Sound on Sound (February 2005)

In This Issue In This Issue February 2005 In This Issue Click article title to open Reviews Intakt & Kompakt Instrume...

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In This Issue

In This Issue

February 2005 In This Issue Click article title to open Reviews

Intakt & Kompakt Instruments East West/Zero G Sample Libraries With NI Plug-in Front-end (Mac/PC) Sample and loop libraries have come a long way from being delivered on a humble audio CD — the most recent trend is to include a specific playback front-end for easy integration with your sequencer.

People

Allan Walker & Craig Connor: Grand Theft Audio Sound & Music For Rockstar Games Imagine selling a million copies of your music in just over a week, with no video, no radio play and no touring. Impossible? Not if it's the soundtrack to one of the most successful computer games of all time...

ADK Vintage Valve A48

Business End

Multi-pattern Valve Mic This new mic combines a colourful valve sound with the flexibility of nine different polar patterns.

MPG producers evaluate reader tracks Listen online to the tracks whilst reading what some Music Producers Guild members think of the latest collection of SOS reader recordings.

ART Tube MP Studio V3 Valve Preamp A pint-sized preamp with anything but a pint-sized sound, plus the bonus of Presets!

Behringer BCA2000 USB 2.0 Audio & MIDI Interface Completing Behringer's B-Control family, the BCA2000 combines audio and MIDI interfacing with routing/mixing features, and preamps — and all for under £200.

Bellari RP503 Valve Recording Channel This new front end offers a clean, warm sound and is surprisingly easy to use.

BIAS Sound Soap Pro Noise-reduction Software (Mac/PC) This cross-platform package from BIAS offers high-quality denoising at an affordable price.

DAV Electronics BG4 Compressor/Limiter A premium compression sound lurks behind the no-frills cosmetics of this new Broadhurst Gardens processor.

Digitech GNX4 Guitar Workstation Is it an amp modeller? Is it a multitracker? Is it an audio/ MIDI interface? We stomp the facts out of Digitech's new box?

Composing & Recording With Vanessa-Mae Walter Taieb Walter Taieb enjoyed worldwide success as a dance producer, before reinventing himself as a classical composer. Working with violin prodigy Vanessa-Mae tested both his orchestration skills and his music technology expertise.

Crosstalk: Your Communications Readers Writes Responses to recent articles from SOS readers.

Modelling Classic Hardware In Software Joe Bryan of Universal Audio Since Bill Putnam's sons founded Universal Audio anew, the company have become well known for software emulations of old gear, as well as for their hardware recreations of vintage classics. We find out how they go about it...

Music Software Updates Leader If Apple and Norton can offer automatic updates, then why can't music software companies? asks Editor-in-Chief Paul White.

Sounding Off: Vintage Synths Andy Foster Why are we still in love with vintage synths when all they do is break our hearts?

Studio SOS Aniff Akinola

file:///H|/SOS%2002-05/In%20This%20Issue.htm (1 of 4)9/22/2005 8:16:32 AM

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FAR OBS Active Nearfield Monitors These compact ported monitors from Belgium boast a communicative mid-range and a flattering overall sound.

FXpansion BFD Virtual Drum Instrument (Mac/PC) When it comes to drum sounds, BFD and its XFL expansion pack offer unprecedented levels of detail and flexibility, totalling over 30GB of sample data. Is it overkill, or the rhythm programmer's ultimate weapon?

Hercules 1612FW

Acoustic problems and mains hum were making Aniff's studio difficult to use, so the SOS team stepped in to help.

The History Of Roland Part 4: 1992-1997 The recession of the early '90s was difficult for many companies — but not Roland. Diversifying still further from their roots as a maker of synthesizers and organs, they emerged stronger, encompassing almost every area of hitech musical instrument manufacture. Technique

Cubase: Working with Multi-Channel

Firewire Audio Interface (Mac/PC) Projects The much-anticipated Hercules 1612FW Firewire interface Cubase Notes supplies an impressive quotient of analogue I/O at a very Returning to Cubase's Mixer window again this month, we attractive price. explore ways to make it more manageable when working with Projects that contain a large number of channels. Hot New Sample CDs Sample Shop Reviews/appraisals of the latest sample CDs: Sample Lab Drum Fundamentals **** MULTI-FORMAT Tekniks Blue Shock *** WAV/EXS24 Smart Loops Pro Drum Works (Volume One) **** ACID Cycles Volume Two: Unnatural Rhythm **** AUDIO?WAV? REX2

Joemeek Three Q Recording Channel The Joemeek product range has recently been completely overhauled, and one of the updated units is this new halfrack recording channel. Have the designers retained what made the original products successful?

Pro Tools: The Mysteries of Shuffle Mode Reavealed... Pro Tools Notes Pro Tools was originally designed as an audio editor, and it still has some very powerful tools for quick and accurate editing. This month's Pro Tools Notes explains the mysteries of Shuffle mode, and looks at the ways in which key commands and memory locations can speed up your editing.

Reasons to be cheerful: v3 Reason Notes Version 3 of Reason is officially on the way, adding some high-spec new devices and improved operational features.

M Audio O2

Sonar: Hands-free Recording & Other Tips

USB MIDI Controller Keyboard Thin is clearly In amongst manufacturers of keyboard controllers. Mere months after the arrival of Edirol's slender PCR1, M Audio weigh in with the equally waif-like O2. But is it slim and trim, or just a lightweight?

Sonar Notes This month, find your way around projects, undertake handsfree recording, configure an external MP3 encoder and create pseudo-Track Folders for MIDI.

Mackie Big Knob

Apple Notes In an extended Apple Notes column we take an exclusive look at the 30-inch Cinema display from a musician and audio engineer's perspective, and evaluate the performance of the new dual-2.5GHz Power Mac G5 with our usual series of performance tests.

Monitor Controller Mackie's new desktop controller offers source selection, monitor switching, and talkback facilities, all in a single robust box.

Mackie Onyx 800R Preamp & A-D Converter Mackie 'up the ante' once again by squeezing eight channels of their class-leading preamplification and A-D conversion into a 1U rack.

MOTU Digital Performer 4.5 MIDI + Audio Sequencer (Mac OS X) The migration to OS X has caused major upheavals for

Apple 30-inch Cinema Display

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CLASSIC TRACKS: The Stone Roses 'Fools Gold'


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both software manufacturers and musicians, but Mark Of The Unicorn's long-established Digital Performer sequencer is better than ever.

Phonic PAA2 Audio Analyser Phonic's new audio analyser is certainly portable, but how well does it work?

Presonus Firepod

Producer: John Leckie. Engineers: John Leckie, John Cornfield, Paul Schroeder As the '80s drew to a close, The Stone Roses made rock music cool again, melding '60s psychedelia and acid house under the production guidance of John Leckie.

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

Firewire Recording Interface (Mac & PC) Unlike most eight-channel interfaces, the Presonus Firepod includes eight mic preamps — at a very appealing Locking Logic Sequences to a Live take price. Logic Notes Although rather time-consuming, this technique for locking Redmatica Autosampler your sequencer to a live take is often the best. Automatic Sample Generator For EXS24 PC Notes If you are keen to move to an all-software setup, but still depend on sounds from your hardware synths, Hints & Tips Redmatica's utility might smooth the path... This month PC Notes comes across CD-burning problems, offers some Gigastudio 3 tips, and warns against a possible Sonic Implants SBC pitfall when updating a Cubase SX3 dongle. 3-DVD Gigastudio-format Orchestral Brass Library Sonic Implants extend their New World Symphony with an Roland VS2480 Masterclass orchestral brass sample library. More Hands-on Power User Tips A collection of imperative tips for harnessing the power of Steinberg Wavelab 5 Roland's monster multitracker. Windows Audio Editor & DVD-Audio Authoring Updating PC Hard Drives: The SOS Guide Package PC Musician The latest version of Steinberg's popular editing program now includes support for multi-channel audio, including You're planning to change your PC hard drives, so you just the ability to create your own DVD-Audio discs. unplug the old ones, plug in the new ones, and off you go, right? Wrong!! There's many a slip, and this month we aim to Submersible Music Drumcore help you avoid all of them. Groove Library/Audition Engine (Mac OS X) Featuring the performances of some of the world's best drummers, Drumcore is a software-based cross between a drum library and a custom loop-auditioning interface, allowing you to assemble rhythm tracks fast.

Trident Audio 4T Celebration Recording Channel Despite its vintage design credentials, this new recording channel has some interesting new tricks up its sleeve. Competition

WIN Arturia Storm & Steinberg Cubase SL software Sound Advice

Q. Can I feed a 24-bit signal into a 16-bit device? Q. How can I get my multisampled synths to sound more even?


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Q. Should I always pan my vocals to the centre of the mix? Q. What kind of mic setup should I use to record a choir? Q. Will connecting a high-quality D-A converter improve the sound of my CD player?


Intakt & Kompakt Instruments

In this article:


Zero G Wired: The Elements East West/Zero G Sample Libraries With NI of Trance Who Stole My 'C' key? (Mac/PC) East West Vapor Published in SOS February 2005 NI's UIs Print article : Close window Minimum System Reviews : Software Requirements Zero G Afrolatin Slam Format Foibles Zero G Nu Jointz Conclusions Sample and loop libraries have come East West Stormbreakz

Kompakt/Intakt Libraries

Plug-in Front-end

a long way from being delivered on a humble audio CD — the most recent trend is to include a specific playback front-end for easy integration with your sequencer.

pros Both Kompakt and Intakt are easy to use and offer plenty of flexibility. Lots of samples! Generally, these libraries offer good value for money.

cons Samples cannot be used with other sample- or loopplaying software. Documentation could be improved. Some of the libraries could have been configured to make better use of the frontend.

summary If you are already an NI user or don't mind being restricted to using these libraries with either Intakt or Kompakt, then each of these libraries has plenty to offer... just make sure your hard drive has plenty of spare capacity!

John Walden

One of the more recent developments in the world of sample and loop libraries is for library producers to include some sort of playback device with the sample set. Two well-known sample companies, Zero G and East West, are now taking this route with a number of their products and, by teaming up with Native Instruments, are supplying libraries with a dedicated version of either NI's Intakt or Kompakt devices. Kompakt is a slimmed-down version of NI's flagship sampler instrument Konkakt. Supplied as a front end to something like East West's Vapor — described later in this article, but essentially consisting of a collection of synth pad, bass, lead and arpeggiated sounds — the principle is not dissimilar to that at work in something like Spectrasonics' Atmosphere or Trilogy. In contrast (or should that be kontrast?), Intakt is a sampler device aimed primarily at those working with loops, and both Zero G and East West now have loop libraries that provide Intakt as their front end.

information £79.95 each including VAT. Arbiter Music Technology +44 (0)20 8970

Two observations are worth making before digging deeper into these libraries. First, the versions of Kompakt and Intakt supplied here are specific to these products and, while both offer considerable playback and processing capabilities, they are not able to import samples from outside the library they are supplied

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1909. +44 (0)20 8202 7076. Click here to email www.soundsonline.com www.nativeinstruments.com www.zero-g.co.uk

Test Spec 2.4GHz Pentium 4 PC with 1GB of RAM running Windows XP Pro and Service Pack 2, plus Echo Mia 24, Egosys WaMi Rack 24 and Yamaha SW1000XG soundcards. Steinberg Cubase SX v3.0.0.

with. Second, the sample and loop libraries themselves are rather like a Reason Refill and can only be accessed via the supplied version of Kompakt or Intakt, or from the 'full' versions of these applications, which has to be purchased separately from NI. NI, Zero G and East West describe these libraries as either 'Kompakt Instruments' or 'Intakt Instruments'. Five such instruments are under consideration here: Zero G's Wired, Afrolatin Slam and Nu Jointz (all based on Intakt) and East West's Stormbreakz (using Intakt) and Vapor (which uses Kompakt). For further details on each individual library, check out the boxes throughout this article.

Zero G Wired: The Elements of Trance Wired provides over a Gigabyte of sample data consisting of both loops and samples aimed directly at dance styles such as trance. Within the collection, the various presets include construction kits, drum loops and individual instruments. The latter use Intakt's Sampler mode and include drum kits suitable for trance and house with plenty of 808 and 909-style samples. Other instruments include a good collection of bass synth sounds, a The Intakt user interface — here seen in small number of bright and brash Zero G's Wired instrument with a pianos, some analogue-style pads construction kit preset loaded. and a huge number of one-shot effect, hits, blips and bleeps. The loops are split into two main groups; instrument loops featuring basses, pads, and pianos, and drum loops that include full kits and individual elements such as kick, snare and cymbals. The loops are organised in a number of ways. Some presets contain a single instrument loop mapped with pitch across the entire keyboard and use Time Machine mode to keep it in sync. Others user the Sampler mode and simply contain a collection of different loops, each mapped to a single key — there are both instrument lines and drum loops arranged in this way. Other presets use the Beat Machine mode and consist of a single drum loop beat sliced across a series of keys. However, for instant gratification — and perhaps exploiting Intakt most effectively — the 14 construction kit presets are hard to beat. These tend to have a selection of keys operating in each of Sampler, Beat Machine and Time Machine modes, and typically contain some drum loops, individual hits, perhaps an few instrument loops and a bass or melody sound. While construction kit formats are seen by some as a 'paint by numbers' approach to music, Intakt allows you to 'play' the loops, hits and instruments in real time, resulting in a more creative process — and, frankly, it is also downright fun. All the construction kit contents are derived from various loops and samples used



elsewhere within the library, and this just emphasises why it is a shame that users cannot load individual loops and samples from within the library to create their own construction kits. This is not just a comment on Wired, as it applies to the other Intakt libraries covered in this review. That said, an upgrade to the full version of Intakt is not that expensive, and for anyone buying two or three of these libraries, upgrading ought to be seriously considered. In terms of content, Wired has plenty to offer trance producers and all the essential elements are here to put together complete tracks. Given that there is over a Gigabyte of sample data, and despite one or two frustrating limitations of the dedicated version of Intakt, the combination of sample data and playback frontend is certainly very good value for money.

Who Stole My 'C' key? Versions of both Kompakt and Intakt have been reviewed in SOS in recent months. The full version of Intakt was reviewed by Simon Price in the June 2004 issue (see www.soundonsound.com/sos/jun04/articles/intakt.htm) while in SOS August 2004 Paul White reviewed an earlier Kompakt-based 'sample library with virtual instrument front-end', Morphology, from Zero G (see www.soundonsound. com/sos/aug04/articles/zerog.htm). While interested readers can refer to these earlier reviews for context, a brief recap will be provided below. All five libraries are supplied in the same physical format. Each box contains a DVD-ROM and a slim printed manual. The manual covers installation and operation of the NI software, whether Intakt or Kompakt, and is pretty much identical whether supplied in a Zero G library or an East West one. Installation of each product proved straightforward, and the user is provided with options for installation of both software and samples. Each instrument is supplied as a standalone application and VST, DXi, Audio Units and RTAS formats. As with a Reason Refill, each library is installed as a single file in NI's NKS format — readable by NI software samplers, but not by others. The final stage of the installation process requires authorisation of the library. This is based upon the combination of a supplied serial number and a system ID number generated by the installer that is specific to the host computer hardware and OS. If the computer has web access, the installer will make a connection to the NI web site where an Authorisation Key (essentially a long string of numbers) is generated. On my test PC, the Authorisation process took just a couple of minutes for each instrument. Authorisation can also be completed from another computer, or by post.



East West Vapor As mentioned in the main part of this review, East West's Vapor is supplied with Kompakt at its frontend rather than Intakt. At over 2GB, this sample library is huge. It also contains a massive number of preset instruments — I counted well over 300 Multis and some 800 individual instruments. The library is the work of programmer Marc Van Bork, who has a film and multimedia background, and this is reflected in the sounds. While they cover bass, Kompakt in cool blue with East West's Vapor. lead, arpeggiated sounds and pads, there are many sounds that would work as a sound bed just to provide an atmosphere. There is also a hi-tech influence, so if you work to picture, these sounds would be more suited to a contemporary setting — modern horror, action or sci-fi rather than costume drama. That said, many of the individual instruments, such as the bass or lead sounds, would work just as well in a number of modern dance styles. Given the number of sounds provided, prepare a thermos and some sandwiches before you start to browse! However, exploring is well worth the effort, as the sounds are always interesting and engaging. Many of the pads and sound beds just seem to go on and on evolving with few obvious loop points. Much of this is down to the length of the original samples used and, while this can mean Vapor is a little CPU-hungry at times, given the complexity of the results, one or two notes are usually enough. One obvious plus for this Kompakt-based instrument, however, is the ability to build your own Multi instruments by combining individual samples of your choice from the library. Add in the basic, but very functional, processing offered by Kompakt and the range of sounds that could be conjured up by Vapor is staggering.

NI's UIs As shown by the various screenshots in this review, the Intakt user interface is identical for each instrument, with only the overall colour scheme varying between the two sample library suppliers — a nice cool blue for East West, while Zero G have gone for pale green. The same NI look is retained in the Kompakt user interface for East West's Vapor. Despite some obvious operational differences between the two types of instrument, due to their somewhat different functions, each contains the same basic four elements to the user interface. From the bottom up, the first three of these are the Keyboard, Effects and Modulation sections. The fourth element, at the top of the window, is where the various samples or loops are loaded. In



Intakt, this is termed the Source Edit section and, amongst various controls, it contains a waveform display for the currently selected sample or loop and tempo controls (including a button to sync tempo to the host sequencer). In Kompakt, this top-most element is split into two; the Multi section and the Instrument section. The Multi section allows up to eight samples (each termed an 'Instrument') from the library to be loaded, while highlighting any one of these allows its tuning, filter and amp control to be adjusted within the Instrument section. Given Simon Price's June 2004 of the full version of Intakt, only the edited highlights need to be repeated here. Functionally, the dedicated version supplied with each Intakt-based Instrument (sample library) is the same as the full version, with the exception that these dedicated versions do not contain the File Browser. Each of the libraries contains a series of preset 'instruments' and these might contain a combination of loops, loop slices, individual samples or a mixture of all of these, with each mapped to a particular key on the keyboard. The samples can only be loaded via the drop-down File menu above the waveform display. Unfortunately, given the absence of the File Browser in this dedicated version of Intakt, it is not possible to mix and match loops or samples between presets even from within the same library. So, for example, if you wanted to use three drum loops from one preset along side a bass synth sample from another preset, you would need to run two instances of the plug-in. Upgrading to the full version of Intakt (which costs about £90 but includes a further 1GB sample library) would resolve this limitation. However, as each instance of Intakt only added a two- or three-percent CPU overhead on my test system, it can be worked around. Intakt has three modes of operation for the samples loaded within a preset; Sampler, Beat Machine and Time Machine. Sampler does Minimum System pretty much what you might expect and, via the Requirements Keyboard section, samples within a preset can PC be mapped to zones across the keyboard. Windows XP, Pentium III/ Zones can overlap to form layers if required. Athlon 700MHz CPU or The Beat Machine uses Recycle-style beat faster, 512MB of RAM. slicing to allow loops to be chopped up, either so they can be played back at different tempos, MAC or so the elements within the loop can be Mac OS 10.2.6 or later, G4 mapped to a series of keys for the creation of processor 733MHz or faster, new loops. Users of other beat-slicing 512MB of RAM. applications ought to find Intakt capable of all the essential 'slice and dice' functions. The Time Machine mode provides an approach more akin to that of Sony Acid or Ableton Live for tempo-matching loops based upon granular resynthesis and time-stretching. Time-stretching can be achieved without altering pitch and, as long as things are not pushed to extremes, the results are very good. Within a given preset, different samples can operate in different modes. Helpfully, the keys within the Keyboard section are colour coded to show which mode each file:///H|/SOS%2002-05/Intakt%20&%20Kompakt%20Instruments.htm (5 of 11)9/22/2005 8:16:51 AM


sample is operating in; yellow for the Sampler, blue for the Beat Machine and red for the Time Machine. When individual slices from a Beat Machine loop are mapped to a series of keys, these appear in a slightly different shade of blue. Editing and settings for the various Modulation and Effects controls can be made at either the level of a single sample, groups of samples or globally. The Modulation section includes a Pitch Envelope, AHDSR envelope, two LFOs and an Envelope Follower. The last four of these can be assigned to modulate a range of other parameters such as filter cutoff or resonance, volume, pan or tuning. While this is not the most sophisticated set of modulation options ever created, this section is very straightforward to use and is capable of transforming a sample or loop in both subtle and extreme ways. The Effects section adds some further sound-shaping options but, while the Filter is perfectly capable, the Lo-Fi and Distortion effects are somewhat basic. If required, the order of these three effects can be adjusted by selecting one and then dragging left or right as required. The Delay effect is, again, somewhat basic but does the job and can be tempo sync'ed to the host sequencer. If more adventurous effects processing is required, Intakt features eight stereo output pairs, so particular samples can be passed to the host sequencer for processing via other plug-ins. All the various controls can be assigned to a MIDI controller for real-time parameter tweaking by holding down the Control button (on the PC) or the Command button (on the Mac), and then clicking on and moving the required hardware controller. This worked fine within Cubase SX3. While the individual features described above are all available in other software applications, their combination within Intakt makes for a very effective and efficient loop manipulation tool. For construction kit-style work, the ability to map a series of loops and samples to individual keys makes it very easy to 'play' these loops in real time. This experience is considerably enhanced by the 'Legato' mode provided in the Time Machine. With this engaged, loops will play back in sync with each other. For example, imagine that two eight-bar loops are assigned to adjacent keys. If one is triggered on its own and the second is then triggered three bars in, playback of the second loop will start from bar three rather than bar one. This makes combining loops on the fly and dropping them in and out of playback to generate new loop combinations really easy. Kompakt, on the other hand, is more of a sample-based synth device than a loop tool. Up to eight samples (each termed an instrument) can be loaded from the library into a Multi — and both East West's Vapor and Zero G's Morphology are supplied with large sample libraries and a comprehensive range of Multi presets. Each of the eight instruments can be edited independently not only for filter settings, tuning and effects, but also in terms of MIDI channel. This means that Kompakt can be eight-part multitimbral or that instruments can be layered by assigning them the same MIDI channel. The Mod Wheel can be configured to control crossfades between two instruments assigned to the same MIDI channel. Key ranges can also be specified, so splits can also be created. The Reverb, Chorus, Delay and Master Filter within the Effects section are basic but effective and operate in a similar fashion to the effects in Intakt, while the



Modulation section provides three envelopes and four filters. A lot has been crammed into a small space here — Kompakt's controls are... er... compact. Somewhat oddly, and despite downloading the latest version of the Kompakt software via the web, I could not find a way in the version of Kompakt supplied with Vapor to allow assignment of MIDI controllers to particular parameters. This had worked fine in Intakt — although as far as I could see, there is no description of how to do this in either manual. As with Intakt, Kompakt has multiple outputs and individual instruments can be assigned as required if you need to add further processing to particular instruments via the host sequencer. Given the many long, evolving samples supplied with Vapor, the sample layering and the internal processing, it is possible to create some really complex sounds — and the Kompakt user interface makes this very easy to achieve. However, as a result, Kompakt does demand a little more CPU resource than Intakt. Usefully, the Multi section includes a measure of CPU usage that changes dynamically as the plug-in is playing.

Zero G Afrolatin Slam Afrolatin Slam, also a Zero G Intaktbased title, provides a collection of loops and single hits of African and Latin percussion, although there are the occasional instrumental loops include as well. The producers, Francis Fuster and Sultan Makende, have some impressive credits including Paul Simon, Hugh Masekela, Miriam Makeba and Leftfield to name but a few, so the contents ought to be the genuine article. As with Wired, the loops and samples are provided within a single NKS library file (just over 460MB in size) and can only be accessed via the supplied version on Intakt.

Beat slicing within Afrolatin Slam — one slice or 56?

The preset instruments are presented in various categories based upon style. These include African, Caribbean, Afro-Latin fusion, Latin American and various 'miscellaneous others'. The biggest groups are the African and Latin American, with more congas, bells, talking drums and tambourines than you could shake a shaker at. In the main, the instruments are either Sampler-based kits or Beat Machine loops, where a single loop is presented in both a complete form and as slices mapped across they keyboard. Some loops are also given the Time Machine treatment. The various kits are well recorded, and the drums have plenty of body, but real aficionados of these musical styles will probably find the lack of multisamples somewhat limiting. Individually, the loops are excellent and I found that adding a couple of well-chosen percussion loops from this selection added just enough



flavour of the style to easily spice up a track based on more conventional instrumentation. While the loops themselves are good, I'm less convinced that they have been presented at their best with Intakt. The majority of the presets contain a single loop and, as there is no way to audition loops without loading the instrument into Intakt, browsing through the library to find the two or three loops that might work together is, frankly, a chore. This could have been partially resolved by providing more presets that contained Time Machine loops, with different loops mapped to each key. These could then be played alongside each other in much the same way as the construction kit presets within Wired.

Format Foibles There is no doubting the power, flexibility and sheer 'bang for buck' of many of the current crop of software samplers. However, a consequence of the expanded choice of sampler (hardware or software) has been a proliferation of commercial sample formats. While not that long ago audio CDs and Akai libraries used to dominate, now sample and loop libraries can appear in a wide range of formats — and format conversion can be both problematic and time-consuming. As typified by the five libraries described here, a more recent trend is to bundle a dedicated front-end with a sample library. Of course, this can be genuinely useful to the user, particularly for those still in the early stages of building their computerbased music system. For the company supplying the front-end, they will be hoping that the user acquires a taste for their software and moves on to their more upmarket offerings. In the case of these Intakt and Kompakt-based libraries, the advantage for the sample-library developers is that the material has a degree of copy protection — the samples can only be used with the supplied front-end, and that front-end has to be registered to a single user via a unique serial number. From the perspective of the user, however, I do have some concerns about this approach. Yes, I know that 'exclusive' sample libraries have been around for some time (Reason's Refills for example), but will those who have invested both money and time in their own particular choice of software sampler and loop manipulation tools really want to invest in a sample library that they cannot use with those tools? Some potential purchasers might find this restrictive.



Zero G Nu Jointz In contrast to Afrolatin Slam, Zero G's hip-hop-inspired Nu Jointz library makes better use of the construction kit possibilities of Intakt. With about a Gigabyte of sample data, there is a good range of downtempo loops featuring drums, electric and acoustic bass, guitars, piano, turntable scratches and even the occasional trombone loop for good measure. The preset instruments are clearly organised into a number of categories such as 'Construction Kits', 'Drum Loopz', 'Instrumentz', 'Single Hitz' and 'Bitz N Pieces' which, aside from rather too many 'z's, make it very easy to get at what you might want to audition.

Nu Jointz being used as a VST plug-in within Cubase SX. In both Intakt and Kompakt, each sample can be assigned to one of the eight stereo outputs.

The material itself is quite gritty, so this is definitely hip-hop territory rather than R&B. In particular, the drum loops, many of which have had some lo-fi processing applied, would sit quite happily within some of the less cheery contemporary hiphop and rap-based tracks around the charts at present. The majority of the drum loop presets include two or more loops mapped in various ways across the keyboard, and this does make it easy to mix and match between each small collection of loops. The construction kits, of which there are over 40, take this a little further. Most of these include a variety of drum, percussion and instrument loops with the occasional sample or three thrown in for good measure. In many of them, a 'master' loop is included to illustrate one way in which the individual loops might be put together. However, as with Wired, the fun is in triggering the loops from a keyboard, and just experimenting to see what can be created. While some of the kits work better than others, it is easy to see how a complete bed could be created from the loops provided and, by running two instances of Intakt, I was easily able to piece together a pair of the construction kits for further variety. If I could just invite Eminem around for a couple of hours to supply a top line... Whatever your views might be on music by construction kit, the way the Nu Jointz library has been implemented with Intakt certainly makes it easy and adds a certain creative element.

Conclusions Intakt and Kompakt are, on the whole, easy to use and flexible tools for loop or sample manipulation. In addition, the sheer size of some of the sample libraries discussed here suggests a huge amount of work on the part of their developers — there is plenty of material to inspire musical production — albeit that some of the libraries are better configured than others to work with the supplied front-end.



On a less positive note, do you want to be constrained to having to use a dedicated interface? This said, it is very difficult to argue that something like Nu Jointz, Stormbreakz or Vapor do not offer very good value for money and, if you are already a user of NI's software samplers, then you should certainly pick the library that suits your musical style and give it a serious listen. If several of the Intakt libraries might appear in your shopping cart, then make sure you also budget for the upgrade to the full version of Intakt, as the additional flexibility would be well worth the extra outlay.

East West Stormbreakz Stormbreakz offers a subset of East West's 6GB Stormdrum title, although at well over 2GB of sample data, Stormbreakz itself is not exactly small. Two of the six loop categories available in the larger title are featured here; Big Beats and Fastbreaks, although there is also a smaller selection of other loops and a number of individual drum hits. All the loops tend to use Intakt's Time Machine mode, while the hits use Sampler mode and are Stormbreakz makes excellent use of Intakt's organised into a number of Time Machine Legato mode. In this preset, electronic and processed kits, each every key has a loop assigned to it. of which is stored as a preset instrument. For the loops, the majority of the presets contain a number of different loops, each mapped to a separate key and with Legato mode switched on. Many presets contain 10 or more loops, while some also contain processed versions of the same loops for additional variety. As described elsewhere in this review, this organisation of the material is ideal for using Intakt to lay loops beside each other or in a sequence. In terms of style, the Big Beats are exactly that. Mostly mid-tempo, loud and aggressive, they would definitely work with straight rock (the Red Leppelin preset, for example) or more contemporary rock/nu-metal styles (such as Linkin Park). Played at lower tempos, some loop sets might also work in a hip-hop context. The moods are quite dark and ominous and the majority of the loops sound very live, with plenty of ambience. The loops in the uptempo Fastbreaks category are more electronic-sounding and busier than the Big Beats but, again, the mood is aggressive with just the right amount of lo-fi edge. At slightly slower tempos, these could also work in contemporary rock pieces while, at 140bpm at faster, some of these would not be out of place in dance-orientated material — provided it needed a little bad attitude. Throughout, both playing and sounds are excellent. There is plenty of choice provided and, given the way the loops are organised into the presets, Stormbreakz makes good use of Intakt's Time Machine mode. For those impressed (as I was) by this 2GB collection, East West offer an upgrade price to



the full version of Stormdrum. As well as more loops, Stormdrum also includes Kompakt and a bigger collection of single-hit drum kits. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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



In this article:

Opinion

ADK Vintage Valve A48 £400 pros Affordable in the UK. Multiple polar patterns. Solid sound that cuts through mixes.

cons May be too colouredsounding for some tastes. No pad or low-cut filter switches.

summary This attractively priced mic is designed to add flattering coloration to vocals. If you need to add density and edge to your voice, this could be a great choice.

information £399.99 including VAT. Turnkey +44 (0)20 7419 9999. +44 (0)20 7379 0093. Click here to email www.turnkey.co.uk www.adkmic.com

ADK Vintage Valve A48 Multi-pattern Valve Mic Published in SOS February 2005 Print article : Close window

Reviews : Microphone

This new mic combines a colourful valve sound with the flexibility of nine different polar patterns. Paul White

ADK's Vintage Valve A48 is, as the name suggests, a valve microphone, though the valve in question is a current 12AX7 dual triode, not a hard-to-find antique. The affordable price of this mic is of course due to its Chinese provenance, and it comes in a nice lockable camera case complete with shockmount, PSU, and seven-pin cable. Oddly though, the shockmount comes in two pieces, one part shipped fitted to the mic and the other in the camera case, so you have to arrange the included elastic belts into the suspension hooks yourself. If this is the Chinese trying to save on labour costs, it has to be a first! A rotary pattern switch on the PSU gives away that Photo: Mark Ewing this is a multi-pattern mic, in this case going from omni to figure of eight via cardioid, with six further intermediate stages. No response plot was provided in the box, just a spec sheet for a completely different ADK mic (A51 series!), but checking with ADK revealed the mic to have a nominal 20Hz-20kHz response and a sensitivity of 12mV/Pa. The maximum SPL is 128dB — perfectly adequate, but not exceptional — and the EIN selfnoise is 20dBA, which is slightly noisier than an equivalent solid-state mic of similar quality, but acceptable for close-miked vocals. However, there are much quieter mics around if you need them, with Rode's current range being about the quietest at the 'affordable to medium' end of the UK price range. Mechanically, the mic certainly looks the part, with a chunky nickel-plated brass body and a large grille protecting its one-inch-diameter, dual-diaphragm capsule.

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A knurled nut at the bass of the mic secures the shockmount, and the seven-pin XLR connector has gold-plated pins. There are no pads or roll-off filters, and the utilitarian PSU houses the conventional three-pin XLR, providing a balanced feed to your mixer or preamp. An IEC cable connects the PSU to the mains, though the review model came with a cable fitted with a European plug, not a British one.

Opinion This mic is designed with close-miked studio vocals in mind, so that's how I tested it. The sound is definitely coloured in a way that's designed to thicken the sound and also to add a little edge to it. This gives the mic a big, up-front sound, though it can exacerbate sibilance in those singers who have problems in that area. This mic also works fine for recording acoustic instruments, though the 'tube sound' has been slightly overdone for my own taste. The sound gets thicker still when the mic is used close up in cardioid or figure-of-eight modes, due to the proximity bass boost that affects these patterns, but I can still see a lot of singers liking the sense of density and authority this mic conveys. If you need a tube mic with attitude at a bargain price, then this is a model to try, but as with all 'character' mics, you need to try it to see if it suits your particular vocal style. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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ART Tube MP Studio V3

In this article:

Preset Voicings Verdict

ART Tube MP Studio V3 Valve Preamp Published in SOS February 2005

ART Tube MP Studio V3 £117

Print article : Close window

Reviews : Preamp

pros Easy to operate. Well-designed presets. Good overall sound.

cons No balanced-jack output option.

A pint-sized preamp with anything but a pint-sized sound.

summary The Tube MP Studio V3 is a good solution for anyone who simply needs mic preamp and DI facilities, but who also likes the idea of having different sonic characters available.

information £117.49 including VAT. Sonic8 +44 (0)8701 657456. +44 (0)8701 657458. Click here to email www.sonic8.com www.artproaudio.com

Paul White

The ART Tube MP Studio V3 is a more sophisticated incarnation of previous, similarly styled ART valve preamps, and it has the same preset OPL (output protection limiter) settings and Variable Valve Voicing tonal enhancements used in ART's DPS II. Hybrid tube/solidstate circuitry is used, the dual-triode 12AX7 tube being run from a relatively low voltage. Mains power comes via an AC adaptor. This preamp can accept mic-level signals, line signals, or high-impedance instrument signals on balanced XLRs or unbalanced jacks, while the output is available both on a balanced XLR and an unbalanced jack. A balanced output jack would have made more sense for connection to typical home-studio equipment. Up to 70dB of mic gain is available, and there's a clear, illuminated moving-coil VU meter to read the output level. Separate gain controls are fitted for the input and output, and there's a switchable 20dB gain boost, 48V phantom power, and phase reverse. A multi-colour LED shows that the unit is powered, as well as indicating clipping/limiting.

Preset Voicings The rotary control to the right of the meter is what makes this unit a bit different. Here you'll find 15 different switchable voicings (plus a neutral position), each

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using varying degrees of tube warmth, limiting, and subtle EQ. The presets are: Neutral, Vocal, Guitar, Bass, E-Keyboard, E-Guitar, Vocal (warmth), Valve, Multi, Vocal (warmth and OPL), Acoustic Guitar (warmth and OPL), Piano, Bass, Acoustic Guitar (OPL), Percussion, and Limit. When the signal level is high enough to approach clipping, the green power LED turns red, though this also signifies that the limiter is operating when OPL is activated. Other than selecting presets, the unit operates like any other mic/line preamp. All you need to do is decide whether or not you need the gain boost switch (which you can think of as being a pad in reverse), then set the input gain and the output level with the Clip/Limit LED and meter to help you along. Selecting a preset introduces a noticeable but still quite subtle tonal change (more noticeable on some of the electric-instrument settings) and most have been well chosen to suit their respective sources. However, with any unit of this type, it pays to try all the presets regardless of what sound you're working with, as you might find, for example, that the electric-guitar setting works well on a hip-hop vocal. You should also keep in mind that the output limiter is there primarily as a safety feature, though you can create more aggressive tonal effects by hitting it harder. Overall the sound has a solid, comfortable feel to it, with plenty of detail but no rough edges. The frequency response is quoted as 10Hz-30kHz (±0.5dB), and the equivalent input noise is -129dBA on the mic input, which is comparable to what you'd expect from a well-designed mixer mic preamp. The dynamic range is typically 100dB and the output level on the XLR is a very healthy +28dBu, which should keep even the greediest soundcard input happy. I liked the confident sound of this little preamp, and the presets really do offer a useful choice of characters without getting silly. It's almost like having several different mics to choose from. The metering is better than most small preamps of this type offer, and both the gain and level controls are nicely progressive over their entire range.

Verdict I found the ART Tube MP Studio V3 to be a versatile and sweet-sounding little mic preamp/DI box. While it might not have the finesse of some of the more upmarket, elite models that use high-voltage tube circuitry, it is rather nicer than the preamps found in most small mixers, and tonally a lot more versatile. I'm not a big fan of presets, but these do the job well without detracting from the musicality of the sound, and there's enough choice to suit just about anything you might want to record. Low-voltage tube circuits like this don't offer the same flavour of warmth as highvoltage designs, but the results here are still very musical and don't tend to go muddy as some low-voltage designs do. I think the Tube MP Studio V3 would make a great companion for a digital desktop studio, where its high output drive capability and tonal versatility should help the user inject a little welcome file:///H|/SOS%2002-05/ART%20Tube%20MP%20Studio%20V3.htm (2 of 3)9/22/2005 8:17:02 AM


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


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Behringer BCA2000

In this article:

USB 1.1 Operation Description In Use Conclusion Software & Installation

Behringer BCA2000 USB 2.0 Audio & MIDI Interface Published in SOS February 2005 Print article : Close window

Reviews : Computer Recording System

Behringer BCA2000 £180 pros Very good value. Has insert points. |ncludes a pair of Behringer's good Invisible Mic Preamps, with independent phantom power.

Completing Behringer's B-Control family, the BCA2000 combines audio and MIDI interfacing with routing and mixing features, and preamps — and all for under £200.

Surround monitoring option.

cons PC-only at the time of writing (late 2004). Built-in dynamics processing compromised. Analogue inputs always mixed to stereo en route to host software.

summary It's a shame about the current lack of Mac compatibility, and the compromises on the analogue inputs, but at what would be considered a fair price for an older-style USB 1.1 audio interface of this spec, the BCA2000 offers great value for money.

information £180 including VAT. Behringer +49 2154 9206 6441. +49 2154 9206 321. Click here to email www.behringer.com

Test Spec 3.06GHz P4 PC with 512MB of RAM & Windows XP. BCA2000 firmware version

Derek Johnson

There's a fairly widespread impression that USB 1.1 may not be up to audio and MIDI interfacing. Although that reputation is largely undeserved these days, it has to be said that those devices which are the most modest in their demands on USB bandwidth are also the most successful. But what about USB 2.0? This is a much faster format than earlier versions of USB, and if Behringer's latest computer-oriented release is any indication, the newer format is going to be found at the heart of very affordable gear.

Photos: Mike Cameron This is a busy unit for under £200, with dynamics processing (controls are on the left at the top), reasonable metering (top right), a clear indication of signal flow (see the block diagram with LEDs, below the dynamics controls), and built-in mic preamps. There are even dual headphone sockets and a guitar-friendly high-impedance input on the front edge of the unit.

The company's BCA2000 is part of their new B-Control family, the first two members of which were reviewed in SOS last month. The BCA seems a little anomalous in the company of these two MIDI controllers; it may be mounted in a box of the same shape, but it offers control of a different kind — over the audio and MIDI data moving in and out of your computer. However, if Behringer's early publicity led you to think that the BCA2000 is also a MIDI hardware controller, it categorically isn't — none of its knobs or faders transmits MIDI data of any kind.

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Behringer BCA2000

reviewed: v1.9.5.1.

What we have is a device that offers just enough analogue ins, comprehensive digital I/O, and a fair amount of analogue outs. The BCA2000 allows you to record up to eight audio channels, selected (with a few restrictions, of which more in a moment) from three analogue sources and two, four or eight digital audio channels, and routed into your computer via the USB 2.0 connection (if your computer can't handle USB 2.0, operation can take place over USB 1.1, but the slower nature of that protocol means you can interface fewer simulataneous channels — see the box above on the right for details). A maximum of eight channels of audio returning from your ASIO2- or WDM-compatible host software can be monitored in stereo, or in a multi-channel surround mix, via analogue or digital connections. Quality mic preamps, built-in dynamics processing and a zero-latency monitoring option add up to a pretty good spec for what is a budget interface, one that can operate at up to 24-bit/96kHz. And that's before I make mention of the built-in one-in/two-out MIDI interface, which allows up to 32 MIDI channels to be sent from your host software alongside the audio. Currently, that software will be running only on the PC: no Mac drivers were available at the time of this review (late November 2004), and there's not even basic Core Audio compatibility at the moment. It seems that Mac drivers are on their way, but with no ETA.

USB 1.1 Operation When using USB 1.1, rather than the faster USB 2.0, only three analogue input signals and one stereo digital signal can be recorded at once, although the analogue audio is actually a mixed stereo signal again, as discussed elsewhere in this review. The ADAT input can still be used, too, but only four of the eight channels are available. All digital formats supported in USB 2.0 mode are available here, bar the 24-bit/96kHz ADAT S/MUX format. Two audio outs are parallelled on the analogue and digital outs; again, four channels of an ADAT stream can be output as an alternative. The MIDI interfacing, in contrast, is available in full — in other words, two 16-channel streams.

Description As already noted, the BCA2000 adopts the same outward appearance as the rest of the B-Control family: it comes in a compact plastic table-top box, slightly sloped, with a raised panel at the top, and is finished in a gun-metal blue-ish grey, with white highlights. On the top panel, the layout is clearly understandable, with the controls for the analogue inputs, configured as one mono and one stereo, dominating the lower panel. Though superficially simple, there's more going on than you might think. First of all, the mono channel can route a mic-, line- or instrument-level input to your computer: the balanced line and balanced XLR mix inputs are at the rear of the interface, and there's a high-impedance guitar input located handily on the front panel, just under the mono input controls. The three input choices are selected via a pair of buttons. The mono channel is file:///H|/SOS%2002-05/Behringer%20BCA2000.htm (2 of 6)9/22/2005 8:17:08 AM

Behringer BCA2000

equipped with a trim control and a 100mm level fader, plus a pan pot. A final switch activates 48V phantom power when using the XLR input with a condenser mic, whilst Signal and Clip LEDs keep you visually informed of audio moving through the channel and any overloading that might occur. The mic-input electronics take the form of Behringer's Invisible Mic Preamp (or IMPs), worthwhile circuitry inherited from the company's UB mixer range. If the first thing you look for on computer audio interfaces is an insert point, in order to integrate your favourite hardware signal processing, then I'll save you the time: the BCA is so-equipped. That's a nice touch on an affordable interface. At first glance, the stereo input channel looks pretty much like the mono, with its trim control, 100mm fader, simple metering, 48V switch and so on. But it has more cunning flexibility, with routing switches that let you choose a second line or mic input, in mono, or allow you to hijack both line inputs for true stereo operation. Making this choice does throw up some issues, however. For example, this channel's pan pot works as a left/right balance control when working in stereo, and the mono input no longer has access to the (left) line input. It's also not possible to apply insert processing to both sides of the stereo signal appearing at this channel, since the mono channel's insert remains wired to its mic input. Thus it is possible to record a stereo performance, either miked or via the line ins, with insert processing, but only by setting the stereo channel for dual-mono operation. And although the BCA2000 can capture three analogue sources — one mono and the two sides of a stereo input — what goes to your audio software is always stereo. The theoretical three channels remain theoretical, and are always mixed to stereo. For users without external signal processing, Behringer have thoughtfully included some simple 'analogue input dynamics' circuitry which can be switched in and out of the input signal path. This consists of a one-knob noise gate and a one-knob limiter (plus threshold and limit indicator LEDs respectively), and is no less effective for its simplicity. Noisy basses or guitars are kept quiet when you're not playing, and rogue peaks are easily managed, helping you to avoid clipping while recording to your computer.

The rear panel is equally busy, as befits an interface with this many options. The two mic pres are accessed on the right via the XLRs, and the insert points are also to be found here. The multiple analogue outputs are in the centre above, and the digital I/O is to the left of these. Underneath are the USB connection, and the control-room and line-in jacks.

But there is one small problem: the dynamics processor is a fixed stereo device. All analogue audio passing into the BCA2000 will be processed if the circuitry is enabled, which is not necessarily what you'd want if you were recording two separate parts at once, such as might be the case if guitar and bass were being recorded, or a miked vocalist and stereo keyboards. It doesn't offer a dual-mono option, nor simultaneous stereo plus mono operation. Still, it's worth having.


Behringer BCA2000

The BCA's two input channels are joined by a master section, comprising a master level fader, monitor/control room level knob, and two headphone output knobs — the interface is equipped with two headphone outputs, plus monitor outputs, in addition to its main stereo output. One final knob is labelled 'monitor balance'; this rather usefully lets you alter the relative levels of the audio being playing into the BCA and the return from the host audio software, in some modes of operation. A handful of switches help manage the BCA's digital interfacing, which is actually pretty well specified. Optical and co-axial digital connectors are provided, and between them they can accept or transmit S/PDIF or AES-EBU digital audio; the optical connectors are also compatible with eight-channel ADAT, or four channels of the enhanced 24-bit/96kHz ADAT S/MUX format. Not all choices can be made on the panel: bundled control panel software provides access to the missing parameters. Interestingly, the co-axial output will still function in stereo when the optical interfacing is set to work in ADAT format. Digital channels 1-2 or 7-8 can also be output by the co-axial connection. Behringer didn't stop there: the front panel offers a mono mix audition button, plus control room/headphone mix dim and mute switches. Further monitoring sophistication, of a digital kind, is provided by a Monitor switch: this enables a 'direct monitoring' option, adding latency-free monitoring of audio returning from your software host. Metering is simple, with two 12-segment bar-graph meters for analogue audio, and six pairs of signal/overload LEDs for digital audio. The metering responds to input or output audio depending on the position of a switch. Other LEDs show the presence of MIDI data and USB operation, and a few more light up in a helpful front-panel flowchart. Round the back, there are some interesting discoveries. All the connections I've mentioned are located here: two XLR connectors, with an insert point each (though this functions on the line input as well), a pair of balanced line ins, the control room output (you can connect monitors here), co-axial and optical digital ins and outs, and three MIDI sockets. There are also no fewer than six analogue outputs. Two are balanced jacks, suitable for analogue mastering, while the remainder are simple, unbalanced, phono connectors. This may seem a strange option, in this digital world, but are provided essentially to allow surround monitoring to take place. The digital outs are surround-capable, too, if you have the right hardware. Not bad for a sub-£200 unit.


Behringer BCA2000

In Use Once everything's in place, working with the BCA2000 is straightforward. It'll show up as a MIDI and audio source in all Windows software — it even worked on Open Labs' Neko, from within their shell program. It was instantly available to play back audio from Propellerhead Reason, and was happily hosted by Cakewalk's Sonar and Steinberg's Cubase SX MIDI + Audio sequencers. Of course, you have to play around with settings a bit, to make sure all the audio channels are available, and to adjust latency. The BCA certainly exibits latency, and rather more than I would have expected. There are two issues: first of all, there's the delay that occurs between playing some audio via the interface and hearing it come back from the target software during recording or overdubbing. Then there's the delay between playing a note on a MIDI controller and hearing the audio output of a software instrument hosted by your target software being played back via the interface. Both types of delay are very much in evidence with everything freshly installed. Luckily, the recording and overdubbing problem can be easily dealt with, since the interface offers latency-free hardware monitoring options, as explained elsewhere. Tweaking the monitor balance knob and the track monitoring features of your host software should eliminate all audible delays. However, I found that although I could create acceptable latency values for playing software instruments via MIDI by tweaking buffers, the additional hit this caused on my CPU brought me closer to audio problems and break-ups than I would normally be happy with. I found settings that provided a compromise, but it wasn't the best. Should you wish to record more than three analogue sources independently, then adding an ADAT-equipped analogue interface would do the trick. One good candidate would be Behringer's own UltraGain Pro8 ADA8000, which features eight IMP-equipped XLR mic inputs feeding a single ADAT output. It's not expensive (£187 in the UK), and provides an ideal add-on for BCA2000 users. Of course, any ADAT-equipped device could be interfaced with the BCA, not just Behringer's. Audio coming back from your software can be monitored in stereo, or in a surround mix. Doing so will involve a little fiddling about in your host software, but you'll be able to manage a six-channel surround mix alongside a stereo downmix, plus stereo monitoring (and a switchable mono compatibility audition!). The digital outs support Dolby Digital/AC3 and DTS surround formats, if you have the right decoding hardware.

Conclusion USB 2.0 interfacing stood up well when I tried everything the BCA is designed to be capable of, recording multiple audio channels, monitoring in surround, and playing virtual instruments while beaming MIDI data out of the host software. Sessions with a normal level of activity were glitch-free once I'd adjusted latency in my hosts and on the BCA. Lack of Mac OS X drivers aside, the BCA2000 is a fine mid- to low-price audio interface, and the use of USB 2.0 allows more simultaneous audio channels to file:///H|/SOS%2002-05/Behringer%20BCA2000.htm (5 of 6)9/22/2005 8:17:08 AM

Behringer BCA2000

be interfaced than you would achieve with a USB 1.1 interface, to say nothing of the dual-channel MIDI stream. The input and output circuitry sounds great, too — I was a fan of the IMPs. The unit is very flexible, though serious users may need to add an external box or two to get the most out of the BCA: those eight channels of digital audio really do cry out to be exploited, and require further hardware. Some of the apparent flexibility is also compromised by the steadfast stereo nature of the analogue input. Niggles placed firmly to one side, the BCA2000 shows what USB 2.0 can do for music, and offers a lot of interface for the money. And if the retail price seems keen, some of the UK street prices have to be seen to be believed! Mac users excepted, be prepared to want it once you've tried it.

Software & Installation There are a handful of functions that can't be accessed from the front panel, requiring a little software widget. This is supplied, along with drivers, on a CDROM. Installing the ASIO-compatible driver can be a little odd: a couple of errors will arise on most users' systems, but according to Behringer, this is normal, and we're supposed to simply ignore them. Once installed, the driver works fine, and plants a little control panel icon on the Windows XP task bar. This panel lets you customise latency, select the digital clock source, choose an ASIO monitoring mode, and set up the two-channel digital output format (either AES-EBU or S/PDIF). In addition, you can customise the meter ballistics to offer peak hold or faster decay reading. The rest of the control panel offers information: a second page summarises what ins and outs are available with USB 2.0 and USB 1.1, and a third is simply a useful block diagram. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



BIAS Sound Soap Pro

In this article:

The Correct Use Of Sound Soap Hum & Rumble Washing Machines Broadband Noise In Use

BIAS Sound Soap Pro Noise-reduction Software (Mac/PC) Published in SOS February 2005 Print article : Close window

Reviews : Software

BIAS Sound Soap Pro £529 pros Generally easy to use. Effective against most broadband noise as well as hum, clicks and crackles.

cons The spectrogram display isn't always easy to interpret.

summary Sound Soap Pro is a good denoising solution for anyone wanting flexibility without undue complexity.

information £529 including VAT. SCV London +44 (0)20 8418 0778. +44 (0)20 8418 0624. Click here to email www.scvlondon.co.uk www.bias-inc.com

A cross-platform package from BIAS offers highquality denoising at an affordable price. Paul White

BIAS's Sound Soap Pro is a four-in-one suite of noise-reduction plug-ins for VST, RTAS, Audio Units, Audiosuite, and Direct X host applications on both the Apple Mac and Windows platforms, designed to deliver highquality audio clean-up capability with the minimum of user complication. Sound Soap Pro is based on the technology behind BIAS's original Sound Soap, but this version is rather more sophisticated and allows the user access to parameters that were preset in Sound Soap. Each of its four modules is designed to deal with a specific kind of noise, though one of these modules is really just a conventional noise gate, so most of my attention will be directed towards the other three. These are designed to tackle hum and rumble, click and crackle, and broadband noise. The four separate denoising elements in the Sound Soap Pro suite are all accessed via a single, tab-based user interface, with a specific denoising tool accessed by each tab. As you have probably gathered from the description of the modules, Sound Soap Pro is designed to reduce unwanted hiss such as tape or circuit noise, electrical hum, click and crackle (both vinyl surface noise and digital clicks), though it can also deal with many other types of wide-spectrum noise providing they are constant in level, such as camera noise. Though the four modules are very different, some elements are common to all, such as the spectrogram display, Global Presets, a Noise Only audition mode and bypass. Where the host application allows, parameters may also be automated.

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BIAS Sound Soap Pro

The Correct Use Of Sound Soap Sound Soap Pro, once installed, appears like any other effect/processor plug-in and I tested it in its Audio Units incarnation via the insert points in both BIAS Peak and Logic Pro. The best way to apply the processes is simply to follow the tabs, which are labelled from left to right as Launch, Hum & Rumble, Click & Crackle, Broadband and Noise Gate. Where there is adequate processing power, the tools can be set up individually then run simultaneously. If a tool is not needed, it can be bypassed to save on CPU power. Because Sound Soap Pro comprises four different processing tools, presets can be saved for individual tools or for the complete suite of tools, though where the host software is capable of saving presets, only the settings of the entire plug-in are saved there. A compare feature allows different sets of settings to be compared, and while most plug-ins only offer 'A' and 'B' settings, Sound Soap Pro has four. A PDF manual is supplied, and clicking the Help button uses your web browser to contact the BIAS web site, where you can obtain various additional tutorials on Sound Soap Pro. The Noise Only button is a means of monitoring that part of the signal that Sound Soap Pro deems to be noise based on the current control settings. If this contains wanted signal as well as noise, then you have the opportunity to revise the parameter settings so as to remove only noise. The spectrogram readout utilises a colour display that scrolls as the audio file plays. The white areas depict the original audio signal, while red shows the processed audio after passing through Sound Soap Pro. Reset and display Freeze buttons are available. In essence, the spectrogram shows where in the frequency spectrum noise is occurring and gives an indication as to which frequencies are being removed by the processing, though I didn't find it awfully helpful as I wasn't quite sure what I was looking at half the time. Beneath the spectrogram is a horizontal fader used for scrolling backward and forward in time and for increasing or decreasing the display zoom magnification. The frequency scale of the spectrogram changes according to the type of noise module being used — for example, the hum and rumble scale extends only up to 4kHz while the broadband noise tools show the complete audio spectrum.

Hum & Rumble The hum and rumble tool includes a useful hum meter which displays the level of hum passing through the filters as you adjust the frequency, so you can use this to visually tune the Hum Frequency slider for the strongest signal over its 20Hz to 200Hz range — this indicates you've tuned to the fundamental frequency of the hum, and all the harmonic filters are derived from multiples of this. Fractions of a Hertz may be typed in but I couldn't find a key command that would let me scroll the value at any higher resolution than 1Hz steps. A Q (bandwidth) slider sets the file:///H|/SOS%2002-05/BIAS%20Sound%20Soap%20Pro.htm (2 of 6)9/22/2005 8:17:14 AM

BIAS Sound Soap Pro

width of the notch filter from 10 to 50. High Q values create narrower notches, but unless the hum frequency is very stable, it may drift out of the range of the filter if the Q is set too high, so the trick is really to set it as high as you can without the process becoming ineffective. A Depth slider sets the amount of attenuation in the filter band up to a maximum of 50dB. In all probablity, attenuating only the fundamental frequency will reveal further problems at harmonics of that frequency, so the Harmonics slider is used next to attenuate whatever number of harmonics (up to nine) is necessary to get a subjectively good result. Again, the fewer harmonics you notch out, the less the wanted sound will be affected, though with such narrow frequency bands, the filtering is actually pretty transparent. The Harmonics Tilt knob allows the degree of cut to be made progressively less at higher frequencies, where the hum harmonics are usually less prevalent. The Rumble Reduction slider is designed to attenuate very low-frequency noise using a highpass filter that can be set between 20Hz and 100Hz. This has a slope of 12dB/ octave and a Q value of 0.5. Clicks, crackles and pops can't be addessed by simple filtering, as the artifact has to be identified, removed and then filled with a spectrally correct signal based on what the signal looks like either side of the click. Clicks can be caused by dirt and scratches on vinyl records, and can also be introduced in digital audio by copying and pasting without crossfading the edits; sometimes they also appear randomly (and infuriatingly!) because of clock sync problems or data errors. SSP's click and crackle module has separate sliders for click and crackle detection, so if you have both large clicks and smaller crackles, they can be treated at the same time. The Click Threshold slider adjusts a detector threshold over the range 0 to 25 dB to determine which clicks should be treated and which ignored. Moving the slider to the left will remove progressively more (lower amplitude) clicks. The Crackle Threshold slider does much the same for crackle and has the same range. To aid setting up, a Click Meter illuminates in red to show that a click is being processed. This is really a very simple part of the system to use, but again you need to check the noise-only signal to ensure that nothing is being removed that shouldn't be.


BIAS Sound Soap Pro

Washing Machines Sound Soap Pro is a dual-platform application and on Apple machines it is recommended that it be run on a G4 or G5 Macintosh computer with a minimum processing speed of 500MHz. It only runs under OS X (10.2 or later) and can be used within any host application that supports the RTAS, VST, Audiosuite, or Audio Units plug-in protocols. Windows users require a Pentium III or Pentium 4 computer with a minimum of 800MHz processing speed, and Sound Soap Pro can run under Windows XP Home or Professional Editions, again with the proviso that you are running a compatible RTAS, Direct X, Audiosuite or VST host application. Hard drive and RAM requirements are modest but you will need a bare minimum of 128MB RAM (256 or more is recommended) and at least 20MB of available hard drive space. Authorisation is via the BIAS web site but there's also a USB key with this version for added security.

Broadband Noise The types of noise that can be tackled by the broadband noise section encompass any kind of constant noise that covers a broad frequency spectrum, common examples being tape hiss, air-conditioning noise, camera noise, lighting buzz and mechanical noise from a recorder. It's probably fair to say that the broadband noise-reduction tool is the most complex of the modules, but BIAS have still done as much as possible to make it easy to use. When using this tool, it is particularly important to listen to the noise-only signal to make sure nothing wanted is being thrown away. If it is, you can make manual adjustments to the threshold sliders. Sound Soap Pro's broadband noise reduction works on the familiar principle of splitting the audio signal into a large number of frequency bands, then processing each independently before recombining them. Unlike the other elements of the system, for this process to work, it has to know what the noise spectrum looks like and how loud the noise is, so the material ideally needs to contain an exposed section of noise that you can use to create a 'noise fingerprint'. A section of noise as short as half a second or even less can be analysed manually using the Learn button, or using the timed mode indicated by a stopwatch icon. Sometimes looping the noisy section is the best way to work. Playback must be initiated in order for the learning process to work, after which the noise reduction automatically switches on. The 12 Threshold sliders then move to the appropriate position to subtract out the noise based on the average noise profile created during the Learn phase. Note however that these 12 sliders are simply for display and adjustment — there are actually many more frequency bands under their control. The 12 slider meters indicate the level of audio in each frequency band and the frequencies covered by the bands adapt optimally depending on the measured noise profile's shape. Once the noise has been learned, the Threshold/Reduction View slider allows the display to show the Threshold sliders, the Reduction sliders or both at the same time, where the maximum reduction level is -40dB and the threshold can be set from 0 to -70dB. The Reduction sliders control the amount of noise reduction being applied to each frequency range; by default, all file:///H|/SOS%2002-05/BIAS%20Sound%20Soap%20Pro.htm (4 of 6)9/22/2005 8:17:14 AM

BIAS Sound Soap Pro

the sliders are locked together and moved as one, but they can be temporarily unlocked to allow them to be moved to different relative positions. Furthermore, they can all be unlocked and left so that they can move independently, or you can choose to lock some sliders and not others. The process can be envisaged as having a separate expander/gate for each of the frequency bands being processed, with Attack and Release controls having initial default values of 75ms and 100ms respectively. Sometimes it is desirable for the process to come in or decay faster in some parts of the audio spectrum than others, so Attack Tilt and Release Tilt knobs provide an easy means to adjust the ratio of attack and release between the low and high frequency bands. The last line of defence is the gate, and like all gates, it can only eliminate noise when no wanted signal is present, as everything gets muted or attenuated. If the noise is low enough to be masked by the wanted sounds, then the gate can be used to achieve complete silence during pauses. It has a display that shows the audio waveform as well as the gate threshold. A Threshold slider sets the level above which the gate opens, and the amount of gain reduction applied when the gate is closed is set using the Reduction slider in the usual way. A Threshold Attack knob adjusts how quickly the gate opens once a signal has passed the threshold, and the Release control sets how fast it closes again once the signal has dropped back below the threshold. As far as I can see, this is a quite conventional gate.

In Use My tests involved cleaning up some tracks I'd taken from a favourite vinyl record, and though there was little hum to deal with, there were clicks and crackles in abundance. The clicks were dealt with very effectively without any obvious artifacts, and I'd say that the crackle level was reduced by a subjective 95 percent without messing up the wanted sound. You have to be a little more careful with the broadband reduction tool: my strategy was to move the threshold (locked together) up while listening to the noise-only signal, and then see how high I could get it before recognisable audio


BIAS Sound Soap Pro

started to appear. At this point I backed it off slightly so that noise was once again the only thing being monitored; I then adjusted the noise-removal Depth sliders so that the noise was just low enough to be inaudible beneath the quieter sections of the music. Providing you apply only as little treatment as is necessary using this module, the level of apparent hiss can be reduced very significantly without compromising the audio quality, but if you try to go too far, you can start to remove top end from the audio being treated. Once all the settings are correct, all the processes can be bounced into a new file containing your denoised audio. Sound Soap Pro strikes a sensible balance between lack of complexity and effectiveness, though I'm still trying to find which key I need to hold down to let me roll through the hum frequencies in 0.1Hz steps instead of 1.0Hz steps. The click and crackle tools are certainly effective in improving the sound of vinyl transfers and the broadband noise tool is as easy to use as any other snapshotbased system. As a package for restoring vinyl, removing clicks from digital recordings or reducing the effect of camera noise, Sound Soap Pro is pretty straightforward and generally effective. Inevitably there will be audible artifacts that it doesn't recognise as clicks or crackles, but on the whole it does a very good job and catches most things without too much of a problem. If you need something that's more flexible than a basic clean-up program but you don't want the complexity of a high-end, all-singing solution, then Sound Soap Pro is an excellent compromise. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



DAV Electronics BG4

In this article:

Construction Controls Down The Garden Path

DAV Electronics BG4 £1750

DAV Electronics BG4 Compressor/Limiter Published in SOS February 2005 Print article : Close window

Reviews : Processor

pros The Decca sound character is evident. Gentle and transparent dynamic control. Simple switched controls enable easy operation and accurate resetting.

A premium compression sound lurks behind the nofrills cosmetics of this new Broadhurst Gardens processor.

Flexible side-chain EQ facilities.

cons No independent channel bypass. Gain Make Up control is always active, even when compressor is bypassed. No external side-chain access. No handbook supplied with review model — again!

Photos: Mark Ewing

Hugh Robjohns

The DAV Electronics range of Broadhurst Gardens audio processors has already made a couple of appearances in the pages of Sound On Sound. The BG1 dualA simple dual-channel channel mic preamp was reviewed in SOS December 2003 and the BG5 channel compressor/limiter derived strip appeared in August 2004. The latter unit combined elements from several from bespoke Decca designs BG processors including the BG1 preamp, BG3 equaliser, and BG4 limiter, and dating back to the 1970s and carefully optimised for serious the last of these is now the subject of this review.

summary

acoustic music applications.

information

The Decca Recording Company was, at one time, based at Broadhurst Gardens in West Hampstead, where Mick Hinton — the owner and designer of DAV £1750 including VAT. Electronics — was an employee. Mick has used the prestigious address on the KMR Audio +44 (0)20 BG product range to signify the heritage of these products, which are based 8445 2446. directly on original circuit designs and principles used by Decca (albeit updated +44 (0)20 8369 5529. to use more modern components). A common characteristic of the BG Click here to email processors is a slightly 'larger than life' sound quality, and all the units boast www.kmraudio.com www.davelectronics.com flawless technical specifications and switched control functions to enable all settings to be recreated precisely. As I have said in previous BG reviews, the overriding impression of all of the BG units is of equipment designed to do the required job very well, but without being flash or equipped with 'marketing features'. These are products undoubtedly

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DAV Electronics BG4

aimed at high-end acoustic music applications and for serious engineers — and the company already claims such high-profile users as the former Sony/Whitfield Street Studios, The Decca Record Company, Townhouse Studios, Metropolis Studios, the Master Room, and many others.

Construction The BG4 dual compressor/limiter is contained in a 1U rackmounting box with a modest 250mm case depth behind the rack ears. As with other BG products, the case itself is unpainted steel, with a matt-black front panel carrying white and light-blue control legends. Each unit is constructed by hand, but to high standards, although a peek inside the box does betray a slightly old-fashioned approach to wiring and fabrication methods. Also, just as with the BG5, there was no handbook supplied. The rear-panel connections comprise two pairs of XLR connectors for the linelevel stereo inputs and outputs, plus an IEC mains inlet which incorporates a fuse holder and mains voltage selector. Internally, the majority of the electronics are housed on a single large PCB, with two smaller daughterboards mounted vertically behind the front-panel controls. All components are standard sized — no surface-mount technology here — and everything is well spaced out to avoid unwanted interaction and crosstalk. The circuit design is apparently derived from bespoke equipment developed and used by Decca from the late '70s, and which acquired a fine reputation for having unusually low noise and distortion. In this updated version, a CA3083 highcurrent transistor array is employed in each channel as the basis of the voltagecontrolled amplifier (VCA), while the supporting circuitry for each channel includes a CA3140, an LM1458, and a TL072 — all three being op amps with distinct properties obviously carefully chosen for their specific role. The audio path and output buffering are handled with four OP275 high-quality op amps, and the ubiquitous SSM2017 provides the balanced input (this chip being socketed for easy replacement).

Controls The controls are arranged in separate but identical groups for each channel, enabling either independent two-channel or linked-stereo operation, as required. Each channel is equipped with a small black button (and associated red LED) to select the operating mode (compression or limiting), and a quintet of rotary switches. The compression ratio switches between a nominal 2:1 (apparently with a soft-knee characteristic) for the compression mode and 10:1 for the limiter mode. The limiter isn't a 'brick wall' design, and when driven by loud transients there is some overshoot. While this may have been acceptable thirty years ago in the heyday of analogue recording, digital recorders are considerably less file:///H|/SOS%2002-05/DAV%20Electronics%20BG4.htm (2 of 5)9/22/2005 8:17:21 AM

DAV Electronics BG4

tolerant, so it would be wise to set the limiter threshold to leave a few decibels of headroom, thereby avoiding possible A-D overloads. The dynamics section itself is manipulated by three rotary controls. Gain Make Up is the first and provides up to 16dB of gain in 2dB increments. The second control adjusts the recovery time between 0.3s, 1s, or 3s, while the third sets the Threshold from zero to +20dBu in twelve steps. The first increment is from 0dBu to +1dBu, followed by 2dB intervals up to +19dB, with +20dBu as the final position. When I encountered this threshold control on the BG5 Channel unit I initially assumed from the panel legend that the scale went from -20dBu to 0dBu — meaning that you would have to turn the control anti-clockwise to lower the threshold and thus increase the compression. In fact, the control is scaled from +20dBu to 0dBu, and you have to turn it clockwise to lower the threshold and increase the compression. With hindsight, the latter makes perfect sense, but I was thrown completely the first time I used it! The last two controls weren't incorporated in the BG5 channel's incarnation of this design. The first selects a side-chain filter type (low pass, high pass, or wide band), and the second determines the turnover frequency (0.1, 0.25, 0.5, 1.0, 2.5, or 5.0kHz). The final element is a green bar-graph meter to show the gain reduction — no input or output level metering is provided at all. The meter is scaled to show a maximum of 12dB of gain reduction with 1dB increments to 4dB, and 2dB increments thereafter. In the centre of the unit are two black buttons (with associated red LEDs again) that affect both channels — although the panel layout suggests they relate to the right-hand channel. The first activates the stereo-link mode, which couples the side-chains of both channels to ensure stereo images remain stable during compression or limiting. The second switch is labelled Limiters Off, and this disables the VCAs — a kind of bypass mode, if you like. The fact that this affects both channels simultaneously makes it slightly harder to use this machine as a pair of independent dynamics processors, but in practice I doubt this will be a concern in the applications for which it is likely to be used.

