins within a host sequencer, with the plug-in handling both the editing and playback tasks. Although we're looking at Recycle, the moves are similar in other software — for example, when using Cubase's Hitpoint system. Prepare & Import The Sample: None of the beat-slicing programs are really audio editors in their own right, so they'll need a file that's already edited up to a point. You'll be able to set in and out points, but if the loop you want is part of a longer sample it makes sense to first trim it to size in your main audio program. When you're happy, export or bounce the sample as a single file (stereo interleaved if it's a stereo sample). Wave or AIFF formats



will do the trick for any of the slicing applications. In Recycle, choose Open from the File menu and navigate to your sample. Your sample's waveform will be displayed, and the left and right loopboundary markers will be set automatically to the start and end of the file. Set Up For Beat Detection: A good first step is to Normalise the sample (choosing the command from the Process menu), which will adjust the gain to use the full possible range. As well as giving you plenty of level in the finished loop, this makes the individual 'hits' in the audio more obvious. When you've normalised, enter the length of the loop into the Bars field in the top centre of the screen. The tempo will be calculated and displayed automatically based on the length of the sample in seconds. Detect The Beats: Recycle (like its equivalents) uses an automatic transientdetection system to save you much of the work of chopping the sample into individual hits. However, your input is needed to adjust the detection sensitivity until the program correctly discriminates between relevant peaks and background noise. This is easy for drum loops, unless there is something loud and continuous interrupting the quiet bits between the hits. In the example shown in the screenshot there was no problem: I just dragged the Sens slider to the right until all the correct markers obediently appeared. Preview Different Tempos: Click the Preview Play button (which in Recycle is to the left of the transport's Stop button) and have a listen to the results. Adjust the tempo control and listen to what happens at significantly higher and lower speeds. Listening at a low tempo will reveal any problems, such as undetected beats or slices that cut off abruptly. For now I'm assuming that everything sounds fine... Save The File: Before saving, put the tempo dial at a speed of your choice, because this information will be saved by Recycle within the file as its default. Saving leaves the original sample untouched and creates a new REX 2 file. If, however, you've actually used an existing library REX file instead of raw audio, it will update the file. It is possible at this stage to export the file as unsliced audio at a new tempo (where you left the tempo dial). This is useful if you don't have anything that can play back REX 2 files. Test The File: For this demonstration, I made a fresh song in Reason and file:///H|/SOS%2004-06/Beat-slicing%20Masterclass.htm (4 of 12)9/22/2005 7:46:17 PM


created a Dr:Rex module, the Reason instrument that exists solely for playing back and manipulating REX files. When opened from the Dr:Rex's file browser, the sample appears in the central display, along with the slice markers. The small preview button plays back the loop using the current tempo of the song. If Reason is playing back, the loop will trigger in sync. Phatmatik and Intakt have exactly the same operation in this regard. The REX player will stay locked to the tempo of the song, and provides a host of sound-manipulation options for messing with both the whole loop and the individual slices. Export A MIDI File: Each of the loop slices is now mapped to a MIDI note in the Dr:Rex's sequencer track, and you should be able to play them back individually from a MIDI keyboard. To play back the loop properly (without using Preview mode) you need a string of MIDI notes that will trigger all the slices and recreate the loop. The Dr:Rex module can handle this automatically by pressing its To Track button, which writes the necessary MIDI within the current locator positions in the sequencer. Once written, this data can be moved around to arrange where the loop is played — and can also be used to do a whole lot more, as you have probably guessed.

Integrating REX 2 Files With Your Sequencer Reason is not the only software studio to make elegant use of sliced loop files. Some programs provide an alternative method to that of Dr:Rex, supporting import of REX files directly into audio tracks, while keeping the benefits of slicing. For example, let's take a look at how Cubase handles things. Cubase allow you to import REX files like normal audio files. The file appears in the audio track as a single block, as you can see in Screen 1a, but the slice information remains — Cubase calls the breaks between the slices Hitpoints. The slices slide together or apart so that the whole loop automatically stays in tempo with the song. The loop can be arranged and looped as a block, like any other audio file. Double-clicking the loop will open the Part Editor, giving access to many slice-based tools. Each slice can now be moved around, and even groove or grid quantised. While in the Part Editor you can trim slices, adjust their gain, and draw fade-ins/ outs, as shown in Screen 1b. By choosing Shuffle Edit (Screen 1c), you can quickly rearrange the slices to create new patterns, variations, and



fills. Dragging slices down to a new row, as in Screen 1d, reveals overlapping regions. The loop shown in the screenshots was originally 102bpm, and the overlaps show how the slices have been squashed together to fit the Cubase Song's 120bpm tempo. When returning to the Arrange window, the loop still appears neatly as a single block. Not all sequencing platforms have this level of integration, however, so there are many situations where you need to use MIDI tracks to play back sliced loops. Staying with Recycle files for the moment, most current software samplers can read REX 2 files, so they can take the place of the Dr:Rex unit in the first example. Some samplers, like Phatmatik and Intakt, can generate the necessary MIDI file to trigger playback. For other samplers you need to use Recycle to do this, using the Export option in the File menu. In the following example, I'm going to look at importing a REX file into Intakt. First create an Intakt instrument in your sequencer and load in a REX 2 file, as shown in Screen 2a. Like any sampler that supports the format, the beat slices are recognised. Intakt (like Phatmatik) has a couple of advantages, because it can create slices on its own and can generate MIDI files to play the loop without needing Recycle. Once you've decided which slices to map to which MIDI notes (Screen 2b), you can generate a standard MIDI file which you can then import into your sequencer of choice for limitless rearrangement — you can see how the MIDI file I generated looks within Cubase in Screen 2c.

Beat Slicing Within Your Sequencer Although dedicated beat-slicing applications provide the quickest way to chop up your loops, that doesn't mean that you can't achieve similar results using just your sequencer. All of the regular SOS sequencer columns have now featured information on beat slicing, so rather than repeating things, here's where to find each specific article. MOTU Digital Performer

The new Beat Detection Engine is touched on in Performer Notes in SOS April 2004. www.soundonsound.com/sos/apr04/articles/performernotes.htm Emagic Logic

Logic can import and use REX files, and provides a Recycle File Import window for determining how to deal with them. Slices are grouped together into a folder in the Arrange window, with crossfades automatically applied to smooth transitions. Although you can't do beat slicing in a fashion that's directly comparable to that of Recycle or Cubase, similar results can be achieved by



using Strip Silence to chop audio into individual hits. The resulting regions are tied to their bar/beat positions, so will maintain their musical relationships if the tempo is changed. www.soundonsound.com/sos/feb04/articles/logicnotes.htm www.soundonsound.com/sos/may04/articles/logicnotes.htm Steinberg Cubase

Cubase can import and manipulate REX files, but it also has sophisticated functionality for slicing up any audio loop. Automatic and manual transient detection can be used to generate Hitpoints, after which the audio clip will behave as already detailed in this article. www.soundonsound.com/sos/sep02/articles/cubasenotes0902.asp www.soundonsound.com/sos/oct02/articles/cubasenotes1002.asp Cakewalk Sonar

Although Sonar can use Acid loops (using time-stretching rather than slicing), it doesn't support REX files or have any beat-slicing functionality. However, you can still chop beats up manually to accommodate tempo manipulation, and there's a workaround for bringing in REX loops if you have Recycle. www.soundonsound.com/sos/jul02/articles/sonarnotes0702.asp www.soundonsound.com/sos/may04/articles/sonarnotes.htm Digidesign Pro Tools:

Pro Tools does not support REX files, but the TDM version of the software does feature the Beat Detective tool. Beat Detective includes functions for automatically detecting and separating beats, and altering tempo and timing. www.soundonsound.com/sos/aug03/articles/protoolsnotes.htm

Beyond The Basics The basic process of creating and using sliced-up loops is fairly straightforward when you understand how it works, but there are times when things don't go as smoothly as in my examples so far. For example, although the automatic slicedetection system in Recycle can often work out all the boundaries between slices when you're working with clean drum and percussion loops, in many other cases you need to lend it a bit of a helping hand. Depending on the nature of the material, you may get markers appearing where they shouldn't, while hits that need separating are overlooked. Sometimes it's best to set the sensitivity low so that the obvious hits are detected, and then to add the rest in manually. If there are only a few 'false positives' you can set the sensitivity higher and delete what you don't need. A good technique is to try listening to the loop at a low tempo, which should reveal slices that contain more than one hit and need further division. All the beat slicers also provide the option of auditioning each slice individually for further scrutiny.



Another problem that can arise is that, especially where there are sounds with slower attacks, you can get late beat markers causing clicks at the ends of slices. These can be moved forward slightly so that the click becomes the start of the next slice. Phatmatik always seems to put its markers a little early, which may be to stop this problem occurring. If you're getting clicks during playback, or if

More common causes of jerky there are obvious gaps between slices at slow tempos, then you can use Bitshift Audio playback are sounds with long decays Phatmatik Pro's amplitude envelopes to that hang over into the next slice. For improve the situation by smoothly rounding example, a long cymbal decay might off the end of each slice. still be ringing when the next hit occurs. You must have a new slice to keep the second hit in time, but now the decay on the cymbal is getting broken into two or more parts. This has a greater effect when the loop is slowed down, because you hear a gap in the middle of the decay. In fact, whenever there is sound lasting to the end of a slice there is potential for an obvious gap when you slow the loop down: this is the drawback of beat slicing as a method. However, this doesn't mean that you have to give up, because there are some compromises you can make. The first is to use volume envelopes on the slices to reduce the decay of each one. In Recycle this is as simple as just pulling down the Decay control. Other devices have more complicated envelope generators and can act differently on each slice. The other tool available is the 'end stretch'. All the beat slicers we've been looking at have methods of padding out the end of each slice by looping some portion from the end of the slice. Recycle's Stretch command artificially extends the tail of each slice to fill any audible gaps that appear after slowing the loop down. This is a key trick for extending the use of beat slicers into less obviously rhythmic material. With careful editing it's possible to get more continuous audio, like pulsing pads and textures, to shift tempo.



Beat Slicing Versus Warping Ableton Live, Sonic Foundry Acid, Apple Garage Band, and the Native Instruments range of samplers are among the software products that use an alternative approach to altering loop tempos, known as granulation. These programs can independently adjust the tempo and pitch of an audio file without chopping the file into large chunks. The advantage of this approach over beat slicing is that you can often apply it transparently with continuous sounds, such as vocals and pads. However, depending on the nature of the material, and the amount that it's being 'warped', there can be effects on the sound quality. These audible artefacts are generally less serious than with less sophisticated timestretching, and decent software will have various parameters that can be tweaked to get the best results. Granulation-based time warping is fast, needs no pre-processing by the user, is versatile and seamless for manipulating timing, and can be used in a track-based context like regular audio. However, it lacks the beat slicer's ability to trigger and manipulate individual components of the loop. Also, warping is a highly complex DSP task, so using this method is much harder on your computer.

It's Not All About Tempo So far we've looked at the most common applications of beat slicing, namely tempo shifting and re-sequencing of loops. Of course the whole point of slicing when changing tempo is to avoid changing the pitch of the audio. Equally, beat slicing allows you to process the pitch of some or all of the audio without changing the tempo. This can be useful with drum loops, where you might want to re-pitch some of the more ringing sounds to sit with the rest of your track. With musical loops, you can even pitch individual notes to create a new melody or to match the key of your other parts. Another application is to 'pick and mix' particular components of a loop. For example, you might have a loop that contains a snare sound and pattern that you like, but you don't want the rest. Once the loop is sliced up it's usually fairly easy to just discard the slices that you don't want. Or maybe you want to treat some sections with a particular effect. This is possible too by opening up two copies of the loop in different tracks and triggering different slices in each track. I've already mentioned that slicing makes it possible to quantise the hits file:///H|/SOS%2004-06/Beat-slicing%20Masterclass.htm (9 of 12)9/22/2005 7:46:17 PM

Another solution to the problem of gaps between slices at low tempos is offered by Propellerhead Recycle — sections at the end of each slice can be looped to fill in the gaps, and you can adjust this effect using the Stretch control.


within a loop. A particularly subtle yet powerful trick is to use groove quantising with loops. Say you want to use two recorded loops in your track, but slight variations in the performances mean that they don't quite seem to gel or lock up nicely. With them both sliced, and triggered from MIDI tracks, there's a clear solution. Select the MIDI notes from one loop and then use your sequencer's groove-extraction functionality — most of the major sequencers now have one — to groove quantise the MIDI notes that are triggering the other loop. This simple trick alone adds huge usability value to your loop collection.

Adventures In Sound Design OK, so using programs such as Recycle can solve a lot of production problems. Don't get me wrong, this stuff is essential and is being used every day in studios by programmers whose job it is to make producers look clever. But of course the first thing you really want to do with tools like this is see how much you can completely mangle stuff up and stumble upon something totally unexpected. By breaking up audio into rhythmic components, and then providing you with envelopes, filters, LFOs, and bit crushers, these beat slicers let you do really synth-like stuff. A favourite trick of mine starts by grabbing a rich pad sound or other atmospheric 'texture'-type sound, chucking it into Recycle and making up a bar length value. Both Recycle and Phatmatik have quick ways of adding slice markers on, say, the 16th-note grid. The slice markers trigger your envelopes, so add some attack and decay to the amplitude or filters and you've instantly got a new pulsing gated pad sound. Finally, beat slicing provides a link between the studio and live worlds. If you are producing music that you think you might take out and perform in some way, it's worth working like this from the outset. By having your rhythm tracks as sliced samples triggered by MIDI, instead of as continuous recorded audio, you are leaving yourself options. At the basic level, if you use these elements as backing tracks you are not tied down by tempo or linear transport like you are with tape or continuous tracks. By working with loops you also have real-time control over rearrangement, and can easily fly bits of different songs in and generally mess about with stuff. If you play completely live with a band, using beat-sliced audio tracks in the studio can open up production possibilities that would have been difficult to recreate on the stage. Say you used a distinctive-sounding drum loop as the basis of a song, and you sliced it up and rearranged it in different ways throughout the track. When you play live you can just load the REX file into a sampler and trigger the individual slices from drum triggers. It's the best of both worlds: it's still a genuine performance, but you're getting the same sounds as you used in the studio.



Useful Software Propellerhead Recycle

The program that started it all, Recycle creates the widely recognised REX 2 file format. Originally reviewed all the way back in SOS May 1995, it has recently been rejuvenated, with the Mac version making it to OS X. The new version is the first to support 24-bit files, with the REX Shared Library being released a few weeks ago to support playback from supporting third-party applications and plug-ins. This year Propellerhead opened up the REX 2 file format so that any manufacturer can add REX 2 playback and manipulation into their products. This is a very popular sample library format, as the inherent versatility means you can reuse the loops over and over. You know, like, recycle them... www.propellerhead.se www.soundonsound.com/ sos/1995_articles/may95/ steinbergrecycle.html Propellerhead Reason

This sequencing environment, reviewed in its v2.5 incarnation in SOS December 2003, includes a REX 2 file playback device called Dr:Rex as one of its built-in instruments. www.propellerhead.se www.soundonsound.com/sos/ dec03/articles/propellerhead.htm

Some less well-known beat-slicing applications (top to bottom): Concrete FX Dicer, GForce Beat Burner, and Basement Arts Reflex.

Bitshift Audio Phatmatik Pro

Until Intakt, Phatmatik Pro was the only big contender for the beat-slicing crown. Reviewed in SOS September 2002, it performs the same functions as Recycle, but as a plug-in rather than as a stand-alone application. The plug-in generally acts as the playback device, but can also export loops in a proprietary format for use in other supporting devices. Phatmatik has a great range of modulation and effect possibilities, making it one of the most fun slicers to use, and it's great for mashing up and changing your loops.



www.bitshiftaudio.com www.soundonsound.com/sos/sep02/articles/phatmatik.asp Native Instruments Intakt

Intakt takes technology available in NI's top-flight sampler Kontakt and packages it into a streamlined looping tool. Functionally it is quite similar to Phatmatik, with a few extras. For a start it can create tempo-shifted loops using granular resynthesis as well as beat slicing. In addition to the usual keyboard mapping, complete (but sliced) loops can be assigned to single keys for simple triggering and layering. For full details, check out the review in this issue. www.nativeinstruments.de Concrete FX Dicer

Dicer is a simple but powerful PC-only VST instrument plug-in. There are lots of modulation options, which can be assigned to individual slices as well as globally. There's also automatic beat detection, tempo-synced preview, MIDI export, MIDI controller learning, randomiser, and more for only 48 Euros (around £35). www.concretefx.com GForce Beat Burner

Another new and exciting young upstart in the beat-slicing arena, the PC-only Beat Burner has some innovative features that promise to be useful for a wide range of material in addition to drum loops. Of particular note are the 'wave shaper', the click-and-drag definable envelopes, and the ability to morph between presets as in Reaktor. www.gmediamusic.com Basement Arts Reflex

This is another PC-only VST instrument that also comes as a free LE version. It has all the stuff you'd expect, plus a slice-position editor, anti-click 'micro envelope', high sample-rate support, and tempo-sync'ed envelopes. www.basementarts.de Published in SOS June 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



CLASSIC TRACKS: 'Wuthering Heights'

In this article:

Artist Development Starting At The Top Virgin Territory Mr. 87 Fader Movement Separation Anxiety Scaling The Heights

CLASSIC TRACKS: 'Wuthering Heights' Artist: Kate Bush; Producer: Andrew Powell; Engineer: Jon Kelly Published in SOS June 2004 Print article : Close window

Technique : Recording/Mixing

Kate Bush's smash hit debut single was also the first major project Jon Kelly had recorded. It proved to be a dream start for both artist and engineer, and a perfect illustration of the benefits of working with talented session musicians. Richard Buskin

The 1977 sessions for The Kick Inside marked the debut not only of a new artist named Kathy Bush, but also of Jon Kelly as a fully fledged engineer. He had spent the previous couple of years as a tape-op and assistant engineer at the original AIR Studios facility on central London's Oxford Street, during which time he'd assisted the already-legendary Geoff Emerick on recordings by Gino Vannelli, Robin Trower and Gallagher & Lyle. He also "did as many jingles as I could, because I knew that would teach me to be quick", and had worked on several smaller projects with producer Andrew Powell before the two of them joined forces for The Kick Inside. Thereafter, Kelly would engineer Kate Bush's second album Lionheart, co-produce Never For Ever with her, go on to produce and/or engineer for the likes of Paul McCartney, Chris Rea, Tori Amos, the Damned, Deacon Blue, New Model Army and Prefab Sprout, and form a notable relationship with the Beautiful South which has so far yielded nine albums.

Artist Development A beautifully tender yet haunting musical setting of Emily Bronte's classic love

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story, 'Wuthering Heights' wrapped swelling keyboards, strings and guitars around a lead vocal delivered in a sustained, almost child-like soprano by the song's 18-year-old composer, Kate Bush. Mentored by Pink Floyd lead guitarist Dave Gilmour, Bush had been signed to EMI at the age of 16 on the strength of 'The Man With The Child In His Eyes' and 'Berlin' (later retitled 'The Saxophone Song') — both recordings, engineered during an 'artist test' by Geoff Emerick, would be included on her first album, The Kick Inside. Thereafter, she had been allowed to Photo: BBC / Redferns study dance, mime and voice while Kate Bush performs 'Wuthering Heights' on developing her self-evident keyboard Top Of The Pops in February 1978. and writing talents, and by early 1977 she'd penned 'Wuthering Heights' and numerous other numbers and was ready to enter the studio to record The Kick Inside. This was achieved with producer Andrew Powell and engineer Jon Kelly behind the 24-channel Neve console in AIR's spacious Studio Two, which also housed Tannoy monitors and a 3M M79 two-inch 24-track tape machine, as well as the similar equipment in Studio One, whose vast live area was utilised for the string sessions. "As a Geoff Emerick protegé, my early miking choices basically mirrored his," says Jon Kelly. "For instance, on drums he loved the Coles 4038s for overheads, as I still do now, and at that time his snare mic was an AKG D19 — he liked that punchy dynamic on the snare, and the D19 provided that kind of definition while the overheads captured most of the size. His tom mics varied between D19s, Sennheisers and Shures; and bass drum mics were usually D12s, D20s and sometimes a [Neumann] FET 47. I can't remember ever putting up any ambients or room mics with Geoff, because during the mid-'70s everything was pretty dry. It was always that Westlake/Eastlake sound, with people taping up cymbals so they didn't ring too much. "Geoff took immense care positioning the mics. He used to say 'The microphone is like a camera lens. Imagine it's taking a picture.' Having assisted a number of engineers at AIR, the difference I noticed with Geoff was that he always used the cheapest dynamic mics on the drums, whereas others like Bill Price used things such as KM86s on the snare, 84s and 87s as overheads — much more classy condenser microphones. Geoff would use the old dynamics and then bring the sound out with EQ."

Starting At The Top All of this served as Kelly's starting point for the very first Kate Bush session, file:///H|/SOS%2004-06/CLASSIC%20TRACKS%20%20%27Wuthering%20Heights%27.htm (2 of 8)9/22/2005 7:46:28 PM


during which he was "learning as I went along and dreadfully insecure. I give full credit to Andrew [Powell] and the great musicians, who were very supportive, while Kate herself was just fantastic. Looking back, she was incredible and such an inspiration, even though when she first walked in I probably thought she was just another new artist. Her openness, her enthusiasm, her obvious talent — I remember finishing that first day, having recording two or three backing tracks, and thinking 'My God, that's it. I've peaked!'" The live rhythm section that Jon Kelly recorded for 'Wuthering Heights' consisted of Kate Bush playing a Bösendorfer grand piano, Stuart Elliott on drums, Andrew Powell on bass and Ian Bairnson on a six-string acoustic. And in terms of the miking, Kelly adhered pretty closely to Geoff Emerick's favoured choices while adding some of his own. "For the drums I used a D19 on the snare, Sennheiser 421s on the toms, a D12 on the bass drum and a [Neumann] KM84 on the hi-hat," Kelly Photo: Vicky Brown recalls. "The bass was DI'd and amped Jon Kelly today. — at the time I was very keen on the Susan Blue DI box, while a Marshall cabinet and Marshall head were miked with an FET 47. Ian Bairnson's acoustic was recorded with a Neumann U87, as were Kate's piano and vocal — I was a big 87 fan, I used to use them on everything. I still think it's a really under-rated microphone. When people listen to one on its own they often think it's a bit hard and doesn't have such a huge sound as some of the valve or softer-focus mics, but it's so efficient once you place it within the mix. "Kate always recorded live vocals, and they were fantastic, but then she'd want to redo them later. In the case of 'Wuthering Heights', she was imitating this witch, the mad lady from the Yorkshire Moors, and she was very theatrical about it. She was such a mesmerising performer — she threw her heart and soul into everything she did — that it was difficult to ever fault her or say 'You could do better.'"