Down The Garden Path Since the dynamics section of the BG5 Channel Unit was based closely on the BG4, many of my earlier comments regarding the operational 'quirks' of the BG5's dynamics section apply equally here as well. In particular, the Gain Make Up control is always active, even when the compressor/limiter processing is switched out with the Limiters Off button. This makes it harder than it really should be to assess the appropriateness of the dynamics processing, since the unprocessed signal will be louder than the processed signal. file:///H|/SOS%2002-05/DAV%20Electronics%20BG4.htm (3 of 5)9/22/2005 8:17:21 AM

DAV Electronics BG4

With its fixed 2:1 ratio and soft-knee characteristic, the compressor is both subtle and transparent, and ideally suited to the gentle control of complex instruments or mixes without it becoming heavy-handed. The simple three-position releasetime control seems entirely adequate, with well-chosen options, although I suspect a degree of automatic programme-related adjustment is also going on here. Personally, I'm not a big fan of using compression while recording. While it was a necessary evil in the analogue days to maximise the relatively poor signal-tonoise ratio of magnetic tape, modern 24-bit digital systems don't require such cosseting. To my way of thinking it makes far more sense these days — at least when multitracking — to record everything flat and unprocessed (other than correcting obviously unwanted LF rumbles and so forth), and then to determine the necessary dynamic control (if any) afterwards in the context of the complete mix. The BG4 is ideally suited to such post-production dynamic control, either on individual instrument tracks, across stereo submixes, or even across the entire mix. However, if mixing straight to stereo on location, or if mixing a live concert with potentially unpredictable levels on some channels, the BG4's ease of use and inherently transparent nature could prove invaluable. Subtlety and gentleness is the name of the game here though — don't expect to find any rock & roll attitude here, no matter how hard you drive it! The limiter affords a useful degree of peak control — again without trampling all over the sound character — but I detected some overshoot on loud, brief transients. Consequently, I would suggest taking the threshold calibration as an optimistic guide rather than as an absolute protection level. When used with an AD calibrated for 0dBFS at +18dBu, I obtained reliable results from the limiter with the threshold set to 15 (a setting of 17 produced occasional digital overloads). Although there is no external access to the dynamics side-chain, the unusual side-chain equaliser proved surprisingly effective in a variety of situations. In the wide-band mode, the dynamic processing reacts to all frequencies equally — as with most compressors. However, when necessary, the side-chain can be desensitised to extreme high or low frequencies by choosing the low-pass or high-pass filter modes, respectively, and tweaking the turnover frequency control as required. This feature can be used to help prevent the dynamics processing from reacting excessively to, for example, bass drums or cymbal clashes when these are present within a composite mix. Overall, then, this a high-quality compressor/limiter ideally suited to delicate serious music applications and for use by the kind of engineer who still wears tweed jackets when mixing — that'll be me then... The BG4 doesn't draw attention to itself, either in terms of its sound character or its aesthetic design, but that is sometimes precisely what is needed.


DAV Electronics BG4

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



Digitech GNX4

In this article:

Digitech GNX4

What's Under The Guitar Workstation Floorboard? Published in SOS February 2005 Model Performance Drumming Up Support Print article : Close window Compact Flash Multitracker Reviews : Multitrack Recorder USB Audio & MIDI Interface All-round Performer

Digitech GNX4 £599 pros Excellent range of amp modelling for both guitar and bass use. The term 'workstation' is accurate — the GNX4 is a real jack of all trades. Very good value for money.

cons No Compact Flash card provided. For just live/studio amp modelling, there are cheaper, more streamlined options available.

summary The Digitech GNX4 packs an awful lot of features into a very small space. If the very full feature list matches your particular needs, then there is no doubting the quality of the amp modelling or the value for money.

information £599 including VAT. Harman Pro UK +44 (0) 1707 668222. +44 (0)1707 668010. Click here to email www.harmanprouk.com www.digitech.com

Test Spec X-Edit editor software for Windows v1.21

Is it an amp modeller? Is it a multitracker? Is it an audio/MIDI interface? We stomp the facts out of Digitech's new box... John Walden

Unless you have just returned from an extended period in a sensorydeprivation unit, it will have been impossible to miss the rise and rise of the guitar amp modeller, whether hardware or software based. Indeed, there are probably a whole generation of recording guitarists who have never even felt the urge to stick a microphone in front of a real guitar Photos: Mike Cameron amplifier/speaker-cabinet combination. In terms of hardware amp-modelling units, two formats dominate: desktop/rack units such as the PodXT or Behringer V-Amp, aimed predominately at studio use, and pedalboard units such as the new PodXT Live or the Vox Tonelab SE, which are equally at home either live or in the studio. Digitech, in one form or another, are a well-established player in this market. With products such as the rack-based 2120 Artist, the J-Station (under the Johnson brand name), the relatively budget RP-series of floor units, and the Genesis desktop amp modellers (the Genesis 3 was reviewed by Paul White back in SOS October 2001), they have a good pedigree. The GeNetX ampmodelling technology used in the Genesis series has continued to develop and has now found its way into the GNX floor units. Latest in this line, and at the top of the pile in terms of features and price, is the GNX4. However, the GNX4 is more than just a studio amp modeller. While its floorboard format means it is a live-performance unit, the GNX4 also includes a drum machine, an eight-track audio recorder, a mic preamp with phantom power, and a USB-based audio/MIDI

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Digitech GNX4

Cakewalk Pro Tracks Plus v.2.2 3.4GHz Pentium IV PC with 1GB RAM running Windows XP Pro (SP2). Echo Mia 24, Egosys Wami Rack 24, and Yamaha SW1000XG soundcards.

interface with software supplied for both PC and Mac. Apparently, there wasn't room for a kitchen sink, but it is pretty clear why Digitech have given the GNX4 the 'guitar workstation' title.

What's Under The Floorboard? As you can probably tell from the photographs, the hardware layout consists of three elements. The first of these is made up of the seven large-format (well, large enough if you don't wear size-12 boots!) footswitches and the expression pedal. These controls provide access to key features during live performance — although they also serve a useful function in the studio, as they can be configured to provide hands-free control for a number of functions. The second operational area, filling the remainder of the top surface, contains the controls for the eight-track recorder (on the left); the amp, cabinet, and effects modelling (centre); and the drum machine (on the right). Sensibly, these are all well out of the way of all but the clumsiest stamp in the dark. The third hardware element is the equally well-stuffed rear panel that contains all the input and output connectivity. Aside from the usual guitar input and quarter-inch line and headphone outputs, perhaps the most notable features here are a pair of balanced XLR outputs, a USB connection, a slot for inserting a Compact Flash card and, most surprisingly, a balanced XLR mic input supplied with phantom power. However, the package doesn't stop there. Included in the box is the X-Edit patch editor utility (for both PC and Mac), sequencing software (Cakewalk's Pro Tracks Plus for PC or BIAS Deck SE for Mac), Lexicon's Pantheon reverb plug-in (again, for both PC and Mac platforms), and some decent printed documentation. A USB cable and the wall-wart external power supply are also included. Perhaps the only notable omission is a Type-1 Compact Flash memory card — users are left to provide their own for use with the built-in eight-track recorder. Installation of the drivers, X-Edit and Pro Tracks Plus all proceeded as described in the documentation using my PC test system, and 15 minutes from opening the box I was ready to experiment with any of the considerable array of the GNX4's functions.

Model Performance Despite the truckload of features, the core job of the GNX4 is, of course, to provide convincing guitar-amp and cabinet modelling. While the quality of the modelling and the processing power available have moved on, the basic functions of the GNX4 modelling are not that dissimilar to those of the Genesis 3 reviewed by Paul back in October 2001. For example, the GNX4 includes the same Warp option, where it is possible to blend between two different amp/ cabinet model combinations to produce new tones.


Digitech GNX4

As with most of the up-market amp modellers, the GNX4 provides models of a large number of effects. These include various classic distortions, wah, compression, pitch-shifting, noise gate, chorus/modulation effects, delay, reverb, and a selection of more synthlike processes such as Auto Ya, Ya Ya, and Synth Talk. In addition, pickup modelling allows a single-coil guitar to emulate a humbucker and vice versa. Side-by-side comparisons with my own PodXT suggested little to pick between the two units in terms of the range of amp and cabinet models or the quality of the modelling, and I'd be more than happy to use either unit in a recording context. The quality of the effects is also excellent and, if anything, the range of effects on offer with the GNX4 is somewhat wider than that supplied by default with the PodXT. While guitar tones are always a matter of personal taste, on the whole the presets are very usable and do a good job of demonstrating the capabilities of the GNX4. All the sonic bases are covered, from sparkling cleans (such as the Twinverb preset) through to murderous metal (try Rectify) and the usual line-up of tones in the style of well-known guitarists are present (Carlos, Angus, and Eddie VH for example). I also found a number of the more effectsbased presets served to get the creative juices flowing (such as Stutter, Trembo, and Divebomb). Of course, for any floor unit like this, a minor irritation can be accessing the various hardware controls when something other than switching presets or turning effects on/off is required. Fortunately, the GNX4 is supplied with the very competent X-Edit software. While this does not provide access to all the functions of the unit (for example, the audio routing or the eight-track recorder), for patch editing it is very comprehensive. Editing is performed through three main screens. The GeNetX screen provides access to the two channels of amp/ cabinet modelling and the Warp function — the two models can be blended via the X-Y controller at the centre of the window. The Effects screen does pretty much what is says on the tin, while Expression Assign allows the footpedal and the two LFOs to be routed in various ways. While patch editing using the GNX4 control surface is straightforward enough, for those with a suitable computer XEdit is even better and will certainly reduce the potential for backache!

Drumming Up Support


Digitech GNX4

The built-in drum machine contains more than 100 internal patterns, covering styles from rock, funk, and jazz through to Latin, hip-hop, and dance. Fortunately, there are also eight different drum kits suitable for use with these various musical styles. The combination of patterns and drum sounds gives the feel of a respectable GM-style drum machine — although solid enough for practice purposes or assembling a song demo, something more sophisticated might be required for the final master. While patterns cannot be chained, there is a facility to use the footswitches to select patterns and to set the tempo while in Recorder mode. However, the drum machine will play back MIDI files, either stored on a Compact Flash card or received via MIDI from a sequencer. Interestingly, the drum machine also includes a function to play back an MP3 stored on Compact Flash. This would certainly be useful for practice or for a solo performer during a live performance. MP3 playback cannot be used simultaneously with the drum-machine functions.

Compact Flash Multitracker The built-in eight-track audio recorder is fairly simple in operation, but can record either one or two tracks simultaneously at 16-bit/44.1kHz resolution, provided that the user acquires their own Compact Flash card. A suitable 256MB card can be purchased for under £30 and would provide about 48 track minutes — just about enough for a couple of threeminute eight-track hits! Either a drum pattern or MP3 files can be triggered to playback with a particular song, and these can also be recorded as audio to a mono track or stereo pair. Recording to a particular track or track pair simply The bundled X-Edit software editing utility requires you to arm the tracks using provides easy access to the GNX4's the appropriate track buttons and to modelling and effects parameters. press the Record button. Punching in and out is provided, and can be done 'hand-free' via the footswitches. However, subsequent audio editing is somewhat limited, aside from the ability to digitally bounce tracks. Usefully, it is possible to monitor the signal being recorded with effects applied while recording 'dry'. This also forms part of the re-amping function, where a guitar part is monitored through all the amp modelling and effects processing, but is actually recorded with this processing bypassed. This recording can then be replayed through the amp/effects modelling and the final guitar sound adjusted to taste. Given that the GNX4 is a 'guitar workstation', the inclusion of the mic


Digitech GNX4

preamp might seem like a bit of a gimmick. However, in tests recording vocals and acoustic guitar with a mid-priced condenser microphone, it actually produced very respectable results. While I would not perhaps choose to use it over a dedicated channel strip or audio interface with phantom power, it certainly does make the built-in eight-track capable of surprisingly good demos. One further useful feature is the Compact Flash Storage mode. With this mode engaged, the Recorder is disabled, but if the USB connection is present then the Compact Flash card becomes visible as a removable storage device and files can then be transferred to and from the host computer. This would obviously be useful for data backup, for moving MP3 and MIDI files to the GNX4; or for moving audio form the GNX4 for further editing/mixing within a more powerful software editing environment.

USB Audio & MIDI Interface As mentioned earlier, the GNX4 acts as a USB-based four-in/two-out audio interface to either a PC or Mac, and is capable of 16-bit or 24-bit operation at 44.1kHz. MIDI In and Out sockets are also provided. I tested the PC version via Pro Tracks Plus. Audio routing within the GNX4 allows the user to specify which of the inputs (guitar/mic/ line-in) get passed to the PC via USB — either with or without effects. Audio Cakewalk's Pro Tracks Plus integrates well from the drum machine can also be with the USB-based MIDI and audio I/O on the GNX4 hardware. In the foreground of the passed to the PC via USB. Pro Tracks Plus is a functional enough sequencer, screen above you can see the included Lexicon Pantheon reverb plug-in. supporting up to 63 audio tracks as well as MIDI tracks. It is, however, limited to Direct X-format effect or instrument plug-ins plus Rewire support. That said, someone new to computer-based recording is unlikely to see the lack of VST support as too much of a problem. The supplied Lexicon Pantheon reverb plug-in also does a very good job. I was able to get down to 10ms latency via the supplied WMD drivers, and this gave a suitably responsive performance for the Tassman SE DXi synth supplied with the sequencer. Latency while recording audio via the GNX4 is not a major issue, as direct input monitoring can be toggled on and off within Pro Tracks Plus. Usefully, the GNX4 footswitches can also be configured to provide handsfree control of some of Pro Tracks Plus's recording functions — great for the solo musician who often finds they do not have enough hands to operate both the guitar and their recording software. With 24-bit A-D/D-A conversion and quoted signal-to-noise ratios of greater than file:///H|/SOS%2002-05/Digitech%20GNX4.htm (5 of 6)9/22/2005 8:17:31 AM

Digitech GNX4

106dBA on all inputs and outputs, the combination of the GNX4 and Pro Tracks Plus is undoubtedly capable of producing very decent multitrack recordings. Perhaps the only downside of this aspect of the package is that, at present, no ASIO drivers are supplied — nor is there any clear indication on Digitech's web site about plans for this. ASIO drivers would certainly make the GNX4 more appealing to those already using a PC for recording based around an alternative sequencer.

All-round Performer Having experimented with Digitech's flagship amp modeller, I can say two things without hesitation: that the GNX4 is an excellent amp-modelling device capable of results on a par with the best of the rest; and that, given the sheer number of features packed in, it represents excellent value for money in the UK. However, there may be a question of how many users would need all the features on offer? If all that is required is a studio/live amp modeller, Line 6's PodXT Live or the Vox Tonelab SE are probably the most obvious competition, and would undercut the GNX4 in price, although Digitech's own GNX3 might be the more direct competition on that front. That said, for guitarists making their first foray into the hi-tech world of amp modelling and digital recording, the GNX4 is an extremely good option, with all bases covered. Despite being a relatively new unit, I've already seen the GNX4 advertised for considerably less than the recommended retail price listed here — if it does fit your particular requirements, then such pricing turns a bargain into an absolute steal. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



FAR OBS

In this article:

Reflex Design First-order Crossover Filters

FAR OBS Active Nearfield Monitors Published in SOS February 2005

FAR OBS £600 pros Involving sound. Good mid-range clarity.


Reviews : Monitors

cons Tonally inconsistent off-axis. Sibilant tweeter. Limited bass extension.

summary Worth a listen. While it's Idiosyncratic, and not the most analytical of monitors, you could fall for the communicative mid-range.

information £600 per pair including VAT. Digital Village +44 (0)20 8440 3440. +44 (0)20 8992 4550. Click here to email www.dvproaudio.com www.far-audio.com

These compact ported monitors from Belgium boast a communicative mid-range and a flattering overall sound. Phil Ward

The history of compact monitors goes back to the BBC-designed LS3/5A of the mid-'70s. The LS3/5A, itself derived from a speaker used in one-eighthscale architectural acoustic modelling, was intended specifically for monitoring in cramped outside-broadcast trucks. It was designed in-house by the BBC, partly because there was no suitable commercially available product — imagine, a world with no small monitor speakers! The LS3/5A was licensed for Photos: Mark Ewing commercial manufacture, originally by Rogers, and subsequently by a whole string of UK speaker manufacturers — Spendor and Harbeth to name but two. Stirling Broadcast currently hold a license and manufacture the speaker, one that's held in very high regard in hi-fi circles, and it's not unknown for fans to get quite weepy over the LS3/5A. But what, you might ask, has that paragraph of audio history got to do with the subject of this review, the FAR OBS? Well not only is outside broadcast one of the intended applications for the OBS, but it also uses drivers with dimensions almost identical to those of the LS3/5A, and a cabinet that differs only in proportions — it's a little squatter and deeper. Despite the obvious 'active' difference between the LS3/5A and the OBS, there's no ignoring its lineage. The OBS follows the familiar basic constructional pattern of similar active nearfield monitors. The rear panel incorporates an amplifier module that carries

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FAR OBS

the mains and signal inputs, the latter on combi jack/XLR (jack unbalanced, XLR balanced). The module here also carries low- and high-frequency equalisation knobs that each offer positions labelled Flat, +1dB, and +2dB. Strangely, the controls offer only gain, no cut, and still more strangely the owner's manual doesn't refer to them at all, so I've no idea at what frequencies they operate. And while I'm on the subject of the manual, it's a pretty humdrum document that's of little use to man or beast. Around the front of the OBS there's a green LED power indicator, an indented gain control (the indents being the only way of matching the level of a stereo pair), and the noisy bits — a 130mm coated-paper-cone bass/mid-range driver and a 25mm textile-dome tweeter. The drivers are mounted on a truncated pseudo-elliptical sub-panel that provides a little relief from the rectangular aesthetic, and may offer some diffusion of edge-diffraction effects. Positioned between the drivers, and either side of the panel, are twin reflex ports. The ports have a modest flare moulded into their exits, but are of a small size that means compression and distortion from turbulence are almost impossible to avoid. The front panel, like the remainder of the OBS, is finished in a dark-grey textured paint — 'functional' is the term that immediately springs to mind.

Reflex Design Regular readers will know of my dislike of reflex ports, but with a very small cabinet and bass driver, the need to squeeze a quart into a pint pot in terms of bass extension and power handling means that there is little option. Or is there? The LS3/5A is a closed-box speaker, and so is the similarly dimensioned ATC monitor, the SMC7. The LS3/5A manages without port loading simply by having no ambitions other than to offer medium-volume, accurately balanced audio between 100Hz and 20kHz. While it was originally designed as a genuine 'monitor', it simply doesn't have the power handling or bass extension to cope in our contemporary wide-bandwidth, high-volume world. It doesn't attempt to reproduce low bass, so doesn't get into difficulties by trying. The ATC, though, is a different kettle of fish. It both has a relatively extended low-frequency bandwidth and goes reasonably loud, without resorting to a ported cabinet. The secret is heavyweight and expensive engineering in the bass/mid-range driver that enables it to handle more power both thermally and mechanically. A port (or two) is unfortunately no substitute for a bigger magnet, thicker top plate, largerdiameter voice coil, and edge-wound voice-coil wire. Ports do extend bandwidth and increase power handling, but at the cost of distortion, compression, and degraded transient response. However, despite the fact that I'd much rather see a closed box and more money thrown at the bass driver, the bass quality of the OBS is, in practice, pretty reasonable for the speaker's size. The response alignment chosen seems relatively benign — in other words the ports are not asked to work too hard. At higher listening levels, especially with the LF EQ turned up, there's some port noise and compression, but it is delayed until the speaker as a whole is really


FAR OBS

working hard. There's no real extension to the OBS's low end, though, and it's not a monitor you'd choose if judgment of bass level and quality were of critical importance. The balance further up the band is neutral tending towards bright onaxis (though it's inconsistent off-axis) and I felt no need to use the HF EQ other than to check that it worked — I'd actually have preferred a decibel or so of high cut. The tweeter drew attention to itself a little through displaying an occasionally over-sibilant nature on vocals, but the mid-range quality of the OBS was noticeably good — the bass/mid-range driver had a clean, uncoloured natural sound.

First-order Crossover Filters The OBS specifications reveal that it employs first-order filters to integrate the drivers. This seems an odd decision to me, and one which has repercussions. First-order filters offer a clean transient response, a slow rate of change of phase with frequency, and, in theory, zero phase error at the crossover frequency. They also, in passive speakers, involve the minimum of distortion and colorationinducing components between amplifier and drivers. Unfortunately, however, the slope of first-order filters is so shallow (6dB/octave) that they provide precious little attenuation of any out-of-band driver resonances and result in a very wide band where the outputs of the two drivers overlap. One of the great advantages of active speakers is that higher-order crossover filters can be used, properly attenuating the drivers outside their optimum operating band, without the necessity to introduce a multitude of lossy capacitors, resistors, and inductors between the drivers and the amplifier. In using active first-order filters, FAR have consciously chosen not to take advantage of this. Proponents of first-order filters, presumably including FAR, would perhaps argue that the good transient performance and 'perfect' phase response are the most significant characteristics. The trouble with this point of view, however, is that the 'perfect' phase response of a first-order crossover network only comes fully to fruition if the drivers are coincident in space (dual-concentric is the closest we can get). With the non-coincident drivers of the vast majority of speakers, while perfect phase integration will occur at specific points in space, at other points the overlap output of the two drivers will 'rotate' in and out of phase, resulting in wideband off-axis frequency-response errors. Of course the same basic phenomenon occurs with any filter topology and non-coincident drivers, but the wide overlap inherent in shallowfilter slopes makes matters worse. Unfortunately, a pretty good demonstration of the resulting effect can be heard with the OBS — its perceived high/mid-range tonal balance varies significantly above and below axis. If ever there were a monitor not to use laid on its side, it's the OBS.


FAR OBS

So why, you might ask, have FAR decided to go with first-order filters? Well, primarily, I suspect it's because there is empirical evidence that low-order filters can endow speakers with a subjective sense of clarity and naturalness, and while I'm not convinced by aspects of the OBS monitor's engineering, they do have an engaging and transparently musical sound about them when heard on-axis. Perhaps this sound flatters to deceive — after all, 'flattering' is not the first characteristic you'd choose for a professional monitoring tool — but, on the other hand, the OBS is a highly communicative speaker. Without straying too far into hifi review language, if there's an emotional, human element to be found in a piece of music, I'd bet on the OBS to reveal it. And funnily enough, that's exactly the kind of thing traditionally said about the LS3/5A. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



FXpansion BFD

In this article:

Devil In The Details Groove Librarian The User Interface In Use Big Friendly Drums?

FXpansion BFD Virtual Drum Instrument (Mac/PC) Published in SOS February 2005 Print article : Close window

Reviews : Software

FXpansion BFD £199 pros Easy to use. Sounds fabulous! Doesn't dribble or fall off the stool.

cons No way to add extra toms to kits.

When it comes to drum sounds, BFD and its XFL expansion pack offer unprecedented levels of detail and flexibility, totalling over 30GB of sample data. Is it overkill, or the rhythm programmer's ultimate weapon?

summary This is a great-sounding, funto-use program that gives you all the realism of a well recorded acoustic kit without the hassle. XFL adds to the range while maintaining the quality.

information £199; XFL expander £149. Prices include VAT. Sonic8 +44 (0)8701 657456. +44 (0)8701 657458. Click here to email www.sonic8.com www.fxpansion.com

Paul White

FXpansion's BFD is a sample-based sound generator dedicated to producing authentic acoustic drum sounds, and can be operated as either a plug-in or in stand-alone mode on Mac OS X and Windows systems running hosts compatible with VSTi, DXi, RTAS, Audio Units or Rewire. Its core library is based on seven acoustic drum kits recorded at multiple velocity levels in the same sympathetic room using the same mic setup, but rather than just offering basic samples, the 14 mics around the kit are layered with user controls to allow, for example, the balance of close mics and overheads or room PZM mics to be adjusted. In practical terms, it's more or less the same as being in the control room with a good drum kit in the studio, where you can adjust the levels from the various mics to get the sound you want. There are also various tricks that let you try different mic placements and distances — for instance, the kick has a balance control for internal and external mics — while the core kits have up to 46 velocity layers and the optional XFL expander kits even more. A library of standard kits is included, but the way BFD is managed allows you to take individual drums from one kit and put them in another (toms always come as a set) and to tune individual drums. On top of

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FXpansion BFD

that, there are MIDI drum grooves that can be imported into your songs to drive BFD, and also a self-contained groove mode where different grooves are assigned to different keyboard keys, rather like the Groove Menus in Spectrasonics' Stylus. According to FXpansion, BFD should work under Windows 98SE and Me, but they only officially support its use under Windows 2000 and XP, and of course Mac OS X. You will need a fairly powerful computer with plenty of RAM and hard drive space to run BFD: Windows users should be looking at a PIII equivalent or faster, while Mac users need a G4 733MHz or above.

Devil In The Details Before getting deeper into the way the plug-in looks and works, it is worth examining the way in which the original drums were recorded. A variety of well known mics including the Sennheiser MD421, Neumann KM81 and M49, Electro-voice RE20, AKG C451 and Shure SM57 were used as the close mics. These were apparently amplified via custom-modified API preamps. As with a real studio-recorded kit, each The core library contains only seven kits, but these are sampled in heroic detail. drum was picked up by a number of mics at the same time, and AKG C12s were used as overheads via Summit MPC 100A tube preamps, while the room ambience was captured using Neumann U87s through Avalon preamps. In addition, a pair of Crown PZM microphones was set up at floor level and compressed via an Empirical Labs Distressor. The room ambience that you hear is all from the natural recording space, and to capitalise on this, there is control over the distance of the room mics from the kit, and the width of their stereo field. I'm not sure whether this was achieved using multiple mics or by simply delaying one pair of room mics, but it works well enough and the amount of ambience can be adjusted on a per-drum basis rather than globally. The mixer section where the kit sounds and mics are balanced also has dedicated controls to set the balance between mics positioned inside and outside the kick drum, and above and below the snare drum. The snare hits include straight hits, flams, drags, rims and sidesticks, while the hihats include closed and half-open using the stick tip and shank, open played with the stick tip, and pedal closure. Suitable muting modes are employed to give natural-sounding hi-hat playing. When you look at the number of miked layers and the number of velocity layers, it soon becomes apparent why a 9GB library is needed to do justice to a relatively small collection of drum kits. Where the host software supports it (and most of the popular sequencers do), file:///H|/SOS%2002-05/FXpansion%20BFD.htm (2 of 7)9/22/2005 8:17:54 AM

FXpansion BFD

each individual microphone buss and each individual dry (close-miked) drum sound can be routed to separate physical outputs for further processing. Modified kits and mixer settings may be saved in a user library and further kits can be added from the XFL expander library, increasing the basic 9GB library by a further 22GB, adding a greater range of cymbals, brushed kits, snares, tambourines and so on.

Groove Librarian BFD makes a wonderful drum sound module for anyone who needs to capture the real miked acoustic kit sound, but for those less gifted in the percussion department, it also comes with a large library of 'Grooves' in a number of styles. These include variations and fills, so it is relatively easy to create a naturalsounding performance, and because these grooves control only the single hits, they can be played back at any tempo. They are based around Standard MIDI files, so you can also use your own MIDI grooves, or commercial MIDI files. BFD is mapped in accordance with GM (though exceeds it in some areas where more variations are available) so the files should always trigger the right parts. In addition to the groove library, BFD also has tools to vary the feel of the rhythms with controls to add timing and velocity variations or to add swing. The Groove Librarian, which contains almost 1000 patterns, is accessed by clicking the pull-down bar at the top of BFD's window to reveal three 'Banks' of grooves with a browser on either side for locating the MIDI patterns for grooves and fills. You can then drag any groove 'bundles' from the browsers into Bank A and Bank B: a bundle is a set of 12 complementary grooves, each triggered by a different MIDI note. Fills go into the third bank. Normally, when triggered via MIDI notes, the grooves play through only once at the tempo set by the user, but there is a repeat button for auditioning or jamming along. The groove can be changed using a variable quantise control that is illustrated by a quirky-looking beatnik who crossfades into an equally quirky-looking robot as the timing becomes more rigid. Two graphical windows also allow you to add varying degrees of randomness to the timing and levels of each hit. Swing is also variable. As intimated, grooves are normally triggered directly from MIDI notes, both in free-standing and in plug-in mode, but clicking directly on a groove allows the grooves loaded into the banks to be cycled and switched randomly, including adding fills, if you like a little controlled chaos/synthetic improvisation. This sounds surprisingly good and is also a great way to audition sounds while you're finetuning the kit balance. A 'sync to song' function allows grooves to follow the host sequencer's song tempo and if Sync Groove Phrase is ticked, any grooves that are layered will play in perfect sync. There's also a means to change transitions between grooves so that instead of happening exactly at the start of a bar, they can happen part-way through to allow for upbeats and so forth. In monophonic mode, triggering a new groove kills the previous one, while polyphonic mode allows groove layering. The other way to trigger grooves is to copy the MIDI patterns from BFD's grooves and fills folders and place them in your sequencer tracks. Originally I couldn't get the sync to song function to work in Logic, but an updated set of BFD AU Components fixed that.


FXpansion BFD

The User Interface There's only one main window to this plug-in, and though sections of it change depending on what you're doing, operation is pretty intuitive. Normally you see a picture of a drum kit in the upper window, with the mixer section below and buttons for selecting kits and individual drums down the left. This is replaced by tables of MIDI note allocations when you're working on grooves. Further buttons on the right access the Groove and Feel functions as well as preferences and MIDI assignments. The button with the kit logo loads in a complete kit and shows you pictures of the available kits (and some technical spec) as you do it, while the individual buttons allow you to replace individual drums or cymbals, again with pictorial references. The mixer section has nine sections for the Kit Parts with mute and solo buttons as well as (Ambience) trim, tune and MIDI dynamics controls, plus additional trim and pan controls for the close-miked sounds. To the right of the mixer are the faders for the close mics and the three stereo pairs, each of which has mute and solo buttons, as well as a master fader. Further knobs provide separate width and distance controls for the three stereo pairs: overheads, room and PZM. There's no kit loaded by default, so you have to select a kit before you can hear anything. This takes a few seconds because of the size of the samples. Red lights underneath the solo buttons in the Kit Piece area of the mixer section turn orange, then yellow and green as the parts are loaded. The current kit layout can be seen by clicking on the main drum kit graphic to show an overhead view. Included in the Kit Piece selectors is a display showing which types of hits are provided for each drum, which is important when the snare might have straight hits, rimshots, sidesticks and so on. To save an edited kit, click the save kit combo icon, name the kit and then decide where on your drive to save it. The only serious limitation is that you can't add extra toms to kits, even by adding a detuned version of an existing tom. Sometime four toms is nice! When BFD is used as a plug-in, it shows up as three different versions: BFD Stereo, BFD Groups and BFD All. The only real difference between these is the output buss configuration: as you might imagine, BFD Stereo mixes everything to one stereo output pair. Group gives you separate outputs for the four main mic groups, while All gives you 14 separate outs (all the close mics plus three stereo pairs corresponding to overheads, room and PZM), which is the most flexible option for external processing. Playing the drum sounds from a MIDI keyboard is easy enough — you just treat BFD like a super GM drum kit — though you can also play it from a set of drum pads. If the sound or drum balance isn't right, there are plenty of ways to fix it. The Kit Piece and Direct mixer controls the levels of individual Kit Pieces: you can tweak the kick close/distant mic balance, change its pitch and overall level and reduce the amount the ambient mics contribute. Solo (red) and Mute (yellow) file:///H|/SOS%2002-05/FXpansion%20BFD.htm (4 of 7)9/22/2005 8:17:54 AM

FXpansion BFD

buttons let you work on individual sounds and pan controls allow the direct signal to be placed in the stereo mix. As with a real studio session, you can't pan drums within the overhead or room mics though you can alter the overall width of these. In addition to volume, a Dynamic control varies the velocity of incoming MIDI notes for the various Kit Pieces. An additional Master Dynamics control works for the whole kit, and it is possible to load and save mixer settings. Where BFD is being used as a straightforward MIDI-triggered drum sound module, a learn function enables other mappings than GM to be created and stored and some pre-made maps are provided for various electronic drum kits. If you don't like learn functions, you can make the assignments manually.

In Use Kits take around 20 seconds to load, and once loaded, most seem to need a little balancing to make them sound right. For my taste, the cymbals tend to come up a little too loud and the ambient mics make too much of a contribution, but it only takes moments to resolve this and then save the changes. But most important is the sound quality, and these kits sound exactly as though you have a real drum kit on the other side of the glass. The cymbal samples are very long and both the drums and cymbals are beautifully sampled, with more velocity layers than you could wish for. Being able to tune the drums within a kit is useful, as some of the tom sets aren't quite tuned ideally, and it's also wonderful to be able to change out cymbals, kick drums and snares. I loaded the XFL expansion and found it to be every bit as well sampled; it more than doubled the number of drum kits and added a wealth of new cymbals. All these kits sound great and you can adjust them from tight and dry to big and roomy — anything from John Bonham to jazz. There's no artificial processing so you can add your own if you feel the drums need it, but I found they sounded excellent as they were. In fact the only addition I'd suggest is a control for virtual gaffer tape to allow the damping to be changed — something like a software equivalent of SPL's wonderful Transient Designer would be perfect. I liked the feel functions for modifying grooves (see the 'Groove Librarian' box on the previous page) and in general, the grooves are pretty well designed, though some include gratuitous cymbal crashes that soon become wearing if you repeat them. The random backing mode is great for jamming and also for fine-tuning the drum sounds, but I prefer to create my own in real time where possible. If you work by dragging the MIDI files into the sequencer, you can easily edit these to sort out such annoyances as excessive cymbals so that's really not a problem. I also found some of the fills a little pedestrian, but then that's better than having everything over-embellished! I discovered that you can combine automatic Groove mode triggered from the higher keys with manual playing on the lower keys, so there are lots of possibilities. In fact the Groove section is far more flexible than I can go into here, so if you like that way of working, you shouldn't be disappointed, especially as you can add your own MIDI loops to BFD's file:///H|/SOS%2002-05/FXpansion%20BFD.htm (5 of 7)9/22/2005 8:17:54 AM

FXpansion BFD

repertoire. Just make sure you read the manual thoroughly as there's more to BFD than you might initially imagine.

Big Friendly Drums? BFD is definitely aimed at those users who want the sound of a real drum kit in a real room. If you want pre-treated, pre-produced, pre-digested sounds, this may not be the best place to look. The kits are very well recorded and the choice of three different stereo miking setups combined with the close mics gives you plenty of tonal scope. Furthermore, the range of quality drum kits is exceptional, especially if you add the XFL expander — and by the way, the letters BFD and XFL apparently stand for exactly what you'd expect them to! With a real kit, normally I start with an overhead mic and then add the close mics, but with these kits it's usually better to start with the close mics and then add the appropriate stereo mics to complete the picture, otherwise the sound can get too ambient. The PZM mics have a nice bright edge to them while the room mic adds a sense of space. Increasing the PZM mic distance also gives a nice doubling effect for those '80s pop sounds. Personally, I probably wouldn't use the automatic groove play function much, but the MIDI groove files are well worth having for copy-and-paste composing. In all, BFD is a lot of fun and mainly intuitive, though it does have a few hidden wrinkles that aren't explained as deeply as they might be, so you have to be prepared to explore. If the idea of getting real miked drum sounds without the real drums appeals, then this is one piece of software you have to try out — just make sure you have plenty of drive space available, especially if you go for the equally wonderful-sounding XFL expansion!

You can view the current kit from overhead, and swap out individual items.


FXpansion BFD

BFD's groove functionality is highly sophisticated. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Hercules 1612FW

In this article:

Looks Aren't Everything Hercules 1612FW Brief Specifications Installation & Mixer Utility Audio Quality Driver Tests Summing Up

Hercules 1612FW Firewire Audio Interface (Mac/PC) Published in SOS February 2005 Print article : Close window


Hercules 1612FW £439 pros Excellent value for money for an interface with 12-in/ eight-out analogue format. Balanced analogue I/O throughout. Low-latency ASIO and GSIF drivers.

cons Zero-latency monitoring is restricted to two inputs at a time. Frequency response only extends to 21kHz at a 96kHz sample rate. Can't be powered from the Firewire buss. Lacklustre WDM drivers.

summary Despite a few idiosyncracies, the Hercules 1612FW offers more analogue I/O than any other Firewire interface I can think of at the price, and is bound to win an enthusiastic following.

information £439 including VAT. Et Cetera +44 (0)870 873 8731. +44 (0)870 873 8732. Click here to email www.etcetera.co.uk www.hercules.com

Test Spec

The much-anticipated Hercules 1612FW Firewire interface supplies an impressive quotient of analogue I/O at a very attractive price. Martin Walker

Guillemot Corporation are already very well known for their range of ATI-based 3D Prophet graphics cards, Prophetview LCD monitors, Smart TV tuner cards, consumer soundcards including the Fortissimo, Game Photos: Mark Ewing Theater and Digifire ranges, and Hercules multimedia speakers. However, they have only occasionally dipped their toes in the waters of 'professional' audio interfaces — I reviewed their Maxi Sound 64 Home card way back in SOS February 1997 and John Walden looked at their Maxi Studio Isis in SOS June 1999. Both retailed at under £250 and were aimed at musicians on a budget, but this time round Guillemot have launched a rather more ambitious product. The Hercules 1612FW is, as its name suggests, a 16-in/12-out Firewire audio interface, yet is still offered at an enticing retail price of just £439. The bundle also includes Steinberg's Cubase LE and a special edition of Ableton Live, and sounds almost too good to be true.

Looks Aren't Everything The smart 1U-high case with its silver-sprayed front panel and black metalwork is significantly narrower than a normal 19-inch rack at about 13.5 inches wide, but is bundled with optional rack brackets, or you can attach a set of self-adhesive rubber feet if you want to use it as a desktop device.

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Hercules 1612FW

Windows XP driver version 5.3.4.5. Intel Pentium 4C 2.8GHz processor with Hyperthreading, Asus P4P800 Deluxe motherboard with Intel 865PE chip set and 800MHz front side buss, 1GB DDR400 RAM, running Windows XP with Service Pack 2. Tested with Cakewalk Sonar 4.0, NI Pro 53, Rightmark Audio Analyser 5.4, Steinberg Cubase SX 3.0 and Wavelab 5.00a, Tascam Gigastudio 160 3.03.

As with the majority of audio interfaces, the total of 16 inputs and 12 outputs implied in the name includes both analogue and digital I/O: there are 12 analogue inputs and eight analogue outputs, plus stereo co-axial and stereo optical S/PDIF, along with two MIDI Ins and Outs. Two of the analogue inputs have mic/instrument preamps with controls on the front panel, although unusually, the preamps are connected to channels 11/12 rather than 1/2. These comprise a Neutrik Combi connector for balanced/unbalanced mics on XLR plugs, or balanced/unbalanced instruments on TRS/TS-wired quarter-inch jacks. The mic preamps can provide up to 55dB of gain via their rotary controls, and you can bypass them completely to switch to line level by turning them to their clickable 'off' positions. The associated Instrument buttons change the input impedance from 10k(omega) to a high 100k(omega) impedance more suitable for most guitars, and there's also a global button for +48 Volt phantom power. The front panel is completed by a stereo headphone output and its associated volume control, hardwired to outputs 1/2 and sited on the extreme left, an on/off switch with associated blue LED power indicator, a pair of MIDI In and Out sockets and two rows of LED indicators — the inputs use tricolour green/amber/ red units to display signal present, -6dB and clip levels, while the output LEDs light up green when a signal is present. The back panel contains the other 10 analogue input sockets plus the eight analogue outputs, all on TRS-wired quarter-inch jacks for balanced or unbalanced operation and software-switchable between +4dBu and -10dBV levels, along with pairs of phono and Toslink optical sockets for the co-axial and optical S/PDIF, BNC connectors for word clock in and out, another pair of MIDI sockets, the six-pin Firewire port, and a socket for the supplied 15V AC wall-wart power supply. Unlike many other Firewire audio interfaces, the Hercules 1612FW can't be powered from the Firewire buss. The bundle is completed by an extremely generous 4.5-metre-long six-pin to sixpin Firewire cable, a six-pin to four-pin adapter for those using a PC laptop, plus a full version of the Cubase LE MIDI + Audio sequencer and special edition of Ableton Live 2.04 with a discount offer on the full version. Overall this is an impressive array of features in a smart-looking case. All the jack sockets are goldplated, and the only disappointing aspects of the review model were that two of the three rotary knobs chafed on the front panel slightly as I turned them, and the rear-panel jack sockets didn't feel very secure.


Hercules 1612FW

Hercules 1612FW Brief Specifications Mic/instrument inputs: two, balanced XLR with switchable global +48V phantom power, or balanced/unbalanced TRS quarter-inch instrument jack, 0 to 55 dB gain range, 10k (omega) or 100k(omega) input impedance, or line level at +4dbU or -10dBV. Analogue inputs: 10, balanced/unbalanced TRS quarter-inch jack, +4dBu or -10dBV sensitivity. Analogue outputs: eight, balanced/unbalanced TRS quarter-inch jack, +4dBu or -10dBV sensitivity. Digital I/O: S/PDIF in and out on phono co-axial, S/PDIF in and out on Toslink optical, word clock in and out, two MIDI Ins and Outs. Dynamic range: 105dB (A-D), 114dB (D-A). Supported sample rates: 32, 44.1, 48 and 96 kHz from internal clock. Frequency response: not stated.

Installation & Mixer Utility As always I paid a visit to the manufacturer's web site before installing the interface, but for once the bundled version 1.0 CD-ROM drivers were the most recent on offer. These support Windows XP with Service Pack 1A or Windows 2000 with SP4 on the PC, or Mac OS X 10.3. As usual, you install the drivers before plugging in the rackmount unit for the first time, but I've never known Windows detect so many devices on a single audio interface before — there are six in total, comprising ASIO, WDM, and GSIF analogue, plus optical, co-axial and MIDI digital. The Hercules 1612FW Mixer utility is functional rather than decorative, but of course it doesn't need a faux brushed-aluminium finish to do its job. It's divided into three main areas: at the top are monitoring and clock options, while the other two main areas are devoted to the Analogue Inputs and Analogue Outputs. Each of the 12 inputs has a slider with five gain settings in 3dB jumps over a 12dB range, the lowest being calibrated as +4dBu sensitivity, and the -10dBV setting being provided with an extra 9dB gain (strange, since there is actually 11.8dB difference between these two standards, so the full +12dB gain would seem more appropriate). All other settings are regarded as Custom, and this extra flexibility can be useful when optimising recording levels. Another unusual feature is that each of the input pairs can be software-switched between balanced and unbalanced operation. The eight analogue outputs have similar gain sliders to set output level, although this time the +4dBu position is at 0dB, -10dBV at -9dB, and the sliders continue right down to -125dB, so they act as digital volume controls. Input and output pairs can also be linked for stereo operation. The outputs can be used either as balanced or unbalanced, and for a unit of this price the option of balanced analogue I/O throughout is very welcome.


Hercules 1612FW

Even before I received my review unit, several SOS readers had asked me to investigate the Hercules' zero-latency monitoring, and I can see why. Unlike most Control Panel utilities, the Hercules' has no level meters, and indeed the unit itself has no onboard DSP mixer for monitoring purposes, in contrast to interfaces from the likes of Echo, Emu, M Audio and Terratec. Instead, you can choose a single input channel pair to be routed to hardware outputs 1/2 with 'zero' latency (the actual latency value is that of the converters, plus that set by the buffer size in the Audio Transfer page, which defaults to 1ms but can be dropped to 0.5ms).

Despite the lack of metering and a restriction to a single pair of inputs for 'zero'-latency monitoring, the Hercules 1612FW Control Panel utility still provides plenty of I/O options.

This two-input limitation probably won't bother those recording racks of synths, while those recording with more than two mics will need additional preamps anyway, and may therefore require a small mixer and be able to use that for monitoring purposes. However, if you intend to record more than two simultaneous inputs and need to monitor them simultaneously through your computer, you'll need to disable hardware monitoring and rely on the higherlatency path through your DAW's buffers. As is becoming common with other recent audio interfaces, WDM support comes in the shape of a single multi-channel driver rather than a set of stereo pairs, and an Advanced Properties page lets you choose how many channels are associated with it (ranging from two for stereo to eight for 7.1 surround). This is ideal for surround playback applications, but while some applications like Sonar will let you access all the individual channel pairs from a multi-channel driver, others such as Wavelab will default to using channels 1/2, which can make life trickier. However, the S/PDIF optical and co-axial inputs and outputs get their own stereo drivers, and the S/PDIF page in the mixer utility provides settings for the dreaded SCMS flags so you can attempt (if you so desire) to prevent your digital copies being further cloned.

Audio Quality Auditioning the converters and clock of the Hercules 1612FW against my Echo Mia and Emu 1820M proved to be a closer race than some I've attempted, but after matching their output levels to within 0.1dB, setting all three up in Cubase SX and then switching between them blind, I eventually picked out the Emu 1820M as the winner once again with its tight imaging and open, natural sound, particularly on vocals. The Echo Mia came in a close second with a slightly file:///H|/SOS%2002-05/Hercules%201612FW.htm (4 of 7)9/22/2005 8:18:00 AM

Hercules 1612FW

warmer but clear sound and good imaging, while the Hercules was just slightly behind them both, with a subtle but still noticeable harshness and slightly less focused image. However the differences between all three were fairly small, and most potential users will be delighted with the Hercules' audio quality, given its excellent price. My Rightmark Audio Analyser results confirmed the Hercules' published spec of a 105dBA dynamic range at 24-bit/44.1kHz (this is the overall loopback result; the D-A converters by themselves are even better at a published 114dB). The frequency response measured 0.5dB down at a good 8Hz and 21kHz with a 44.1kHz sample rate, but the top end didn't extend at all when running at 96kHz, still being 0.5dB down at 21kHz, which limits the advantage of using higher sample rates; as a consequence, the dynamic range remained almost the same at 96kHz. It doesn't support 192kHz sample rates at all, although I don't personally see this as much of a disadvantage. Stereo crosstalk was fairly good at about -103dB, while THD levels were low at 0.006 percent, although this is still a factor of 10 higher than some others such as M Audio's Firewire 1814 and Emu's 1820M.

Driver Tests It's hardly surprising that Guillemot's Isis sold very well, with eight-in/four-out capability at £250, but sadly it left many users disgruntled about motherboard incompatibilities, high latency, and the lack of any Windows 2000/XP drivers. This has understandably left some musicians wary of buying another Guillemot interface, so I paid particular attention to its driver performance. I had no problems running the GSIF 1.7 drivers with Gigastudio, while the Direct Sound drivers managed a 35ms Play Ahead setting in NI's Pro 53, compared with 30ms for the latest Echo Mia drivers and just 10ms for Emu's 1820M. I did try the lowest-common-denominator MME drivers, and the Hercules scored a poor 70ms against 45ms for both the Echo Mia and Emu 1820M, although few people have to use these nowadays. After running the Sonar 4 Wave Profiler for WDM Kernel Streaming I couldn't get the WDM drivers to run below 23ms without glitching, which is disappointing. However, these lacklustre Direct Sound, MME and WDM results are largely academic, since for the vast majority of musicians it's the ASIO The 1612FW features plenty of balanced performance that counts, and it's here analogue I/O, with extras such as word clock that the options also open up. From the being very welcome in a unit at this price. ASIO Control Panel you can sync the Hercules 1612FW from its internal clock, or from the word clock, S/PDIF optical or co-axial input signals. The buffer size defaults to a conservative 1024 samples (23ms at 44.1kHz), but in both Cubase SX 3 and Sonar 4 I was able to drop this right down to the lowest 64file:///H|/SOS%2002-05/Hercules%201612FW.htm (5 of 7)9/22/2005 8:18:00 AM

Hercules 1612FW

sample setting with no glitching, for a latency of just 1.5ms at 44.1kHz. This is an excellent result. The drivers are multi-client in that you can allocate the same channels simultaneously to ASIO and GSIF drivers (I had no problems running Cubase SX and Gigastudio on outputs 1/2 for instance), but you can't currently use ASIO drivers with two applications simultaneously. Hercules do promise this functionality later on, and also hint at firmware updates to improve other aspects of performance, although as always you should make buying decisions based on what's available now. Overall I thought the technical performance of the 1612FW very good for the price, although you won't really benefit from 96kHz sample rates, and I did notice a 0.5dB gain discrepancy between the channel 1/2 output levels on the review unit. This is unlikely to cause many problems in practice, but shouldn't really happen.

Summing Up If you want a Firewire-based audio interface, the most obvious competitors to the Hercules 1612FW are probably M Audio's Firewire 1814 and Edirol's FA101, both at very similar retail prices. Of the three, the Firewire 1814 offers the greatest potential number of audio channels because of its ADAT I/O, and it also provides slightly better dynamic range than the Hercules, supports 192kHz sample rates, and has a more extended frequency response at sample rates of 96kHz and above. However, there are only eight analogue inputs and four analogue outputs, and only the analogue outputs are balanced, where the Hercules scores highly for those without ADAT requirements in offering 12-in/ eight-out analogue I/O that's totally balanced, plus twice as many MIDI ports. The FA101 provides a closer spec, offering eight-in/eight-out balanced/ unbalanced analogue, plus S/PDIF optical and single MIDI I/O, and like the 1814 it does support 192kHz sample rates. However, there's no co-axial S/PDIF, and only offers a high-impedance option on one rather than two inputs. Overall, the biggest strength of the Hercules 1612FW is that it offers four more analogue inputs than either of its main competitors, plus twice as many MIDI ports. On the other hand, its audio quality lags slightly behind them both, its hardware monitoring is less flexible, it doesn't support 192kHz sample rates, and it can't be powered from the Firewire buss, which won't make it so popular for mobile recording. Nevertheless, at £439, the Hercules 1612FW is remarkable value for money for a 12-in/eight-out analogue audio interface with co-axial, optical, word clock, and two MIDI In/Outs, and it's hardly surprising that so many musicians are already interested in it. Published in SOS February 2005


Hercules 1612FW

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



Hot New Sample CDs

In this article:

Drum Fundamentals **** Blue Shock *** Pro Drum Works Volume One **** Cycles Volume Two: Unnatural Rhythm ****

Hot New Sample CDs Sample Shop Published in SOS February 2005 Print article : Close window

Reviews : Sound/Song Library

Star Bart Simpson Prank Calls ***** Amanda Hugginkiss

Drum Fundamentals ****

**** Hugh Jass

MULTI-FORMAT

*** Bea O'Problem ** Mike Rotch * Ivana Tinkle

This Sample Lab release comprises more than 2000 one-shot drum and percussion samples recorded at Rockfield Studios. Acoustic drums start things off, and they are much more varied sonically than in many drum libraries, especially the cymbals — these sound pleasantly 'wide-screen', fanning out harmonics across the stereo image, and I particularly liked the gong-like mallet hits and mysterious rolled swells, as well as the extravagant brush sweeps. By comparison, the hi-hat selection seems a little spartan, and I'd have liked a few more of the nuances between 'open' and 'closed'. The kick section, on the other hand, provides a versatile line-up of sounds, ranging from round and roomy to close and choked, which should cover most situations. The snare samples use several different drums (including DW and Ludwig models) and a bewildering array of playing and production styles: not only nice little performance techniques like brush drags, rolls, flams, and rimshots, but also different mic balances and even the odd gated-ambience sound. There's not much in the way of different velocity levels here, but that's not really the point of this library — it's not for creating realisticsounding kit performances, but more for supplementing live drum tracks and loops, or for programming in modern styles where realism isn't a primary concern. In such contexts, the snares here are mightily usable. The 808 and 909 sounds are everything you'd expect, with murderous low end and abrasive transients all present and correct, and some of the samples have also been overdriven for extra attitude! Then comes a bit of a 'lucky dip': the 900 files in the Breakbeat Drums section. Although these are named to indicate hihat, snare, and kick types on the WAV files CD-ROM, they aren't grouped that way on any of the discs. So you'll probably find your eyes glazing over some time

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Hot New Sample CDs

around auditioning file 183, despite the fact that there's lots of great stuff in there. It's a shame, but I suspect many users may simply skip to the following Kicks and Snares sections, the usefulness of which I found rather patchy by comparison — a playground for the more extreme effects and distortions. The Percussion section is good, though, combining a solid base of standard/ethnic percussion with lots of electronic and processed hits. The overall sound is very satisfying and meaty, as well as being clean and quiet where it matters, although I did feel that some of the sample edits were a little too abrupt. (I was also unable to use the EXS24-format instruments in Mac OS 9, because of the OS's truncation of the CD-ROM's file names.) Given the broad range of sounds, this library would make a good general-purpose workhorse, but would also be especially useful for electronica and dance styles. Mike Senior Audio CD and WAV, EXS24, Halion, and Reason Refill CD-ROM 3-CD set, £59.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.samplelab.com

Blue Shock *** WAV/EXS24 This latest offering from UK-based Tekniks is a collection of acoustic/electronic drums, bass, and percussion loops and samples, provided on a single CD containing both WAV files and programmes for Apple's EXS24 sampler. The sleeve notes state that all recording and subsequent processing was done to analogue tape rather than a digital format, and this is presumably to give an element of the warmth commonly associated with analogue recordings. The material is organised into five folders containing, in total, over 700 loops, nearly 300 single drum, percussion, and cymbal hits and, in the Settings folder, a series of EXS24 programmes for those who wish to access the material via a suitable software sampler. The loops are organised into Loops With Music, Mono Beats, and Stereo Beats folders. As might be expected, the latter two are mainly drum and percussion loops, while the former contains some 170 drum/bass loops. All the samples and loops seem to have been well recorded, and I'd be more than happy to use them in a commercial context — although I'm not certain I could really hear the influence of that analogue tape, even when auditioning some of them alongside other material that I knew had been recorded


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direct to digital. The titles of the individual loops give a clear steer as to the musical territory they would slot into; Acidrock, Acidhiphop, Chemical, and Pop appear in the names of a good number of the WAV files. At medium-to-higher tempos, these would work well for a Chemical Brothers or even Moby mood. At a more down-tempo 8090bpm there is plenty of material that would form a groove for hip-hop productions. I'm less convinced about the Pop reference — Will Young or Westlife it is not — although the loops could work with something with a bit more angst, perhaps Avril Lavigne or Christina Aguilera. While there are some straight loops, the majority have been processed in some way — reverbs, delays, and filters abound — and this makes for some interesting rhythmic textures. While the drum and percussion loops are very useable, I was left scratching my head a little over the contents of the Loops With Music folder. In one sense, these are instantly gratifying; just add your own top line or a chord structure and start to mix. If instant results are required to meet a deadline, then these loops could form an ideal starting point, and I've no qualms about their musical content — they would fit right in alongside current commercial releases. However, if you want to tweak things, the lack of separate loops for the drum and bass parts is somewhat frustrating. For maximum flexibility, perhaps these loops could each have been provided as a paired drum loop and bass loop. Being able to mix and match, re-balance or re-process either bass or drums independently would increase the creative options considerably. This said, for producers of styles such as hip-hop or Chemical Brothers-influenced dance, Blue Shock provides an inexpensive source of some very useable grooves. John Walden WAV and EXS24 CD-ROM, £39.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.tekniks.co.uk

Pro Drum Works Volume One **** ACID This latest Smart Loops offering contains a whopping 9000 Acidised WAV drum loops, as well as nearly 300 individual hits, many of which include multiple velocities. The 9000 loops actually contain 3000 unique grooves and fills, each played on three different drum kits; a dry Acoustic Kit (suitable for pop, rock, and funk), a Trap Kit (also fairly dry using smaller drums and with a distinctive hiphop-friendly sound), and a Thunder Kit (which sounds like the Acoustic Kit, but with plenty of ambience for a big rock sound). Given the amount of material, it is good to see a very clear folder organisation. A series of Basic+Funk folders goes from fairly simple eighth and 16th patterns file:///H|/SOS%2002-05/Hot%20New%20Sample%20CDs.htm (3 of 6)9/22/2005 8:18:06 AM

Hot New Sample CDs

through to loops containing more complex playing with variations on the snare, and further sub-folders contain loops with floor tom, open hi-hat, and ride cymbal. Other loop groups contain cut-time (snare only on beat three), double-time (snare on all the upbeats), double-kick (real head-banging stuff!), train (plenty of nice snare work here), and side-stick (with rim shot replacing straight snare) patterns. On top of this lot are folders containing jungle beats (tom-tom playing with a tribal feel), reggae loops, flams, and a large collection of four-beat and two-beat fills. The library is completed by some excellent one-shots for each element of the kit, including some very useable crashes, rides, and hi-hats, as well as some choked cymbals. In testing these loops within Sony Acid Pro v4, I noticed great consistency between the sounds of the three kits. While you might not wish to combine a double-time train loop with a reggae pattern, they would at least sound like they had been played by the same drummer, in the same space, and on the same kit. This makes it extremely easy to mix and match the loops, fills, and cymbal crashes to produce a really convincing, complete drum part. Within the basic and funk loops, there is also a tremendous amount of choice — so if some subtle variations are required then it is easy to find something to fit. The three different kits certainly add a distinctive flavour and are uniformly well recorded. For my own taste, the Thunder Kit was just a touch too wet, although it would certainly work in some hard-rock styles. For more routine rock, funk, and pop duties, I found the Acoustic Kit more appropriate, and adding just a touch of ambience to this worked really well. Indeed, given the unique sound the different kits bring to the loops, I wonder whether Smart Loops might not also make them available individually to allow users to purchase just the kit that suits their own particular musical needs. Pro Drum Works: Volume One may not offer anything radically new, but as a 'bread and butter' drum-loop collection, it has plenty to offer and represents good value for money. John Walden Acidised WAV CD-ROM, $249 (around £132). Click here to email www.smartloops.com

Cycles Volume Two: Unnatural Rhythm **** AUDIO/WAV/REX 2 Cycling '74 are probably best known for the Max/MSP audio development environment and the Pluggo collection of AU/VST/RTAS-compatible plug-ins, both of which are distributed by DACS in the UK. However, Cycling '74 also have a growing collection of sample CDs under the Cycles series title. Volume two of


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this series, Unnatural Rhythm is under the spotlight here, and comprises an audio CD and a DVD-ROM, the latter carrying 24-bit WAV and REX 2 formats. For the WAV files, both 44.1kHz and 48kHz versions are included — I experimented with the 24-bit/44.1kHz WAV versions, almost 1GB of data spread over 300 loops, using Sony Acid Pro. As might be guessed from the title, the content of Unnatural Rhythm is a little leftfield. There are no straight drum loops, synth melodies, or bass lines here. Instead, the focus is on rhythmic, pulsating, and, it has to be said, rather unsettling soundscapes. The material is divided into six folders and their titles confirm that we are as much in sounddesign territory here as music composition. For example, Back Alley Tribal contains a series of industrial, metallic-sounding percussion loops, while Contraptions contains various real and synthesised machines humming, buzzing, screeching, and pounding away. The Grooves folder is more obviously rhythmic in nature — but even here plenty of processing has been applied, while Cyclic Waves concentrates upon slowly evolving sound beds. Found Sound features some very heavily processed natural sounds (glass, marbles, and taps, for example), while Vintage Electronics contains all sorts of bleeps, blinks, and clicks, all with a rhythmic element. While most of these loops are between two and eight bars in length, a final Xtended Versions folder contains (surprise, surprise!) longer versions of a small number of the loops, some of these running to over a minute. This collection would work really well in a music-to-picture context — think gritty urban drama, dark sci-fi, or straight horror and Unnatural Rhythm would fit right in. It could also work within trip-hop or abstract electronica. In these contexts, many of the loops would need little in the way of additional material adding (just enough to comply with the terms of the licence!) but, as the emphasis is on rhythmic sounds rather than pitched or melodic phrases, I found it very easy to mix and match loops from other libraries to build a complete piece. Indeed, putting together a couple of Unnatural Rhythm loops underneath a more straightforward melodic phrase (a string section for example) created a contrast and a musical tension that worked well — in a disturbing sort of fashion! Clearly, given the above description, this library is not necessarily going to appeal to everyone, but, for media composers who like the 'dark side' or those with a taste for abstract soundscapes, this collection is well worth exploring — just don't leave anyone alone in a dark room with the music you create! John Walden Audio CD and WAV and REX 2 DVD-ROM 2-disc set, £63.99 including VAT. T DACS Audio +44 (0)191 438 2500. +44 (0)191 438 2511. Click here to email www.dacs-audio.com www.cycling74.com


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Joemeek Three Q

In this article:

Overview In Use

Joemeek Three Q Recording Channel Published in SOS February 2005

Joemeek Three Q £135 pros Transparent, quiet preamp. Vintage-sounding EQ. Classic optical compressor sound.

cons Limited mid-band EQ control range. The character of an optical compressor may not suit every application.


Reviews : Recording Channel

The Joemeek product range has recently been completely overhauled, and one of the updated units is this new half-rack recording channel. Have the designers kept what made the original products successful?

summary This re-launched Joemeek processor has been updated and improved in many areas without compromising the original opto-compressor sound.

information £134.99 including VAT. PMI Audio UK +44 (0) 1803 215111. +44 (0)1803 215111. Click here to email www.pmiaudio.com

Paul White

The Joemeek Three Q is the first product to bear the Joemeek name since the company was acquired by the American PMI Group, and the circuitry has undergone a complete redesign. Although the new owners are American, this redevelopment has Photos: Mark Ewing been overseen in the UK by Malcolm Toft, who was instrumental in the design of the original Trident consoles.

Overview Built into a half-rack 1U metal case and powered from an included adaptor, the Joemeek Three Q combines a mic/line preamp with an optical compressor and a simple but musical equaliser. Everything about the Three Q is different from the old Joemeek range, other than the fact that the front panel is green, and even that is now a stylish extruded metal component with a curved front, embellished with small, tasteful knobs and pale-yellow legending. A new preamplifier based around the same low-noise chip used in many high-end preamps now feeds the optical compressor section, and this too has been revised to provide a greater dynamic range and to avoid saturation in the side-chain. Directly following the compressor is a three-band equaliser with a swept mid-frequency control, and the EQ circuitry is now based on the classic Wien Bridge topography (a variation

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Joemeek Three Q

on which was used in the old Trident consoles) to provide more of a vintage British sound. On the rear panel is a switch to activate the 48V phantom-power source (though there's no front-panel status LED) and, as expected, the mic input is on a balanced XLR connector. A quarter-inch balanced jack takes care of the line input, and a front-panel button with status LEDs selects which is active. An insert point (unbalanced on a TRS jack) comes between the mic preamp and the compressor, and there's also a balanced mix input that allows an external linelevel signal to be added into the Three Q's output stage prior to the Output Gain control. That leaves dual outputs on balanced quarter-inch jacks with a +4dBu/10dBV level selector switch. The reason there are two outputs is that one is available to be used as a direct monitoring source in computer recording systems that exhibit excessive latency. Here the circuitry is designed to give the same level into balanced and unbalanced loads provided that unbalanced loads are connected using a single-pole (unbalanced) jack. Controlled by a single knob with a gain range of 10-60dB, the mic preamp has no pad or low-cut filter, though it can stand up to +19.5dBu without clipping, so a pad is very unlikely to be needed. On a number of preamps I've reviewed recently, a lot of gain has been crammed into the last few degrees of the control, but here a specially designed pot has been used to make the control of gain much more even. As line signals are generally much greater in level than mic signals, the gain range is from +35dB to -15dB when the line input is selected. In terms of noise, the official spec for the EIN is -128dB, which compares well with other high-quality mic preamps. A warning LED comes on 6dB below clipping. As with most good mic preamps, the frequency response extends well beyond the range of human hearing (10Hz-70kHz, -3dB), but not so far as to run into unnecessary radio-frequency interference problems. Signals below the compressor threshold have a distortion figure better than 0.001 percent, but above that some intentional compressor distortion is part of the character of the unit. There are just three compressor controls: Compress, Attack, and Release. The ratio is fixed at a nominal 5:1 and, in common with other optical compressors, the threshold tends to be progressive, giving it a soft-knee characteristic. In essence, the ratio automatically rises from very low to its full 5:1 value as the signal level approaches the threshold. When higher levels are presented, the ratio eventually falls again, which is another characteristic of optical compressors that contributes to their 'dynamic' sound. Compress essentially sets the level of signal relative to the threshold, where clockwise equals more compression. No calibrations are provided for the Attack and Release controls as, in this design, the actual attack and release values vary depending on the nature of the incoming programme


Joemeek Three Q

material. A simple four-LED meter monitors the amount of gain reduction being applied, up to a maximum of 16dB, and a button bypasses the compressor when not needed. Both the compressor and EQ bypass buttons have status LEDs. That leaves the EQ section, which comprises three 'bell curve' equalisers rather than using peaking filters for only the mid-range. The mid-band covers 300Hz5kHz, while the low and high sections are centred at 80Hz and 12kHz. All have a ±15dB gain range. I prefer my mid-band to go lower by around an octave where possible, but this range seems to be a good compromise for a three-band EQ. Like the compressor, the EQ can be bypassed, after which the overall output level can be controlled from 'off' to +14dB of gain, the actual level being monitored by a nine-section LED meter reading up to +12dB. With a low 75 (omega) output impedance, the unit can drive almost any load, even over long cables.

In Use My first impression when trying the Three Q was that the mic preamp was very clean and uncoloured for a product in its price range, but what I really wanted to find out was whether the compressor redesign had lost the character of the original. I needn't have worried, though, because although the compressor behaviour now seems a little better controlled it is still possible to go from mild gain-control compression to quite aggressive compression that's meant to be heard. The heavier compression works well on things like rock vocals and bass guitar, but it can be made to work on most sources, including acoustic guitar and drums. One of the interesting things about 'lamp and photocell' compressors is that their gain-reduction characteristics are quite nonlinear, and once the signal has risen to a certain level the compression ratio falls back, increasing again once the level gets even higher. This kink in the compression curve helps the compressor impose a character without killing the transients in the signal. I found the equaliser more musically useful than Joemeek's previous fixedfrequency four-band design, and I prefer the sound of 'bell curve' high and low equalisers in most situations, as they let you boost more of the frequencies that need boosting, while affecting everything else to a lesser degree. The only frustration for me was the lower limit of the mid-range equaliser, as I find the ability to apply EQ cut down as low as 150Hz to be quite useful. Overall, though, I liked the sound of the EQ. The compromises inherent in this design are of course the availability of a single swept mid-range EQ band, the fixed compression ratio, and less comprehensive file:///H|/SOS%2002-05/Joemeek%20Three%20Q.htm (3 of 4)9/22/2005 8:18:12 AM

Joemeek Three Q

metering than you might expect from a top-shelf processor. Nevertheless, given the low UK price of the Three Q, it performs extremely well, combining the essential character of the original Joemeek compressor with a more transparent mic preamp and a more useful equaliser. It also looks a lot smarter. Optical compressors are designed to be heard unless used very sparingly, so they may not be the best choice if you need heavy compression with minimal side effects. Clearly the Three Q is a 'character' compressor, where the sound being processed is audibly thickened, and you can coax it, if you choose, into gain pumping to give an aggressive rock or hip-hop sound. In other words it is as much a creative tool as a means of regulating level. There's also a lot more to this unit than a colourful compressor, as it includes a very respectable mic preamp and a nice sounding EQ that has more than a hint of vintage feel to it. If the Three Q points the way for how the rest of the range will be presented, the Joemeek name can look forward to an interesting future. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



M Audio O2

In this article:

Thinner Conclusions Comparisons

M Audio O2 £119

M Audio O2 USB MIDI Controller Keyboard Published in SOS February 2005 Print article : Close window

Reviews : MIDI Controller

pros Excellent value for money. Ultra-thin profile mimics the latest laptops it will be used with. Full-width keys for chord and interval playing.

cons Half-distance travel a little strange at first. Mastering the unusual pitch and mod controls needs practice.

summary M Audio have done it again in terms of price and spec. If you need a compact controller keyboard and your budget is very limited, but you would rather sacrifice key-travel distance than make keys narrower and therefore make chord and interval playing unpredictable, the O2 is the one for you.

information £119 including VAT. M Audio +44 (0)1923 204010. Click here to email www.maudio.co.uk

Thin is clearly In amongst manufacturers of keyboard controllers. Mere months after the arrival of Edirol's slender PCR1, M Audio weigh in with the equally waiflike O2. But is it slim and trim, or just a lightweight? Paul Wiffen

It's hardly original to observe that everything's getting smaller in today's technological world, but what's interesting about the most striking recent changes in the size of musicmaking equipment is that they've involved a lessening of physical thickness. I recently found my old Wall Street Apple laptop and was amazed to find that it weighs a ton and that Photos: Mike Cameron most of the apparent bulk comes from Like a couple of other recent keyboards, the thickness of the computer. If you such as the Alesis Micron reviewed last month, the O2 makes use of the musical think about it, the third dimension of keys to provide extra switches to access any type of gear is the one that brings certain functions, which are listed in the no advantage to users — whereas the 'scooped-out' section above the musical keys. width and depth of a laptop, for example, give you useful viewing area and room for extended keypads. Edirol were clearly thinking along the same lines when they released their ultra-slim PCR1 controller (see SOS September 2004, or www.soundonsound.com/sos/sep04/articles/edirolpcr1.htm), and M Audio have now made the same kind of design leap to produce the O2 (for a little on how the two compare, see the box opposite).