Virgin Territory David Paton, who was the bass man on the other songs, overdubbed 12-string acoustic guitar on 'Wuthering Heights', and after Ian Bairnson redid his six-string part, Jon Kelly double-tracked them and tweaked the Varispeed on the machine to provide some breadth together with a chorusy feel. Then Andrew Powell hired a celeste and played the chime-like arpeggios that double with the piano motif file:///H|/SOS%2004-06/CLASSIC%20TRACKS%20%20%27Wuthering%20Heights%27.htm (3 of 8)9/22/2005 7:46:28 PM


during the song's intro and the sections preceding the chorus... all of which was virgin territory for the fledgling recording engineer. "There was a fair bit of fun involved in working with instruments like that," Kelly recalls. "Kate would certainly get involved, poking her head all around to see where it sounded nice. There was a good feeling of camaraderie, so I never felt nervous... just insecure! I recorded the celeste with a Coles ribbon mic positioned on the soundboard at the back, and that worked out fine. "You couldn't keep Kate away from he sessions even if you had wild dogs and bazookas. She was just drinking it all up, learning everything that went on. The first moment she walked into the control room, I could tell that's where she wanted to be, in control of her own records. She was so astute and intelligent, and she was also phenomenally easy to work with. An absolute joy. I can't remember any bad moments at all." Next to overdub some parts was percussionist Morris Pert, who spent an entire day working on songs for The Kick Inside. "The only things he played on 'Wuthering Heights' were crotals, which are like disc-shaped glockenspiels," Kelly explains. "Again, these were doubled with the piano motif throughout the song."

Mr. 87 Then came the strings recorded in AIR's Studio One — eight first violins, six second violins, six violas and six cellos — as well as three French horns. These comprised the section that was used on 'Wuthering Heights', whereas a smaller section was used for some of the other songs — the parts for a couple of numbers were recorded in each three-hour session. "That was a huge room, twice as big as the live area in Studio Two," Kelly remembers. "It could accommodate between 60 to 70 musicians, and had high ceilings and a lovely, bright sound. Everything sounded great in there. I miked the first violins with a couple of 87s, as I did for the second violins, the violas, the French horns and as overheads — back then you could have called me Mr. 87. At least there were FET 47s on the cellos. I'd try to use as few mics as posssible in Studio One because the room sounded so good and there was this phase thing going on — the more mics you used, you could fool yourself into thinking it sounded better, but things would cancel one another out and you'd lose the vibrancy.



"Nothing was slaved, everything was kept 24track on this album, and that was fortunate because slaving was a really laborious process in those days — before Q-lock enabled us to efficiently run two machines together, we'd have to physically get two tapes in the right position to start a song. Tracks one through five were hi-hat, bass drum, overhead left, overhead right and snare — hi-hat would always be the first casualty if we needed an extra track — and tracks seven and eight were the tom-toms. Track six was missed out because you couldn't pan between odd and even on the Neve desks in AIR, while some of the groups had faders on them and some weren't normalised. You had to be careful about getting groups caught between the two, because there were cancellation problems. Meanwhile, the strings were mixed to two tracks and the French horns went to just one track."

The layout of AIR Studio Two for the recording of the 'Wuthering Heights' band.

Ian Bairnson's electric guitar solo, which winds its way through the closing stages of 'Wuthering Heights', was played in the Studio Two control room, his Les Paul going through a Marshall head and Marshall 4 x 12, miked with... yes, a pair of 87s, one close, the other about four feet away. "Ian warmed up and developed that solo while I got the monitoring right, and there was one take that was just great," says Kelly. "Being in the control room, he missed the feedback from the amp, and I can remember telling him to get close to the speakers, expecting this to do the same. You can tell I was pretty naïve..."

The string section was recorded in the larger AIR Studio One.

Kate Bush, meanwhile, re-recorded her 'Wuthering Heights' vocal late one night, miked with a Neumann U67. "I liked the clarity of the 67," Kelly explains. "For me, the top end was a little better suited to vocals than the 87, helping with diction, and to that I added some [Urei] 1176 compression. At that point, there was only one track left, and Kate did just two or three passes, and that was that. There was no comping, it was a complete performance."

Fader Movement



Mixing of The Kick Inside took place inside AIR's Studio Three on a brand new, custom-designed Neve console, featuring innovative NECAM moving-fader automation that, according to Kelly, was something of a double-edged sword. "Some artists don't like to see things move because they become overly aware of them," he says. "Moving faders can be distracting, and that's why SSLs work well with the VCAs — you don't see the fader movement, you're not distracted by it and you can concentrate on the music. Because sometimes, when you do a move, you automatically look at the fader to see if it's gone where you wanted it to go, instead of concentrating on how it sounds." Nevertheless, over the course of a weekend Jon Kelly completed 'L'Amour Looks Something Like You' and 'Moving' before, at about 11 o'clock on the Sunday night, Andrew Powell informed him he'd need at least three tracks to play for the EMI execs the following day. Forget any ideas about going home. He wanted 'Wuthering Heights' to complete the trio. "We started that mix at around midnight, and Kate was there the whole time, encouraging us," Kelly remembers. "She was the shining light of the entire sessions. You couldn't deny her anything. So, we got on with the job, and we finished at about five or six that morning. It was a fairly straightforward mix — among the only effects were a pair of EMT 140 echo plates, one straight, the other delayed Photo: Jorgen Angel / Redferns with a 15ips Studer — but it gelled and Kate Bush's only major concert tour took it had a whole vibrancy to it, and full place in 1979, and included this performance credit for that goes to Andrew. He got in Copenhagen. the arrangement exactly right; he got the chords right, he translated Kate's work beautifully, and everything that was on the multitrack deserved to be there." For his part, Kelly would subsequently regret not mixing Ian Bairnson's guitar solo a little louder. "I always used to apologise to him whenever I saw him afterwards," he says, even though the balance actually works perfectly well, said guitar soaring subtly above all else during the track's instrumental outro. Still, 20/20 hindsight is a wonderful thing. "I love the fact that performance was our main concern back then, and that everything had a distinctly human feel," Jon Kelly now states. "These days, that whole album would be approached differently — it might end up with a Logic sequencer somewhere — but in the final analysis Kate's talent would shine through anything. It would shine through an old dustbin lid and a rubber band. And that's what I loved about working recently on the Tom Baxter album, which had that spirit of performance. It was almost like coming full circle. Wouldn't it be great if we have an era of real playing and performance? I think the kids are file:///H|/SOS%2004-06/CLASSIC%20TRACKS%20%20%27Wuthering%20Heights%27.htm (6 of 8)9/22/2005 7:46:28 PM


screaming and gagging for that."

Separation Anxiety After some screens had been strategically placed, there wasn't too much effort expended on attaining separation when the band was playing. "We weren't as precious about that sort of thing back then," Kelly asserts. "Since the advent of multi-channel recorders and people's ability to solo stuff, if they suddenly hear some drums on the piano they'll go into a panic. During the Kick Inside sessions, the piano wasn't all that far from the drums, yet it really wasn't an issue. For us, the performance was the thing." In most cases, this was captured within just a handful of takes — it wasn't unusual to complete two or three backing tracks in a single day, and so, allowing for several breaks as well as the mix, the entire album took about six weeks to complete. "I can't remember doing any editing on Kate's sessions," says Kelly, "but I can remember 'Wuthering Heights' being a performance-y type song. Stuart was a brilliant drummer, he absolutely adored Kate's songs, and the all-round enthusiasm and will to play well on those sessions was just fantastic. They were great musicians, and everything they did was of a very high standard."

Scaling The Heights 'James And The Cold Gun' was orginally scheduled to be Kate Bush's debut single, yet when Bush pleaded the case for 'Wuthering Heights' EMI deferred to the teen prodigy's intuition, and by January of 1978 the song was topping the UK charts. It would be a massive hit almost everywhere except America, where the diminutive musician would have to wait several more years before achieving her breakthrough. And the song also provided the British press with ample opportunity to dismiss her as little more than an eccentric novelty, before her subsequent releases dispelled that notion. One of Kate Bush's best-loved songs, 'Wuthering Heights' was accorded a new vocal by her when included on her 1986 greatest hits compilation The Whole Story. Nevertheless, despite the furore which that provoked amongst purists, it is still the original that garners the most airplay. Published in SOS June 2004






Creative Synthesis with Yamaha XG

In this article:


A Closer Look XG Masterclass: Part 3 At The XG Modulation Matrix Published in SOS June 2004 Track Mixer Print article : Close window Scripts For Technique : Synthesis Cubase VST & SX Dynamic Detuning & Sweeping Chords In the final instalment of our series of XG programming Modulation a look at how the advanced modulation parameters can Matrix Parameters layered sounds to life. Auto-wah & Advanced Filter Sweeps Mike Senior Programming XG Effect Last month I looked at how you can unlock the sound-design Parameters potential of your XG sound source by layering voices. One Tremolo & problem with layering voices, though, is that the normal MIDI Crossfades controllers for pitch-bend, filter cutoff, and volume control XG Xtreme adjust all parts on the same MIDI channel in the same way.

tips, we take bring your

Therefore, in this final instalment of Yamaha XG programming tips, I'm going to look at how you get around this using the more advanced modulation facilities tucked away in the XG MIDI specification.

A Closer Look At The XG Modulation Matrix The modulation possibilities available within the XG format are remarkably flexible: six MIDI message types can feed the matrix, and each of these can be routed to any or all of six destinations in the voice architecture, with individual modulation amount control — that makes 36 parameters in total. However, because only a few (if any) of these parameters are actually accessible from the front panels of most of the hardware XG modules, MIDI SysEx messages are pretty much the only way to take full advantage of the power available. The six MIDI control sources which can feed the matrix are Mod Wheel, Pitch-bend, Channel Aftertouch, Polyphonic Aftertouch, Assignable Controller 1, and Assignable Controller 2. Some of the simplest XG modules may not recognise anything more than MIDI Mod Wheel and Pitch-bend messages (and I'll mostly be using those controllers for my examples in this article), but most of the studio units I've encountered will recognise all of them. The two Assignable Controller sources are MIDI Continuous Controllers, the numbers of which can be set using the hexadecimal MIDI SysEx strings F0 43 10 4C 08 mm 59 xx F7 and F0 43 10 4C 08 mm 60 xx F7 respectively, where 'mm' is the multitimbral part and 'xx' the desired MIDI Continuous Controller number. Any of these modulation sources can control pitch, over a range of up to 24 semitones (two octaves); file:///H|/SOS%2004-06/Creative%20Synthesis%20with%20Yamaha%20XG.htm (1 of 7)9/22/2005 7:46:32 PM


filter cutoff, starting from the main cutoff setting; and amplifier level, starting at the main volume setting. All of these three modulation destinations can be modulated with positive or negative polarity, so the filter can close or open as the mod wheel is raised, for instance. The MIDI control sources can also modulate the degree to which the single LFO affects any or all of pitch, filter cutoff, and amplifier level, although here you can only make a positive-polarity setting. All six modulation destinations can be seen in Figure 1, which shows how MIDI Mod Wheel messages are routed to them using the six relevant XG parameters. A complete list of all the modulation-matrix parameters, along with their proper XG-format names and hexadecimal SysEx messages, is given in the 'Modulation Matrix Parameters' box for reference purposes, although all these messages can also be found in Yamaha's XG MIDI specification document mentioned in the first article of this series. Obviously, most of the modulation-matrix parameters are set to zero by default, but there are two exceptions: MIDI Pitch-bend messages normally adjust pitch over a range of ±2 semitones, and MIDI Mod Wheel messages control the LFO pitch modulation amount to a small degree. Bear in mind that MIDI Pitch-bend messages are different to Continuous Controller messages — Pitchbend messages can have both positive and negative values, where Continuous Controller messages have only positive values. So if you have, for example, set Pitch-bend messages to modulate filter cutoff with positive polarity, pushing the pitch-bend wheel up will raise the cutoff point beyond its nominal setting, while pulling the pitch-bend wheel down will cause the filter cutoff to fall below its nominal setting. However, if you try to create a negative-polarity LFO effect by controlling the LFO modulation amount using Pitch-bend messages, I'm afraid it won't work — negative Pitch-bend values are simply treated as positive ones for the purposes of LFO control. Another thing worth mentioning is that, although the XG specifications say that the Filter Control parameters span -9600 cents to +9450 cents and that the Amplitude Control parameters span ±100 percent, the actual data byte which conveys the information still only covers 128 discrete steps, so you might as well think of both parameters as encompassing the usual range of -64 to +63, in my opinion. The calibrations certainly seem very little use in practice, as I find it's best to set up modulation ranges by ear. You'd be forgiven for nodding off during the above theoretical discussion of the modulation matrix, as it's not immediately obvious how all these parameters can be used to make your XG sounds more involving. So, I'm going to spend the rest of this article giving lots of examples of ways you can use all these parameters in practice to spice up layered XG sounds.

Track Mixer Scripts For Cubase VST & SX If you have been having trouble finding XG Track Mixer Scripts for Cubase VST or the first version of Cubase SX, then you'll be pleased to know that SOS reader Julian Slade has emailed in to say that he's now found an MU100 set on a Japanese site, including effects and EQ parameters. http://popup10.tok2.com/home2/mitsubamushi/komonoda_Cubase.htm

Dynamic Detuning & Sweeping Chords To start with, let's have a look at some of the possibilities made available by pitch modulation. Always

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one for a challenge, I've layered three voices of the exceptionally cheesy HyprAlto alto sax patch to try to make a useful pad synth line. (I showed how to layer voices together in last month's article, so I won't go over it again here.) You can hear my preliminary efforts in Example 1a, where I've pulled down the filter resonance, adjusted the attack and release times, and applied some judicious Pitchbend to try to disguise the worst vices of the original waveform. (As with last month's article, you can download all the sound files relating to this workshop from www.soundonsound.com/soundbank.) Hardly the most inspiring line, I'm sure you'll agree. However, I'm deliberately going to avoid going any further with filter and effects controllers at the moment in my attempts to de-cheese it, and instead I'll use MIDI Mod Wheel messages to apply some dynamic detuning. First of all, I've switched off the mod wheel's routing to the pitch LFO modulation amount by using a zero-value MW LFO PMod Depth message for each of the voices in the layered sound. Now MIDI Mod Wheel messages have no effect on the sound at all. Then I've used MW Pitch Control messages so that one of the layers sweeps up a semitone as the mod wheel is raised and another shifts down a semitone, while the remaining layer remains stationary. The result of waggling the mod wheel around can be heard in Example 1b, and this instantly gives the composite sound more life and movement, especially given that the voices are already panned to different positions. The great thing about doing this pitch modulation from the mod wheel is that any pitch-bend work you've already done is unchanged — the detune thickening effect can be applied completely independently of the pitch sweeps. Now have a listen to Example 1c, where I've added an extra lead line onto the end of the previous rhythm part. Again, not awfully inspiring, so this time I've going to use CAT Pitch Control messages to create a much wider pitch modulation effect using MIDI Channel Aftertouch messages. First I've instructed two of the voices to shift 14 and 24 semitones respectively over the aftertouch control range. This means that all the voices are in unison at an aftertouch value of zero. However, I want to be able to pitch-shift in both directions around an aftertouch value of 64, so I've adjusted the Note Shift parameters of the two voices to -7 and -12 respectively. The final result is that the two shifted voices are in unison with the unshifted one at an aftertouch value of 64, but they also create a variety of different chords above and below the unshifted voice at all the other aftertouch values. But enough of the theory; Example 1d shows one way of using this approach, and you can hear that I've managed to make the part much more complex than it started out, because even single notes can become chords. Note again that the chordal shifts work independently of, and in addition to, the existing pitch-bend sweeps triggered from MIDI Pitch-bend messages. Now let's see what the LFO can bring to the party. For the added lead line, I'm going to hijack the mod wheel from its detuning task, and instead route it to controlling how the LFO affects the pitch of each voice using the MW LFO PMod Depth parameter. As we discussed last month, each layer of your composite sound can have its own synth settings, and this of course applies to the LFO. However, the MW LFO PMod Depth parameter also lets you set how much the LFO controls pitch in each case. In Example 1e, I've set the voice which is unaffected by the aftertouch messages to be modulated heavily with a fast LFO as the mod wheel is raised, while the shifted voices are treated to a more modest amount of slower LFO modulation. Hopefully I've demonstrated how you can use pitch modulation to create a variety of different timbres from even the most uninspiring raw materials. If you want even more variety then you can still, of course, play with all the filter and effects parameters as well — in Example 1g I've tweaked filter cutoff and resonance in real-time, and there's also some dynamically applied flanger, phaser, and reverb.



Modulation Matrix Parameters MIDI PITCH FILTER AMPLIFIER LFO PITCH LFO FILTER LFO MODULATION MODULATION CUTOFF LEVEL MODULATION CUTOFF AMPLIFIER SOURCE MODULATION MODULATION MODULATION LEVEL MODULATION Mod Wheel

Pitch-bend

Channel Aftertouch

Polyphonic Aftertouch

Assignable Controller 1

Assignable Controller 2

MW Pitch Control

MW Filter Control

MW Amplitude MW LFO Control PMod Depth

MW LFO FMod Depth

MW LFO AMod Depth

F0 43 10 4C 08 mm 1D xx F7

F0 43 10 4C 08 mm 1E xx F7

F0 43 10 4C 08 mm 1F xx F7

F0 43 10 4C 08 mm 20 xx F7

F0 43 10 4C 08 mm 21 xx F7

F0 43 10 4C 08 mm 22 xx F7

Bend Pitch Control

Bend Filter Control

Bend Amplitude Control

Bend LFO PMod Depth

Bend LFO FMod Depth

Bend LFO AMod Depth

F0 43 10 4C 08 mm 23 xx F7

F0 43 10 4C 08 mm 24 xx F7

F0 43 10 4C 08 mm 25 xx F7

F0 43 10 4C 08 mm 26 xx F7

F0 43 10 4C 08 mm 27 xx F7

F0 43 10 4C 08 mm 28 xx F7

CAT Pitch Control

CAT Filter Control

CAT Amplitude Control

CAT LFO PMod Depth

CAT LFO FMod Depth

CAT LFO AMod Depth

F0 43 10 4C 08 mm 4D xx F7

F0 43 10 4C 08 mm 4E xx F7

F0 43 10 4C 08 mm 4F xx F7

F0 43 10 4C 08 mm 50 xx F7

F0 43 10 4C 08 mm 51 xx F7

F0 43 10 4C 08 mm 52 xx F7

PAT Pitch Control

PAT Filter Control

PAT Amplitude Control

PAT LFO PMod Depth

PAT LFO FMod Depth

PAT LFO AMod Depth

F0 43 10 4C 08 mm 53 xx F7

F0 43 10 4C 08 mm 54 xx F7

F0 43 10 4C 08 mm 55 xx F7

F0 43 10 4C 08 mm 56 xx F7

F0 43 10 4C 08 mm 57 xx F7

F0 43 10 4C 08 mm 58 xx F7

AC1 Pitch Control

AC1 Filter Control

AC1 Amplitude Control

AC1 LFO PMod Depth

AC1 LFO FMod Depth

AC1 LFO AMod Depth

F0 43 10 4C 08 mm 5A xx F7

F0 43 10 4C 08 mm 5B xx F7

F0 43 10 4C 08 mm 5C xx F7

F0 43 10 4C 08 mm 5D xx F7

F0 43 10 4C 08 mm 5E xx F7

F0 43 10 4C 08 mm 5F xx F7

AC2 Pitch Control

AC2 Filter Control

AC2 Amplitude Control

AC2 LFO PMod Depth

AC2 LFO FMod Depth

AC2 LFO AMod Depth

F0 43 10 4C 08 mm 61 xx F7

F0 43 10 4C 08 mm 62 xx F7

F0 43 10 4C 08 mm 63 xx F7

F0 43 10 4C 08 mm 64 xx F7

F0 43 10 4C 08 mm 65 xx F7

F0 43 10 4C 08 mm 66 xx F7

Auto-wah & Advanced Filter Sweeps Using MIDI Continuous Controller number 74 to sweep the filter cutoff of all voices in a layered sound together can give you a lot of useful sounds, as I demonstrated in the second part of this series. However, uniform filter control across all layers only gets you so far, and there's a lot more mileage in filter modulation once you get stuck into the serious modulation matrix parameters. Take a listen to Example 2a, which is a simple two-layer composite sound comprising two identical RndGlock (a variation on the Crystal patch) voices panned hard left and right. To get an interesting stereo effect, reminiscent of a rotary speaker, I've set each voice to have a different LFO rate, and then I've adjusted the amount of LFO filter modulation in real time to create Example 2b. This is made possible using the file:///H|/SOS%2004-06/Creative%20Synthesis%20with%20Yamaha%20XG.htm (4 of 7)9/22/2005 7:46:32 PM


MW LFO FMod Depth parameter, set to its maximum value of 127. You could equally well use this parameter to create simple auto-wah effects for single-layer voices or those with identical LFO settings. You can also use the MW Filter Control parameter to set up different filter responses for each part, which means that one voice can have its filter closing as another voice has its filter opening, for example. Have a listen to Example 3a, which is a simple lead synth phrase using a two-layer sound comprised of two variations on the ChoirPad patch: Heaven 2 and CC Pad. There's only a bit of pitchbend and expression control at the moment and, although the patches have a little movement in them, there's more that can be achieved with independent Filter Control settings. Example 3b uses opposite-polarity MW Filter Control messages for the two layers, and the mod wheel has been swept in real time. Not only do you get the motion of the two different filter peaks, but you also get the change in timbre between the two different voices as they dominate at different ends of the mod wheel's travel, which makes this one of the most pleasing and complex modulation effects, as far as I'm concerned. Finally, in Example 3c I've used MW LFO PMod Depth to add a little independent vibrato to the parts as the mod wheel is raised.