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M Audio O2

The initial response to the O2 tends to be that something so insubstantial can't be a serious professional product. I have noticed over the years that despite all the moves to smaller and smaller technology, we still tend to instinctively equate weight with value. However, when you consider the most recent Mac Powerbooks or Sony Vaios — hardly unprofessional pieces of hardware — this is clearly not a fair response. What's more, the O2's slim, light form makes it ideal for mobile work — in fact, it fitted into one of my laptop bags with plenty of room to spare on its own, and, handily, with just enough room to squeeze in my 12inch Powerbook as well. Of course this is all fairly academic if the resulting product is so compromised that it doesn't do the job required of it. So; how does the reduced height impact on the usability of the keyboard? Well, the keys remain full size, so that the width of one octave on the keyboard is the same as on a 'normal' keyboard, and this is critical for players who have traditional keyboard skills, and are used to the lateral distance required to jump standard intervals like fifths and octaves. The main restriction that has had to be placed on the keyboard is that the distance the keys pass through when you play them is about half what you would normally expect. Like Simon Price when he reviewed the PCR1, when I first tried playing the O2's restricted-travel keyboard, even without it connected to a sound source, it felt very strange. However, there is a musical precedent for this type of action. While I was writing this review, I went to visit a writer/composer whose musical film I am directing, and he has an antique harpsichord. Sitting down to play harpsichord for the first time in years, I was immediately struck by how similar the action was to that of the O2! Despite this, I did find that it was initially more difficult to play the O2 and achieve the same subtle control of velocity which you can on a keyboard with full key travel (harpsichords, of course, aren't 'velocity sensitive' either). But as with so many things, once I started working on a song, I soon forgot about the actual keyboard I was using and found myself accomplishing everything I needed from a point of view of velocity sensitivity. As with so many things, the human body is fairly adaptable and this is certainly much less of a compromise than minikeys which, for me at least, completely mess up my sense for note intervals.

The rear panel is stripped back to the basics, with just the all-important USB connection, a lone five-pin MIDI Out and a Sustain Pedal jack.

I suspect that the thickness of the O2 also dictates the use of rubber rocker switches as performance controllers in place of the more usual wheels for pitchbend and modulation — clearly, wheels require a fairly deep casing. But the choice M Audio have made is an excellent compromise, both in terms of its low profile and usability, particularly on a unit which is designed to be regularly

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M Audio O2

chucked in and out of laptop bags — a Korg-type joystick or Roland three-way 'bender' would be too vulnerable in this respect. I had never tried this type of rocker switch before, but it definitely represents a new approach to controlling pitch bend, and I actually found it to be quite inspiring. Of course the range of pitch-bend over which the rocker operates is set on the target devices, but the Octave rocker allows you to shift up five octaves and down four from the nominal central position, so the O2 will cover an 11-octave range. The Octave rocker switch also doubles as a Transpose control — pressing both sides simultaneously puts it into Transpose mode, and then you use the '+' and '-' sides of the rocker to shift pitch in semitones instead of octaves. The remainder of the controllers on the O2 are much more conventional, with the standard fully assignable knobs and switches (eight of each) which are to be found across the rest of the M Audio range of controllers (and, prior to their buyout by M Audio, Evolution's controllers, from which many of M Audio's current range are derived). Five more such buttons below the three-digit LED display allow you to access Advanced Functions (in tandem with the lower half of the musical keyboard), Global Channel, Mute, Decrement and Increment controls respectively. The single slider to the left of the display is not hardwired to volume (still my favourite feature of the Evolution 461C) but can be set to that function in any of the five non-volatile memory locations. Once again, M Audio/Evolution's generic 'once-size-fits-all' editor/librarian software Enigma can be used to archive these memories, name them and view their settings. The version on M Audio's web site has already been updated to support the O2, and worked fine for me under Mac OS X (see the screengrab on the next page). The back panel of the O2 features a reasonable number of connectors, namely for USB, the 5V power supply (in case you are not powering from USB but using the O2 as a stand-alone MIDI controller), a single MIDI Output, a sustain-pedal input and a power switch. There is also a Kensington keylock port to prevent the highly desirable O2 going for walks (sensible — you could almost fit it in the inside Editing profiles using M Audio's splendid pocket of a greatcoat!). The MIDI (and free!) software editor, Enigma. Output allows the O2 to act as a 16channel MIDI interface output from the computer as well as a stand-alone MIDI controller. Unfortunately, as there is no MIDI Input, it does not allow you to input MIDI data to your host computer unless it can be generated on the keys, rocker switches, knobs or buttons of the O2 itself (I am always looking for a way to plug my MIDI guitar in without needing to run a second USB MIDI Interface). I've covered the way that M Audio and Evolution products deal with the assignment and range of controllers to knobs, buttons and sliders in my recent reviews of their products (see October and June 2004's reviews of M Audio's Keystation Pro 88 and Evolution's 461 at www.soundonsound.com/sos/oct04/


M Audio O2

articles/keystation.htm and www.soundonsound.com/sos/jun04/articles/ evolutionmk461c.htm respectively). The only difference here is that there is no room for a numerical keypad on the O2, so the upper half of the keyboard is used to enter numerical values, in the same way that the lower half is used to decide which of the Advanced Functions you want to access. The names of these functions and the numbers are printed above the relevant keys on a 'scoopedout' concave edge below the main panel (see the main picture at the head of this review). This is nice, as you can nearly always read the legending, no matter what angle you're at to the top panel, which is not always the case on keyboards with silver casings, especially under bright lighting. This use of the musical keyboard to access functions makes it very easy to select and enter new values for all the Advanced Functions, by simply holding the Advanced Button with one hand, selecting the function with the other, and then hitting the keys that represent the values immediately afterwards. Of course, if you have a computer connected, then it is always going to be quicker and easier to create and edit presets with the Enigma software, but for quick changes where you don't want to bother opening up Enigma up, the system works very well. The five memories which are preset as standard in the O2 cover a General MIDI Preset, two Reason presets, one for Yamaha XG or Roland JV synths, and one which M Audio describe as 'Undefined CCs for MIDI learn'. This last one is particularly useful, as applications like Logic and Ableton Live do not have default settings for controllers. Recommended practice for such applications is to use controller numbers which do not have a specific function associated with them, so this preset contains controller numbers which are less commonly used. You then put your application into Learn Mode, move the control on the O2, and then the application will use that MIDI controller without you having to spend time deciding which number you should be using. The Quick Start Guide which comes with the unit is extremely basic and really only covers those functions which any reasonably intelligent MIDI- and computerliterate person can figure out for themselves. To learn anything which you can't instantly figure out, you need to open up the fuller International User Guide, which is supplied with the O2 as a PDF (but only print out the 20 pages in your own language, otherwise you will end up wasting a hundred sheets of paper!). This does go into quite good detail on topics like 'Assigning RPN/NRPNs to a Fader/Rotary Controller' and 'Assigning MMC Controls to a button', but I do wish the info was supplied on paper, rather than in a PDF.

Conclusions The O2 is a great example of how good things can come in small packages — just like modern laptops. With a little bit of familiarity, the reduced travel of the keys becomes second nature, and personally, I found the rocker switches for pitch, mod and octave a positive joy to use. Finally, there's the price to consider — and at just £119 including VAT, you'd have to say that it's great value for file:///H|/SOS%2002-05/M%20Audio%20O2.htm (4 of 5)9/22/2005 8:18:20 AM

M Audio O2

money.

Comparisons The obvious comparison for the O2 is the Edirol PCR1, which was the first of the ultra-thin controllers launched earlier this year. Unsurprisingly, the travel distance on the PCR1's keys is of a similar order to that of the O2, and it also does away with conventional wheels, opting instead for a rubber pad for pitch, and a rollertype arrangement for the mod wheel. The PCR1 has a suggested retail price of £229, which is considerably more than the O2 but firstly, there are some great discounts to be had on this price in the UK at the moment, and secondly, the PCR1 also acts as 24/96kHz-capable audio interface. You have analogue ins and outs on RCA phone jacks and a mini-jack headphone out. On balance, I would say that if you need the audio interface, then the PCR1 is the obvious choice, as you are killing two birds with one stone (and saving one USB socket for your sequencer dongle or other device), whereas if you just need the keyboard triggering, the O2 represents killer value for money. Apart from the PCR1, it's not wholly fair to compare the O2 with other controllers, as they're too dissimilar. There are full-travel mini-keyboards for around the same price, but they don't have the same dimensions and sleekness of the O2. If you want full-size and full-travel keys, then of course you will have to pay more, and also be less concerned about the space taken up by the keyboard. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Mackie Big Knob

In this article:

Whipping It Out Of The Box Under The Hood The Full Frontal At The Helm Polishing It Off

Mackie Big Knob Monitor Controller Published in SOS February 2005 Print article : Close window

Reviews : Accessory

Mackie Big Knob £325 pros Solid build quality with large controls and good LED indication. Excellent audio quality. Sensible routing options. All balanced connectors with gain switches and trims where required. Includes talkback.

Mackie's new desktop controller offers source selection, monitor switching, and talkback facilities, all in a single robust box. Will it turn out to be a Dirk Diggler or a Dark Helmet? Paul White

cons No surround mode.

summary The Big Knob stands proud as an ideal control centre for any mixerless DAW system, with a good choice of source selection, including phono record-deck inputs.

information £325 including VAT. Mackie UK +44 (0)1268 571212. +44 (0)1268 570809. Click here to email

Apparently the phrase 'big knob' doesn't have the same connotations in the US as it does in the UK, but Carry On-style jokes aside, Mackie have come up with the most elegant DAW monitoring controller I've come across to date, at least at the kind of price a UK home-studio owner can afford. It doesn't support surround monitoring, but offers pretty much everything you could wish for in a stereo, computerbased home studio.

Photos: Mike Cameron

www.mackie.com

Whipping It Out Of The Box Essentially, the Mackie Big Knob gives you all the features you'd expect in the master section of a medium to large mixing console, allowing you to select monitor sources, choose between different monitors, and provide a talkback facility to the studio. It also has a couple of two-track outputs and a DAW output, to all of which the selected input source is simultaneously routed. While it's not the only product to offer such facilities, the Big Knob is a little more flexible than most insomuch as multiple sources can be mixed and rear-panel gain trims allow different monitoring systems to be balanced in level.

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Mackie Big Knob

Designed for desktop use and powered from the mains rather than from a power adaptor, the Big Knob can select between four stereo input sources, one of which is the stereo monitor mix from your DAW's soundcard or audio interface. These can be routed to any one of three sets of monitors plus four sets of stereo outputs, and there's also a built-in headphone amp with two headphone outputs for headphone monitoring in the control room, as well as a separate stereo studio output that can be sourced from the selected input source or from an external stereo source such as a DAW's aux send. Most of the inputs are switchable from +4dBu to -10dBV, or have gain trims, and all the signal connections other than the Phono inputs are on balanced jacks. The Phono inputs are designed to be used with a record deck fitted with a magnetic cartridge and have the necessary RIAA equalisation. Selected sources are routed to the control-room buss, which in turn also feeds the DAW output and two two-track outputs on balanced jacks, allowing the output from a record deck to be recorded back to the DAW or to a two-track recorder. All four sources and all three monitors have rear-panel gain-trim pots, and all the outputs (2-track A, 2track B, DAW Output, and Phones Amp) are switchable from +4dBu to -10dBV. The whole unit is built in traditional Mackie tank-like style, and slopes forward for ease of use when positioned on a desktop. The cosmetic styling matches Mackie's original mixers and Mackie Control, and the unit is solid enough and heavy enough not to get dragged off the table by the weight of any connected cables.

Under The Hood Mackie have been very careful to make the signal path clean and quiet so that your monitoring won't be compromised. The spec shows the frequency response to be essentially flat from 10Hz to 50kHz, and only 3db down at 100kHz, while EIN is 119dBu maximum and the dynamic range 112dB minimum on all paths except the phono ins (which have a 93dB minimum dynamic range). The input trims have a ±10dB range, while the output trims go from -14dB to 0dB. Certainly the unit seemed as quiet and as clean as a Mackie mixer.

The Full Frontal Input-level meters and the four input source-selector buttons are at the top left of the panel, with the phones and studio output section beneath. Multiple sources can be selected at the same time, in which case they are mixed to the controlroom buss. This is most useful if you want to play back a CD or sequencer track and jam along with your guitar preamp, for example. Both the phones and the studio output are fed from the same source, which can be switched to follow the main control-room source or the stereo DAW Phones Mix Input on the rear panel — this could be fed from a DAW stereo aux send or other source to provide the performers with a different mix to the one the engineer is hearing. Both phones


Mackie Big Knob

outputs and the studio outputs have their own level controls. The centre of the Big Knob is, predictably enough, dominated by a big knob, the purpose of which is to allow you to keep a firm grip on your monitoring levels. This is accompanied by Mono, Mute, and 20dB Dim buttons, though there's no option to solo the left and right monitors independently or hear them out of phase in mono, which I know would not please our Mr Robjohns — these were facilities the BBC insisted upon. A small perforated section of panel hides the talkback mic, and there's also a small power LED as well as red status LEDs on all the buttons. No blue-LED posing here! The internal talkback mic can be routed to the DAW output or used to feed the studio headphone amp for communicating with the musicians in the live room. Two Talkback buttons are provided, one for 2-track and one for Studio Phones, and both of these are nonlatching to save embarrassment when you slag off the drummer thinking the talkback is off! Monitor selection is to the right of the knobicus maximus, with the talkback buttons and talkback level controls directly below. On the rear panel there's also a footswitch jack for hands-free talkback switching.

At The Helm At 3.5Ibs (1.6kg), Big Knob sits firmly upon the desk and is well set out from an ergonomic point of view. Although simple to use and pretty foolproof, the four stereo outputs are always active, which means inappropriate use of your Big Knob might get you into trouble. Specifically, you can end up in a situation where you create a feedback loop — for example, where a stereo tape machine is connected to both the inputs and the outputs and the tape machine's input monitoring is turned on. Avoiding such problems requires only a little care on behalf of the user, and the Big Knob's approach here is certainly less rigid than that of competing devices that preclude certain routing options. Not only does the Big Knob fulfil all the basic stereo requirements of a DAW studio, it also allows CD players and guitar preamps to be played through the monitors without having to switch on your DAW. In my case it also makes it easier for me to review and compare different sets of active monitors, as I can connect these to the spare monitor outputs, trim the levels to sound the same, then switch between them.


Mackie Big Knob

Polishing It Off While the Big Knob might face some stiff competition on cost, it is the bestdesigned, best-packaged controller I've tried at anything like this price. In picking the right features, I think it's safe to say that Mackie have really pulled it off this time. You can route signals to the DAW from external devices, route mixes to other hardware recorders, and even transcribe vinyl. Passive solutions may retain the moral high ground insomuch as they put no circuitry other than relays in the signal path, but the ability to mix multiple inputs as you can here is occasionally very useful. It's no good — I'm going to have to buy one! Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Mackie Onyx 800R

In this article:

Ins & Outs A Peek Inside Controls & Operation Built-in M&S Decoder Specifications & Digital Functions Using The Onyx 800R

Mackie Onyx 800R Preamp & A-D Converter Published in SOS February 2005 Print article : Close window

Reviews : Preamp

Mackie Onyx 800R £1099 pros Onyx preamps provide superb sound quality. Analogue and digital outputs. Flexible digital output rates and formats. M&S decoder and DI facilities.

Mackie 'up the ante' once again by squeezing eight channels of their class-leading preamplification and AD conversion into a 1U rack.

cons No word-clock output. M&S selector on rear panel with no front-panel indicator. Questionable A-D alignment.

summary This is a remarkably flexible unit in a very compact box. The preamps provide superb sound quality in a class far higher than the price would imply, and the converters translate that quality to the digital domain.

information £1099 including VAT. Mackie UK +44 (0)1268 571212. +44 (0)1268 570809. Click here to email www.mackie.com

Hugh Robjohns

The market for straightforward, multi-channel mic preamps has grown considerably in recent years as more and more people have grasped the idea that the best way to use digital recorders and DAWs is to record all sources without processing, and then to process them as necessary during the mixing stage. Thus high-quality front ends are big business these days, covering every possible variation from mono recording channels to multi-channel preamps like the Mackie Onyx 800R reviewed here. Mackie have packaged eight of their latest Onyx preamps into a surprisingly heavy 1U rackmount box, along with a high-quality eight-channel A-D 24bit/192kHz converter and several useful 'bells and whistles'. The preamp circuitry is taken directly from the Onyx-series mixers, which incorporate a new mic preamp design claimed to 'meet or surpass the fidelity and dynamic range of ultra-expensive' stand-alone preamps. This Onyx design builds upon the acknowledged quality of the previous VLZ XDR preamps, to provide greater transparency, lower distortion, more headroom, and a slightly larger-than-life character.

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Mackie Onyx 800R

Ins & Outs The 800R is housed in a relatively deep 1U case which extends 325mm (12.8 inches) behind the rack ears and weighs nearly 5kg (over 10.5lbs). The case is grey-painted steel, but the front panel of the unit is a brushed aluminium, with matching sculpted knobs and grey buttons. Power is supplied by an internal 'universal' power supply which can accommodate mains voltages between 100V and 240V AC. The rear-panel connections are well laid out and clearly labelled, with eight XLR mic input sockets which, when viewed from behind, go from channel eight on the left to channel one in the centre. A button adjacent to the channel-one input activates an M&S decoder in channels one and two, of which more later. Balanced line inputs and balanced analogue outputs are presented on two 25-pin D-Sub connectors, wired to the ubiquitous Tascam eight-channel analogue wiring standard. Suitable cable harnesses are readily available, and probably cost less than buying a bunch of separate cables, not to mention being a lot tidier! A frontpanel button for each channel selects between the mic and line inputs, so both may be wired permanently to appropriate sources if required. The remainder of the rear panel is taken up with the digital interfaces — the A-D is a permanent feature, not an optional extra as it is with so many other multi-channel preamps — supporting all the standard sample rates between 32kHz and 192kHz, with 24-bit or 16-bit resolution. A pair of Toslink optical ports (complete with dust shutters) provide ADAT-format signals. At the lower sample rates these both provide all eight channels. At double sample rates they are configured according to the SMux II protocol, with channels one to four on one port, and channels five to eight on the other. If the quadruple rates are selected then the SMux IV protocol is used, outputting channels one and two on the first port, and channels three and four on the second. There is no access to channels five to eight via the ADAT ports in this mode. Next is a BNC socket to accept an external word-clock reference, complete with a button to select high-impedance or 75(omega) termination modes. Sadly, there is no word-clock output socket, which seems a bit of a shortfall for anyone wanting to use multiple 800Rs synchronised together. It is still possible to synchronise multiple units, but the clock has to come from the DAW or recorder and be distributed to each 800R. The main danger of this approach (which is not pointed out in the handbook), is that if the DAW slaves to its digital inputs (as most do by default), this is an instant recipe for word-clock howlround. Finally, another 25-pin D-Sub socket provides AES-EBU or S/PDIF outputs, and file:///H|/SOS%2002-05/Mackie%20Onyx%20800R.htm (2 of 7)9/22/2005 8:18:34 AM

Mackie Onyx 800R

this is wired according to the Yamaha mini-YGDAI digital wiring standard. A point worth noting here is that standard Yamaha DB25-to-XLR breakout cables are generally wired with four male XLRs for the four stereo AES outputs, plus four female XLRs for the corresponding AES inputs. The 800R only provides the outputs, of course, so it would be better to either make your own cable or have a suitable cable made. The handbook provides all the necessary pin-out information. Three buttons above the connector provide various operating modes. The first determines 110(omega) or 75(omega) source impedance to suit whether the connector is being used to feed AES-EBU or S/PDIF inputs. The second button switches between professional (AES-EBU) or consumer (S/PDIF) status data formats. Although most equipment doesn't seem to care how these bits are formatted these days, it can be a useful facility useful should you need to strip out copy-protection bits from a master created on a consumer DAT machine, for example. The third button configures the connector for dual- or single-wire operation. Most modern interfaces support single-wire connections these days, where each AES-EBU channel carries a pair of signals in the usual way, regardless of the sample rate. However, older interfaces couldn't operate at the higher rates, so the dual-wire mode re-configures the outputs with only one channel per AES connection. Thus, only the first four channels become accessible when the interface is switched for dual-wire operation at the double and quadruple sample rates.

A Peek Inside Given the unusually heavy construction of the 800R I felt an uncontrollable urge to remove the lid and take a peak inside. A large, separately enclosed powersupply module runs the full depth of the unit along the left-hand side, and the whole of the remainder of the case floor is covered with a large circuit board constructed almost entirely with surfacemount components. The various subsections of circuitry are clearly delineated, with input stages, control sections, output buffers, and A-D — all with plenty of space and ground planes to minimise crosstalk. Clearly, this is a very carefully designed product with well-specified components and lots of attention to detail. The A-D circuitry occupies an area of the circuit board alongside the PSU, using four high-spec AKM converter chips. The area of real estate employed for this section is far greater than could be accommodated by most plug-in A-D card options I've seen used by most other multi-channel preamp manufacturers, which I take to be an indication of the quality of this design.

Controls & Operation Moving now to the front panel, each of the eight separate channels share the file:///H|/SOS%2002-05/Mackie%20Onyx%20800R.htm (3 of 7)9/22/2005 8:18:34 AM

Mackie Onyx 800R

same basic features and controls, although the first two channels also incorporate a switched input-impedance facility, and the last two provide frontpanel DI inputs. The rotary gain control for each channel spans zero to 60dB for the microphone input, and -20dB to +40dB for the line input — the source selection being performed by a grey button adjacent to the control. Three more buttons above each gain control provide phantom power to the corresponding XLR socket, a high-pass filter (turning over at 75Hz with an 18dB/octave slope), and polarity reversal — the last two being functional for both mic and line inputs. Only the phantom-power button has an associated LED: all the others have a white collar at their base which becomes invisible when the button is pressed in. The function of each button is marked on the button cap in a very clear form. A trio of LEDs above the buttons give an indication of signal level for each channel, with green for signals exceeding -20dBu, orange for signals exceeding 0dBu, and a red O/L LED for when the signal gets to within 3dB of clipping.

Built-in M&S Decoder I mentioned earlier that the first two channels incorporate an M&S decoder, activated with a button on the rear panel. So, if you choose to set up an M&S microphone array using a forward-facing omni, cardioid, or hypercardioid mic, together with a sideways-facing figure-of-eight mic, you can decode the output to normal stereo by simply pressing the button. The gain of the Sides-mic channel (channel two) effectively controls the stereo width — and because the decoder is immediately prior to the output stage, it is accessible to both mic and line inputs, making it a very useful facility. You could, for example, record the M&S signals in their native format, and then decode them as a post-production process, running the recorder's line outputs through the unit, to control the stereo width as required. However, there are two frustrations with this M&S facility: firstly, the decoder button is relatively inaccessible on the rear panel; and secondly, there is no front-panel indication to let the user know when the M&S decoder is in circuit. I would have preferred this function to be allocated to inputs three and four (instead of one and two), as there is sufficient panel space on these channels to accommodate the extra button. Failing that, an extra pair of LEDs on the panel above channels one and two would have been handy to indicate when the decoder was working — there is already a panel graphic here to remind users which channels should carry the Middle and Sides signals. The first two channels also incorporate rotary switches to select the microphone file:///H|/SOS%2002-05/Mackie%20Onyx%20800R.htm (4 of 7)9/22/2005 8:18:34 AM

Mackie Onyx 800R

input impedance — a fashionable facility these days! The four options provide 300, 500, 1300, and 2400(omega) — the last figure being the standard impedance provided by the other six mic inputs. Microphones with transformer outputs often benefit from lower input impedances, but there are no real rules — look on this control as a kind of subtle tone control. While talking about impedance, the line inputs are loaded with an undemanding 20k(omega) for balanced inputs (10k(omega) for unbalanced connections). Channels seven and eight are equipped with a front-panel quarter-inch socket (unbalanced) to accept guitar inputs, presenting a 1M(omega) input impedance. A button is provided alongside to activate the DI input when required, and the channel gain structure is the same as that for the line input.

Specifications & Digital Functions The input headroom is impressive for all sources. The microphone inputs can accommodate signals up to +22dBu (with gain at the 0dB minimum), as can the line and instrument inputs (with gain at the -20dB minimum). The balanced analogue output can provide signal peaks to +24dBu. Crosstalk between channels is better than -100dB at 1kHz, the EIN figure for the mic input is 129dBu with a 150(omega) source, and the residual output noise at the line output is -102dBu (-113dBFS on the digital output). The overall dynamic range is quoted as better than 123dB (mic input to line output) and better than 113dB via the A-D converter — both very good figures for a device where each preamp and converter channel effectively costs less than £140 in the UK! The left-hand side of the front panel contains the digital converter controls and the mains power button. An octet of LEDs display the sample rate and external clock status, and the mode can be selected with an adjacent rotary switch. When the sample rate is stable, a Lock LED above the switch illuminates steadily — it flashes if there is a problem. Another grey button switches all of the digital outputs between the full 24-bit resolution, or a truncated and re-dithered 16-bit output — a pair of LEDs show the current mode. The handbook provides no information at all about the nature of the dither signal, and I assume it is a simple TPDF application.

Using The Onyx 800R My own 3U flightcased preamp rack contains the slightly unusual combination of a GML four-channel mic preamp partnered with a Focusrite ISA428 — the GML outputs being hooked through the Focusrite's eight-channel A-D converter board to provide eight channels via AES-EBU. The 800R is, in comparison, a far neater proposition, and it was instantly obvious that its performance gave little away. I have been using the XDR preamps in a Mackie 1402 VLZpro mixer for quite a file:///H|/SOS%2002-05/Mackie%20Onyx%20800R.htm (5 of 7)9/22/2005 8:18:34 AM

Mackie Onyx 800R

while and have found them to be remarkably good — not top league, admittedly, but not far off either. OK, so the bottom end doesn't have the solid expansiveness of high-end preamps like the GML, or the slightly warm and cuddly sound of the ISA428's transformers, and the 1402's mix amps and output drivers restrict the headroom (I never run the Mackie desk hotter than +12dBu for that reason). But even so, with care and attention to detail, it is perfectly possible to extract excellent-quality recordings from the XDR preamps. However, these new Onyx preamps certainly represent a step up from those in the VLZ range. The 800R really has bags of headroom — I could drive the analogue line outputs to over +20dBu without any problems at all — and the sound, while still detailed and clean, does have a touch more warmth and bottomend weight than the VLZ design. Overall the 800R produces a very classy sound which is in a league far higher than its pricing would imply. It's still not GML calibre, but I found myself preferring it over the ISA428 in certain situations where a more transparent, less 'flavoured' sound was required. There were a few odd things about the performance though. I noticed that changing the impedance settings on channels one or two also changed the phase response at the low end — something shown clearly on my DK Audio meter. When aligning the input gain of channels one and two for a stereo source, instead of the expected straight vertical line I saw a narrow ellipse — clear evidence of a phase shift between the two channels — and it turned out that I had one channel switched to 500(omega) and the other to 2400(omega). I doubt such technicalities will concern anyone, and I couldn't hear the effect anything like as easily as I could see it, but it was an interesting discovery all the same! The digital outputs are very flexible, with ADAT, S/PDIF, and AES-EBU options, and compared to my reference Apogee PSX100 I judged the converter quality to be extremely good. Using a BCD Jitterbug, I measured a very good signal jitter figure of 7ns (compared to 5ns from the Apogee). However, I couldn't get full 0dBFS modulation from the 800R's digital outputs, as either the preamp or converter input stages appeared to clip at -0.5dBFS. I suspect this is actually an internal calibration error, but it is disappointing nonetheless. If you are relying on the digital meters of your recorder or DAW, you could potentially be clipping the 800R inputs without knowing it, although in practice — assuming that you leave plenty of headroom when recording — this minor discrepancy won't be a problem at all. Overall, the Onyx 800R is an impressive piece of kit: well built, well specified, classy sounding, and versatile. It has a few quirks and frustrations, but if you need more good-quality inputs for a digital mixer, or an audio interface with ADAT or AES-EBU connectivity, or even a simple rack of eight good mic/line preamps with analogue outputs, the Onyx 800R will fit the bill very nicely. Published in SOS February 2005


Mackie Onyx 800R




MOTU Digital Performer 4.5

In this article:


The Right Tools For The Job MIDI + Audio Sequencer (Mac OS X) Installation Published in SOS February 2005 The DP 4.5 Environment One Window To Rule Them Print article : Close window All Reviews : Software Working With Data Mixing In DP Track Types In DP Getting Around File Formats The migration to OS X has caused major upheavals for The MOTU Audio System both software manufacturers and musicians, but Mark DP 4.5 And Pro Tools of the Unicorn's long-established Digital Performer Added Value sequencer is better than ever. Further Exploration In Use

MOTU Digital Performer 4.5 £599

Robin Bigwood

Digital Performer goes back a long way, and has its roots in Performer, The best version of Digital one of the very first Mac sequencers. Performer ever! Throughout its history, MOTU have Full latency compensation, gradually developed and refined the across MAS, AU, TDM and application rather than ever RTAS formats, with no 'reinventing' it, and as such it has over limitations. time accumulated a vast range of Broad plug-in support with excellent third-party options. features and working methods. One Performance enhancements thing that has never changed is DP's valuable for all users, but fundamental 'character', which remains especially owners of extremely customisable, very musicianPowerbooks and older Macs. friendly, and unashamedly Consolidated Window is a individualistic. Even the OS Xgreat asset for workflow and compatible version 4.0 retained some organisation. stubbornly non-standard GUI features cons that sometimes seemed to owe as Non-standard user interface can take some getting used to. much to System 6 as Apple's Aqua technology. Some bundled plug-ins pros

rather ordinary. Compatibility with Pro Tools systems still some way from ideal.

summary With version 4.5, MOTU have reaffirmed Digital Performer's big-league status. Several

Digital Performer 4.5 shows off one of its newest features: the Consolidated Window. This allows users to flexibly configure their workspace whilst ensuring that windows never overlap or become hidden — a very nice user-interface tweak, and a vast improvement over previous versions of DP. Also shown is the Tools palette and transport, otherwise known as the Control Panel.

Though some of its new features had been trailed for a while beforehand, DP 4.5 was delivered at the end of 2004 with many surprising and very welcome new capabilities, some of them major advances, and some more subtle but equally valuable. DP 4.5, as you'd expect, can do just about anything any of the other 'big'

file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (1 of 14)9/22/2005 8:18:41 AM


new features should delight existing users whilst putting the competition on the back foot.

information £599; upgrade from previous DP versions £98.12; crossgrade from other sequencers £249. Prices include VAT. Musictrack +44 (0)1767 313447. +44 (0)1767 313557. Click here to email www.musictrack.co.uk

sequencing packages can do. It has extensive MIDI facilities, audio recording and editing, a virtual mixing environment, music notation, soft-synth hosting and Rewire compatibility — in fact everything you'd need to take almost any musical or sound-related project from inception to completion. Of course, it's the details of how DP does all this that give it a character that's distinct from Pro Tools, Cubase SX, Nuendo or Logic, and which generate so much strong feeling in advocates for or against it. Choosing a sequencing platform can sometimes feel like choosing a religion, so it pays to consider all the options carefully! DP still isn't as well known in Europe as it is in the US, but it's a strong contender for anyone looking for a platform to commit to, or, indeed, defect to...

The Right Tools For The Job

www.motu.com

Test Spec Digital Performer version 4.5. Apple dual 867MHz G4 with 1.25GB RAM and Powerbook 1GHz G4 with 1GB RAM, both running Mac OS 10.3.6. Tested audio hardware: MOTU 896HD Firewire interface, Digidesign Pro Tools HD core with 192I/O, Mac builtin audio.

DP 4.5 will run on pre-G3 Macs, but my advice would be not to try this at home, or anywhere else for that matter! A much more realistic minimum spec is a G4 with no less than 512MB RAM, but having said that, even old single-processor models will work quite well, and vastly better than they did under DP 4.1, thanks to a new feature in 4.5 that modifies the way that it integrates its processes with OS X's multi-threaded processing. Most Macs produced in the last two years or so should be good hosts for DP 4.5, but clearly if you want the best performance you need to be looking at a G5, and preferably a dual-processor model with as much RAM as you can afford. DP 4.5 isn't a processor hog, but it's not a miracle worker either. Staying on the subject of hardware, DP 4.5 can of course use the Mac's built-in audio facilities for recording and playback, but you get more flexibility and better sound quality by pairing it The Sequence Editor is where you do most up with a dedicated audio interface. of your non-destructive audio editing, Clearly, MOTU would love you to opt working with so-called 'soundbites'. A range for one of their own models, but any of editing options is available directly using Core Audio-compatible audio interface the mouse, but additional actions can be will do, and DP will also work with applied via the menus. Digidesign hardware such as Pro Tools Mix, HD and Accel systems via DAE version 6.2 or later. For more on using Digital Performer as a front end for Pro Tools systems, see the box on page 140. As for MIDI, DP will use any USB interface that is Core MIDI-compatible, and if you choose a compatible MOTU model you get sub-millisecond timing accuracy courtesy of their Midi Time Stamping (MTS) technology. If you're not a mouse lover you can control DP from various hardware control surfaces including the Mackie Control and HUI, and the Radikal Technologies SAC 2.2, and there are sections in the DP 4.5 manual describing the use of each of these. DP is also extensively controllable by keyboard shortcuts, or indeed MIDI messages, and version 4.5 introduces mouse scroll-wheel support for window scrolling and altering parameter values.



Installation DP 4.5's rather imposing box includes two CDs and two manuals. The first CD is the main installer, and also includes an up-to-date set of hardware drivers for MOTU audio and MIDI interfaces, a couple of demo projects, and the obligatory Read Me file. The second CD contains about 600MB of audio loops from various companies. Of the two manuals, the slim Getting Started guide has lots of good information on installation and setting up the rest of your system, while the 996page User Guide is where all of DP's features are covered in detail. Both manuals are very nicely produced and are surprisingly readable, though as they don't exist as PDFs, users of mobile rigs might end up lugging round the User Guide more often than they'd like. As far as copy protection goes, DP 4.5 is easy to live with. After installation, and with the installer disc still in the drive, you're asked for a key code, which is printed in the back of the User Guide. Enter this along with your name and you're away, with not a dongle or Internet connection in sight.

The DP 4.5 Environment In essence, the look and feel of Digital Performer haven't changed very much since the days of DP3, though version 4.5 has introduced some user interface changes that are quite far-reaching whilst still retaining complete backwards compatibility for the luddites amongst us. For a complete discussion of DP's basic sequencing concepts, and a thorough look at its main editing windows, I'd urge you to read the review of DP3 in the November 2001 issue of SOS, available on-line at www.soundonsound.com/sos/nov01/articles/motudp3.asp, but I'll summarise the main features again now. Whenever you work with DP 4.5 you're working on a Project, which can contain one or multiple individual Sequences. Projects automatically get a folder on your hard drive, and this contains a single file for your Sequence data, plus other folders for audio files, analysis files, and other bits and bobs that go to make up a Project. It's a sensible, flexible and well organised file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (3 of 14)9/22/2005 8:18:41 AM


system. DP 4.5 offers a typical range of windows for controlling and editing your Sequence. The Control Panel acts as a transport control and gives access to many other basic functions, whilst the Tracks Overview and Sequence Editor windows offer Some users may never have cause to use it, but DP's support alternative but equally for surround mixing is extensive. Mono and stereo tracks can useful 'timeline' views of output to multi-channel 'bundles', in which case they're your Sequences very equipped with one of four surround panners that can be swapped at any time. much in the manner of other sequencers' Arrange pages. There's a single Mixing Board which displays MIDI tracks alongside audio tracks, and a range of other windows provide more specialist environments for certain tasks. These include the MIDI Graphic Editor (a classic 'piano roll' design), the Drum Editor, Event Lists, the Soundbite window, which keeps track of audio use in your projects, and the Audio Monitor window, which shows input levels prior to recording. Pretty standard stuff, you might think, but it all becomes unmistakably DP when you look at the window title bars. MOTU replace OS X's 'traffic lights' with a couple of square buttons with triangle icons and another which sports some superimposed squares. In addition there's a 'mini menu' and various other buttons which all relate to specific functions. Whilst this scheme is non-standard it's certainly not unfriendly, though the familiarisation with button functions and mini-menu contents that new users have to go through probably does represent a brief steep section of the DP learning curve.

One Window To Rule Them All So how easy is the DP environment to get along with? The answer to that is: a lot easier than it used to be! This is down to the new Consolidated Window, one big über-window that can contain a whole host of others in user-configurable cells. It's very easy to work with: windows that generally need to be longer than they are tall (like the Tracks Overview and Sequence Editor, for example) open up in a wide, central area, with other 'portrait-format' windows opening in two sidebars. There's no limit to the number of windows that can be accommodated vertically in each sidebar or the central area, and combined with the fact that the majority of windows can be popped in and out at will, the Consolidated Window is a very powerful. It revolutionises the use of DP on a 12-inch Powerbook screen, but is equally useful to users of multiple-monitor setups. If it has a failing it's that it doesn't play nice with OS 10.3's Exposé feature — but then the whole point of the Consolidated Window is to avoid the need to use Exposé anyway...



The Consolidated Window may be the most important user-interface development in DP ever, but even previous versions have had some good features for managing appearance and workflow, and these remain in DP 4.5. One is Window Sets, which basically provide a way to save a workspace, with a particular combination of windows in specific locations. Many of DP's editing windows also have an Expand button, which opens up a list of tracks to the left of the window. It's then easy to show and hide tracks, and is particularly useful for hiding all the MIDI tracks in a Sequence, say, when you want to do some audio editing in the Sequence Editor. There are a few minor discrepancies in appearance between the editing windows, such as tracks' play enable buttons looking quite different in the Tracks Overview and Sequence Editor, but it's nothing serious, and in use is almost unnoticeable.

Working With Data In a world increasingly overrun with loop-based music applications, DP stands out as resolutely 'linear' in its approach. Although it can easily handle loop-style material, and even import audio in REX format, DP makes no particular concession to dance music over, say, voiceover editing or film scoring. Notably there is no provision whatsoever for object-based editing, whereby you can duplicate a phrase several times and have all instances of it update when you change just one. Consequently a DP Tracks Overview window looks quite different to a Logic Arrange page, and works in a different way. It would be a foolish person who could say that one approach was better than the other, but they're certainly different. Indisputably, though, DP has very nice audio-editing features. It displays audio content in tracks with so-called soundbites — visual blocks which are just pointers to parent audio files. This way of working is standard for most modern sequencing packages, but the implementation in Digital Performer is particularly intuitive and easy to use. By pointing to different parts of a soundbite with the mouse, the entire soundbite or just a region of it can be selected, 'edge-edited', dragged, time-compressed or expanded, or have fades applied. Many more editing options open up via the tools in the Tools palette, each of which can be accessed with a single keystroke. Edits can be applied to many soundbites simultaneously, and DP is intelligent in its handling of soundbites that are abutting or overlapping, especially in applying various kinds of crossfades. MIDI editing is similarly sophisticated. Using many of the same tools and techniques as for soundbites, the manipulation of MIDI data is straightforward but super-flexible. Continuous data can appear in one of three ways — as points, bars or lines — and whilst Bars mode in particular is particularly informative, Lines mode harmonises the display of continuous data with the way that DP's mix automation data appears. The Reshape tool comes into its own here too, making it easy to impose complex transformations on existing data. file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (5 of 14)9/22/2005 8:18:41 AM


I do have a gripe about MIDI editing, though. It's possible to engage an edit grid which confines selections and event drags to a selected time value, but dragged MIDI events never snap to the grid, and instead retain their relative positions to it. I'd like the option to work both ways, but at present this doesn't exist.

Mixing In DP DP's Mixing Board automatically generates a mixer 'track' (ie. a channel) for every track in your Sequence except the Conductor track. As that's potentially a lot of tracks, there's a track selector list as for some of the MIDI editing windows, and this makes the mixing environment very customisable. DP can also coordinate the order in which tracks appear in the Mixing Board, Tracks Overview and graphic editing windows, such that dragging a track to another location in one of them causes all the others to be instantly updated. The Mixing Board doesn't really contain any surprises, and offers a well thoughtout set of features. There are pop-up menus for configuring inputs and outputs on each track, mirroring the settings in the Tracks Overview and elsewhere, and the faders and metering work in intuitive ways. MIDI faders generate Controller 7 (channel volume) data, whilst the meters light to indicate the relative velocity for individual MIDI notes. They also have insert slots, which can accommodate any of 13 MIDI plug-ins. Most useful amongst these are a newly souped-up Arpeggiator, plus nondestructive Change Duration, Change Velocity, Quantize and Transpose plugins. Audio channel faders offer -infinity attenuation to +6dB boost either side of a unity-gain 0dB setting, and the Looks familiar? The new Masterworks EQ MAS plug-in bears an uncanny resemblance accompanying meters show channel to the Sony 0 and sounds great too. output levels with 0dBFS at the top of their range, though sadly these don't meter incoming audio as a track is being recorded. As is common with modernday DAWs, an 'over' on an individual channel is not necessarily cause for concern, as the headroom offered by DP's 32-bit floating-point internal processing means that any resulting distortion can be eradicated by attenuation later on in the signal path. Audio channel insert slots can accommodate plug-ins in MAS or Audio Units format (more on this below) and can be configured pre- or post-fader.



Ever since version 3, Digital Performer has had extensive surround support. As well as its multi-channel track, Aux and buss facilities, there are four serious surround panners on offer, each fulfilling a slightly different role. Any one of them is available for mono or stereo tracks feeding multi-channel output bundles. Perhaps the most important new feature in the Mixing Board, as of DP 4.5, is the vastly improved audio track Aux send facilities. Tracks can have up to 20 sends, and they now work in a similar way regardless of whether they're of the mono, stereo or multi-channel variety. In fact the flexibility here is staggering, with options to sum or choose individual channels from multi-channel tracks to source a mono send, and even to open surround panners for mono or stereo tracks feeding multi-channel sends. Send level is displayed graphically via a send knob, and numerically as the knob is adjusted, and sends can be muted and designated pre- or post-fader. Mix automation is not purely a Mixing Board affair, as it applies equally to plug-in parameters, but it's certainly visible here in the form of faders and pan pots zooming around. DP marries automation data to individual tracks, and it can be edited in the Sequence Editor in a very similar fashion to MIDI Controller data. Mode selector pop-ups in the Mixing Board determine how automation data is written in each pass — whether it overwrites or modifies any data already in the track — and overall behaviour can be controlled using the dedicated Automation Setup window. DP makes extensive provision for grouping tracks, and can tie together fader and pan movements, mutes, and even editing actions. This can be done on a temporary basis just with a little mouse-clicking, or more permanently by using the Track Groups window, which can form part of the Consolidated Window. It's not immediately obvious when a track is part of a Track Group, but it's easy enough to enable and disable groups when necessary. Another feature of the Mixing Board is Mix Mode. This allows multiple alternative mixes to be set up, with automation and plug-ins all included, but unless extensive use is made of DP's mix Snapshot facilities, fader and pan positions don't get remembered automatically, so Mix Mode is not as easy to use or intuitive as it could be.



Track Types In DP There are eight basic track types in DP 4.5, all of which can be viewed and configured in the Tracks Overview and edited in some or all of DP's editing windows: Conductor track: Every Sequence contains one Conductor track, which handles tempo, key and time signature information. MIDI track: for recording and playing back MIDI information to external hardware synths, hosted software synths and audio plug-ins running in DP which can receive MIDI information. MIDI tracks can also drive devices (in Reason, for example) connected to DP via Rewire. Only one device can be selected at a time, unless you use the Device Group function, but this provides no control over relative volume, or any other aspect, of MIDI devices grouped together in this way. Mono audio track: for recording a single channel audio signal, either from the outside world or from elsewhere in DP. Audio tracks can output into a hardware output or a buss, configured as anything from mono right up to a 12-channel 10.2 surround 'bundle'. Stereo audio track: a two-channel track with a single mixer 'track strip', appearing in editing windows in a single track but with two parallel waveforms. Surround track: these tracks record and play back multi-channel audio, from fourchannel Quad and LRCS, through 5.1 and up to 10.2. Surround soundbites appears in editing windows with their component channels' waveforms 'stacked', not running in parallel. Aux track: an Aux track doesn't record or play back audio itself, but acts as a sort of conduit from a source to a destination. For example, you could set up a reverb plug-in on a single Aux track, and then send dry signal to it from multiple audio tracks, via DP's sends and busses. Alternatively, Aux tracks can be used as virtual input channels, handling audio from a hardware input, processing it with plug-ins on its insert slots, before outputting the signal, wet, for recording on an audio track. Instrument track: a specialised kind of Aux track that carries a software instrument in its uppermost insert slot. It doesn't have an audio input, and you can't record MIDI or audio on it — it just provides a platform for soft synths. Master Fader track: Master Fader tracks provide a 'master' level control for a bundle of hardware outputs — hence they can be anything from mono to 10.2 — and they're also ideal for carrying mastering-style limiters and other effects, coming right at the end of the mixing environment audio chain.

Getting Around DP's Mixing Board features are fairly extensive, but they're by no means the whole story when it comes to signal routing. Actually, with MIDI tracks what you see is pretty much what you get. A single MIDI track can drive only one device, unless you assign it a Device Group (see the 'Track Types' box below). Support for inter-application (and, indeed, 'intra'-application) MIDI is excellent, with all DPhosted software instruments appearing in MIDI track output pop-up menus, along with Reason synths connected by Rewire, and any audio plug-ins that accept a MIDI input (like Auto-Tune, for example). The flexibility of audio routing is an area in which DP really shines. All audio and file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (8 of 14)9/22/2005 8:18:41 AM


aux tracks can access a potential 198 busses (ie. 99 stereo busses) as either inputs or outputs, and these can gather signals from multiple sources and route them to multiple destinations, including other tracks or plug-in side-chain inputs. Busses can be mono, stereo or have as many channels as are necessary to support all of DP's surround track types.

File Formats When recording audio, DP creates files in the uncompressed Sound Designer II format, at sample rates between 44.1kHz and 192kHz (the current maximum sample rate offered by MOTU's line of audio interfaces) and at either 16- or 24bit resolution. DP can import audio in Sound Designer II, WAV, Broadcast WAV, AIFF, REX, Acid, MP3 and a few other formats, but it can't play back files with multiple sample rates and resolutions without first converting them to be compatible with the audio settings chosen for the project. This conversion process can be done automatically, so you can freely drag audio files from the Finder into tracks without having to find out their specs first, but it would obviously be a great development if DP could, at least, cope with playback of 16and 24-bit files without having to convert one of them. DP has had various Export and Bounce to Disk functions for some time, but DP 4.5 makes things much more convenient. Previously it had been a multi-step process to export a mix to, say, AIFF format, but the Bounce to Disk window now allows bounces of individual soundbites or entire mixes to formats including SDII, AIFF, WAV and MP3, the latter courtesy of the LAME encoding framework. Bounce to Disk settings can be saved for later recall, and you can set up a kind of 'batch bounce' whereby multiple Sequences and Projects can be left alone to bounce themselves.

DP's bundled plug-ins are all in the MOTU Audio System (MAS) format, and there are some good ones on offer. Shown here are the Plate reverb, the very flexible Delay, and the multi-band Masterworks Compressor.

DP also has a Freeze Tracks function, so that processor-intensive plug-ins and soft synths, for example, can have their output 'printed' to disk and then be temporarily disabled, freeing up valuable CPU resources. Sadly this is not a faster-than-real-time function, nor will it run in the background, but it's certainly useful if you've not got processor cycles to burn.

The MOTU Audio System file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (9 of 14)9/22/2005 8:18:41 AM


DP has long been a 'front end' option for owners of Digidesign Pro Tools hardware, but the majority of users probably use other interfaces with the MOTU Audio System. MAS is the engine that drives DP's audio, but it's also the name given to MOTU's own plug-in format. For some users the proprietary nature of the MAS format is seen as a disadvantage, but having worked with it for some time I would argue just the opposite. To begin with, MAS support amongst thirdparty developers is surprisingly wide — some of the industry's heavyweights such as Waves and Antares are amongst the most active supporters of MAS. Also, the fact that MAS plug-ins only work with DP (and its little brother, Audiodesk) makes for a degree of stability and compatibility which is probably unmatched elsewhere in the world of 'native' sequencing software. A key new feature in DP 4.5 is plug-in latency compensation, which feeds audio ahead of time into tracks which are carrying plug-ins that impose a processing delay. This works exactly as you'd want it to, ie. seamlessly throughout the mix environment, on every type of audio track. It's a superb implementation, and something that earns DP considerable kudos. The MOTU Audio System also supports the Audio Units plug-in format, which opens up the entire Logic-compatible plug-in market, including software from the likes of Native Instruments. Technically DP is 'wrapping' Audio Units in a MAS shell, but this is not apparent in use, and parameter automation works as for MAS plug-ins. What is not possible under DP 4.5 is the loading or saving of Audio Units presets in the Apple standard 'aupreset' format, unless the plug-in has its own file management. That's something I hope MOTU will be able to remedy before too much longer. You can also add extensive VST plug-in support courtesy of Audioease's VST Wrapper, which has recently been updated for DP 4.5 to incorporate plug-in latency compensation. Added to that, there's full support, including latency compensation, for Mackie's UAD1 and TC's Powercore plug-in platforms. Almost as useful as latency compensation is dynamic CPU management. This effectively disables plug-ins when they're not actually processing audio, and whilst it doesn't make your Mac capable of running any more plug-ins or audio tracks in total, it certainly helps to ensure that the user interface and background processes are kept snappy when a Sequence is not actually playing.



DP 4.5 And Pro Tools Digital Performer has long been known as an alternative 'front end' application for users of Digidesign Pro Tools hardware, and with version 4.5 MOTU promised greater functionality than ever before for DP running under DAE rather than MAS. Features include plug-in automation and latency compensation, instrument track support, compatibility with RTAS and Audiosuite plug-in formats, mono and stereo sends and send automation, and a whole host of other smaller enhancements.

A veritable gaggle of Pro Tools plug-ins. DP 4.5 will work as a 'front-end' for Pro Tools systems, and then gets access to any TDM and RTAS plug-ins installed on your Mac. DP 4.5 can't run MAS or Audio Units plug-ins alongside them, though.

I was able to test out DP 4.5 with an up-to-date Pro Tools system at Studio Spec in West London, and whilst it proved to be a useable combination, some aspects didn't work as well as I'd expected. For example, whilst TDM and RTAS plug-in hosting seemed very reliable, DP didn't categorise plug-ins into types in the way that Pro Tools 6 does, so insert slot plug-in lists were often unmanageably long. Strangely, too, only TDM instruments can be accommodated in DP Instrument tracks; RTAS instruments have to go in audio tracks. A more serious problem is the absence of surround support under DAE — it was possible to create surround output bundles, but they could never be configured. And despite what is written in the DP 4.5 manual, the automation modes Trim Touch and Trim Latch are not available. MOTU are aware of these problems, and I imagine fixes are in the pipeline, but there are no promises yet as to when or even if they will happen. In the meantime DP 4.5 is quite capable of handling complex work under DAE, but not with the same level of refinement as is delivered with MAS.

Added Value The bundled plug-ins that form part of a DP 4.5 installation are all in MAS format, and range in quality from excellent to somewhat average. Virtually all basic effect and processor types are covered, though some more unusual treatments like granular synthesis are absent. Special mention must be made of the new Masterworks EQ plug-in, which bears an uncanny resemblance to the Sony Oxford EQ plug-in available for TDM and TC Powercore platforms, and is really outstanding. The Plate reverb, Masterworks Gate and multi-band Masterworks Compressor are also all very good, and some 'utility' plug-ins primarily designed for surround mixing work — Bass Manager and Calibration — are worth their weight in gold. Some are less impressive, though. Preamp-1 turns out a range of distortion, EQ and compression effects, but it's not really comparable to specialist amp modelling plug-ins like Amplitube or Guitar Rig. Similarly, eVerb, although file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (11 of 14)9/22/2005 8:18:41 AM


highly configurable, is trounced by several freeware and shareware reverbs in terms of sound quality. Chorus is disappointing, never really producing a very musical result, and I particularly miss having a good, solid single-band compressor. It's worth mentioning, though, that the majority of bundled MAS plug-ins are available in mono, stereo and multi-channel versions for surround sound work. In particular, the Delay plug-in, which in stereo applications is a rather nice pingpong-type effect, can pull off some wonderful tricks when used in its multichannel guise, bouncing signals around between all the channels.

Further Exploration One of the most heralded features in DP 4.5 is its new Beat Detection Engine. This analyses audio, looking for transient content and then flagging the resulting 'beats' without splitting up the original audio. Various actions can then be applied, such as time-stretching or compression, audio quantising, groove extraction and tempo analysis. It's a sophisticated feature that, when switched on, runs in the background and is not at all intrusive. It's far from being just a Recycle wannabe, and definitely repays some thorough investigation and experimentation. I found it very effective in working with a variety of rhythmic and monophonic melodic tracks, though occasionally it finds beats where, for a listener, there clearly aren't any. A more long-standing feature, POLAR (the Performance Oriented Loop Audio Recorder) is like a Lexicon Jam Man or Electrix Repeater in software, and with much greater memory capacity and more features. It's perhaps one of DP's bestkept secrets, and is a fantastic tool for musical inspiration and experimentation as well as tracking backing vocals, for example. It would be very wrong to not mention here Digital Performer's A/V facilities, as this has always been one of its strongest areas. Individual Sequences can import Quicktime Movies, which then play back in a dedicated Movie window as well as being shown in a timeline format in the Sequence Editor. DP 4.5 can compensate for the latency introduced by external Firewire-based video output equipment so that video displayed on dedicated monitors remains synchronised with the Sequence. Also, there are extensive facilities for synchronisation with external video equipment via SMPTE timecode, a sophisticated Find Tempo function which helps to tie musical content to cuts and hit points in video, and integrated support for Synchro Arts' Vocalign ADR system.

In Use So what's it like to use DP 4.5? Obviously that depends very much on what you're trying to achieve, but my experience of it, and that of several other users file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (12 of 14)9/22/2005 8:18:41 AM


whose opinions I sought as part of this review, is overwhelmingly positive — working with DP 4.5 is very enjoyable. Like previous versions before it, DP 4.5 doesn't channel users in to one way of working, and strikes a good balance between being quick and fairly intuitive in use and providing tremendous depth when it's required. A willingness to read the manual is crucial in learning DP 4.5, though, because one or two important features are not at all obvious to the newbie. For those who find this an unpalatable prospect, several companies now provide support for learning DP in the form of Quicktime movie demos, an on-line course or a 'how-to' book. MOTU provide more details of these on their web site. As I mentioned previously in this review, DP is perhaps not the first choice for those who want to work exclusively with loop-based material, but for virtually any other genre, from dance music to classical editing, there are plenty of bread-and-butter features. Of course, DP 4.5 isn't perfect. I've noted several niggles whilst covering The Beat Detection Engine is one of DP 4.5's DP's features above, but there are a major new features. Audio files are analysed few others. Although I experienced no for their transient content and the resulting problems during the testing period, beats 'flagged'. Audio can then be timeother users have reported some stretched or compressed with minimal artifacts, quantised, and subject to a range of inconsistencies with DP's Undo beat-driven editing functions. The Waveform system. Some multitimbral virtual Editor, shown here, is where you get to the instruments (like NI's Kontakt) can't heart of the Beat Detection process. make use of their multiple audio outs, and it's not possible to simultaneously view multiple takes within tracks. Then there are the tiny things like audio export in MP3 format being rather slow. These things may be a big deal for a few users, but there's nothing really important amongst them for most of us. The bottom line is that, as of version 4.5, MOTU have virtually all of what Digital Performer is about working very well. It's flexible, has plenty of powerful features, and it's reassuringly stable on the latest releases of OS X. The Consolidated Window and performance enhancements represent a huge improvement over previous versions, third-party support is already extensive and continues to grow, and the same goes for the DP user base. Clearly, DP 4.5 doesn't attempt to offer the all-in-one solution, with samplers, soft synths and all, that Logic 7 does. But buyers who can make use of DP 4.5's crossgrade pricing offer (and if you can't, you're not really trying!) can get it for £249 as opposed to Logic's £699. That leaves enough for MOTU's Mach Five sampler and plenty besides. The first-class integration of MOTU's audio and MIDI hardware also works in DP's favour, especially since Apple no longer sell Emagic hardware. With the advent of version 4.5 there's now absolutely no excuse to overlook DP when considering a Mac sequencer, and ever more file:///H|/SOS%2002-05/MOTU%20Digital%20Performer%204.5.htm (13 of 14)9/22/2005 8:18:41 AM


reason to choose it over the opposition. DP 4.5 is a mature, sorted application, and a safe bet for Mac-based musicians working in a wide variety of areas. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Phonic PAA2

In this article:

Software Editor In Use Final Analysis

Phonic PAA2 £325

Phonic PAA2 Audio Analyser Published in SOS February 2005 Print article : Close window

Reviews : Accessory

pros Easy to use. Large, clear display. Very portable.

cons PC software disappointing. Some useful test signals on CD-ROM only. Memory can't store readings for line input signals.

summary A practical product which contains all the important basic sound-analysis tools an engineer might wish for when setting up a sound system.

information £325.01 including VAT. +44 (0)20 8808 2222. +44 (0)20 8808 5599. Click here to email www.shure distribution.co.uk www.phonic.com

Test Spec PAA2 editor software for PC v1.0. 3GHz Pentium IV PC with 256MB RAM running Windows XP Home.

Phonic's new audio analyser is certainly portable, but how well does it work? Tom Flint

The PAA2 is a portable sound analyser designed to aid sound engineers, particularly those who have to set up audio rigs on a regular basis. Very roughly speaking, half of the tools on offer help the user identify and measure signal and wiring problems, while the rest analyse the actual sound itself within the acoustic environment. At the heart of the PAA2 is a 31-band realtime spectrum analyser which allows the user to see exactly what is going on in any given burst of audio. Although the entire spectrum is shown across the screen, individual bands are easily selected so that their specific level is displayed (in dBu). This can be done in real time, where values change according to incoming audio, or, alternatively, a snapshot reading can be taken and analysed at a later date. A memory function allows up to 10 readings to be saved, and, if required, an average figure can be calculated and saved, using any combination of the 10. Having taken an average SPL reading, an EQ function can be used to analyse the result, after which the display provides the positive or negative gain value required for each frequency band in order to flatten the response. Even if you don't plan to use EQ to sort out acoustic problems, this display is useful for making the extent of the work clear. Taking SPL readings is made possible by an onboard condenser mic, which conveniently folds away when not in use, but the PAA2 will also measure line-

file:///H|/SOS%2002-05/Phonic%20PAA2.htm (1 of 4)9/22/2005 8:18:50 AM

Phonic PAA2

level signals input via a balanced female XLR connector. Readings can be made in units of dBu, dBV, or Volts, and for each measurement the weighting, level range, and response time can be set. Weighting options are 'A', 'C', and Flat, and response-time settings are 35ms, 125ms, 250ms, and 1s. The level range, for optimising the reading to suit the signal level, differs depending on whether the mic or line signals are being used. Next to the input is a male XLR output, which delivers the signals created by the PAA2's own internal signal generator. On offer are pink noise, a 1kHz tone, and a polarity signal — the last has been included for checking speaker wiring when using the Phase Check mode. Having just the three audio test signals seems a little restrictive, but the PAA2 does ship with a CD-ROM which plays 26 test signals, including the three mentioned above. The 1kHz tone is joined by six other frequencies, ranging from 250Hz up to 12.5kHz, and there is also a handy 31band tonal sweep which is ideal for identifying the shortcomings of a system's output.

Software Editor PC owners may want to take advantage of the complementary software, also included on the CDROM, which offers an on-screen control interface. Connection is made by way of a serial interface cable which plugs into the mini-jack of the PAA2 at one end, and the serial port of a PC at the other. Once setup, the PC's mouse is used to navigate menus and to select functions, and the program itself enables data storage on the computer hard drive. It's also possible to use the computer printer to run off hard copies of saved settings. Unfortunately, the included lead isn't much longer than a domestic cat's tail, so remote operation is out of the question. Only those with portable laptop computers would be able to position the PAA2 freely around file:///H|/SOS%2002-05/Phonic%20PAA2.htm (2 of 4)9/22/2005 8:18:50 AM

Some of the PAA2 display screens: the main settings menu (left); the Memory Store page (centre), where you can save analyser

Phonic PAA2

a room and still have it connected.

readings; and the EQ Setting screen (right), which suggests what EQ settings would be required to flatten the frequency response of the audio system under analysis.

The software itself is easily installed, and my PC synchronised with the PAA2 without any messing about. The interface is basically a colour version of what appears on the display, possessing the same menus and features. It doesn't offer anything much in the way of additional features, but the main function of storing memory data works well, and future revisions will no doubt expand on the features.

In Use Although an AC power adaptor is provided as standard, four AA batteries power the PAA2 quite comfortably for a few hours, making it eminently portable. Even with the batteries, the PAA2 is comparable in size to a cassette-based personal stereo. Mobile phone and digital cameras with more features come much smaller, but they usually have fiddly buttons and tiny displays. For its size, the PAA2 has a huge backlit screen, so data can be read clearly at arm's length, and the buttons are large enough that even the most sausage-fingered engineer won't get frustrated. Curiously, there are two ways to navigate through the menus. You can either scroll through and select items using a single rotating wheel (found on the lefthand side of the case) which doubles as a button, or you can use the cursor and Enter buttons situated just below the display. I found myself using a combination of both, depending on how I was holding the device at any one moment. In the palm of my left hand, for example, the scroll wheel was ideally placed for thumb operation. Operating the PAA2 is a matter of diving in and out of various menus and submenus, changing settings, and then returning to the current mode of operation. Although doing this proved to be very intuitive, repeated use revealed the menu arrangement as tiresome. I suspect that testing the PAA2 required far more flicking back and forth than a user might do in normal practice, but I still wonder whether another button of two might be worth adding in future versions, so that reliance on the menus is reduced. Although there is a master power switch under the mic, the manual recommends using the Off item in the Power menu in normal use. I found it a pain to have to navigate to a submenu to turn off the PAA2, and would have liked the front-panel On/Backlight switch to have had a press-and-hold function to take me straight to the Off menu.

Final Analysis


Phonic PAA2

The PAA2 is one of those products that doesn't need flashy gimmicks to sell; its strength lies in its ability to do a variety of useful tasks simply, effectively, and to a high standard, and in that respect it is a success. In terms of operation, it's a pretty straightforward bit of gear, so not much learning is required. There are a few shortcomings. As mentioned above, the PC software is lacklustre, the serial lead is 'lack-length', and I think there is room for improvement in the menu structure. For some reason, the internal memory is only able to record SPL readings, and not any of the line voltage measurements, which is a shame. I also think that some more of the test signals found on the CD-ROM should be generated by the PAA2 itself, for when a CD player is not available.

The measurement mic built into the PAA2 locks into a stowing position flat against the case, which makes the system easier to carry. In the stowing recess can be found the main power switch and an LCD contrast control.

Even at its comparatively reasonable UK price, the PAA2 is likely to be considered a bit of a luxury by the typical home-studio user, especially one who doesn't test audio systems on a regular basis. However, if you rely on these kinds of analysis tools day to day, then this unit would prove a very portable and convenient solution. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Presonus Firepod

In this article:

It's The Little Things Ins And Outs The Sound Of Three Knobs Turning In Use A Firebox For All Seasons Minimum System Requirements

Presonus Firepod £599 pros Eight clean preamps with good analogue interfacing. Efficient drivers. Good value for money.

Presonus Firepod Firewire Recording Interface (Mac & PC) Published in SOS February 2005 Print article : Close window


Unlike most eight-channel interfaces, the Presonus Firepod includes eight mic preamps — at a very appealing price. Alan Tubbs

cons Only stereo digital I/O. Main and Cue outputs always pass the same signal. Test unit ran hot.

summary Containing almost everything needed to get good sound in and out of your Firewireequipped computer, the Firepod strikes in the middle of the firebox price range, delivering more features than lower-priced systems and sound comparable to higherpriced ones.

information £599 including VAT. Hand In Hand +44 (0) 1579 326155. +44 (0)1579 326157. Click here to email www.handinhand.uk.net www.presonus.com

Test Spec Presonus 1394 audio driver v1.12.0. AMD Athlon 1.7GHz PC with 512MB RAM, running

Much like the USB 1 format a couple of years ago, Firewire for music has finally come of age, providing products which meet most budgets and needs. Presonus brought out their Firestation Photos: Mark Ewing (www.soundonsound.com/sos/feb03/ articles/presonusfirestation.asp) at the dawn of audio Firewire, and have recently introduced a replacement for it, the Firepod. Some features have been dropped, and others added, but their firebox idea remains fixed. The Pod not only replaces the audio and MIDI computer interface, but the need for an outboard mixer in a small studio, too. It will send and receive 10 audio channels at once plus MIDI, through a single Firewire cable. The Pod contains eight analogue ins, all with mic preamps, and eight analogue outs, as well as S/PDIF and MIDI I/O. The analogue half of the Pod functions as a line mixer, with or without a computer.

It's The Little Things When I first opened the shipping box from Presonus, I was blown away by the size and weight of the Pod. I lifted it out of the box with one hand — easily. And despite its metal construction, it is still light enough to carry around in one hand. So you won't have to worry about weight issues when putting it in a rack — as it isn't even six inches deep, excluding the front knobs, it would be hard to secure it at the rear anyway. The unit would be smaller still if it wasn't for the panels on either side, which look as though they're intended to act as heat sinks. Despite its diminutive nature, the Pod is well constructed. It continues the blue-on-brushed-

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Presonus Firepod

Windows XP. Tested with Sony Vegas 5.0b, Cakewalk Home Studio 2XL and Image Line FL Studio 4.5.2.

aluminium/grey colour scheme found on other Presonus products — maybe their design team is stuck in a rut, but the scheme is sharp-looking, with enough contrast to work under less-than-optimum lighting conditions. The box also contained the Presonus software/driver CD, as well as a second CD containing Steinberg's Cubase LE sequencer. Cubase LE is cross-platform and matches the Pod's recording specs, supporting resolutions of up to 24-bit and sample rates up to 96kHz. So, if you don't already have a Digital Audio Workstation (DAW) you can be up and going right out of the box, whether you're a Mac or PC user. Like most Firewire interfaces, the Firepod provides an additional Firewire port for chaining further devices.