Programming XG Effect Parameters Once you've overcome your natural fear of hexadecimal messages, there's nothing to stop you spicing up your effects processing as well as your synth programming. For a start, there are sometimes more effect parameters available within your XG module's MIDI implementation than are adjustable from the front panel. But the main reason to get into the SysEx is so that you adjust the effects in real time. All the useful effects SysEx messages begin with F0 43 10 4C 02 01, followed by a single address byte to denote the specific effect parameter. The one or two data bytes of the parameter value (depending on the range of the parameter you're adjusting) then precede the F7 byte which closes the SysEx message. The algorithms for Reverb, Chorus, and Variation effects are set using address bytes of 00, 20, and 40 respectively, and this message uses two data bytes. After that, most of the messages only need one data byte, which is handy if you have a MIDI controller which prefers sending out only 128step data ranges. In each case, the 10 subsequent address bytes access the first 10 parameters of each effect, followed by the effects return level and pan setting. Address byte 2E lets you send the output of the Chorus block to the Reverb, while address bytes 58 and 59 let you send from the Variation effect's output to the Reverb and Chorus blocks. If you've got a fancy-pants XG module with lots of extra effects parameters, you can access six more of these using address bytes 10, 30 , and 70 for the Reverb, Chorus, and Variation blocks respectively. So much for the maths. What kinds of things can you do? Well, unfortunately there's not much mileage in rapidly changing effects algorithms, as the output of the effect is muted briefly during this process — whatever effects patches you select will always give you a variation on tremolo! Even changing algorithms between larger musical sections is hampered a little with the reverb effect, where the change in algorithm won't, of course, let the previous reverb tail ring on. However, that doesn't stop you being able to flip between different Chorus and Variation effects for different phrases to make the most of the available effects provision in your particular model of synth. The most interest is probably to be found in real-time control of parameters rather than algorithms, though. It's particularly worth targeting the Chorus and Variation blocks' LFO parameters. Chorus, Celeste, Flanger, Phaser, Symphonic, Tremolo, Rotary Speaker, Auto Pan, and Auto Wah are just some of the algorithms which use an LFO, so there's lots of fun to be had here, especially as the LFOs can be set for the Chorus and Variation blocks separately. Another favourite of mine is to use the 3band EQ algorithm's mid-band as a sweepable peaking filter, setting the resonance and gain to taste or tweaking those on the fly too. The drive and output level settings of the Distortion, Overdrive, and Amp Sim algorithms are also obvious contenders, but don't let that stop you flicking madly between a selection of different amp models! The feedback controls of any of the delay-based algorithms can be used both subtly for massaging repeat tails on lead lines at mixdown, or ham-fistedly for turning Flanger file:///H|/SOS%2004-06/Creative%20Synthesis%20with%20Yamaha%20XG.htm (5 of 7)9/22/2005 7:46:32 PM


or Phaser into a howling mess. One problem that you may encounter, though, is that parameters for reverb decay and delay times will create more-or-less noticeable clicks at the effects output, because of the way such effects processes work. In the case of reverb processing, there's not much you can do to get around this, but with delay effects you can try using the multi-voice MIDI delay effects I discussed back in the first part of this series, as these are much more creative in lots of other respects anyway.

Tremolo & Crossfades The final section of the synthesis architecture which can be controlled through the XG modulation matrix is the amplifier, and the LFO can provide some nice tremolo effects here. Have a listen to Example 4a, a chord using two different layered XG guitar patches — Jazz Gtr and MelloGtr. I've slowed down the envelope for the Jazz Gtr voice and increased its filter resonance, as well as adding a little not-so-subtle flanging, so there is already some movement in the sound straightaway, but adding heavy tremolo using the MW LFO AMod Depth parameter livens things up in Example 4b, especially as the flanger's LFO is slightly out of sync with the tremolo LFO. What I've done, though, is apply the tremolo only to the brighter, more high-resonance Jazz Gtr part. This means that the duller MelloGtr part remains in the tremolo troughs, making the modulation effect more akin to filter modulation. The potential of this technique should not be underestimated, as you can make the tonal changes as subtle or extreme as you like simply by selecting different patches. And, of course, you can also throw some additional controllers at the sound, to sculpt it even further — a few MIDI Expression messages give the sound a new slow-attack envelope in Example 4c, and a filter sweep picks out individual harmonics in Example 4d. The Amplitude Control parameters can also be very useful, especially if one voice fades up as the other fades down. Take Example 5a, for instance, where I've layered the MutePkBa bass patch with a sub-bass SineLead patch, in order to get both attack and low-end welly. This is alright as it stands, but we can make things more adventurous by using the MW Amplitude Control parameter to crossfade the MutePkBa patch to FastResB on some of the notes.

Figure 1. The six possible modulation routings for MIDI Mod Wheel messages.

The first step is to set the MW Amplitude Control on the MutePkBa voice to +63, and that of the FastResB voice to -64. Then the main volume setting of the FastResB voice needs to be set to a value of one — you have to set it above zero, otherwise the amplitude modulation will have no effect. Now that this has been done, you can crossfade between the two voices in real time using the mod wheel. I've used extreme mod-wheel positions in Example 5b to just switch between the two voices, rather than crossfading, but I could have crossfaded more subtly if I'd wanted. Notice that, because I've left the SineLead voice unaffected, the low end weight doesn't change as you crossfade between the two other voices. Example 5c combines all of the audio examples from this article, and you can download an XG-format MIDI file of them all from the SOS web site if you want to have a closer look at the exact data I've used in each case. For my money, the ability to crossfade between voices using the Amplitude Control parameters is the most powerful facility offered by the XG format. It means that you can create a pair of complex layered sounds (each complete with its own in-depth synthesis, effects, and modulation settings) and then crossfade between them in real time using, say, Assignable Controller 1. And you could crossfade to a third voice using Assignable Controller 2. And to a fourth on a note-by-note basis file:///H|/SOS%2004-06/Creative%20Synthesis%20with%20Yamaha%20XG.htm (6 of 7)9/22/2005 7:46:32 PM


using Polyphonic Aftertouch, assuming you can find something to generate the appropriate MIDI messages! The only limitations are the number of MIDI control sources recognised by your module and the number of notes of polyphony available — most studio XG modules can handle all six modulation sources and at least 32 voices of polyphony, so you have the potential for an enormous range of evolving and expressive sounds.

XG Xtreme In this short series I hope I've been able to convince you that there's more to XG than its dodgy General MIDI legacy. As long as you modulate any parameter that isn't nailed down, and resolve to waste polyphony gratuitously at every opportunity, then any XG module should provide more than enough inspiration to earn its keep in your studio. Published in SOS June 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:

Ndot Doctor's Advice: Vocal Variations End Of May QUICKIES

Demo Doctor Readers' Recordings Assessed Published in SOS June 2004 Print article : Close window


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

Ndot Recording Venue: Home Recording Equipment: Apple Mac G4, Steinberg Cubase VST sequencer, BIAS Peak editing software, PPG Wave synth, Propellerhead Reason software, Shure microphone (unspecified model). Track 1 Both compositions on this CD are full of interesting 1.4Mb production touches, such as clever use of stereo Track 2 delay and reverb. For example, the snare drum on 1.4mb the second track, 'Pain', punctuates like a whip crack in some places and in others is preceded by backwards reverb. Like some other demos we've received recently, this one attempts to go for a big sound with expansive delay and reverb. It's mainly successful on this front, but a more punchy, direct sound could be achieved with a little less use of effects. Likewise, more light and shade could be given to the vocals by applying less of the delay and phase effects used throughout the entire track — the occasional drier vocal line would have some impact. However, there are many instances of good variation in the use of vocal effects, like the backwards delay and reverb applied to certain phrases. I particularly enjoyed the moments where the voice deliberately sank into the backing and became part of the sound texture rather than the main focus of attention. Obviously, the vocal delivery is structured to be performed with effects, and singer Mel Skye does a fine job of sounding ethereal yet menacing!

A matter of more concern is the tendency for the mid-frequency area of the mix to see a lot of action. This is a result of similar equalisation being applied to most of the instrumentation, as well as the sheer number of parts with a lot of midfrequency content which are thrown into the mix. Listening to the first mix, there file:///H|/SOS%2004-06/Demo%20Doctor.htm (1 of 6)9/22/2005 7:46:37 PM

Demo Doctor

seems to be an emphasis on the 2kHz region and a bit of a hole between the upper midfrequencies and the low end. In fact, the bass sound is pretty indistinct unless the track is played very loud and even then it's not great. A bass sound with more definition, similar to the one used on the second song, would also deal with that hole in the lower mid-range because it has more energy in that frequency area, and would bring the whole mix together nicely. As it is, the over-use of upper-mid EQ emphasises the lack of quality in some of the sounds, especially on the first song. There is a grainy quality to the drum loop and you can hear something which could be high-frequency interference or the higher frequencies of a rather harsh-sounding keyboard. Looking at Ndot's gear list, which is not short of quality synth sound sources, I think it's more likely to be a poor-quality sample that is the culprit, possibly one downloaded from the Internet. A similar upper-mid EQ on the vocals pushes it towards sibilance, especially on the second mix, and it wouldn't reduce the impact of the vocal to back off the EQ. The EQ may be attempting to compensate for using a dynamic microphone on the vocals, but the microphone is not specified on the gear list. Even so, this is a good demo with the second song getting my vote as the better of the two tracks, both technically and in terms of composition. Ndot have been around for quite a while and in that time have submitted a few CD's to SOS. They seem to be taking their music seriously too: they've got their own web site, which is simple and clearly laid-out, and their live show sounds well worth checking out. www.ndot.co.uk

Doctor's Advice: Vocal Variations In reviewing the demos this month I was struck by the different approaches taken to the use of vocals and effects. While the guitar-driven rock and pop tracks feature standard treatments of delay, reverb and occasionally overdrive, the dance mixes tend to be more innovative, using filtering, pitch-change and vocoder effects to manipulate the sound. While I'm not suggesting that a rock or pop vocal should be completely altered throughout by such techniques, perhaps it's time for a less conservative approach, at least on a line or two of a song.

End Of May Recording Venue: Home Recording Equipment: Apple Titanium Powerbook 667, MOTU 828 MkI audio interface, Rode NT1, Shure SM58 Beta, AKG C1000 (x2) and AKG D112 mics, Soundcraft Folio FX8 mixer, TL Audio valve mic file:///H|/SOS%2004-06/Demo%20Doctor.htm (2 of 6)9/22/2005 7:46:37 PM

Demo Doctor

preamp, Genelec 1031 monitors. The covering letter for this demo, was brimming with enthusiasm about new technology and the recording process. Vocalist and instrumentalist Chris Wang extols the virtues of portable computer recording and the joys of working with like-minded individuals.

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

The rock mixes on the resulting CD have many good points, and one of them is the excellent vocal sound. The Rode NT1 mic and TL Audio preamp combination is certainly one which suits Chris's voice. In the absence of an outboard compressor, he must have used a plug-in for compression. It's also possible that the vocals, which are placed right at the front of the mix, are being compressed quite heavily by the mastering process. Certainly, there are points in the first song where the voice rises in level and it's possible to hear the compression cutting in. Backing vocals are used sparingly and, in my opinion, they're mixed too low to make much of an impression when they arrive. I'm thinking particularly of the second song on the CD, where you can hear a nice harmony line tickling away in the background on the chorus. On the first song, harmony vocals would have been a bonus, and could have added to the strong melody line of the chorus, but I liked the emotive vocal cries blending into the keyboard mix towards the end of the song. The weak point of this demo is the drum sound. In the first track, the snare has been mixed too low and, for a rock band, it's a very weak sound with no attack. It's possible that the slow action of a noise gate could contribute to this, robbing the snare of bite. However, many plug-ins have a 'look ahead' control to compensate for this, so there is no real excuse. If the plug-in has no such parameter you can copy the snare to another track, pre-delay it fractionally and use that snare to trigger the gate on the original snare track, making it open a fraction of a second before the original is hit. In fact there are a number of other methods available, but this has always proved the most reliable for me. I was also struck by the strange EQ applied to the snare, which seems to have a cut around 900Hz-1kHz. This allows more of the sound of the actual snare to be heard but at the expense of that valuable attack. This is particularly noticeable on the second track. In spite of this, the songs are rather good, although none of them stand out as singles. Some of the acoustic guitar sounds are well recorded and it's a strong debut recording. www.endofmay.com

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Demo Doctor

QUICKIES

Moonflower Here the drum sound and level of the opening track tend to bring down what is a fine rock song. The tightly compressed snare sound, in particular, is too loud and detracts from both vocal and guitar parts. In fact, contemporary rock mixes would have the guitars louder and would not allow the drums to sound so obviously sequenced. With a louder guitar mix, especially on the chorus, the whole song would have more impact and the fine vocal performance of Jen Wolstenholme would then be fully supported by the backing. The other tracks on this CD tell a similar story. Even though a variety of drum sounds are used (probably too many for continuity's sake, in a rock context at least) the emphasis is always on the snare. I also noticed some double triggering of the kick and snare on the third mix, resulting in a bit of phasing, and this needs to be removed. A reference for further productions is probably the last song on the CD, which had the best overall sound.

Aural Perception This duo offer a variety of different styles of electronic music, and are at their most successful on the fourth composition on the CD. Here the introduction of a vocal sample (I think it's the word 'me') adds an element of humour missing from their other mixes, and the choice of sounds and rhythmic drive are somewhat reminiscent of Yello. Elsewhere, they pick up the influences of Leftfield and Underworld without bringing in the ethnic instrumentation of the former and have the skill to combine sounds well. I particularly enjoyed the kick drum-triggered synth in the opening of the third mix. It's a real attention-grabber and the mix balance at this point is just about perfect. Elsewhere, I was concerned with the overload on the drum loop of the first mix, which seems to increase once the growling synth bass arrives. I wasn't entirely convinced that this was deliberate — a highly resonant filter sweep also overloads about a minute into the arrangement. In contrast, the other mixes are clean and punchy, comparing favourably against the professional mixes I listen to on my monitors.

Jonah The dance and chill-out CD market is where composer Craig Simmons aims his work and it has to be said that he faces tough competition! The 'anthemic' tracks he seeks to emulate must have both drive and hook, and Craig's mixes seem to have the component elements but lack the energy. It's partly a question of confidence in the mixing and partly a case of recognising where the track's strengths lie and exploiting them. If Craig makes more of the hooks by using stronger sounds, and emphasises the dynamic lifts already hinted at, it will give the guys on the decks something to work with. Chillout mixes are probably easier to achieve with a fairly basic setup and this is where this demo hits the spot. His clever choice and manipulation of vocal samples on the third mix demonstrate an ear for a hook and the ability to use the equipment well. The use of a harpsichord sound here is unexpected but evocative, and the simple piano file:///H|/SOS%2004-06/Demo%20Doctor.htm (4 of 6)9/22/2005 7:46:37 PM

Demo Doctor

line could have had a longer reverb but is still effective. This is the best track on the CD and has some potential, but it needs a bit more work.

Martin Rigby In the review of Martin's last demo, I praised the sequencing and instrumental performances, but I also suggested that there was too much action in the mid-frequency range (800Hz-2kHz), leading to a lack of clarity, and that the rhythm guitar was too low in the mix. In response Martin has mastered this demo with what sounds like a mid-EQ cut, when I was really suggesting that he look more closely at his choice of sounds and their arrangement in the mix. In fact all three tunes I listened to on this demo have the same problem. The guitar is meant to take the lead melody, but is being obscured by the string lines at crucial points in the arrangement even when the string part is a simple harmony line. So, bring the strings down in level where necessary and look again at your guitar sound. On the second mix, the guitar has more attack and so doesn't disappear into the more mellow keyboards, and a reverb gives the guitar its own identity and results in a more successful production sound.

AD This CD has got everything right but the mixing! The artwork and sleeve design are excellent but the general sound is very tinny and must have been over equalised in the upper mid-region at some point in the recording process. I only hope it was at the mastering stage and can be easily fixed, because there's plenty of good music here. Worthy of mention is the way composer Aiden Gallagher uses real instruments such as trumpet and violin, in what is essentially an art pop context. I think this lends a freshness to the production and has an energy that works as a foil to his melancholy vocal performance. So a remix would be a good idea, and why not make more use of the analogue sound of that Soundtracs Jade console to bring out the organic qualities inherent in the compositions? If the setup allows, try mixing the signal through the channels on the Jade with some minor equalisation. That may be enough to bring back some warmth! Published in SOS June 2004


Demo Doctor




FP Numbers

In this article:

Audio And Decibels Computer Storage Fixed-point Systems.

FP Numbers How do they affect your music? Published in SOS June 2004 Print article : Close window

Technique

A reference guide to understanding the terms used in SOS articles and technical documentation. David Nash

Many of the articles in SOS, not to mention the specifications for audio hardware and software, use the terms fixed point numbers, floating point numbers and decibels. But what do terms like this actually mean? And what are the consequences for our music? This guide is intended to help SOS readers who would like to know a bit more about the relationship between the sounds that we hear, and the numbers that are pushed around inside our PCs. Sound Intensities And Decibels. The range of sound intensities which can be processed by the human ear covers about 14 powers of 10 (1014). If we exclude all outside sounds, we can derive a threshold intensity at which sound can barely be heard. This is of the order of 1012 Watts per square metre at frequency of 1kHz, but itÕs not very practical for audio engineers and musicians to use these units from physics. It is more useful to have a measure which uses the threshold level (or some other reference level) as a starting point, and states the factor by which a given sound is larger than the threshold. That is, if Ip is the particular sound intensity of interest, and I0 is the threshold intensity, we can use the ratio Ip/I0 as a measure. In order to bring the large numbers given by powers of 10 into a more manageable range with our subjective perception of sound, we take logarithms to base 10 of the ratio, and use the following formula to measure sound in terms of its intensity level. Intensity level = 10 log(Ip/I0) This unit of intensity is the decibel, abbreviated dB, and measures the amount of sound power per unit area. From this definition we can obtain a useful scale with which to measure sounds as shown in table 1. This puts the level of the threshold file:///H|/SOS%2004-06/FP%20Numbers.htm (1 of 16)9/22/2005 7:46:42 PM

FP Numbers

of hearing as zero decibels (0dB). Table 1: Sound intensities expressed in decibels Intensity, Watts/m2 Ratio Ip/I0 Level, dB Threshold of hearing. 1 0 10-12 ppp (very soft)

10-8

104

40

p

10-6

106

60

f

10-4

108

80

fff (very loud)

10-2

1010

100

Threshold of pain.

100

1012

120

Very painful.

102

1014

140

We now have a manageable range of numbers (0 to 140) which covers the range of our aural processing. It can be seen that that the range from fff to ppp is 10040 or 60dB, a very manageable set of numbers to use in music. Doubling the intensity of a sound increases its measured level by 3dB, regardless of its current intensity: dB = 10 log(2/1) ie. 10 x 0.3010 =3 and halving the intensity decreases the measured level by 3dB: dB = 10 log(1/2) ie. 10 x Ð0.3010 = -3 Microphones respond to pressure amplitude. The intensity (power) of a sound wave is proportional to the square of the sound pressure level (amplitude, analogous to voltage), therefore when used to measure sound pressure level (SPL), the decibel becomes: dBSPL = 10 log(SPLp/SPL0)2 = 20 log(SPLp/SPL0) Psychologists have discovered that at a level of 30dB using a 1kHz signal, the smallest perceived change in sound level is about 1dB.

Audio And Decibels In audio engineering, the decibel still describes how much larger or smaller one value is than the other. It can also be used as an absolute unit of measurement for lining up audio equipment if the reference is fixed and known. (See SOS Q&A file:///H|/SOS%2004-06/FP%20Numbers.htm (2 of 16)9/22/2005 7:46:42 PM

FP Numbers

January/October 2003 for defined references). The decibel is now defined as 10 multiplied by the logarithm to base 10 of the ratio between the powers P of two signals: (Equation 1) dB = 10 x log(P2/P1) As before with sound intensities, a doubling of the power will be an increase of 3dB. If we want to use the decibel to measure increases/decreases in signal levels using voltage or current, we need to take into account the relationship between power (W) and voltage (V) or current (I). OhmÕs law gives W = V2/R, or I2R, so to compare two voltages: (Equation 2) dB = 10 x log(V2/V1)2 = 20 x log(V2/V1) If we double the voltage of a signal from V1 to V2, the gain in dB will be: (Result 1) dB = 20 x log(V2/V1) = 20 x log(2/1) = 20 x 0.3010 =6 That is to say, the doubling of a voltage (in a permitted range) is a change of +6dB in the level of the audio signal that voltage represents. A halving of the voltage will be Ð6dB change. We can now use decibels to describe the voltage gain of an amplifier. If the gain is quoted as 40dB, we use equation 2 to find the ratio R by which the input signal is multiplied. (taking anti logs) 40 = 20 x log(R) 2 = log(R) 102 = R 100 = R Therefore a 40dB gain is the equivalent of multiplying the input voltage by 100.

Computer Storage In the digital domain, we typically capture sound by sampling an incoming signal voltage and converting the sample value to a digital value to be stored in some binary form. So we need to know how computer storage systems can store

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numbers in different forms to represent these signal samples, and how we can increase or decrease them by a specified decibel value.

Fixed-point Systems. Storing integers in a computer is a relatively simple matter. A 16-bit storage unit has 216 combinations of values. If we are not interested in storing negative numbers, then the largest number that can be held in 16 bits is 216-1, which is 65,535. If we need to store negative numbers, then we split the available combination into two halves. The lower half where the leftmost bit is zero is used to store positive numbers, and the upper half where the leftmost bit is always set to one is used to store negative numbers. Thus we still have the same combination of numbers, but instead of being in the range 0 to 216-1, they are now in the range -215 to 215-1 (-32,768 to 32,767). Table 2 shows how various numbers are stored. Table 2: Decimal value Binary value +12

0000000000001100

+255

0000000011111111

-1

1111111111111111

-21

1111111111101011

To find the positive value of a negative binary number, we invert all the bits, and then add 1. This representation of negative numbers is called the twoÕs complement system. Fractions can also be represented if space is allocated: binary fractions to the right of the binary point take the values 1/2, 1/4, 1/8, 1/16É and so on, so decimal 20.625 is binary 10100.101, because 0.625 is 5/8, which is 1/2 + 0/4 + 1/8. Unfortunately, unless the denominator of the decimal fraction is a power of 2, the binary fraction will not convert exactly. In the decimal system, we can increase a number by a factor of 10 by shifting it one place to the left. For example, 21.6 becomes 216.0. In the binary system, we increase a number by a factor of 2 by shifting it one place to the left. For example, 1100.1 (12.5) becomes 11001.0 (25.0). It can also be seen that in order to double the capacity of a binary storage system, one extra binary digit is needed. As we have seen, to double the value of a number in that system, we shift it 1 place to the left. From Result 1, the doubling of a signal is an increase of 6dB, therefore if we use a binary integer system to store say, a 16-bit sample, each binary digit represents a potential change (increase or decrease) of 6dB. A 16-bit storage unit can therefore store values in a (16 x 6) = 96dB range. This number is known as the dynamic range of the storage system, since whatever actual (sensible) values we use to record signal levels, they can be changed within a 96dB range. We could decide, for example, that when all the bits are set to 1 (as this is the only valid reference


FP Numbers

point), this represents a level of 0dB, the maximum value handled by the system (here we are talking about 0dBFS, where FS means Ôfull scaleÕ). When all bits are set to 1 there is no more headroom to store any further increases. All other values are counted down from this level, and when the value of just 1 is present, this could represent Ð96dB, the minimum value handled by the system. As we shall see later, it is wise to leave some bits for headroom overflow, as well as some bits to accommodate fractional values of numbers. It is clearly a relatively simple operation to change the value in a 16-bit storage system by 6dB by shifting the number one place to the left (+6dB), or to the right (-6dB). For example, let's consider a 16-bit sample containing the value 27. 0000000000011011 (27) shifting one place left gives 0000000000110110 (54) The gain in decibels is: dB = 20 x log(54/27) = 20 x log(2) =6 Note that if we attempt to reduce the value 27 by 6dB, by shifting one place to the right, the value becomes: 0000000000001101 This is 13 in decimal representation. The rightmost bit has been lost; we should have 1101.1, which is 13.5. Moreover, a reduction of 12dB (two shifts right) would have lost two bits, resulting in a value of 6 instead of 6.75. Further arithmetic on these numbers could introduce even more loss of accuracy. What can we do about it? And how can we make changes which are not multiples of 6dB, for example, 5 dB, or Ð4dB? Table 1 shows that our ears can manage a dynamic range of about 140dB. In order to store values of this order, we need storage units of (140/6), ie. 24 bits. But both 16-bit and 24-bit integer storage can still only record changes in multiples of 6dB. Special-purpose hardware like digital mixing desks, and some applications development languages, allow integer arithmetic on 32-bit, fixedpoint numbers, where the hardware determines the position of the binary point (if there is one), or permits the management of this to be left to the programmer. Table 3 shows how 24 bits (three bytes) may be assigned to store the results of signal processing. Such an assignment would normally have to be managed by the applications programmer.