Whilst the Firestation used Yamaha's mLAN Firewire protocol, the Firepod uses Presonus' homegrown Firewire software and drivers, which support ASIO, WDM and Core Audio. On my PC, the software loaded in with no problems, and in a couple of minutes I was up and running. After installing the drivers, I plugged in the Firewire cable and turned on the Firepod. The blue sync light blinked a few times, then glowed steadily. After the software installation, a Presonus icon appears in the Windows toolbar; rightclicking on this brings up the CPU use choice (high, medium or low), as well as restore and exit commands, while left-clicking opens a small program box. This is where sample rate (44.1 to 96 kHz), clock source (internal or S/PDIF) and latency (down to 1.5 milliseconds) are set, with bit depth being chosen in your music software. On my system, I set the Firepod up for ASIO operation with the internal clock, and had no problems with a 6ms latency, even on a less-thanoptimum home computer. There were no clicks or pops in recorded audio or playback. The Firepod showed up in my various audio programs, recorded and played back audio and MIDI with nary a glitch and hasn't crashed once. In other words, it did nothing special and only what it was supposed to do — just the way recording equipment should work!

Ins And Outs Around the rear of the Firepod is where most of the interface action takes place. There are the obligatory two Firewire ports, so you can chain another Firewire unit. There are the eight line outs, S/PDIF and MIDI In and Out, and inserts for channel 1 and 2 on separate quarter-inch jack plugs, so you can add an outboard EQ or dynamics processor before passing the signal to the converters. As well as the eight line outs, the Pod includes separate Main and Cue outputs. Both of these reproduce the DAW output that also appears at line outs 1 and 2, and allow you to blend this with a summed mix of the analogue and S/PDIF inputs for zero-latency monitoring whilst recording. This is better than having to work


Presonus Firepod

through a single stereo out, but it's a shame that there is no separate volume control for the Cue out; the space taken up by this output could perhaps have been used to provide another function, although this probably would have raised the price. The mic/line inputs are on the front, on XLR/TRS Neutrik Combi connectors. Again, channels 1 and 2 have an extra feature in the shape of instrument inputs. No switch is necessary; just plug your guitar or bass into either of the two and the Pod automatically bypasses the mic preamp and becomes an active instrument preamplifier. For some people, front inputs may be a problem, and if you have invested in a patchbay and abhor dangling cables, you'll either have to live offended or plug in the inputs needed for every session. I kept my main keyboard plugged into the rear of the Firepod via the inserts on channels 1 and 2, leaving the unsightly cables out of the way and view. This left the front inputs disabled, so I had to disconnect the keyboard whenever I wanted to DI a guitar or bass, but was otherwise a fairly elegant solution. Phantom power is switchable in groups of four — again, a comprise, but one that makes sense, and much better than the all-or-nothing situation one finds on many units. There are no phase switches for the mic inputs, but most DAWs include this feature, so you are not paying for something you probably don't need. Finally, there are no mic pads. It is always nice to find one or two of these, but on the Pod you'll just have to turn the knobs.

The Sound Of Three Knobs Turning More smart compromises are made on the front-panel controls. The inputs line up on the left of the unit, while the pots are on the right. The main out has a volume knob, as well as a separate gain knob for the front headphone jack. Every mic/line in has its own knob, with an associated clip light. Having separate gain knobs for each preamp is not ideal for stereo recording, but all the knobs step through increments so you can match levels between them, and they seemed to track correctly. However, the increments are small, and unnumbered except at either end, which makes it hard to see the relative levels. One of my pet peeves with the 'incredible shrinking mixer' syndrome is that densely packed pots are harder to adjust, and make it harder to pick out the right one in a hurry, but Presonus have tackled this by positioning the knobs diagonally to one another, leaving enough room between them to make adjustments without jogging neighbouring pots. The last front-panel knob controls the Main and Cue output balance between the direct analogue mix and the playback from channels 1 and 2 — the direct analogue mix always reflects the levels at the preamps. The fact that you can only monitor the first pair of DAW output channels at the Main and Cue outputs may be limiting in some situations, but except for surround sound mixes, most people use outs 1 and 2 for their monitor mix anyway, and the other outputs have a variety of uses. If your DAW has the busses, you could set up six analogue auxiliary outputs to your favourite outboard gear and return them. You could bring in your MIDI tracks through an outboard mixer via input 1 and 2, while


Presonus Firepod

running the entire mix through yet another outboard unit via those channels' inserts. Outputs 1 and 2 could send this final mix to an analogue recorder, and you could monitor the project through the Main outs. The digital in/out could send a mix to a digital recorder, or add hardware reverb. Not bad for a line mixer. The simple Presonus driver box, with the

Ergonomics are nice, but they don't included Cubase LE software. matter if the sound isn't. So, how does the Firepod sound? In a word, superb, especially for the price. A-D/D-A converters have reached the point where most mid-priced units are acceptable for pro work, and there is not that much difference between hardware in a given price bracket. Over an audiophile monitor system there are detectable differences, but in the real world — and listening to a whole mix, rather than a single track — these tend to shrink. If you do have a high-end unit, you could always feed it through the Firepod via the S/ PDIF input. This should firm up the clock timing for the Firepod's converters, too. The Firepod's preamps are also remarkably clean and precise when you consider that you get eight of them for your money. I compared them with the pair that came with Yamaha's i88X Firewire interface, which has outstandingly good preamps derived from their upscale DM2000 digital mixer. The biggest differencee between them was that Yamaha's had 6dB extra gain (60 compared to 54). They also sounded a tad better — as well they should, since the i88X lists for over half again what the Firepod does, and only has the pair of preamps. After sending the i88X back, I added backing tracks and an extra vocal chorus to a song I had recorded with it. I pressed the Firepod into service, and it delivered. There was no practical difference, even on the lead vocal. So, I don't think any wicked record executive is going to trash your precious demo, saying "They should have used a Double-plus GoodSound Tube on the lead vocal!" The Pod's preamps match mid-priced mixers, which have been used to sell a lot of music. Finally, compared to my old Ramsa desk, the difference is easy to hear. Digital converters are not the only elements of music technology to have benefited from an increase in price/performance ratio.

In Use As an interface for mobile recording, the Firepod excels. The sound quality is up to scratch and the Pod is light and small enough to slide into a gig bag instead of a rack —simply unplug it and go. I did just that for some friends who needed a demo. They are a garage band, which is where I recorded them. Using the Firepod saved me two trips each way, since I didn't have to schlep a rack or mixer in and out. This was especially appreciated after the late load-out, as any


Presonus Firepod

gigging musician/ soundman knows. The Pod performed the same duties a stand-alone mixer did on a previous demo recording session: the three-piece band went down over seven inputs into the Pod, The Firepod's analogue interfacing includes while the main outputs went into a PA an insert send/return loop on inputs 1 and 2 for monitoring between takes. The (right), plus eight balanced outputs and the Main and Cue outs, which allow you to vocalist was shunted off to a closet monitor input signals at zero latency. with a mic for scratch vocals (which also got recorded — you never know). The Cue out, with vocal, went to a headphone amp/distributor for monitoring. The Firepod had no problem capturing the 'garage' sound of the band, warts and all, and made it easy to finish the job in one session. More delicate sounds were also faithfully captured. For one particularly wispyvoiced female doing a quiet backing vocal, I had to crank the preamps up to 9, but there were no adverse sonic effects. And though my one-room studio is seldom used to record everyone at once, having eight preamps helped with setup. With Presonus' original Firestation, I had to set up acoustic players sequentially or use a submixer. However, I did miss the Firestation's tubes, and had to rely instead on external units. The only problem I encountered was that my early production unit ran hot. When I contacted Presonus about this, they said they hadn't heard of any other units doing this, and offered to exchange it. As I had already budgeted my time, I declined and made the Pod soldier on. The heat didn't seem to have any adverse effects on the sound, and the unit hasn't failed. Previously, Presonus had spent hours on the phone with me and innumerable emails trying to get my Firestation to work after upgrading it to mLAN 2, despite it being a discontinued product running with software that wasn't their own. This is the way support should be, but, unfortunately, isn't always.

A Firebox For All Seasons If you are looking for a centrepiece for computer recording, Presonus seem to have hit the mark with their Firepod. Price-wise, it is at the lower end of the eightchannel interface market, yet the inclusion of eight preamps is something you won't even find on all the higher-priced units. Sound-wise, it is solid, not exotic, but as it offers plenty of analogue I/O, it's easy to patch in other preamps and processors if you so desire. The digital interfacing is, however, less impressive, with only a single S/PDIF in/out. If you are trying to put a bloated studio on a carb-light diet or build a mobile studio from scratch, the Firepod deserves a hard look. And if eight inputs and outputs are a requirement, the Firepod is quite cost-efficient. You won't even file:///H|/SOS%2002-05/Presonus%20Firepod.htm (5 of 6)9/22/2005 8:18:58 AM

Presonus Firepod

need a mixing desk until your studio starts putting on weight again.

Minimum System Requirements Windows XP, Firewire port, 900MHz or better CPU, 256MB RAM. Mac OS 10.3.5, Firewire port, 800MHz G4 or better, 512MB RAM. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Redmatica Autosampler

In this article:


How It Works Automatic Sample Generator Layering Synths Published in SOS February 2005 Using Autosampler Sampling For Fun And Profit Print article : Close window

Redmatica Autosampler 100 Euros

For EXS24

Reviews : Software

pros Great automated results. Easy to use at a basic level. Lots of creative options.

cons The user still has the chore of looping sustained pad sounds.

If you're keen to move to an all-software setup, but still depend on sounds from your hardware synths, Redmatica's utility might smooth the path... Paul White

summary Autosampler is really a mustbuy for any EXS24 user who wishes to create samples using their hardware or software synths, or indeed any combination of the two. The more complex modes demand a little thought from the user, but basic multisampling is a doddle.

information 100 Euros (approx £70). www.redmatica.com

A few years back, I was bandying around an idea for a software utility that would automatically sample MIDI instruments for itself by firing out MIDI notes, recording the audio and sorting out the resulting files into key maps. At the time, no-one seemed very interested in the idea, but now Redmatica have developed a utility called Autosampler, which is designed to allow Logic's EXS24 sampler to automatically sample hardware or software synths. Ideally, such a program would also loop the samples automatically, but that's not so easy as machines aren't particularly good at finding the loop points that are subjectively the best, and Autosampler leaves this task to the user. However, it does go beyond what I had envisaged by offering the additional ability to create instruments that switch or fade between key maps or layer multiple synth sources, using graphic mix volume/controller envelopes to create dynamic fading between the layers. What's more, Autosampler will automatically create multisample sets at multiple velocities. The user chooses how many samples to take across the keyboard (if the default setting isn't to your liking) after which Autosampler does all the tedious sampling and mapping for you. It even looks at the audio coming in and sets the EXS24 release time to match. A suitable host program is needed when sampling software instruments, but as anyone using this program will already have Logic, that shouldn't be a problem.

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The program also offers a few randomise and tweak functions for creating or modifying envelopes and treats the sounds being sampled rather intelligently by incorporating specific normalisation modes, a limiter, automatic sample truncation and trimming, audio overload protection and the aforementioned release-time detection. If you want, you can even set it to sample whole banks of synth patches while you leg it up to the pub! Patch names can be pasted from text files or from Sound Diver, if you have it in your system, and the samples automatically created can all be given unique names that are partly based on the date and time so you don't end up with any duplicate names. All you need to use it is EXS24 on an Apple Mac G4 or above running Mac OS 10.3 or later.

How It Works Autosampler sends a series of MIDI notes at various pitches and velocities (where needed) to the instrument and records the stereo audio output, after which each sample is trimmed, normalised and saved into an EXS24 instrument. Sampling a virtual instrument works in a similar way. A simple setup window allows you to tell Autosampler which MIDI port it should be sending data to and on what channel, and also on what audio input to expect the signal, so you don't even have to repatch your audio interface to make it work. If you don't like the default settings, you can also decide which patches and notes you want to sample and how many velocity levels you want. It is also possible to create what the designers call a multi-dimensionally sampled (MDS for short) instrument by sampling separate key maps for different value combinations of two user-specified MIDI controllers affecting the instrument being sampled. The example given is that if you auto-sample an instrument using MDS, specifying the filter and resonance knob settings as the designated controllers, then on playback you will be able to switch/fade in real time between multiple key maps sampled at various filter/resonance knob positions. As you'd expect, an EXS24 instrument created in this way comprises a number of sample sets, and which one plays back depends on the values of the corresponding MIDI controllers on playback. Alternatively, you can create complex sounds by sampling from multiple synth sources, using graphically driven mix volume envelopes to dynamically vary the balance between the contributing instruments. For ease of use, the software includes a virtual keyboard and on-screen MIDI controller pad. A VU meter displays the current peak and RMS input levels and holds the highest detected peak to show how much headroom is available, much like the meters in Logic itself. These can be reset using a button, and audio monitoring can be enabled or disabled at any time by toggling the Mon button. The Scan tab is where you set the operating mode for your sampling session: here you can decide to sample a single patch, a whole bank or a consecutive set of patches. Clicking Scan Parameters brings out another information panel where the way the program samples can be tweaked in enormous detail, though I think it's fair to say that most people will use it much more simplistically. You can file:///H|/SOS%2002-05/Redmatica%20Autosampler.htm (2 of 5)9/22/2005 8:19:04 AM


arrange to send bank change messages and define the maximum MIDI volume the synth should be sampled at, and you can even have patch names imported automatically from Sound Diver if you use it. The default settings work well for the vast majority of synth sounds, and though you can plunder entire banks, I would imagine most people would sample one patch at a time manually. Still, it's great to have all this extra functionality in such a user-friendly sampling tool.

Layering Synths It's possible to use Autosampler to create single EXS24 instruments based on up to 64 dynamically layered additional sound sources, controlled from up to four additional MIDI ports. For each source you can set up a multi-point graphical volume envelope so that the blend of sounds changes with time. A random function randomises the level envelopes to produce unexpected transitions and fades, should you want this. If you have the patience, you can create whole patches Wavestation-style using individual synths as oscillators, but Autosampler can be used to create multiin most cases, crossfading between layered EXS24 instruments using multiple three or four synth sounds should hardware instruments as the sound sources. be enough. Certainly that's all I had the patience to deal with and the results were adequately complex-sounding! As far as I can tell, all the synths being layered must be mixed to the same audio input port, after which you can double-click on the envelopes in the Layers page to create level or controller changes. This is dead easy and works fine, but if I could make make a suggestion, it would be some sort of solo function on the MIDI and layers pages. A more direct way of sending program changes to the individual parts would also enable you to browse layers more easily.

Using Autosampler Once the MIDI synth is connected to the computer's audio and MIDI interfaces, starting the process involves opening Autosampler, selecting the MIDI and audio ports you have your synth connected to, and setting a destination folder into which the newly created instrument's samples will be stored. This new instrument folder should be placed inside the existing EXS24 Instruments folder in order for Logic to see it. Next you select the required patch on the connected synth, check the levels are OK, then click Start. Autosampler measures the ambient audio noise level prior



to sampling so that it can create a threshold allowing it to detect when a note has started and ended. Each note is held for a user-definable maximum time while being sampled, though if the sound has a natural decay after the note is released, Autosampler does not simply truncate the sample — it keeps sampling until the sounds decays to a point just above the noise floor. Additional trimming and normalising are done automatically using floating-point arithmetic, and if the user specifies 16-bit files, the bit-depth reduction and dithering is done as the last step in the process to maintain quality. Furthermore, in order to completely avoid any possible clipping during sampling, you can set an automatic overload protection option that discards clipped samples and samples them again at a lower volume. Normalisation can be selected so that the final sample level will always be consistent across all the sampled notes in a set, even if the automatic clip protection has made gain changes. Both conventional and one-shot samples can be specified. It seems the designers have thought of just about everything to keep the user out of trouble. I tried sampling one of my hardware Wavestations using a pad sound and Autosampler dealt with it perfectly once I'd optimised the audio level going into my audio interface. I used the default keygroup spacings with two velocity layers, and a few minutes later I had a playable EXS24 Instrument, though being pads, the patches needed looping, and I had to do this manually. I was well pleased with the result, though, as it sounded very faithful to the original and also allowed me to use EXS24's filters to shape it further. Multi-dimensional sampling is a little more involved, as there are two controllers that can be set prior to sampling. MDS Controller 1 value is in effect latched by each Note On event so that only its initial value is read. MDS Controller 2's value is continuously checked, and on playback, changing this controller will fade between the various sample layers you created to give an approximation of what would happen when that controller was moved on the real synth. Of course, the more layers you sample, the smoother and more accurate the transitions will be. During sampling when MDS is enabled, Autosampler samples all the userdesignated combinations of patch, note, velocity, Controller 1 and Controller 2 until the process is done. Of course if you get too carried away with this feature, you can generate a huge number of samples per instrument, and EXS24 only supports 64 layers maximum, so you need to set the step sizes sensibly, but being realistic, you'd probably use only a fraction of what is available in normal use. The controller settings are saved in a Scan Parameter set, and Autosampler can store one global scan parameter set plus 128 individual-patch parameter sets, so you can either use the global set for all patches, or use the individual ones.

Sampling For Fun And Profit Autosampler is an extraordinarily useful add-on for EXS24 and it takes all the tedium out of sampling single or combination hardware instruments. Its more creative functions demand a little more of the user, but are still infinitely easier



than trying to achieve the same thing by other means. The only spade-work the user has to do is to loop any pad sounds that need it — sounds with a natural decay are taken care of automatically. For most routine work, you can rely on the default setting, which samples for seven seconds every six semitones and provides two velocity levels, but it's easy to change these values if you need to. The more advanced applications stray into sound-design territory, but then why not? Often you can combine synth sounds that result in something greater than the sum of the parts. In the short time I've spend with this program I've come to like it very much, and if you're in the position of phasing out your hardware synths to go completely soft, then this is a great way to assimilate you favourite synth patches so that you can hang onto them. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Sonic Implants SBC

In this article:

Sonic Implants SBC

Library Overview 3-DVD Gigastudio-format Orchestral Brass Library Triple Whammies Published in SOS February 2005 Box Of Tricks Installation Print article : Close window Opening Fanfare Reviews : Sound/Song Library Symphonic Brass Collection: Instrumentation Dem Bones Release Triggers Sound The Horns Sonic Implants extend their New World Symphony A Tale of Two Tubas with an orchestral brass sample library. A Whiff Of Corruption? Programmed Ensembles Dave Stewart Conclusions

Sonic Implants Symphonic Brass Collection

Sonic Implants' 12GB Symphonic Strings Collection, which was reviewed in SOS in November 2002 (see www. pros soundonsound.com/sos/nov02/articles/ Top-quality brass samples sonicimplants.asp), established the performed by leading American East Coast outfit as a orchestral players. purveyor of high-class orchestral Contains ensembles, duos samples, and remains one of the best and solo instruments. string libraries in the business. Well recorded in a pleasantly reverberant, Encouraged by the success of their natural-sounding concert-hall- strings collection, the company The library's classy packaging resembles an style recording space. old 12-inch classical music vinyl box set. commissioned the Symphonic Brass Offers a good range of Collection (or SBC), a 10.9GB opus in performance styles and the same mould. To maintain the excellent sonic quality of the strings and ensure programming options. musical compatibility between the two libraries, Sonic retraced their steps and Sonically and musically hired the same engineers, the same hall (Sonic Temple Studio in Massachusetts) compatible with the excellent Symphonic Strings Collection. and players from the same orchestra to record their new brass project. cons No solo tenor trombone or The studio's splendid appearance (seen on the documentation accompanying piccolo trumpet. the library, reproduced opposite) is matched by its pleasant, natural sound. With No grace notes, and the a 60-foot wooden floor, plaster walls and a high arched ceiling, the space creates trills are played by trumpets a lively hall ambience which imparts acoustic depth and excitement to only. instruments without smothering them in reverb. Under the direction of producer Apart from the trombones, and Sonic Implants co-founder Jennifer Hruska (a former head of sound design the performance styles are at Kurzweil), award-winning engineers John Bono and Antonio Oliart recorded not fully implemented across players from the Boston Pops orchestra, using the same front pair of B&K 4011 the different sections. mics used on the string sessions back in 2001. The review copy has a few tuning issues.

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Sonic Implants SBC

summary Very creditable, solid and professional, this orchestral brass collection offers some excellent samples, and is well organised, practical and musically versatile. Considering the scale of the project, it's also fairly priced, and would be a good investment for musicians, composers, studios, and anyone involved in the production of orchestral music.

information US$995 excluding shipping (about £515 at time of going to press). Sonic Implants +1 617 718 0202. +1 617 718 0227. Click here to email www.sonicimplants.com

Library Overview Unlike some recent orchestral libraries, SBC doesn't offer multiple mic placements resulting in a choice of listening perspectives — however, the extensive use of optional release samples (see the box towards the end of this article) does give users some control over the amount of 'hall' in the sound. The B&K mics were used as a front stereo pair throughout, supplemented by a variety of 'spot' (ie. close) microphones, for example Cole ribbon mics on the trumpets, Lawson tube mics on the trombones, and a Neumann TLM170 on the tuba (some of these mics can be seen in the picture at the end of this article). The samples were recorded at 24-bit/48kHz into Pro Tools. Following a convention established by Miroslav Vitous in 1992, the library provides both ensembles and solo instruments. The inclusion of multiple section sizes gives flexibility for orchestration, but it should be noted that some of the larger sections have been programmed; the 'six French horns' programs, for example, don't feature recordings of six horns playing together, but were created by layering a horn duo over four horns. You can see SBC's complete instrumentation and actual section sizes listed in the box over the page. Stylistically, the library concentrates on straight multisamples, with no played crescendos or diminuendos, grace notes, chords, licks, phrases, runs, impressionistic effects or overt sonic processing. As with the string collection, the delivery and recording style is naturalistic. The programmed ensembles (layered sections of different instruments) are a handy starting point for anyone looking for a general-purpose orchestral brass sound, while the individual ensembles and solo instruments offer a large range of detailed musical alternatives. On a technical note, congratulations to the US company for looping all the sustained notes in the library, a gargantuan task which they also undertook for their strings. It must have taken an eternity, but the peace of mind that comes from knowing that a sustained chord will go on sounding until you lift your fingers more than justifies the time and effort!

Triple Whammies In sampling sessions, it's good practice for musicians to play each note a few times before moving on to the next pitch. The repetition is reassuring for the player and the sound engineer, and provides an opportunity for a bad sample to be immediately corrected. Good musicians tend to produce repeated notes of a consistently high standard, so rather than agonising over whether the first, second or third take constitutes the definitive sample, producers have started to include all the alternatives and leave it to users to pick their favourites! This appears to be the case with the Symphonic Brass Collection — throughout the library, the staccato and double-tongue 'ta' and 'ka' performances are presented in a choice of three takes, thus avoiding the 'same sample every time' syndrome, and also creating the opportunity for double- and triple-tracking.


Sonic Implants SBC

Box Of Tricks SBC is manufactured on three DVDs and packaged in a capacious, 12-inch cardboard box. This gives the product the upmarket feel of a 1960s classical LP box set, but also places a strain on music room shelf space! Two booklets are included, one a general overview of the library and its playing styles, the other supplying a complete program list and some useful Gigastudio programming tips (the latter throwing much-needed light on the impenetrable mysteries of the program's instrument editor). Both booklets are highly informative, although baffling references to 'First Violins' in the second indicate that its text (obviously derived from the string library's documentation) has not been properly proofread... As this review was going to press in late 2004, the library was available in Gigastudio format only. Sonic Implants have recently announced that they are planning a Gigastudio 3 compatible version of the library, which has an expected shipping date of January 2005, so it may be coming out just as you read this. However, this review was conducted using the 16-bit version under Gigastudio 160. The Sonic Implants web site generously states that anyone who buys the 16bit set after November 15th is eligible for a free upgrade to the 24-bit Gigastudio 3 version. Finally, EXS24 and Kontakt versions are also apparently under consideration, but these are unlikely to appear for quite some time.

Installation The Giga samples are saved in eight big, compressed files, conveniently grouped by instrument type so you don't have to install everything in one sitting; if you suddenly feel an overpowering urge to hear French horns, you can load them separately and relatively quickly. Installation is handled by a Wizard which prompts you for a serial number, postal address, your inside leg measurement, and so on. I'd strongly advise against installing a big library like SBC in a time-sensitive situation such as a recording session; the process invariably takes longer than you think (in this case, over an hour), and it's easy to lose track of which parts you've installed (especially since the uncompressed files have bland, nonmusical names like Trombones_A.exe). A further consideration is that there are way too many cryptically abbreviated performance options to get your head round immediately. Do yourself a favour — set aside some quality time, install the instruments carefully and methodically, and use the loading time constructively by reading the booklets! Once the installation was complete, I found myself looking at five folders labelled Trumpets, Trombones, French horns, Tubas and Ensembles, each containing hundreds of programs. With so many files vying for screen space, it was hard to


Sonic Implants SBC

achieve any kind of overview, so I created individual folders for each section type ('Solo Trumpet', 'Two Trumpets', 'Three Trumpets', and so on), and sorted the instruments into their respective categories. The end result — 12 folders dedicated to individual sections and solo instruments, plus a 13th for my personal setups — was a lot easier to manage.

Opening Fanfare On to the samples, starting with the ensembles called 'three trumpets'. Their four main styles (legato and marcato sustains, staccato and double-tongue short notes) are not actually recordings of three trumpets playing together, but threeplayer 'virtual ensembles' constructed by layering two-trumpet samples with the library's solo trumpet. That said, these performances sound very strong and convincing, and the layering in no way distracts from their musical power and realism. The legatos and marcatos sound bright, confident and imposing, the latter given extra weight by a heavier, emphatic attack which subsides into a quieter sustain. Both styles have four dynamic layers, and sound nicely in tune throughout the trumpets' three-octave range (E3 to E6). Layering the legatos and marcatos creates a majestic fanfare sound, fit to greet the arrival of the Queen of Sheba. The three layered trumpets' tight staccatos also comprise four dynamic levels, and their timbral nuances and precise, energetic delivery will enhance any arrangement. The staccatos' abrupt cut-off enables Sonic Temple's hall ambience to be heard clearly, its bright, natural acoustic perfectly suited to the tone of brass instruments. The same goes for the 'double-tongue' performances — a recurring feature of the library, these are very short notes with a clearly articulated attack. The sound (designated 'ta', 'ka' or 'spit') is determined by the player's tongue position. Users can play these short attacks on their own (the 'spit' variety being particularly effective), or use them as an overlay to add definition to the front of legato notes. The remainder of the three trumpets' performance styles are performed by three unison trumpets. Their bright, raspy sforzando samples come in three flavours: 'sfz hit only' (a shortish stab), 'sfz soft hold' (the same layered over a quiet, threesecond sustain), and a 'sfz mod wheel crescendo' option where the sforzando hit is combined with a long sustain, the level of which is controlled by the wheel. This programming trick enables the simulation of the dramatic 'sfz crescendo' effect beloved of brass arrangers — but sadly, the layering introduces some


Sonic Implants SBC

minor tuning issues. Other trumpet trio styles include semitone and tone trills (the latter rather sloppily played) and some well-executed flutter tongue sustains. While I always enjoy hearing this somewhat specialised delivery, it's arguable that the time taken to record it could be better spent covering more mainstream areas. The addition of straight mutes renders the three trumpets' broad, handsome tone thinner and more pointed (though still evocative), reminding me of the brass on 1930s recordings. 'With mutes' styles comprise legato, staccato, flutter tongue and some whole-heartedly played big-band style 'falls' (one-second hits which end in a falling pitch). Jazz and pop arrangers should enjoy those. The two components of the programmed trumpet ensembles are also presented separately, playing legato, marcato, staccato and double-tongue deliveries. Heard alone, the two-trumpet section lacks the commanding tone of the programmed trio, but that's OK; in orchestral music, not every sound needs to leap out and grab the listener's attention, and these two-player unisons are quite capable of handling supporting lines, filling in backgrounds, and generally doing the more subtle things in an arrangement. The solo trumpet sounds very strong, and it's easy to see why the producers chose to incorporate it in their three trumpets programs — all its performances are executed with panache, giving the ensembles a lot of their strength, character and brilliance. As a bonus, the lone trumpet player also whips out some fine 'solo melodic' samples, featuring four-dynamic legato sustains played with a very subtle vibrato. This is far preferable to the over-excited, braying wobble of a trumpet in Rio carnival mode — in an orchestral context, such overstated vibrato would stick out like a sore thumb.

Dem Bones One problem with sampling brass instruments is that their tone can change dramatically with volume; quiet notes sound soft and enclosed, while loud ones are ear-splittingly bright and open-sounding. This can lead to disconcertingly obvious changes in timbre when moving between dynamic layers, but SBC have managed to side-step this problem with their two-trombone ensemble; their tone progresses quite smoothly from piano to forte, and even when the loudest fortissimo layer kicks in, there is no obvious tonal discontinuity. Sensibly, the producers reserved the brightest, most rasping trombone timbre for the sforzando performances. I found that these 'sfz' samples were very useful for adding a strong attack to file:///H|/SOS%2002-05/Sonic%20Implants%20SBC.htm (5 of 11)9/22/2005 8:19:12 AM

Symphonic Brass Collection: Instrumentation TROMBONES — 3.54GB Solo bass trombone — 808MB. Two tenor trombones — 980MB. Three trombones* (two tenor and one bass) — 1750MB.

TUBAS — 0.69GB Solo tuba in 'C' — 583MB. Solo tuba in 'E' flat — 103MB.

BRASS ENSEMBLE* — 1.24GB

Sonic Implants SBC

the two trombones' legato and marcato sustains, resulting in a powerful, brilliantsounding trombone ensemble ideally suited to underscoring triumphal Hollywood moments!

Trumpets, tenor and bass trombones, French horns and tuba — 1240MB.

TRUMPETS — 2.77GB Solo trumpet (709MB).

As well as trotting out the library's trademark staccatos and double tongues, the two trombones do some entertaining performance stuff: slowish, up and down tritone glissandi (perfect for salacious intros or on-screen pratfalls), lively, robust falls and rips (the latter's pitch rising a tone or so), and flutter-tongue samples which might fool you into thinking your speakers have blown. Nostalgia reigns when the trombonists fix their mutes and squeeze out that attenuated, 1940s on-the-radio tone, evoking the golden era when Stanley Matthews was the forces' sweetheart and Vera Lynn played on the wing for England [surely some mistake? — Ed]. But when the muted 'bones whip out their rips and falls (as we say in the trade), the mood changes — though no doubt played in all seriousness, these are great samples for comedy music.

Two trumpets (506MB). Three trumpets (507MB). Three trumpets* (two trumpets and a solo trumpet) — 1043MB.

FRENCH HORNS — 2.72GB Solo horn — 260MB. Two horns — 1110MB. Four horns — 1080MB. Six horns* (four horns and two horns) — 268MB.

Note: '*' = Programmed ensemble.

Strangely, no solo tenor trombone is included, which slightly impairs the library's musical flexibility. However, there is a solo bass trombone (incorrectly named 'baritone trombone' in the documentation). Sadly, this fruity, melodic-sounding instrument suffers from the obvious jumps in timbre mentioned earlier, especially when moving to its loudest dynamic in the low register — having said that, every sampled bass trombone I've heard does the same! The instrument's bottom note is G1, but Sonic Implants have promised a 'pedal tones' program descending to a shudderingly low B flat 0 in a future upgrade. The bass trombone plays the same set of performance variations as the two tenor trombones, sounding particularly energised in its glissandi and muted rips. Having captured a matching range of performances from a pair of unison tenor trombones and a solo bass trombone, Sonic Implants layered the two sets of samples to produce 'virtual' three-trombone sections. In these, you'll only hear three trombones playing in the C2 to E4 range; high notes from F4 up to E flat 5 are played by the tenors only, while the bottom few notes are handled by the bass trombone. Within their range of over two octaves, the three trombones' marcato sustains sound glorious, but the double-tongue performances suffer from bad tuning — the two trombones' 'ta' and 'ka' samples have been tweaked well sharp of concert pitch, and sound pretty sour when combined with the bass trombone's in-tune versions. Hopefully, this will be fixed in a forthcoming articulation file update, or the Gigastudio 3 version of the library. The 'three trombones' programs duplicate all the performance variations played


Sonic Implants SBC

by the two trombones and bass trombone, making the trombones' performances the only ones to be fully implemented across the different sections.

Release Triggers Like many contemporary sound libraries, the Symphonic Brass Collection offers users the option of programs with built-in release samples. In case you're not familiar with the term, 'release samples' are basically samples of reverb trails. When recording in an ambient room or concert hall, the reverb (ie. the room ambience) can be heard dying away when a note stops sounding; during the editing stage, the initial note is trimmed off, and its reverb is turned into a separate sample. Sample players like Gigastudio and Kontakt permit these auxiliary, reverbonly samples to be triggered at the point when a key is released, thus coining the name 'release triggers'. As you might expect, the release samples are usually appended to the note from which they were derived, thus ensuring continuity of sound and avoiding tuning mismatches! The big advantage of using release samples is that they cause a real-life room ambience to occur when a sample stops sounding, irrespective of note length. This is particularly useful for looped sustains, as without the help of a release trigger, a looped sample (which, being looped, never reaches its natural end point) can sound unnaturally dry and abrupt on note-off. Where memory conservation is an issue, 'no release' programs use fewer samples and so consume less RAM, but once you've heard release samples in action, no-release versions can sound flat and uninspiring by comparison. However, users who want to add their own reverb may prefer their dryer, more neutral effect.

Sound The Horns Buyers of SBC should get a lot of mileage from the French horn ensembles — they're very classy, and some performances sound quite magnificent. The four horns' five-layer legato sustains are among the most playable and versatile samples in the library — their lower four dynamics range from a tender piano up to a strong (but still fairly mellow) fortissimo, with smooth, practically undetectable timbral transitions. When the fifth, fff layer takes over, the horns' timbre flares into that unmistakable, thrilling orchestral brass sound which combines breadth of tone with a fierce cutting edge. In cinematic terms, this is the kind of exultant din required for the scene when the ship's lookout sights land, the chariot enters the gladiatorial arena, or the herd of dinosaurs trample the bad guys. According to the documentation, the horns' fabulous fifth layer is created by a delivery called 'bells up' (I've never come across this term before, though I do have considerable experience of the performance style we English call the 'balls up'). As the booklet explains, the French horn's bell is held 'up in the air facing the audience instead of against the body facing away'. Fair enough — it certainly makes a great, exciting racket, far preferable to the noise produced when the bell file:///H|/SOS%2002-05/Sonic%20Implants%20SBC.htm (7 of 11)9/22/2005 8:19:12 AM

Sonic Implants SBC

is thrust into a vat of lentil soup. For anyone who wants to cut straight to the chase, there are 'bells up only' programs, stirring marcato sustains and sforzandos with powerful, dominant attacks. After some time playing the velocityswitched sustains, I found myself wishing I could program a realisticsounding crescendo. This proved easy — the horns' long notes include 'mod wheel crossfade' programs which use the wheel to move smoothly between the dynamic layers. Hearing the horns' tone grow steadily brighter as the volume increases is very realistic, and feels very much like turning a knob to open or close a synth's filter. The great advantage of this method over real-life crescendo and diminuendo performances is that users can control the timing of the swell!

This photo was taken during the sessions for the library at Sonic Temple Studio, while the musicians were on a break, and shows the variety of mics and the acoustic baffles set up in the beautiful old hall.

The four horns distinguish themselves further with some fine glissandi, the twooctave ascending variety giving a satisfying, rowdy whoop and the descending type sounding more like a jazz/pop fall. Played with mutes, the horns lose their warm, mellow quality and sound more martial and trumpet-like, while handstopped performances utterly transform the horns' timbre into that intense, buzzy sound traditionally used in arrangements as a musical wake-up call. Like a tribute band lacking a couple of members, a duo of two French horns copy everything the four horns do, with the exception of glissandi. The duo performs creditable (if less spectacular) versions of each performance style, their more subtle sound suitable to supporting roles and/or more intimate music. By combining this horn duo with the four horns, the library's programmers created a virtual six-horn ensemble offering staccatos, sforzandos and 'bells up' legatos, each sounding very strong. Almost as an afterthought, a solo French horn makes an appearance, offering only two deliveries: restrained, subtle legatos and perky staccatos, both essential for arrangements. If the solo horn's contribution appears slender, it's worth remembering that its 260MB of sample data would have filled over half of an Akai-format CD-ROM!

A Tale of Two Tubas The tuba often plays the bass note in brass chords, performing the vital role given to the bass guitar in rock music. As owners of car stereos across South London will testify, bass notes need a strong sound, good attack and a steady pitch, and that's exactly what this tuba player brings to the table. The large contraption's four-dynamic legatos sound satisfyingly big and rounded, their file:///H|/SOS%2002-05/Sonic%20Implants%20SBC.htm (8 of 11)9/22/2005 8:19:12 AM

Sonic Implants SBC

stately tones given a boost by Sonic Temple's felicitous acoustics. As with the horns and tenor trombones, moving up through the dynamic layers causes no crude jumps in timbre, merely (in this case) a slight change in the intensity of the attack. As well as being a good source of strong, stable root notes, SBC's tuba sounds convincing and eloquent playing melody lines throughout its threeoctave, C1 to E4 range. The smart, emphatic delivery of the tuba's staccatos mean that you can program passages of short fast notes with no fear of the samples sounding indistinct. For a heavier, more brassy delivery, there are one-dynamic marcato sustains, which could be used to add a fifth, 'fff' layer to the legatos. However, the most impressive samples are the tasty 'sfz' performances, which show this tuba at its most assertive, stately and sonorous. In its small but enjoyable effects section, the tuba's upward 'rips' put me in mind of a surprised elephant, and the mad 'extended rips' contain a couple of samples which seem to end with the player suppressing a giggle. I had the same reaction. The aforementioned samples were all played on a 'C' tuba, but for the muted performance styles, the player switched to a smaller, E-flat instrument. Once again, the mute radically alters the instrument's tone, giving the marcatos a real cutting edge, and transforming the sforzando low notes into a collection of magnificent, flatulent trumps. Overall, the tuba performances are delivered in some style, and their weighty, musically poised samples will add depth and stability to orchestral brass arrangements.

A Whiff Of Corruption? In the Gigastudio review copy of the Symphonic Brass Collection, a fault in the GIG file '4 Horns Mute Staccato' on the first DVD of the set consistently caused my copy of Gigastudio 160 to crash, the host PC to freeze and the reviewer to utter a string of unprintable curses. Sonic Implants have promised to send an update disk directly to any users who should experience the same problem.

Programmed Ensembles Orchestral libraries very rarely feature mixed ensembles (violas and cellos playing at the same time, unison trombones and French horns, and so on), and SBC's ensembles are no exception — they were created by layering and mapping the library's individual instruments and sections. Not true multiinstrument brass ensembles, then — but close your eyes and you'll never know you're not hearing the real thing. Mapping the orchestra's brass instruments across over five octaves is not a straightforward job. Sonic Implants' programmers approached the task scientifically, dividing the wide compass into six zones and allocating the instruments thus: Tuba (C1-A1), tuba and bass trombone (A#1-D#2), bass file:///H|/SOS%2002-05/Sonic%20Implants%20SBC.htm (9 of 11)9/22/2005 8:19:12 AM

Sonic Implants SBC

trombone and two tenor trombones (E2-A#2), two tenor trombones and four French horns (B2-A#3), four French horns and two trumpets (B3-F4), two trumpets and a solo trumpet (F#4-E6). This formula is repeated more or less identically in six performance styles (legato, marcato, staccato, sforzando, muted legato and muted staccato). Though realistic, fairly powerful and musically logical, the ensembles' overall effect is not as spectacular as I would have hoped, and there are some discontinuities of tone at the low end (this is especially noticeable in the sforzandos, where the 'trombones only' zone sounds thin and unsupported). Layering the largest ensembles (three trumpets, three trombones and six horns) might yield more exciting results (albeit within a more limited pitch range), but this is where users need to get their hands dirty and experiment — Sonic Implants have provided the individual ensembles, and all you have to do is combine them!

Conclusions SBC's producers put a lot of effort into providing programs which facilitate dynamic expression, while managing to avoid swamping users with too many options. Their programming scheme focuses on practical choices — as described earlier, there are 'mod wheel crossfade' and 'mod wheel swell' options for the creation of realistic crescendos and diminuendos, programs which use keyswitching or the mod wheel to quickly switch between two dynamic layers, and also programs containing switchable alternate takes. For musical situations requiring a more limited dynamic response, users can select a program containing fewer dynamic layers, and there are also single-dynamic programs for passages marked piano, or forte. Implemented throughout the library, such features make it easier to play the samples in a musically expressive way. Since entering the sampling world in 1986, Sonic Implants have produced an interesting catalogue of titles comprising ethnic drums and percussion, Middle Eastern instruments, drum kits, multisampled guitars, basses and organs, and more. Orchestral libraries are a relatively new departure for the company, but on the strength of this collection and the excellent string library that preceded it, you'd have to say that it's something they do well. The playing, recording and programming are all of a high standard, resulting in a set of samples which sound polished, powerful and assured. I have only one gripe — the title. Is it just me, or are the names of orchestral sample libraries beginning to converge? It's all Symphonic this, Orchestral that or Classical the other, to the point where it's getting hard to tell them apart. Can someone please show some initiative and call their library Phineas Foghorn's Brazen Brass Blasters or Musicians In Dickie Bows Making A Big Pompous Noise? Joking apart, the title is the only dull thing about Sonic Implants' latest orchestral offering — scratch the surface of their brass collection, and you'll find gold shining through.


Sonic Implants SBC




Steinberg Wavelab 5

In this article:

Steinberg Wavelab 5

Multi-channel Montage Windows Audio Editor Madness Published in SOS February 2005 Video Montage 1, 2, 3, DVD-Audio Print article : Close window Wavelab Through The Ages Reviews : Software It Takes A Montage... Getting Into Details Conclusion

information Steinberg Wavelab 5 £500 pros Now an affordable and comprehensive DVD-Audio authoring package. The new multi-channel support in Montage window, Master Section and real-time analysis windows is particularly welcome. The Video Track offers some great features for those who want to cut against picture in the Montage window.

cons You can't open and edit multi-channel audio files in a sample editor window in the same way you can with mono or stereo files. No support for encoding audio using MLP on DVDAudio discs.

summary In addition to being an elegant, feature-rich audioediting and CD authoring solution, Wavelab is now a comprehensive tool for DVDAudio authoring, with added support for video and multichannel work. £499.99; upgrade from version 4 £129.99; upgrade from v2/3 £199.99. Prices include VAT.

& DVD-Audio Authoring Package

The latest version of Steinberg's popular editing program now includes support for multi-channel audio, including the ability to create your own DVDAudio discs. Mark Wherry

2004 marked the eighth year since Steinberg introduced their Wavelab audio editing software. Over this time the application has attracted a loyal band of users, so much so that even the most hardcore audio-oriented Mac user will justify the purchase of a full Windows system just to get their hands on Wavelab. And indeed, after all this time Wavelab remains a Windows-only application, having been one of the very first audio applications to be written from the ground up for Windows 95 (with NT compatibility).

It's now possible to edit your Audio Montages against video in Wavelab thanks to the video track feature that is new in version 5. Here you can see two Video Clips from the same file (the first Clip ends where you can see the narrower thumbnail) — notice how the outer video players above the Montage show the start and end points of the selection, while the central player shows the frame at the cursor position.

As most users are aware, Wavelab is mainly the work of one programmer, Phillipe Goutier, and each new release has tended to focus on one particular area, providing a comprehensive, carefully thought-out and well implemented solution. After the original release, for example, 1.5 added the Master Section with real-time effects, 1.6 offered CD authoring, 2.0 was aimed at those wanting to use Wavelab in conjunction with hardware samplers, with associated looping and analysis tools, and 3.0 added the Audio Montage window.

file:///H|/SOS%2002-05/Steinberg%20Wavelab%205.htm (1 of 10)9/22/2005 8:20:38 AM

Steinberg Wavelab 5

Arbiter Music Technology +44 (0)20 8970 1909. +44 (0)20 8202 7076. Click here to email www.arbitermt.co.uk www.steinberg.net

The last release, 4.0, broke from this tradition by adding a slew of useful facilities such as real-time analysis, enhanced plug-ins for audio restoration, the OSQ (Original Sound Quality) lossless compression format, backup features and better CD project management. However, in version 5.0 Goutier has again focused on one particular area, this time handling multi-channel audio — and, in particular, providing a full suite of features for authoring DVD-Audio discs (as well as standard data DVDs) without the need for any extra third-party software.

Test Spec Wavelab 5.01a (build 231). AMD dual-246 Opteron system with Tyan 2885 motherboard, 4GB RAM, Nvidia Quadro4 380 XGL graphics, Pioneer DVR-106D DVD-RW drive and RME Hammerfall DSP 9652 card, running Windows XP Professional with Service Pack 2. IBM Thinkpad X40 with 1GB RAM and Echo Indigo I/O card, running Windows XP Professional with Service Pack 1. Pioneer DV45A DVD player.

Multi-channel Montage Madness Although Wavelab's terribly handy Montage window has always allowed multiple tracks, these have always been submixed to stereo for output, usually through the Master Section. In version 5, Wavelab has an eight-channel that mode that enables you to output up to eight channels of audio from the Montage window, and work with an eight-channel Master Section with support for multi-channel VST plug-ins. A nice touch is that eight-channel mode also extends to Wavelab's real-time analysis tools, which means each of the eight channels gets its own dedicated display. Although you could potentially use Wavelab v5's eight-channel mode with the Montage window to make Wavelab an eight-track recorder, the real point is to allow support for surround formats. This means it's now possible to work with 5.1 audio in the Montage window (or any surround format up to a maximum of eight channels — I don't think not being able to work with 10.2 is a big disadvantage here!), which is what facilitates the new DVD-authoring features that we'll look at in the next section. In order to use Wavelab's eightchannel mode, you'll need an audio card with at least six outputs (for 5.1) that offers ASIO drivers, and this is configured in the Audio Card page of the Preferences window. A new ASIO Connections window allows you to set how Wavelab routes its surround channels to the physical outputs on your audio hardware, and its eight input channels to the physical inputs on your audio hardware for multiThe Surround Panner window allows you to channel recording. Now, in the pan mono or stereo audio tracks to the Montage window, you can switch to a surround output. multi-channel configuration via the Mode menu in the Edit page, and here you can choose between CD and DVD-Audio options and various surround formats, plus a generic eight-channel mode where Wavelab simply uses eightchannel labelling without any specific surround channel naming assignments.


Steinberg Wavelab 5

As with Wavelab's main sample editing window, the Montage window still works in terms of mono and stereo tracks. This means that if you're going to work with a 5.1 mix, it needs to be split into a minimum set of three stereo files: LF and RF, LS and RS, C and LFE, (where 'F' represents the front channels and 'S' the surround), although it tends to be better practice to split the centre and emitter channels as two mono files. Obviously you can drag split-mono files into the Montage window, but Wavelab also offers an Import Surround Audio File command so you can take an interleaved six-channel file (such as one you might export from a program like Cubase) and have Wavelab automatically figure out how to bring it into the Montage window. Once you're in a multi-channel mode, you route a track to a set of outputs by clicking on its output label, just below its numeric identifier and pop-up menu button, to specify the outputs via the Audio Dispatching window. For imported surround files spread across multiple tracks, you can assign the relevant outputs to each track so the front left and right parts of the audio are routed to the front left and right channels. Alternatively, if you want to take a mono or stereo track and output and pan it to the surround channels, if for instance you're doing an upmix, you can assign a track to all available outputs in the Audio Dispatching window, although an 'assign to all available outputs' command in that window would also be a nice touch to save a few mouse clicks. When you route a track to all available surround outputs, Wavelab displays a surround panner in the track, which you can right-click to edit more precisely. It's also possible to create surround pan envelopes for individual clips. As mentioned earlier, the Master Section offers control for your multi-channel outputs, although it's worth noting that it won't switch to this mode until you actually play back in eight-channel mode for the first time. Multi-channel plug-ins can be used in the Master Section, although a plug-in must be specifically designed for multi-channel support in order for Wavelab to allow it to be used on the Master Section. Many of Wavelab's are already supplied with multi-channel compatibility, such as Leveler, Puncher and EQ1, and the application is compatible with Steinberg's Surround Edition bundle of multi-channel plug-ins. Still, it would be nice if Wavelab enabled multiple stereo plug-ins to be used across different sets of output channels as in Nuendo, since the current behaviour really limits the usefulness of the Effects part of the Master Section.


Steinberg Wavelab 5

Video Montage One particularly neat feature added to the Audio Montage window in version 5 is the ability to add a video track so you can edit your audio alongside video. With Wavelab 5's support for multi-channel audio in the Montage window, the added video functionality becomes quite important since you might now use Wavelab to master your final audio tracks for DVD-Video, in addition to using it to burn DVDAudio discs. Once a video track has been added to a Montage, it works pretty much like an audio track: simple right-click in the lane and choose Insert File from the pop-up menu to add a video file to that track. The file selector that appears to choose your video files lists *.avi, *.qt, *.mov, *.mpg, *.m1v, *.wmv, *.vob, and *.asf as suitable formats, but I wasn't able to open Quicktime movies. However, MPEG and all Windows-related formats (AVI, WMV and ASF) opened fine. After you've chosen a file, a Video Clip is added to the video track. This shows a thumbnail overview of the video file, and a new Video tab in the Montage window provides an on-screen video display, which can be enlarged to fill the whole window by toggling Shift+A. A nice touch, though not original to Wavelab, is that the video thumbnails can actually play back the video while the playback cursor moves over a given thumbnail. This is useful when the Video tab isn't in view, and the behaviour can be toggled from the Options menu in the Video tab of the Montage window. Another nice touch is that the Video tab actually displays a total of three video players. When you make a selection, you'll notice two additional video displays open either side of the main player (which remains in the centre) to show the frames at the start and end of the selection you've made. The appropriate start or end display also updates in real time as you make your selection (or if you adjust a selection later), which makes cutting against video a piece of cake. You can work with Video Clips in almost the same way you can work with Audio Clips, in so much as you can cut them up and move the start and end points, as well as move a clip's position in the timeline. It's possible to have multiple video files and Clips on a video track, and it's even possible to have multiple video tracks, although without any mixing or crossfade facilities (that's what applications like Avid are for, after all!) — the lowest video track always has playback priority. Finally, perhaps as a show of Steinberg's closer association with Pinnacle since the last release of Wavelab, version 5 supports Xsend and Xreceive for exchanging audio files with video applications such as Pinnacle's Liqud Edition.

1, 2, 3, DVD-Audio As mentioned earlier, the highlight of Wavelab v5 is undoubtedly the new DVDAudio authoring features, which work in conjunction with the Audio Montage window and the new multi-channel features described in the previous section. Creating DVD-Audio discs is slightly more complicated than creating audio CDs, due to the richer feature set of the format, but existing Wavelab users familiar with the workflow will have no problem in creating simple DVD-Audio discs in a very short time, before diving into the array of options and configuration


Steinberg Wavelab 5

possibilities. Taking a 'bull in a china shop' mentality, I created my first DVDAudio disc in Wavelab in about 15 minutes before really figuring out how it worked properly — a fact that says more about Wavelab than it does me! However, before looking at the way Wavelab deals with the format, it's important to consider what exactly a DVD-Audio disc is in the first place. A typical Red Book CD contains a maximum of 99 stereo 16-bit/44.1kHz audio tracks (each of which can contain a further 99 indices): you pop it into your CD player, select the track you want and press Play — pretty simple. By contrast, the basic organisation of a DVD-Audio disc is a maximum of nine groups, each containing 99 tracks, with each track, as on an audio CD, containing a Both Wavelab's real-time analysis tools and maximum of 99 indices. While there Master Section offer support for multiare extensions to the Red Book channel functionality. standard for including graphics on an audio CD (CD-G and CD-EG, for example) and CD TEXT for extended text-based information, you'll probably know from your own experience that the use of these extensions is fairly rare: the former much more so than the latter. The use of extended pictorial and textual data is an intrinsic part of the DVDAudio specification, although not a prerequisite, along with graphical navigation menus for those who use a video display with their DVD-Audio player. DVDAudio discs can contain text information that appears as the audio plays, and also a series of pictures that you can define to appear at certain points, allowing you to create a slide show that plays along with the music. A DVD-Audio disc is divided into two basic folders, or title sets (TS), in terms of its organisation: VIDEO_TS and AUDIO_TS. The AUDIO_TS folder contains DVD-Audio-centric data that can only be played back on DVD-Audio players, while the VIDEO_TS folder contains video information that can be played back on DVD-Video players. In this way you can create a disc that contains information that can be played back on both types of players, and while you can't author video content in Wavelab, the application lets you add VIDEO_TS content created in another application when you come to write your DVD. In terms of audio data, a DVD-Audio disc can support multi-channel audio data (up to 5.1) at a resolution of either 16-, 20- or 24-bit, and sampling rates of 44.1, 48, 88.2, 96, 176.4 or 192 kHz. However, there is one giant caveat that has a big impact on these specifications, in that the maximum bandwidth for the audio stream on a DVD-Audio disc is 9.6Mbits/s. You probably don't need a calculator to figure out that six channels of audio data at 24-bit/192kHz requires more than 9.6Mbits/s, and indeed, the maximum amount of multi-channel audio you can fit on a DVD-Audio disc without compression is six channels of 48kHz at 24-bit (or


Steinberg Wavelab 5

six channels of 96kHz at 16-bit, but I'd take the extra bit depth over the sampling rate here), or, at a pinch, four channels of 96kHz at 24-bit.

Wavelab Through The Ages Wavelab 1.01 www.soundonsound.com/sos/ 1996_articles/aug96/steinbergwavelab.html

To overcome this limitation, the working group for DVD-Audio Wavelab 1.5 chose MLP (Meridian Lossless www.soundonsound.com/sos/ Packing) as a format for 1997_articles/feb97/steinbergwavelab.html compressing audio in a lossless Wavelab 1.6 way to maintain quality and basically get more audio data on www.soundonsound.com/sos/ a disc. In most cases MLP 1997_articles/oct97/steinbergwavelab.html achieves less than half of the bit Wavelab 2.0 rate required for uncompressed audio, which is important both for www.soundonsound.com/sos/ higher-quality surround mixes, jun98/articles/wavelab.html and for the total amount of data Wavelab 3.0 you can fit on a disc. A DVD is www.soundonsound.com/sos/ limited to 4.7GB per layer, and mar00/articles/wavelab.htm since not every drive either reads or writes dual-layer discs, this will Wavelab 4.0 be a real limit for most people. www.soundonsound.com/sos/ Using MLP, you can get two may02/articles/wavelab4.asp hours of two-channel 24-bit, 192kHz or a staggering 13 hours of 16-bit 44.1 audio data on a single-layer DVD.

There's just one problem: like other major encoding algorithms such as Dolby Digital and DTS, MLP costs a reasonable amount of money to license. These licence fees would push the cost of Wavelab into the thousands of pounds price range, and hence MLP has not been bundled as standard. This is understandable, but it is a shame Steinberg don't provide it as an optional plug-in for high-end customers in the same way the company have provided optional Dolby Digital and DTS encoders for Nuendo customers.

It Takes A Montage... To give you an idea of how simple it is to create a DVD-Audio disc in Wavelab 5, let's consider the process of creating the most basic DVD-Audio disc possible. A DVD-Audio Project in Wavelab consists of a collection of Montages, where each Montage translates onto a DVD-Audio disc as a group — a DVD-Audio Project can therefore contain up to nine Montages. To create a DVD-Audio group as a Montage, you simply create a Montage of the appropriate sample rate and set its Mode to DVD-Audio, via the Mode pop-up menu in the Edit page of the Montage window, described earlier, where you can also define a configuration for the


Steinberg Wavelab 5

Montage, in terms of the number of channels. All the audio in a Montage uses the same channel format, although each Montage (or group) on the DVD-A disc can be of a different sample rate and configuration. The next step is to import the actual audio into the Montage window and create DVD-Audio Track Start and End markers to define the start and end point of the Tracks within a group, just like when you're using Wavelab to create an audio CD. You can do this manually from the Markers page, or using the DVD Wizard option in the DVD, which enables you to go through and create a series of Markers based on the length of Clips in the Montage window. Once you have defined some tracks, you can check that you've created a Montage that can be a valid DVDAudio Group by using the Check command from the DVD-Audio page to make sure you haven't gone above the maximum 9.6Mbits/s of available bandwidth or the maximum capacity of a DVD-Audio disc. If Wavelab confirms everything is all right, you can now go ahead and save this Montage and add it to a DVD-Audio Project by either creating a Group from the current Montage (by clicking the Create DVDAudio Project command in the DVDAudio page) or starting a new DVDAudio project and dragging in your Montage manually.

Wavelab's new Menu Generation window allows you to customise how menu screens will look when you play back your DVDAudio disc on a system attached to a video screen.

A new window is opened for the DVD-Audio Project and you'll be asked whether you want the Montage to use the same Master Section settings as other Montages, a specific set of settings for that Montage, or to simply ignore the Master Section altogether. Even if you choose Ignore and Master Section is disabled, Wavelab tells you that you have to enable the Master Section to continue, which seems a little quirky, but this is a very minor point. Now you can use the Create command, which renders your DVD-Audio disc (make sure you have plenty of hard disk space for working with DVD-Audio Projects) and opens the disc burning window, similar to when creating CDs in Wavelab. Simply click Write DVD and Wavelab will go ahead and burn your DVDAudio Project onto a blank DVD — I guess it would be stating the obvious to say that Wavelab's minimum requirements now include a DVD writer for the DVDauthoring features. However, if you don't have a DVD writer on the system used to create the DVD-Audio Project, you can create an ISO image file that you could take to another system to write the DVD-Audio disc, which is pretty handy if you don't want to tie up your audio machine with DVD-writing duties. After burning the Project, I used a Pioneer DV45A DVD player (which is also file:///H|/SOS%2002-05/Steinberg%20Wavelab%205.htm (7 of 10)9/22/2005 8:20:38 AM

Steinberg Wavelab 5

capable of playing both DVD-Audio and DVD-Video discs, along with SACDs) to test the disc, and it played back pretty much as expected. Some of the styles for selecting tracks didn't always come out the way I intended, and I'm not quite sure who's to blame here, although I'm guessing the Pioneer player may have problems in handling certain graphical menu commands (more on this in the next section). Overall, while creating a simple DVD-Audio disc in Wavelab isn't quite as quick and easy as the Basic Audio CD creation, thanks to the added complexity of the DVD-Audio format, Goutier has done a commendable job of adding DVD-Audio authoring facilities into the Wavelab workflow, allowing to tremendous control and flexibility. In a future update, it might be good if there was a 'Create Basic DVDAudio' command that allows you to create a simple, one-group disc by dragging a few audio files into a window, just for speed, but this request definitely falls well beyond the 'icing on cake' territory.

Getting Into Details Once you've got to grips with the basic process of creating a DVD-Audio disc, you'll soon realise that Wavelab's feature set goes way beyond creating basic DVD-Audio discs, and actually includes full support for adjusting almost every possible setting on a DVD-Audio disc, including the ability to work with text and pictures. To start with, each Montage has a DVD-Audio settings page that allows you to define whether a given Montage should end up in either 20- or 24-bit resolution on the disc, along with how the DVD-Audio player should handle the stereo downmix if the Montage is multi-channel and ultimately played on a non-surround system. These down-mix settings can be previewed when you enable the Stereo option on the Master Section, which is pretty handy. Wavelab handles the picture and text elements of DVD-Audio discs by providing two extra track types in the Montage window: yes, you've guessed it — Picture and Text Tracks. Adding Picture or Text Clips to these tracks is just like adding an Audio Clip to an The ASIO Connections window allows you to audio track: right-click on the track's set which surround channels are output to lane and select either Insert Text or which physical outputs on your audio File depending on the type of Clip hardware. you're adding. If you select Insert Text, Wavelab will display a dialogue for you to configure a Text clip by typing in suitable text and defining how long it should be visible. If you select Insert File, Wavelab will prompt you to find a picture file, and double-clicking a picture file enables you to set various options, including various transition effects such as fades, wipes and dissolves, although Wavelab file:///H|/SOS%2002-05/Steinberg%20Wavelab%205.htm (8 of 10)9/22/2005 8:20:38 AM

Steinberg Wavelab 5

is keen to point out that not all DVD-Audio players actually support such transition effects. Once you've added all of your Montages into the DVD-Audio Project window, an additional step to polishing off your DVD-Audio disc is to customise the menus that appear on the video screen of a DVD player, allowing the user to navigate your disc. Wavelab actually prepares a basic menu by default, which is what your disc will contain if you don't make any other adjustments, but the application also contains quite a comprehensive Menu Generation tool that can be accessed from the DVD-Audio Project window. In the Menu Generator you can basically customise how the names for the Album, Groups and Tracks are displays, setting the fonts, colours, and so on; and since menus are stored on a DVD as bitmap information, you don't have to worry about font availability on DVD players. For the actual tracks, you can specify whether only the names are displayed, or whether Wavelab adds the length as well, and when it comes to set the colour, you can set the basic colour, the selection colour (to show the current selection in a list of tracks), and the action colour (for when the user actually chooses to play that track from the menu). Each group is displayed on its own menu page, and each menu page can have either its own static-colour background, or a unique picture. Wavelab comes with some fairly attractive and generic backgrounds, which definitely add a more professional touch than a solid colour, and once you come up with combinations of colours and pictures that you like, you can create presets for use with other DVD-Audio Projects in the future. Finally, the Menu Generation window also provides a simulation mode so you can try out your menu system with playback of the actual audio content. As you can see, Wavelab's support for almost every intricacy of the DVD-Audio standard is pretty much exemplary, and the only really significant feature not implemented is support for MLP.

Conclusion Wavelab 5 is a great update to an already great and well-respected audio editing package, although it's also fair to say that perhaps not every existing user will view this as a must-have update. Unless you want to be able to author your own DVD-Audio discs, work with multi-channel Montages, or add video to a Montage, there aren't really any other major features to warrant updating, unless some of the more minor improvements and the desire to have the latest version prove attractive. In terms of alternatives to the DVD-Audio authoring features, the main competition comes from Minnetonka's Discwelder series of applications, which


Steinberg Wavelab 5

are available in three metal versions: Bronze, Steel and Chrome. Bronze is a basic $99 solution for selecting a few PCM surround or stereo files and writing them to a DVD-Audio disc, while the original version of Steel was reviewed back in March 2003 (www.soundonsound.com/sos/mar03/articles/discwelder.asp). Chrome is the high-end $2995 solution, and provides a similar feature set to Wavelab for working with pictures, text, and so on, with a few extra advanced features such as DSD conversion and the ability to import content encoded in MLP. For encoding audio into MLP in the first place, Minnetonka also offer Surcode MLP, which retails for $2499 (explaining why Steinberg didn't integrate MLP into Wavelab!). As long-term Wavelab users know, Goutier is consistently improving Wavelab and version 4, for example, saw a constant stream of free minor updates, tweaks and fixes over its lifespan. This trend looks set to continue and the original Wavelab 5 release has already been superseded by the freely downloadable 5.01a update containing a fair list of small fixes, mostly concerning the newer DVD-Audio authoring features. Wavelab is possibly the best application of its kind on the Windows platform, and one of the most powerful all-in-one solutions in its price bracket on any platform. At this point, it's a pretty stable and mature application that remains a pleasure to use, and if you need the functionality offered in this new release, Wavelab 5 is unlikely to disappoint. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Submersible Music Drumcore

In this article:

Drum Heaven Creating & Editing Kits Journey To The 'Core Further Options Conclusions

Submersible Music Drumcore Groove Library/Audition Engine (Mac OS X) Published in SOS February 2005 Print article : Close window

Reviews : Software

Submersible Music Drumcore £169 pros Features the performances of real, world-class drummers. Ultra-fast and flexible audition interface. Innovative 'Gabrielise' randomiser function for when inspiration runs dry. Exports to all popular Mac audio sequencers. Rewire compatibility.

cons No separate outputs for individual drum sounds from loops. Audio loops are only available at certain tempos. No Mac OS 9 compatibility.

summary Combining the work of real 'name' drummers and a set of smartly chosen performanceoriented editing parameters, Drumcore is a unique drum library which allows you to produce rhythm tracks with real personality and flair.

information £169 including VAT. MI7 +46 40 630 6970. +46 40 699 2509. Click here to email www.mi7.com

Featuring the performances of some of the world's best drummers, Drumcore is a software-based cross between a drum library and a custom loop-auditioning interface, allowing you to assemble rhythm tracks fast. Paul Wiffen

It feels to me as though we're finally starting to see proper drum and percussion products for computers. Of course, it's been possible for a long time to buy drum sounds or loops for samplers (hardware or software) and trigger them straight from our sequencers, but over the past few years, there have been more and more software-based products specifically tailored to cater for rhythm, whether to tailor loops or simply build kits which respond like real drums do. Of these, I particularly like the look of NI's Battery, now out in a revamped version, Stylus RMX from Spectrasonics, reviewed by Paul White in last month's SOS, and FXpansion's BFD, which Paul tackles on page 226 this month. And now I have one more to add to the list, because the first time I heard about Submersible Music's Drumcore, I knew I had to have it.

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Loop selection can be carried out from a list ordered by Drummer, as shown here, or by Style. Depending on which tab is selected at the top of the window, the Styles or Drummers then appear in the left-hand box, while the loops that fit into the selected category, or which were performed by the selected drummer, appear in the box on the right. The lower panel contains the details of and variations for each loop (fills, individual instruments, and MIDI performances, for example).


Drum Heaven I recently set up a computer recording system based around an iMac and Apple's Logic for a friend who simply wanted to trawl through loads of drum samples till he found one he wanted to write or play over, and then get on with composing and performing. He found it very frustrating having to import loops as AIFFs or WAVs from CD-ROM into Logic and matching their tempos to hear if they would be suitable. He wanted to audition loops the way you can whizz through presets on a synth, then simply pick the one he wanted, try different tempos till it was right, and have Logic use this as the start point for his new song. His frustration lasted until I showed him Drumcore. It's a software instrument containing the performances of many very gifted drummers, presented as a series of audio loops in various styles, with each loop recorded at many different tempos. For further flexibility, individual drum hits recorded at the same sessions as the loops are also supplied, as are MIDI sequences derived from the performance loops — so if you wish, you can create a completely tempo-flexible version of the performances based on the supplied MIDI sequences and kit samples. In addition to its library of audio and MIDI loops, the program has excellent auditioning facilities. You can step through the available patterns very quickly, arranged either by artist or style of music, until you find something that piques your interest, vary the tempo with a big user-friendly control and then export the loop so that it opens in your sequencer and works with the tempo you've set. You can even customise Drumcore's excellent auditioning interface so that it works for any sample libraries you already own. It really takes the hard work out of auditioning and tailoring drum loops for use in your own music. My interest in Drumcore stemmed from having all these famous drummers at my disposal. Alan White (the drummer with Yes since the early '70s, and before that with John Lennon's Plastic Ono Band) is one of my favourite drummers of all time, and I was irresistibly drawn to the idea of being able to use his most famous rhythms to write and arrange my own material without falling foul of the publishers and lawyers for breach of copyright. Now, I wouldn't expect everyone else to feel the same way about Mr White's drumming, but I would be surprised if this collection does not contain at least one of everybody's favourite drummers. The other one I immediately checked out was Michael Shrieve (the drummer with Santana in their classic Abraxas phase); but that's my roots showing again. For those who like a different type of roots, there is Sly Dunbar (who played with Bob Marley and Black Uhuru, as well as being one half of reggae production team Sly and Robbie) and for those who like their black music on the rockier side, there is Zoro, whose credits include Lenny Kravitz. Plenty of other styles are represented here, from AOR (Ben Smith from Heart) and rock (Matt Sorum from Guns'n'Roses and Velvet Revolver) at one end of the spectrum, through Jeff Anthony (Sheryl Crow's drummer) and Tony Braunagel (who has played for both Bonnie Raitt and Taj Mahal), Ned Douglas (who has worked with both Dave Stewart and Simply Red) to DJ Size-up, whose grooves feature in Ultra Naté's stuff. If you can't find someone whose playing you admire in that lot, you must have very exclusive tastes indeed! And if you do like file:///H|/SOS%2002-05/Submersible%20Music%20Drumcore.htm (2 of 5)9/22/2005 8:20:45 AM


what you get on the main disc, Submersible are apparently making other Sly Dunbar, Zoro, and Matt Sorum add-on discs available in the near future.