FP Numbers

Table 3. 4 bits

16 bits

4 bits

¥

Binary point The 4-bit headroom allows for ÔoverflowÕ calculations of up to 24dB. The 4-bit fraction permits calculations to be made to an accuracy of 1 decimal digit. The theoretical dynamic range, including the headroom is 20 x 6 = 120dB. LetÕs assume that the above 24 bits contain the integer value 9, and that we want to increase its value by 4dB. The ratio for this is clearly less than the ratio 2 which gives a 6dB increase; it is 1.585. The required result is 9 x 1.585 = 14.264. The integer value 14 can easily be accommodated in the allocated 16 bits. But what about the fractional part 0.264? When we convert this to a binary fraction, it has the following bit pattern (up to 24 bits): .010000110110110010001011... and cannot convert exactly If we store the most significant (leftmost) bits (0100) of the fraction result in the 4 bits allocated, the actual value stored is: (0 x 2-1) + (1 x 2-2) + (0 x 2-3) + (0 x 2-4) = 1/4 = 0.25 The resulting fraction is accurate to one place of decimal, but it has already lost 0.014 from the actual result. Further arithmetic processing will cause more loss of accuracy in the results. Floating-point Systems The basic problem in the fixed-point integer/fraction system for a computer implementation is how many binary digits to allocate to the integer and fractional parts respectively. An astronomer might want to record numbers as large as 1 billion. A physicist might want to use numbers as small as 1 millionth. Audio engineers use 16, 24, or 32-bit systems to store numbers in a relatively small range. To store numbers as large as 1035 would require some 128 bits just for the integer part of the number. Another 128 bits might be allocated to store the fractional part of the number. This is undesirable in several respects. The accuracy (discussed later) of 76 decimal digits afforded by the system is larger than most normal requirements; the system would use too many storage units (32 bytes); arithmetic operations on such units would be slow, and could still incur unacceptable rounding and truncation errors. A practical solution to storing numbers which may vary over a large range of noninteger values is to store the number as two components: a fractional part, always justified in a particular manner, and an exponent which describes the file:///H|/SOS%2004-06/FP%20Numbers.htm (6 of 16)9/22/2005 7:46:42 PM

FP Numbers

justification of the fractional part (ie. the number of places it should be shifted to the left or right). The number of storage units allocated to the fractional part determines the accuracy of the system. The range of numbers that can be stored is determined by the number of bits allocated to the exponent, but since there is the requirement to store very small and very large numbers in the same system, the maximum value possible by the exponent is split into two, the upper half storing numbers greater than or equal to the smallest value of the normalised fraction(ie. exponents that shift the fraction to the left)

, and the lower half storing numbers less than the smallest value of the normalised fraction (ie. exponents that shift the fraction to the right). Computer systems, especially the PC, have to be all things to all men, and implement their floating-point storage systems in different ways with regard to the assignment of storage units for the fraction and exponent. However, using 32 bits for the entire number is very common, and this can give a generally useful guarantee of range and accuracy. If you know what this is, you can decide whether its implementation is suitable for the numbers used within your work. Two such implementations are described. Note that outboard equipment often deals with numbers in a special range on which limited arithmetic operations will be performed, and can therefore have specialised storage systems to meet the particular processing requirement. In this case, scaled fixed-point systems may be preferable to floating-point systems, as the implementation of arithmetic calculations is far faster than it is in floating-point systems. What Do They Look Like? In the decimal system, the number 326.42 can be represented as: 0.32642 x 103 (103 = 1000) In other words, it becomes a fraction, 0.32642, multiplied by an exponent, 3, to an exponent base, 10. This is the basic form of a floating-point number in a computer, except of course, the components of the number are stored as binary numbers. For numbers less than 1, the exponent would be negative, thus the number 0.00234 can be represented as 0.234 x 10-2 (as 10-2 = 1/100). Two computer implementations are described. Implementation 1 is relatively straightforward, and facilitates an understanding of the theory and implementation of floating-point numbers. It was used by various third-generation computers in the Õ60s and Õ70s. Implementation 2 is defined by IEEE Floating Point Standard 754. This implementation is more complex. It has a Ôhidden 1Õ bit to give the fraction more storage, and the exponent reserves particular (extreme) values to cope with overflow, underflow, true zero and infinity. Single-, double- and extended-precision forms are defined, the latter, perhaps, anticipating 64-bit hardware. MicrosoftÕs Visual Basic, SteinbergÕs Cubase and


FP Numbers

Wavelab , Fortran and C++ use this implementation. Implementation 1 A 32-bit system (four bytes) could have its bits assigned from left to right as follows to store a useful floating-point number: Bit 0 Sign bit (set to 1 if the number is negative) Bits 1 to 7 E, the exponent to base 16, in excess 64 format, of the number Bits 8 to 31 F, the fractional part of the number, always kept in the range 1/16²F, which begins with a binary point followed by a 1 bit when


FP Numbers

normalised in the range 1/2 ² binary fraction < 1. This means that in a normalised binary fraction, the bit to the right of the binary point must always be 1. (For example, 1/2 as a binary number is 0.1, and any number greater than this, but less than 1, will always have this bit set.) As it must always be 1, the standard does not store it (wasted space!), it assumes it to be present with an implied value of 1, and is often called the Ôhidden 1Õ bit. The significand therefore comprises: the hidden 1 bit, with an implied value of 1 the implied binary point the remaining 23 bits of the binary fraction. This arrangement allows 1 + 23 = 24 bits for the full significand. Its value is therefore 1 plus the value of the binary fraction in the remaining 23 bits. To be normalised, the significand lies in the range 1 ² significand < 2, which means that its value must be greater than or equal to 1, and less than 2. In the following examples, the hidden 1 bit is to the left of the binary point. With all the F bits set to 1: 1.11111111111111111111111 is just less than 2 With the F rightmost bit set to 1: 1.00000000000000000000001 is just greater than 1 With all the F bits set to 0: 1.00000000000000000000000 is equal to 1. In the basic notation, numbers are stored in the form 1.F x 2E. For example, the decimal number 24, which as a binary number is 11000, could be represented as 1.1000 x 24. In the IEEE standard, the exponent E is stored in excess 127 format, therefore the actual value of the exponent would be 127 + 4 = 131. To normalise a number in the IEEE standard, we shift it left or right by 1 bit until it is a significand in the range 1 ² significand < 2. It is shifted by 1 bit because the base of the exponent is 2, not 16 as in implementation 1 where each shift was 4 bits. For each of these 1 bit shifts, we add or subtract 1 to or from the exponent excess of 127. Using the earlier example of +12.625, how would this be stored in the IEEE standard? +12.625 = 1100.101 x 20 = 1.100101 x 23 130

(when the exponent excess of 127 is added).

To derive the normalised significand, we shift the binary number three places right, and for each of these shifts we add 1 to the exponent excess obtaining the value 130. Thus the 32 bits will be assigned as follows: Bit 0 0 (+ve)


FP Numbers

Bits 1 to 8 10000010 (130) Hidden 1 bit (Not stored, but with the implied value of 1) Bits 9 to 31 10010100000000000000000 To check the value stored, we have a significand of ( 1 + {1/2 + 0/4 + 0/8 + 1/16 + 0/32 + 1/64}), and an exponent of 2130-127. = (1 + 37/64) x 23 = 101/64 x 8 = 12.625 Using the earlier example of a small number, how would 3/1024 be stored? 3/1024 = 0.0000000011, as a binary fraction. (i.e. 1/512 + 1/1024) To derive the normalised significand, we shift the fraction 9 places left to obtain 1.1, and for each of these shifts, we subtract 1 from the exponent excess. Thus the 32 bits will be assigned as follows: Bit 0 0 (+ve) Bits 1 to 8 01110110 (127 Ð 9 = 118, representing 2-9) Hidden 1 bit (Not stored, but with the implied value of 1) Bits 9 to 31 10000000000000000000000 To check the value of the stored number, we have a significand of 1.1, and an exponent of 2-9. = 3/1024 Reserved Values Of The Exponent.

The exponent absolute binary values 0 and 255 are not permitted for normalised numbers, and are reserved for special occurrences thus: 1. True zero Sign Exponent Fraction

+ or -

0

0

2. Denormalised number 3. Infinity


FP Numbers

+ or 0 -

Non zero bit pattern

+o -

NAN (Not A Number) Denormalised numbers: In computation, problems arise when the result of a calculation is less than the smallest normalised number that can be represented in the system, but which is far enough from zero to be useful in further calculations. Essentially, denormalised numbers expand the range, and gives a gradual underflow in the range 2minimum value to zero. A denormalised number has an exponent of zero, but the fraction is non-zero, and there is no Ôhidden 1Õ to the left of the binary point. The smallest value that can be stored is represented by an exponent of value 2-126 , and a 1 in the rightmost bit of the fraction having the value 2-23. The smallest positive number that can be represented is therefore 2-126 x 2-23 = 2-149, which is approximately 1.4 x 10-45.

+ or -

11111111 Non zero bit pattern

NAN(Not a number): These are values that do not represent a real number, for example, when an operation is not defined, as in the case of the square root of a negative number. Range And Accuracy

The decimal digit accuracy of the single-precision system is the number of bits in the significand divided by 3.32, ie. 24/3.32 = 7 decimal digits. The largest positive number that can be stored is represented by the following bit assignment: Bit 0 0 (+ve) Bits 1 to 8 11111110 (254, representing 2127) Hidden 1 bit (Not stored, but being an implied 1) Bits 9 to 31 11111111111111111111111 (representing almost 2 with the hidden 1 bit) Thus the largest value is approximately 2127 x 21 = 2128, which is approximately 3.4 x 1038. The smallest positive number has already been identified as a denormalised number whose value is 2149, approximately 1.4 x 10-45. Double-precision Format

Operations in double-precision format follow the same principles as those for single-precision format. In this case, the number of bits used to store the number is 64, and both the exponent and significand are extended using the following bit assignments. Bit 0 0 (+ve) Bits 1 to 11 Exponent in excess 1023 format


FP Numbers

Hidden 1 bit (Not stored, but being an implied 1) Bits 12 to 63 Fraction, normalised as before The accuracy in decimal digits of the system is 52/3.32 = 15. The largest positive number that can be stored is represented by the following bit assignment. Bit 0 0 (+ve) Bits 1 to 11 01111111111 (representing 21023.) Hidden 1 bit (Not stored, but being an implied 1) Bits 12 to 63 1ÉÉall 63 bits set to 1 ÉÉ1 (representing almost 2 with the hidden 1 bit.) Thus the largest value is 21023 x 21 = 21024, which is approximately 1.8 x 10308. The smallest positive value is the denormalised number 2-1074 , which is approximately 4.9 x 10-324. Other Formats

The standard also includes Extended and Quadruple formats. The interested reader can find further information about these and the IEEE Standard in general by placing the text, IEEE754, in an Internet search engine. Published in SOS June 2004




Gigging Safely With A PC

In this article:

Shock Treatment Some Practical Tips Rackmounting Drive Protection Using Laptops Live Removable Drives Avoiding Power Problems Anti-vibration Measures Final Thoughts

Gigging Safely With A PC PC Musician Published in SOS June 2004 Print article : Close window

Technique : PC Musician

Increasing numbers of musicians want to gig with their computers — but home PCs are fragile and laptops may not always be powerful or adaptable enough. What are your alternatives, and what measures can you take to protect the centrepiece of your live set? Martin Walker

Given the amount of great music that's emerging from PC-based project studios, it's hardly surprising that musicians want to take their computers on stage too. Gone are the days when we had to rely on DAT or Minidisc backing and play live guitars or keyboards over the top; nowadays many musicians want to perform more creatively on stage with a PC, either mixing a bank of sequenced loops in real time, playing back sequenced songs to be treated while they're triggering soft synth lines from MIDI keyboards, or performing with liveperformance software instruments. So far, so good, but as Craig Anderton pointed out in SOS January 2004, having your live performance totally dependent on a computer is not a secure feeling.

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A rackmounting case like these ones used by (top to bottom) NuSystems, Red Submarine and Millennium Music will protect your PC far better than any desktop or tower case when on the road, and they have lockable front panels to prevent tampering while they're on stage.


Shock Treatment No doubt a few of you do take a desktop or tower PC to occasional gigs, but these really must be treated with kid gloves, as they're not robust enough to survive rough handling. Whether your PC is being taken to a gig in a car, put in a van for longer journeys, or subjected to courier delivery, the two big problems during transit are individual shocks and vibration. For the occasional short car journey, my desktop and tower computers have always been quite safe strapped into a car seat, but for longer journeys (especially when moving house), return them to their original boxes with their properly engineered foam inserts to absorb vibration and shocks. After a long journey, I still remove the side panel of my computers and have a quick check that the PCI cards, RAM sticks, cables and so on are still firmly in place before I switch the computer on. However, you simply can't do this kind of checking every time you arrive at a gig, so I'd strongly advise against regular gigging with a standard desktop or tower PC.

Some Practical Tips See clearly: Everyone has their own way of using software when performing live, but the last thing most people will want to be thinking about is navigating through menus and launching new dialogue windows when strobe lights and dry ice may be involved! Instead, you ideally want to have every window you need on screen all the time, and preferably split across multiple screens to make everything as clear as possible. Take a shortcut: I daresay some brave souls use a mouse live, but you want to minimise the chances of clicking in the wrong place, so keyboard shortcuts seem far safer to me. The best solution must be performing with one or more external MIDI controllers having dedicated switches, knobs, and faders — plus, of course, a normal MIDI keyboard. Latency Live: While we're on the subject of safe use live, don't be tempted to run your audio peripheral at anything lower than a rock-solid buffer size — it might be tempting to set a lower latency to make your software feel more responsive, but if your soundcard starts clicking and popping through the PA you'll wish you hadn't, and if your sequencer stops altogether due to a CPU 'maxout' the stage lights should ensure that the audience sees your red face!

Rackmounting So if it's inadvisable to take a standard PC on the road, how about buying or building a PC in a 19-inch rack case, and bolting it into a rack, with hardware synths, effect units, and so on? Many musicians do this, and most of the specialist music retailers, including Academy Of Sound, Carillon, Digital Systems, Digital Village, Millennium, and



Red Submarine, offer a 4U- or 5U-high rackmounting system option with their PCs. Compared with most floor- or desk-mounting PC cases, industrial rackmount cases have various things in their favour. They tend to be far more rugged, generally with 2mm-thick steel bodywork that will withstand accidental impacts from other objects (the Carillon rack PC's front panel is particularly sturdy, being die-cast in aluminium alloy), and they also tend to have reinforcing bars to prevent PCI cards becoming dislodged in transit. If your PC is bolted into a heavy rack with other gear, it's also far more difficult for a casual thief to walk off with, especially if you use security bolts. Do bear in mind, though, that the final weight of a rackmount PC will exceed that of most power amplifiers, so not only will its front panel need bolting to the rack, it will require further support at the rear.

Fitting your hard drives into removable 'caddies', as seen in this system from Red Submarine, means that you can remove these most fragile PC components from your PC before taking it on the road.

Most rack cases have front fans, with dust filters to prevent cigarette smoke causing internal damage (another bonus in a live situation), and many have lockable front doors to prevent stealing. In some cases you won't even be able to switch the PC on or off without opening the door, which is ideal for gig security, although for the same reason you should keep the keys in a very safe place, and never leave them inserted when the PC is unattended — that's just asking for trouble! The cons for the typical rack case are higher cost (a moulded PC case costs about £30, an aluminimum MIDI tower about £70 and a rackmount case £150200), and slightly less expansion potential for further drives than in a MIDI tower or desktop case — rackmount case dimensions tend to be smaller. Moreover, for the live user a rackmount case doesn't automatically guarantee greater robustness. Yes, the case itself is stronger, and your PCI cards may be strapped down more securely, but the other internal components are still prone to shocks and vibration. (The Carillon's rack ears are backed with vibration-absorbing rubber gaskets, which will certainly help.) The PC components inside a rackmount simply aren't all that robust, and may not survive the shock of being accidentally dropped a few feet by a roadie, whereas I've known racks of MIDI synths and hardware effects happily carry on even after being dropped down a flight of stairs. Nevertheless, the dealers I spoke to confirmed that many customers do gig with their rackmount PCs, for convenience and security, but admitted that they're not ideal for regular touring unless enclosed in a dedicated flightcase with a foam lining designed to absorb knocks. Unfortunately, such cases aren't cheap, especially custom-built ones, so many musicians cross their fingers and carry on without. Having given all these warnings, there are ways to protect the most sensitive PC components, so let's look at some DIY solutions to safeguard the mobile PC.



Drive Protection The most vulnerable parts of any PC are its hard drives, which can be damaged by excessive vibration and sudden mechanical shock. This can result in them developing bad sectors that can't be read or written to — then these need to be specially marked as such by a utility such as Windows 98's Scandisk or Windows XP's Check Disk, to avoid these sectors being used again. After a bad shock the drive may not even spin up again at all. Drives can withstand much larger shocks when inactive than when powered up, as shown in the 'Environmental' section of spec sheets. A typical figure for 'non-operating' shock is 350G for 2ms, whereas operating shock might be just 60G for 2ms, equivalent to a drop of just a few inches onto a hard surface. Vibration levels of around 1G RMS can often be tolerated when the computer is powered down, but only around 0.6G when spinning.

Always fit a good-quality power supply to any PC that's on the move — cheap models are notoriously unreliable. This high quality SilenX PSU has had its capacitors treated to prevent vibration problems.

Molex SilentDrive sleeves from QuietPC (www.quietpc.com) are primarily designed to absorb acoustic noise, but their thin decoupling foam lining is also quite effective as a shockmount, and absorbs external vibration during transit. They are widely used by specialist music retailers for silencing, installing into a 5.25-inch drive bay. You then mount your 3.5-inch hard drive inside them. However, some drives run too hot to be enclosed in this way, and the sleeves don't suit the new SATA (Serial ATA) drives. Another solution might be the similarly-priced NoVibes cages from NoiseMagic (available in the UK from www. chillblast.co.uk), which can be even more effective at absorbing vibration. Again, they mount in a 5.25-inch drive bay, but suspend the hard drive in a cradle of three high-tension rubber bands. You won't be concerned about your drives overheating in one of these cages, but I (and various dealers I've spoken to) have always been a little wary of drives jumping out of the cradle altogether during transit.



Using Laptops Live Plenty of musicians seem to be taking rackmount PCs on the road, so that they can benefit from the fastest processors and multiple large hard drives, and the ability to rackmount their audio breakout boxes, MIDI synths and hardware effects alongside the PC. However, many others have already found a somewhat more obvious solution for the smaller gig: using a PC laptop. You may not be able to buy a laptop as fast as the fastest standard PCs, but for most purposes they are perfectly adequate for live use, as well as being far easier to transport. You can take them as hand luggage on a tour (even abroad) and a suitable padded case is far less expensive than buying a flightcase for a standard PC. Desktop-replacement laptops are popular live because the fastest models offer more power than Centrinos, as well as having larger If you're thinking of using your PC laptop live, 17-inch screens that are easier to make sure you invest in a security cable like see on stage. And, of course, the the one advertised here, to lock it to a heavy noise of their cooling fans won't stand. worry you in a live environment. However, a Centrino model will be perfectly adequate for many gigs, if you don't want to run loads of plug-ins and soft synths simultaneously. Because laptops are routinely moved about, the design of their hard drives may already have benefited from extra attention paid to shock and vibration protection. For instance, the 40Gb Seagate Momentus drive in my Centrino laptop is not only extremely quiet, but also incorporates G-Force Protection for non-operating shock protection and 'QuietStep ramp load technology' for operating shock protection. Together they provide operating protection against shocks of up to 225G for 2ms (typical 3.5-inch hard drives only offer around 50G to 60G) and a staggering 800G for 2ms when switched off. Still more reassuring is that laptops are fairly immune to mains problems, since even if there's a complete blackout they will automatically carry on with their integral battery power for perhaps an hour in the case of a desktop replacement model and several in the case of a Centrino. Do make sure you've chosen 'Always On' for your Windows power scheme, though, to avoid a sudden drop in CPU performance when the battery kicks in. Placing your laptop on a foam pad is still a wise move to avoid vibration problems, and a dedicated and rugged stand should help it last the course, too. For security reasons it's also well worth investigating some form of anti-theft device — nearly all laptops feature a Security Cable Connector on their back panel to thwart opportunists, and to this you can connect a galvanised steel cable that you loop around a heavy and preferably immovable object and then lock with a key or combination. Some models can also be used to protect



external drives at the same time, while other devices can fix your laptop directly to a stand or desk.

Removable Drives The ultimate solution to the problem of drive protection during transit has to be removable caddies, also known as drive drawers or racks. You bolt the outer docking station into a 5.25-inch drive bay, just like a CD-ROM drive, plug your PC's internal IDE and PSU cables into its rear connectors, then mount your drive into the caddy itself and connect its IDE and PSU connectors to the internal caddy cables. The caddy is then pushed into the docking bay, and you turn the supplied key in the caddy's lock before switching on your PC. Quite a few musicians now use these caddies, primarily as a way to easily move from one audio project to another, by unplugging one drive and replacing it with another. Buying multiple drives and using a new one per project is also an ideal way for studio owners to cope with clients returning to remix some tracks, especially now that drives are relatively cheap. For live use, caddies mean that after powering down your PC you can simply remove its most vulnerable components before transportation and place them in a well-padded case. At the gig, as long as you keep your drives with you until the soundcheck, you have the reassurance that if your PC gets stolen your data is still intact (although, as always, you should still have it backed up elsewhere for safety). You can already buy PC systems featuring drive caddies from specialist music retailers, including Carillon, Philip Rees and Red Submarine. It's also possible to purchase caddies to fit to other PCs. It's well worth buying a high-quality caddy. The cheaper ones are made of ABS plastic and generally support IDE drives up to 7200rpm spin speed and ATA133 standard, while more expensive aluminium ones are stronger and designed for drives up to 15,000rpm. The majority are for parallel IDE drives, but you can now buy models suitable for SATA drives as well. Prices range from around £30 to £45. Incidentally, anyone who has tried in the past to buy one docking station but multiple caddies to use with it (they are nearly always sold as a pair, so you end up having to pay more for parts you don't want) will be pleased to hear that as a result of my enquiries Red Submarine are now offering this option for Lian-Li's RH-32 model. By the time you read this you'll be able to buy the complete unit from them for £27.95, and extra caddies for £17.95 each. Since you're (in effect) placing a hard drive in a sleeve, again beware of overheating. Some caddies have cooling fans for this reason, and that might make them less suitable in the studio where you want minimum noise. This won't



matter for live use, of course.

Avoiding Power Problems We all know that, despite the great improvements in the launch speed of Windows XP compared with its predecessors, it still takes a couple of minutes to boot up a PC from cold. Then there's the time it takes to launch the music applications themselves, and load up the required song plus plug-ins, soft synths, and so on. One way to avoid some of this delay during your soundcheck is to use the Hibernate feature of modern PCs to save the entire contents of your system RAM onto your hard drive, so that when you return to the desktop it will be exactly as you left it, complete with all running applications. Unfortunately, some PCs don't recover properly from Hibernation, and some soundcards also seem to have problems with it, so you should thoroughly test out this idea before relying on it live — if anything goes wrong, you'll have to reboot anyway. Having to reboot the PC is the most dreaded scenario for a live musician, as the audience is unlikely to be very impressed with minutes of silence while you reload everything — and, of course, unless you're a really laid-back performer, a computer crash could seriously unnerve you. Obviously, you should make sure your PC is as reliable in everyday use as possible, by installing latest driver versions and generally keeping it 'clean' and virus free. Even if you do this, though, there's another potential cause of problems on stage. If you get a mains 'brown-out' (a drop in mains voltage normally signalled by the lights dimming) or a blackout (when the power disappears altogether, either momentarily or for a longer period), your rackmount PC may crash, either locking it up and requiring you to reboot, or causing it to spontaneously reboot. Unless you really like living dangerously, the answer is to buy a UPS (Uninterruptible Power Supply), which can filter out mains spikes and even survive several minutes of complete mains blackout. Essentially, a UPS contains batteries that are normally trickle-charged by the mains supply, and when the mains fails they kick in to power an 'inverter' circuit that generates the normal AC mains voltage for as long as the battery power lasts. In most cases, the UPS will only need to supply a few hundred watts for a couple of minutes. Some UPS units run their inverters continuously, to generate a clean interference-free 'mains' signal. In a live situation, this will also provide greater protection from incoming spikes, such as those caused by some stage lighting systems. However, most PC power supplies do have some interference filtering built-in, although it may be worth adding a filtered distribution board to provide a little more protection. Finally, one of the most annoying power problems has to be unexpected hum or background noises from your CPU/mouse/drive/graphics card suddenly being heard in the PA. Even if your PC audio sounds perfect in the studio it's worth buying a DI (Direct Injection) box to cure such ground-loop problems live. Suitable stereo models, such as Behringer's DI20, are available for around £20.



Anti-vibration Measures Having dealt with the PC's most shock-prone components, let's turn to the smaller details and see what else can prevent transit damage. During regular journeys, vibration may eventually cause normal screws to work loose. The most potentially troublesome components are PCI and AGP expansion cards: if these become unseated the result may be obscure problems ranging from intermittent crashes to an inability to boot up at all. However, hard drives, cooling fans, PSUs and even the motherboard will most likely be held in place by screws. Even if your PC rackmount case has a reinforcing bar to hold expansion cards in place it may be worth adding anti-vibration measures if the PC goes on regular journeys. The simplest and cheapest measure is a dab of paint between the screw and whatever it's securing, which should stop it rotating. Alternatively, it may be worth trying the rubber grommets sold for fan and drive mounting. Be a little careful using these with fixed hard drives, however: they may reduce the amount of vibrational noise from the drive that enters the rest of the chassis, but they may also allow the drive to bounce about more during transit, increasing its quota of shocks. I/O connectors can also come loose during transit, but this can largely be prevented by careful attention to wiring up the loom with ties to various points on the chassis to prevent the cables from pulling on the connectors. You should also tie down any unused PSU connectors, to prevent their exposed pins accidentally touching other components. Other large internal components, such as FanMate controllers, should also be tied down to the chassis if possible. RAM sticks can't be tied down, but you could try putting rubber bands around their clips to prevent them popping out of place. Modern motherboards don't tend to have large unsupported components (such as capacitors and inductors) on them, but power supplies do, and cheap ones are notorious for breaking down anyway, even before one considers problems due to vibration fatigue of component leads. When I recently reviewed a new SilenX PSU I was pleased to see that its internal components had been immobilised with rubber or potting compound, or glued to other components nearby. These measures not only made it quieter, but will also ensure that it's less prone to 'travel sickness'.

Final Thoughts Even if your computer is rackmounted, once it's in position on stage it's worth trying to prevent vibration — from a PA, nearby drumkit, over-active vocalist or dancer — from causing problems. As explained previously, this can permanently damage hard drives. If you find yourself on a 'bouncy' stage, try to choose a setup position that minimises the problem, even if this means you're not centre



stage! Again, having your PC mounted in a properly-designed flightcase will help. It may pay you to set up two drives in a RAID (Redundant Array of Inexpensive Disks) configuration, not to 'stripe' them for faster performance, but to mirror your data so that if one drive fails the other can carry on. Also, avoid placing your PC near to any large speaker cabinets that contain strong magnets. Hard drives and magnets don't mix! By the way, don't rule out taking a Minidisc or CD player with you as a backup: if the worst happens and you need to reboot your PC in the middle of a set, at least the music can carry on while you sort out the problem. Some musicians find Minidisc players more reliable than CD players on stage, as they tend to have larger buffers that will survive longer periods of vibration without the audio stream being interrupted. Both CD and Minidisc players will benefit from being placed on a foam pad to isolate them from vibration. Published in SOS June 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



Hyperthreading & Spring-cleaning

In this article:

Tiny Tip: MIDI Latency Testing Clean Sweep PC Snippets Hyperthreading Tests

Hyperthreading & Spring-cleaning PC Notes Published in SOS June 2004 Print article : Close window

Technique : PC Notes

This month, we find out whether Hyperthreading is hyper-helpful to the musician and discover some new freeware. First, though, it's time to spring-clean that Windows Registry... Martin Walker

The December 2003 issue of SOS contained a feature of mine about installing a new PC motherboard, and since I wrote it I've discovered something else that may well be useful if you do this with your PC at some time. It's a way of cleaning up the Windows XP Registry (and those of Windows NT and 2000, for that matter) after the motherboard change-over, to remove references to hardware that no longer exists. Even if you're not considering a new motherboard, this same procedure will help you clean out references to hardware that's been unplugged, such as old soundcards, graphics cards, hard drives, monitors and so on, plus any installed USB devices that aren't currently plugged into any USB ports. You won't normally spot these references in Device Manager, or even when performing the Safe Mode Cleanup that I described in PC Notes June 2000, and although your PC may be working well despite these invalid references, there's still a chance that at some stage you may try to install a new hardware item that has similar properties. The old driver files could then spring into action and cause problems. So, for the cleanest and most reliable PC, it's safest to remove them all. Here's how.

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Tiny Tip: MIDI Latency Testing I recently answered an SOS forum query about MIDI timing irregularities, in which the poster's soundcard buffer setting indicated a latency of under 10ms, but after a few minutes the time lag seemed to increase, making his soft synths unplayable. I explained that since this latency was connected to the buffer size it shouldn't change over time, but came up with a simple way for him to test out his MIDI timing. It worked a treat, confirming an unworkable 100ms latency that seemed related to a fault in his soundcard's driver. The test is really easy to try. First, connect a standard MIDI lead between the chosen MIDI In and Out that you want to test, then set up a MIDI track in your sequencer. Paste in regular short notes every second or so for as long as you want to run the test, and then play back this 'Send' track, along with the rest of a typical song, while recording the MIDI input onto another 'Receive' track. When you get to the end of the song, stop and compare the Send and Receive MIDI tracks to see if there are any timing differences between them, to confirm what you're hearing. You'll probably find the notes in the Receive track slightly behind the Send one, but any suspected timing irregularities should show up as one or more Received notes visibly further behind. This test essentially measures MIDI In and Out driver performance, exactly like the Miditest utility I mentioned in last month's column. Although you don't get the same easy-to-read figures that you get when using Miditest, you do get the advantage of measuring with the stresses of audio tracks and soft synths running in a real-world song — which, after all, is when most timing problems occur.

Clean Sweep Normally, Device Manager only shows devices that are currently connected, even if you use its 'Show hidden devices' option. What we're going to do is to force the redundant devices to appear inside Device Manager, so we can delete them once and for all. First, start the Windows Command Line Prompt, by selecting the Run option in the Start menu, entering 'cmd.exe' into the text dialogue and pressing return. This will launch a DOS window into which you type the following commands, pressing return at the end of each line (see screen, bottom of page): set devmgr_show_nonpresent_devices=1 start devmgmt.msc Device Manager will automatically appear after the second command, and now if you go into its View menu and select 'Show hidden devices' you should see quite



a few greyed-out items referring to missing hardware. After a motherboard upgrade these will usually include the old processor type (in my case, a Pentium III that was replaced by a Pentium 4), as well as the motherboard chipset, IDE controllers, USB Host Controller, IEEE1394 host controller, network adaptor, and so on. You could well also have duplicate entries for your various hard drives, optical drives and display adaptors. It's probably just as well to uninstall any 'Unknown devices' too, as well as any greyed-out items titled 'Generic volume' in the Storage volumes section. The latter are likely to be due to partition changes after using utilities such as Partition Magic.

After a major hardware change, such as upgrading a motherboard, or after removing various PCI and AGP expansion cards, it's best to clear out the hidden entries in Device Manager.

However, you should leave any devices in the section labelled 'Non-Plug and Play Drivers' and any Microsoft filters in the 'Sound, video, and game controllers' section. Windows also treats each USB port individually, so for a system with six ports you may see a valid attached device and up to five ghost devices (one for each of the ports), which you can also ignore. Don't delete any greyed-out USB device that you're still using but isn't currently plugged in; if you do, you'll need to re-install its drivers the next time you want to use it. Otherwise you can uninstall each item in turn that refers to a known piece of hardware that's no longer connected, and when you next reboot you'll have a cleaner machine.

PC Snippets Crakbone is a 2.5Mb Windows utility that turns text into music. You enter the text (typically a single word), and then Crakbone takes the ASCII values for each letter and converts them into a tune played by a small ensemble of instruments. Simple but fun! www.azzer.com Capable of somewhat more sophisticated results, Nicolas Fournel's freeware AudioPaint generates sound from pictures, turning each pixel from a JPEG, GIF or BMP file into frequency, amplitude and pan information. The canvas in effect becomes a frequency/time grid, while the colours determine stereo positioning, so AudioPaint is a sort of additive synth. Just the job for turning paintings, local maps or deep-space images into futuristic audio landscapes. www.nicolasfournel.com Peersynth is the brainchild of Dr Joerg Stelkens


Do you fancy a caring, sharing creative session with other musicians? Peersynth lets you jam with other musicians live


online. Registered users can (whose crusherX-Live! granular synth already has consult the Event Calendar to a healthy following) and is described as a "multifind out when the next jam is user Internet synthesizer". Running under happening. Windows 9X, ME, 2000 and XP, the 10Mb shareware stand-alone synth can generate extreme sounds with just a few mouse clicks or via a MIDI controller or sequencer, using either a DCO (Digitally Controlled Oscillator) or sampler, with its own database to store and sort WAV and MP3 files. Once you're online and running Peersynth, other Peer users can be seen and heard live via pop-up instruments, while your performance appears on their screens in the same way. The sounds from each Peersynth are generated locally, to minimise bandwidth requirements (300 samples are included in the download) and if another Peer is using a different sample this will be distributed in the background while the others carry on playing. You can also capture a combined live performance as a local MP3 file and distribute that to others during the session. Registration costs just 29 Euros; registered users can consult the Event Calendar to join future online sessions. www.peersynth.de

Hyperthreading Tests Back in PC Notes August 2003 I explained the ins and outs of Intel's Hyperthreading technology, which lets Xeon and Pentium 4C processors appear to Windows XP Home and Professional or Linux 2.4x as two 'virtual' processors instead of one physical one. They each share the various internal 'sub-units', including the all-important floating-point unit, but can run two separate processing 'threads' simultaneously. I've had HT disabled on my own Pentium 4C 2.8GHz machine until now, partly because I still run Windows 98SE partitions alongside my XP ones. Since 98SE doesn't recognise HT technology it may cause problems to leave it enabled in the BIOS. In addition, as I reported in PC Notes January 2004, GigaStudio 2.53, which I use, won't run with HT enabled either. I recently had a Dual Xeon PC from Red Submarine to review for SOS. Since this has true multiple processors, I was interested to see how the virtual processors of an HT-enabled machine compared to it. I used my own Pentium 4C 2.8GHz PC, with Hyperthreading enabled, as the test bench and began my tests by running Waves plug-ins, inside Wavelab 4.01a, on this PC. The plug-ins included the Renaissance and Trueverb reverbs, C1 compressor/gate, C4 multiband parametric processor, and Renaissance EQ running six bands. For each one I made measurements with and without Hyperthreading enabled, and apart from experimental error the results were identical. This confirmed what I had expected. Hyperthreading makes no difference to single apps that are not aware of it, nor those that run a single stereo audio stream rather than multiple audio tracks, each with their own complement of plugins or soft synths. (In the latter case multi-processing is more likely to be of benefit). However, you may well notice performance improvements when running several such applications simultaneously.



Next, I ran Steinberg's 'Five Towers' Performance Test, the 'Five Towers' version 2.0 test, which has higher CPU overheads, and Steinberg's Cubase SX 2.0.1 'Heaven And Hell' demo. This provided more interesting results, because SX 2 relies on running multiple 'threads' and has been optimised for HT. I measured all three songs in 'Stop' (only plug-ins running) and 'Play' (with soft synths as well), at both 23ms latency and 3ms latency, to isolate the effects of interrupt overheads. To explain this a little further, as you drop latency below 23ms CPU overhead rises, simply due to the massive number of interrupts per second. Within the bounds of experimental error, most results for each of the three songs were almost identical, except for the 3ms 'Play' values, which showed a significant improvement of between 10 and 12 percent. Although these figures aren't as high as some I've seen, they're still not to be sneezed at and they demonstrate that Hyperthreading works best with Cubase SX 2 just where musicians already need a helping hand — running soft synths with low latency. Cakewalk's Sonar 3.1 also has a new multi-threaded engine that works with Hyperthreading as well as true multi-processor PCs. Cakewalk's Ron Kuper has recently published a 'White Paper' on its performance, and while it provides significant improvements with a true dual-processor PC (see my review of Red Sub's Dual Xeon PC starting on page 90 for more details), it doesn't do so well with Hyperthreading. Cakewalk themselves measured a six to seven percent improvement on a Pentium 4 3.2GHz HT system when they switched on the new engine, and I measured a similar four to five percent improvement on my own P4 2.8GHz machine. However, after disabling Hyperthreading altogether in the BIOS and measuring Sonar 3.1 overhead again, the results were less than two percent better than those with HT on and Sonar's new engine enabled. So overall Sonar 3.1 doesn't seem to noticeably benefit from HT being enabled. Unfortunately, there are further stings in the tail. Despite HT having been available since 2002, there are still audio applications that disagree with it, and if you're currently running one of these you won't be able to activate HT in your BIOS until an update appears. As previously mentioned, for me the most serious is GigaStudio, but this should be solved when version 3.0 is released in June. More seriously, Digidesign state specifically on their web site that HT must be disabled if you're running any of their hardware under Windows XP. I've come across other users who have problems with NI's Kontakt, and Cakewalk have recently posted a warning on their web site that some Pace-protected software can cause a crash or complete lock-up with HT enabled. Antares plugins are specifically mentioned. Waves plug-ins are apparently happy with HT, which fits in with my own experience. Published in SOS June 2004






More Creative Synthesis with Delays

In this article:

Creating Chorused Sounds Without Chorus A Basic Chorus Effect Improved Chorus Effects Stereo Chorus Some Real Examples Epilogue

More Creative Synthesis with Delays Synth Secrets Published in SOS June 2004 Print article : Close window

Technique : Synthesis

In the penultimate instalment of this long-running series, we delve deeper into what can be achieved with just a few delays and some creative routing... Gordon Reid

Here we are, at the far end of our synthesizer's signal path. We've generated the waveforms, created multiple signals, filtered and sculpted them, applied modulation, mixed the results, and... well, it all sounds a bit thin, doesn't it? Despite the techniques we've employed, the results totally lack the depth of a nine-foot six-inch Bösendorfer, or a four-manual cathedral organ. Yet, if you think back to the early 1960s, there was — apart from a choir or the string section of an orchestra — little that was musically lush, and the electronic sounds that we now take for granted were still in the future. If you wanted to record the semblance of multiple instruments playing at once, you either paid a few dozen instrumentalists to do their thing simultaneously, or you bought a MkII Mellotron. Either way, the costs were crippling. Things began to change in the mid-1960s, when the affordable 'chorused' organ was born. Consider the way in which a cheap electric organ creates its sound. In general, the outputs of high-frequency oscillators are 'divided down' by integer factors to create the correct pitches for all the notes of the top octave of the keyboard, and these are then further divided by factors of two to generate each octave beneath. However, organ designers discovered that they could divide the master oscillators in different ways to generate two frequencies for each note that were almost, but not exactly the same. These small discrepancies — which were not even identical from note to note — were not dissimilar to the differences between the pitches of two pipes tuned to the same pitch, or between the three strings that comprise a note on the aforementioned Bösendorfer. Consequently, manufacturers began producing electric organs that generated a primitive chorus effect using dual sets of dividers. More sophisticated designs incorporated two independent sets of

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master oscillators, each with a set of frequency dividers, and some even offered a global detune for the second set (see Figure 1, on the previous page). Of course, there was also the principle of using multiple electronic sound sources playing in unison to recreate the effect of multiple physical sound sources playing in unison. So, long before the appearance of the modern chorus effect in 1975, keyboards were using multiple oscillators per note to thicken up what would otherwise have been bland and uninteresting patches. Of these, the most sophisticated was the prototype of Ken Freeman's String Symphoniser. This used three banks of detuned oscillators, applying vibrato to each to create a rich chorus effect. To this day, synth programmers often use detune, pulse-width modulation and frequency modulation to obtain richer timbres than would otherwise be possible (see the box on the previous page). Nevertheless, this is not what we now mean when we use the word 'chorus' and, of course, it can't be applied to externally generated signals such as a human voice or a note played on a guitar. What's more, despite the complexity of detune/ vibrato/PWM programming, timbres generated in this fashion still don't sound as lush as even the cheapest and cheesiest 'string synths'. On the surface, this is rather surprising. In contrast to a sophisticated multiple-oscillator-per-note synth, the initial waveform produced within a string synth is almost always a single, 'divide-down' sawtooth which, at the best of times, sounds weedy and uninspiring. Yet, passed through the instrument's internal chorus/ensemble unit, it sounds animated, lush and full of body. So... what is chorus, and why does it sound so good? To answer that, and before Synth Secrets departs through the output socket of history, it's time that we took a look at that most popular and most useful of all keyboard effects: the chorus/ensemble.

Creating Chorused Sounds Without Chorus If a polyphonic synth has dual oscillators (or better still, three) per voice, it will be capable of creating thick, quasi-chorused sounds, even without a chorus unit. To create these sounds, you must first select the sawtooth wave option on one of the oscillators (because this has the correct harmonic content for a string sound) and the pulse wave on the other. Secondly, you must detune one oscillator against the other to create a 'detuned' sound. Next, you must add pulse-width modulation (or PWM) to the pulse wave.

Figure A: Thickening the sound by modulating the duty cycle of a pulse wave.

Figure B: Adding another signal to further thicken the sound.

As I showed in the March 2003



instalment of this series (see www. soundonsound.com/sos/mar03/ articles/synthsecrets47.asp) pulsewidth modulation generates two Figure C: Creating a lush sound using two 'virtual' signals, with one being oscillators per voice. pitch modulated with respect to the other (see Figure A, right). PWM alone creates a 'chorused' timbre, but if you detune the sawtooth wave with respect to the pulse-width modulated wave, there are, in effect, three pitches present in the output, and this further thickens the sound (see Figure B). Finally, adding vibrato to the sawtooth wave will complicate the relationships between the three pitches, especially if the synth can modulate the vibrato and PWM at different rates (see Figure C). The key here is to ensure that there is so much activity that your ear becomes unable to recognise the limited number of pitches present. When programmed carefully, the sounds produced by this method can be superb, as evidenced by the remarkable 'ensemble' patches produced by the Prophet 5, OBX and, in particular, the Roland Jupiter 8.