Creating & Editing Kits The Drumkit Editor window (shown below) allows you to change the samples and their assignments in the MIDI kits. You simply call up the name of the kit you want to edit and then you can assign new samples or swap around the ones you already have. Editing is on a fairly basic level and certainly does not have the same level of complexity as NI's Battery or Spectrasonics' Stylus. Having said that, you can Drumcore's easy-to-use Kit Editor. add as many layers as you have samples at different velocities. Most of the 'famous name' kits seem to have three samples for most of the instruments, so there are a significant number of possible variations.

Journey To The 'Core At the time of writing (late November 2004), the single Drumcore install DVD only works on Mac OS X (minimum requirements a 400MHz G4 with 256MB of RAM and 10GB of disk space to install and audition the library from). However, Submersible recently announced version 1.5, which will work on the PC as well when it ships in the New Year. Any Mac users who bought Drumcore before the end of the year are automatically entitled to a free v1.5 upgrade when it is released. Whether you choose to browse by artist or style, you can instantly skip from one category to the next using the left and right arrow keys, or from one pattern to the next using the up and down arrow keys. In case you cannot hear the difference between the audio grooves and those made from kit samples triggered via MIDI, the two distinct categories are represented by a waveform and five-pin DIN icon respectively. Original tempos are shown beneath MIDI icons, whereas the audio loops default to the tempo of recording when you first open them. You can also open a Video Window to see Quicktime footage of your selected drummers in action and read their biographies. One of the important things to note is that when you change tempo in Drumcore, you are actually selecting a different audio loop recorded by that same drummer at the new tempo. This means that the looped audio performances are not available at every possible tempo, although you can of course edit them in your



sequencer after importing them if you need them to run at a slightly different speed. However, this does mean that the grooves you get have been properly played at all the different available tempos. This is important, because rather than just playing the same thing slower or faster like a drum machine, good drummers will play a pattern differently at different tempos, especially where there is a lot of 'swing' or 'feel'. This means that Drumcore will give you a more 'human' performance at the selected tempo — in fact, some of the patterns change quite drastically at different tempos. But then you try playing 16th-note triplets at 150 beats per minute! Of course, if this doesn't offer enough flexibility for you, then as mentioned earlier, each drummer's kit is also available as a MIDI-triggerable set of samples with velocity layering. Using the supplied MIDI files of the famous drummers' loops, you can set the tempo to whatever you like. The kit samples also allow you to add fills and other embellishments with instrument sounds which match with those in the audio loops, as the individual hits were recorded at the same sessions.

Further Options The other really cool thing about Drumcore is that you can use the audition engine to go through all your existing audio loops. You just drag and drop your files into Drumcore's Date/Content folder and then select 'Import Files' from the Import menu inside Drumcore. You can decide what criteria you use to sort the loops (for example, if you have files with tempos in their filenames, this can be used to order them), and you can even make your own Styles and Groovesets if you want. Once you have done this, your loops become available to Drumcore's browser interface, sorted in the way you've determined. If none of the patterns appeal, and you are bored of all your own loops, then Drumcore still offers a way to please. It offers an excellent option which randomises the content of audio files in a musical way, named after Peter Gabriel. Although the 'Gabrieliser' doesn't give you a useable result every time, every once in a while it comes up with a cracker of a rhythm which you wouldn't have thought of by programming or chopping up a loop Selecting loops ordered by Styles. yourself. It works virtually instantaneously, too, so you can just keep clicking on the Gabrielise button until you get something which appeals — I saved off a few real corkers which turned up while doing this review which are going to come in nicely one day.



Of course, none of this great stuff would be that much use if you could only run it inside the Drumcore program. Fortunately, Drumcore can output in Logic, Cubase/Nuendo, Digital Performer, or Pro Tools formats, or generic ones like WAV, AIFF or Sound Designer II. Files can be 16- or 24-bit and at sample rates from 44.1 up to 192kHz. I tried using Drumcore files with several different sequencers, and everything worked swimmingly. Rewire is also supported, so you can feed the output of the Drumcore engine through your audio software if it supports this.

Conclusions As you can probably tell, I was really impressed by Drumcore, but as mentioned at the start of this review, it's up against some stiff competition at the moment, all at around the same price. Stylus RMX was reviewed in detail in last month's SOS, and a review of Battery 2 is still forthcoming, but the obvious other contender is FXpansion's BFD, which is only 30 quid more expensive. BFD does come with some great drum sounds and arguably offers more flexibility, with the ability to mix and match the levels from overhead, room and close mics, and to process the different instruments in the kit through different output channels, which is not possible with Drumcore. However, auditioning kits is much slower with BFD, as it takes a lot longer to switch from one kit to the next. Like Drumcore, BFD comes with a lot of MIDI patterns to make preset grooves possible, but it is really about programming your own patterns and using your sequencer to trigger the patterns. And whereas BFD's raison d'être is the different classic kits they have painstakingly set up and sampled, with Drumcore it is the performers who are the focus, not the individual instrument sounds. After some time spent with both, I think I can categorise the difference as follows. If you want a particular drum kit or combination of specific bits of drum hardware, then BFD offers you a level of mixing and matching, and of subtlety in the triggering department, which no other library can touch. But if you want classic drummers to bring their own special something to your music, or you want to make selecting from your own library of loops and grooves a speedy process which does not get in the way of writing, then Drumcore is the way to go. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Trident Audio 4T Celebration

In this article:

Peering Around The Back Side-chain I/O & Stereo Linking Controls Signal Processing In Use

Trident Audio 4T Celebration Recording Channel Published in SOS February 2005 Print article : Close window

Reviews : Recording Channel

Trident Audio 4T Celebration £652 pros Good-quality sound. Versatile EQ and effective dynamics processing. Stereo replay/monitoring input. Useful EQ Magic control on DI input, with flexible routing options.

cons Lots of key technical information missing from instruction leaflet. Phantom power appears on TRS input socket. Lacks a separate headphone level control.

summary An innovative product with several unique features which extend its versatility well beyond that of most channel strips. Brimming with Trident heritage, this is a quality product which addresses several different markets and applications superbly well.

information £1086.88; promotional pricing for a limited period, £652.13. Prices include VAT. Trident Audio +44 (0) 1474 815300. +44 (0)1474 815400. Click here to email www.tridentaudio.co.uk

Despite its vintage design credentials, this new recording channel has some interesting new tricks up its sleeve. Hugh Robjohns

John Oram has now been involved in the design of mixing consoles and related products for forty years, and to celebrate that fact he has launched the 4T Celebration channel strip. This new product offers a very interesting range of facilities, with Oram's famed sound quality, at a very attractive UK price for a limited period.

Photos: Mark Ewing

This isn't the first time John has made a special product to celebrate his years in the business. He launched the Hi-Def 35 equaliser five years ago to celebrate his — you guessed it — 35 years in the industry. The 4T Celebration is manufactured entirely in the UK and built to high standards, which would normally force a high price. However, Oram has been able to manufacturer it in large volumes thanks to a very significant order from a well-known US retail chain. This has brought the price back down to the current very competitive level. This 1U rackmounting channel strip incorporates a mic preamp along with both DI and line inputs, a three-band equaliser, and a compressor/limiter. That might sound pretty typical for the genre, but the 4T goes on to offer some unique and very useful features including a stereo input, a stereo output, some natty sourcerouting modes, and headphone monitoring.

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Peering Around The Back My engineering background means that whenever I pick up a new product, I always start examining it from the back. The connectivity of a product tells you a lot about it long before you start working through the front-panel controls. The 4T starts off on the right-hand side with the main input: an XLR socket and a TRS quarter-inch socket, wired in parallel. I really do mean 'wired in parallel' too — if you switch phantom power on, it appears at both the XLR socket and the TRS socket. Since the same input connectors are used for both mic and line inputs, this potential appearance of phantom power might well upset some line-level devices with electronic output stages, especially those with ground-compensated outputs. It is very unusual to find this situation. Normally a couple of DC-blocking capacitors are used to isolate the TRS socket connections from the phantom power applied to the XLR socket. So when plugging things into the 4T, care is needed to ensure phantom is switched off. The phantom-power supply is completely separate from the main PSU that supplies the rest of the machine's electronics, and even has its own miniature toroidal transformer. However, I noticed that, as with most preamps, the phantom voltage takes quite a while to discharge if there is nothing connected to the input socket to help drain the supply's reservoir capacitors. So if you disconnect a mic, switch off the phantom, and then plug in a line output, there may well still be the best part of 48V on the terminals which will discharge through the output stage of the connected source. While I'm nitpicking, it is interesting to note that the XLR socket is mounted upside down — a necessity brought about because of the internal construction of this unit. However, since there is no latch on the socket this isn't a practical problem, although I found myself continually having to rotate the XLR connector in my hand as I tried to plug things in!

Side-chain I/O & Stereo Linking Moving on from the input connectors, we have a pair of quarter-inch sockets which provide an external link buss for the dynamics side-chain. Again, it's usual to require a pair of sockets for this function, and they are clearly marked Link In and Link Out. There is no front-panel facility to activate the side-chain link, so I presume it works as soon as the cables are plugged in. The Technical Specifications & Features leaflet that is supplied with the product makes absolutely no mention of these connectors at all! Next we have a pair of phono sockets which provide the unit's stereo input, file:///H|/SOS%2002-05/Trident%20Audio%204T%20Celebration.htm (2 of 6)9/22/2005 8:20:53 AM


although care is needed when plugging a source in, as the left input channel is on the right phono socket (as viewed from the rear), and vice versa. Finally, we have the stereo output connections, with both XLR and TRS sockets. Again, these are wired in parallel and the XLRs are upside down. In this case it is more of a problem, though, as the release catch on the female XLR cable connector is underneath the plug and quite hard to get at. Like the stereo inputs, the leftchannel output is provided on the right-hand pair of sockets. To the left of the rear panel is a mains-voltage selector, a fuse holder, and an IEC mains inlet.

Controls The front panel is very simple, with a brushed-metal finish, dark-red labelling, and elegant knurled knobs. At the extreme left is the DI instrument input section, comprising a quarter-inch unbalanced input socket, a gain control, an EQ Magic control, and a white button labelled Direct. The input circuitry presents an unusually high 10M(omega) impedance, making it ideal for use with piezo pickups on acoustic guitars as well as with conventional magnetic pickups, and the input can accept signals up to a massive +15dBu. The EQ Magic control is a clever constant-power design that either boosts the top and bottom while cutting the middle, or boosts the middle while cutting the top and bottom (turnover frequencies are 100Hz and 10kHz), all the while maintaining the apparent loudness. The control has a centre-detent at the flat position, and it is a very useful facility for creative tonal tweaking of a DI'd guitar.

Equally useful is the Direct button, which routes the DI input directly to the output stage (appearing equally on both outputs, effectively panned centre) instead of passing it through the EQ and dynamics processing. This enables the user to modify the guitar DI signal with the EQ Magic control, while leaving the EQ and dynamics facilities for the exclusive use of the microphone signal, if required — a great solution for a gigging solo musician who wants a simple, compact, but very flexible and high-quality solution. The next front-panel section is the main mic/line input. As already mentioned, the two rear-panel sockets are wired in parallel and there is no mic/line switching as such. The transformerless input stage, which is identical to that of the Trident S20 and S40 units (derived from the Trident TSM console, but based on the ubiquitous SSM2017 chip) is able to accommodate input signals up to a massive +23.5dBu, yet still manages an EIN figure of -127.6dBu. Up to 60dB of gain is available via a detented knob.



Also provided are a white polarity-reverse button, and a red phantom-power button. The mic/line input signal always passes through the entire EQ and dynamics processing chain, and appears equally on both output channels (in other words, it is effectively panned centre again). The stereo input is equipped with a level control and a white Mon button. This facility can be used for a number of different purposes, but perhaps most obviously it allows a stereo backing track to be mixed with a microphone and/or DI input. The stereo signal is passed straight through to the stereo outputs via the level control, where it is combined with the DI and mic/line signals (which appear in the centre of the stereo image). Again, this is useful for the gigging musician, who can use the 4T to process and control not only a DI guitar and vocal mic, but also the backing track from a CD player, computer, or whatever as well. In a home-studio setting, the stereo input can be used to monitor the output from a computer workstation. Pressing the Mon button removes the stereo signal from the outputs and routes it to the headphone monitoring section, where it is mixed with the main output signal as a cue feed. In this way the user can hear their own performance live (and without any latency) along with the replay cue tracks from the workstation.

Signal Processing The 4T's signal processing is, as you would expect, also derived from previous Oram products and designs. The three-band EQ is derived from the Trident Series 80 console, but has only one swept mid-band instead of two. The top and bottom shelf sections have switchable turnovers (50Hz or 150Hz for the bass and 7kHz or 12kHz for the treble), with a ±15dB range on the gain controls. The swept mid-range The rear-panel connections include a stereo section spans 100Hz to 10kHz with the input on RCA phonos for mixing a stereo same gain range, and a red button monitoring signal in with the mic or DI inputs. (with associated green LED) switches the entire EQ section into the signal path. All the gain controls have centre detents at the unity-gain position, and all the knobs have a heavy, solid feel to them. The dynamics section is based on Oram's solid-state Sonicomp design — a favourite of Al Schmitt, apparently — with rotary controls to adjust the Threshold (from off to -25dBu), Attack (0.1ms to 40ms), and Release (0.05s to 3s). The ratio is determined with a white button, offering 4:1 compression or 30:1 limiting. A red button (with associated red LED) switches the dynamics section into the signal path.



Finally, the output section features a level control (±15dB), a headphone socket, a VU meter, and a white button to select either output-level or gain-reduction metering. A nice feature here is that the meter illumination changes colour depending on the mode: blue for level and green for gain reduction. The maximum output level is a massive +28.5dBu, so there is plenty of headroom even when driving professional A-D converters. Sadly, though, there is no separate level control for the headphone output, which may be a frustration to anyone using the 4T as the front end to a workstation.

In Use As we have come to expect from products carrying the Trident badge, the unit performs very well and has excellent specifications. The sound character on the mic input is very slightly on the warm side of neutral, a quality revealed most clearly by sources containing complex transients — the classic 12-string acoustic guitar test, for example — and I would say that the effect is musically flattering, if not always very exciting. The DI input worked very well, and I was able to achieve some excellent results recording an acoustic guitar fitted with a high-quality piezo pickup (if such a thing really exists!). The EQ Magic facility is superb, allowing a nice range of tonal shaping to be performed very simply and effectively, and I liked the flexibility to route the DI signal straight to the output or through the main EQ and dynamics processors. Both the EQ and dynamics sections worked entirely as advertised, and were very effective and easy to set up. The EQ had sufficient flexibility and range to cope with every source I tried, and the compressor was rather more versatile than I was expecting. I feared that, with only two fixed ratios, the unit would easily be caught out, but in fact it appears to have a broad, soft-knee slope, and so careful adjustment of the Threshold control can provide less heavy-handed gain reduction if required. With most contemporary vocal styles, though, a ratio of 4:1 is pretty much ideal anyway, and I found the dynamics section actually worked very well almost regardless of the source. To nitpick again, the instruction leaflet leaves out a lot of important operating details that have to be fathomed by trial and error, and I would have preferred a separate headphone level control — although panel space is rather tight and I can't think of any other facility I would sacrifice to provide the space. However, the flexibility of routing the DI and stereo inputs is excellent and makes this a very versatile, and probably unique, product. If the 4T's innovative feature set appeals, you won't be disappointed with the performance. Published in SOS February 2005






Allan Walker & Craig Connor: Grand Theft Audio

In this article:

Station To Station Genie In A Bottle Into The Buffers Ped Power Mood Music Two Billion Conversations Breaking Down Career Options Checks & Balances

Allan Walker & Craig Connor: Grand Theft Audio Sound & Music For Rockstar Games Published in SOS February 2005 Print article : Close window

People : Artists/Engineers/Producers/Programmers

Imagine selling a million copies of your music in just over a week, with no video, no radio play and no touring. Impossible? Not if it's the soundtrack to one of the most successful computer games of all time... David Greeves

When the Playstation 2 game Grand Theft Auto: San Andreas was released in the UK at the end of October 2004, it took just nine days for sales to reach the one million mark. With 677,000 copies sold in its opening weekend, GTA: SA became the fastest-selling games title ever in the UK, breaking the record set by its predecessor, Grand Theft Auto: Vice City, which has sold over 10 million copies worldwide. It would be a rare thing indeed for an album or DVD release to sell a million in a little over a week, and, at £40 a pop, GTA: SA saw more money changing hands than most blockbusters can expect to bank in the same period. It's just one of many signs that computer games have risen to the level of, and may already have surpassed, films as the diversion of choice of the masses. And in terms of production values and time spent in development, today's blockbuster games are creeping up on their movie counterparts.

Photos: Richard Ecclestone Craig Connor (left) and Allan Walker (right) of Rockstar Games.

Having worked for Rockstar North in Dundee, and now Edinburgh, for the past 10 years, Allan Walker and Craig Connor have seen the transformation of the games console from children's toy to grown-up's hobby at first hand and, in overseeing the music and audio for all of the five GTA games to date, they've played their part in it — the GTA franchise was aimed at adults from the off, and has arguably done more than any other to make it OK for adults to talk about

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gaming. This change in attitudes is not lost on the pair. "I remember even six years ago people would kind of laugh and say 'Oh, so you do bleepy bloppy noises,'" says Craig. "That's what people used to think of sound for games. The Playstation 2 has changed that now." "It isn't like we're doing the sound for a game any more," Allan adds. "We're trying to make an imaginary world come alive. That's where the game becomes immersive, when you believe you're there, and the sound can really let you down on that front — it's all got to be believable. Working within the limitations of the machine, that gets pretty tricky." Given the scale of the game, with its hundreds of square miles of virtual cityscapes and countryside to explore, and hundreds of individual interactive characters, I'd say this was something of an understatement.

Station To Station All five GTA games share the same essential premise. Starting out as a lowly petty crook, the player must work his way (it's always a he, incidentally) up the underworld ladder by stealing cars and performing a variety of unsavoury tasks for a colourful cast of mobsters, pimps and hustlers. Crammed full of movie and pop culture references, the games bombard the player with an irreverent, mischievious and downright sick sense of humour, manifested in the characters you encounter and the commercials and DJ chatter on the car radio. The car radio provides the bulk of the musical content of the game, and any time the player is inside a car, there are 11 different stations to choose from. The mammoth task of mastering this material, all of it existing music licensed by Rockstar, fell to Craig. "In total, there's 155 tracks in the game and I had to master all of them, plus the ones that didn't get used, as well as 70 commercials and the DJs' dialogue. I think it's about 12 albums' worth of stuff, and not 12 separate albums — it all had to work together. You can flick between the different stations in the game at any time, so getting a consistent level was really important. I did the mastering in Pro Tools, mainly using Waves compressor and limiter plug-ins, Focusrite Red EQs and sometimes a BBE Sonic Maximizer. The game is set in 1992, so there's a lot of music from around then — lots of hip-hop and new jack swing — but there's also a rare grooves station which is all '60s stuff, there's some '80s stuff, some classic rock, even a country and western station. There's all these different styles of music that needed to be mastered into one game, so that was a bit of a nightmare. I used different compressor and EQ settings on the DJ dialogue, too, because I wanted them all to sound different, to sound like they were coming from different stations, so the DJs on the hip-hop stations had really fat bass and the country one would sound like it was a shitty mic in a shitty studio."



The task of mastering all of the material for the different radio stations was made more complex by the fact that much of it is interchangeable. "In all the previous titles, each radio station was a loop of about an hour and a half with the adverts and the DJs all mixed in, so I could just EQ the whole thing the way I wanted. This time the tracks are selected randomly from each station's playlist and there Craig Connor's studio at Rockstar North. are three different DJ intros and outros for each track. Most of the ads appear on several different stations too. We wanted people to be able to play the game for a long time before hearing anything repeated. We knew that there was going to be about 100 hours of gameplay in the game, so we wanted to really break it up so everyone gets their own experience from the game. But it meant that if I made an advert really bassy to fit in with a hip-hop station, when it's played on a station that sounds much more tinny it'll really pop out at you. So I had to keep a level head and not go overboard on that front." Different cars even offer different qualities of radio reproduction, and as the player progresses through the game they can purchase subwoofers and other upgrades for certain vehicles. Colin Entwistle, Matthew Smith and Alastair McGregor were in charge of programming the audio elements of the game engine. "Basically we come up with creative stuff and say 'Can you do this?' and they say 'No,' and we say 'Why not?' but eventually they'd find a way to do it," Craig jokes. "They're really keen to push the boundaries of what we can do with the Playstation 2, as are we."

Genie In A Bottle Perhaps the most unexpected tool in the Rockstar North studio is an empty water bottle. "When we need to create sound effects from scratch," says Allan, "somehow we always end up finding a way to use a water bottle! For example, in San Andreas, we used the sound of an empty bottle being jiggled up and down in the opening of a desk drawer to creat the sound of a blown-out car tyre flapping along the road." "And for Manhunt," Craig adds, "when we needed to get that spooky pizzicato sound from The Shining, we did that just by plucking rubber bands stretched out around a water bottle, and then putting on really nice reverbs and a bit of distortion — it adds such a great sound."

Into The Buffers Battling with the constraints of the format is a recurring theme in Allan and Craig's line of work. One might have thought that with the capabilities of the file:///H|/SOS%2002-05/Allan%20Walker%20&%20Craig%20Connor%20%20Grand%20Theft%20Audio.htm (3 of 10)9/22/2005 8:21:19 AM


current generation of games consoles and a whole DVD game disc to store data on, such limitations would be less pressing, especially when compared to the early days of computer games where the audio designer had only a few bytes and a hardware sound chip to play with. Not so, according to Allan. "You had to work down to the last byte in those days and it's amazing but Allan Walker's studio, with Parker Fly guitar we're still working down to the byte "for stress relief". now, literally trimming down the tail of every sample. The job hasn't got easier — it's just as hard but it's a bigger job. There's more space to fill but there's so much to fit in there. And with the next generation of consoles it'll be the same — the first thing we'll do is up the quality of all the samples and that'll eat up a lot of space. "The main constraint in the past has always been the amount of memory you've got to play with. In this case there's a seperate audio RAM. When we take into account all buffers required for specific tasks, what we have left could store around nine seconds of CD audio. So we have to fit whatever could be going on in terms of audio at any one time, which is often an awful lot, into that nine seconds. We're also careful to arrange groups of sounds associated with specific things into packets to minimise reading off the disc. For example, when you jack a car in the game, we've got the time between the player pressing the button and the character on screen getting to the car to load all the sounds that go with it — the sounds of the door opening, pulling the driver out of the car, any dialogue they might have, slamming the door again and so on. So while memory was previously the main constraint, now it's actually the disc — apart from anything, there's so much dialogue in there the disc isn't big enough, and it's a DVD!"

Ped Power He's not kidding. Some 4000 lines of scripted dialogue advance the plot of the game, voiced by an impressive cast of movie and music stars from Samuel L Jackson and James Woods to Ice T and Shaun Ryder, while the lead character, CJ, has 7700 different comments he can make according to the situation at hand. San Andreas is also populated by 300 different types of 'ped' who'll interact with CJ according to his actions as he makes his way through the game, and can even strike up conversations with each other, using 60,000 possible responses. "Every ped was given his or her own personality," Allan explains, "and that dictated what we needed to provide for them in the way of audio. They'll react in different ways if you get in their way on the street or start a fight with them — they'll even comment on CJ's clothes or his haircut. But when there are 300 peds who can talk about the colour of your shoes or whatever, you can't just have



them all saying one thing because you'd hear it all the time, so you need variations. So that's 300 peds times three or four variations on the comment, or eight or nine if you count the versions that didn't get used. So just a little detail like that would flow into hundreds of samples. CJ's also got about 10 different moods which govern his comments. So if something happens in the game which he's going to say something about, the version of that comment you'll get will depend on his mood. If you've just failed a mission in the game, he might be angry and you'll get the pissed-off version, or if you've just completed one he'll be wisecracking and being a bit pleased with himself. A lot of code had to be written to make all that work and the game can track 150 of the last bits of dialogue, so that covers most repetition — it can really break the illusion if you hear something then you hear it again straight away."

Most of GTA: SA's spoken parts were recorded in the US, but this vocal booth at Rockstar North's Edinburgh studios was used for recording jingles for the game's fake radio commercials, with the Rode Classic mic.

"We did two sessions in New York and one in LA to record the ped comments, each with 100 actors coming into the studio in four days," adds Craig. "That was pretty manic — getting the actors to scream 'Get out of the way!', 'Move it, asshole!' and that sort of thing. Then we brought everything back here on an iPod and on Lacie drives as well as extra backups, and set about editing it all with the rest of the team [audio designers Will Morton and John McCavish]. There's a wide dynamic range between the really loud shouting and the quiet chatting, and we had to keep it all consistent. We also had to bear in mind that there might or might not be all sorts of things happening at the same time — guns going off, the car radio and so on. Thankfully we kept up close communication on what everybody was doing. We also used an Excel spreadsheet to keep track of what was finished and what we still needed — with tens of thousands of bits of dialogue, we had to."

Mood Music For Rockstar's horror-themed stealth/action game Manhunt, Craig was called on to create a more traditional movie-style soundtrack, while Allan oversaw the game's range of alternately gory and unnerving sound effects. "Basically," Craig explains, "we sat and watched every horror film we could find so that we could get a feel for the format and see what made horror movies scary. Because it's not the man who's sneaking up behind you or hiding in the bushes or whatever else, it's the music that tells you how to feel. So we just separated out all of the different elements of soundtracks and then thought 'Right, how can we put this into a game?'" The idea they hit upon in the end was to design four pieces of music for each level which could be looped and layered one on top of the other to incrementally raise the levels of tension and excitement according to the action. Allan explains: "Once file:///H|/SOS%2002-05/Allan%20Walker%20&%20Craig%20Connor%20%20Grand%20Theft%20Audio.htm (5 of 10)9/22/2005 8:21:19 AM


we'd broken down the elements of the music that we needed for the game, we were listening to horror soundtracks on their own, and you can tell what's going on just from the music — you know, now they're hiding, now they're being chased, that's the fight bit. So we had these four different moods for the soundtrack that are dictated by what's happening in the game, five if you include the times when you're hiding in the shadows, when the music drops out entirely and you just hear a heartbeat. There was this sort of mellow wandering-around music, then when you've been spotted it goes up another notch, then again when you're being chased and finally when it turns into the final fight the whole tune comes together, and all four stereo loops are streaming at the same time. The crossfades going Craig's Korg Mono/Poly formed the from one piece of music to next were backbone of the Manhunt soundtrack. all done in code and the timing of them was something that we really had to work on, checking if this bit was going to work with that bit and setting up the volume fades, but somehow whenever it went into code and we could test it in the game it just worked so much better." The music itself was created almost exclusively on a battered old Korg Mono/Poly analogue synth which was passed through some distortion and lo-fi effects, combined with some similarly grimy drum samples, and sometimes truncated to eight-bit for good measure. The soundtrack's unrelentingly dark and sinister mood perfectly reflects the content of the game, and also struck a chord with a number of drum & bass and electronica producers — an entire album of Manhunt remixes from up-and-coming producers was released on Aphex Twin's Rephlex label. As pleased as Craig is to see his music recognised, he wasn't sorry to move on from the project: "I loved doing the Manhunt stuff but I'm glad that I've finished it now — I got to a very dark place by the end!"

Two Billion Conversations Because of the interactive nature of the game and its environment, it's impossible for any one player to hear everything that has been put into the game, but this doesn't seem to bother the creators. "We're basically supplying the world with a library of sound," says Allan. "It isn't like we're making a film which is linear, we're just putting options and possibilities into the game allowing the world to do what it does — it doesn't matter if you don't hear everything. Just the fact that you can just wander about listening to peds talking to each other, and they'll talk to you and you can talk back, brings the whole thing to life. That alone adds another couple of hours of gameplay. There's two billion conversations that can happen,



so nobody'll ever hear everything — we worked out that if one happens every two minutes it would take 76 centuries for them all to take place!" "It seemed insane at the start," says Craig. "When we started talking about all the things we wanted to put in, we were thinking 'Will you actually see, will you ever hear it?', but then it started to Some of the rack gear in Craig's studio. make sense. For instance, the DJs will From top right: Ensoniq DP4+ effects, BBS sometimes comment on the weather 862 Sonic Maximizer, Behringer Ultracurve and other things that are happening in Pro EQ, Emu ESI4000 sampler, Drawmer the city. Getting that to work was a DL231 dynamics and BBE 462 Sonic Maximizer. huge task for the programmers, and maybe people won't notice it until they've been playing a few hours, but then I was walking past one of the game testers and he was like 'Craig, I think the DJ just said it was raining — how does that happen?' and I thought, 'It's worked, and it's worth it.' It's little things like that that make the game feel more real."

Breaking Down The sound effects, another crucial element in the game, which range from car engine and tyre noises, gunshots and explosions to rain and other atmospheric sounds, are rendered in 5.1 surround, panned to correspond with the action taking place on the screen. Allan explains: "We use Pro Logic II to do the surround stuff — we used to use DTS but that actually halved our channel count so we gave up on that. The Playstation 2 has 48 hardware audio channels. We use four to stream the radio and the background ambience, so we end up with 44 channels to cater for everything else that's going on in the world — all the sound effects and dialogue that might be going on at any one time. But once again, our amazing programmers have come up with some awesome stuff where the game keeps track of up to 200 channels in memory, and you only hear the 44 loudest ones. If a sound gets pushed out by louder sounds, this means that, whether it's a loop or a one-shot, it'll be tracked for its duration until the louder sounds fade and it can come back in in the right place. So, you'd never notice it while playing the game, but if a character is talking and there's a much louder sound, that dialogue would be cut and then come straight back in at the right time. The sentence would seem like it was getting played back all along, but there was maybe a gunshot in the middle which voided the need to hear it at that moment." The sounds themselves were mostly culled from commercial sound libraries, but were then split into much smaller constituent parts which could then be controlled by the game engine, giving maximum flexibility from as little data as possible. Allan continues: "The sound effects are broken down to a level where there are so many small samples and so much code going into controlling them that



they're almost being synthesized. The rain sound, for example, uses nine channels, with two loops and seven little one-shot samples that play all around you in surround. It sounds like rain but it's actually coming out of tiny little samples so that we can squeeze all of them on and have that happening at the same time as streaming radio off the disc. If a car drives by you, you'll hear the different sound of the rain on the roof of the car as compared to the road, and that's all programmed in surround as well." This deceptively complex approach to incidental sounds is exemplified perfectly in the treatment of gunfire, a frequent occurrence on the streets of San Allan's studio is also Pro Andreas. "The gunshots use up to eight or nine Tools-based, with further audio channels each — as far as I know this hasn't outboard gear including Avalon DI box, TC Electronic been done in a game before. We've got five Reverb 4000, Behringer Tube different sounds that will be attached to a single Ultra Q equaliser, SPL gunshot so it can be rendered in surround, and the Vitalizer enhancer and dynamics of the sound change depending how Drawmer DS231 dynamics. close the character you're controlling is to the gun being fired. The gunshot itself is made up of several different sounds — a little bass sound, a dry 'click' sound and so on — and these all get mixed differently depending on the distance from the shot. So if your point of view is far away from the shot, all you'd really hear is the shot's echo — the reverb tail of the gun that'll fade in on its own. As you get closer to it there are other sounds that are layered and crossfaded. When you're right up close to it you can almost hear the air getting sucked out of your ear, there's such a nice pop on it. Then there's the bullet itself — we've got different sounds for the bullets hitting concrete or metal and even just going through the air near you. We also use a slightly random pitch element just so it doesn't sound completely repetitive and mechanical when someone's firing repeatedly."

Career Options As in all areas of the music industry, there are lots of people wanting to get into audio design for computer games and a relatively small number of jobs available. When I ask Allan and Craig if they've any advice to offer, they're definitely clear on what not to do. "Don't just send in a music CD to show off your music abilities. It is a major part of the job, but there's a whole lot of other stuff as well. Go out and play the games that are out there at the moment and see what's happening. A lot of the CVs we'll get sent are attached to a CD with a note saying 'Here's some background music I thought you would like to use in your game.' Come on! Do you know what year it is? Do you know how far games have come since then?" Allan agrees: "We sent them a tape — don't do that any more. That was 10 years ago! Try and understand what's going on the music and audio side of games these days. When you write a letter and a CV you need to show that you've at least some idea about what's involved in the job. There's musical ability, and file:///H|/SOS%2002-05/Allan%20Walker%20&%20Craig%20Connor%20%20Grand%20Theft%20Audio.htm (8 of 10)9/22/2005 8:21:19 AM


knowledge of equipment but there's a lot more to it than that — it's all about creativity and ideas."

Checks & Balances Besides monitoring via the Dynaudio BM6s in their office studios, Allan and Craig were sure to check all the audio in the game on a variety of domestic hifi and TV systems. "I'm religious like that," says Craig. "I check with the same TV that I've had since 1982 and the same speakers. You've got to do it because once the game's finished, you're always going to hear it somewhere coming out of the worst setup in the world." Testing for bugs in the game was equally thorough. "We've recently got a new tester, George Williamson, specifically for audio, so he'll play the game all day and report all of the bugs he finds. We've got a debug setup here where we can do some nasty things to the game just to see what'll happen. But the sheer scale of the game was really scary — it's absolutely massive and there's bound to be things that we're going to miss. The scariest thing of all is when it's done and it's in the shops, and you know your mates are going to come round and tell you the truth!"

Craig's recording and editing system is based around a Digidesign Pro Tools rig with 888/24, HD96 and 882 interfaces. Between those are Behringer Ultrafex II and Ultra Q, and the Dolby Pro Logic II Gaming Encoder used to render the surround elements for the GTA: SA soundtrack.

It's not just what's in the game: what was left out is also a cause for sleepless nights. "It does just come down to time constraints at the end of the day," says Allan. "We could go on forever — we could easily have kept working on the game and improving it for months, but you have to draw the line at some point and start putting things on the list for next time." "Normally we'd come up with really crazy ideas when we were working late," says Craig, "and then in the morning we'd come in and say 'You know what, that's a stupid idea, that's fucking insane.'" "Yeah," agrees Allan. "Let's do it!" Published in SOS February 2005






Business End

In this article:

Business End

Cloud 149 Rotating MPG producers evaluate Published in SOS February 2005 Leslie Uncle Print article : Close window Johnny People : Miscellaneous This Month's Panel

reader tracks

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

Cloud 149 Tim Speed (TS): "It sounds ridiculously '80s, doesn't it? Like Gary Numan or Ultravox." Paul Hollis (PH): "They've got that Frankie Goes to Hollywood bass drum down well though, haven't they? The first track's very polished, you can imagine it being used on a corporate video or something — that's how clean it is." Nikki Harris (NH): "I think the first track might be good to introduce them when they're playing live or something but I don't think it's particularly representative of the rest of the songs." Brian Campbell (BC): "Yeah, they're really different, I thought it was a different band to start with. When you've got a track that's obviously supposed to be an intro — like it's only one minute long and there are no vocals — you've got to wonder why it doesn't sound like the rest of the songs. "I'd like to see it like dirtied up a bit, it's too clean at the moment and that makes it sound even more '80s." PH: "It is like a homage to the '80s, though, isn't it? It just sounds like so many '80s bands, you know, like Visage, the Human League, Dave Stewart and the Eurhythmics. It's like a study on the '80s." NH: "That's it though isn't it? But at least they've done that well and they've been successful in that." TS: "They've completed the coursework then. It's just not original though, is it?" NH: "No, not at all, and it doesn't even have the hooks that the likes of the Eurhythmics have." file:///H|/SOS%2002-05/Business%20End.htm (1 of 6)9/22/2005 8:21:24 AM

Business End

TS: "Liverpool's full of bands that are '60s retro. You've got to blame the La's for it really — a lot of them are copying those sort of Gerry and the Pacemakers sounds like the La's did. This is just as bad, in the sense that it just doesn't have an original take on it, it's not like it's moved on or progressed — it's just a straight copy." BC: "Apart from the distorted guitars maybe, but then that was still there at the time to some degree." TS: it just sounds like a homage to that sort of style — it's just like what the Rutles did." NH: "The '80s stuff was going to come back and that's just what they're doing. It was kind of inevitable." BC: "Ladytron are doing this sort of thing but they're being much more original with it and I think they come across as being a lot more credible because of that." TS: "I do quite like it though." NH: "Yeah, I think it would sit well in an Evol night, which is the club that the Ladytron DJs do." BC: "I just don't think there's anything original about this and I find that really off-putting. I think the recording's far too clean, it sounds like '80s compressors and reverbs being used on it — and maybe that's the point but it's exactly the kind of thing I don't like." PH: "They're good — they're competent and they're very good at what they do. There's just not a lot of emotion to it though." NH: "I think the tunes are good but they're going to need much better hooks if they're going to compete with the Eurhythmics or anything like that."

Rotating Leslie BC: "This sounds a bit like Hope of the States. It seems a bit too muso for me, like they're trying too hard.

Track 1 2.6Mb

"With the recording, it seems like everything's in its own separate place rather than going along together as one song." PH: "I hate the vocal, it sounds really soulless and forced, like he's trying to be someone else instead of himself." NH: "I like the beginning of the third track, the bit with the guitar part — that's really nice. I'm not very keen on the rest of it though. It'd be interesting to hear something acoustic by them, just to like strip it down and see what there is." BC: "I never trust bands who use Trace Elliots on the bass. Not that I would have known that they were file:///H|/SOS%2002-05/Business%20End.htm (2 of 6)9/22/2005 8:21:24 AM

Business End

using one if I hadn't read their letter. I just don't like them, they've got no drive to them, it's a really clean sound and I don't like that." TS: "The recording's really un-warm, it's got a really cold sound with no bottom end to it. There's just no bass. I think I actually prefer the sound of a rotating Leslie — they should record one of them, you get a nice little click every time it goes round. "All bands seem to sound like somebody else, unless they're really great. It's very hard for bands to get away from that. It's either Radiohead or Oasis, they're the bands that everyone's sounded like for the last 10 years, that or the Beatles. Y'know what I mean — they all sound like someone else. "The singer on this sounds like Paul Draper out of Mansun." PH: "Yeah, the vocalist out of Hope of the States, his hero's Paul Draper, he used to interview him and do fanzines and stuff and his vocals are just like this." TS: "The best advice you can give them is to just be themselves more. I mean they're talented, they can obviously play and write tunes, but you've got to be yourself or you'll just sound like everybody else." BC: "There aren't any original sounds on this but I don't think the songs are that original either — they're not taking any risks." PH: "It doesn't seem like they're a band that practice in a room every day, it sounds like they've been practicing once a week and then they've got together for this recording. They're hardly going to set the world alight with this, are they?" BC: "I think the recording's too clean, and again it sounds like they're not taking any risks. Be bolder!" TS: "I think the recording's good, I think they've probably got a problem with their monitors because the bass isn't loud enough. I think it sounds good, the bass is just far too quiet though. I recommend they book in here. Yeah — be yourself and spend a couple of days in Elevator, that'll make all the difference."

Uncle Johnny BC: "This is great, I like this. it's really cool."

Track 1 3.3Mb

TS: "It's very lo-fi, isn't it?" BC: "Yeah, but I like that, it's interesting. There are different elements of detail in the production and everything. It sounds like a cross between Beck and Elliott Smith."

file:///H|/SOS%2002-05/Business%20End.htm (3 of 6)9/22/2005 8:21:24 AM

Business End

NH: "You can definitely hear the Beck in there and maybe a bit of Beatles too." BC: "This is so much more refreshing than the last two we've listened to 'cause it's something different — it's someone taking a few risks." NH: "He says in his letter that he's just done at home on quite a basic PC setup and it's a really good recording." TS: "He's got loads of compression on this, it sounds really nice." PH: "It's hard to analyse this, you just want to sit back and listen to it for what it is. It's hard to give him any advice 'cause he seems to being doing really well already." BC: "I think the sleeve is fantastic — it's brilliant." PH: "It's definitely eye-catching." BC: "With the other two bands we've listened to you can imagine them just having a shot of the band members or something for their publicity, but this shows a bit more tact and credibility." PH: "It's always the way with demos — you get loads where it looks like they've spent more time on the cover than the music and then you get one with a handwritten label and they're always the best." NH: "Sometimes people can try far too hard with cover art, but with some bands it just comes naturally. When we got the Little Flames' demo, they'd just done the cover themselves and it was a bit abstract and it wasn't too over the top and that combined with the music did make it stand out. Cover art really isn't a necessity though. We listen to all the demos we get — regardless of how good the sleeve is." BC: "Apparently John Peel got sent 'Teenage Kicks' on a cassette tape wrapped in a piece of paper with 'John Peel, London' and a stamp on it. And he said that that attracted him more. I think bands can definitely try too hard with this sort of thing." NH: "If it's something really glossy and interesting and then the band's actually quite normal you just feel disappointed." TS: "They should be spending all that time on the music." NH: "It's not like it's got to look like a real CD or anything, and most people are looking for something that's earlier on than that stage, d'you know what I mean?" TS: "I think the artwork's important because it's the first thing you see and it's like a first impression. We had this band in called the Altericks and when they dropped their CD round the first thing you noticed was the artwork. It wasn't amazingly high-tech or expensive, it was just a good idea and that's what's important. It was the same with the Little Flames, actually. I really liked the name and they had some artwork on the wall of their rehearsal room and I thought 'that's great, that is', and you just know that there's a creative mind behind it." file:///H|/SOS%2002-05/Business%20End.htm (4 of 6)9/22/2005 8:21:24 AM

Business End

NH: "Yeah, Matt from the Little Flames does their artwork and Ian from the Coral does their stuff, that's just them — that's what they do." BC: "I love it when a band come up with their own videos and their own artwork and you get a whole package."

This Month's Panel

Photo by Nigel Jopson

Brian Campbell is the bass player in the band Clinic. Clinic have now recorded three albums, co-producing each one along with Gareth Jones, Ben Hillier and Ken Thomas. Their second album, Walking With Thee, was nominated for a Grammy in the Best Alternative Album category. They have toured extensively in their own right as well as supporting Radiohead on their Kid A tour. Brian is not hideously disfigured beneath his mask. Nikki Harris is the Label Co-ordinator for Deltasonic Records, the Liverpool-based label which is home to the Coral and the Zutons. Nikki's career in music began with a part-time job engineering for a local cable TV company whilst studying for her 'A' Levels. She went on to study for a degree in Sound Technology at the Liverpool Institute for Performing Arts where she developed the interest in the business side of music that would lead to her current job. Paul Hollis has endured a life-long obsession with music. His interest led to him learning to play the saxophone, studying jazz for three years and later going on to become a session musician, playing for numerous bands. More recently he has worked as a tour manager for EMI, looking after Mansun and many others on the road. Today he manages Liverpool-based band the Open (pictured above) who have recently signed to Polydor. Tim Speed was signed as an artist to EMI in the '90s and recorded two albums for them before signing to London Records. Five years ago Tim had a small private studio in Liverpool which he has developed into a fully commercial facility named Elevator Studios. Over the last two years Elevator has played host to the Coral, the Zutons, Ian Broudie, Clinic, Ladytron and many others. Most recently Tim has been producing the Altericks. Many thanks to Tim Speed and Elevator Studios (www.elevatorstudios.com) for organising and hosting the session.


Business End




Composing & Recording With Vanessa-Mae

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From Rave To Ravel Walter Taieb Mac Versus PC Published in SOS February 2005 Symphonic Samples Gathering Pace Print article : Close window Transfer Troubles People : Artists/Engineers/Producers/Programmers Classical Sounds, Pop Mixing

Walter Taieb enjoyed worldwide success as a dance producer, before reinventing himself as a classical composer. Working with violin prodigy Vanessa-Mae tested both his orchestration skills and his musictechnology expertise. Paul Tingen

Although you've probably never heard Walter Taieb's name, you have, in all likelihood, heard his music. He was responsible for a large number of dance music hits, such as 'I Luv U Baby', (seven million sales worldwide, number two in the UK charts in 1995), 'It Feels Good' and 'On Top Of The World', and also did remixes for the likes of Cher and Gloria Estefan. None of these were released under Taieb's own name, hence his relative obscurity. In addition, some intrepid readers may know him as the composer of The Alchemist, a symphonic work based on the eponymous, bestselling book by Paolo Coelho. Most recently, however, he's been working with violinist Vanessa-Mae on her latest classical-rock crossover album, called Choreography. Perhaps surprisingly, it turns out to be rather good, courtesy of several big-name composers, such as Vangelis, Riverdance's Bill Whelan, and India's best-known film composer AR Rahman, along with talented but less wellknown figures such as Taieb and Tolga Kashif, who wrote The Queen Symphony.

From Rave To Ravel

file:///H|/SOS%2002-05/Composing%20&%20Recording%20With%20Vanessa-Mae.htm (1 of 7)9/22/2005 8:21:29 AM


Like Vanessa-Mae, Taieb was once a child prodigy, constructing his first studio at the age of 16, in 1985, around a Commodore 64 computer. After an impressive career in dance music in France, he moved to New York City in 1991 and created his own studio at Times Square, based eventually around an Euphonix desk and a Pro Tools system. Dance music was still the name of his game, but something pulled him towards classical music and the legendary Juilliard School of Music, where he studied composition. "I had one major disappointment with a dance music record, and I somehow felt that my personal taste didn't match the public's taste any more," explains Taieb. "Dance music was more and more financing my classical music, and when it was no longer profitable, I had no reason to do it any more. The Alchemist was the first orchestral work I ever wrote, and when Paolo Coelho supported it, I was lucky to be signed by RCA Red Seal just before I was about to finance the recording myself from the proceeds of 'I Luv U Baby'." The Alchemist (1997) did well for an unknown composer's work, but as it was full of romantic, 19th century-like string writing and based on a novel that's popular with New Agers, it became bracketed as New Age music. Taieb had another stab at the classical limelight with a track for cellist Caroline Dale, 'Empires Of Light', which was released on Dale's 2002 album Such Sweet Thunder. As it happened, the album's executive producer, Rob Dickins, was creative consultant for Vanessa-Mae's Choreography. The album was and is intended to be a celebration of dance music from around the world, and the legendary film composer Michael Kamen was set to compose a tango. When he suddenly died in 2003, Dickins asked Taieb to substitute. "They wanted a rock band with some strings," Taieb explains, "but I wasn't interested in doing such a crossover record. So I wrote a bolero for orchestra, inspired by Ravel, but with my sound. I did an orchestral demo, with lead violin, and Vanessa loved it. After that 80 percent of the album took this direction: accessible, new classical compositions."

Mac Versus PC Taieb began his relationship with computers as a teenager, working avidly with the Commodore 64. After this he migrated to the Atari, with Notator, and then the Macintosh. With most Mac lovers being true believers, the French composer explains how and why he made the switch to PC. "Initially my Mac wasn't really working very well, so I was still going back and forth between my Atari and my Mac. I composed most of my dance music on the Mac, but the platform was always a frustration for me, because every time you bought a new model you had to re-buy everything. Nothing appeared to be compatible, whether hardware or software, everything changed all the time. "I had PC technology in my Euphonix, even though it didn't have Windows. But it was so solid compared to the Mac, it never crashed, ever. I could work for two years without rebooting. But I had major crashes on my Mac system. This definitely had some influence on me. What prompted me to seriously consider PC



was when Emagic developed programs that were exactly the same for Mac as PC. I was on OS 8 when I made the switch. I began by buying a PC for my office, and I put a music program on it to see if it worked, and then slowly the PC entered my recording studio in New York. I first used Windows 98 with Logic. "I prefer the PC platform, it has more choices and more manufacturers build things for Windows. I always loved the Mac hardware, but to me the operating system was never as good. I find that it's more difficult to make a system work on PC, but when it's working, it's really solid. Also, it's easier with PC to work on the software like a mechanic. With Mac you don't have a lot of options to go inside of a program and make it work perfectly for you. On PC you can fix things and open a window and write a few lines of programming. However, I've heard good things about Mac OS X."

Symphonic Samples Choreography ultimately features three Walter Taieb compositions, 'Bolero For Violin And Orchestra', 'Tango De Los Exilados' and 'Tribal Gathering'. He composed and demoed them in his Paris studio, which has an unusual setup, featuring four PCs, all with Windows XP and clock speeds of around 2.5GHz and over 1GB RAM, but no mixing desk. "I never understood why people have such a great need to touch buttons," Taieb comments. "I love to work virtually, and the moment Logic had the virtual mixer inside its program, I sold my Euphonix." Taieb's main PC handles sequencing and percussion, PC2 handles strings, PC3 brass and PC4 woodwinds. Most of his 200GB of samples come from Vienna Symphonic Library and the Miroslav Vitous collections. Outboard gear in his studio includes Lexicon 300 and TC M5000 reverbs, Dbx 165 and 160XT and Aphex 661 compressors, Focusrite ISA 115HD voice channel and an SPL Vitaliser enhancer, inserted into Logic using two MOTU 2408 and one 24 I/O interfaces, plus a 16 I/O analogue patchbay. According to Taieb, creating realistic orchestral demos has only become possible during the last couple of years, in particular because of streaming samplers such as Gigastudio and the VSL orchestral library. "Of course people have been making pretty good orchestral demos for a while," Taieb concedes. "But they were using technology at its limits, and this affected the compositions. When I wrote The Alchemist and the Caroline Dale track I was still doing a lot with pen and paper, but for the VanessaMae album I could for the first time input right away into my system, and hear something that's very close to the end result. In the past you could make things sound like an orchestra, but you couldn't compose like you normally

Walter Taieb's Paris studio is based around four PCs running Cubase, Sibelius and Gigastudio, with the Vienna Symphonic and Miroslav Virtous orchestral sample libraries.



would for an orchestra. It was also impossible to make very fast passages work, or to make the strings sing, and so people often simply had held chords in their demos. The VSL library has made a huge difference, even though the sounds themselves are sometimes better in other libraries. But the VSL actually contains played intervals, and so the passages from note to note sound a lot more realistic now." For Taieb, the process of handling the computer screen like score paper has simplified and accelerated the process of writing for orchestra, and improved his sample-based demos as well. "I always begin composing playing my keyboard and using a piano sound," the Frenchman explains. "Then I sketch the general harmonic, arrangement and orchestration structures on paper. Once I have them I immediately enter the notes manually into Logic, using the Score and Matrix editors. I find that entering notes this way saves a huge amount of time when preparing the score. It may not be as listenable as when you're playing things in, but there are a lot less mistakes in the score — the notes are all the right length. Software programs have tremendous problems with things like triplets, and when you play things and then quantise, the score will look like a mess. "I used Logic for Vanessa-Mae's album, but I've since switched to Cubase, which I think is better now; but at the time I didn't want to waste time learning a new program. From the first moment of working in Logic, or Cubase, I'm aware that the score will eventually end up in Sibelius, and so I make sure that it will be easy to export. First of all, I lay out the score on my screen just as an orchestral score, all the instruments exactly in the same order. The other thing I do is make extensive use of the MIDI channel splitter. I specify a musical articulation for different channels, so one channel is for legato, another for staccato, yet another vibrato, and others will be for long notes, short notes and so on. I have those different articulations in a very precise order in the splitter, so I can change the MIDI channel very quickly for each part on a note-by-note basis, and with that trigger the appropriate samples, making the demos sound more realistic. "The moment I take the MIDI splitter out, it will look like one channel, and will be very easy to export to Sibelius. I'll then later write the musical expression comments and dynamics on the print-out from Sibelius. Another advantage of this way of working is that normally, you can put a lot of your time and imagination in making your sample-demo sound realistic, and yet you'll still expect the orchestra to improve on your demo a lot. This often leads to disappointments. But if it sounds good when you're basically playing it like a robot, you can only be pleasantly surprised later on in the recording studio."

Gathering Pace Taieb's slightly unorthodox way of working appears to have made an impression, because he got the gig for Vanessa-Mae's album on the basis of the demos he made of his bolero and tango. "These demos were very elaborate," recalls Taieb," and I had written in every level change and expression, trying to make file:///H|/SOS%2002-05/Composing%20&%20Recording%20With%20Vanessa-Mae.htm (4 of 7)9/22/2005 8:21:29 AM


them sound as real as possible. If I want a more realistic-sounding demo, I will play the lead line in, or have someone overdub it. "Later, when Vanessa came to rehearse these two compositions, we had three days to spare in Paris, and we came up with the idea of doing a kind of Santana track, with lots of percussion, some minimalist orchestration, and the violin on top. It was a strange concept, and I did a minute-long demo for it with some percussion loops and orchestral sounds played on my keyboards. It was sent to London, and I was told that if I finished the piece before the recording dates with the Royal Philharmonic Orchestra, which were a week later, it would make the album. "This meant that I had to do the composing and orchestration in a couple of days. This is where the Vanessa-Mae and Walter work with the computer was extremely useful. It was Taieb at Abbey Road during like using a word processor — I could copy and the recording of Choreography with the Royal paste parts across and so on. When working on the Philharmonic Orchestra. track, which was called 'Tribal Gathering', I didn't bother with making the demo very realistic, since noone would have time to listen to it, and it was only for my own reference. What's nice with my way of working is that after entering the notes, I just press Play and listen to it. I could test the orchestration really fast, and try different combinations in the orchestra, things you wouldn't attempt during a recording session with an entire orchestra waiting for you. This definitely helped me being more creative. "In the past you learned orchestration from studying other people's work, and so you know that the combination of this instrument with that instrument will sound beautiful. Learning orchestration takes a lifetime, because there are always new tricks to consider. But when you work with computers you can experiment much more. I think that as long as you still have the same state of mind as when you were working with pencil and paper, computers are a great tool for orchestration. But if you do it without having a clue about orchestration, never have studied it, quality will go down. This is what happened with a lot of Hollywood scores, where composers have great sounds to work with and do MIDI versions of the orchestra, but end up creating a lot of middle-of-the-road music, without melody or real composing talent."

Transfer Troubles After Taieb finished his composing work his copyist took over, double-checking the Sibelius score, extracting individual instrument parts and transposing some of them (Taieb prepares his conductor's score un-transposed), creating a conductor's score, and printing everything out. Armed with a suitcase full of file:///H|/SOS%2002-05/Composing%20&%20Recording%20With%20Vanessa-Mae.htm (5 of 7)9/22/2005 8:21:29 AM


paper and a Firewire drive with his Logic demos, Taieb travelled to Abbey Road Studios for rehearsals and recording with the Royal Philharmonic. Pro Tools was to be the recording medium for all the orchestral sessions, which necessitated transferring Taieb's Logic files via the dreaded OMF. "There were several reasons for the transfer. First I simply wanted to give the engineer an idea of the feel of the demos and the balance of the orchestra I had in mind. But we ended up not even listening to them! Then, all the orchestral percussion you hear on the final tracks came from my Gigasampler, so we needed these parts. Third, 'Tribal Gathering' was written for a 90-piece orchestra, and we only had 60 musicians, so I had to find a way of making it sound like 90. Obviously, the orchestra played to a click track, to be in sync with my material. "With regards to the OMF format, I was surprised to hear at the three different top studios where the album was recorded, SARM, Metropolis and Abbey Road, that they all had problems with it. Somehow the audio tracks never end up in the right place. I found a way of transferring data from Logic to Pro Tools, which involved first transforming all my track files into one big audio file, and then transferring this from Logic to Nuendo, using OMF. They are compatible and so this was easy. In Nuendo I added an empty audio file to each track, starting all empty audio files at beat one of bar one, and then merging these empty tracks with the audio tracks. In this way all audio files begin at the same point and fill in all the blank spaces as well. If you export this from Nuendo as individual WAV files and then import them into Pro Tools, all the tracks are in sync."

Classical Sounds, Pop Mixing Taieb conducted the orchestra himself for his three tracks, and was also involved in the overdubbing, first of Vanessa-Mae's parts at SARM West studios. "We had the luxury of a four-day lockout. After a few days she felt confident enough to improvise some cool violin parts on the 'Tango' and compose her lead part on 'Tribal Gathering'. I think we put about 10 microphones around her, and inserted all of them in the board so that we didn't know which microphone was on which channel. We gave all microphones marks out of 10, and ended up using a combination of a B&K 2006 and a Microtech Gefell UM92.1S, both placed about one metre above her." Back in Paris, Taieb recorded some additional percussion in his studio, and a bandoneon at Mega Studios. The bandoneon turned out to be tuned slightly flat with respect to the orchestral recordings, so the entire orchestral recording was pitched down for monitoring while the overdubs were recorded, with the bandoneon pitched up again afterwards. Mixdown took


Walter Taieb's main outboard equipment racks. From top: MOTU Digital Timepiece synchroniser, Ensoniq DP4 effects, Lexicon 300L reverb, Dbx 165 and 160XT (x2) compressors, Aphex 661 compressors (x2), SPL


Vitalizer enhancer, place at Metropolis in London, where Vanessa-Mae's Focusrite ISA 115HD violin was, says Taieb, "treated as if it was the lead input channel, TC singer in a pop record. The engineer used a mixture of M5000 reverb and expensive vintage gear like Pultec EQs and cheaper Tascam DA45 DAT equipment like the Focusrite Platinum Voice Master, recorder. and plug-ins like the Sony GML emulation EQ. We also sent the violin through a quarter-inch analogue tape machine to make it sound warmer."

Anticipating that the record company would want to make the completed album sound as 'pop' as possible, Taieb had made sure he recorded all the sections of the orchestra together during the Abbey Road sessions. "On all the other tracks the orchestra was recorded in sections, but I wanted to have the sound of an orchestra playing together, so it would also sound good with just two overhead microphones, and so that the engineer would not be able to make it sound too rock & roll. As it turned out, the final mix still uses too much of the direct microphones for my taste. You need a mixture of room and direct mics in my opinion. Perhaps I'm just too much of a purist." Taieb has since returned to club music, acting as DJ in Paris's largest dance clubs, and being DJ of choice for Marilyn Manson, Audioslave and Black Eyed Peas. Purist he may be, but he clearly still understands both sides of the classical/pop divide. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Crosstalk: Your Communications

In this article:

High Voltage Controllable Urges

Crosstalk: Your Communications Readers Writes Published in SOS February 2005 Print article : Close window

People

High Voltage I am a keen music-technology enthusiast, and have been reading the review of the Chandler TG Channel in January's Sound On Sound. I should also explain that my main job is director of an electrical safety testing company. My eyes were drawn to the picture of the inside of the TG Channel's power supply [reproduced below]. As far as I can tell from the photo, the mains connector looks like it is a two-pin device (live and neutral) and there is no provision for earthing on the power supply. This is potentially very serious as it would mean that the case would almost certainly become live if one of those hand-soldered joints gave way and a mains wire came adrift! I would be interested in knowing whether the product has any CE approval or has gone through any kind of type testing before sale. It may well be that the design of the production model has changed from that given to the magazine for review. I should point out that I have no commercial interest here, nor am I connected to any official or enforcement body. I am passing these comments to you in the hope that they will be useful to you. Tim James Technical Editor Hugh Robjohns replies: In the nicest possible way, I'm very pleased to say you are mistaken here! It's a little difficult to see from the angle at the which the picture was taken, but the mains inlet (at the bottom right) is a file:///H|/SOS%2002-05/Crosstalk%20%20Your%20Communications.htm (1 of 3)9/22/2005 8:21:33 AM


standard three-pin IEC socket, complete with earth terminal. The incoming mains earth is connected directly to the power-supply case adjacent to the mains socket, via a loop of insulated wire. Admittedly, the wire used to do this is coloured blue, which is confusing, but rest assured, there is a decent safety earth connection to the chassis.

Controllable Urges I was dead chuffed to open the latest Sound On Sound to find reviews of the Behringer BCF/BCA2000 hardware MIDI controllers. After reading a generally very informative review, I was disappointed, because for me, the most important factors determining the overall usefulness of the product weren't covered in enough detail. But this appears to be the case in nearly all previous reviews of control surfaces, such as the Mackie Control, Radikal Technologies SAC2.2 and so on. What I'm talking about is just how well a controller functions within a particular DAW, perhaps looking at three of the big names, say Logic, Cubase and/or Sonar. By this I mean how functions such as control assignments and fader-bank switching are implemented, and how the control surface follows on-screen selection. And while I know many people will immediately say that this isn't really the domain of the hardware manufacturers but down to how the DAW developers provide support, and thus not within the scope of the review, I'd respond by saying that this integration is the single most important thing in determining how useful your £157 or £999 control surface will be when used day-to-day. I've found further coverage of the Behringer controllers on the Internet, but there were still plenty of questions left unanswered. I know that web-site-based reviews have more potential 'column inches' to devote to reviews, and no deadlines, but I feel that any review of the products really need to address the usability questions in depth in order to be of any real value to those of us who are looking for guidance. SOS Forum Post Managing Editor Matt Bell replies: By their very nature, generic controllers are designed to be used with as many pieces of recording software as possible, and this immediately makes writing a definitive review difficult. At one extreme, testing a given controller with every possible software and hardware combination is beyond the resources of any magazine, especially within an acceptable timescale. At the other extreme, using it with just one sequencer setup, on one computer platform, is likely to give an unbalanced picture of the device's functionality. Moreover, like most of its readership, SOS's reviewers are musicians and engineers, and like most of us, they tend to use one or two major recording and file:///H|/SOS%2002-05/Crosstalk%20%20Your%20Communications.htm (2 of 3)9/22/2005 8:21:33 AM


sequencing applications which they know well. This makes it hard to write reviews of generic controllers with all of the major sequencing and recording packages. You'd either have someone commenting in detail on what operation is like with one or two packages — the ones with which they're most familar — and being very sketchy about the others, or you'd have to pass the review unit around multiple reviewers to cover the behaviour with all applications. This would immediately turn the piece into a talking shop, as reviewers obviously don't all necessarily share the same opinion. For this reason, as well as logistical complexity (shunting the review unit between three or four reviewers instead of one would add considerably to the time it takes for a review to be printed), we've avoided taking the latter approach. Instead, we compromise, and find reviewers who can comment on a generic controller's behaviour in detail with as many packages as possible. In practical terms, this usually amounts to three or four of the major applications; so in the case of Mackie Control, we tested with Cubase and Nuendo, Samplitude, Digital Performer, and Pro Tools (Logic was not supported at the time of the review). With the Radikal 2.2SAC, we used Logic, Cubase and Pro Tools, and with the B-Controls, Derek Johnson used Reason, Pro Tools LE, Sonar and Cubase. To date, this is the most far-reaching approach we can come up with while still allowing authors to comment about the behaviour of the controller under review with software they know well, and while still remaining within the bounds of what's practical each month. SOS contributor Derek Johnson adds: As Matt has said, I, and other reviewers, do try products such as the B-Controls on a wide range of software, and if tests turn up something nasty or inconvenient, we'll flag it. The B-Controls, however, were well behaved, although I understand some users are now reporting niggles with some flavours of Cubase. Generally, I feel that the BControls are great value, and offer remarkable control in a wide range of circumstances. And from my experience, the company are doing what they can in terms of drivers and emulation from their end: a number of updates appeared while I was conducting my review. I hope that a combination of this natural upgrade process and lobbying on the part of users can iron out any remaining problems between these controllers and the software we're using with them. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


file:///H|/SOS%2002-05/Crosstalk%20%20Your%20Communications.htm (3 of 3)9/22/2005 8:21:33 AM

Modelling Classic Hardware In Software

In this article:

Mighty Joe Bryan Audio Unit Compatibility Of Plates & Latency Interested Third Parties Against Piracy New Vintages Analogue Parts In The Digital Age To The Future

Modelling Classic Hardware In Software Joe Bryan of Universal Audio Published in SOS February 2005 Print article : Close window

People : Industry/Music Biz

Since Bill Putnam's sons founded UA anew, the company have become well known for software emulations of old gear, as well as for their hardware recreations of vintage classics. We find out how they go about it... Paul White

It's no secret that Universal Audio, as we know them today, were founded by Jim and Bill Putnam Jr, sons of the the highly respected engineer Bill Putnam Sr, inventor of the 1176 FET compressor, and studio engineer to such greats as Frank Sinatra, Nat King Cole, Ray Charles, Duke Ellington and Ella Fitzgerald (for more on Bill's life and work, see the article on studio pioneers in SOS October 2004, or surf to: www.soundonsound.com/sos/oct04/articles/ rocketscience.htm). A large part of the modern Universal Audio's business is in building authentic reproductions of vintage analogue studio processors. Joe Bryan in Universal Audio's travelling studio/demonstration When Bill Sr passed away in 1989, the brothers bus. were clearing the family home and came across their father's design notebooks, which documented every technical problem he'd ever solved. That was the trigger for the brothers to set up Universal Audio and to recreate these designs using the original components and circuits. The company name came about because one of Bill Putnam Sr's companies back in the '50s was also called Universal Audio. Today they have an impressive complex in Santa Cruz, California.

At the heart of the Universal Audio range are the LA2A and 1176 compressor/ file:///H|/SOS%2002-05/Modelling%20Classic%20Hardware%20In%20Software.htm (1 of 11)9/22/2005 8:21:37 AM


limiters. The LA2A 'leveling amplifier' as it was orignally called, uses valve circuitry, hand-wired components and has just three controls on the front panels. The design dates back to the 1960s and uses a photocell as the gain-reduction element, illuminated by an electro-luminescent panel. The design was originally patented by Jim Lawrence, and the unit was manufactured by Teletronix in Pasadena, California, but the rights to the design passed to Bill when he bought the company that owned them in the mid-'60s. Three different versions of the LA2A were produced before it was discontinued around 1969. The 1176 'limiting amplifier' was designed by Bill Putnam himself, and he experimented with Field Effect Transistors (FETs) as gain-control elements. In effect, they can be used as voltage-controlled resistors. After a great deal of research, Bill Putnam found a simple solution using the FET as the bottom leg in a voltage divider prior to a preamp stage. The output transformer used in this design is also critical, as it uses additional windings to provide feedback, making it an integral part of the output amplifier. This had to be accurately recreated for the modern version of the 1176. Brad Plunkett subsequently revisited the circuit in the early 1970s with a view to making it less noisy, which is where the 1176LN came about. Apparently further design improvements followed, resulting in at least 13 revisions of the 1176, though aficionados still claim that the D and E 'black-face' versions sound the most authentic.