A Basic Chorus Effect The secret to this effect is fooling the ear into thinking that it is hearing multiple performances of the same note when it is not. This may sound tricky, but the key to doing so already exists in my explanation of modulated BBD delay lines, which I introduced in March's instalment of this series (see www.soundonsound.com/ sos/mar04/articles/synthsecrets.htm). In short, if you don't have access to multiple, closely related timbres and pitches, why not split a single signal into multiple paths, apply pitch modulation to one or more of these, and then remix them? It's a simple idea, and it works beautifully. Figure 2 (left) shows the structure of a modulated delay line. You can present any signal to the input, whereupon it will be low-pass filtered to eliminate aliasing, sliced into samples, and then passed through the line before being reconstructed at the far end. This, by the way, is as true for a digital delay as it is for an analogue BBD. The speed at which the samples travel down the line, and their precise temporal relationships (ie. how far apart they are spaced) are determined by the clock generator and the oscillator modulating its speed. I hope that it is obvious that, if the clock is running faster when a bunch of samples reach the end of the line than it was when they entered, the samples will be closer together, so the pitch will be higher. Conversely, if the clock is running slower when those samples reach the end of the line, the samples will be further apart, so the pitch will be lower. Clearly, this allows us to modulate the pitch of the signal, and if the LFO in Figure 2 were generating a sine wave, the output from this diagram would exhibit a pronounced 'wow' effect, like a vinyl record with the hole punched in the wrong place. Now, let's add a second signal path to Figure 2, which allows the unaffected file:///H|/SOS%2004-06/More%20Creative%20Synthesis%20with%20Delays.htm (3 of 10)9/22/2005 7:47:16 PM


signal to pass to the output, as shown in Figure 3 above (from which, for clarity, I've omitted the anti-aliasing and reconstruction filters). We now have the situation where the modulated signal is sometimes at a higher pitch than the 'straight-through' signal, sometimes at a lower pitch and, on two occasions in each cycle, at the same pitch. Figures 4 and 5 (below) demonstrate this, showing how the delay in the upper signal path changes in time, and how this affects the pitch relationships between the upper and lower signal paths. We can patch Figure 3 very easily using just four modules from a modular synth: a multiple to split the incoming signal into two paths; an LFO to modulate the rate of the Echo unit (which combines the delay line and clock generator in a single module); and a Mixer to recombine the two audio signals. In Figure 6 (left), I have shown the original signal and output in blue, the modulation path in green, and the modulation signal in red, just to make things clearer. Of course, this patch will produce nothing like the desired effect unless we choose our parameters sensibly. Firstly, you'll find that the barest minimum of modulation is needed. If you can't obtain a low enough modulation level from the Level knob on the LFO, place an attenuator in the modulation signal path to reduce the amplitude even further. In contrast, you have a wide choice of modulation rates. A slow sweep at a fraction of 1Hz will provide a gentle chorus, while a faster rate — say, 5Hz to 7Hz — will result in a more typical 'synth' ensemble. The other vital factor is the delay time. Set this to be too long, and you'll hear a distinct delay. Set it too short, and you'll obtain a version of another effect: flanging. But get it right

Figure 1: Creating a chorus effect by 'dividing down' master oscillators.

Figure 2: A modulated delay line.

Figure 3: Adding the unaffected signal to the pitch-modulated signal.

Figure 4: A simple delay modulation.



— somewhere in the range 10ms to 50ms, as your taste dictates — and then mix the two signal paths in equal measure, and you'll obtain a serviceable chorus, reminiscent of the cheapest '70s string synths. Hang on... the cheapest '70s string synths? The individual modules in this patch (or an equivalent software modular synthesis application) could cost a couple of hundred quid, so you've a right to expect something a bit better. What's going on? Unfortunately for us, our ears — which evolved to locate the rumbling tummy of a sabre-toothed tiger at 500 paces — are not fooled by the two signals generated and mixed in Figures 3 to 6, so this implementation of chorus is not perceived as particularly deep, nor indeed as particularly lush. In effect, it's the equivalent of hearing just two singers, or just two violinists, when what you're after is the choir and orchestra performing Mahler's Ninth in the Royal Albert Hall. So we overcome the problem by adding more signal paths to the existing scheme, and by modulating all of them differently. I have shown an efficient way to do this in Figure 7 (above left), which shows that we can use a single LFO to modulate each of the delays, provided that the phase of each instance of the LFO waveform is shifted by some amount. Without these shifts, the three audio paths would be modulated identically, and we would create vibrato, nothing more. If the three paths are modulated at relative phases of 0 degrees, 120 degrees and 240 degrees, we obtain the pitch relationships shown in Figure 8 (left). As you can imagine, this is a far more complex sound, and the relationships between the three modulated signals provide a thicker and warmer chorus effect.

Figure 5: The pitch-shifts resulting from the modulation in Figure 4.

Figure 6: A simple, two-path chorus unit.

Figure 7: Using a single LFO to modulate three delay lines.

Figure 8: The frequency modulations of the three signal paths in Figure 7.



Improved Chorus Effects The chorus described in Figures 7 and 8 is a classic '70s design, and was used in numerous string ensemble keyboards, but it still does not have the richness and depth that we have come to associate with the best of such effects. This is because synthesizer designers continued to dream up better ways to modulate the input signal, the first of which was to modulate each delay line with not one LFO, but two. This was conceptually simple, although debate raged over the speed and depth that creates the most pleasing effect. Many manufacturers opted for two very different speeds — one of the order 0.5Hz to 0.7Hz, with a second closer to 6Hz to 7Hz, but with no discerned integer relationship between the two, so that the modulation didn't repeat for a long time. Figure 9 (left) shows two such waves, and the resulting modulation signal. As you can appreciate, it's going to be more difficult for Ugg the Caveman to perceive this as a simple, repeating waveform. Now consider Figure 10 (below). This shows the input split into four paths — the original signal plus three delayed versions of it. Each of the six LFOs in the patch can have a different modulation rate, and each of the mixers in the modulation paths can be designed so that each of the LFOs contributes a different depth. The result is a lush, complex swirl of sound that is forever evolving, and which adds movement and texture to even the most basic of initial waveforms.

Figure 9: Creating a more complex modulation from two sine waves.

Figure 10: A four-path chorus unit.

Figure 11: Reconfiguring the LFOs to produce a modulation signal with vibrato.

Figure 12: Modulating the speed of the modulator.

Another variation on this approach



rearranges the LFOs so that one of each pair modulates the frequency of the other. This arrangement, which I've shown in Figure 11 (bottom) produces the waveform in Figure 12 (on the next page), creating yet another form of pitch modulation and, therefore, a subtly different chorus. One could go even further, for example using another LFO to modulate the depth of the modulating waveform as well as its frequency and, if cost were no object, you could keep slinging LFOs, mixers and, where necessary, VCAs at the problem to create the most complex modulations imaginable. You can even use a random waveform as a modulator, which goes some way towards imitating the genuine pitch instabilities of human singers and players. As you might imagine, circuits such as Figure 10 can be expensive to build, and although this design produces a superb ensemble sound, it may not be economical. To overcome this, many manufacturers combined the ideas set out in Figures 7 and 10, employing a trick that fools the ear into believing that it's hearing multiple, complex modulations, when in fact only one is present (see Figure 13, on the next page). This involves the use of just two LFOs (which cuts costs) and four phase-shifters (which are cheap), and generates three instances of a single complex delay modulation. As before, these are out of phase with one another, typically by 120 degrees, and the result, while not quite as lush as you can obtain using six independent LFOs, is nonetheless gorgeous. This is why this method — or close variations of it — became the standard for almost all the best-loved chorus/ensemble keyboards.

Figure 13: The classic three-phase chorus unit.

Figure 14: The output stage of a triple-path stereo chorus unit.

Figure 15: A 1978 chorus design by Roland Corporation.



Stereo Chorus Nice as monophonic chorus might be (and is), it doesn't make the most of the techniques described above. Indeed, when Roland first designed the chorus/ensemble pedal, they created a CE1 mountain, because nobody bought the things. It wasn't until a handful of players discovered the 'Stereo' output (which sits less than an inch to the right of the Left/Mono output) and directed this to a second amplifier and speaker that the world sat up and started to take notice.

Figure 16: Converting the Roland chorus in Figure 15 into a stereo unit.

As we now appreciate, stereo chorus is the classic synthesizer effect, and it's easily created if you have access to the signals being produced by the multiple delay lines. Figure 14 (bottom) shows a triple-path configuration in Figure 17: The modulations generated by the which the signals generated by the first clock and dividers in Figure 15. and second delays are mixed and sent to the left output, while the signals generated by the second and third delays are mixed and sent to the right. In this scheme, you can leave out the 'straight-through' signal, because the dual inputs to each mixer will be chorusing differently, and — far from contributing to the result — the original might actually damage the impression of width and depth.

Some Real Examples Although some people think that string synths sound much like one another, this is not true. Sure, they all share the same class of sound, they are almost without exception based upon quasi-sawtooth waves generated by organ-style 'dividedown' technology, and if you switch off their chorus/ensemble effects, they are all about as interesting as a bunch of boring things on a very boring day. Nonetheless, there are marked differences in the way that they generate their chorus effects. For example, the Eminent Solina (1974) uses the chorus structure depicted in Figure 11, with two LFOs, one running at around 1Hz and the other at about 6Hz, phase-shifted to produce modulating signals at 0 degrees, 120 degrees and 240 degrees. In contrast, the Roland VP330 (1978) has a thinner string ensemble sound generated by just two delay lines with dual LFOs. The altogether richersounding Korg Polysix synthesizer (1981) incorporates another triple-delay chorus, but dispenses with the phase-shifters and uses a configuration of file:///H|/SOS%2004-06/More%20Creative%20Synthesis%20with%20Delays.htm (8 of 10)9/22/2005 7:47:16 PM


independent LFOs similar to that shown in Figure 10. One of the cleverest of chorus designs was developed by Roland (see Figure 15, on the next page). This uses just a single square wave LFO (generated, would you believe, by one of the flip-flops that I described a couple of months ago) and three frequency-dividers that output modulation signals at the clock frequency Fc, at 1/2 Fc, 1/4 Fc and at 1/8 Fc. These 'square' signals are low-pass filtered to 'round off' the waveforms into approximations of sine waves, and these are in turn used to modulate the clocks driving four BBDs. The outputs of the delay lines are then mixed into a single sound, and emerge in glorious mono. There is no 'straight-through' signal present in the output. The major advantage of this configuration is one of cost; clocks, dividers and simple low-pass filters are cheap. However, there is an additional benefit, which was the true raison d'être of the design: due to the nature and relationships of the four modulating waves (shown in Figure 17, left) the chorus effect is less susceptible to the 'warbling' effect that you hear generated by other low-cost circuits. Roland even suggested ways in which Figure 15 could be improved. For example, you could mix the outputs from the first and third BBDs and from the second and fourth, to create a stereo chorus (see Figure 16, above). The designer also suggested inverting the phase of the second and fourth modulating waves to create an even richer 'spread' of the ensemble effect, and using frequency dividers that used other integer factors of Fc, such as one third or one fifth. Yet another Roland design dispensed with sine-wave modulators and substituted sawtooth waves, eliminating the pitch discontinuities by modulating a set of VCAs that silenced the delay lines while the modulating waves reset to the start of their cycles. Yet another incorporated signal gates that disconnected the modulation waveforms when there was no audio signal present at the input, thus eliminating the characteristic 'swishing' noise that mars many chorus effects. The permutations are almost endless.

Epilogue Despite the obvious benefits of chorus units, BBDs are rather noisy, so many (although not all) 1970s manufacturers treated them as a necessary evil that added interest to cheap, single-oscillator keyboards and synthesizers, but which were not suitable for top-of-the-range instruments. But while the high-brow approach was good in principle, the synthesizers that eschewed chorus and relied on other techniques to create lush sounds were expensive: multiple VCOs per voice, the ability to detune independent banks of oscillators against one another, the ability to modulate the pulse width of one bank and the frequency of the other... it all added up. Then, sometime in the 1980s, chorus effects became legitimate, and started to



appear on multi-oscillator-per-voice synths as well as single-oscillator instruments. Ask yourself, what's the thing that most synth fans mention first when discussing the Elka Synthex? The oscillators? The filters? The modulation? No... it's the superb chorus unit at the end of the signal path. Even Sequential Circuits conceded defeat in the late '80s, by adding chorus to the Prophet VS. And what do people most often decry about some of the most powerful 'pure' synthesizers yet developed? It's that they didn't sound lush, so Yamaha redesigned the DX series, restoring the chorus unit that they had removed when they discontinued the GS1 and GS2 in 1983 or thereabouts. By the late '80s, it had become clear to everyone that you could take the less animated sounds from DCO-based synthesizers, or the relatively sterile waveforms from early digital synthesizers and workstations, add a well-designed chorus unit, and the thing would sound gorgeous. Hmm... a bland waveform enlivened by a chorus/ensemble... we used to call that a string synth, and it's one of the enduring absurdities of our industry that instruments such as the Solina and ARP Omni now sell for more than double the price of a well-preserved DX7.




Real-time Jam Sessions in Logic

In this article:


Communal Working In Logic Logic Notes Have Your Say! Creating A MIDI Destination Published in SOS June 2004 Menu Print article : Close window Remote Switching Of The Technique : Logic Notes Port Routing Logic Tips Selecting MIDI Channels Dynamically Finishing Touches Learn how to set up

Current Versions Mac OS X: Logic Pro v6.4

Logic's Environment so that you can jam with other musicians in real time on a single system.

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

Ingo Vauk

A complete Environment setup for selecting any of your sound sources, internal or external, for a single incoming MIDI stream. It also lets you route the MIDI input to the sequencer input for recording purposes if you wish. By having multiple copies of this setup in your Environment, you can easily work with several different MIDI musicians simultaneously.

Everyone knows the scenario: a studio full of people all staring into one computer screen. After the initial fascination, the interest quickly dwindles, and the programmer gets increasingly irritated by everyone else laughing and joking around while he is 'doing all the work'. And the only thanks he'll get for that is to be accused of being a control freak, the one that holds all the power of the session. Let's face it — modern recording techniques, especially when MIDI based, don't exactly enhance that 'band feeling', where everyone can chip in and the writing and production process becomes a collective effort.

Communal Working In Logic So when I was preparing for a session with multiple writers involved I thought that it would be nice to be able to have everyone jamming on an instrument without interfering with my programming. I wanted to be able to go into record without being inundated with MIDI information coming from all corners of the room. Also, I wanted to allow the players to play whenever they wanted, and I didn't want to have to stop three of them in order to record the fourth. That way all the players could vibe off each other.

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Although Logic allows for the recording of multiple master keyboards/MIDI controllers, there are a few drawbacks when recording in the usual default MIDI Environment. Firstly, it is impossible to route the MIDI data to multiple ports without getting into a complex wiring situation that has to be disentangled should a player decide that he/she wants a different sound from a module that hangs off the end of a different MIDI lead. By the time you've rebuilt the Environment, the fickle musician will have moved on, or the session will have ground to a frustrating halt. The other disadvantage of using the usual MIDI Environment for multiple masters is that it isn't possible to record only one of the players at a time without lengthy demixing procedures, again a bit of a turn off for the discerning groove meister. So my idea was to have an Environment that prepared you for all of these eventualities and, once built, could stay part of the general setup of your studio. The configuration I ended up with is quite complicated at first glance, but don't let that put you off, because it only takes about an hour to construct. As an example I'm going to show you how to go about setting things up for a single Unitor 8 system, with eight MIDI inputs and eight MIDI outputs, but this basic layout can be expanded using copy and paste to cover whatever setup you use. It would have been nice to be able to do everything in a modular form using Macro objects, but they don't allow for multiple outputs so unfortunately it wasn't possible. Essentially, what you want to be able to do is feed any connected MIDI input directly to any of your sound modules (either external MIDI modules or internal virtual instruments) at the touch of a button. There are two different ways of implementing such a router, one which allows multiple players to record at the same time, and another which lets the programmer pick and choose who to record at any given time. Plus, you can combine the two where, for example, you want two players to jam along while two others are being recorded.

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.

Creating A MIDI Destination Menu So here's what to do. Start by setting all the master keyboards to output on



different MIDI channels, so that you can separate the incoming MIDI signals using a Channel Splitter object. It is also useful to write down the port numbers (for example, 'USB1 in') of the keyboards, since this will make the labelling process a lot easier. Next, prepare the cabling for each output port. You need to be able to address all individual instruments on one port with one cable. So in the case of Multi Instrument objects you don't need to do anything, but when a port is configured using single Instrument objects these should all be connected to the appropriate channel outputs of a Channel Splitter. The Channel Splitter objects should be labelled clearly with the name of the port they are addressing (for example, 'Split USB7') and should be on the same Environment layer as the instruments.

Although it's a bit tedious to do, entering all the names of your sound sources into the Destinations Fader object makes it really quick to select sounds in the think of a session.

The next step is a little tedious, but makes the whole system very quick to use in the end. Create a new Environment layer, by selecting Insert from the Options menu's Layer submenu, and call it Direct Routing. Now, select Text from the New menu's Fader submenu, and name this new object Destinations. Double-clicking the fader will cause a list with values from zero to 127 to appear. Write into this list the names of the instruments in your system, starting with MIDI channel one on output port one, which goes into list position zero, and working your way up. Single-timbre instruments that occupy a port of their own only need appear once. This means that you can address more than eight output ports. In my example the Roland Jupiter 8 is the only instrument on port one and the Korg M1 only takes up eight channels of port seven. Therefore I still can address 16 different Audio Instrument objects and have a few slots unused. The last entry in this list should be labelled Record, and will be used to route the signal back to the sequencer for recording. Each name in the list now has an individual controller number assigned to it, allowing the fader to act as the master switch that routes the incoming signal to the correct port and sets the MIDI channel for it. Set the Out field in the Fader object's Parameters box to an unused controller number (I've used 86 in this example), and limit the range to the maximum number of Destinations in the list (121 in this example).

Remote Switching Of The Port Routing Now that the Destinations Fader has been programmed with all the different routing destinations, you need to make a Cable Switcher object that will send the MIDI signals to the correct port according to your routing selection. In this file:///H|/SOS%2004-06/Real-time%20Jam%20Sessions%20in%20Logic.htm (3 of 6)9/22/2005 7:47:23 PM


example we need ten outputs from the switcher: eight USB MIDI ports, one cable to supply the 16 Audio Instrument objects, and one cable to route MIDI back into the sequencer input for recording. Pull down the Environment window's New menu again, and head into the Fader submenu — you can find Cable Switcher in the Special submenu. Name the new object Port Switch and set the Range fields in its Parameters box to zero and nine respectively. Also set the In fields in the Parameters such that the object will respond to MIDI Continuous Controller number 86. If you change the object's style from Auto Style to As Text, then you can Double-click on the switch and enter a list of the names of the ports or groups of Audio Instrument objects you want to address. Once this has been done, create a Transformer object from the New menu, and name it Prog2Port. This will re-map the controller information from the Destinations fader so that any groups of instruments using the same port generate the same value — in other words, all USB1 instruments get mapped to a value of zero, all USB2 instruments to a value of one, and so on. In the Transformer's Conditions set the Status field to '=' and the associated value to Control, and in the Operations set the '-2-' field to Use Map. Select the Destinations Fader, the Prog2Port Transformer, and the Port Switch Cable Switcher, and then select Cable Serially from the Environment window's Options menu. Once you get the Cable Switcher outputs connected up it should display the correct port for each destination you dial up on the Destinations fader. If it doesn't, check your Transformer's Universal Map and your controller settings, and tweak them accordingly.

Logic Tips When you click and hold in any window while holding the Control and Shift keys, the cursor becomes a four-way arrow, allowing you to scroll the window contents in two dimensions. You can also invoke two-dimensional scrolling by dragging just to the left of the window's horizontal scroll bar. Len Sasso You can scrub audio in the Sample Editor window by dragging in the Time Ruler. If you hold the Shift key when releasing the mouse button, the closest region boundary will snap to the cursor position. Len Sasso

Selecting MIDI Channels Dynamically Next you need to create a Transformer that dynamically sets the MIDI channel of a signal, again under the control of the Destinations Fader. For this you can make use of the fact that Logic allows you to set Transformer parameters from an external source, using Meta messages. Create two new Transformers and one Fader; name one Transformer Set CH Range, the other Set Channel, and



the Fader CH. On the Set CH Range Transformer set the Status Condition to '=' with an associated value of Control, and put Sub in the '-2-' Operations field. Set the Fader's Out Parameters fields so that it outputs Meta message number 127, and also set the In fields so that it responds to MIDI Continuous Controller number 86. This means that the fader will convert the input MIDI messages from the Destinations Fader into suitable Meta messages. On the Set Channel Transformer, set the Operations Cha field to Fix. Connect the CH Fader to the Set Channel Transformer and move the fader up and down. The Operations Cha value should move from one through to 16. Now cable from the Destinations Fader to the Set CH Range Transformer and on to the CH Fader. Moving the Destinations fader should now move the CH Fader, which in turn sends out Meta messages to set the destination MIDI channel for the Set Channel Transformer's processing. Create a Monitor object (again from the Environment window's New menu) and connect it to the output of the Set Channel Transformer. With all four elements selected, hold Alt and drag them to copy the ensemble as many times as you have ports to address. Connect the copies to the Destinations fader as before, and name the Monitor objects to reflect the names of the ports you want to address.

Finishing Touches

This is how you set up the CH Fader object's Parameters so that it converts MIDI Continuous Controller number 86 into Meta message number 127.

Connect the Sum output of your Physical Input object to the Port Switch Cable Switcher, and cable the individual outputs of the Cable Switcher to the output port Set Channel Transformers. Playing your master keyboard while moving the Destinations fader should then route the data to the different Monitor objects. The MIDI channels will also be changing, but probably not correctly. Use the Sub values in the Set Ch Range Transformers to offset the MIDI channels to sort this out.

All that remains to be done is to connect the output of the Monitor objects to the Channel Splitters in the Instrument and Audio Environment layers and to check that everything is happening as it should. The port that is addressed by the Record setting in the Destinations Fader list should be connected back to the sequencer's input. This allows you to record that individual master keyboard back in the usual way, effectively bypassing the whole routing setup. Note that the sequencer has to be running for virtual instruments to play — when Logic's transport is stopped the only Audio Instrument that will play is the one that is selected in the Arrange window. By copying this ensemble and taking the input from the individual ports of the Physical Input object rather than from the Sum connection, you can get the file:///H|/SOS%2004-06/Real-time%20Jam%20Sessions%20in%20Logic.htm (5 of 6)9/22/2005 7:47:23 PM


separation of the different sources. It is also useful to connect a few copies (as many as you have MIDI keyboards/controllers) to a Channel Splitter with the icon enabled, which means that you can record onto a MIDI track (with the splitter being the Track Instrument in the Arrange window) and then separate the different signals from the sequencer afterwards. This Logic Song file contains an Environment setup which allows you to jam with multiple musicians, as discussed in the article. Note: PC users running Logic 5.x should download Tim Rainey's PC version of this Environment instead, which avoids crashes associated with Style 5 Meta faders. Published in SOS June 2004 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Recording A Live Choral Performance

In this article:

The Task Ahead The Soundfield Microphone Go To Plan B The Final Setup Did It Work? Kit List Mixing The Project Finishing Off

Recording A Live Choral Performance From miking to mixdown Published in SOS June 2004 Print article : Close window


The story of a multi-miked location recording session, from pre-concert setup to post-recording, softwarecontrolled mixdown. Paul White

When recording an all-acoustic ensemble, the job is much easier if the instruments achieve a reasonable natural balance and if the room acoustics are sympathetic to the music. Assuming these criteria are met, ensembles of any size — from orchestras down to duos — can be recorded using a simple stereo mic technique, adjusting the mic-to-performer distance to obtain the correct balance of direct to reverberant sound. The job becomes more difficult, however, when you're recording a mixture of acoustic and amplified instruments, some of which are amplified separately via the house PA — which exactly describes a recording project SOS Technical Editor Hugh Robjohns and I undertook recently.