Mighty Joe Bryan After putting the old analogue designs back into production, the brothers Putnam turned their attention to creating equally accurate-sounding digital replicas of the vintage gear in the form of plug-ins that run on their UAD1 DSP card. I was interested to know how a company with such a solid background in analogue, high-end signal processing got into the DSP card, software plug-in market, and the man to provide those answers was Joe Bryan. Prior to joining Universal Audio, Joe worked with Ensoniq, and then as part of the Korg R&D group working on the Wavestation and OASYS projects. Joe claims that he was electrocuted under his mum's piano at the age of four and from then on, he knew he had to work with electronics! According to Joe, one of the challenges of the Korg OASYS project was to get sufficient DSP power to make it respond and play like a Universal's 2001 recreation of the classic musical instrument, and not like a 1176 compressor. computer. He clearly has a lot of respect for what was achieved back then, even though the only part of OASYS that came to market was a DSP card. "For particular reasons, the OASYS keyboard was never realised — it never shipped, so I left and went to join a startup company in silicon valley that was doing very high-performance DSP targeted at desktop computers. I reasoned file:///H|/SOS%2002-05/Modelling%20Classic%20Hardware%20In%20Software.htm (2 of 11)9/22/2005 8:21:37 AM


that if this worked, then there would be high-performance DSP available on everybody's desktop, but the company went out of business and the project was abandoned. So I picked up the technology and brought along some of the engineers I'd been working with back in 1998, and approached several different companies — and got some interesting responses! I can't mention names, but some formerly high-profile plug-in companies thought that what we were proposing was impossible. "We had been working with Jonathan Able and Bill Putnam Jr for years, so we approached them at Universal Audio and they recognised immediately the potential of teaming up high-performance DSP with quality algorithms of the type that the Kind Of Loud guys were doing at the time. That was the start of the UAD1. The concept was to use as much DSP as we could possibly get, then focus on the quality of the algorithms." Of course, native platforms are getting more powerful every few months. What can the UAD1 offer that isn't already available in the native plug-in domain? "One difference is that the performance available from our DSP is typically an order of magnitude higher than for the DSPs commonly used in other products. It is a proprietary DSP that was designed for a totally different The LA2A plug-in for the UAD1 card. purpose — graphics — and the company spent around $60,000,000 developing it. That level of investment is not something that can can be supported by the audio industry — technology is rarely a limiting factor, just the economics of the application of the technology. It's very different from the Motorola 56K-series DSPs, because ours is floating-point and much faster. We wanted to achieve the fastest performance for the minimum cost, which we did, and then we used that as a basis for developing the highest-quality plug-ins without worrying about running out of power." I imagine the greatest challenge was to measure the characteristics of those classic pieces of hardware and then to create algorithms to reproduce their sound with sufficient accuracy. Also, there must be some artifacts that are crucial to the sound, while others may be difficult to model and have no significant effect on the sound. How do you know which parameters are important? "It was a very big deal, though removing the DSP performance constraints helped us considerably. As you start to approach the ideal scenario, the performance requirements go shooting up. Fortunately we have Dr Dave Berners and Dr Jonathan Able, who are world-class algorithm designers. Focusing on the parameters that had to be modelled was a big challenge, as there's always a corner of the performance space that can't be reached by the model. For example, if you're going to run a +32dB level into an 1176 and turn everything file:///H|/SOS%2002-05/Modelling%20Classic%20Hardware%20In%20Software.htm (3 of 11)9/22/2005 8:21:37 AM


up, the plug-in isn't going to replicate that! That would saturate all the amps and transformers and be almost impossible to model — but then nobody would use an 1176 like that in practice. Also, do you model line noise and hiss? Is that desirable? We decided no, because there's no advantage of having that." I guess that you do have to look at dynamic distortion mechanisms and also the precise attack and release curves, as well as the quirky way ratio changes with level. "Those are very important attributes, and are modelled as closely as we can possibly get them. Basically it is within the tolerance of unit-to-unit variation. The 1176 is a very difficult process to model because of its feedback structure, and the attack time is actually less than one sample. In the mathematical realm where everything is quantised to single samples, you have to play very complicated mathematical games to replicate that. It doesn't actually require up-sampling because of the shape of the knee, and the DSP guys are very good with these things — they have all these tricks they can play using a predictive approach that basically calculates all the possible paths and then selects one as the starting point for the next sample. The LA2A even has a photon value in there that goes between the L panel and the cadmium cell." That sounds almost like quantum computing! "The multi-stage release is actually a quantum process, and that's what they model. Different products have different modelling requirements — the Pultec EQ is a very clean passive EQ with make-up gain and plenty of headroom, so it is a fairly linear process. What is challenging is that because it is a passive EQ, in order to match the amplitude and phase responses, you have to up-sample. There's a few other plug-ins that up-sample too, so we look at each case individually to see what that algorithm needs. By contrast, some of the algorithms, such as the Fairchild, turned out to be relative easy to do after the experience we'd gained doing the LA2A and the 1176."

Audio Unit Compatibility The introduction of Apple's Logic v7 with its Audio Units checking utility tripped up a lot of plug-in designers, but the UA plug-ins now seem fully compatible. I wondered if this caused Joe and the UA team any problems. "We have been working with Apple very closely and they've made changes to their AU engine based on feedback we've given them, though we still had to modify our existing code to work with Logic 7. Initially they were somewhat handsoff with us, because they thought native plug-ins were the right way to go, but once they heard what we were doing with our algorithms, they realised that they would benefit from the added quality we could bring. However there are some remaining issues related to the way audio data is buffered in Logic. That causes problems if you run a UAD1 plug-in in a live track at the same time as having UAD1 plug-ins in playback tracks. You should really only use one or the other — not both together. In fact, I had a conversation about this with the former Emagic file:///H|/SOS%2002-05/Modelling%20Classic%20Hardware%20In%20Software.htm (4 of 11)9/22/2005 8:21:37 AM


guys yesterday — they've worked with us in the past, and I'm hopeful that this will be resolved in the future."

Of Plates & Latency You've just announced a vintage plate reverb plug-in, Plate 140, which relies on modelling, despite the success of convolution-based reverbs. What are the advantages of doing it your way? "There are two or three benefits. We did sample the reverbs and measure the impulses as a base-line reference, then after we had come up with our modelling algorithms, we took them back into the studio and made changes to get the sound as close as possible to the original plates. A plate is essentially linear, though there are some non-linear aspects to it, but I think the advantage of modelling is that the processing load on the DSP is a fraction of what it is with a convolution reverb. Also, you can adjust the parameters of a model, whereas an impulse response is essentially fixed. You can't adjust the damping on an impulse reverb; you have to switch to a different impulse. Also the group delay of an impulse reverb is huge compared to our modelling algorithm, so there's a big benefit to the user in doing it our way." A plate reverb has a very fast buildup of complexity, whereas a typical digital reverb algorithm starts with an FIR — in effect, a multi-tap delay line — to create the early reflections, then recirculating comb and all-pass filters are used to build the late decay. How do you achieve an adequately fast build-up of density with modelling? "That was probably one of the biggest challenges — getting the basic tonality of the plate was more straightforward. The UAD1 DSP card. Getting the late field smoothness was also relatively straightforward, but achieving that initial build-up, as you say, was the biggest challenge, and we had to play around with several different reverb architectures to realise that. The filtering also has a tremendous impact on the sound, and we did a combination of listening and analytical optimisation to get the most out of the DSP we had available. With an impulse-convolution approach, you have to process every sample, but you don't actually need every sample, just the samples that matter. So we used an analytical approach to optimise our algorithms to reduce the number of taps and feedback networks that were required. That meant we could allocate the remaining DSP to the more difficult parts of the task. For example, within a plate, high frequencies travel much faster than low frequencies, so we could do the early build-up just on the higher frequencies using an analytical approach. Jonathan Able is just brilliant on file:///H|/SOS%2002-05/Modelling%20Classic%20Hardware%20In%20Software.htm (5 of 11)9/22/2005 8:21:37 AM


reverbs and also, Will Shanks spent days listening to all the different plates so we could get the right voicings for various applications. As plates are pretty linear within a broad range, any transducer-based non-linearities, and other nonlinearities, can be emulated using some of the standard processes that we've added on top of the reverb." DSP-powered plug-ins generally have latency issues which double whatever latency you would normally have with the application running native plug-ins. Is this also true of the UAD1? "Yes, that's the way it works. As we look at it, there are three primary applications for DSP within the digital audio environment — tracking, where you're monitoring in real time, mixing and mastering. We have focused initially on mixing, and we're looking more closely at mastering. Frankly, tracking is something that personally I don't see being low enough latency if you're really serious about it, even with native plug-ins. If you want to process as you track, use an analogue processor. Even if you just need to hear some vocal EQ, compression and reverb while tracking, those can be supplied using outboard hardware relatively easily. I like using our Nigel guitar processor, but for me, nothing really beats an amp."

Interested Third Parties Unlike the competing TC Powercore system, UA have not opened up the UAD1 platform to third-party plug-in developers. I asked Joe if that was a deliberate strategy, and if so, why. "Now it is, but initially we were planning to announce some third-party relationships. It's not something we're ruling out entirely, but at the moment we don't see the need to open the platform up, as it seems to serve our customers best to focus on giving them everything we can and not diverting our development resources towards third-party support. Also, from a piracy point of view, if there's a native plug-in available now, it makes no sense to port it to our platform, as there's a high likelihood it's already been cracked. That means the only plug-ins that would be worth considering are ones that aren't out yet or ones that are only available on other DSP platforms that have robust protection. It would also mean converting the code for our floating-point processor, though it's easier to go from fixed-point to floating-point than the other way around."

Against Piracy I suppose one of the attractions of the DSP platform is that you can protect your software from piracy more effectively. "When I first designed the card, we hadn't figured out how we were going to do the copy protection, but we knew that we would be doing it, so we added the hardware that gave us the capabilities. Then, when we released the Cambridge EQ plug-in, which was the first plug-in we sold separately, we turned on the

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protection. Software piracy is a function of how easy it is to steal versus how easy it is to buy. We make it as easy to buy and as hard to steal as possible. Some of the copy-protection schemes I've had to deal with are so unwieldy that it's easier to use the crack! Our authorisation has nothing to do with the computer — it's keyed to the UAD1 card, so if you want to load the software onto multiple machines and then move the hardware around, you can do that. In fact, I do that myself — I have a Magma chassis that I move between my desktop machines and my laptop. "It also makes things a lot easier when you come to change computers. It's a challenge system where you have to register only once, then pay using a credit card or whatever. The other important thing to bear in mind is that if The new Plate 140 plate reverb emulation for the UAD1. a software company goes out of business and you're using a system where you have to reauthorise every time you change computers or swap out a hard drive, you're stuck with a plug-in that no longer works. Our authorisation is a simple text file that you can copy from computer to computer, or you could even write it down! We don't want copy protection to penalise the existing customer." Talking of portability, have you considered putting the UAD1 in an external box connected by Firewire, for use with laptops? "Laptops and compact desktop computers are the fastest-growing parts of the marketplace, so this is important for us. We are a small company, and so far we've been focusing on producing the plug-ins, but we've also been looking at Firewire for a long time. I'm on the AES Firewire committee and have been watching all the protocols, and frankly, it's not really 'soup' yet! All the Firewire solutions have problems, whether it is stability, latency or capacity — which is a limiting factor, particularly at high sample rates. It's just getting to the point where it's do-able. The whole Firewire thing is, shall we say, under active consideration."

New Vintages Your main focus so far has been on recreating vintage equipment, and the new plate plug-in follows in that direction, but what other vintage areas are left for you to conquer? My own view is that nobody has got tape echos and tape-flanging emulations quite right yet, but does this interest you? "Absolutely. The focus we've had up to now has been on more linear processes such as EQ. Compressors are non-linear, but still lend themselves to being physically modelled quite easily. Things like saturation distortion are also fairly straightforward, but taking it to the next level, where you have distortion with no time dependent, such as in transformers, it gets more complicated.



"Going to tape modelling is also a big step, because nobody really figured out how tape worked. The best models that were available were two dimensional, but it's actually a threedimensional process. To do that correctly requires a tremendous amount of effort, and it is something we're working on, but we don't want to do a compromise solution." One thing that nobody seems to have modelled is the way the tape recording process changes phase — and few put the record equalisation before the saturation process. If you put a square wave into a tape machine, you don't get anything like a Testing a dual-channel 2-1176 square wave out of it, so perhaps that needs compressor/limiter at Universal to be modelled first before the time Audio's California factory. processes, wow and flutter and feedback filtering are applied? What's more, the saturation elements need to be in the feedback loop so the sound deteriorates properly as it is recirculated. "And many people do take the simple approach of just recirculating via a filter, perhaps adding some pitch modulation, but I think it's getting there now, and more accurate modelling will eventually happen. The other thing that's appealing to us is going into non-linear equalisers that colour the sound. This coloration comes from several different effects. Obviously you have amplitude and phase characteristics, but adding the right type of saturation is the next dimension to get into. One of the things we're trying to do first is cover all the bases so that the card we're offering gives the user a broad palette of processing they can use immediately, and then provide extra options within those areas. Frankly, a lot of the things we build is stuff that we want to have for ourselves." So, is there any area that you have a strong personal interest in? "Yeah, my background is in synthesis, and I still think the next generation of synthesizer that performs like a real instrument has yet to be built. I've built acoustic instruments, I play acoustic instruments; I know what they sound like. Synthesizers are convenient and they can make some interesting sounds, but even modelling a subtractive synth properly is a challenge. Going to the next step, where physical modelling is happening effectively still hasn't been done properly. The last synthesis project I was involved in was the Korg OASYS, which was a great-sounding product, and I think that was as close as anybody has come to it yet." From what I can see, part of the problem is that the market tends to be customer driven, yet customers don't often know what is possible, and most don't know that they want something until they hear it.



"It's a big problem, and companies have to have a lot of balls and put a lot of money into what they believe in. Mr Kakehashi, the head of Roland, is one of my heroes, and one of things he's done — and I'd like to get to the level where we can emulate that — is that he has products that make a lot of money in the mass market which he uses to fund research, and he knows they may never make a profit. They're fun to do, they push the envelope, and the benefits find their way into other products later. The guitar-synth stuff he's done took a lot of investment, and probably only benefits a handful of people, but it would be wonderful to be able to do something like that. We're a growing company, so hopefully we'll get to the point where we can have a fun projects group."

Analogue Parts In The Digital Age Though Joe lives and breathes DSP circuitry and code, he is also very passionate about great analogue equipment. Universal Audio are still best known for building faithful replicas of the LA2A and 1176 compressors, and I wanted to know how they manage to source the original components, especially things like the photocells and electro-luminescent light sources used in the LA2A. Obviously, this isn't something they have to worry about when designing algorithms, as digital bits are not known for becoming obsolete! "It's a big challenge, because the cadmium photo-resistors and the electroluminescent panel are both absolutely critical to the design. We went to the manufacturers after measuring the exact performance and response time of the originals, and they asked us how many we wanted. When we told them how few it was, they almost fell about laughing — it just wouldn't have been cost-effective to make them to that spec in that quantity! However, after we did the algorithms for the LA2A, we learned for the first time how they actually worked and what gave them their characteristic sound, and once we'd done the physical modelling, we were able to go back to the manufacturer and tell them exactly what we needed. This was a more realistic spec that they could meet in our quantities, and the consistency was far better too. Now it's not a big voodoo thing, where nobody dare change any part of the process for fear of ruining the batch. "The electro-luminescent panels are also problematic, and we've just gone through the process of qualifying a new vendor to meet the demand that we have. Again, being able to draw on the DSP technology to find out what parameters really matter has helped enormously."

To The Future Where else do you go with the hardware now that you've replicated the vintage analogue products that you wanted to do? "Mic preamps are like shoes — you can never have enough of them. Selling mic preamps is like selling shoes to people with foot fetishes! The same is true of microphones, and there's no mic or preamp that is perfect for all applications. You have two things that interact with each other, so our new mic preamps were designed to complement the existing product line. You can either have an



extremely clean and transparent preamp, or you can have one that interacts with the mic to provide more of a coloration to the sound. There are so many dimensions in the physical space of the signal itself — it's not just phase response and frequency response. One number that I hate is the specification that quotes total harmonic distortion plus noise. Frankly it's the level of distortion at each harmonic that matters, not the overall figure. In fact, some of the old tube gear was specified with a distortion of one percent, and it sounded great. It's the way the harmonics come up with a tube — it's the second, then the third, and there's pretty much nothing else. With a modern op amp, the level of the harmonics may be much less, but those harmonics go all the way up to the Nyquist frequency and beyond." And the more feedback you use in an opamp circuit to keep the steady-state distortion low, the more dynamic distortion you get under real-world signal conditions. The performance might look great on a test sine wave, but still sound bad with a real signal. "That has a huge, huge impact and in hi-fi circles, the trend is towards minimum feedback. Feedback has benefits in lowering the output impedance and lowering the distortion, but an amplifier does not have infinite bandwidth, and so any corrective signal is necessarily going to be wrong to some extent. The circuit may An 1176 (below) and a 6176 measure absolutely perfectly with a sine wave, channel strip (above) undergoing but a sine wave is the laziest signal an amplifier an extended 'soak-test' at the UA factory. Several LA2As and a 2can handle. You put a complex signal in there 1176 can also be seen in the and everything goes to hell, because of the background. temporal relationship of the harmonics within the signal, the crest factor and the phase response — all these things add up to affect the sound, and there's a lot of subjective and anecdotal evidence to point at what matters, but we don't really know exactly what it is for sure. "There are two approaches, the first being the analytical approach where you say that if the numbers are right, then it must sound good. Then there's the subjective approach, where you start from a basic functioning circuit and then modify it until it sounds good. It's like tuning an instrument. We tune all our stuff in our studio, but then there's a difference between listening to things in isolation and listening to them in a mix. If it's a microphone preamp, we try to select the applications that we are going to be using it for, pick the mics, run them into the unit and then see how they sound. We try to get as broad a palette of applications as we can where it sounds good in the mix. Rich Williams, who designed the 4110 and 8110 units, had the designs for the microphone preamps in the studio for months, and he had trim pots on his prototypes that he'd tune as he mixed a session. There's no other way to do it. We do all the analytical stuff as well, but we have to get the



sound right. Ultimately, the sound is all that matters." Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Music Software Updates

Music Software Updates Leader Published in SOS February 2005 Print article : Close window

People : Industry/Music Biz

As you probably know by now, I am for the main part a big fan of software-based studios, not least because they are compact and have total recall of absolutely everything. If you miss knobs and faders, you can always buy a control surface, but they're by no means essential. However, while the studio itself may be compact, the way in which software products are sold tends to be far less so. Why is it that so many software packages come in boxes containing a thin manual, a couple of CDs or DVDs — and a huge amount of empty air? I've just installed Native Instruments Komplete, and I have to tell you that whilst it is a fabulous bundle, the box is simply huge. OK, the Logic Pro 7 box is almost as big, but at least that's completely filled with paper manuals and needs to be the size that it is. Other packages are smaller but still far bigger than their contents, and the amount of shelf space you lose to these things can be enormous. It goes against the grain to chuck away the boxes and if you do, you may later find that some vital registration code has been thrown away with them, so please guys, give us slimline boxes. Whatever your marketing department might tell you about large boxes making things look like they are worth more, we all know how big CDs and DVDs really are! If packaging is an inconvenience, the need to constantly update music software is an even greater one. If companies like Apple and Norton can offer automatic update facilities, then why can't the music-software companies? Essentially, these systems log on at regular intervals, check for updates and download them automatically if they are required. From my standpoint as an end user, this is far more friendly than having to visit all the music software web sites every few days to check whether or not updates are available and whether they are applicable to your particular operating platform. An automated approach could also help to slow the tide of pirated software, as the updaters would only be available to registered users, and the software company in question would be able to check the legitimacy of the installed software during the process that decides which updates are necessary. Better still, if the updater can read what music software is in our system, it could also warn of known compatibility issues. The more paranoid may see this as bit too much 'Big Brother' (in the sense of Orwell's 1984, not the TV show!), but the legitimate user should reap only benefits. After all, the main downside of computer-based studios is that you can easily end up spending far more time maintaining them than you ever do creating music in them. Finally, the spring-cleaning theme popularised by so many TV shows has now spread to my own studio computer, where I'm actively removing software I don't use in the hope that this will give me a simpler and more stable system to work with. Even though I'm fortunate enough to be allowed to leave a lot of review software installed, I still tend to remove it after a review, unless I'm planning to continue using it. This is in direct contrast to the people who fill their drives with as many pirated plug-ins as they possibly can, use hardly file:///H|/SOS%2002-05/Music%20Software%20Updates.htm (1 of 2)9/22/2005 8:21:41 AM

Music Software Updates

any of them and then end up with a system that's unstable. The same thought processes can also be applied to sample CDs. There's no denying that there are some truly excellent sound libraries out there these days, but there does come a point where you realise that life is too short to allow you to audition everything you have, in which case why bother keeping it? A simple, straightforward studio that does exactly what you need is generally far more useable than one that does everything — and in my own case, I've discovered that less really can be more. Paul White Editor In Chief Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Sounding Off: Vintage Synths

In this article:

About The Author

Sounding Off: Vintage Synths Andy Foster Published in SOS February 2005 Print article : Close window

People : Sounding Off

Why are we still in love with vintage synths when all they do is break our hearts? Andy Foster

I suppose over the years I have been bitten by the vintage analogue synth bug as much as anyone. As a young lad growing up in the late '80s and early '90s, my musical diet was a combination of rave and Vince Clarke's analogue rumblings with a smattering of Jean-Michel Jarre thrown in for good measure. Of course it was not long before I yearned for the machines that created the sounds on the records I listened to. A long stream of classics passed through my fingers — ARP 2600s, a brace of Minimoogs, the entire RSF Kobol line and even things like the Synton Syrinx and Steiner-Parker Synthacon. Each synthesizer would quench my lust for a while before I would have to hunt down something else. However, after a while an ugly problem reared its head; that of keeping these machines working fully. It quickly became apparent that these items of desire could cost a lot of money to maintain on top of their initial purchase price. Good technicians are few and far between these days, and the cost even of fault finding can be £50. I soon realised that each machine was basically a time bomb. The RSF Kobol is a prime example, stuffed full, as it is, of hard-to-find SSM chips. These are the chips that filled the early massproduced synths such as the Prophet 5. What people never tell you when you're buying these old synthesizers is that these chips are very hard to get hold of these days. Even if you do find a replacement it will more than likely not be new, but will have come from a non-repairable unit. The trouble is that these

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About The Author Andy Foster has been hooked on synthesis since he discovered his school's MS10 at


the age of 15. He is chips were only designed to last around 20 years or currently working on so, and many are now reaching that age. There's two albums and also another problem — due to the way these chips getting to grips with work, the more you use them the more they fade some of the larger away, so even your replacement chip could be on its modular synths. last legs. It's not only the chips that are the problem; knobs, sliders and even displays are all becoming scarce. I used to live in fear that one of the displays in my Oberheim Xpander would fail — and it had three!

I spent some time trying to work out what it would take to maintain my collection of vintage synths, but the answer I eventually came up with was not the one I imagined. I don't want to start the old 'analogue versus digital' debate, but I do want to say that there are alternatives to these ageing beasts. I have cleared out almost all of my vintage synths; a lone Jupiter 8 and a customised Oberheim OBSX are now my only repair worries. I have replaced everything else with either virtual-analogue synths like the Clavia Nord 2 or new analogue units. The surprising thing is that my friends, the general CD-buying type, cannot tell the difference between a real Minimoog and a good copy of one, either live or on a recording. I know some will say that the sound is just part of the appeal of these units, and that the tactile control surfaces and the look and style are also a big reason for their popularity. I would have to say that synths like the Alesis Andromeda and the range of large-format modular systems that are now available, such as the MOTM, Oakley and Modcan ranges, look just as good and give you just as much control as the old Prophet 5 or indeed a big Moog System 55 ever did, and without the long-term repair bill. I find it unbelievable that anyone would still want to purchase something like the Memorymoog — a dire unit that didn't even work properly when it was first made. I recall reading an interview in the mid-'80s with a band who said that their two Memorymoogs kept breaking down on them — and that was when the machine was two years old, not 20! I can appreciate that some people will never change their opinions of vintage synths, and I cannot blame them for that, but is it really worth hanging on to these machines any more when there are good alternatives that will continue to work for the next 20 years, and also be easy to maintain? There seems to be a lot of prejudice against the newer-generation synthesizers at times. I can only liken this to when you were a child and you refused to eat something; you would say you didn't like it, even though you had never tried it. Maybe it's about time we tried some of the new things, and left these ageing machines to the collectors and museums of the world. Published in SOS February 2005






Studio SOS

In this article:

Preliminary Listening & Subwoofer Placement Killing Undesirable Sonic Reflections Curing An NS10 Hum Tidying Up The Wiring Aniff's Comments Last-minute Soldering Tips

Studio SOS Aniff Akinola Published in SOS February 2005 Print article : Close window

People : Studio SOS

Acoustic problems and mains hum were making Aniff's studio difficult to use, so the SOS team stepped in to help. Paul White

Aniff Akinola has been working in music for around 20 years and has enjoyed a certain amount of commercial success with his productions, including collaborations with artists such as the late Kirsty McColl, Steve Hillage, Ian Brown, The Beloved, A Guy Called Gerald, Urban Cookie and Backyard Dog. During that time he's run two independent record labels, been signed as an artist to four major labels, and been involved in the creation of three top-20 hits. He's currently signed as a writer to BMG Publishing. As if this weren't enough, Aniff is currently working part time at Manchester's Sound Control music store. He called us in because he wasn't happy with the monitoring environment in his studio, located on the top floor of his Old Trafford home. Prior to our arrival, he'd bought a pair of Genelec 8040As and a monstrous Genelec 7070 subwoofer — along with a pack of Primacoustic room-treatment foam. The subwoofer was serious overkill for file:///H|/SOS%2002-05/Studio%20SOS.htm (1 of 9)9/22/2005 8:21:51 AM

Here you can see Aniff's setup as Paul & Hugh found it on their arrival, and this configuration was causing several problems. For a start, the Genelec monitors were too far apart, compromising the stereo imaging at the monitoring position. The central monitoring position wasn't helping the bass response, and the subwoofer (to the left of the desk in the lower picture) also needed a little repositioning.

Studio SOS

the size of room, but it was a bargain secondhand buy, and Genelec equipment is always easy to set up in almost any combination. Our job was to help him optimise his monitoring environment, sort out his subwoofer placement and balance, and fit the foam where it would do most good. When we arrived, Aniff had set up his Genelecs on stands behind a pair of Quiklok studio desks on which he'd placed his Yamaha NS10s plus a pair of flatscreen monitors for use with his Apple Logic/G5 studio setup. His G5 is fitted with both TC Powercore and Universal Audio UAD1 DSP expanders and he uses a Rode Classic 2 as his main recording mic. To keep the Genelecs clear of the NS10s, he'd had to set them very wide apart, which was less than ideal from a stereo-imaging point of view, and he'd also reported some hum problems with his NS10s, which were driven from an old Quad 405 MkII amplifier. He'd parked the subwoofer almost exactly halfway along the left wall of the room, and the desks were a couple of feet from the front wall to make space for a large MDF box that housed his Mac G5 and a PC. This arrangement resulted in the listening position being almost exactly in the centre of the room.

Monitoring tests proved frustrating working directly from the outputs of Aniff's MOTU 828 MkII audio interface, so Hugh quickly plugged up Aniff's new Samson C*Control monitoring controller to make switching and level control easier. The next step was to make sure that the Genelec monitors were suitably set up, and this involved experimenting with the subwoofer positioning, level, and phase.

The wiring behind the desk resembled the pit of snakes from an Indiana Jones film, and a rather lost-looking Samson C*Control was perched hopefully on the desk with no cables yet attached to it. A MOTU 828 MkII handled the audio interfacing, and a Focusrite ISA430 front end was connected to the S/PDIF input of the MOTU when required. One of the few hardware synths to survive Aniff's transition to software was a Roland JV2080 fitted with a number of expansion cards, though Aniff also has an Akai MPC4000 that he likes very much.

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Studio SOS

Preliminary Listening & Subwoofer Placement As usual, our first job (after attending to the tea and chocolate biscuits) was to listen to the monitoring system exactly as it was. Initially, this was done using the MOTU 828 MkII's volume control, but this proved frustrating to control, and the resolution fell off at low volume settings. So Hugh quickly set about patching in the Samson C*Control instead, with the MOTU monitoring outputs set at full level to achieve the best D-A resolution, and with Aniff's CD player hooked up to one of the two-track inputs so that he could play back CDs without re-patching.

Following a quick trip to B&Q, Paul constructed an acoustic panel using a sheet of MDF, some adhesive, and elements of a Primacoustic foam room-treatment kit.

As expected, the stereo imaging was less than ideal, because of the wide speaker spacing, the very reflective side walls, and the relatively large amount of clutter in front of and between the monitors on the desk — in fact it was so bad that we could detect very little difference when we pressed the C*Control's Mono button! The bass end also seemed rather too loud and lumpy. We reckoned that the imaging would be improved by moving the speakers and by putting up the Primacoustic foam Aniff had bought, so Hugh set about adjusting the subwoofer while Aniff and I made a trip to the local B&Q to buy some fixings for the MDF panels on which we'd decided to stick the foam. As there was a fair bit of gluing and lifting to do, Aniff conscripted his friend Ben from Manchester's branch of Sound Control to help out. If the subwoofer isn't in the right position in the room and relative to the listener, the level of differently pitched bass notes can vary enormously, so Hugh and Ben used the old trick of lifting the sub into the place where the engineer's chair normally goes, then crawling around on the floor alongside the walls listening for the spots where the bass sounded most even. There were a couple of places which seemed quite good, and one turned out to be not far from where the sub had been located in the first place — it just needed moving forward by a foot or


Studio SOS

so, which got it away from the 'dead centre of the wall' position, which is usually less than optimal. Having located the ideal place, the subwoofer was dragged back over and then the DIP switches that control the relative phase were checked to get the best crossover response with the 8040s. Finally a listening test was performed to establish the correct bass Aniff held a vacuum cleaner to catch the dust while Paul drilled holes for mounting the balance. It was necessary to drop the acoustic-foam panel he'd just put together. sub level by around 2dB from its Once the hole was drilled, it was fitted with a original setting to get a subjectively wall plug to accept the mounting screw. A picture bracket was then screwed to the back sensible bass balance, though this of the panel so that it could be fitted to the wasn't easy to judge, as Aniff's wall screw's head. collection of predominantly drum & bass music seemed to vary rather a lot when it came to bass content. Hugh managed to find some more 'known quantity' pop albums and we eventually agreed on a setting that was a happy medium.

Killing Undesirable Sonic Reflections Aniff decided to fix the Primacoustic foam to MDF panels so that they could be moved around if necessary and when redecorating the room. As luck would have it, we were able to make up a pattern of foam blocks that exactly fitted the 2 x 4foot MDF panels he had bought. Using a panel adhesive and a mastic gun, we fixed up four identical panels to be used in pairs on the side walls to tighten up the imaging and kill flutter echoes. A further panel was used directly behind the monitors as a horizontal absorber, with four thick foam blocks overhanging the panel at either end. The remaining foam blocks were fixed to a sheet of MDF and placed on the rear wall, though some additional rear-wall absorption would be beneficial. Aniff said that he planned to get a settee to put at the back of the room, so that should also help kill reflections. In addition, we discussed the idea of hanging a couple of single duvets across the angled part of the ceiling coming down to the back wall, which Aniff said he'd try to do later on.


Studio SOS

In an attempt to even out the room's bass response, some foam bass traps were also installed directly into the room's corners using adhesive.

We fixed the panels to walls using brass picture-hanging plates screwed to the backs of the panels. These simply slotted over screws in the walls that were fixed using plastic wall plugs in the traditional way. The two triangular foam bass traps that were included in the Primacoustic kit were glued between the front wall and the ceiling. Fortunately, bass trapping wasn't a serious issue in this room, as there was a panelled chimney breast in the front right corner, a door enclosing the hot water system in the front left corner, a wooden entrance door (back left) and a window (back right), all of which soaked up bass quite effectively and in a well-balanced way. Once we had the panels in place, we experimented with the speaker positioning. Although the side panels had certainly improved the stereo imaging a little, it was still way below that which we expected from these Genelec speakers — and the NS10s sitting on the desk shelf produced a far better stereo image. Although it is generally better to place speakers on dedicated stands, rather than on the desk, in this case we found that with the Genelecs on stands behind the desk the clutter on the desk shelves (NS10s and LCD monitors) wrecked the imaging completely. So, we eventually ended up with the NS10s directly to the sides of the two computer screens, then the Genelecs placed on the top shelves next to file:///H|/SOS%2002-05/Studio%20SOS.htm (5 of 9)9/22/2005 8:21:51 AM

Aniff's nest of adaptors and mains plugs were cluttering up the floor underneath the

Studio SOS

main desk, so the SOS team improvised a these, so the stands were no longer long suspended mains connection strip using required. Fortunately, the Isopod a piece of wood and several six-way stands that are supplied with the new plugboards. This cleared the cables off the Genelec speakers are very good at floor, making access behind the desk (for rehelping to isolate vibrations to and from patching or cleaning) much easier. Because the six-way plugboards were connected to a the desk, and the stereo imaging was single surge-protected plugboard in a 'star' now perfectly good. Aniff was worried configuration, this setup also had the effect that the NS10s might not sound right, of reducing the level of mains hum in the as they were now standing vertically, system. but this is the way they are actually intended to be used. Mounting them on their sides is a convenience issue (to help lower their profile when mounted on a big desk meterbridge, obscuring the main soffit-mounted monitors) and narrows the sweet spot very noticeably.

We have noticed before that there can be a bass suck-out near the centre of a square room and, because the two desks were a fair distance from the wall, Aniff's chair was dangerously close to this dead spot. In fact, this centre-room 'hole' became very noticeable as you stood up or sat down. The solution was to remove the computer box from behind the desk to allow us to move the two desks closer to the wall. This improved things considerably, and the monitoring balance, especially at the bass end, was far more acceptable in this new location. Ideally, we should have used balanced cables between the 828 MkII and the C*Control, and also between the C*Control and the Genelecs. However, Aniff didn't have any spare balanced cables, so we had to make do with unbalanced cables between the MOTU and Samson boxes as a temporary solution, with a promise that he'd change them as soon as he could. The main speaker output from the C*Control used balanced cables to the Genelec subwoofer and from there back to the 8040s.

Curing An NS10 Hum Having sorted out the main monitoring, we turned our attention to the humming NS10s. There was, indeed, a constant low-level hum, and we first assumed this was a ground-loop problem, as the Quad amp's inputs are unbalanced. However, disconnecting the inputs didn't change the hum level or quality at all, and neither did shorting the tip and ring of the phono connectors — if shorting the inputs had killed the hum, then the hum may have been getting in at the inputs.


The finished setup, with the monitor controller plugged up, the desk and monitors repositioned, all the acoustic treatment in place, and the hoover safely back under the stairs...

Studio SOS

This all pointed to a problem in the amp itself — possibly faulty power-supply smoothing capacitors — and would require the services of a good repair centre. However, there was also a second possibility. The amp had been modified at some stage to install an input level control, and it was possible that the wiring for this was picking up a stray hum field within the amplifier. Either way, Quad Electroacoustics maintain a very good repair service, so we suggested Aniff contact them to have the amp overhauled. The speaker leads to the NS10s were also enormously long, which was pointless and could only reduce the sound quality, so Hugh quickly shortened the cables and we then hooked up the Quad amp inputs fed from the Monitor B output of the C*Control. The Samson has facilities to balance the levels of the two main pairs of speaker outputs, so that switching from the Genelecs to the NS10s maintained nominally the same level.

Tidying Up The Wiring While we were at it, we decided to have a go at tidying up Aniff's mains wiring. To get the mains connections off the floor, we bought some new six-way mains distribution blocks and taped these to narrow planks of wood that would fit between the upright legs at the rear of the Quicklok desks. Ordinary brass hooks were screwed into the ends of these, then they were simply hooked onto Velcro straps fastened around the rear desk legs, just above one of the horizontal supports to stop them sliding down. We did this on both desks, which gave us four sets of six-way distribution boards around 18 inches from the ground, and we also had enough Velcro straps left to help tidy up the cabling. To minimise the risk of ground loops and to offer the maximum protection for the equipment, we ran all the distribution boards back to a central surgeprotected distribution board plugged into a wall socket. This meant the whole setup was running from a single power point with the mains cabling in a 'star' configuration. Running everything from a single wall socket wasn't a problem, as the overall power consumption was fairly small. Aniff had labelled all his mains adaptors


Aniff's Comments "I still can't get over how different the room sounds just when I walk in, never mind when I'm playing music! I have now got the quilts for the rear ceiling and will put them up in the next couple of days. I've also put in a bypass switch for the sub for when I'm working late at night. Now it's time to go and practice my soldering!"

Studio SOS

and plugs, so it didn't take long to get everything up and running again. The result was a hum-free system (with the exception of the low-level hum on the NS10s from the suspect power amplifier) and, although the wiring was still far from being a work of art, at least it meant you could get a vacuum cleaner under the desks without encountering piles of cables! As we'd made a fair bit of mess sorting out his system, I tested this by vacuuming the room in readiness for our photos! By the way, please don't tell my wife that I know how to do this... With the desks shuffled back, the C*Control repositioned centrally below the dual LCD screens, and the surplus cabling removed, the whole setup was better both ergonomically and sonically. The stereo imaging was actually very good, and the level of bass end seemed appropriate.

Last-minute Soldering Tips Our final task was to try to teach Aniff to solder using a propane-powered soldering iron we'd picked up at B&Q. I demonstrated the basics by replacing a stereo jack plug on his headphones, and I pointed out that the iron shouldn't be used to carry solder to the job. Instead, you heat the two parts to be joined separately and feed solder onto them to tin them. Then, the two tinned parts are brought together, heated again and more solder fed onto the Before he and Hugh headed for home, Paul joint as required, until the solder flows freely across the joint. At this point, the used a propane-powered soldering iron he'd picked up at B&Q that morning to show Aniff heat is removed and the joint held still how to go about basic soldering. until the solder has cooled. Aniff tried this on a couple of scraps of mains wire and seemed to be getting the hang of it by the time we were ready to go, though he had to be reminded to keep the heat on the joint and not dab at it with the iron. It only takes a few minutes of practice to learn to solder, and it is a hugely valuable skill, especially if you need to repair cables or make up custom connections. As the sun set over the sleepy suburbs of Manchester, Hugh punched the route home into his trusty GPS navigator, and we headed for that perpetual car park that is the M6! Published in SOS February 2005


Studio SOS




The History Of Roland

In this article:


1992 Part 4: 1992-1997 The JV Series & Its Published in SOS February 2005 Expansion Options Milestone: The Compact Print article : Close window Drum System People : Industry/Music Biz 1993 Milestone: Atelier Organs 1994 1995 Milestone: The VG8 The recession of the early '90s was difficult for many 1996 companies, but not Roland. Diversifying still further Milestone: The VS880 Digital from their roots as a maker of synthesizers and Studio Workstation organs, they emerged stronger, encompassing almost 1997 every area of hi-tech musical instrument manufacture. Milestone: V-Drums Epilogue Gordon Reid

In the first three parts of this history, I described how Roland emerged from Ace Electronics and Hammond International in 1972, and grew in just two decades to be one of the leading hi-tech music manufacturers in the world. In this part, we'll see how the company sidestepped a dramatic slowdown in the MI industry, and evolved into a corporation very different from the maker of analogue synths and organ rhythm accompaniment boxes it had been in the early '70s.

1992 MAJOR PRODUCT LAUNCHES AMPS & MIXERS M160 MkII line mixer. MA7 & MA20 micro monitors.

BOSS PRODUCTS DR550 MkII Dr Rhythm. DR660 Dr Rhythm. DS330 Dr Synth GM/GS synth module. ME6 guitar multi-effects. ME6B bass multi-effects. ME10 guitar multi-effects. PQ3B bass parametric EQ.

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At the start of the 1990s, the world was dipping into a severe recession. You may remember this as an era of plummeting house prices and 'negative equity', but manufacturers remember it just as keenly as the end of the boom of the 1980s, and the start of a tough period in which it became much harder to find customers and sales. Some of the smaller companies in our industry did not survive, while other struggled on, hoping that there would eventually be light at the end of the tunnel. Those with the wherewithal to do so diversified, looking for new markets and new customers. This is the path that Roland followed in


PQ50 parametric EQ.

DIGITAL RECORDERS DM80 multitrack disk recorder system.

HOME/ACCOMPANIMENT PRODUCTS RA90 real-time arranger. GUITAR SYNTHS GR1.

MASTER KEYBOARDS A30 keyboard controller. AX1 keyboard controller. PC150 keyboard controller. PC200 MkII keyboard controller.

PIANOS EP9.

1992... Up until this point, there was a branch of digital music production that still lay beyond Roland's reach. This was multitrack hard disk recording which, long before Pro Tools and even cheaper computer-based systems, was still the domain of high-end systems such as the AMS Audiofile and the DAR Soundstation. But by 1992, the cost of random-access digital recording and signal processing was dropping rapidly. Hard disk recorders and editors had started the decade costing well in excess of £30,000, dropped below £20,000 in 1991, hit £10,000 in 1992 and carried on plummeting to the amazing prices that we enjoy today.

HP2900G. HP3800. HP5700. HP7700 Micro Grand. KR650 Intelligent Piano.

RHYTHM PRODUCTS FD7 hi-hat control pedal. KD7 kick trigger unit. MDS7 drum stand. PD7 drum pad. R70 Human Rhythm Composer. R8 MkII Human Rhythm Composer. TD7 sound module.

SAMPLERS DJ70 workstation. SP700 sample player.

SEQUENCERS MC50 MkII Micro Composer. MT200 music player.

SOUND CANVASES SC7 GM module.

Digital recording technology was changing and becoming cheaper at a frantic rate in the early '90s, rendering many products obsolete overnight. The DM80, at around £10,000 for a fully equipped system, was a third the price of many existing systems of a similar spec. Just four years later, Roland's own similarly equipped VS880 cost less than a sixth that of the DM80.

Once prices started to collapse in 1991 or thereabouts, companies such as Akai, Korg and Roland decided that they wanted to get involved. Korg did so while prices were still in excess of £20,000, releasing a fully featured post-production system, whereas Roland waited a few more months, and released a considerably smaller and less powerful system aimed more at the music industry.

SC33 Sound Canvas. SC155 Sound Canvas.


The DM80 appeared in two flavours: a


four-track version with a single 100MB drive, and an eight-track version with SYNTHS & HI-TECH double the storage and a sample-rate CM300 GS sound module. converter on one of the digital inputs. There was also a software package for CM500 GS/LA Sound Module. the Macintosh, Track Manager. In JV30 synth. principle, this allowed you to cascade JV80 synth. four system units for 32 channels of synchronised playback. The DM80 also JW50 GM workstation. offered a limited form of automation for its two-band EQs, level and pan controls. SCC1 GS/GM soundcard.

Unfortunately, the eight-track version was merely two four-track systems, with little interaction between them, and you were limited to just 128 The JV80, the first of the massively 'Takes' per system. Furthermore, the successful JV synth range. DM80 offered no waveform display, limited crossfade facilities, slow and cumbersome menus, no video output for an external monitor, TRS jacks instead of XLR connectors, and was capable of storing just 18 track-minutes per system! Oh yes... and with insufficient storage for multiple projects, you had to back up everything via the SCSI ports provided on the back panel. Given these limitations, it's hardly surprising that the DM80 was not a commercial success. What's more, the UK retail price of £7455 for an eight-track recorder with just 200MB of storage (or nearly £10,000 if you purchased the essential DM80F fader unit and DM80R remote control options) quickly started to look hugely overpriced and under-specified. So why — with more than 90 hard disk recorders already on the market in 1992 — was the DM80 important? In itself, it wasn't, but this was the 'missing link' between Roland's 'S'-series samplers and the 'VS' recorders that form the backbone of the company's range in the 21st century. Were it not for the DM80, the big news of 1992 might have been the replacement of the 'D' and 'U' series synths with a new family of synthesizers, modules and workstations. These were the 'JV' series, the first two of which were the JV80 and its little brother the JV30. Despite the similarity of their designations, these were very different synths. The JV80 was the 'pro' model, with a powerful, 28-voice engine that let you combine up to four unique tones for each voice. It sported a semi-weighted, pressuresensitive keyboard, eight editing sliders, an external slot to take 2MB PCM Expansion Cards, another slot for a RAM card, an internal slot for one of Roland's new range of 8MB PCM Expansion Boards, and a competent set of master keyboard functions. Consequently, it was an immediate hit with Roland aficionados, and it proved to be a fine workhorse. Sure, it was not quite the workstation that many had anticipated but, as the first of the illustrious line of JV synths (see the box opposite), it deserves its place in history. Strangely, the company did launch a workstation in 1992, but it's largely lost to



history, because, as they had with the W30 a couple of years before, Roland had again judged the market incorrectly. The JW50 was a good idea in principle, taking the 16-track sequencer technology from the MC50 These days, there are countless similar Micro Composer, and adding 30 music devices, but despite innovations such as the styles derived from the MV30 'Studio novel 'scratch wheel' controller, the DJ70 M' Music Production System. proved to be a little ahead of its time. Unfortunately, its sound source was a 24-voice 'one tone per patch' GS/GM (General MIDI) engine, which placed the JW50 firmly in the domestic, pub and karaoke class of keyboards, not the professional end of the industry. In retrospect, the JW50 was decent enough, with a large screen, the ability to edit and store user patches, a good selection of edit faders, and the ability to record and replay some surprisingly sophisticated sequences, but it failed to take the world by storm. The JV30, meanwhile, despite the JV name, was very much an entry-level synth. Another GM synth, it could produce only one tone per voice, and offered just 24 voices. It lost the semi-weighting and pressure-sensitivity of the JV80 keyboard, as well as all the other goodies mentioned above. Roland seemed to be finding no end of ways to repackage their GM engine. There were two more Sound Canvases released in 1992, two more 'CM' desktop music system (DTMS) products to replace the aging CM32s and CM64, a GM/GS sound card, and the SC7 sound module. More repackaging was evident in the SP700 Sample Player (which was, in essence, an S770 without the sampling capability, but with the ability — at last — to read Akai S1000 disks) and the DJ70 Sampling Workstation, which drew both its sampling and editing capabilities from the S770. Designed to make high-quality sampling straightforward for DJs, the DJ70 offered a number of innovations (not all of which, such as the 'scratch wheel', worked quite as well as hoped) but was rather limited in other areas: there was no built-in storage, and a maximum of just 4MB of RAM. Again, the idea was good, but the implementation was wide of the mark. So why does the DJ70 deserve The Boss DR660 was a popular 'starter' anything other than a passing drum machine, and remained in the mention? Well, ask yourself — in these catalogue for over half a decade. days of cheap, ubiquitous Grooveboxes — what was the first product to combine real-time sampling, real-time phrase sequencing, and a device that lets you 'scratch' samples? Yes... Roland had done it again, and foreseen the way the world was going. It's just that the DJ70 arrived a little too early to be technologically mature or cost-effective.



Another idea that came just a little soon was that of the digital grand piano. The HP7700 was a beautiful instrument, with an excellent hammer action and a speaker system designed to imitate the soundfield generated by a 'real' grand. Nevertheless, this technology, too, was immature, and the idea would only come properly to fruition a few years later, when improved DSP power would make Roland's KR-series digital grand pianos a reality. Elsewhere, Boss were encroaching on traditional Roland territory, with the DR500 MkII and DR660 drum machines. Together with Roland's own R8 MkII and R70 Human Rhythm Composer, these were doing brisk business in the MIDI percussion markets. But the biggest news of 1992 was probably the demise of Roland's five-year-old PD11, PD21 and PD31 percussion system, and its replacement with the world's first complete MIDI drum kits, the TBD and TBE Compact Drum Systems. These arrived complete with a sound-generating 'brain' and pads that could act as cymbals or hi-hats as well as kick drums, snares and tom-toms. Suddenly, a drummer could set up an electronic system, plug it in, and play a full kit without needing acoustic drums or metalwork to fill the gaps (see the box above).

The JV Series & Its Expansion Options There were three families in the JV series and its direct descendent, the XP series. The low-cost instruments were all GS compatible, and there were eventually to be four of these, the JV30 (pictured below), JV35, JV50 and the XP10. Although the specifications improved as each model was superseded by the next, these were all limited to a single tone per patch, and they lacked the sonic depth of their more expensive siblings. They also lacked the slots for PCM Expansion Cards, data cards, and Expansion Boards, although the JV35 and JV50 (and, strangely, the JV90) let you install either the VEGS1 or VEJV1 board. As its name implies, the first of these was a GS sound card that added a further 28 voices of polyphony and a further 16 multitimbral parts. The second added 28 voices, eight multitimbral parts, and was in many ways a JV80 on a card. The second family was more powerful, and included the original JV80, its successor the JV90, the JV880 rackmount and, finally, the JV1000 workstation. These shared the JV80 sound engine, although it's The basic, entry-level JV30. worth noting that their internal ROMs changed as each model replaced the previous. All accepted 2MB PCM Expansion Cards and one 8MB Expansion Board, as well as commercially produced data (patch) cards that used the waves in the internal ROM and Expansion Cards as the basis of new sounds. The third family replaced the second in its entirety, and expanded hugely upon it. Polyphony was increased to 64 voices and, depending upon model, between one and eight Expansion Boards could be accommodated. Furthermore, the standard chorus/reverb effects structure was improved to include an EFX (expansion effect) that you could use in a primitive multitimbral way to add effects to some patches, but not others. The models in this family included the JV1080 (which was the only



one to retain the earlier PCM Expansion Card and data card slots), the XP30, XP50, XP60, XP80, the JV2080 (which offered three EFX units) and the JV1010, which proved to be the final model in the JV range. At the time the JV80 was launched, there were just two of the 2MB PCM waveform cards available, Piano Selection and Guitar & Brass. Roland promised that others would soon become available, and a further six were added before the series was discontinued in 1996. The additional cards included drums, accordions, orchestral sounds, baroque instruments, and country/folk/bluegrass, with some of the sounds they produced remaining unique to this day. Shortly after the JV appeared, Roland started to produce the 8MB Expansion Boards that, for many years, formed the backbone of its synthesizers. There were 21 of these, with the complete listing running as follows: SR JV80 (8MB) EXPANSION BOARDS SR JV8001: Pop. SR JV8002: Orchestral. SR JV8003 : Piano. SR JV8004 : Vintage Synth. SR JV8005 : World. SR JV8006 : Dance. SR JV8007 : Super Sound Set. SR JV8008 : Keyboards of the 60s & 70s. SR JV8009 : Session. SR JV8010 : Bass & Drums.

The themed card-based SR JV sound expansion libraries (of which just a few of the earliest are shown here) were compatible with nearly all of Roland's sample-andsynthesis based products of the '90s, and were hugely successful.

SR JV8011 : Techno. SR JV8012 : Hip-Hop Collection. SR JV8013 : Vocal Collection. SR JV8014 : World Collection: Asia. SR JV8015 : Special FX Collection. SR JV8016 : Orchestral 2. SR JV8017 : Country Collection. SR JV8018: Latin Collection. SR JV8019: House. SR JV8098 : Experience 2. SR JV8099 : Experience.

Of these, SR JV8007 (Super Sound Set) is perhaps the most interesting, because it contains most of the waveforms and patches found in the 2MB Card series. However, for collectors, SR JV8098 (Experience 2) and SR JV8099 (Experience) boards are the most intriguing. These were bundled with some synths, and contained a selection of waves and patches from the other boards. Such was the popularity of the JV/XP sound sets that Roland released some of their sounds on CD-ROMs for their samplers, and later based a complete series



of rackmount sound modules — the Sound Expansion Series — upon them. Most recently, many of these waveforms have reappeared on the SRX expansion board series for Roland's newer XV-series synths.

Milestone: The Compact Drum System By 1992, electronic drums had been available for more than a decade, but there had never been a complete MIDI drum system that you could assemble in the form of a conventional kit, play like a conventional kit, and which — when used appropriately — would sound similar to a conventional kit. Roland changed that when they launched the Compact Drum Systems. There was just one type of pad, the PD7, but this had dual sensors — on the pad and the rim — and you could configure as many of these as you wanted to provide the snares, toms, hi-hats and cymbals that you required. Meanwhile, kick-drum duties were fulfilled by the dedicated KD7 pedal/trigger units, and hi-hat control was provided by an FD7 pedal. The heart of the system was the TD7, a halfrack sound generator with hundreds of samples, 32 programmable kits, a built-in phrase sequencer, and nine stereo inputs for the trigger signals generated by the pads.

Like their earlier electronic drums, Roland's Compact Drum Systems looked great, but the pads didn't play well enough to be considered attractive to drummers, and they were unsuccessful.

With everything mounted on the dedicated MDS7 stand and the pads hooked into the TD7, you were ready to play. Unfortunately, the tactile response of the PD7 pads was not good enough for serious players, and after an initial surge of excitement, sales collapsed, and the outlook for electronic kits seemed bleak. Even the introduction of a range of pad sizes — the 7.5-inch PD7 was later supplemented by the 8.5-inch PD5 and 10-inch PD9 — failed to generate much interest. Eventually, some players opted to add one or more PD7s and a TD7 to their acoustic kits, allowing them to augment the sounds they obtained from conventional drums and percussion. Roland didn't really reap the rewards of their efforts until four years later, when they released a much improved set of drum products.

1993 In contrast to the previous year, 1993 was quiet by Roland's standards, although a slew of new products continued to appear. There were more JV synths, more Boss effects, more amplifiers and mixers, more E-series accompaniment keyboards (the most expensive of which adopted the JV80 sound engine), and new digital pianos, rhythm products and sequencers. There were also the first of



MAJOR PRODUCT LAUNCHES AMPS, SPEAKERS & MIXERS MA7/MA20/CS30 micro monitors. RM106 & RM166 mixers. SRA50 stereo power amplifier. SV50R & SV100DR 'Solid Valve' guitar amps.

BOSS PRODUCTS DR5 Dr Rhythm Section. FZ2 Hyper Fuzz. HM3 Hyper Metal. MEX expandable multi-effects. SD2 dual overdrive. SE70 effects processor.

EFFECTS SDE330 Dimensional Space Delay.

a range of processors that adopted elements of Roland's RSS (sound space) technology, and used this to generate effects that belied their modest price tags. But, 1993 should be remembered as the year in which Roland finally, finally released their first true synth workstation, the JV1000. First to appear, however, was the JV880, a rackmount version of the JV80 keyboard synth. This shared its predecessor's 28-voice polyphony, eightpart multitimbrality, dual effects, and its selection of PCM samples. Likewise, it also accepted a single SR JV80 Expansion Board, and had slots for a PCM Expansion Card and a Data card. Replacing the U220, the JV880 became one of Roland's most successful products of the early '90s.

SRV330 Dimensional Space Reverb.

HOME/ACCOMPANIMENT PRODUCTS E16/E56/E86 'Intelligent Synthesizers'.

Even better, and still

PIANOS EP7II digital piano. P55 piano module.

RHYTHM PRODUCTS AT4 acoustic trigger interface. SPD11 total percussion pad.

SEQUENCERS MT120 music player.

SOUND CANVASES SC55 MkII Sound Canvas. SD35 MIDI player.

The JD990 was Roland's best sample-and-synthesis sound module ever in 1993, but Roland were on such a roll in the early '90s that it was surpassed just one year later by the JV1080.

SYNTHS & HI-TECH JD990 Super JD synth module. JV1000 music workstation. JV880 synth module. SMPU/AT Super MPU interface. SRC2 sample rate converter.


respected to this day, was the JD990. While similar to the 'JV' series, this was really an enhanced JD800 in a 2U rackmount, to the extent that you could program many of its parameters directly from the JD800's front panel. Soundwise, it was a bit of a hybrid, being


TA10 IBM-AT soundcard.

based on the JD800's ROM, but with almost double the number of waveforms courtesy of much of the JV80's innards. It also had the JV's ability to host a SR JV80 Expansion Board, Expansion Cards, and Data cards. There were many other enhancements. For example, the filters were not merely resonant but selfoscillated, the large LCD provided graphics that showed exactly what was happening to whatever you were editing, there was oscillator 'sync' and ring modulation, and there were Structures that let you combine Tones in different ways. There were also lots of real-time control options, and no fewer than eight outputs. There was even an attempt to add limited multitimbrality to the effects in a Performance. When it appeared, the JD990 was described by Roland as their 'best synth yet' and, for once, the claim was justified. Shortly after the JV880 and JD990, the JV1000 arrived. Its synth engine was another hybrid, with the standard 28-note 'JV' polyphony, but an enhanced set of waves including a better set of piano samples in its ROM, and far more patch and preset memory than the JV80. Of course, the JV1000 offered the now standard facility to install an 8MB Expansion Board, and offered slots for 2MB Expansion cards and Data cards. There was also space for a VEJV1 or VEGS1 board, either of which doubled the synth's polyphony to 56 voices, which was a respectable total in 1993. Furthermore, as I would discover a few years later, the JV1000 would also (with a few glitches) host the superb VERD1 piano board, even though Roland claimed that it was not designed to do so, and that it would not function. Nevertheless, the thing that made the JV1000 special was the inclusion of its 'Super MRC' sequencer; the first that Roland had installed in one of their top-ofthe-range synthesizers. This offered all the power of the 'MC'-series sequencers, with eight tracks, each capable or recording and replaying all 16 MIDI channels, plus a dedicated rhythm track for pattern-based programming. What's more, it was compatible with Standard MIDI Files (SMF), so you could write sequences to, or play them from, all manner of other machines. The sequencer even had its own screen and a second MIDI output, so the JV1000 was capable of transmitting no fewer than 32 MIDI channels to the outside world. There were other goodies, too. For example, you could automate real-time parameters such as Level and Pan, and the big JV even included a synchroniser so that you could use it with other recording systems. All this was hosted by a super-sleek instrument based on Roland's excellent velocity- and aftertouch-sensitive 76-note keyboard, which featured eight assignable edit sliders, and which offered numerous master keyboard functions, Yet, despite its length, the JV1000 was surprisingly light and portable; indeed, an ideal 'live' instrument. If there was a limitation, it lay with the 3.5-inch floppy disk drive, which served only the sequencer. This meant that, to save patches and performances, you had to record the data first as a SysEx file, and then write this to floppy disk. Actually, there was a second limitation... rather than adopting the more powerful effects section introduced on the JD990, the JV1000 retained the JV80's limited chorus/reverb effects structure. But these were minor niggles. It was fair to say that — five years after their competitors — Roland had finally



arrived in the workstation marketplace.

Milestone: Atelier Organs Roland founder Ikutaro Kakehashi admits that he views the organ as the "most complete musical instrument". Despite this, or maybe because of it, the first of Roland's home organs, the Atelier AT70, was a strange hybrid of ideas. For example, its lower manual was 76-notes wide, was velocitysensitive, and responded to a pianostyle damper pedal. Unlike traditional organs, the Atelier also offered many of the rhythm and accompaniment features found on Roland's E-series accompaniment keyboards, and it even accepted Roland's existing Style cards. In addition, it included a 40,000-note sequencer. However, it didn't The first Roland Atelier organ, the AT70. incorporate some expected organ facilities such as drawbars, and its sound-editing capabilities were decidedly limited. Nonetheless, there was no mistaking that the complete package was a large home organ, and it certainly packed a punch: a 240W punch, to be precise. The AT70 (which cost a hair under £10,000) and its more affordable little brother the AT50 may not have been the most successful of Roland's instruments, but they were to spawn a dynasty that would become highly respected in its field. The initial range was fleshed out by the AT30 and the huge AT90 in 1996, and then the AT80 in 1997. There were no fewer than four new models in 1999 (the AT20R, AT60R, AT80R, and AT90R) followed by the introduction of the current range two years later. There are now seven of these: the AT10S, AT20S, AT60S, AT80S and the mighty 256-voice AT90S models that appeared in 2001; plus the AT15 and the baby of the range, the single-manual AT5 launched the following year.

1994 I think that it was in 1994 or thereabouts that some people started to feel that Roland were no longer at the forefront of synthesizer design and manufacture. They were right... although the company were making very popular, commercially successful synths, they had begun to recycle ideas. Each year, although the specifications of their sample-and-synthesis synths improved, with more sounds, higher polyphony, more effects, and so on, the underlying technology and synth engines remained very similar (and have done so for all the company's S&S synths until the present day). Roland were hardly alone in this — similar developments had occurred at Yamaha by 1994, and would occur at Korg a year later, so you could argue that all three arrived at well-rounded implementations of file:///H|/SOS%2002-05/The%20History%20Of%20Roland.htm (10 of 25)9/22/2005 8:21:57 AM


MAJOR PRODUCT LAUNCHES BOSS PRODUCTS PS3 pitch-shifter/delay.

S&S synthesis that didn't require fundamental changes. But this did mean that there was less to distinguish the sample-based synths that Roland produced as time went on.

RV3 reverb/delay. RV70 stereo reverb.

EFFECTS RE800 digital echo. SDX330 Dimensional Expander.

HOME/ACCOMPANIMENT PRODUCTS E66 'Intelligent Synthesizer'.

GUITAR SYNTHS GR09.

ORGANS Atelier AT50. Atelier AT70.

PIANOS RD500.

RHYTHM PRODUCTS KD5 kick trigger unit. PD5 pad. PD9 pad. TD5 percussion sound module.

SAMPLERS MS1 digital sampler. S760 digital sampler. DA400 D-A converter for S760. OP760 1 option board for S760.

SOUND CANVASES SC50. SC88.

SYNTHS & HI-TECH JV1080 Super JV sound module. JV35 synth. JV50 synth. JV90 synth. MCR8 MIDI controller.


Consider the three keyboard synths launched in 1994. None of them was bad, but none of them was groundbreaking or exciting, either. The JV35 was merely a GS synth that replaced the JV30. Sure, it had more memory, and was able to host a VEJV1 or VEGS1 board, but it offered nothing new. Likewise, the JV50 was merely a JV35 with a sequence player. Then there was the JV90, a 76-note version of the JV80 with more memories and a larger ROM. This astonishingly light synth was an excellent gigging keyboard, but it lagged behind the instruments being produced by other manufacturers. Yet another repackaging job, the Rodgers W30 was a curious beast; it was a JV50 with an expansion board dedicated primarily to classical organ sounds. Aimed at small churches, chapels and perhaps even evangelical meetings, it certainly sounded different from the other 'JV' synths and, in retrospect, it's a shame that Roland did not pursue the idea of voicing and branding instruments for specific purposes. Much more successful was the module that replaced the JV880, the so-called 'Super JV', the JV1080. This improved hugely on Roland's existing PCM-based synthesizers, with an internal wave memory of 448 waveforms rather than a hundred and something, as found in all previous JVs and JDs. It was also the first Roland synth with a JV name that incorporated the advantages of the JD engine, including multiple Tone structures and ring modulators. In addition, it introduced a new 'Boost'


RAP10 PC soundcard & ATW10 PC software.

facility that generated overdrive, clipping, and distortion for a range of more aggressive timbres than would normally Rodgers W30. be associated with Roland's digital synths. With 40 'insert' effects, 64-voice polyphony, six outputs, and space for no fewer than four Expansion Boards, it instantly rendered every previous Roland 'samploid' synth engine obsolete. The S760 sampler was yet another new kid on the same block, updating the S770 engine with more RAM, an improved editing system, and the ability to read Akai S1000 samples (although only over SCSI, not directly from floppy disks). The S760 was much more affordable than its predecessors, but it accomplished this in part by dispensing with the digital I/O, the CRT interface, and the mouse input. Fortunately, these were available on an optional expansion board, the OP760 1. Together with the optional DA400 D-A converter, the S760 and OP760 1 combination was Roland's best sampler to date. Elsewhere, there were numerous improvements and extensions of other existing technologies. These included the SDX330, which was another in the 'S' series of effects units that used RSS technology, more guitar products, another E-series keyboard, more Sound Canvases, more pads for the Compact Drum System... basically, just more, more, and more. So where was the hot news of 1994? As it turned out, there were some significant developments and changes afoot, but not where you might have expected them. By 1994, domestic 'console' organs were becoming an endangered species; either replaced by accompaniment keyboards, or simply Roland got the multitimbral rackmount synth consigned to Granny's back room. But concept just right with the JV1080, and sold in the view of Roland founder Ikutaro them by the lorryload. Kakehashi, the home organ market had been shrinking not because people were less enthusiastic about playing this type of instrument, but because manufacturers had been failing to deliver what customers wanted. So, at that year's Frankfurt Musikmesse, Kakehashi achieved another of his dreams when Roland presented the first of its large, Atelier organs (see the box on the next page). However, he didn't do so unaided, having gathered to the Roland fold many experts in the field. For example, Carlo Lucarelli — once the Technical Manager of Farfisa, and later one of the founders of SIEL — became the President of Roland Europe following Roland's acquisition of his company. Likewise, Dennis Houlihan, now President of Roland US, was once Vicepresident of Marketing at Lowrey. Then there was Alberto Kniepkamp, a former Director of R&D at Lowrey, who became the head of Roland R&D in Chicago in the late 1980s. It's hardly surprising that, when Roland entered the organ market, they did so at the top, and far from tentatively. The year also witnessed the launch of the company's Video Canvas Desktop Video System, although this was only released in Japan. A precursor to today's



PC-based video editing systems, this was a subtle, but in retrospect important pointer toward Roland's view of the future of music technology and hi-tech. Roland and their subsidiaries also opened a number of new companies in 1994. In Europe, Roland DG founded a new division to market its products to France, Italy and the Balkan countries. Meanwhile, in the USA, a second Edirol company was established. But even bigger changes were happening in Japan... Perhaps the most fundamental of The S760, with its OP760 1 video-output these was a new method for board and optional mouse. conceiving, designing and manufacturing new products. Called the 'Producer' system, this required that the planning and design stages for each new product were carried out by just one team under the management of a single team-leader (or 'Producer'). The team acted almost as a separate company within Roland, but with access to the Group's corporate facilities. Then, when the planning stages were complete, a new team was chosen from within the Group's manufacturing base, and the product moved into production. To complement this, Roland adopted an ambitious development strategy called the '301 Project'. At the time, the company said of this project, "its goals are to develop the number one product in each category with a 30-percent increase in value compared to previous models". Today, Roland claim that the project has been very successful, and that products such as the XP80, VS1680, and V-Drum systems (of which more later in this article) exist as a consequence of adopting '301'. But not all the changes were designed to increase revenue. In 1994, Kakehashi established the Roland Foundation, with the intention of bringing music to people worldwide. The Foundation opened music schools, and funded competitions and concerts in all musical genres. One of the earliest examples of the Foundation's work was the exchange of three Japanese pieces of music and three Spanish pieces of music, all in Standard MIDI Format, between schools in Japan and Spain. These were then arranged by the other countries' students and returned to their original country for review and comments. The aim was not just to encourage musical creativity, but also to promote understanding of two cultures using music as the medium. More recently, in April 2004, the Foundation became Japan's Secretariat of the Associated Board of the Royal Schools of Music, the British organisation that provides the standard framework for music grading and examination the world over.

1995 file:///H|/SOS%2002-05/The%20History%20Of%20Roland.htm (13 of 25)9/22/2005 8:21:57 AM


MAJOR PRODUCT LAUNCHES AMPS AC100 acoustic guitar amp. BC30 Blues Cube. BC60 Blues Cube. MA100 powered speaker. ST50R & ST100DR amps.

BOSS PRODUCTS BD2 Blues Driver. CEB3 bass chorus. DD5 digital delay. GEB7 bass EQ. HR2 harmonist. LMB3 bass limiter. ME8 guitar multi-effects. MX5 stereo mixer. OD2R Turbo overdrive. ODB3 bass overdrive. TM7 guitar monitor.

DIGITAL RECORDERS AR100 announcement recorder. AR2000 announcement recorder. DM800 multitrack disk recorder.

EFFECTS AP700 EQ/feedback suppressor. GP100 digital guitar preamp & FC200 MIDI foot controller. RSS10 Sound Space Processor.

GUITAR SYNTHESIS GI-10 guitar MIDI interface. VG8 V-Guitar system.

HOME/ACCOMPANIMENT PRODUCTS E36 'Intelligent Synthesizer'. E400R 'Intelligent Synthesizer'.

By the mid-'90s, it was impossible to think of Roland as merely a synthesizer company. In truth, they never had been this alone; products such as the JC series of guitar amplifiers had long demonstrated their commitment to other types of customer. Indeed, you could say that by 1995 Roland no longer had a single main market, and that they were just as committed to small studio owners, pianists and organists as they were to rock & roll musicians. Perhaps it was for this reason that the year's offerings in its original fields of endeavour were so unremarkable. Firstly, there was the further evolution of the 'JV' series of synths and workstations into the 'XP' series. At first, there were just two of these. The XP10 replaced the short-lived JV35, adding extra memory and an arpeggiator, but retaining the GS/GM sound engine. More interesting was the XP50, which was in many ways a JV1080 with a 61note keyboard and a sequencer. However, the XP50 lost the PCM card and Data card slots that had existed since the JD800 had appeared in 1991. It also lost two of its pairs of stereo outputs, which made it less suitable for studio use. There was also the introduction of the 'Sound Expansion Series' modules which were, in truth, little more than SR JV80 expansion boards in 1U rackmount boxes. Initially, the series comprised five units, all of them designed primarily for use as preset systems, although the JV1080's SysEx commands were supported, so you could edit them using appropriate software if you wished. However, there were no onboard memories in which to store your results.