The Task Ahead The performance we had to record was of a modern choral composition entitled Rites of Passage, written and conducted by Richard Chew. It was around 35 minutes long and was to be performed by a large youth choir made up of choristers from six schools, plus five professional and semi-pro soloists supported by a small wind and brass orchestra, two electric pianos and a generous percussion section, including a drum kit. The performance venue was Malvern's Forum theatre, which (since its extensive redesign) has a remarkably dead acoustic, despite its cavernous size! Our job was to commit as faithful a representation of the performance as possible to 16 tracks of an Alesis HD24 recorder, using a selection of mics and direct injection. The HD24, as its name implies, does offer 24 tracks, but the mixer we were taking with us could only

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accommodate 16, so that's what we had to work with. Within this restriction, we needed to build in as much flexibility as possible, to allow us to create a good mix at the end of the project. The aim was to produce a CD for the composer, with tentative plans to also make it available to the schools whose pupils took part, any proceeds to go to charity. The stage layout at the Forum was more conducive to live performance than to recording — and to make matters more difficult, we also had to move all the mics after the soundcheck, then reposition and check them during a 15minute break in the programme. This was because the first half was taken up by several smaller musical items — something we weren't made aware of until immediately prior to the performance! The choir was positioned at the rear of the stage on a raised plinth, with the five soloists to the centre and in front of the choir. About ten The layout of the performers and feet in front of them were the two electric microphones for the Malvern pianos, which were fed through their own Forum concert. amplifiers, with the drums and percussion to the right and the bass guitar to the left. The small orchestra was set up to either side of the pianos, in front of the other instruments. (See the diagram below for a better idea of the layout of the performers.) Because the concert would be amplified through the house PA system, the balance we'd hear on stage would not be the correct front-of-house mix, so we couldn't just record the whole thing in stereo: multiple mics would be needed to achieve an acceptable balance. The original plan was to use a Soundfield mic above the conductor (see 'The Soundfield Microphone' box for more details), to provide the main stereo coverage, with additional mics to cover the left and right wings of the choir, plus three mics spaced in a row in front of the soloists (ie. one mic per pair of soloists). The two electronic pianos would be DI'd in mono and the electric bass guitar miked or DI'd as necessary. While Hugh and I rigged our equipment for the recording, the theatre's PA engineer set up five Shure SM58 mics in front of the six soloists, placed a mic on the bass guitar amp, and took DIs of the two electronic pianos in order to provide a little extra support via the house PA system.



The Soundfield Microphone The Soundfield Microphone has been around for about 25 years, in various guises, and is arguably one of the most versatile microphones in the world. Sadly, it has struggled to gain widespread acceptance despite its remarkable qualities — but both Paul and I are sufficiently impressed with it that we each own SPS422-B Soundfield systems. This is the basic studio version of the threemodel range (the others being the ST250 portable unit and the MkV flagship studio processor).

The top-of-the-range MkV Soundfield mic and processor.

The first Soundfield microphone was developed by the National Research Development Corporation (NRDC) and Calrec Audio, based on the work of mathematician Michael Gerzon. It formed the heart of what became known as Ambisonics — an advanced surround sound system which suffered (unfairly) through association with the much inferior quadraphonic systems of the late 1970s, and is only realising its true potential now with the increasing use of DSP-based decoders. In 1993 the rights to the Soundfield microphone technology and patents were bought by Soundfield Research, which has been marketing and developing the microphone and its associated systems ever since. For anyone not familiar with the technology, the Soundfield microphone comprises four sub-cardioid capsules in a tetrahedral array (see photo, right). The outputs from these elements are 'transformed' in the associated control unit to provide signals equivalent to those generated by four first-order virtual microphones, all four being 'coincident' with one another. This combination is known generically as the 'B-format'. The four signals equate to an omnidirectional microphone (known as the 'W' signal), and three orthogonal figure-of eight microphones facing front-back (X), left-right (Y) and up-down (Z), respectively. You could think of it as a kind of three-dimensional MS (Middle & Side) array. The strength of this composite B-format signal is that it contains all of the directional information of any incident sound wave in all three dimensions, and can either be recorded to a four-track machine, or processed directly to produce any number of outputs, corresponding to virtual microphones pointing in any direction and with virtually any polar response. All three Soundfield mic processors can provide the raw Bformat signal for recording purposes, but only the MkV and SP451 processors have the facilities to manipulate the signal in that domain as well (although file:///H|/SOS%2004-06/Recording%20A%20Live%20Choral%20Performance.htm (3 of 10)9/22/2005 7:47:31 PM


Soundfield have produced a very clever plug-in for the SADiE and Nuendo DAWs (Digital Audio Workstations) that provides sophisticated manipulation and surround-sound decoding functions). Although the Soundfield mic is ideally suited for surround-sound acquisition, most people use it as an ultra-flexible stereo microphone or even as a highquality mono mic (Pete Waterman is said to favour the Soundfield mic for recording pop vocals). All three Soundfield controllers provide a stereo output and the signal can be manipulated and configured in the MS domain, with controls to adjust the notional 'M' mic's polar pattern and the relative level of the virtual 'S' mic. The internal MS signal can then be transcoded to conventional left-right stereo for recording. In effect, the 'Pattern' control determines the amount of ambience and the 'Width' control sets the size of the stereo image — both adjustable in real time, remotely, and without having to fiddle with the microphone itself. We chose to use the Soundfield mic for the Malvern Forum project partly for convenience, but also because of an interest in acquiring surround sound material for future experimentation. The Soundfield microphone is compact and easy to rig — like any single-bodied stereo mic — but it has the advantage that once it's suspended in roughly the right place above the orchestra, the direct/ambient sound balance and stereo width can be adjusted from Soundfield B-Format signals. the control unit, saving the user from having to physically mess around with the microphone too much. For this particular recording, I configured the Soundfield's stereo output with a near-cardioid polar response for the 'M' contribution, and slightly more than half width for the 'S' contribution, the result providing the equivalent of crossed cardioid mics at roughly 90 degrees. We also recorded the B-format outputs to four additional tracks, and although these signals were not used for the stereo mix, they offer the potential in the future to rework the material for surround sound. Furthermore, had we been dissatisfied with the stereo output configuration we could have re-engineered the stereo output from the source Bformat, using either a MkV processor or the Soundfield software plug-in as mentioned earlier. Hugh Robjohns

Go To Plan B During the rehearsal, it became clear that the orchestra was rather smaller (and quieter) than we had anticipated and the choir rather larger (and louder!), so the plan quickly had to be amended. We also realised that some of the youth choir were sitting along the front of the rostra where our three soloist mics were placed, and there was therefore a danger that we might pick up a lot of file:///H|/SOS%2004-06/Recording%20A%20Live%20Choral%20Performance.htm (4 of 10)9/22/2005 7:47:31 PM


mechanical thumps and bumps from the children hitting the stands. In addition, it became apparent that some of the soloists tended to move around and often favoured the SM58s directly in front of them. The quickest and easiest way to address these issues was to relocate the two mics originally intended to cover the choir sides, to help capture the woodwind and brass, respectively. We also added an extra mic to provide a little more definition to the percussion ensemble, and put a further mic in front of the small bass amp, rather than using a DI. As a final safety fall-back we thought it prudent to arrange a mixed feed of the five SM58 mics from the house PA. I also lowered the Soundfield mic to about eight feet high (I had initially set it up at about ten feet), in order to give a little more prominence to the orchestra. As the drum kit was being used in more of an orchestral context, close miking of that was not thought necessary.

Here it's possible to see the five SM58s feeding the PA set up in front of the soloists (front row of choir), plus the three condensers that were capturing their performance for the recording. To the left and the right are the BLUE Baby Bottles, with the SE1 in the centre.

All but one of the soloist and orchestra mics was connected to a 25-metre, eightway multicore back to the desk, and there was a separate dedicated cable for the Soundfield mic. An Alesis 3204 mixer was used to accommodate the microphones, as it has quiet mic amps and also features direct outputs on all 16 channels. Furthermore, it has an in-line monitoring system, so we were able to listen to a mix of the recorder outputs and also solo individual microphones without disturbing the recorder feed. This whole rig was located in an aisle at the side of the choir rostra. This meant that loudspeaker monitoring was out of the question, so we monitored via headphones. To avoid low-frequency thumping caused by stage noise, we used the low-cut switches on the mixer for the soloist mics, and also asked the conductor to get the choir members to remove their shoes during the performance, which they did. This simple measure was well worth taking, as it dramatically reduced the amount of stage noise.

The Final Setup The final configuration of sound sources occupied 16 tracks of the Alesis HD24 hard-disk recorder, recording at 24-bit, 44.1kHz. The first four tracks carried the B-format outputs from the Soundfield mic, as Hugh thought that we might be able to experiment with some surround-sound remixing of the concert at a later date. file:///H|/SOS%2004-06/Recording%20A%20Live%20Choral%20Performance.htm (5 of 10)9/22/2005 7:47:31 PM


Next came the two piano DIs, the woodwind and brass mics, and the three soloist mics. The bass guitar and percussion mics were next, followed by a mixed feed from the house PA's array of SM58s and, finally, the stereo output from the Soundfield mic. Pianos were DI'd using a dual-channel passive EMO box, and a third singlechannel DI box was used to isolate an auxiliary output from the house PA's desk (a Soundcraft 6000) of the five soloist SM58s. The two orchestral mics were Sennheiser MKH40s on high stands, with Audio Technica mics for the bass guitar amp (an AT4040) and the percussion (an AT4033a). Originally the soloists were covered by three BLUE Baby Bottle mics, but one Hugh Robjohns with the Alesis 3204 mixer developed a fault and had to be used during the recording. replaced with an SE1 at the last minute. The mics used were, inevitably, condensers, chosen for their full frequency range and high sensitivity, with the exception of the SM58 dynamics placed by the house engineer. The mics' cardioid (directional) pickup characteristic helped us to exclude spill from neighbouring sound sources and gain some separation, as a cardioid pattern picks up sound most efficiently from one direction and tends to reject off-axis signals. The orchestra mics were moved to achieve a reasonable balance between the orchestra and choir during the soundcheck, and because of the close proximity of the soloists to the centre of the choir, we judged that the solo mics would also reinforce the choir, to the extent that additional choir mics would not be needed. Recording levels were set up during the soundcheck to leave between 6dB and 12dB of headroom, which was established by using the indefinite peak-hold feature on the Alesis HD24. During the rehearsals we were able to confirm that the mics were all providing the coverage we expected of them, with reasonable separation, and that everything was hum and buzz free.

Did It Work? The simple setup outlined above worked reasonably well, but it did throw up a few interesting points that are applicable to most jobs of this kind. The first is that as we had to remove the mics before the show and then reposition them in the interval, there was no guarantee that their positions would be exactly the same as during the soundcheck. We marked the positions of the mic stand bases on the floor with crosses of gaffer tape, but that's about as precise as we could be. Perhaps the biggest challenge was capturing the solo vocalists adequately, as they tended to move around, and in some sections some were noticeably louder file:///H|/SOS%2004-06/Recording%20A%20Live%20Choral%20Performance.htm (6 of 10)9/22/2005 7:47:31 PM


than others. As we were covering all five of them using just three mics, we had a degree of control at the mixing stage, but not always as much as we'd have liked. The other balance issue was to do with the relative level of the choir. We'd set up the mics during the afternoon rehearsal and soundcheck to achieve what we felt was a natural balance between the choir and orchestra, but we had very little further control over the choir/orchestra balance at the mixing stage, though we did have the option to automate the levels on the soloists' mics when the soloists weren't singing, to bring up the choir.

Kit List Alesis HD24 24-track Harddisk recorder Alesis 3204 16-channel mixer Audio Technica AT4040 mic Audio Technica AT4033a mic 2 x BLUE Baby Bottle mics 2 x Sennheiser MKH40 mics Soundfield SPS422-B mic system

Inevitably, some of the amplified electric pianos and bass guitar bled through to the orchestra Studio Electronics SE1 mic mics and to the stereo Soundfield mic, but Selection of DI boxes fortunately it wasn't so loud that it caused us any major balance problems. One of the pianos was switched to an organ sound for about the last third of the performance, and this had to be increased in level, using mix automation, in order to achieve a correct balance. Another balance-related challenge involved the drums and percussion, which fed through into all the mics to such an extent that the separate percussion mic wasn't really needed. With hindsight, that mic would probably have been better utilised for miking the choir. As it turned out, we ended up with an acceptable balance, but had it not worked out that way, there was very little more we could have done about it. Just as with rock bands, everyone seems to play with more exuberance (read level!) during the performance than during the rehearsal.

Mixing The Project To mix this project, I transferred it from the HD24 ADAT recorder to my studiobased Mac system, using the Alesis Fireport adaptor, which allows the ADAT's system drive to be read by the computer as an external Firewire drive. Its support software includes a utility for selecting which tracks to copy over, and in what audio file format. This is straightfoward, and I'd estimate that moving the whole 16-track project to my Mac's hard drive took less than 10 minutes. The reason for mixing in the Mac was so that my Logic Audio sequencing software could be used to automate mix levels — and so that we could use plug-in processors, specifically compression and EQ. The first step after importing the audio was to audition the individual tracks, to see if there were any problems. The solo vocalist group would obviously require some juggling via the track automation, to keep the overall level even, and one of file:///H|/SOS%2004-06/Recording%20A%20Live%20Choral%20Performance.htm (7 of 10)9/22/2005 7:47:31 PM


the sopranos had a particularly piercing voice that had to be tamed slightly, using a notch filter at around 1.8kHz. The low-cut shelving filter was also used, adjusted to remove as much low-end spill as possible without materially affecting the tone of the voices. We created a separate subgroup of the three vocal spot mics plus the feed from the house PA, and inserted a compressor into this group just to tame any vocal excesses. We set the compressor to give around 6dB of gain reduction on the peaks, which had the additional benefit of lifting the choir level slightly when the soloists weren't singing.

The Soundfield mic, positioned high above the conductor, was intended to provide the main stereo coverage of the choir and orchestra. Two spot mics (Sennheiser MKH40s) were also used on the choir.

The basis of the mix was to be the Soundfield mic's stereo output, naturally, but surprisingly the balance between choir and orchestra favoured the orchestra a little more than was ideal, due to their previously mentioned exuberance! This was confirmed by the peak-reading meters, which showed that our headroom was down to 3dB in some cases, where it had been in excess of 12dB during the rehearsal! With hindsight, Hugh suspected that he might have set the height of the Soundfield mic a little too low when we adjusted the rig during the rehearsal, but it's very hard to make this kind of critical judgement when monitoring on headphones in the same space as the performers! The three orchestral spot mics (wind, brass and percussion) were panned to match the Soundfield's stereo image and balanced to provide a degree of clarity and focus for the mix. We compressed and equalised the bass guitar signal — to produce a consistent sound that underpinned the mix but was still audible in its own right — and panned the two pianos slightly. Otherwise, the pianos were more or less left alone, although the level of one of them had to be increased when it switched to a church organ sound, as mentioned earlier. The three soloist mics were also panned only slightly across the centre of the stereo image, to minimise unwanted image shifting when we automated the levels of those mics. Our three-mic configuration actually delivered a pretty respectable sound that also helped to complement the level and definition of the youth choir quite well. The mixed feed of SM58s helped to bolster the soloists a little when they occasionally wandered off-mic, but as this feed was a mono source it compromised the stereo imaging if used too high in the mix, so we added it sparingly. Getting a reasonable ball-park balance of all the elements of the ensemble turned out to be fairly straightforward, but because the soloists had been close miked they sounded rather dry. To help everything 'gel', as well as to compensate for the unusually dry hall acoustic, we added a couple of different



convolution reverbs, rather than using conventional artificial reverb. Convolution reverbs are based on the sound of real rooms, and when you have genuine room acoustics present already to a certain extent, as we did at the Malvern Forum, it sounds more natural to add the sound of another real room than an artificial ambience. In the end we used one convolution Logic Audio's automation was used reverb plug-in (Emagic's Space extensively during mixdown, to maintain a Designer) on the channel sends, to correct balance between elements. impose the acoustics of a real concert hall on the recording, and the Waves IR1 (in the vocal mic subgroup insert) to add a slightly more rounded concert-hall reverb to the vocal soloists. The miked orchestra already included some natural room ambience, so we were careful not to muddy the sound by adding too much reverb. The addition of appropriate concert hall reverb made a huge difference to the subjective quality of the recording and also knitted the sounds together in a very natural way. When it was time to fine-tune the mix, most of the work went into automating the levels of the three soloist mics to achieve a consistent vocal balance. In sections where none of the soloists were singing, we could increase the mic level subtly when required, to boost the level of the junior choir behind them. We then sent a preliminary CD mix to the composer, and he made a note of any further adjustments he deemed necessary, prior to attending the final mix session.

Finishing Off The finished mix was topped and tailed and burnt to a master CD using Roxio's Jam. With hindsight, I would have liked a couple more mics on the choir, but because we mixed in software we were able to easily and extensively automate levels, allowing us to compensate to a large extent for not having quite as many mics as we would have liked. The end result sounded very natural and met with the approval of everyone involved. Published in SOS June 2004






Synchronisation in Cubase

In this article:

MIDI Clock Caveats Master Of The Universe Slave To The Rhythm Receiving Timecode From Tape Machines Frames Of Reference

Synchronisation in Cubase Cubase Notes Published in SOS June 2004 Print article : Close window

Technique : Cubase Notes

If you get that sync'ing feeling when using Cubase in conjunction with external hardware devices, you may need to know more about its synchronisation options. This month's column explains. Mark Wherry

The basic principle when synchronising multiple devices in the studio is that you have one Master device that sends sync data to all the other devices, which are said to be Slave devices, in as much as they are 'slaved' to the Master. When you press play on the Master device, for example, all the slaved devices will begin playback at the same time, and a continuous clock signal from the Master will make sure the Slave devices stay in sync. Cubase's synchronisation facilities are actually pretty comprehensive, and Cubase can act as either the Master or the Slave — or, in fact, as both at the same time. In this latter configuration, the software can control devices while itself being controlled from another device — for example, Cubase might be slaved to a video machine while simultaneously sending clock data to a synth's arpeggiator. For Cubase's synchronisation features to be active, the Sync button on the Transport panel must be enabled, which you can do by simply clicking it, pressing 'T', or selecting Transport / Sync Online. To configure the synchronisation settings in Cubase, you need to use the Synchronisation Setup window, which can be opened by selecting Transport / Sync Setup or Control/ Apple-clicking the Sync button on the Transport Panel.

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MIDI Clock Caveats Since the release of Cubase SX/L, many users have reported problems when slaving other devices to Cubase via MIDI Clock. This doesn't seem to affect every user, but if you notice your external devices not synchronising correctly to your Project, Steinberg first recommend making sure that the SysEx option is enabled in the Thru group of the MIDI Filter panel in the Preferences window — which it should be by default. There's a reason for this: many devices send out MIDI Clock data whether you ask them to or not (consult your manual to see if your device needs to be set to only receive MIDI Clock, rather than transmit it as well), and if this information is sent to Cubase and then resent, via the MIDI Thru facility, back to the device, it's understandable that some confusion can occur. Enabling the SysEx Thru Filter option forces Cubase to filter out all incoming System Exclusive MIDI data (including MIDI Clock) and ignore it. The second thing to bear in mind when using MIDI Clock is that the MIDI Clock commands supported by Cubase include Start, Stop, and Continue, which all do pretty much what you'd expect them to by their names — Start tells a device to start playback, Stop tells a device to stop playback, and Continue tells a device to continue playback from the point where it was last stopped. However, it turns out that not all equipment that can be synchronised via MIDI Clock supports the Continue command. Many drum machines fit into this category. So if you notice the device you have slaved to Cubase via MIDI Clock not resuming playback when you're part-way through a Project, it's worth enabling the 'Always Use MIDI Clock Start' option in the Transport panel of the Preferences window, forcing Cubase to use the Start command instead of sending Continue commands.

Master Of The Universe In most cases, you'll want to set up Cubase as the Master device, so that all your other hardware is slaved and kept in time with the Project that's playing in Cubase. A common example is where you want to enable the arpeggiator or other tempo-based feature on a synth to play in time with the Project in Cubase — but there are many other uses for the software's synchronisation facilities. Cubase can send out two types of MIDI-based clock, namely MIDI Timecode (MTC) and MIDI Clock, to other devices. The fundamental difference between these types is that MTC is time-based, so the timecode is based on the SMPTE hours:minutes:seconds;frames format, while MIDI Clock is based on bars and beats, consisting of 24 pulses for every quarter note — a 4/4 bar would therefore consist of 96 (4 x 24) synchronisation pulses. For this reason, MTC is typically used to sync video machines, other multitrack recorders, and so on, whereas MIDI Clock would be used to sync the arpeggiator on a synth, for example.



If you're going to use MTC, it's important that the frame rate (expressed as frames per second — fps) is set to the same value on your Master (Cubase) and Slave devices. Frames are sub-divisions of each second, and each second must be subdivided in the same way on each device. (If you're plannning to use MIDI Clock, you can jump straight to the setup instructions in the next paragraph.) To set the frame rate for a Cubase Project, open the Project The Synchronisation Setup window is where Setup window by selecting Project / you can configure how Cubase sends sync data to other devices as the Master, or Setup (or pressing Shift+S), click the Frame Rate parameter and choose the receives it from other devices as a Slave. required rate from the pop-up menu. The Project Setup window is also used to define the start SMPTE position, which you can do by adjusting the Start parameter — it's set to 00:00:00:00 by default. To set Cubase so that it acts as a Master device and sends synchronisation data to other devices, all you need to do is decide which MIDI ports you want to send either MIDI Clock or MTC to, and enable the relevant check boxes for those MIDI ports in either the Send MIDI Timecode or Send MIDI Clock groups in the Synchronisation Setup window. That's literally all there is to it!