G800 'Intelligent Synthesizer'. KP24 acoustic keyboard pickup RA30 real-time arranger.


There was also an addition to the Eseries range, and two new Arranger


RA95 real-time arranger.

PIANOS FP1 digital piano. HP330. HP530.

RHYTHM PRODUCTS TD5K affordable kit.

SAMPLERS JS30 sampling workstation.

SYNTHS & HI-TECH MPU401 AT MIDI processor. XP10 multitimbral synth. XP50 music workstation.

modules, the more expensive of which (the RA95) was in effect an E86 in a box. Roland even produced a two-octave Acoustic Piano Pickup, which detected key movements and sent the information to any of the E and RA series for conversion into backing tracks. More interesting, perhaps, was the G800, which was Roland's entry into the top end of the home keyboard market. Like the E series that continued alongside it, the GS/GM-compatible G800 was an 'Intelligent Synthesizer', but it sported a 76-note keyboard, a large LCD, and more styles. Unfortunately, it lacked a sequencer, and there was little scope to edit sounds.

MDC1 dance sound module (Sound Expansion series).

The same could also be said of the DM800 which, despite being seeming MGS64 GS/GM sound module very appealing, ultimately failed to (Sound Expansion series). impress. This long, slim box was MOC1 orchestral sound module perhaps the first fully self-contained (Sound Expansion series). DAW, needing neither a computer in MSE1 strings sound module (Sound tow, nor a bank of external rackmount Expansion series). processors, nor even external hard disk MVS1 vintage keyboard sound drives if you installed the optional module (Sound Expansion series). internal drives that it supported. What's more, it offered real-time two-band EQ, time compression, pitch correction, and even mix automation. Add to this outputs for external video monitors to augment the onboard graphic LCD, a proprietary expansion buss, dual SCSI ports, a slate of MTC and LTC synchronisation options, project management and backup to DAT or MO drives, and everything looked pretty hunky-dory. But the DM800 was not a success. Perhaps this was because each SCSI port was limited to just four channels of recording/playback, or because the digital I/O was limited to S/PDIF rather than the more professional AES-EBU interface. Perhaps it was because of the limited fade and crossfading facilities, or more likely it was because editing was clunky, and the waveform display was of low resolution and difficult to use. Whatever the reason, the DM800 promised much, but survived just a handful of months before Roland announced a product that would sweep it away at a stroke. So, where was the big news in 1995? In retrospect, there was one overriding development that year, and it would soon have an impact on all manner of Roland products. It wasn't a keyboard, a synth, a sound module, or an effects unit; it was a digital signal processing technology that Roland called Composite Object Sound Modelling. COSM was a method of combining the physical modelling of various elements in



the sound generation chain, and reducing the results to a single algorithm. It first appeared in the GP100, which offered emulations of Marshalls, Fenders and of course Roland's own JC-series amplifiers, and was perhaps the world's first physically modelled guitar preamp but, much more notably, it was also the basis of the VG8 Virtual Guitar system (see the box on the left). Inevitably, the VG8 didn't quite live up to its promise, and Roland were to improve the V-Guitar system on numerous occasions over the next few years. The VG8 would be superseded by the VG8EX in 1998, followed by the VG88 in 2000, the VGA series of modelled guitar amplifiers in 2001, and the V-Bass in 2002. Likewise, the GP100 would soon make way for the GX700, and then the floodgates would open, as COSM infiltrated many of Roland's product ranges. Indeed, the company now incorporate COSM technology in almost every synth, organ and multi-effects unit that they produce. Even if it was in some respects one of the company's less eventful years, at least one thing must have put a smile on the face of Roland's shareholders and bankers in 1995 — this was the year in which the company sold their millionth GS sound module. Given that the format only appeared in 1991, this was no mean achievement.

Milestone: The VG8 The VG8 confused many when they first saw it. Was it an effects unit? Not exactly. Was it a guitar synth? Well, not exactly. Was it a preamp? Yes and no. In fact, the VG8 proved to be the precursor of what has since become a common element in our musical armoury: the physically modelled guitar amplifier, speaker cabinet and recording environment. With the GK2A pickup mounted on your existing electric guitar, the The VG8, core of the V-Guitar system, the Virtual Guitar Modelling (VGM) first of Roland's commercially released (and system in the VG8 allowed you to highly successful) 'V' products (the V-Piano, decide whether the resulting sound supposedly the first mooted 'V' product, would be that of a single-coil or remains unrealised at the time of writing...). humbucking pickup, and where that lay on the body or neck. You could then choose which kind of amplifier the system would model, and whether the output was reproduced through a virtual 1x12, 2x12, 4x12, or a stack. Finally, you could choose whether or not the sound was DI'd, or recorded using one of selection of 'virtual microphones' positioned at user-defined distances and angles from the virtual speaker cab. COSM provided a fair degree of control over all of this, allowing you to obtain a good range of rock timbres using standard pre-amp and amplifier controls, as well as Boss-style dynamics processing, overdrive, and effects. The factory programs



tended to concentrate on the mainstream nature of the sounds thus produced, with patches named Jimi Hendrix this and Jeff Beck that, but the VG8 also offered the hex-distortion that made the GR300 so distinctive, so it was able to produce another wide family of sounds not obtainable from conventional guitar, amp, and speaker setups. In addition to all the above, the VG8 included a second form of DSP processing called Harmonic Restructure Modelling, or HRM. This was surrounded by quasiscientific marketing gobbledygook but, in essence, it was a means of processing the string vibration in different ways to produce new harmonic structures. Roland pointed out that, because the actual string vibration was the sound source, the results were far more expressive and organic than using the guitar to trigger synthesized sounds. This may have been true, but few players made full use of the brass, bowed strings, lead synths, bass, pads, organs and all manner of weird sound effects models that HRM offered.

1996 MAJOR PRODUCT LAUNCHES AMPLIFIERS & MIXERS KC100/KC300/KC500 keyboard amps. SRA260/SRA540/SRA800 power amps.

BOSS PRODUCTS GT5 effects processor. GX700 effects processor. ME8B bass multi-effects. PW2 Power Driver pedal. SX700 effects processor. SYB3 bass synth pedal. VT1 Voice Transformer.

The VG8 marked the end of a fallow period for Roland that had lasted from 1993 to 1995, and the next two years were to see an explosion of advances, many of which were fuelled by the magic letter 'V'. Of these, the first and one of the most significant was the V-Studio, more commonly known as the VS880. Announced in 1995, and immediately nicknamed 'the digital portastudio' (although not by Tascam, who had registered Portastudio as their trademark), the VS880 brought digital recording, mixing and multi-effects together in an affordable, self-contained package that was portable enough to carry around in a carrier bag (see the box below).

XT2 Xtortion pedal.

DIGITAL RECORDERS VS880 digital multitrack & VS8F1 effects board.

EFFECTS EQ131/EQ215/EQ231 graphic EQs.

GUITAR PRODUCTS VG8S1 system expansion for VG8.

HOME/ACCOMPANIMENT PRODUCTS E12/E28/E38/E68/E96 'Intelligent


With a built-in MC-style sequencer, JV1080 synth engine (including effects) and 76-note keyboard, the XP80 was a fine successor to the JV1000 workstation.


Synthesizers'. RA800 'Intelligent Module'.

MASTER KEYBOARDS A33. A90. A90EX (plus VERD1 piano expansion board).

ORGANS Atelier AT30. Atelier AT90.

PIANOS HP135/HP235/HP550G/HP555G digital pianos. KR370/KR570/KR770 'Intelligent pianos'.

SAMPLERS DJ70 MkII workstation.

SOUND CANVASES SC55ST. SC88. SC88ST. SC88VL. VSC55 Virtual Sound Canvas.

SYNTHS & HI-TECH MC303 groovebox. PMA5 personal music assistant.

Elsewhere, the evolution of JVs into XPs was continuing apace as the JV1000 was replaced by the XP80 workstation. This shared the big JV's 76-note semiweighted keyboard, but it was far more than simply a wider XP50. It offered a much larger LCD that could display extensive graphics as well as menus, and there were more real-time controls, twice as many outputs, inputs for twice as many controller pedals, a more powerful sequencer, and a new arpeggiator. With its four expansion slots, the XP80 was the closest thing to a JV1080 with a keyboard and, as such, it justified its continued status as a powerful and popular workstation. Roland also expanded their range of Atelier home organs with the AT30 and the new head of the range, the AT90. This offered the oversize manuals of its predecessor, provided similar autoaccompaniment facilities, stuck with the GS/GM sound source, and again blasted 240W through no fewer than 11 speakers! But there were many extra features, and although there's no room to discuss them here, one bears special mention. It's the price: a mere £19,999. It's clear that, when Kakehashi entered this market, he wasn't messing around.

SN700 noise eliminator. XP80 workstation.

VIDEO PRODUCTS LVC1 lyrics video converter.

Perhaps the best keyboard launched in 1996 was the replacement for the A80 controller keyboard. The A90 looked 1996's A90, one of Roland's finest controller keyboards. fantastic, and its superb 88-note piano action simply begged you to play. In addition, it had many unexpected and sophisticated functions under its large, clear controls and, with the VEGS1 or VEJV1 board installed, it became a powerful GS/GM or JV sound source in its own right, with limited but nonetheless useful editing capabilities. Nevertheless, the jewel in the A-series crown was the A90EX, a standard A90 with the VERD1 board pre-installed. This contained what remains for some players the finest digital piano yet developed, complemented



by numerous electric pianos, D50 and JV-style sounds, all editable to a degree, enhanced with EQ, chorus and reverb, and reproduced with generous 64-note polyphony. The A90 was large, heavy, beautifully built and expensive, but I suspect that 1996 will be best remembered for a product at the opposite end of the spectrum. Small, light, and relatively cheap, it was something that Roland had spent more than a decade claiming that it would never produce. It was the successor to the chosen tools of the techno brigade, the TB303, the TR808 and the TR909. It was the MC303, which, depending upon your perspective, either helped to hurry the destruction of popular music, or was a wonderful, liberating box that brought Groove to the masses. The MC303 comprised two major sections: a synthesizer that included numerous rhythm sets, and a sequencer. The former was 28-voice polyphonic, and many of its sounds were culled from the TB303 and various TRs. There were complete sound banks devoted to synth bass, as well as whole banks of individual percussion sounds such as kick drums, snares, handclaps and so on. Roland said they'd never re-introduce the However, there were also numerous TB303, as it was too expensive to masspianos, organs, brass patches, and produce its all-analogue design in the guitars. Consequently, you could use mid-'90s — but eventually they did concede and produce a digital version, the MC303, to the MC303 to create far more than just great success. drum & bass. The sequencer offered just eight tracks and a capacity of 14,000 notes, but it was far more flexible than it seemed, with phrase sequencing, an arpeggiator, several quantising functions, a few hundred preset patterns and their variations, and space for a further 50 patterns of your own. Roland had inadvertently helped to invent house, acid, trance, techno, and most other forms of '90s dance music, so it came as no surprise to find that the MC303 was an instant success. Analogue purists tried to claim that it sounded nothing like the real thing, but played through a 10,000W PA system producing little sound energy above 150Hz, it's doubtful that anybody could really tell the difference. So, despite a few anoraks moaning about how things weren't as good as they were in the good ol' days, the MC303 simultaneously defined the nature of the Groovebox and re-defined the way in which many people produced dance music. A few other developments from 1996 deserve a quick mention. One was the Boss VT1 Voice Transformer, which soon popped up in all manner of Roland products, and provided endless hours of fun, turning men into women, women into children, children into robots, and robots into Bugs Bunny. Then there was the LVC1 Lyrics Video Converter, which read the lyrics that could be stored



within the SMF data format, and displayed them on a video monitor. Compatible with the G800, RA800, the KR570 and KR770 'Intelligent' pianos, plus the E68 and E96, this allowed you to turn your expensive keyboard into a karaoke machine. No doubt it was big in Japan! Finally, just to prove that Roland don't always think of the good ideas first, there was the PMA5 Personal Music Assistant. This hybrid of a Palm Pilot, a sequencer and a GS/ GM sound source was a great little product that would have been worthy of far more than a mere mention... had Yamaha not done something rather similar five years before.

Boss's VT1 was one of the first affordable formant-based vocal processors, allowing you to transform male into female-sounding vocals and vice versa, as well as offering many vocal special effects.

There were a couple more important snippets of news in 1996. Firstly, after the failed attempt to relaunch Rhodes pianos and keyboards in digital form, it appears that Roland transferred the brand name back to Harold Rhodes, who said that he hoped to restart production of electro-mechanical pianos. Unfortunately, Rhodes was an elderly man, and died just four years later, aged 89, without realising that dream. Secondly, there was the appointment of Mr Katsuyoshi Dan as President of Roland Corporation. Ikutaro Kakehashi was to stay on as Chairman of the Group and of the Roland Foundation, but his health had declined and, after 24 years, Dan's appointment signalled a lessening of Kakehashi's involvement in the day-to-day running of the company.

Milestone: The VS880 Digital Studio Workstation A handful of Roland employees first saw the VS880 in 1994, when it appeared as a Styrofoam dummy called the VR88. Of course, the development was highly confidential because, had it leaked out, it would have killed sales of the existing DM80 and the DM800 that was due to be launched the following year. But when the VS880 finally appeared in 1997, nothing in it was new. We had all seen eight-track hard disk recording before, as well as digital mixing, automation, and digital effects. We had also been introduced to Roland's Sound Space processing (RSS) and COSM, so the optional VS8F1 dual file:///H|/SOS%2002-05/The%20History%20Of%20Roland.htm (20 of 25)9/22/2005 8:21:57 AM

It wasn't the first compact digital multitracker, but it was the first really good one, combining random-access digital recording, advanced COSM effects, and digital mixing facilities in one sub-£1500 product.


effects board did not cross any boundaries either. Nonetheless, the VS880 and VS8F1 combined all these elements in a new way that was both affordable and simple to use. With a 540MB drive installed, the VS880 offered around 100 minutes of audio at 44.1kHz, but for applications where recording time was more important than audio quality, three data-compressed recording formats were offered. Not so critical in today's world of Terabyte drives, this was a significant consideration when external 1GB drives were not only state-of-the-art, but seriously expensive. The VS880 could only record six tracks at a time, but each of its eight tracks could up to hold eight 'Takes'. A limited form of snapshot automation was provided within the VS880 itself, but by connecting it to an external MIDI sequencer, you could enjoy dynamic automation of many parameters. The VS880 fell short in two areas. Firstly, its editing model (unassisted by a waveform display) was even less capable than that of the DM800. Secondly, there was its inability to replay all eight tracks when using the uncompressed audio mode. In this case, just four tracks were available. But as the healthy sales attested, many people were happy to live with these restrictions, perhaps because the VS880 was the cheapest way to obtain all of these features in one package. And though it lasted just a year before being superseded, it was a seminal product for Roland.

1997 MAJOR PRODUCT LAUNCHES AMPS, MIXERS & SPEAKERS GC405/GC408/GC405X & GC408X guitar amps. JC90 Jazz Chorus amp. KC100 keyboard amp. SST151 & SST251 speaker systems. SSW351 subwoofer.

BOSS PRODUCTS AC2 acoustic simulator. AD5 acoustic instrument processor. CT6 guitar/bass tuner. DB12 Dr Beat. DB88 Dr Beat. FZ3 fuzz pedal. ME30 guitar multi-effects. OD3 overdrive pedal. SP202 Dr Sample. TR2 tremolo pedal.


It was Roland's 25th birthday in 1997, and whether through coincidence or otherwise, it was to prove to be a bumper year for the company. Despite Kakehashi's departure from the hot-seat, Roland continued to establish new companies and joint ventures worldwide, adding Roland France SA and the Roland Taiwan Enterprise Company Ltd. (which is distinct from the manufacturer, Roland Taiwan Electronic Music Corporation) to the roster. Back in Japan, a new factory in Miyakoda was opened, primarily to build larger products such as Atelier organs. Roland were very proud of this plant, declaring that production was carried out using a 'cell' system in which each worker assembled a single unit from start to finish. Given that the rest of the world embraces the production line (in which each worker is responsible for a single element of all units, rather than all the elements of one unit) this is very strange — Henry Ford would have been dumbfounded.


DIGITAL RECORDERS VS880 V-Xpanded digital studio workstation.

GUITAR SYNTH GR30.

HOME/ACCOMPANIMENT PRODUCTS E14 'Intelligent Synthesizer'. E500 'Intelligent Synthesizer'. G600 Arranger Workstation.

MASTER KEYBOARDS A70.

1997's VK7 was Roland's best tonewheel organ emulation to date.

This was also the year in which Roland applied the letter 'V' to three more areas of music technology, with their first modelled organ, a virtual analogue synth, and most important of all, the 'VDrums' (see the box opposite).

ORGANS Atelier AT80. VK7 'Virtual Tonewheel' combo organ.

PIANOS EP75 & EP85 digital pianos. HP130 & HP730 digital pianos. RD600 digital piano.

RHYTHM PRODUCTS PD100 'V-Pad'. PD120 'V-Pad'. TD10 V-Drums percussion sound module.

SOUND CANVAS SC88 Pro.

SYNTHS & HI-TECH AF70 anti-feedback processor. JP8000 analogue modelling synth. JV2080 synth module.

Given Kakehashi's major contributions to the development of electronic organs in the 1960s, it's remarkable that Roland's efforts in this field had never lived up to their heritage. The company sold few VK6s and VK9s (1977), and the later VK1 (1979) and VK09 (1981) were horrid affairs. Released in 1990, the samplebased Rhodes VK1000 was a much finer instrument but, by 1997, numerous companies had produced better variations on a theme by Hammond, who had themselves developed the XB2 and XK1 keyboards, plus the XM1 module. Consequently, it was pleasing to discover that the VK7 'virtual tonewheel' was superior to these, chunky and robust, with a sound that was — thanks to its COSM overdrive, amplifier, and rotary speaker effects — even closer to that of a vintage Hammond. Unfortunately, the VK7 was let down by a handful of minor niggles.

MBD1 sound module. SMPUII AT MIDI interface.

The year also saw the pinnacle of the development of the 'JV' series. The JV2080 was a better, more powerful JV1080 that employed the XP80's large LCD, provided three EFX units, and was capable of hosting no fewer than eight expansion boards. With more patches and performances, and an improved operating system, the only deficiency with respect to the earlier model was the loss of the PCM card slot. But this was an insignificant loss, and the JV2080 immediately became Roland's 'pro' module, one that remains highly desirable to this day.



Powerful though the JV2080 was, it was overshadowed by the launch of Roland's first 'physically modelled analogue' synthesizer, the JP8000. And, as had happened when the company announced the JD800, the Together with the earlier JV1080, the JV2080 was the bread-and-butter synth rumours began. It had a 'J' in the module of the '90s, and can be heard on name, and an '8'... was this the longthousands of records. awaited digital Jupiter 8? Certainly, its knobs and sliders were in the grand tradition of big analogue polysynths and, in many ways, the JP8000 looked and felt vintage. Seven analogue-style waves lay at its heart, and these included the first appearance of the Super Saw waveform that has since become a standard ingredient in virtual-analogue synthesis. Like the Jupiter 8, the JP8000 offered 12dB- and 24dB-per-octave filters, and like the Jupiter 6, these were multi-mode affairs, with low-pass, band-pass and high-pass options. Other nods to the past included an effects section with chorus and delay (but no reverb), and an arpeggiator. There was also 'Motion Control', which was a grand name for recording front-panel adjustments as miniature eight-bar control sequences that could then be replayed as loops, or output to an external sequencer. Inevitably, not all was rosy, and the JP8000 had its deficiencies. Foremost among these was the unfathomable decision — and, in my view, serious mistake — to limit it to a four-octave keyboard. Secondly, it could sound a bit thin when compared to the Nord Lead and other modelled analogue synths of the day and, thirdly, its duo-timbrality and eight-voice polyphony were already way behind the times. Nonetheless, the JP8000 carved a niche for itself, and many remain in use today, not because they are the most powerful tools for the job, but because — like some classic synths — players simply like their sound. Elsewhere, Roland were continuing their habit of recycling existing technology in new ways. A sixth 'Sound Expansion' module, the MBD1 was a rackmount version of the SR JV10 Drum & Bass Expansion Board, while the RD600 stage piano (a superb instrument, by the way) was a hybrid of an RD500 piano, the A90's VERD1 The JP8000 wasn't quite the 'Digital Jupiter' board, and a selection of sounds from the synth fraternity had been waiting for, but it was a sufficiently impressive physical the SR JV80 expansion board series. modelling synth that many of them remain in As always, there were numerous new use today, eight years on. Boss pedals, numerous guitar amplifiers and effect units, another guitar synth, another Atelier organ, another Sound Canvas, another master keyboard, a significant redesign of the E-series keyboards to create the E300 and E500, and... well, you get the idea. There was even an upgrade to the previous year's VS880, which became the VS880 V-Xpanded. This was the same hardware with improved software, so existing owners could upgrade the VS880 to the later specification with the file:///H|/SOS%2002-05/The%20History%20Of%20Roland.htm (23 of 25)9/22/2005 8:21:57 AM


VS880 S1 System Expansion Kit. This included the new software, new manuals, and stickers to annotate the new functions on the front panel. The improvements were not trivial. Most notably, you could now replay six tracks in the uncompressed audio mode, and there were 10 new effects, including the Boss Voice Transformer algorithm, a COSM-generated Mic Simulator, and a Vocoder. To further improve the usefulness of these — and the previously included effects — Roland added new Insert options, as well as on-board automation and improvements to the editing capabilities. With a host of other improvements, the VS880 V-Xpanded was a much better buy than the original VS880, so the VStudios deserved their continuing success.

Milestone: V-Drums Throughout the mid-1990s, Roland's 'Compact Drum Systems' had proved to be a disappointment, both to drummers and to Roland, with sales far lower than anticipated. However, Ikutaro Kakehashi remained committed to manufacturing instruments in all the main musical groups and, recognising that the problem lay not in the sound generated by the kits, but the way that they responded to the drummer's technique, he never gave up on the idea of producing an electronic kit that both sounded and felt like an acoustic kit.

Drummers had failed to warm to Roland's electronic drums since the late '80s, but with 1997's V-Drums, the first COSM-based drum system, Roland finally cracked the market. The V-Drums have been revised a couple of times since, and remain some of the company's most successful current products.

After years of experimentation, Roland's engineers developed a drum pad based on what the company called a 'mesh drumhead'. This special fabric could be tensioned in the usual fashion, and therefore adjusted to suit each player's taste. Most importantly, it responded in a way that was much closer to that of a genuine drum skin. Unfortunately, Roland did not know how to manufacture these heads in quantity, so Kakehashi turned to Remo Belli, the head of Remo, the worldrenowned manufacturer of acoustic drums and percussion. Remo's expertise was crucial to solving Roland's problems, and soon a new generation of electronic drum pads were born: the 10-inch diameter PD100 and the 12-inch diameter PD120. To complement the new heads, Roland introduced the 'V-Drum' brain, the TD10. Another product based on COSM, almost overnight this made all previous electronic drum sources obsolete. The TD10 provided a huge selection of drums (more than 600 of them) with detailed editing and fine-tuning of the bass, snare and tom-tom sounds, plus a wide selection of accompaniment sounds, extensive effects, and sequencing — but it wasn't the specification that made the V-Drums special. It was simply that the TD10 and a selection of PD100 and PD120 pads formed the basis of the most expressive electronic kit ever created at that point, one which provided the feel, controllability and expression that professionals file:///H|/SOS%2002-05/The%20History%20Of%20Roland.htm (24 of 25)9/22/2005 8:21:57 AM


demanded. With a couple of KD7 kick drum units, a FD7 hi-hat controller, plus a bunch of the earlier PD5, PD7 and PD9 pads for cymbals and other metalwork... the fully electronic drum kit had finally arrived!

Epilogue At the start of the '90s, Roland were emerging from a golden age in which it seemed that they could do little wrong. Then a recession hit... and whether this simply reflected the worldwide economic slowdown or pointed to a malaise in the music industry itself, the result was the same: sales in the company's traditional markets plummeted. Roland responded to this by purchasing existing companies, opening new companies, and expanding into new markets. They added home organs, General MIDI Sound Canvases, electronic drum kits, hard disk recorders and a huge range of electronic pianos to their existing synth, sampler, guitar synth, amps and effects product lines. Nevertheless, it was the emergence of COSM and the 'V' products in 1995 that pointed the way forward. Kakehashi and his teams had recognised that digital signal processing and physical modelling were the future of the electronic audio industry, and Roland would soon be applying these to everything from the humblest sound modules and dance-oriented Grooveboxes to their most expensive digital mixers and audio workstations. Which is, of course, where we'll pick up this story next month... Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Cubase: Working with Multi-Channel Projects

In this article:


Linked Channel Tips & Tricks Cubase Notes A Channel With A View Published in SOS February 2005 Little & Large The Weakest Link Print article : Close window Sub-mixing With Group Technique : Cubase Notes Channels

Returning to Cubase's Mixer window again this month, we take a look at some ways to make it more manageable when working with Projects containing a large number of channels. Mark Wherry

Once you start working with large Projects — which is to say, Projects with a large number of channels — you'll find that it can become quite difficult keeping track (no pun intended) of them on the Mixer. To help make life easier, Cubase allows you to do two things: firstly, you can open a second or third Mixer window simultaneously from the Devices menu, to look at different parts of what is the same mixer. Secondly, you can customise either Mixer window to show only what you need to see at a given moment.

In Cubase SX 3 you can open and configure three separate Mixer windows. In the example above, one Mixer shows MIDI channels, another shows audio, input and output channels, and the third window displays FX channels.

As you've probably noticed, the order in which Cubase displays channels on the Mixer from left to right is exactly the same as the order in which tracks are displayed on the Project window from top to bottom. And since Cubase has many different varieties of tracks and channels that can be viewed on the Mixer these days, this window easily becomes overcrowded. To make things easier, you can configure the Mixer to display only the channel types you require, using the Hide Channel buttons in the Common Panel. This can be useful if, for example, all of your MIDI channels are playing through VST Instruments and you only need to see the channel strips for the VST Instruments file:///H|/SOS%2002-05/Cubase%20%20Working%20with%20Multi-Channel%20Projects.htm (1 of 5)9/22/2005 8:22:13 AM


during mixing. Enabling the Hide MIDI Channels button (Hide Channel buttons glow orange when enabled) hides all the MIDI channels on the Mixer. In addition to hiding and showing channels of different types, you can also set up a user-definable hide-and-show group across different channels of any type. Click the Channel View pop-up menu for a channel you'd like to assign to the hide-and-show group, and make sure the 'Can Hide' item is ticked — click to tick if it isn't already. After doing this you can hide and show the channels you assigned as 'Can Hide' by toggling the 'Hide Channels Set To 'Can Hide'' button. That's the lowest icon in the set of Hide Channel buttons.

Linked Channel Tips & Tricks When you're working with a group of linked channels, it's still possible to make changes to individual parameters without having to first unlink the channels. Simply hold down the Alt/Option key as you adjust the parameter. Since all the channels in a linked group are automatically selected when one of the channels in that group is selected, you can select just one of the channels in a group by holding Alt/Option as you click the name at the bottom of a channel strip. Note that this only works if the linked channels aren't already selected on the Mixer. You'll have to deselect all of them, by selecting another channel, before you can select just one of them. Although it's possible to have multiple groups of linked channels in the Mixer, it's worth remembering that a channel can only belong to one linked group. While you don't have to unlink a channel before you include it in another group, bear in mind that a channel will no longer be part of the linked group it previously belonged to once it's been reassigned. The original group will remain otherwise intact, however.

A Channel With A View Being able to configure what you see in the Mixer is, of course, very useful; but continually reconfiguring the Mixer to see different sets of channels can be rather tedious. To overcome this, you can create what Cubase term Channel View Sets, taking a snapshot of the current Mixer configuration that's saved with the Project and can be recalled at any point. When the Mixer has been set up to display a certain configuration you'd like to be able to get back easily at a later date, click the 'Store View Set' button (located at the top left of the Common Panel in the lowest section of the Mixer), enter a name in the politelytitled 'Type In Preset View' window and click OK or press Return. You can now

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recall this Mixer configuration at any time by clicking the 'Select Channel View Set' button and selecting the configuration from the pop-up. It's possible to delete a Channel View Set preset by recalling it and clicking the 'Remove View Set' button on the Common Panel. If you want to overwrite an existing Channel View Set preset, simply follow the procedure for creating a new preset and give it the same name as the preset you want to overwrite. When you click OK, Cubase will ask you if you want to overwrite the existing preset by clicking Overwrite, or abort the whole procedure by clicking Cancel. It's worth noting that the Channel View Sets you create are available independently of each other in both Mixer windows, enabling you to view two different Channel View Sets simultaneously: one in each Mixer window. Usefully, Channel View Sets are stored globally with the application, meaning that any Project you open will be able to use the same Channel View Sets.

Little & Large One way to identify all the channels on the Mixer is simply to display more channels. For those who can't justify a bigger monitor, Cubase offers two different widths for displaying each channel strip: narrow and wide. Wide is the usual setting. Although the narrow view limits the number of parameters visible for each channel, it's a useful way of cramming more level faders and meters on to the screen when you're concentrating on balancing levels, for example. To view a channel as either wide or narrow, click the Narrow/Wide toggle button above that channel (the button is located just above the selection strip, which is above the Channel's pan control). Alternatively, to view all the channels on the Mixer as wide or narrow, click either the 'All Wide' or 'All Narrow' button at the top of the Common Panel to the far left of the Mixer. If you have the input or output channels visible on the Mixer, you'll notice that clicking the 'All Wide' or 'All Narrow' buttons makes no difference to input or output channels. To include these channels in the 'All' operation, Alt/Option-click the buttons instead.

The Weakest Link When you're working on large and complicated mixes involving many channels, making manual adjustments to faders individually can become impractical. A common approach to large mixes is to get the balance right within specific sections, such as the balance of a drum kit or string section, and then start mixing these sections together. Since you want to be able to adjust the volume of a section in one go, rather than having to go along and move each fader for the section individually, Cubase provides two options. One way in which you can group channels on the Mixer is to link them together. Wouldn't it be great if you file:///H|/SOS%2002-05/Cubase%20%20Working%20with%20Multi-Channel%20Projects.htm (3 of 5)9/22/2005 8:22:13 AM


could move one of the faders for an audio channel containing part of the drum kit, and have the faders for the other audio channels containing the drum kit adjusted proportionally? This is exactly the kind of thing you can do by linking channels on the Mixer. In the Mixer window, select the first channel you want in the selection, by clicking its name at the bottom of the channel strip. Then Shift-click all the other channels you want to link in the same group. Right-click in an empty part of the Mixer (on a level meter, for example) and select 'Link Channels' from the pop-up menu. Now, when you adjust one of the faders on a linked channel, notice how all the faders on the other channels that were linked move proportionally with the fader you're adjusting. It isn't only faders that are linked when you link channels together; other parameters are too, including the Mute, Solo, Monitor and Record Enable buttons. To unlink a group of channels, select one of the channels in the group you want to unlink and all the others should become selected too. Now right-click in an empty part of the Mixer and select Unlink Channels from the pop-up menu. Unlinking any channel that's part of a group will unlink all channels in that group since it's not possible to unlink one channel and keep the rest of the linked group intact.

Sub-mixing With Group Channels Linking channels can be a useful way of handling the mix in sections, but there's another useful technique for audio-based channels (which can be used in addition to linking channels, incidentally): sub-mix them through Group channels. For example, you could route the individual audio outputs of the channels containing drum-kit sounds to a Group channel, which would enable you to treat the output of the drum kit as a single stereo channel with one fader. It would then be easy to, for example, use a plug-in as an insert effect (a compressor, for example), across the entire output of the drum kit. To create a new Group channel, select Project / Add Track / Group Channel and choose a suitable channel configuration (mono, stereo, 5.1, and so on) from the 'Add Group Channel Track' window. Click OK. Notice how a new Group channel track is added to the Project window within a newly created Group channels Folder track. Cubase will automatically add all Group channel tracks created

The Groups/FX page in the VST Connections window provides another way of configuring Mixer Group channels.



subsequently into this Folder track. (It's worth noting that the Group channel appears as a track on the Project window for automation purposes only — you can't drag any audio onto it directly.) As a side note, since SX3 you can create Group channels in the Group/FX page of the VST Connections window (Devices / VST Connections) by clicking the 'Add Group' button, setting its channel configuration and clicking OK. From here you can also set which output buss the Group channel's audio is routed to. The next step is to route the outputs of audio-based channels to the Group channel, which you can do by clicking the output selector of an audio-based channel (either in the General Section of the Inspector or above that Channel in the Mixer's Input and Output section) and selecting the Group channel from the pop-up menu. And that's basically all there is to it — channels routed to a Group channel are processed through the latter before being sent to the Master channel, or indeed another Group channel. You can use Group channels to create sub-mixes of sections such as drum kits, keyboards, and so on, allowing the final mix to be carried out by mixing a few stereo audio channels. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Pro Tools: The Mysteries of Shuffle Mode Reavealed...

In this article:

Basic Shuffle Editing Mark My Words Shuffle Top Tips More Key Combinations Making Selections Whilst Playing

Current Versions

Pro Tools: The Mysteries of Shuffle Mode Reavealed... Pro Tools Notes Published in SOS February 2005 Print article : Close window

Technique : Pro Tools Notes

6.4: HD, Accel and LE systems on Mac OS X, LE systems on Windows XP. 6.4: HD & Accel systems on Windows XP. 6.4.1: Older TDM systems on Mac OS X and Windows XP.

Pro Tools was originally designed as an audio editor, and it still has some very powerful tools for quick and accurate editing. This month's Pro Tools Notes explains the mysteries of Shuffle mode, and looks at the ways in which key commands and memory locations can speed up your editing. Mike Thornton

There are two main editing modes in Pro Tools: Shuffle and the default Slip. In Slip, when you delete a region Edit modes are selected at the top left-hand nothing else moves, so you are simply corner of the Edit window. left with a 'hole' where the deleted region was. In Shuffle, by contrast, when you delete a region, all the regions on that track after the deleted one move up and so close the hole. This feature is particularly useful for editing speech, say for a radio interview, as it provides a quick way to tidy up all the errors, 'ums' and 'ahs'. It does also have uses in a multitrack setting, but you need to be aware that if you Shuffle edit on one track, everything after that point will go out of sync with the other tracks unless you take steps to prevent it. (Everything I'm going to describe in this column works on both TDM and LE systems.)

Basic Shuffle Editing Select Shuffle in the toolbar. Then use the Selector tool ('I'-shaped cursor) and click, hold and drag across the audio to highlight a section you wish to remove. Holding down Control + Shift and then clicking and dragging the mouse (on Windows, hold down Shift, right-click and drag) will allow you to 'scrub' audition

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the edit point, although with a lot of material it's easy enough to see the edit points from the waveform display. Once you have selected your audio to delete, press Backspace (the one above the Return key) on the keyboard to delete it. Alternatively, you can use Command + E (Windows: Control + E) Here, a section of audio has been selected to make the various edits and then by placing markers and using keystrokes. delete the appropriate regions you create by clicking on them with the Grabber (hand) tool and pressing the Backspace key. There are other ways of using Shuffle mode, too. When you drag regions out of the Region List in Shuffle mode, they will always snap to the tail of an existing region. This allows you to insert a region between two others, but be aware that everything after it on the track will shuffle along to the right to make room for the inserted region. You can also use Shuffle to reorder a section: as you move one region, the gap closes up, and you then can insert it between two different regions. The great thing with Shuffle is that you don't have to be too precise with your placement within a track — so as soon as you see a shimmering effect at a region boundary, you know that is where the selected region will go when you drop it.

Mark My Words Once you start re-editing a section which already has a lot of edits, it can be very helpful to use Markers and Memory Locations to keep track of the correct edit points. Having found your 'in' point (ie. the point where you wish to cut and create the start of a new region), press '1' on the numeric keypad and then press the keypad's Enter key, then check that Marker is selected under Time Properties and then press Enter again. I've used 1 as an example, but you can use any number from 1 to 99. Then, having found the 'out' point (where you will cut to create the end of your new region), press '2' and Enter, again checking that Marker is selected, and then press Enter again. Note that for subsequent edits you can simply press '1' or '2' and Enter twice, as Pro Tools will remember that Marker is selected for these two locate points.

Hitting the down and up arrow keys during playback automatically marks out a selection.

To select the audio between the two markers press '2' then '.' (period or full stop), hold down the Shift key and press '1' then '.' Press Backspace to delete the highlighted selection. file:///H|/SOS%2002-05/Pro%20Tools%20%20The%20Mysteries%20of%20Shuffle%20Mode%20Reavealed....htm (2 of 5)9/22/2005 8:22:33 AM


(Note that these shortcuts to recall a marker point only work if 'Classic' is set in the Numeric Keypad Mode in the Operations section of Preferences window. If you prefer to use 'Transport' then you will need to add another period before the combination, making it '.' then '1' and then '.' again to recall marker 1. Remember that when using numbers in memory locations you must use the number pad to the right-hand end of your keyboard, rather than the number keys on the main keyboard.)

Shuffle Top Tips Now you have mastered making editing selections and using Shuffle edit to tidy your audio up, here are some power-user Shuffle-editing tips... Shuffle editing within a multitrack session can be useful, but can cause problems because it will mess up the sync and/or timing of everything after it in the Session. The trick is to go to the first region on the same track after the section you want to Shuffle edit and lock it using Command + L (Windows: Control + L). Now you can Shuffle-edit your section and nothing later on in the Session will move because of that one locked region. When you have finished, unlock it, tidy up the gap and you're done. Rather than changing to Shuffle mode just to Above: Shuffle editing across an butt two regions together, try this. Still in Slip entire Session can be a useful way of (make sure Tab to Transient is off), click with reordering the structure of a the Selector Tool in region one, then Tab so the multitrack project. cursor is now at the end of region one. Next Control-click (Windows: Start-click) region two and it will snap to where the cursor is — which is the end of region one! Sometimes you need to Shuffle edit 'in reverse' — the final region is in the correct place, and you want to butt up a preceding region to its beginning. To do this, highlight the last region and then Command + Control-click (Windows: Start + Control-click) on the preceding region. Its tail will then snap to the head of the last region.

This works because Pro Tools understands that there are three 'sync' points for each region: the head, tail and one point in between. If you want to snap the head of a region to a mark, Control-click (Windows: Start-click) it. If you want to snap the tail of a region to a mark, use Command + Control-click (Windows: Start + Control-click). Finally, if you want to snap the user sync point, first define your sync point (which can be any sample point between the head or tail of a region but not the head or tail), then use Control + Shift-click (Windows: Start + Shiftclick). Sometimes you need to make a Shuffle edit across a multitrack Session, if for instance you want to delete an entire section of a song. Make your selection on one track and use Command + E (Windows: Control + E) to make sure you have a region that represents what you want to take out of the Session. Then hold down Option + Shift (Windows: Alt + Shift) and click with the Grabber (hand) tool on the region you have just created. This will extend that selection vertically across all the tracks. Now, if you hit Backspace in Shuffle



edit mode, the same amount is taken out of all the tracks, so everything moves up together and nothing slips out of sync! Left: The region at the right has been locked, meaning that it and other regions further to the right will be unaffected by any Shuffle editing taking place to the left.

More Key Combinations There are four other keys which are very useful in making edit selections, and we have used one of them in the example above. These are Tab, Shift, Option and Return, which can be used in combinations to enable you to do a number of very useful tricks. So let's look at what each one does individually and then we can see how to put them together. The Tab key does exactly what you'd expect, provided Tab to Transient is switched off. Place the cursor on a track with some edits, and when you press the Tab key, you will see that the cursor moves to the right to the next region boundary. The Option key (Windows: Alt key) changes the direction the cursor goes, so if you hold down the Option key and press the Tab key, you will see the cursor move to the preceding region boundary. Things start to get useful when you add the Shift key into either of the above key combinations, which makes the cursor highlight a selection as it goes. This is great for making a selection back to an existing edit point. Simply place the cursor at a point, then use Shift + Tab to make a selection to the next region boundary or Shift + Option + Tab make a selection to the previous region boundary. Finally, the Return key takes the cursor back to the start of the Session. Option + Return will take the cursor to the end of the Session, which is great for checking the length of a song or programme. In addition, you can select the whole Session for doing a bounce by first pressing Option + Return to go to the end of the Session and then using Shift + Return to select everything back to the start of the Session.

Making Selections Whilst Playing You can also make selections without stopping playback. When you get to your 'in' point, press the down arrow key, and then when you get to your 'out' point, press the up arrow key. Pro Tools will automatically highlight a selection between file:///H|/SOS%2002-05/Pro%20Tools%20%20The%20Mysteries%20of%20Shuffle%20Mode%20Reavealed....htm (4 of 5)9/22/2005 8:22:33 AM


the two points. You then can stop and fine-tune the edit points before deleting the selection, for example. Make sure you hold the Shift key down whilst you tweak the edit points with the Selector tool, or you will instantly lose your selection! Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Reasons to be cheerful: v3

In this article:


Coming Attractions Reason Notes Master Plan Published in SOS February 2005 Before You Ask... (Re)Fill 'Er Up Print article : Close window There Will Be A Long Delay

Technique : Reason Notes

Version 3 of Reason is officially on the way, adding some high-spec new devices and improved operational features. Derek Johnson

It's been buzzing for a while, but the big news this month is the official announcement by Propellerhead of Reason V3. There's no definite release date as yet, since the software is still in beta-testing (as I write). But, as with the last couple of upgrades, Propellerhead have been punting for beta-testers via their web site, so that some of the user base will have a chance to contribute to the final product. In fact, it was largely the call for beta-testers that alerted users to the fact that something might be on the way!

Coming Attractions Many of you may know that, before its original release, Reason was meant to be more of a live performance tool than it turned out to be, and indeed a handful of functions in the program are left over from those plans. With the new Combinator device, Reason moves back to its performance roots by allowing as many synth, sampler and effects devices as you like to be joined together and to work as one single performance instrument — just like a workstation synth. Combinator even lets you integrate Matrix pattern sequencers and Redrums into the resulting 'Combi' patch. Once saved, a Combinator patch can be recalled with all chosen devices loaded instantly and the signal path intact. A Combi could even consist of just effects — its back panel offers interface jacks that let you use Combinator as a single giant effects processor Devices can be layered, split across MIDI key ranges and set to respond to velocity splits, and ease of use is improved by a large clear display window. The Combinator device even offers a handful of global assignable knobs and buttons, to let you tweak multiple parameters across several devices in a Combi.

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Master Plan One area in which many Reason users show considerable ingenuity is mastering — audio sweetening a Reason mix to sound as fat, vibrant and exciting as possible, without resorting to external hardware or software processing. The tools are there to do the the job in v2.5, but Propellerhead have just made it a whole lot easier, with the MClass suite of 'pro level' signal-processing tools. First up, the MClass Equalizer offers high and low shelving bands, two parametric bands and a low-cut filter. Great graphic feedback of EQ curves is provided by a large display, and the modelled circuitry is designed to offer you the tools to add 'bottom' or 'zing' in as musical a manner as possible. Then there's the Stereo Imager, which is a bit of a surprise. Not only does it provide control over the stereo width of a Reason mix, it does it over high- and The new Combinator turns Reason V3 into a low-frequency bands. The Compressor super performance synth. Any device in the Reason rack can be combined, layered and in the suite is several levels above the controlled from one handy window. original COMP01. It's a single-band device (though multi-band configurations should be easy to set up), complete with side-chain access to allow ducking and de-essing to be undertaken, and a smooth 'soft-knee' operation mode. Finally, the Maximizer works with perceived volume, aiming to add punch to your mix without unwanted artifacts. Reason is a digital system, so the Maximizer's best trick — a look-ahead option — should be no surprise. However, this one option should allow you to 'brick-wall' the level of your mixes without distortion, since it's checking incoming level peaks before they actually happen! Want a more 'round' sound to your maximized mix? Enable the soft-clip section. I haven't even covered all the V3 features yet — it also adds 'out of the box' integration with external MIDI controllers, a new free-text browser for digging around your sound bank, and a new, even larger factory sound bank. There's another new device, too: the Micromix stereo line mixer, for sub-mixing in or out of a Combinator Combi. This is a simple device, but not as simple as a Spider Audio. Six input channels have level, pan, mute and solo controls, plus one aux send. In addition, it'll now be possible to record automation on multiple tracks, there are new mute/solo options on the sequencer, and sample loading is to become much



faster. About the only new 'feature' that might be sad to hear (for Mac users, anyway) is that, based on the preliminary spec for Reason v3, it would appear that the software will no longer run under Mac OS 9. There, that's your reason to upgrade! Visit www.propellerheads.se to hear some audio demos and read about upgrade paths. This will be a chargeable upgrade for existing users, costing 99 Euros/US $129, or about £69. However, the list price of Reason 2.5 has recently been cut by that amount.

Before You Ask... I'll save you the trouble of reading the main text about the new features of Reason V3 again. On the basis of information released so far, audio recording hasn't been added to the Reason studio. Propellerhead seem quite steadfast on this issue, though surely it must come eventually. Even synth workstations, which Combinator is taking a pop at with its super-performance setup capabilities, are moving more in the direction of adding audio tracks alongside synthesis and sampling. I don't know what other Reason users feel is missing, but so much effort is expended in trying to create real-time arpeggiation that I'd also like to think that the Props will add a proper arpeggiator device at some point.

(Re)Fill 'Er Up The reliable Reasonfreaks team (www.reasonfreaks.com) have just released a top free Reason Refill, dubbed RF01. It's specifically for the NNXT advanced sampler and consists of a 10MB collection of multisampled raw synth waveforms and ready-made patches. The waveforms, which play back in high-quality 24-bit, seem to derive from soft synths rather than hardware originals, but that certainly doesn't negate the hard work that's gone into the set. First of all, audition the supplied patch collection: browse through them all, or have a look at the sets broken down into themes. These include pads, bass, leads, organs, rhythmic and noise FX. Then move onto the folder labelled 'NNXT Oscs'. Here you'll find the tidily multisampled raw oscillators, including many layered for pulse-width and phasey effects. There are several of each sawtooth, square, triangle and sine-wave patches, with two-oscillator varieties simulated again by layering.

The four new pro-spec devices that make up the MClass mastering suite: serious EQ, compression, stereo image manipulation and level maximisation.

Creating your own patches involves choosing an NNXT 'Osc' patch that most file:///H|/SOS%2002-05/Reasons%20to%20be%20cheerful%20%20v3.htm (3 of 5)9/22/2005 8:22:43 AM


closely represents the oscillator sound you want, then tweaking NNXT parameters, taking advantage of this device's very powerful synthesis engine. Should you want to use a combination of 'oscillators' that isn't provided in a preset, you can simply open two NNXTs and load one of the patches you want to merge into each device. Now highlight all the samples in one NNXT patch, copy them, and paste them into the second NNXT. The new patch (which you should make sure to save immediately, under a new name) will consist of both sets of 'oscillators'. It's then possible to use NNXT's Grouping function to apply different synth parameters to each 'oscillator', if desired. If the respective waveforms aren't already grouped automatically, highlight the ones that should make up each group, using 'Group Selected Zones' under the Edit Menu. (To select a group, click in the slim column to the left of the waveform list; this will highlight all the grouped waveforms.) Any synth-parameter tweaks will now apply to all the grouped waveforms. (Don't forget NNXT's global offset parameter controls: they can provide you with quick tweaks for any patch you're currently editing.) Reasonfreaks even provide a handy little PDF that goes into more detail about creating patches from scratch with their raw material. This is a very worthy download, covering most analogue synthesis basics in a way that's different from Reason's own Subtractor synth. In total, the set offers 21 waveform types, 80 NNXT oscillator patches, 250 further patches, and 17 Reason Song templates. The last adds effects and/or Matrix pattern sequencers to patches from the Refill, to give you some help in exploring the collection's potential. The 'freaks are also encouraging the Reason community to create patches with this new Refill, and post them at the site. In fact, there's already a download for use with this Refill — the bottom-heavy Xplosive Bass. Not bad for nothing: all you have to do is register! The collection can also be downloaded from Propellerhead's site, but again, downloads are for registered users only.

There Will Be A Long Delay I love playing with delays, temposync'd or otherwise. Delay, or echo, seems to me to be the quintessential electronic music effect, and Reason, with its DDL1 and the echo/delay algorithm options in the RV7000 advanced reverb, is a prime environment for working with delay effects. In fact, it was playing with the RV7000's multitap delay algorithm that reminded me that a sort of multitap delay can be created with multiple DDL1s. Simply patch, for the sake of example, an aux send out from a

A simple multitap delay setup. The DDL1s are linked via the left ins and left outs, with



their right outs patched to Remix input Remix to one DDL1 and pass that channels, to allow each tap to be panned DDL1's output to the input of a second and treated independently. Incidentally, the DDL1, the second's output to a third's NNXT has a patch from the Reasonfreaks input, and so on for as many delay RF01 Refill, mentioned in the main text, taps as you would like. You could stop loaded. here: set each delay's feedback to '0', ensuring it has a completely wet output, whack the delay time up to maximum (2000ms) and enjoy some really long, Frippertronic-like delays. Even the RV7000 delays max out at 2000ms, so a way to access even longer delays is welcome. (Note that the feedback parameters on any devices in the chain won't work in the same way as if the long delay were produced by a single device.)

Alternatively, you could take the next step, which is a bit fiddly but worth it. DDL1 is essentially mono, so repatching its right audio output to a Remix input channel has no effect on what it produces. However, doing this does give us complete control over the level, pan, EQ and individual effects processing for each of our 'taps'. The last DDL1 in the chain will still output a delay at the maximum time of all linked delay devices. The RV7000's multitap algorithm does offer some of the same kind of control, but the final effect is not quite the same. Work with tempo-sync'd delays instead of absolute time and you'll find some excellent rhythmic material that can't be replicated in any other way coming from your Reason rack . Use long or short delays, as befits your current track, and for more variety add an RV7000 to the chain, using its own multitap delay or echo algorithms to spice up the delay mix (why not try the reverse algorithm to add some real bizarreness?). Settings such as diffusion, which add a nice, reverb-like fuzziness to delay and echo taps, aren't available on the DDL1, and can help take some of the dryness from a delay-only effect chain — although dryness, of course, may be entirely what you're after! Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:


An MP3 Encoding Alternative Sonar Notes Look — No Hands Published in SOS February 2005 The Now Time Collapsing Tracks Print article : Close window

Technique : Sonar Notes

This month, find your way around projects, undertake hands-free recording, configure an external MP3 encoder and create pseudo-Track Folders for MIDI. Craig Anderton

Quite a few of the new features in Sonar 4 seem fairly straightforward — until you dig a little deeper. Such is certainly the case for the Navigator view, which is more than just an overview; it's a powerful tool for getting around a project. The Navigator view is a strip that sits in the upper part of the Track View and shows the entire project at a glance. You can make this strip appear and The Navigator view provides another point of entry for accessing commands that relate to disappear with the Show/Hide sizing projects and fitting them to the Navigator button (located above the available space. Tracks Pane), as well as dragging the separator bar between the Navigator view and Track view up or down to change the Navigator view's height. On top of the strip is a resizable rectangle — and that's where the fun begins: the area defined by the rectangle also defines what you'll see in the Clips pane. Make the rectangle smaller to 'zoom in' on the tracks on the Clips pane; enlarge the rectangle and you see more tracks. You can also click inside the rectangle and drag it around, which acts like putting a magnifying glass over the tracks. This is all fairly basic, and a lot of Sonar users could stop here and still get a lot out of the feature. But there's more. Right-click within the Navigator view strip and you'll find multiple options. file:///H|/SOS%2002-05/Sonar%20%20Hands-free%20Recording%20&%20Other%20Tips.htm (1 of 5)9/22/2005 8:22:49 AM


Many of these commands (Fit Project to Window, Hide Selected Tracks, Track Manager, etc) duplicate those found under the 'View Options' button in the toolbar above the Tracks pane. In any event these commands, regardless of how you invoke them, now affect both the Clips pane and the Navigator. You can also select one of six horizontal zoom levels. This may seem redundant — isn't it simple enough just to resize the rectangle? Yes, but making the zoom levels part of the pop-up menu options means that key bindings can trigger them. This is particularly useful for the 'Horz Zoom to Project' function, as hitting a key can instantly fit your entire project within the Clips pane and Navigator view. The other pop-up menu selection involves track height within the Navigator view. For projects without a lot of tracks, I recommend the Tall option, as that makes the tracks much easier to see. There's one other cool navigation technique. Go View / Navigator, and the Navigator appears in a separate window that supplements the main Navigator strip. Resizing the window resizes the contents as well, so you can do anything from floating the Navigator to a second monitor and maxing out its size for easy navigation, to slimming it down and using it as a floatable Navigator strip on your main monitor. It takes a little effort to break old habits and start to think of the Navigator as your primary way to get around a project. But once you do you'll find yourself zipping around in much less time.

An MP3 Encoding Alternative Many Sonar users chose the CWENC MP3 encoder as an alternative to Sonar's 'optional at extra cost' MP3 encoder. Unfortunately, it's not possible (at least, not yet) to get CWENC to work with Sonar 4. However, there's still a free encoder option: the LAME encoder. First, you have to download the LAME software. It's widely available on the web, and I found a copy at www.free-codecs.com. Now, I can't comment as to the legalities of using the LAME encoder without paying royalties to those who invented the MP3 format, and LAME is said to be for educational purposes only. However, it seems that no one's getting prosecuted for doing the occasional MP3 conversion using 'freebie' tools (although if MP3 is a significant part of what you do, you may do your conscience good by getting an 'official' encoder). Second, you need to take advantage of a new Sonar 4 feature, 'Configure External Encoder', to get LAME to work. This involves installing a profile for the encoder and telling Sonar where to look to find the encoder it's supposed to use. Insert the Sonar 4 distribution CD-ROM into your drive, as this is where the encoder profiles dwell. Don't let it auto-run; instead, Explore the CD and open the Utilities folder. Inside that, open the External Encoder Profiles folder. Now double-click on any of the encoder options you want to add (for example, LAME MP3 High Quality Stereo.reg). You'll be asked if you want to add this information to the

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registry. You can now tremble in fear for a bit, because you're adding something to the registry, but go ahead and click on 'Yes' — everything should go smoothly, and you'll be informed that the profile was added. Next, look in the Shared Utilities folder (in C:\Program Files\Cakewalk), and double-click on the EncoderConfig.exe program. In the 'Friendly Name' field, the Profile you just installed (LAME MP3 High Quality Stereo) should be listed; select it. In the 'Path' field, delete the default path and enter the path to the LAME encoder folder installed on your computer. I put the LAME folder within Shared Utilities, but it can sit anywhere. path, then click on 'Save'. If you don't save, Sonar won't be able to launch the encoder. Finally, click on 'Close', to close EncoderConfig.exe.

The next time Sonar is open and you go to Export Audio, any of the profiles you configured will show up as export options. Now go forth and encode!

Look — No Hands I've taken quite a liking to the Digitech GNX4 recorder/guitar processor, for several reasons, but the one that's of the greatest interest to Sonar users is that it's supported as a control surface within Sonar (called 'Digitech hands-free recording'). The GNX4 has a USB interface that communicates with the computer running Sonar. Once connected, the GNX4 footswitches control several functions. When you click on the Record footswitch, this creates a stereo pair of tracks in Sonar that defaults to recording two channels of audio coming in over USB from the GNX4 (guitar, mic, line ins, drum machine, etc), record-enables the two tracks, then automatically commences recording. (The GNX4 can also send a second pair of audio tracks via USB. If a source is also assigned to these, Sonar will create, record-enable and start recording with another set of tracks whose inputs are assigned to this second pair.)

The Now Time can either snap back to where playback started, or can stay where it was when the transport stopped. However, the Loop/Auto Shuttle settings also influence the Now Time position.

The GNX4 Play footswitch works as expected, as it initiates playback in Sonar, and you can also stop recording with the GNX4 Stop/Undo footswitch. But something more interesting happens if you press and hold the Stop/Undo footswitch: whatever you recorded last will be deleted. You can keep deleting back through tracks: just release the footswitch, then press and hold again to delete the previously recorded track.



In some ways this process resembles loop recording, in that recording functions are automatic and you can initiate more tracks for recording simply by pressing the Record footswitch. Unlike loop recording, though, you can fiddle around between takes. For example, what sold me on the hands-free feature was recording a sample CD of guitar licks. It was easy to record track after track, making slight changes to alter the sound of the samples. While I'm not sure that hands-free recording alone justifies the price of the GNX4, if you're a guitarplaying Sonar user in the market for a signal processor that includes a lot of other features, hands-free recording is a major bonus.

The Now Time The Now Time is where the cursor sits, and is the point where recording or playback begins. However, you have some leeway over where the Now Time is going to stop. There are two general options, as selected in the Options / Global / General tab. If you tick 'On Stop, Rewind to Now Marker', when the transport stops the Now marker will return to where it was before playback began. This is the probably the best option to use if you're overdubbing parts over the same section of a tune.

The selected MIDI clips have been copied and are about to be pasted into one track, thus collapsing them into a single clip on a single track.

If you untick 'On Stop, Rewind to Now Marker', when the transport stops the Now Time will stay at its stopped position. If you click on 'Play', the Now Time will now continue playback from where it left off. This is the option I tend to use while composing, where I'll block out sections of a tune without being too concerned about the part itself — the goal is simply to get parts down before the inspiration goes away. There is one other factor regarding where the Now Time ends up, as found under Transport / Loop and Auto Shuttle. This matters only if you've looped a portion of the project and looping is enabled. With 'Stop at the End Time and Rewind to Start' ticked, when the Now Time reaches the loop end the transport will stop automatically and the Now Time will rewind to the loop start. Unticking this turns off looping. If you'd prefer to have looping run continuously, tick the 'Loop Continuously' box. And what happens if you click on 'Stop' before the Now Time hits the loop end? That depends on whether 'On Stop, Rewind to Now Marker' is ticked or not under Global / Options. If it's ticked the Now Time will snap back to the beginning of the loop. If unticked, it will remain at the stop point.



Collapsing Tracks If you still have Sonar 3 but envy the Track Folder feature in Sonar 4, there's a workaround for MIDI tracks. While this technique is nowhere near as flexible as Track Folders, it does help tidy up groups of MIDI tracks. Consider this scenario: you've created a MIDI drum part over multiple tracks, with each drum sound getting its own track. But now you've tweaked the part and you want to collapse all the drum sounds into a single track. Select the clips you want to collapse (note that they all must be MIDI tracks — no audio allowed), then go Edit / Copy. Make sure that 'Events in Tracks' is ticked, then click on 'OK'. Place the Now Time where you want the paste to start, and go Edit / Paste. This dialogue box lets you select the track where you want all the new parts to reside, or creates a new MIDI track for you if desired. Make sure you click 'Paste to One Track', or collapsing won't occur (see screen, left). It's also possible to specify multiple repetitions and, if you click on the 'Advanced' tab, whether you want to blend the data with an existing track. Click on 'OK' and the parts will be collapsed as specified. Archive, then hide, the source tracks; they'll be out of the way, but if you want to make changes later on you'll be able to recall them. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

Out Of The Box What The DVI? How Many Channels? Four Million Pixels, Sitting On My Desk... About The Tests I've Got The Power (Mac)

Apple 30-inch Cinema Display Apple Notes Published in SOS February 2005 Print article : Close window

Technique : Apple Notes

In an extended Apple Notes column we take an exclusive look at the 30-inch Cinema display from a musician and audio engineer's perspective, and evaluate the performance of the new dual-2.5GHz Power Mac G5 with our usual series of performance tests. Mark Wherry

Forget the fact that it's actually just a computer screen. There's just one word that sums up Apple's new 30-inch Cinema display: wow. The company revamped their line-up of Cinema displays back at this year's Worldwide Developer's Conference with a new, sleeker aluminium-styled enclosure, and while the 20- and 23-inch models have been readily available for a few months now, the king of the new range, the 30-inch model, has now also started hitting Apple's distribution channels.

Photo: Abhay Manusmare. Wine: Hans Zimmer If you can afford to drink wine like this, you should be able to afford Apple's impressive 30-inch Cinema display, seen here showing Logic Pro 7.

Out Of The Box As you unpack the 30-inch Cinema display, one very obvious fact becomes apparent: it's big. So big, in fact, that it's not even worth cracking jokes about size, and so big that everyone who sees the display will remind you just how big it is. Specifically, the 30-inch model measures 27.2 inches (68.8cm) by 21.3 inches (54.3cm) and has a depth of 8.46 inches (21.5 cm), including the stand. file:///H|/SOS%2002-05/Apple%2030-inch%20Cinema%20Display.htm (1 of 7)9/22/2005 8:22:54 AM


The display panel itself is actually only about two inches deep. And although it's a fairly substantial unit to pick up, this 30-inch screen is far easier to move around than an old 17-inch CRT display, weighing in at just 27.5 pounds (12.47kg). Offering a resolution of 2560x1600 pixels, the 30-inch display is also a significant step up from the 1920x1200 resolution offered by most 23-inch displays available. It's also perhaps the most aesthetically impressive display Apple have ever produced, and shares a clear design philosophy with the new iMac G5. The iMac and Cinema display models are supplied on a similar stand that allows the display to be tilted up or down between -5 and 25 degrees (where 0 is perpendicular to the desk), although the display looks good when viewed from a variety of angles. Alternatively, the stand can be removed completely so that you can mount the display almost anywhere you like, with the optional VESA mounting kit. One interesting thing to note about the new line-up of Cinema displays is that they don't require fans internally for cooling, which obviously makes them pretty much silent when compared with other displays that do have fans, and ideal for use in music and audio studios. The lack of fans is partly made possible by the fact that the power supply is kept outside of the monitor enclosure, although the displays do get noticeably warm to touch. Still, this didn't appear to be a problem, even when leaving the monitor switched on for extended periods of time.

Logic Pro 7 looks great on Apple 30-inch Cinema Display, and from this diagram you can appreciate how much you can see on the 30-inch display compared with Apple's 12-inch (1024x768), 15-inch (1280x854), 17inch (1440x900), 20-inch (1680x1050) and 23-inch (1920x1200) displays.

For this new range, Apple have moved the controls for brightness and power to the lower part of the right edge. This is alright, but in the unlikely event you wanted to put two Cinema displays side-byside, it could become awkward to reach the controls on the left-hand display. Not a big complaint, obviously, since you can always control the monitor from the computer. (On a related point, it would be nice if the monitor's power button could switch on the whole system, rather than just being able to power it down, put it to sleep or wake everything up again.) In addition to a built-in two-port USB 2.0 hub, the new displays also include a two-port Firewire 400 hub. In order to drive the 30-inch Cinema display you'll need a Power Mac G5, and you'll also need to purchase a new graphics card from Apple, specifically the Nvidia Geforce 6800 Ultra or GT models (see the 'What The DVI?' box for more information). This could pose a problem for some users, since the cooling assembly on the card has an increased width over the majority of graphics cards supplied with a Power Mac G5 as standard. In fact, it requires the space (and back-plates) of both the AGP slot and the first PCI slot. If all the PCI slots in your

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Power Mac are currently full and you want to use the 30-inch Cinema display, an expansion chassis is definitely in your future. For testing, I installed the Nvidia Geforce 6800 Ultra in a dual-2GHz Power Mac G5 (bought after the Power Mac G5's original introduction at the end of 2003), which had 2GB RAM and the factory-installed 250GB SATA drive. I filled the remaining accessible PCI-X slots with a Pro Tools HD Core card and an HD Accel card so I could see what Pro Tools looked like on a 30-inch display, and also to see if there were any obvious problems with using the three cards, including the graphics card, in this way. As a side note, you need to be running at least Mac OS 10.3.5 to use the newer graphics card, and it's therefore imperative to make sure your Mac is up-to-date via Software Update before swapping out the graphics card. Although the Nvidia card is supplied with a driver CD, this is unnecessary if you're running Mac OS 10.3.6 (or higher), since all the software required is now part of a standard OS X installation. A resolution of 2560x1600 pixels is all well and good, of course, except for the question of what exactly you're going to use all 4,096,000 of them to display...

What The DVI? Unlike the previous generation of Cinema displays, the new 20-, 23- and 30-inch models featuring an aluminium enclosure no longer use Apple's ADC interface for connection to your Mac. ADC (Apple Display Connector) was an interface introduced in later ranges of Cinema displays to provide a single cable and connector between the computer and monitor for power, video and USB. It was convenient, but it meant that a converter was needed to use the display with any computer other than a Power Mac (including Power Books and any non-Apple system), or as a second display with a Power Mac. In place of the ADC connection on the new Cinema displays is a single cable that splays out into individual DVI, USB, Firewire and power connections on the other end, providing better compatibility with a wider range of systems. On the downside, now Power Mac users have the opposite problem: you'll need to purchase an ADC-to-DVI adaptor if you want to use two of the newer Cinema displays with your system. While the DVI connections on all Cinema displays look identical, the 20- and 23inch models use what's known as Single-Link DVI, compared to the 30-inch display, which uses Dual-Link DVI instead. Single-Link DVI is the most common type of DVI connection, supported by the majority of graphics hardware currently in circulation, including the graphics cards supplied with all DVI-based Macs, past and present, and is so called because it uses only 12 of the 24 pins available on a DVI connector. Single-Link DVI supports a maximum bandwidth of 165MHz to give a resolution of 1920x1080 at 60Hz (or 1280x1024 at 85Hz), even though most displays and graphics cards can usually push this to 1920x1200 — the resolution of the 23-inch Cinema display, incidentally. To drive the 2560x1600 resolution of the 30-inch Cinema display, Single-Link DVI



obviously wouldn't work, so the 30-inch display uses Dual-Link DVI instead, and needs a graphics card that supports Dual-Link DVI to drive it. Dual-Link DVI carries, as the name suggests, two 165MHz signals for double the bandwidth, making use of all 24 pins on the DVI connector.

How Many Channels? How does an application such as Logic or Pro Tools, for example, actually make use of such a large display? How many mixer channels can you expect to see? How many tracks on the Arrange or Edit window can you have on screen simultaneously? You can get a basic idea from the illustrations in this article, but I can also be a little more specific about this issue. Taking Logic Pro 7 as our first example, the Arrange window can display 74 tracks, assuming you stick with the initial default layout of the Arrange window and track height. With the smallest track height you'll be able to see a staggering 148 tracks, and while the tracks might be considered too small to be practical at this height, it might help you make a useful Screenset to get an overview of the arrangement. Alternatively, setting the track height to the largest setting gives you just six tracks. In the Environment window you can see 42 audio channels horizontally and four vertically, given a reasonable number of Inserts and Sends, which means you could see a total of 168 audio channels (4x42). Moving on to Pro Tools, the Edit window can display quite a substantial number of tracks, and I also had the Bars and Beats, Timecode, Tempo, Meter and Markers Rulers displayed. With the track height set to Mini you can get 73 tracks — or 38 Small, 15 Medium, seven Large, four Jumbo, or two Extreme tracks. On the Mixer window I could see either 63 (61 with the Show/Hide Channels and Mix Groups column available) narrow channels or 35 (34) wide channels.

Four Million Pixels, Sitting On My Desk... Figuring out how to best make use of an application at such a large resolution can be more challenging than you think. Current music software is usually designed with multiple windows where each window is given a specific task: an editing window, a mixing window, and so on. This type of user interface design lends itself well, of course, to multiple monitor systems where you can put a different window on each monitor; but often this approach won't work so well on a large display where you have to arrange multiple windows to make one big work area. Indeed, if you take a look at any one window maximised to fill the 30-inch display (Logic's Arrange or Pro Tools' Edit windows, say) you'll notice that most of the controls get huddled into the top-left corner of the screen to fit snugly with the kind of resolutions used by the larger majority. While it could be argued that most users won't be buying 30-inch displays just yet, it does give software developers an interesting challenge: how do you create file:///H|/SOS%2002-05/Apple%2030-inch%20Cinema%20Display.htm (4 of 7)9/22/2005 8:22:54 AM


an interface that works at all resolutions? One interesting and, in my opinion, successful example of this type of user interface design can be seen in another Apple product, the video compositing software Shake. In this application there's only one main window that's divided into four sections (or should that be panes?) which can be resized to show more or less of the timeline, video viewer, parameters, and so on.