Slave To The Rhythm To set Cubase so that it acts as a Slave device, listening and responding to synchronisation data sent from another device, you need to change the Timecode Source setting in the top-left group of the Synchronisation Setup window. By default, Timecode Source is set to None, meaning that Cubase uses its internal clock as the timecode source. However, you can set Cubase to receive timecode via MIDI, ASIO Positioning Protocol, or VST System Link. MIDI Timecode: MIDI Timecode is basically timecode sent via MIDI cables, based on SMPTE frame-based times, as described earlier. If MTC is chosen as the timecode source in Cubase, the MIDI input port that MTC should be received from is set with the MIDI Timecode Settings MIDI Input pop-up menu. ASIO Position Protocol (APP): This is part of the ASIO technology developed by Steinberg to allow audio hardware, such as soundcards, to work with Cubase, where sample-accurate timing information can be received from another device connected to a digital interface on your soundcard. This can be useful if you're recording multitrack audio into Cubase from an external digital multitrack recorder. However, APP must be supported by your soundcard and driver software in order for this to work in Cubase. file:///H|/SOS%2004-06/Synchronisation%20in%20Cubase.htm (3 of 5)9/22/2005 7:47:40 PM


VST System Link: VST System Link is Steinberg's platform-independent technology for using Cubase with other computers via digital audio interface connections running System Link-compatible software, such as Cubase SX/ SL, Cubase VST 5.2 (which runs in slave mode only), and Nuendo. If you're using Cubase on a network of computers running VST System Linkcompatible software and you want to sync Cubase to the master machine via VST System Link, this is the option to enable. The interesting thing about VST System Link is that it also includes timing and tempo Some important sync-related information, which is important because unlike settings can be found in the Cubase VST, SX/L can't be slaved to MIDI Clock, Project Setup window. These include the Project's SMPTE due to the newer timing engine being based start time and frame rate. around linear time rather than musical time. This means that if you want to set up a Cubase system primarily as a mixer (slaved to another Cubase system, for example) with tempo-dependent effects, VST System Link is your only choice.

Receiving Timecode From Tape Machines If you're slaving Cubase to timecode sourced from an analogue tape machine, which I'm told some people still do, the signal might not be quite as reliable as if it was coming from a synth or another device generating the timecode internally. For this reason, you might want to adjust the Drop Out Time setting in the Options group of the Synchronisation Setup window, which tells Cubase how many frames to keep going for when the timecode signal 'drops out'. If you notice Cubase occasionally pausing when playback is slaved to timecode from a tape recorder, you might want to increase the Drop Out Time value. However, doing this means that Cubase will take longer to cease playback when you stop the tape recorder, since it will assume it's a drop out until the Drop Out Time value expires. So if your tape recorder is actually sending a reliable timecode signal it's a good idea to reduce this value, to make Cubase more responsive when you stop the recorder. A closely related option to the Drop Out Time value is the Lock Time parameter, file:///H|/SOS%2004-06/Synchronisation%20in%20Cubase.htm (4 of 5)9/22/2005 7:47:40 PM


which sets how many accurately-timed frames of timecode Cubase should receive before starting playback based on the location of the incoming timecode. If you notice Cubase starting playback at the wrong timecode position, before jumping to the correct timecode position, you might want to try increasing the Lock Time value. Conversely, if you have a really efficient tape machine you might want to reduce this value so that Cubase locks to the incoming timecode sooner and everything feels a little snappier.

Frames Of Reference When Cubase is being slaved to another device, you can see via the Sync indicator on the Transport Panel whether incoming sync data is being received. When the Sync button is disabled, this indicator will report an Offline status, but when you activate the Sync button and Cubase is slaved to another timecode source, the Sync indicator will report either an Idle status (meaning that Cubase is waiting to receive timecode), or a Locked status, which confirms that Cubase is locked to an incoming timecode, and also displays the frame rate of the timecode. The Frame Rate setting in the Project Setup window is automatically adjusted, and the Project's frame rate is instead determined by whichever device is sending the Master timecode to Cubase. Published in SOS June 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 your MIDI Synth to control Plug-ins

In this article:

Elements Quick Tips Sources & Targets In The News Buttons Control Items In Action

Using your MIDI Synth to control Plug-ins Digital Performer Notes Published in SOS June 2004 Print article : Close window

Technique : Digital Performer Notes

Fancy turning your knob-laden MIDI synth into a control surface for tweaking plug-in synths? DP's Consoles make this kind of application possible. Robin Bigwood

In last month's Performer Notes I started to look at Digital Performer's Consoles, and showed how to set up a Console that redirected MIDI Controller 1 data (from a synth's mod wheel) to the centre-frequency parameter of the Multimode Filter plug-in. The resulting small but actually quite sophisticated Console embodies a large part of what DP's Consoles are all about: namely, making connections between MIDI sources and destinations that would otherwise be impossible to achieve, and changing one type of MIDI message into another (in this case, controller data into note data).

This Control Assignment dialogue box configures a Console slider to send MIDI continuous controller 91 data to a JV1080.

This month it's time to take a more indepth look at how Consoles work, and examine the individual elements that make them up.

Elements When it's first created, a Console (which you create by clicking the Project menu, then choosing Consoles / New Console) is the ultimate blank piece of paper. It's

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only what you put on it that determines its role and usefulness. The bits of the Console that actually do the work are 'Custom Control Items', of which there are three basic types: Knobs, sliders, value boxes and pop-up menus: These can all generate (or redirect) data with a range of values. Buttons and text buttons: These do the same for data having one or two values (like 'on' and 'off', for example). Arrows: These work in conjunction with other Control Items and can be clicked to add one to or subtract one from their current values. Some Consoles use all these Control Items in abundance, whereas others just use one or two. In either case, though, the Console first has to be 'designed', using its so-called Edit Mode, and each individual Control Item configured for a specific task. Edit Mode is switched on by clicking its button — a dotted square with a blob at the lower-right corner — in the Console's title bar. (This mode is also switched on automatically whenever you place a new Control Item.) To draw a slider, for example, you'd just choose Slider from the Insert ('I') mini-menu in the Console's title bar, then click and drag on an empty part of the Console window to place and size the Item. All Control Items are placed in this way, and while you're in Edit Mode Items can be selected, moved and resized as if they were objects in a graphics or page-layout application. Whenever you create a Control Item, a Control Assignment dialogue box appears prompting you to configure the Item for use right away. The dialogue box can take two forms, depending on whether you've just created a slider/knob/value box/pop-up menu, or a button. In each case, the upper part of the dialogue box is the same, allowing Source, Target and data types to be configured. The idea is that each Control Item is capable of redirecting or remapping MIDI data, and that requires both a source of data and an ultimate destination, or 'target'. You don't have to configure a source, though, as you may want to regard the Control Item — a knob or slider, for example — as a source of data by itself. That's a completely valid way of working. What doesn't make sense is a Control Item without a valid target.



Quick Tips If you're suffering from the DP 4 'Powerbook bug', which can cause processor spiking even at otherwise low CPU-load levels, give the following a try. Rather than setting DP's hardware buffer size to 256, 512 or 1024 samples, try 128 instead, and then set the Host Buffer Multiplier to 2, 3 or 4 to achieve the same buffer sizes, but hopefully without the spikes. Watch www.motu.com for a MAS update that fixes this annoying bug. When you're doing processor-intensive work in DP 4, make sure you quit other non-essential applications. Use the Apple-Tab shortcut in OS 10.3 to quickly see what's running, and check to see what processes are being launched at log-in by going to Accounts, in OS X's System Preferences, selecting your account and then clicking the Startup Items tab. I recently tracked down a MIDI timing problem to a background application a printer driver installation had installed as a Startup Item, but which I hadn't been aware of! Disabling it didn't harm the printing, but fixed my timing problem.

Sources & Targets There are two possible sources for a Control Item — either one of your setup's MIDI inputs (frequently your master keyboard or control surface), or MIDI data already present in a track. You can switch between the two using the radio buttons in the Source section of the Control Assignment dialogue box, and then choose from a list of options in the pop-up menus for each. The contents of these menus is entirely dependent on your current MIDI setup, and on the number and names of MIDI tracks already in your sequence. Once you've chosen a source, you then have to choose a data type for the Control Item to 'listen to', from the Receive pop-up below. Options include continuous controller data, note number, note velocities and so on, and when you choose one of these types various other options, most of which are completely self-explanatory, appear. So, for example, to choose your controller's mod wheel as a Control Item source: Select the 'MIDI' radio button. Choose your controller (and possibly its transmit channel) from the pop-up menu. Select 'controller' from the Receive pop-up menu. Type '1' in the Ctrl # field — or just select 'modulation wheel' from the final popup menu. You can also click to highlight the field and then simply move the modulation wheel — DP 'learns' the controller number you're sending it and enters the value automatically. The final option is the 'Only follow source when file:///H|/SOS%2004-06/Using%20your%20MIDI%20Synth%20to%20control%20Plug-ins.htm (3 of 6)9/22/2005 7:47:47 PM


selected' checkbox, which is often best left unchecked — but more about this next month. Now for configuring Control Item targets, which in many ways is exactly the same as configuring a source. Once again, a target can either be a MIDI device or a pre-existing MIDI track, and you can choose to send a variety of data types via the Send pop-up menu. If you're planning to use a Control Item to translate one data type into another, by the way, it's the simple disparity between Source and Target data types that does the remapping — it's as simple as that. What's very important, though, is the 'Echo data from source to target' check-box. If it's selected, any data received by the Control Item is passed straight through to the target, without any of the modification that can be introduced by the next part of the Control Assignment dialogue box. This makes most sense with the settings available for sliders, knobs, value boxes and pop-up menus, as these give the option to limit the range of output values from a Control Item, and also to 'thin' the data stream it produces, both by limiting the frequency at which data is produced ('Minimum time change'), and making its scaling coarser ('Minimum value change'). The bottom line is this: if you want these data modifications to come into play, don't select 'Echo data from source to target'.

In The News PSP Audioware have been one of the most DP-friendly plug-in developers over the last few years, but though their widely admired Vintage Warmer and '80s-style delays are working fine as VST format plugins running under VST Wrapper 4, their newer VST offerings, such as EasyVerb, have not fared so well. But things are looking up, as their new King of the filters? PSP's new Nitro Audio ultra-flexible filter plug-in, Nitro, is Unit plug-in sounds fantastic, is easy to use, available in Audio Unit format, and is and has all the programming depth you'll already running without a hitch in DP ever need. 4.12. On the face of it, Nitro looks quite simple, but the guts of the plugin are hidden away in the pages of a central 'LCD' screen, with only main parameters directly tweakable from the front panel. Despite its complexity, Nitro is not difficult to use and it has, I think, one of the best manuals I've ever come across. As with PSP's other plug-ins, there's a depth and refinement here which puts it right up there with the very best plug-ins on the market. I'd tentatively say, too, that it beats Antares's Filter at its own game. It's a top class plug-in, available from www.pspaudioware.com for $150. The VST to AU Adapter by fxpansion has recently been updated to version 1.3.02, incorporating stability tweaks and offering compatibility with an even wider range of VST plug-ins, as well as sporting a new copy-protection system. Check out www.fxpansion. com Native Instruments' Reaktor is now at version 4.1, and NI are claiming full compatibility for the Reaktor AU plug-in inside DP 4. There's a snag though, in that MOTU's Audio Unit inspection process currently has Reaktor down as a blacklisted Audio Unit, and so doesn't allow 4.1 to be used! Watch for a DP 4 update from MOTU — which could



already be out by the time you read this — to fix this silly little hiccup.

Buttons As I mentioned above, button-type Control Items output just two data values (and sometimes only one). To configure these values, buttons get their own version of the Control Assignment dialogue box, without the range-limiting and datathinning facilities, but with special controls to determine how the button actually works. Basically, there are three types of Console button: 'single state' (momentary action), 'two-state' (again, momentary, but sending one value when the button is pressed and another when it's released) and 'On/Off' (a latching or 'sticky' button). Single-state buttons can only send a single value, and do so only when they're clicked, so the Control Assignment dialogue box only grants them a single Value field. As you'd expect, then, two-state and On/Off buttons have two Value fields, one for their 'on' (or 'down') state, and one for their 'off' (or 'up') state. This is pretty intuitive to set up, although the 'M' option next to the 'off' field is a very specialised function that only makes sense in certain circumstances — again, more about this next month.

Control Items In Action All the flexibility in the world doesn't count for much unless you can do something useful with Consoles. Fortunately, they can be immensely useful, proving themselves in a number of different areas. I like to think of Consoles as having three main roles: first, as control surfaces within DP for external MIDI equipment; second, as data remappers, allowing you to use almost any data source to control any other; and finally, as 'creative' devices providing modular synth-like connectivity for virtually any parameter accessible via MIDI messages. An ideal candidate for a 'DIY' controlsurface-type Console is the effects depth controller, ideally suited to rack synths that don't allow their 'Multi' or 'Performance' mode effects depths to be controlled from their front panel. You can easily set up a Console with 16 sliders or knobs, one for each MIDI channel on the synth, and each transmitting controller 91 values (or whichever is appropriate). During mixing you can use the Console to set appropriate effects depth values for each channel. The Control Assignment dialogue box for a



If you want to try a 'remapper' Console, button control element, in this case a 'twostate' button which sends MIDI Continuous you can use any 'knobby' hardware Controller 72 data to a Powerbook synth you have which is capable of connected via a MIDI interface. sending controller messages when its front-panel knobs are moved, to control a soft synth such as Native Instruments' Pro 53, where the controller numbers associated with each of its parameters are preset. Within a couple of minutes you can have a Console allowing cutoff, resonance and ADSR controls on your hardware synth to control their counterparts on the Pro 53. In the case of 'creative' Consoles, the sky's the limit. For example, several seconds of fun can be had by controlling note events on a MIDI instrument from your pitch bend or modulation wheel. In all seriousness, this can be a superb effect when used with the right patch, particularly unpitched noise-based sounds. You might also want to experiment with tying a synth's reverb depth to its note velocity, for example. Next month, the final instalment on Consoles: using groups, integrating with recording and playback, and making them look nice... Published in SOS June 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



Vocal Manipulation

In this article:

Melodyne Meets Sonar Quick Tips Harmony Generator Take 2 Tweaking The Interface: View Options

Vocal Manipulation Sonar Notes Published in SOS June 2004 Print article : Close window

Technique : Sonar Notes

This month we investigate a new DXi option that makes vocal manipulation as easy as editing MIDI data. Plus the usual haul of Sonar power tips... Craig Anderton

It's a little ironic: due to the initial paucity of DXi devices, Sonar users flocked to wrappers that would allow VST and VSTi plug-ins to work within Sonar: Cakewalk even acquired the technology for VST-DX Adapter and built it into the program. And now that the whole VST wrapping process is working just about perfectly, the DXi floodgates are opening up as more and more companies adapt their plug-ins to this format. Let's look at a couple of the more interesting developments.

Melodyne Meets Sonar

It's not quite a plug-in and not quite an instrument, but MelodyneBridge provides a link between tracks in Sonar and Melodyne.

Amongst the apps that have been optimised for Sonar, Celemony's Melodyne is one of the most important. For those not familiar with Melodyne, it's a program that gives monophonic audio data the same flexibility as MIDI data. You can change pitch and rhythm, cut/paste/copy notes, alter formants, and much more, as well as using MIDI to change note pitches, add pitch-bend, and the like. Of course, other programs let you do timestretching and pitch-shifting, but Melodyne's speciality is a detection process that causes audio to appear very much like MIDI notes in a piano roll editor, which you manipulate in the same

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Vocal Manipulation

way. The bottom line is that it's a brilliant program, particularly for working with vocals. Melodyne started life as a Mac multitrack program that sort of lived off in its own world and didn't integrate well with other applications. Over the years, that has changed, and Version 2.1 includes two Sonar-orientated integration tools: ReWire and MelodyneBridge. ReWire works as expected, with Melodyne essentially acting like a stand-alone multitrack application that streams audio into Sonar. MelodyneBridge creates a link between Melodyne and Sonar so that Melodyne acts more like a DXi instrument or processor. The overall feel is a bit of a kludge, but it works. You open Sonar, select the track you want to 'melodize' then open MelodyneBridge in the track's FX field as a DXi instrument (you can also use the Insert / DXi synth option). Finally, you open the Melodyne application itself. If you then set MelodyneBridge to 'Record', and start playback within Sonar, material in the track transfers over to Melodyne. After using Melodyne's Detect Melody function, you're ready to mess with the audio. Note that you don't have to record the entire track; you can isolate just a few problem measures and record them. On playback, MelodyneBridge will pass through the non-recorded audio, then take over playback when it hits the audio section that was recorded into Melodyne. It's important to check the FAQs and manual. For example, MelodyneBridge didn't register as a DXi, but an FAQ on the Celemony site (www.celemony.com) tells how to register it manually. There are some other important issues, such as setting Melodyne's buffer size to a greater size than Sonar's. One non-issue: the manual recommends turning off multi-processing, but I tried both enabling and disabling multi-processing and Melodyne worked fine both ways. Incidentally, Melodyne is a deep program — don't expect to master it in an hour, although you can do simple stuff such as fixing note pitches within minutes of being up and running. Overall, Melodyne really adds tremendous capabilities to Sonar, and I'm glad Celemony decided to support the program.


Vocal Manipulation

Quick Tips Don't forget that you can assign a buss, not just a track, to any buss. Right-click on the buss and go Insert Send / (destination buss name). Note that you won't see the current buss as a destination: you're not allowed to choose that one, in order to prevent feedback. Rumour has it that a future version of Sonar is going to support REX files. So, hopefully, before too long you won't need to use the REX file workaround presented in last month's issue! It's easy to mix time signatures within Sonar. Click where you want to insert the time signature change, then go Insert / Meter/Key Change. Enter the desired meter in the dialogue box, then click 'OK'.

Harmony Generator Take 2 In the very first Sonar Notes, we mentioned a technique for generating harmony lines using the old Pitch Shifter plug-in (still available on the distribution CD, but not installed by default). Since then, Sonar has added a DSP-based Time/Pitch function, which includes formant preservation so that vocals sound like vocals, not chipmunks, and has more variable parameters for optimising the sound. In view of this fact, let's re-visit the idea of synthesizing vocal harmonies using Sonar: First, create two clones from your original vocal (right-click on the vocal track number and select Clone Track, then repeat for the second track). One will provide the Major third harmony and the other a Minor third. There are two ways to apply the Time/Pitch Stretch DSP to a Clip or Track: Select the Clip(s) or Track you want to process, then right-click on the Clip(s) and go Audio Effects / Cakewalk / Time/Pitch Stretch. Or, don't right-click but instead go Process / Audio Effects / Cakewalk / Time/Pitch Stretch. When the Time/ Pitch Stretch dialogue box appears, proceed with editing its parameters. The Time/Pitch Stretch module has three tabs: Settings, Advanced, and Mixing. For mixing, specify whether you want the output result to be stereo or mono. Generally, you'll want file:///H|/SOS%2004-06/Vocal%20Manipulation.htm (3 of 6)9/22/2005 7:47:52 PM

The Time/Pitch Stretch processor works very well with vocals, assuming you set the parameters correctly. There are presets for Vocal Maj Third Up and Vocal Third Up (minor third). This picture shows recommended parameter values for the Settings and Advanced tabs.

Vocal Manipulation

the same result as the original track (ie. a stereo result for a stereo track). You can save some time by calling up the 'Vocal Maj 3rd Up' preset for major harmonies. This sets Pitch to +4 semitones and specifies 'Vocal' for the Source Material in the drop-down menu. On the Advanced page, make sure that Accuracy is set to 'High' and Algorithm to 'Formant Preserving'. You can just leave Block Rate, Overlap Ratio, and Crossfade Ratio as they are — or experiment with the settings if you're so inclined. To hear the effect, click on the Audition button. Remember, though, that the Audition amount may be limited to a few seconds, so if the vocal Clip has a lot of dead air at the beginning you may not hear the processing. Either slip-edit the beginning of the Clip, or extend the audition time by going Options / Global / General, and increasing the time of the 'Audition Commands For' parameter. Note that it will take the audition process longer to 'start up' with longer audition times. We should now have the original vocal, one track transposed up a Major third, and one track transposed up a Minor third. The next step is to cut out those phrases from the harmonies that aren't compatible with the original vocal. For example, if one phrase requires a major harmony, remove the matching phrase from the minor harmony, and vice-versa. This requires a lot of cutting, but you're going to have to do it anyway, because the Time/Pitch Stretch process doesn't preserve length perfectly — the tracks always seem to stretch a little bit long, and I find it's necessary to cut bits from the harmony lines and line them up with the main vocal. It might appear that there's an easy fix for this: just shorten the time using the Pitch/Stretch's time-stretching function. However, this function can't be accessed if Formant Preservation is on — and if you turn off Formant Preservation the sound quality for vocals is nowhere near as good. The only real workaround is to make the vocal an exact number of measures and convert it into a Groove Clip, but that messes with the stretched sound. Not surprisingly, real harmonies from a human singer usually sound better than this synthetic version. But synthesizing harmonies creates a distinctive timbre that may sometimes be desirable; in fact, sometimes you might deliberately choose not to use formant preservation, just to achieve a weirder effect.


Vocal Manipulation

Tweaking The Interface: View Options The View Options button in the toolbar above the Track Pane is the key to instant customisation features that influence how you interact with Sonar's interface. Here are your options: Show and Fit Selection: Expands selected Clips both horizontally and vertically to fill the screen. With a single track selected (all non-selected tracks are hidden), this is like having a sample editor showing the waveform in great detail. In fact, after the You can make wholesale changes to the February 2004 Sonar Notes, where I Track View with just a few mouse clicks. described turning Sonar into more of a waveform editor by expanding a track to fill the screen, using 'Show Only Selected Tracks' (below), then using the Track Manager to restore the original workspace, I became aware of 'Show and Fit Selection' as a better option. It was brought to my attention by Michael Hoover of Cakewalk, who helped design the view macros that addressed the 'lack of audio editing view' issue users were complaining about. Click on the clip you want to expand, then hit 'Shift+S' — the 'Show and Fit Selection' macro. This expands the view, but once you're finished editing you can undo the view state simply by hitting the 'U' key, bypassing the Track Manager completely. Fit Tracks to Window: Adjusts the height of all tracks so that they fit in the current Track Pane size (within reason, of course... if you've recorded 65 tracks and the Track Pane height is a couple of inches, you'll see only the first few tracks). Fit Project to Window: Adjusts the Project duration so that you can see the entire Project, from beginning to end, in the Clips Pane. Show Only Selected Tracks: Hides all other tracks. This is basically like 'Show and Fit Selection', except that instead of being based on what Clips are selected, it's based on which tracks are selected in the Track Pane. Hide Selected Tracks: Removes any selected tracks from view. Show All Tracks: Shows any tracks that were hidden. Track Manager: Brings up a window that lets you check and uncheck boxes to show/ hide tracks. Show/Hide Inspector: Provides control over the Inspector. Undo View Change: This is exceptionally useful, especially because all you have to do to activate it is type 'U'. Want to zoom in on a waveform? Fine, but after doing any business with it, you'll probably want to return to the previous view. Just hit 'U' and you're there. Similarly, 'Redo View Change' (Shift+U) lets you toggle back to the previous view. Vertical FX Bins: When unchecked, this places FX bins horizontally rather than as a vertical box. Published in SOS June 2004


Vocal Manipulation




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