Running Pro Tools on a 30-inch display is useful, since you can take a split-screen perspective, placing the Mixer window on one half of the screen and the Edit window on the other.

If you run Shake on a larger display, you simply have more space to work with each area of the program, and while running the software on a larger display is more comfortable, the application is equally functional at lower resolutions. One neat trick, admittedly borrowed from X Windows servers on Unix, is the ability to expand the screen with just one section of the main window, such as the video viewer, for example. In this way, even on smaller displays you can always make the most of the pixels you have by filling the whole screen with one part of the application when that's the one you really need to be focused on. It's worth noting that the user-interface design of most of the music and audio applications we rely on every day hasn't changed much since the early '90s, and it will be interesting, especially with Apple providing these larger displays to creative professionals, to see how application developers change tack to make the most of the hardware available to them. In conclusion, the 30-inch Cinema display is an impressive product. Aesthetically it's beautiful to look at, and visually it's bright and sharp. Although you might think it's potentially too big initially, especially when you find yourself putting a window 'out of the way' in the lower part of the screen, it's actually a perfect, workable size. From a cost point of view, at £2549 the 30-inch display doesn't seem overpriced compared to what smaller Cinema displays cost a few years ago, although some users may wonder whether two 23-inch ones for close to the same money (the 23-inch retails for £1549 and doesn't require a new graphics card) may be a better buy. The Nvidia graphics card required for the 30-inch display costs an additional £449 for the 6800 Ultra or £349 for the 6800 GT. However, you can buy the 6800 Ultra with a new Power Mac for £379, or £339 with the dual-2.5GHz model; or the 6800 GT for £279 or £269.



About The Tests As with previous performance tests in Apple Notes, the following guidelines were used. The audio hardware was the Mac's built-in hardware, using the analogue connections with a buffer size of 512 samples, which equates to approximately 12ms latency running at 44.1kHz. The User CPU usage readings were taken as an average from OS X's Activity Monitor application, found in the Applications / Utilities folder. With the Platinumverb tests, the default settings were used. With the Space Designer test, the 'Large Hall 2.6sec' preset was used. For the EXS24 test, the 16-bit HA_ES instrument from the EXS24 edition of the Vienna Symphonic Library was used.

I've Got The Power (Mac) In the February and March 2004 Apple Notes, I wrote about the performance of the then top-of-the-line dual-2GHz Power Mac G5, looking at how many reverb instances and sampler voices could be achieved (see the 'About The Tests' box for information), and over the past year we've looked at similar tests on other Mac systems. This month I managed to borrow a friend's dual-2.5GHz Power Mac G5 (my thanks to Geoff Zanelli), so I thought I'd run the same batch of performance tests to see what's possible with the highest-performing Mac Apple has ever released. Since I've used Logic 6.4.x throughout the last year, I decided to stick with Logic Pro 6.4.3 for the main testing, as using Logic Pro 7 instead would skew the results if it was more or less efficient than previous versions. However, I did install Logic Pro 7 after the tests, just to see whether it is indeed more, less or the same in terms of efficiency. We'll look at this after discussing the main results. The Power Mac G5 in question was the factory-specified dual-2.5GHz model, but with 4GB RAM installed and two 250GB internal SATA drives. It had been updated to the latest software versions available at the time of writing, including Mac OS 10.3.6. Starting with the exciting 'how many Platinumverbs' tests, the dual-2.5GHz Power Mac G5 managed 140 instances, reporting approximately 93 percent User CPU usage in the CPU panel of Activity Monitor, which marks a small improvement over the 120 instances attainable with the dual2GHz a year ago. Staying with reverb, the next test was with Space Designer. The dual-2.5 GHz Power Mac G5 is capable While the dual-2GHz could run 29 of running 140 instances of Platinumverb. instances with 82 percent usage, the dual-2.5GHz model achieved 30 instances without breaking into a sweat (75 percent User CPU). I was able to run file:///H|/SOS%2002-05/Apple%2030-inch%20Cinema%20Display.htm (6 of 7)9/22/2005 8:22:54 AM


up to 32 instances fairly comfortably (on 83 percent usage), although this result was effectively voided by the occasional Core Audio Overload message. The EXS24 sampler voices test was, perhaps, more revealing. Using 16-bit storage mode and no filters enabled, I was able to play 752 voices concurrently (11 instances playing 64 voices, plus one instance playing 48 voices) with 87 percent User CPU usage, compared to the 704 voices attained by the dual-2GHz Power Mac G5 last year. Not bad. However, when switching to 32-bit storage, again with no filter, I was able to play back 3648 voices (57 instances at 64 voices) with 87 percent usage, compared to 3264 voices on the dual-2GHz model. So the dual-2.5GHz really shows a benefit for EXS24 users, although the improvement is very much in keeping with what you would expect the performance boost to be just by looking at the numbers — 2GHz to 2.5GHz. With the filter enabled, the number of voices dropped fairly sharply, as you would expect. With 16-bit storage and filters active, the dual-2.5GHz Power Mac G5 played back 240 voices (three instances at 64 voices and one instance at 48 voices) with 84 percent User CPU usage; with 32-bit storage and filters active the same system was capable of 364 voices (five instances at 64 voices and one instance at 44 voices) at 92 percent usage. This is compared to 192 and 256 voices for 16- and 32-bit storage respectively on the dual-2GHz Power Mac G5. As I mentioned at the start of this section, after recording these results I looked briefly at how Logic 6.x compared with Logic 7 and found that there wasn't any significant difference in terms of audio performance on a Power Mac G5. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Beat Detection in DP 4.5

In this article:

Preferences Quick Tips Horses For Courses Practical Applications

Beat Detection in DP 4.5 Digital Performer Notes Published in SOS February 2005 Print article : Close window

Technique : Digital Performer Notes

The new Beat Detection Engine in DP 4.5 gives you much greater control over audio quantising and tempo manipulation. We explain how. Robin Bigwood

It's notable that the new Beat Detection Engine is being given just about top billing in all the DP 4.5 advertising, but while most DP users are familiar with the concepts of timestretching and Recycle-like loop slicing, this new feature appears to be something else again. So what exactly does it do, and how can it help you? Well, primarily the Beat Detection Engine searches out the transients or 'attacks' in rhythmic audio files and does nothing more than make a note of where and how strong they are. Your audio plays back in the same way whether the Engine is turned on or off, and changing the tempo of your sequence does not automatically result in the sort of tempo manipulation that characterises Ableton's Live. Beat Detection, instead, is just the precursor to a clutch of interesting possibilities, including: Allowing audio in soundbites to be quantised, timestretching audio between beats to maintain the best possible audio quality.

It's not terribly exciting, but the Background Processing Preferences in DP 4.5 are important for configuring Beat Detection.

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Taking the rhythmic groove from an audio file and applying it to other audio files or MIDI. What's more, Beat Detection also allows DP to have a good guess at the tempo of the audio in your soundbites, making it easier than it used to be to work with rhythmic audio of varying tempos — even audio that wasn't recorded to a click. Tempo analysis is a rather different beast to beat detection, though, and I'll be looking at this in more detail next month. As is typical of many a new DP feature, Beat Detection is far-reaching enough to impinge on lots of other existing features, and to manifest itself in a number of windows and menus. One thing's for sure — MOTU never do things by halves!

Preferences Before starting to work with the Beat Detection Engine, it's worth visiting the new consolidated Preferences window, accessible from the Digital Performer menu, and clicking on the Background Processing list item. It's here that several important Beat Detection settings are made. Starting at the bottom, the three 'Beat Analysis Defaults' options determine when DP automatically looks for beats in soundbites. They're pretty self-explanatory, and can be summed up as: never; only for new projects; and for all projects, including old ones. The two checkboxes above are a little less clear, especially as the 'hint' text for one of them doesn't seem quite right. Basically, the top checkbox just turns on beat detection for the current project, while the bottom one enables automatic tempo analysis of soundbites. Audio must have had its beats detected for any tempo analysis to take place, though. The settings in the upper part of the window are perhaps more familiar, but may be just as important. The topmost pair of radio buttons controls when DP analyses audio for timestretching and pitch-shifting (as it has long been able to do). The new default is to analyse as soon as possible, since DP's PureDSP algorithm is used in conjunction with the Beat Detection Engine. Following an 'old' preference that controls how DP deals with soundbites that are being processed, there are some options for DP's background processing window. These have changed in DP 4.5 because background tasks are now deemed either 'high priority' or 'analysis'. If you set the Background Processing Window to show only high-priority tasks, things like beat detection and tempo analysis won't show up, although they're still taking place.

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Quick Tips In the aux send section of DP4.5's Mixing Board, clicking the new, tiny 'P' button next to the send level knob makes a send pre-fader. Pre/post can also be configured in a send's pop-up menu. As this month's Performer Notes neared completion I got some news from MOTU about the impending release of DP 4.51. This adds some intriguing new features, such as dynamic CPU management and better compatibility with Mackie's UAD1 and TC's Powercore cards. Check out www.motu.com for the download.

Horses For Courses The default settings for background processing are a good starting point for using the Beat Detection Engine, but there are plenty of users for whom they'll represent wasted processor cycles — live recording purists, classical music editors, mastering engineers and spoken-word recordists, amongst others, whose audio isn't really compatible with the Beat Detection algorithm. If you definitely don't want to make use of it, you can disable it completely by deselecting 'Automatically analyse beats in this project' and choosing 'Never look for beats in audio files', again in the Background Processing Preferences pane. And if some of your project's audio has already been subject to beat analysis, you can discard this completely by trashing the Analysis folder in your project's folder. Any beat display remnants in soundbites can be eradicated by deselecting 'Show beat grid lines' in the Graphic Editor's Preference pane.

Practical Applications Assuming you're using DP's background processing defaults and you want to make use of the Beat Detection Engine, here are a couple of practical scenarios:

1. The sober (but not so tight) drummer: Most musicians find it hard to really 'groove' with a click track, drunk or not, but for styles which juxtapose live recording with 'programmed' parts, the click is a necessary evil. Using the Beat Detection Engine's ability to quantise live audio, any deviations from the beat can be quickly quantised right back on to it. After you've recorded your drum track, DP tries to detect beats within it. Any that it finds are marked with vertical blue-grey lines superimposed on soundbites shown in the Sequence Editor. DP tries to find as many beats as it can, but the number that are active at any one time can be adjusted by first selecting the soundbite(s), and then choosing the Adjust Beat Sensitivity window via the Audio



menu, or with the keyboard shortcut control-D. Dragging the slider to the left in this little window disables more and more beats, whilst dragging it all the way to the right enables all of them. You're looking to get DP to find beats on the really important transients, particularly ones that fall on the 'strong' beats of the bar. If you can achieve that, you hit Apply and then you're ready to quantise. First, select the A beat-detected soundbite, viewed in the Waveform Editor. Here, individual beats can section of audio that needs quantising, be enabled or disabled, via the Beats menu, perhaps using the I-Beam tool or which is open in the screen above. double-clicking a soundbite and dragging over a section within it — so called 'pop-editing'. You could, alternatively, select a whole soundbite, or several. Then choose 'Quantize...' from the Region menu, or using Apple-0 (zero). Under the 'What to quantize' pop-up choose 'Beats within Soundbites', and below this choose a suitable note value. This setting works differently than it does with MIDI, in that DP doesn't try to quantise every beat in the audio, just the ones nearest to the time-value divisions you choose. Beats occurring between timevalue divisions are 'spaced out', to maintain a natural feel and to minimise the impact of the DSP time-stretching that will occur with quantising. So you can often choose pretty large time values here (crotchets/quarter-notes), or at least larger values than the very smallest sub-divisions of the musical beat within the soundbite, which should look after themselves to a large extent. Finally, click Apply in the Quantize window to fix your audio. But what if DP refuses to find useful beats in very complex audio, or is plagued by lots of 'false positives'? The thing to remember is that you can override its beat decisions, creating new ones and disabling incorrect ones, which should make almost any audio 'quantisable'. This is done in the waveform editor. First, select the troublesome soundbite, then choose 'Edit in Waveform Editor' from the Audio menu. In the Waveform Editor click on the Beats tab, and you should now see all the beats DP detected in the audio (see screen above). Enabled beats are dark blue and have an intensity slider, while disabled ones appear grey. You can again use control-D to access the Adjust Beat Sensitivity window, but here you have full manual control too. If you point to a beat you get a 'pull hand' mouse cursor: clicking selects a beat (and also plays it), while clicking and dragging moves it. With a beat selected you can choose 'Delete Beat' from the window's Beats menu to get rid of it completely, or 'Toggle Active Beat' to disable it. The left/right arrow keys on your keyboard can be used to move between beats, and I set up the keyboard shortcut '/' (slash) to Toggle Active Beat (using the Commands window). This way it's very easy to step through a large number of beats, turning various ones on and off. If you need to create a beat where there just isn't one, click within the waveform to place an insertion point, then choose 'Create Beat From Insertion Point' (in the Beats menu). After coming up with the perfect selection of beats in this way, you're ready to quantise, as before. 2. Editing beat-detected audio: If you're at all familiar with DP you'll already file:///H|/SOS%2002-05/Beat%20Detection%20in%20DP%204.5.htm (4 of 5)9/22/2005 8:24:36 AM


know about the Sequence Editor's Edit Grid. This can be turned on or off by checking the mysterious, unnamed box at the top right of the window. The grid's time division can be set with the pop-up menu next to that box. When the grid is on, a dragged soundbite will snap to positions relative to the edit grid. The same goes for selections you make within and across soundbites with the I-Beam tool, and by dragging in the lower section of soundbites with the arrow tool. In DP 4.5, lurking next to the Edit Grid checkbox is a new Beat Grid checkbox, and this can be used instead of, or in addition to, the Edit Grid. With this turned on by itself, drags, selections and the scissors tool snap to beats. What can this do for you? Well, it makes splitting soundbites, perhaps to cut out a single snare beat, simplicity itself — you just use the scissors tool, click near to the beat markers surrounding a beat, and you should have a perfectly cut soundbite. A section of audio, isolated in this way, can then be dragged over other soundbites, at which point it will snap to their beats. This makes it ludicrously easy to replace individual beats and even notes — in a bass guitar part, for example. With the Edit Grid checkbox enabled as well, you have even more options. If your audio was recorded with a click and is mostly in time, you could try isolating the most useful longer sections and then duplicating them to build an entire track. In this case it may be better to split soundbites at bar rather than beat boundaries, so that timing relative to the original click is maintained, with possibly less need for audio quantising of duplicated sections. We'll be covering lots more about beat detection next month, including working with audio and grooves, tempo detection, and applying drastic tempo changes to recorded audio. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

Battery Powered A Killer Track Cupboard Love The Discipline Of Tape Keeping The Vibe The Best Possible Sound

CLASSIC TRACKS: The Stone Roses 'Fools Gold' Producer: John Leckie. Engineers: John Leckie, John Cornfield, Paul Schroeder Published in SOS February 2005 Print article : Close window

Technique : Recording/Mixing

As the '80s drew to a close, the Stone Roses made rock music cool again, melding '60s psychedelia and acid house under the production guidance of John Leckie. Richard Buskin

At the forefront of the late-'80s 'Madchester' scene that merged '60s guitar pop with dance rhythms, the Stone Roses made an indelible mark on the British rock canon. Their selftitled 1989 debut album opened the doors to a host of other UK bands, including the Inspiral Carpets and Happy Mondays. Featuring the haughtily laid-back vocal style of Jaggeresque frontman Ian Brown, catchy hooks of guitarist John Squire, and infectious rhythms of bassist Mani and drummer Reni, The Stone Roses sandwiched anthems such as 'She Bangs The Drums' and 'Made Of Stone' between trance-like opener 'I Wanna Be Adored' and epic closer 'I Am The Resurrection'. 'She Bangs The Drums' became the Manchester band's first top 40 single, yet it was the funky 'Fools Gold', recorded after the album's UK release and subsequently added to American pressings, that finally cracked the top 10, reaching number eight that November. During a halcyon year, and preceding ruinous legal wranglings with Silvertone Records, this single would prove to be the Stone Roses' crowning achievement, and it was also among the career highs of producer/engineer John Leckie, who in the next decade would be responsible for such classics as the Verve's A Storm In Heaven, Radiohead's The Bends and Kula Shaker's K.

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Originally a tape-op at EMI's Abbey Road Studios, Leckie had assisted on early-'70s projects such as John Lennon's Plastic Ono Band, George Harrison's All Things Must Pass and Pink Floyd's Dark Side Of The Moon, before making his production debut with Be-Bop Deluxe's Sunburst Finish album in 1976. After going freelance a couple of years later, he'd then worked with the likes of XTC, Public Image Limited, Simple Minds and the Fall, and it was on the strength of this resumé that, towards the end of 1988, he became involved with the Stone Roses. "I initially received a demo tape from Geoff Travis at Rough Trade," Leckie recalls, "and by the time I got back to him they'd signed with Silvertone. I then got the same kind of tape from Silvertone, so it was clear that the band had asked for me to work with them... I can't now remember what songs were on the tape, but I know they included 'Waterfall', 'She Bangs The Drums', 'This Is The One' and 'I Am The Resurrection'. "Basically, I was brought in to do my thing. The A&R guy, Roddy McKenna, was quite a tough Glaswegian, and he told me 'You're gonna make this good.' He didn't specify what this meant, but he'd come down to the studio and tell us when he thought something was good or something was bad."

Battery Powered After mixing the Roses' third single, 'Elephant Stone', Leckie spent a few days rehearsing with them in Manchester before the album sessions commenced at North London's Battery Studios, where four tracks were recorded within the space of 10 days: 'I Wanna Be Adored', 'She Bangs The Drums', 'Waterfall' and 'This Is The One'. 'Waterfall' would subsequently be re-recorded at Rockfield in Wales. Utilising an SSL E-series console and Studer tape machines, the Battery sessions took place between about seven at night and nine in the morning, and the day after they ended, the Studio Two complex was dismantled and converted into offices. Thereafter, the relocation to Rockfield saw the aforementioned remake of 'Waterfall', as well as tracking sessions for '(Song For My) Sugar Spun Sister', 'Made Of Stone', 'Shoot You Down' and 'I Am The Resurrection'. There, Leckie pushed the faders on a Neve, alongside Battery in-house engineer Paul Schroeder, who until then had never worked anywhere else. "He had only worked an SSL, and he'd also only really worked with drum machines rather than a live kit," Leckie says. "So, he got quite an experience, file:///H|/SOS%2002-05/CLASSIC%20TRACKS%20%20The%20Stone%20Roses%20%27Fools%20Gold%27.htm (2 of 8)9/22/2005 8:24:54 AM


although I did a lot of the engineering as well." In the meantime, there was also the producer's relationship with the band members, which he now describes as "Very much a collaborative thing. On any given day the guys would come in with songs that Ian and John had demoed on the Fostex, consisting of just guitars and vocals. John always John Leckie today. used a capo on his guitar and he'd spend ages tuning, and there was also a lot of bass tuning as well as time spent on the vocals, although Ian sang a lot better in those days. Regardless of some people's negative comments about his singing, to me he was no different to any other vocalist. He'd perform, say, four takes, and I'd comp them and bounce them down. It certainly wasn't a nightmare. He'd always get what we wanted within a couple of hours, and back then you have to remember there was no Auto-Tune." Ian Brown would actually record guide vocals while guitar, bass and drums were captured live, providing each song's foundation before small sections were dropped in and other parts overdubbed. "They were all competent musicians," says Leckie. "I think the standards these days have gone downhill, but the Stone Roses were a gigging band who had been in the studio before, and the main thing was their optimism and the vibe: 'Yeah, we're gonna make it. Yeah, we're gonna have a great time doing it.' It was all about having fun making music, even if they were also very fussy. We had a good time doing that album, and I think it shows in the music."

A Killer Track Including further recording sessions at Konk and mixing at Battery Studio One and Abbey Road's Studio Two, the album took a total of 55 work days to complete, spanning October 1988 to March 1989. Thereafter, with the Stone Roses playing to sold-out venues across the UK, the band members went back into the studio just under six months later, on August 23, to commence work on 'Fools Gold'. "The record company was looking for another single, but Ian and John were also getting prolific," Leckie explains. "The main thing was, they had a killer track." As it happens, the studio that band and producer entered was quite unique: the Sawmills in Cornwall, one of the UK's first residential facilities and also among the most remote, located within a creek on the banks of the River Fowey. The



only means of access is by boat or, when the tide is out, by walking along a railway line. "If they tell you to be there at four o'clock to get the gear in and you turn up at five and the tide's gone, you have to wait 12 hours before your next chance," Leckie remarks. "Everything fits on the little boat and off you go. Then, once you're there, you can't get out unless the tide's in or you want to take a halfhour walk through the woods. It's a fantastic place." The main building, a 17th-century water mill with low ceilings and plenty of natural daylight, contains a live room, a couple of iso booths and 'the cave', a small bare rock area at the back of the studio, in addition to a control room and machine room. In 1989, as is the case today, the control room housed an 82-channel Trident Series 80B console and 24-track Otari MTR90 MkII recorder.

The Stone Roses: Alan 'Reni' Wren, Gary 'Mani' Mounfield, Ian Brown and John Squire.

"I've worked on a lot of projects there," states Leckie, "including a psychedelic album named Chips From The Chocolate Fireball that I produced in 1987 for the Dukes of Stratosphear. This was XTC under a different persona, and the Roses told me that record was the main reason for them asking to work with me. "The Sawmills' in-house engineer, John Cornfield, largely built the studio, and he was involved in the 'Fools Gold' sessions during the 18 days that we were there. Then we went back to Battery Studio One for four days, where Ian did his vocals and John [Squire] played all of the little wah-wah guitar licks that he'd worked out to go between the main lines. Paul Schroeder and I did the mix at Battery, but it didn't work out — spending three days on overdubs and then the last to do the mix very rarely works. There's a lack of objectivity and distance, because you always tend to focus on the last things you record. Time off helps. And what's more, as Alan Parsons has said, the time it takes to mix is directly proportional to the number of people in the room. So, on October 3 and 4, I went into RAK Studio Three on my own and mixed 'Fools Gold' along with the 'B' side, 'What The World Is Waiting For'. Again, I was working on an SSL, and I took a couple of days to mix both songs." It's interesting to note that, up to this point, the band members actually intended 'What The World Is Waiting For' to be the 'A' side, with 'Fools Gold' in the secondary role. "I thought 'Fools Gold' was fantastic and so to me it was obvious it should be the 'A' side," Leckie says. "But all along the band had different ideas, and in fact early copies of the 12-inch record did have 'What The World Is Waiting For' as the 'A' side. That got changed around pretty quick, yet it meant that throughout production we gave a lot of attention to the 'other' track."



Cupboard Love "John Squire always had his Fostex 16-track recorder in the tape cupboard at the back of the studio, and after we'd gone home he would sit there with headphones and work out all his parts," says John Leckie. "He always did that, right through 'Fool's Gold', up until the second album. In fact, part of the reason for the breakdown of that second album was that John would sit in his bedroom with the Fostex while the rest of us waited in the studio for his guitar parts. He wouldn't improvise or make something up in the studio, but he got to the stage where he was really good. For instance, on 'Bye Bye Badman' there's a guitar that plays all the way through — a kind of counter-lead line, going through a Leslie — and I remember him playing that in about half an hour. However, we had to wait four days for that before he came out of the cupboard." The Fostex was also responsible for the Stone Roses album track 'Don't Stop', which evolved from the 16-track demo of 'Waterfall' being played backwards.

The Discipline Of Tape Piecemeal in nature, the recording approach for 'Fools Gold' was totally different to that employed for the album. John Leckie didn't hear either this track or 'What The World Is Waiting For' until he met up with Ian Brown and John Squire in Cornwall. Then, what he was presented with in terms of the former song was a demo consisting of a four-bar drum loop and tambourine along with a guitar, vocal and some reverb. "The drum loop came from a record," Leckie says, "and the funny thing about it was, when the track ran out on that record, you could hear them lift the needle and put it on again. They'd brought the record with them, so we copied the drum track and made the loop in an [Akai] S1000, sequenced on Cubase. We spent ages tuning that loop, trying to get the right tempo and generally fiddling about, and we quite The control room at Sawmills. enjoyed doing that. After that, the challenge was where to put the band on top, how to represent their individual characters. With any band, I like to give each of the musicians equal importance. "The song didn't really have any structure at this point — it was kind of verse, chorus, jam — and so what we did was lay down about five minutes of the loop onto tape before adding the guitar and bass, trying to determine what to put on without destroying the vibe. Tuning the loop had set up the vibe, and whatever we put on it would kind of change it, so different ideas came up, like a second bass line that floats over the top of the main bass line. Then there was the idea of sticking on the one note, 'E', and then coming back to the loop again. This gave file:///H|/SOS%2002-05/CLASSIC%20TRACKS%20%20The%20Stone%20Roses%20%27Fools%20Gold%27.htm (5 of 8)9/22/2005 8:24:54 AM


us loads of possibilities, which nowadays would be easy using Pro Tools, but back then we had to imagine something, cut the tape, and if we got it wrong, cut it back again." For Leckie, the more laborious method is also the most enjoyable one. "I've done records on Pro Tools, but I just prefer the whole attitude and discipline with regard to tape. Getting your head around things can be a lot of fun, whereas when you're given the freedom of using Pro Tools it can get painful because you spend all your time exploring that freedom without ever really reaching a conclusion."

Keeping The Vibe While Mani's spray-painted Rickenbacker bass was DI'd and fed through an Ampeg SVT amp miked with an AKG D12, he did also try playing an Ashbory bass with its short-scale silicone rubber strings. This, however, never survived the final cut. "Of course, the big problem with us wanting to retain the original vibe of the loop was what the drums were going to do," Leckie comments. "Reni would play along, aware that we didn't want to replace the loop, even though on the final record it's mixed out in several places to reveal Reni's playing and provide more intensity. He actually played great across the loop, using his own custom kit made up of bits and pieces: floor toms painted by John Squire, a Gretsch bass drum and a Ludwig snare, although on 'Fools Gold' we experimented with a smaller Noble & Cooley snare. This was miked with a [Shure] SM57, while there was a D12 on the bass drum along with a Sennheiser 421 to provide a bit more slap and clarity. The D12 was to give it more weight, and we had to position both at the same distance to avoid phase problems. The toms were miked with [Neumann] U87s and 421s, while the hi-hat had an AKG 451, padded down. As the Sawmills live area is quite small with a low ceiling, solid stone walls and lino on the floor, the drums always sound very tight in there. Reni also played bongos, and I think he attempted congas, too, but a lot of 'Fools Gold' has a bongos rhythm." Whereas the drums were positioned in the centre of the live area, Mani's bass amp was closed off in the aforementioned 'cave'. John Squire, meanwhile, played his pink Fender Stratocaster and custom-painted Hofner 335 copy in a far corner, opposite that occupied by the studio's grand piano, with two screens supporting a horizontal one that served as a kind of roof, while a third served as a makeshift door. Inside this setup, Squire used a Fender Twin reverb with JBL speakers, rented from Dreamhire, and played lead parts on the Hofner and rhythm on the Strat, captured with either a Shure SM57 or SM58 together with a Neumann U67. "Things weren't recorded in a big, open space," Leckie confirms. "They were recorded in closed spaces. But then, the main room itself isn't that big anyway, file:///H|/SOS%2002-05/CLASSIC%20TRACKS%20%20The%20Stone%20Roses%20%27Fools%20Gold%27.htm (6 of 8)9/22/2005 8:24:54 AM


and the control room is right next to it, separated by a three-foot-thick stone wall, providing perfect separation." It was in the control room that Ian Brown recorded his vocals, an SM58 having been set up between the main monitors that were only about 15 feet apart. Leckie wasn't concerned about a little spill, and he also had little to worry about in terms of Brown's performance. "It was a pretty simple vocal," he asserts. "The phrasing never changed between the guide track and the final version, and we always had the same reverb on his voice; a Lexicon PCM60, whose reverb is very rich and clear, and which was also used on the main guitar riff."

Leckie at Mushroom Studios, Vancouver, 1990.

The Best Possible Sound According to Leckie, 'Fools Gold' evolved as the sessions progressed, especially the song's second half. It was, after all, released in two versions: one clocking in at nine minutes, 53 seconds; the other, an edited down radio mix, lasting a mere four minutes, 15 seconds. "The playout on the nine-minute, 53second version kept getting longer and longer," he says. "John would have different ideas for the various parts, saying 'Oh, I need to extend this by three bars,' and we were on the leader tape, picking up on code and everything. That meant we had to edit, copy and jam-sync the code, and then fly in everything that had gone before — like the drum loop and bass — sample it onto the Akai S1000, sync it up with Cubase and fly it back in. The object was to eventually get everything onto tape, so when it came to the mix anyone could replicate what we'd been doing.

Situated on a tidal river in remote Cornwall, Sawmills Studio is accessible only by boat for much of the time.

"A lot of effort was put into getting the right sound. They wouldn't play anything unless it sounded good. It wasn't a case of 'OK, turn the mics on and keep going.' We'd spend three days working on the drum sound, and then Reni would be too tired to play so we'd say 'Well, let's do the bass.' Mani would spend two days working on the bass sound, and he'd go back and change it. And then John would take delivery of endless guitars and amps, plug them in and try different file:///H|/SOS%2002-05/CLASSIC%20TRACKS%20%20The%20Stone%20Roses%20%27Fools%20Gold%27.htm (7 of 8)9/22/2005 8:24:54 AM


mics, but it was always the same parts being played and I would share in the learning process. It wasn't a case of wild experimentation, but just getting the best possible sound and something that we all got off on. In fact, whenever there was experimentation, we'd end up going back to the original sound. "Over the 18 days it was a creative process. The second half of the song was all worked out in the studio, and then there were these crazy noises that I highlighted in the mix. We'd messed around with things like sloweddown cymbals, and at the time none of those experiments had worked, but then when I got to the mix I had all these weird tracks to play around with and you get glimpses of them in the second half of the song. Once I got on a roll, there was a temptation to somehow incorporate all of the experimental stuff, but in the end I just retained the most important elements. "Having recorded the album very much as a band, doing the 'Fools Gold' parts one at a time was quite different for us. However, because of the groove of the track and the atmosphere working at the Sawmills, everyone enjoyed it. We all picked up on the spirit of the place. The Stone Roses were, you see, a very vibey band. And it also helped that, on the first day we arrived there, the record company phoned to say that they had an advance order of 30,000, meaning that, no matter what we did, the single was going to chart. It was a good position to be in, and while some bands could have felt under pressure to deliver the goods in that kind of situation, the Stone Roses thrived on it." And they did indeed deliver. Still, the fact that 'Fools Gold' was so different to the preceding album made it stick out like a sore thumb when it was tacked onto the end of the American release. "It was totally out of context," insists John Leckie, who has recently been working with another Manchester outfit, New Order, as well as an American band called Longwave. "In fact, the scary thing was coming up with a single that was nothing like the album, but it enjoyed a lot of success as an 'indie dance crossover', so we obviously got it right!" Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Demo Doctor

In this article:

Centiva How To Submit Your Demo Provocateur QUICKIES

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


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

Centiva Venue: Home Equipment: PC running Steinberg Cubase sequencer and Wavelab editor, Image Line FL Studio soft studio, IK Multimedia T-Racks mastering suite and various plug-ins, Terratec soundcard.

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

The music on this CD could be loosely described as ambient but it also draws inspiration from artists like Moby and Massive Attack. Composer Adrian Kelly achieves a pleasingly full sound with the equipment at his disposal and comes up with some well considered mixes. The first track uses a dry drum sound, putting the drums right up front and giving the production a contemporary dance feel. I liked the way he introduces the drum track with all the bass squeezed out by a high-pass filter and then opens up the filter, creating a dynamic lift. It's a nice change to hear the reverse of the usual trick of applying a low-pass filter opening up from bass to treble. In fact all the mixes feature similarly competent use of both subtle and more dramatic filtering effects and it made me wonder if Adrian might have cut his musical teeth on analogue synthesizers. The beginning of this track is packed with little gems like a fast comb-filtering effect in the background stepping through the harmonics and a tom fill run through what sounds like a dynamic pitch-shift effect. It's always good to hear little touches like this that only happen once in a track and Adrian's file:///H|/SOS%2002-05/Demo%20Doctor.htm (1 of 6)9/22/2005 8:25:03 AM

Demo Doctor

judgement is impressive. His arrangements are also well thought-out, and on this first mix he cannily leaves out the excellent, warm-sounding bass line for a whole minute, confident that it will have a big impact when it arrives. The use of backwards effects for the breakdown about two minutes into the mix is also masterfully executed. As you can gather, keeping these tracks moving with the introduction of different sounds and effects is a strong point of Adrian's approach but this inevitably means that he's got a lot going on in his mixes. His solution to what could easily become a cluttered mix is to use quite wide stereo panning, not only as an aid to separation but also to tease the listener. For example, often a sound in the middle of the frequency range will skip across the stereo field while simultaneously undergoing a filter change. Yet it's on the second track that he's got things just about perfect by splitting the high electric piano notes of the main theme to opposite sides, creating a question-and-answer effect. With the addition of some echo and tasteful use of pitch-bend, this makes for a cracking hook line. In fact the loops, arpeggios and use of rhythmic echo amply demonstrate a strong sense of rhythm which forms a solid base to all the tracks. This, coupled with some excellent mixing and understanding of the equipment, makes for a fine demo.

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

Provocateur Venue: Home Equipment: Apple Mac G4 running Digidesign Pro Tools LE recording software, Digi 001 interface.

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

While searching for Provocateur's web site (the URL given no longer seems to work), I kept being led to the Agent Provocateur lingerie company's site [whatever you say, John — Ed], whose products, male readers may recall, were famously advertised by Kylie Minogue, and there are some similarities to the diminutive one's music in this demo. Provocateur are a five-piece band from the Huddersfield area who manage to conjure up a contemporary sound, whether it's

file:///H|/SOS%2002-05/Demo%20Doctor.htm (2 of 6)9/22/2005 8:25:03 AM

Demo Doctor

the out-and-out dance of the opening track or the more R&B stylings of other cuts on their self-penned CD. Lesa Smith is a capable vocalist who is at her best on their jazz/soul numbers but can also crank up the intensity level when necessary. Even so, a touch of upper-mid EQ boost at around 5kHz would add a bit more energy to the vocal sound on the more upbeat songs. A high-quality outboard EQ like a Neve or even a valve module like the old TL Audio EQ1 would do the job without making the vocal sound harsh and would be well worth renting for mixing if their budget allows. Many of the songs could also be taken to a more complete production level with the addition of block harmony vocals and I'm almost surprised at their omission. Having said that, the individual backing vocals, where they are used, are very effective but are placed a little erratically in the arrangements. For instance, the second song has some very good harmony phrases on the first verse but doesn't give the same embellishment to some of the later verses. Elsewhere, I liked the use of vocal effects like echo, though I was less sure whether the use of a heavy vibrato effect quite came off. On the arrangement side this band create great openings to songs and I just adored the start of the second track, 'Late September'. Here, a heavily effected organ chord and a simple beat lead the listener into the song beautifully. This, followed by some wonderful electric piano chords and laid-back jazzy bass is the sort of sheer class that trendy wine-bar punters and Steely Dan enthusiasts alike would love. Perhaps the bass should be up a little, because the finer points of the playing are getting lost at lower listening levels — but having said that, it was still a pleasure to listen to this track with its mellow vibe and understated muted trumpet breaks. I was also conscious that the piano level was occasionally altered subtly to add the right dynamic touch. In fact, such attention to detail in both the programming and playing is typical of this CD as a whole. I've already praised the way these songs begin but I was surprised that the band have payed so little attention to the endings! These are often abrupt and a bit, well, unmusical to say the least. There's also a tendency to leave very short gaps between the songs on the CD and this, coupled with the odd endings, featuring no fade-out in many cases, doesn't really do justice to the material. In fact, the gap between songs is a very important part of the listening experience which needs to be carefully handled, and some of the songs would actually benefit from fading out on the groove. In last month's column, we gave an incorrect web address for featured demoist Bryce Jonn. The correct URL is www.brycejonn.com.


Demo Doctor

QUICKIES

Johnson Justin Johnson has a good voice, sings in tune and turns in a strong, emotive vocal performance. Unfortunately his voice hasn't been recorded very well. You can't solely blame the gear limitations here, because it's possible to get a pretty decent sound out of a Shure SM58 with a bit of care. Despite the poor recording, the lashings of reverb and delay applied to the vocal do suit Justin's epic style. However, too much reverb has been added to the backing, so the whole mix sounds like it's coming at you from a distance. Tthe instrumentation also sounds very muddy, almost as if it's been pumped into a live room and miked up, or run through an effects patch with the bandwidth set to 2kHz. Such things have been known to happen! The mixes can be considerably improved by a hefty dose of HF boost around 12-14 kHz, which could have been added using Justin's Sound Forge software. However, this doesn't solve the problem of the source sounds. So it's worth taking more care to get a strong, dry signal at the initial recording stage — you won't be able to fix it later on.

Carbone I liked the 'Coalshed Productions' tag on the sleeve and indeed the opening mix is quite dark! Lyrically, it's a kind of stream of consciousness set over an atmospheric backing cleverly put together using Apple Logic and Propellerhead Recycle. Carbone loves to mess with the vocal too, using slight distortion, vocoder and extreme EQ effects to create a threatening mood. Crucially, all this use of technology enhances rather than detracts from the vocal and the audibility of the lyrics. The second mix features a more straightforward approach to the rhythm and arrangement, dropping the slightly Eastern flavour of the previous song for a more underground electro feel. This time the interest comes from his choice of sounds. The vocal is again heavily treated but, wisely, he doesn't tread the same path with the effect as in the first track, this time preferring to use echo and filter EQ. This is a good demo, and Carbone shows that he has a good grasp of the gear at his disposal and a focused approach to the overall sound of the production.

Free Flights Up Drummer Jon James has produced an excellent drum sound for this rock CD. I was particularly impressed with the open, tightly tuned snare that cuts easily through the guitarladen mix. I guess the Noble & Cooley and Ludwig snares were worth the expense! As for the vocals, singer and guitarist Andy Buchanan doesn't seem quite sure whether he wants to be in Queens of the Stone Age or Coldplay, and the group's strong songwriting encompasses this range. Personally, I think he's more comfortable with the pop material, never seeming to quite go the


Demo Doctor

distance with the heavier stuff. I also thought that all the vocals were mixed too loud, especially on the heavier opening song. However the Audio-Technica AT4033 mic, which I've used extensively myself, was a good choice for his voice. The mixes are otherwise pretty sound but lack the low bass energy that you'd expect from a DI'd Line 6 Pod bass preamp. Even so, it's an above-average demo with good songs which, after all, is the most important thing if you want to get signed by a record label. www.freeflightsup.org.uk

Gordon McKracken and David Campbell This is an intriguing approach to the common problem of not having a vocalist — Gordon and David sent a backing track to Nashville by mail where a session singer recorded the vocal. The isolated vocal track was then sent back and added to the original home-produced demo. It seems to me, though, that this service is readily available in the UK — check out the Musicians Union directory or even the classified ads at the back of SOS for session singers offering their services. Anyway, the result is OK but what's missing is the really big, tracked-up backing vocal sound in the choruses, which you'd expect to hear in this kind of pop music. Presumably, they have the backing vocals on separate audio tracks and I suggest they mix these higher in level and even add some more harmonies, particularly one below the main vocal line. The overall sound of the mixes is rather soft, indicating that too many mid-range frequencies have been cut around 500Hz 1kHz, most likely at the post-production stage. If you like '80s Euro pop, the tinkling 16th-note arpeggiator line on the choruses will be welcome, but given the equipment list, some more imaginative sounds could have been used to add an extra edge. Otherwise, it's a solid demo.

Yan Ian Halley claims to be having trouble with distortion on this demo, but I wonder if he's using the term 'distortion' to mean a lack of clarity in the mix. Terminology can be a strange thing. I remember that when I was learning about audio, distortion was the term used to describe literally anything you did to alter the source sound. Now it's more commonly associated with axe-wielding maniacs (except perhaps by the BBC). Suffice it to say that there doesn't seem to be any distortion in the latter sense on this demo, although the synthesizer filters are pretty fizzy. Changing the lower-mid-frequency arpeggios in the backing by making some of them more mellow will compliment the brighter, and very fullsounding melody line. Some experimentation with automated filtering would also give this composition a more contemporary feel, using a slowly opening filter to introduce sounds and bring them to life. Published in SOS February 2005


Demo Doctor




Locking Logic Sequences to a Live take

In this article:


Essential Preparations Logic Notes Display Anomalies Generating The Tempo Map Published in SOS February 2005 Logic Tips Print article : Close window The Rewards Of Patience Technique : Logic Notes Have Your Say!

Current Versions Mac OS X: Apple Logic Pro v7 Mac OS 9: Emagic Logic Pro v6.4.2

Although rather time-consuming, this technique for locking your sequencer to a live take is often the best.

PC: Emagic Logic Audio Platinum v5.5.1

Ingo Vauk

In last month's column I discussed two ways of shifting MIDI sequences into line with live performances: the simpler one using Groove Templates; and the more complex generating a new dynamically changing tempo map. I also showed how to go about using the former approach. This month, I'll be taking a look at how to apply the second of these two techniques. Although generating a tempo map is a more laborious procedure, it is better suited to situations where you want to sync the sequencer to a complete take of a live band. Also, unlike the Groove Templates method used last month, the following strategy lets you lock any number of slave devices, be they hardware or software, to your live take — so your Rewire and VST/AU applications will lock to the drummer just as tightly as any drum machine or slaved sequencer.

Essential Preparations Because the task ahead is very repetitive, I recommend that you set up a number of Key Commands that will make life easier and speed things up considerably. Also, it is important that you set up your work place ergonomically so that you won't inflict a repetitive strain injury on yourself — or if you do, make sure that you can sue the producer!

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The Key Commands you will need are: Set Locators By Objects (don't use the Rounded Locators version), Adjust Tempo Using Object Length And Locators, Split Objects By Locators and Open Sample Editor. In the Sample Editor window have a Key Command for Goto Selection End. I also usually have the Event Float window open (you can get to this from the main Options menu), since it gives all the information I need about a chosen object at a glance. The other thing you should do is make sure that you will not lose the timing information of your original takes. Since Logic references the position of audio regions against tempo information, creating tempo changes will move the relative positions of regions. There are two ways to prevent this from happening: bouncing each audio track so that every track begins on the first beat of the song; or (less time and disk intensive) selecting all tracks and locking their SMPTE positions using the Lock/Unlock SMPTE Position Key Command.

Display Anomalies Logic's graphical interface doesn't follow all the tempo changes in the waveform display of the Arrange window. When you put the original track next to the cut-up version they will not match graphically. This doesn't mean that your cut version is out of sync. You can test this by running both tracks at the same time and flipping the phase of one with the Gainer plug-in. At the same level both signals will completely cancel each other out. Also, when you place a cut somewhere in the original track, the display will adjust to show them synchronised again. Ingo Vauk

Generating The Tempo Map Now the fun begins! If your original audio was recorded to a click track, the grid of the arrangement window should be close enough, but if the live take was done without any clock reference it is helpful to set an approximate starting tempo to work to. Once you've done this, zoom in on the beginning of the audio file in the Arrange window until you can see the first beat of the track (perhaps the beginning of the count-in). Cut the region at the closest quantisation value, delete the region that runs from the cut to the end, and select the remaining region. Set Locators By Objects and then enable Cycle mode. This helps to quickly locate the screen to the point of the track you are working on in case it jumps somewhere else. Now Open Sample Editor and Goto Selection End, adjusting the zoom factor so that you can clearly see the attack portion of the next hit. (This zoom factor will apply to any subsequent Sample Editor windows). Adjust the region length by



dragging the region's End point as close as possible to the attack of the next transient. Close the Sample Editor window (using the default Apple+W Key Command). In the Event Float you will see that the object length has changed to a few ticks longer or shorter than the previous, quantised length. Now Adjust Tempo Using Object Length And Locators. You will be asked by a pop-up dialogue whether you want to set the tempo globally or create a tempo map. In this first instance it makes sense to do it globally, since this will clear any previous tempo information. The tempo indicator will change, and the region will again be of a quantised length (as long as your Cycle).

Where the waveform and the grid markings don't coincide (top), just drag the region end point to the beginning of the audio transient (middle), and then the tempo can easily be adjusted to match them up (bottom screen).

Now extend the region by dragging the bottom right-hand corner to the start of the next transient waveform and let go at the closest quantise value again. It doesn't matter whether this falls slightly before the waveform or cuts into it. Split Objects By Locators and select the righthand object using the right cursor key (the default Key Command for Select Next Object). Set Locators By Objects again, and then open the Sample Editor, where you will see that the new region starts exactly on the beat, but ends slightly off the next waveform transient. Adjust the region length again and close the Sample Editor. When you Adjust Tempo Using Object Length And Locators this time, opt for the Tempo Map option. In the Event Float, the length will be quantised again, and the tempo display will be slightly different, reflecting the fact that you've created the first tempo change. Repeat this procedure until you've created a tempo change for every beat. To summarise: Extend the region to the quantise value closest to the next waveform transient. Split Objects By Locators. Select the right-hand object and Set Locators By Objects. Open the Sample Editor and adjust the region End point to match the waveform transient.



Close the Sample Editor and Adjust Tempo Using Object Length And Locators. It is important to check the timing in the beginning against some MIDI programming, so that you make sure you get the desired effect and choose the right hits to quantise to. Also try to stick to the same zoom factor in the Sample Editor window, and develop a feel for a consistent point at the waveform attack. This will help to get a smooth result, with the beats quantising to the same point of each live hit.

Logic Tips When you double-click a MIDI sequence in the Arrange window a MIDI editor opens. By default that is the Score window, but you can change it to any other MIDI editor using Logic's Global Preferences. Len Sasso Here's a funky effect you can create with the new Direction Mixer plug-in in Logic Pro v7. Create two instances of this plug-in in a stereo track with a Tremolo plugin between them, and change the default Input parameter setting in both of the Direction Mixer plug-ins from LR to MS. This processing splits the signal into M&S format, passes it through the Tremolo plug-in, and then re-combines it into normal stereo; but what it does in practice is modulate the stereo width according to the Tremolo LFO. Mike Senior When you're dragging multiple regions from the Audio window to the Arrange window, open one of the windows as a floating window. That prevents the focus from constantly changing between windows. Which window you choose to float depends on which you prefer to keep available for Key Commands, which will operate on the non-floating window. Len Sasso

The Rewards Of Patience At first this will be a time-consuming procedure, but once you get the flow of the editing sequence it becomes more fluent and almost like typing (the only two actions that you need the mouse for are extending the region to the right and adjusting the length in the Sample Editor). Be prepared to spend about an hour and a half on a four-minute song, depending on how fine you cut the audio. In a lot of cases it will be enough to adjust the tempo at the main beats of each bar and spend some extra time on fills. Don't forget to save regularly, as losing half an hour of this kind of work is a painful experience. Once you have done all this it will be a joy to just run Recycle or Live loops with the material, quantise any takes you are doing on keyboards, or whatever.



Have Your Say! If you want to suggest changes or improvements to Logic, then here's your chance! The Apple development team are inviting SOS readers to send in their suggestions of what they'd most like added or changed in Logic. Email your top five suggestions (in order of preference) to [email protected], and we'll forward your lists on to the Logic team. We'll be asking them for feedback on which changes users deem most important and how these might be addressed. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



PC Notes

In this article:

PC Notes

Gigastudio 3 Update Hints & Tips PC Snippets Published in SOS February 2005 GS3 Plug-in Tips Tiny Tip: USB Devices Print article : Close window Cubase SX3 Dongle Update

Technique : PC Notes

This month PC Notes comes across CD-burning problems, offers some Gigastudio 3 tips, and warns against a possible pitfall when updating a Cubase SX3 dongle. Martin Walker

This month, my CD-R/W drive started to give me problems. It will still read CD-ROMs and CD-Rs, but when I recently tried to burn a CD-R with it, using Ahead's Nero, it refused to read the files I'd just burned, complaining that I'd inserted a blank disc, and my new DVD drive couldn't read its contents either. I experienced no problems during the burn, got the usual 'CD successfully burned' message, and could even see the 'burned' area on the disk.

If you want to run DX plug-ins in Gigastudio 3, try Vincent Burel's freeware FFX4, shown in action here. Note, also, the way in which you can organise Gigastudio's VST plug-in list using folders (see main text).

I burned a couple more discs, with exactly the same problem, and also tried using a CD laser lens cleaner, but eventually had to come to the conclusion that the drive had simply worn out. Laser diodes in CD and DVD players and recorders do age, slowly reducing in power output over time, which is why most of us have experienced hi-fi CD players that eventually refuse to read some CDs, even after careful cleaning.

So I turned to my new DVD burner, bought as the cheaper 'OEM' (Original Equipment Manufacturer) model without any software, cables, or other extras, and tried to burn a CD-R with that instead. Unfortunately, while Nero recognised it as a source device for reading both DVD and CD discs, it didn't appear as a file:///H|/SOS%2002-05/PC%20Notes.htm (1 of 5)9/22/2005 8:25:22 AM

PC Notes

destination device for burning purposes. This can often happen with cut-down, bundled burning software that is only designed to run with the drive it accompanies, but my Nero is the full version. In this situation, I knew the problem was most likely to be that the burning software was released before the CD burner: burning software generally incorporates its own database of CD burners and their capabilities, and if yours isn't recognised it won't appear as a recording device. Sure enough, having uninstalled my previous version of Nero and downloaded and installed the latest free update (from www.ahead.de), I was immediately able to burn a CD-R with my new DVD burner.

Gigastudio 3 Update In last month's SOS, I reviewed Gigastudio 3 and had a few reservations about its operation. However, I'm pleased to report that Tascam have since released a third 'maintenance release' update (version 3.03) that addresses some of them. Firstly, compatibility with Intel's Hyperthreading has been improved, so some users should now finally be able to enable HT in their BIOS (thereby achieving lower CPU overhead when running audio interfaces at low latency), though Tascam say that some systems may still be unstable when HT is enabled. Sadly, my PC is one of them, and it still crashed badly with driver problems each time I enabled HT. I'm also continuing to have problems with Gigastudio 3 periodically refusing to enable my USB keyboard's MIDI In port because it declares it already 'in use' (even though it isn't). Tascam do, however, encourage anyone with problems to send their specs to them so they can make further improvements. Version 3.03 also features a handful of bug fixes, including ones for missed program/patch changes and hanging notes that can occur if you're playing during a program change, as well as possible crashing if hardware settings are changed while the Gigapulse window is open. In addition, NFX and VST bypass states are now remembered correctly after a save, and MIDI Mixer Mute/Solo behaviour has also been Improved, as has the File Save command.

PC Snippets Michael Tippach's famous ASIO4ALL driver (or, more correctly, ASIO-to-WDM Wrapper) is now up to version 2.0, and now has an improved audio engine, looks better and comes with a PDF manual. And it's still freeware! However, the most revolutionary Michael Tippach's latest ASIO4ALL driver now provides ASIO support for multiple addition is its Advanced audio interfaces. Configuration page, which supports multiple interface devices. Yes, you read that correctly — for the first time it's theoretically possible to run multiple interfaces from different manufacturers inside applications such as Cubase SX. file:///H|/SOS%2002-05/PC%20Notes.htm (2 of 5)9/22/2005 8:25:22 AM

PC Notes

You will, of course, need to run them all from the same master clock to keep them in sync, and there are no guarantees, but the PDF manual is very helpful in explaining the various possibilities. www.asio4all.com

GS3 Plug-in Tips Gigastudio 3 has always automatically blocked plug-ins that it deems incompatible, while its comprehensive 'Configure Plug-ins' window lets you manually block others that you don't wish to appear in its list. Because Gigastudio 3 is a 'real-time' application, Tascam can't build in the plug-in latency compensation that most MIDI + Audio sequencers use on playback, so you might prefer to block plug-ins that exhibit high latency (such as some compressors, pitch shifters and convolution reverbs), since these may be almost unplayable otherwise. (If you use Cubase SX you can find out what each plug-in's latency is, in samples, from the 'Plug-In Information' window.) Waves plug-ins are in DX (DirectX) format and require Waves' Waveshell in order to appear in their VST guise. You might be tempted to try to manually block the high-latency Waves plug-ins, such as Ultrapitch, Soundshifter and the Restoration and Masters bundles in Gigastudio 3, but I would advise that you don't. If you do, it's the Waveshell file that's blocked, and all your Waves plug-ins will have disappeared simultaneously next time you launch Gigastudio 3. You can, however, tidy up the plug-in list by creating a new folder named 'Waves' inside the VST Plug-ins folder, and dragging the existing 'Waveshell-VST 5.0.dll' file into it. This will place all the Waves plug-ins into their own folder within Gigastudio, to prevent you being presented with a long list of plug-ins you may not use. Many Gigastudio 3 users have been frustrated at the program's lack of DX plugin support. It's certainly annoying to find only half of your favourite plug-ins appearing in the list of options. Over the last few years there have been quite a few host applications, such as Cakewalk's Sonar, that support DirectX plug-ins but not VST ones, and various developers have stepped into the breech with VST-to-DX 'wrapper' plug-ins that let such hosts load the plug-in variety they don't otherwise support. Products that do this include Spinaudio's VST-DX Wrapper, the Tonewise DirectiXer, and the one that started it all — FXpansion's VST-DX Adapter, which is now bundled with Sonar. DX-to-VST wrappers aren't so easy to find, although I did eventually track one down to try. Vincent Burel's FFX16 is designed as a DirectX 'rack' that lets you chain up to 16 DX plug-ins, each with on/off and bypass buttons, change their order via drag and drop, and load and save your chains. This was a godsend when early DX applications didn't allow plug-in chaining, and FFX16 is still a handy tool today for building complex DX effect chains. For our purposes, the cutdown FFX4 version, with four plug-in 'slots', is perfect as a DX-to-VST adapter, especially as you can download it free from http://vincent.burel.free.fr/download/

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PC Notes

ffx4_FullDemo2.zip. During the install you just need to choose the VST version. Next time you open a VST-compatible host there will be a new VB_ffx4 plug-in, into which you can load any DX plug-in (see screen on previous page). I didn't experience any problems with using FFX4 in Wavelab Montages (another VST-only plug-in zone), but Gigastudio 3 is rather more picky, resulting in some flickering FFX4 graphics and a refusal to let me remove any plug-in once it had been loaded into the rack. However, you can still 'Select Another Plug-In' and load it over the top of the existing one, so this isn't much of a problem in practice. I also experienced some audio glitching after loading a few DX plug-ins, and these could only be cured by increasing Gigastudio's GSIF driver latency. These limitations aside, if you've got a killer DX plug-in that you're desperate to use within Gigastudio, at least now you have a way to try it.

Tiny Tip: USB Devices Many musicians now have quite a few USB devices plugged into their PCs. If you've created several Windows partitions (one for general and Internet use, and another just for music, for instance) you may not want all the drivers for these peripherals to be installed on your music partition. Since each USB Universal Host Controller is in charge of a pair of ports, the answer is to keep non-music devices, such as modems and printers, plugged into one pair, and your USB music keyboard and dongle into another pair. Then you can disable one of your Host Controller chips in Device Manager by right-clicking on it and selecting the 'Disable' option. Your music partition will never even realise that a printer and modem are plugged in. The trick is in working out which Host Controller is in charge of which pair of ports, as this isn't always obvious (mine are simply labelled 24D2, 24D4, and 24D7). The way to do this is in Device Manager: switch from the default view, 'Devices By Type', to 'Devices By Connection', so you can see what's plugged into each port.

Cubase SX3 Dongle Update Although I know that some musicians dislike copy-protection dongles, I've always preferred them to hard-disk authorisations, since a dongle and its contents always survive system crashes and hardware and software upgrades, and can even be moved from computer to computer. However, this month I ran into some problems with a dongle. The Cubase SX3 update comes with an authorisation code to upgrade your existing Syncrosoft dongle from version two to version three status, and requires Internet access to do so. Like many musicians, I don't keep Internet software on my music partition, and I don't intend to, but fortunately Steinberg do provide details of how to update the dongle on another computer that does have Internet access, and they say you can do this from a laptop, or even via an Internet café machine, as long as the PC in question has a USB port. Unfortunately, this is


PC Notes

where I ran into problems, and as some of you may encounter the same ones, I thought I'd tell you what happened. Steinberg always warn you to remove their USB dongle before installing a Cubase update, to prevent Windows attempting to install USB drivers for it. The Cubase SX3 update installs the latest drivers for the dongle, and once this has been completed you can plug it back in again, safe in the knowledge that its drivers are ready and waiting. To update the dongle on another computer you need to first install the 5.5MB 'Copy Protection Driver' software from the Update DVD-ROM, and then you plug in the dongle. I did this on my laptop and, as expected, Windows then found the dongle and offered to search for a driver for it. As usual, I declined. Unfortunately, as I found after ringing the Steinberg helpline in despair after Syncrosoft's License Control Centre refused to detect my dongle, this is the one time when you do need to ignore Steinberg's normal advice and let Windows install a driver, since Syncrosoft's software doesn't do it automatically. Doh! After I did this, I was able to successfully update my dongle. While you're online, it's a good idea to download the new 5MB Cubase SX HTML help file. For some reason that I can't fathom, it isn't included on the DVD-ROM. You can find it at ftp://ftp.pinnaclesys.com/Steinberg/download/. Personally, I don't want to have to install personal Firewall, anti-virus and spyware protection on a music PC, even if I don't run such utilities when making music, but it's getting increasingly difficult to keep the Internet away from a musiconly PC. At some stage I suspect we'll all have to give in, especially since I can't see many Internet cafés letting you install dongle drivers and software on their machines! Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Roland VS2480 Masterclass

In this article:

Gain Extra Aux Outputs! Using The Assignable Controls Using The Oscillator For Drum Mixing Waiting For VS8F3... VS8F2 Effects Tricks Audio Editing Using Stereo Delay Quick Tips Effect Chains & Effectsreturn Aux Sends The Quickest Way To Deess Using Automix Handy Effects Setups Virtual Veteran VS2480 Web Resources

Roland VS2480 Masterclass More Hands-on Power User Tips Published in SOS February 2005 Print article : Close window


More hands-on tips for harnessing the power of Roland's monster multitracker. Mike Senior

Last month's VS2480 masterclass concentrated on mixing and automation advice, as well as discussing some issues of audio quality and the usefulness of the VGA output. Now I'm going to spend a little time exploring some of the useful things you can do with the mixer's routing matrix, before passing on a few of my favourite VS8F2 effects tricks.

Gain Extra Aux Outputs! Because of its eight auxiliary busses and eight direct outputs, there are many more potential sends from the VS2480's internal digital mixer than there are builtin analogue outputs. This may not be a concern for you if you have invested in additional R-Bus interfacing, but there is a workaround to get a couple of extra send outputs at mixdown if you're just running the multitracker on its own. These two extra send outputs are the Monitor outputs. In order to free these up while mixing, you have to do your monitoring via the main Master outputs, if you're using speakers, just re-plug them to the Master outputs if they're not already there. If you're wanting to mix on headphones, then you'll need to press the EZ Routing button and then select the Output tab to get to the Output Assign screen. Once there, you can select the headphone output you're after (with the cursor keys or mouse) and select the MSTL/R buss as its source.

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Linking two aux busses into a single stereo buss lets you send separate signals to each of the two Monitor jack outputs (right). If the aux busses aren't linked, then a single aux buss will be fed to both outputs (left).

You can now access the two Monitor outputs using a stereo aux buss. If you don't have one yet, then link a pair of adjacent mono aux busses from any Ch Edit View screen by selecting one of the aux pre/post switches and then pressing Enter — the pop-up window which follows offers a linking switch. Now if you head over to the EZ Routing Output Assign screen as before, and then cursor all the way to the left, you can connect the aux-buss stereo pair to the Monitor outputs. Some of you may want to get two separate mono sends to come out of the two Monitor output jacks, but unfortunately this is not possible in any straightforward way. An unpaired aux buss will hog both Monitor outputs if you try to select it in the EZ Routing Output Assign page, which is fine if you only need one extra mono send, but a bummer if you want to hear another aux buss on one of the outputs. You can work around this a little by linking the two separate sends together and, although not ideal, it does in practice provide two fairly independent sends if you're willing to spend time juggling aux level and pan controls.

Using The Assignable Controls One of the most useful control facilities offered by the VS2480 is the User Knob/ Fader Assign mode, enabled by pressing the front-panel Knob/Fader Assign button and then the '9' numeric key. This lets you control pretty much any mixing parameter (hold Shift and press the '9' numeric key while in User Knob/Fader Assign mode for the list) from either the channel faders or the row of rotary encoders, as specified using the Knob/Fader Assign Sw parameter in the Global pages of the Utility menu. If you want to be able to see the setting of the adjusted parameter instantly, then fader control is the best option. However, the rotary encoders respond more smoothly to rapid control movements (the faders end up stepping), and also offer greater accuracy. In cases where one 'click' of the rotary encoder doesn't give the finest increment of a given parameter (in other words, for any parameter calibrated in decibels), holding down Shift allows finer adjustment. I find that I usually have the User Knob/Fader Assign mode set up to control the EQ bypass switch from the fader, so that I can adjust the EQ from the rotary



controls as normal while working the bypass switch from the fader, all without touching the cursor buttons or data dial.

If you assign the EQ bypass function to the channel faders, you can keep flipping the EQ in and out of the signal path while you're using the PRM Edit knobs to set it up.

Using The Oscillator For Drum Mixing Although the main purpose of the oscillator in the VS2480 is technical, there is also the odd creative application for it when you're mixing drum sounds. One of the things you can do is add extra low-end welly to your kick-drum tracks. Select the Utility menu's Osc/ Anlyz option to get to the oscillator settings screen, and then set it up to output a sine wave at 40Hz. Make sure it's switched on, but turn off its Mix Sw and assign it instead to a spare Dir buss. Now make sure that the Record Monitor parameter in the Utility menu's The D-Time parameter in the Mastering PlayRec pages is set to Source and arm Toolkit effect algorithm's Input block is set a spare track for recording. From the EZ to 10ms by default, which means the effect Routing View screen, patch the Dir buss adds a delay of 10ms to the signal. When you're using it as a channel insert effect, it's carrying the oscillator sine wave to the therefore usually better to reduce this delay record-enabled track, and then go to to zero. that track's Ch Edit View screen. Switch on the dynamics processor, selecting the Expander option and making sure that the KeyIn parameter is set to feed the side-chain from the kick-drum track. Finally, press Play and adjust the dynamics parameters: go for the more extreme Ratio settings in order to pretty much silence the sine wave between beats, set the Threshold for reliable triggering, and use the Attack and Release controls to tweak the sine wave's level envelope. Note that there isn't any point trying to EQ the oscillator channel — a sine wave only has energy at one frequency, so EQ file:///H|/SOS%2002-05/Roland%20VS2480%20Masterclass.htm (3 of 12)9/22/2005 8:25:30 AM


will only change its level! If you find you need more level, head back to the oscillator settings and raise its Att control, and while you're there you might want to experiment with different oscillator frequencies as well. If you need to reclaim the record-enabled track, bounce the two tracks down to another one and use the result in place of your original kick-drum track — remember that you can always keep the original recording tucked away in your virtual tracks in case you want to redo things. Another variation on this theme is to use the oscillator's White Noise or Pink Noise settings to support snare drums in the same way. Here, though, you can afford to use drastic EQ to isolate just those frequency ranges of the noise signal that you want to add to your snare — perhaps it just needs more weight, or a little more buzz.

Waiting For VS8F3... Roland's new VS8F3 effects card , which offers the capability of running thirdparty plug-ins, has just arrived here at the SOS office, and it's presence already has an impact on what I've been writing in this masterclass. The first thing to mention is that the new card works at 56-bit resolution, which means that you're not losing resolution from the VS2480's internal 56-bit mixer when inserting effects processing into any channel, or indeed the mix buss. The second thing to say is that it looks as if some of the third-party effect algorithms may not be usable without the external VGA display. Although the five Roland plug-ins bundled as standard with the VS8F3 card all work on both LCD and VGA, when you load up any of these plug-ins, there is a line in the Plug-in Information page that reads 'Available On VS2480: LCD & VGA'. Call me suspicious, but this kind of hints to me that 'Available On VS2480: VGA Only' is a possible option here, so my advice last month to spend the money you would have used for a VGA display to buy extra effects cards could end up being a bit self-defeating... Rest assured that any of the third-party plug-ins I get hold of will be reviewed with this in mind! One final thing. Although the heritage of some of the VS8F2 algorithms lives on in the VS8F3, the more recent card doesn't include most of the earlier card's algorithms. So, in practice, you'll probably want to think of the VS8F3 as a complement, rather than a replacement, for the VS8F2.

VS8F2 Effects Tricks Even without the VS8F3, the effects processing is one of the VS2480's real highlights, in my opinion. If you install the full complement of VS8F2 effects boards, you get up to eight stereo processors, which means that you can really file:///H|/SOS%2002-05/Roland%20VS2480%20Masterclass.htm (4 of 12)9/22/2005 8:25:30 AM


go to town with spot effects in your mix. However, there are a couple of bits of general advice which I'd like to offer for getting the best out of them. Firstly, I'd recommend complementing the internal effects with a dedicated hardware reverb unit — you'll probably want something at least as good as the TC Electronic M*One XL or Lexicon MPX550. Not only does this throw more processing power at an important effect which really benefits from it, but it also frees up your internal effects to do all the mad things they do best! The second thing I'd suggest is to avoid inserting effects on your mix buss if you're working at 24-bit resolution. The reason for this is that the VS8F2 only processes at 24-bit resolution (compared to the much more suitable 56-bit resolution of the VS2480's own internal mixer), and it can therefore potentially degrade the sound of your whole mix. If you are a fan of processing the main stereo buss at the mixdown stage, then I'd suggest mixing down to an external digital recorder via a rackmount digital processor — one of the all-in-one mastering boxes would be a good bet for this, but be careful with any multi-band dynamics.

You can mix the oscillator in with drum sounds only when they're playing. First you need to set up the oscillator to feed one of the Dir busses (Screen 1) and route this buss to a spare record-enabled track (Screen 2). Then go to the Utility menu and switch the Record Monitor mode to Source (Screen 3) so that you can hear the oscillator through the spare track. Finally, enable the track's channel expander and key it from your choice of drum track, adjusting the controls for reliable triggering (Screen 4).

Just because you might not want to use the Mastering Toolkit over your whole mix, it doesn't mean that it's redundant, because it can prove very useful on individual channels — just set the D-Time parameter in the algorithm's Input block to zero in this case, so that your track isn't delayed any more than necessary. For example, I recently used it to tighten up the low end of a flabby kick drum: with the Low Split Point parameter in the Input block at about 80Hz, I set up the multi-band expander to operate only in that band, and then used a low ratio (1:1.6) and high threshold to tighten up the low-end envelope of each kick drum, redressing the tonal balance using the Low Level parameter in the Mixer block. The VS8F2 algorithms were developed for the VS1680, so some of the algorithms seem a little irrelevant given the enhanced channel EQ and dynamics facilities in the VS2480. However, the effects-board EQ patches can still be handy in chains of effects, and they can be combined with the dynamics in the



DualComp/Lim and MicModeling algorithms to allow equalisation to follow dynamics — something that isn't possible just using the mixer channel. Furthermore, it's possible to automate dynamics parameter changes by changing effect algorithms, and you can't do this with the channel dynamics processor.

Audio Editing The audio editing on the VS2480 is another of its strengths, despite the inability to edit down to sample level, a restriction shared by the whole VS clan. However, there are some things that it's good to know if you end up editing a lot. First of all, you should prepare the ground a little by setting things up so that you can work as fast as possible. For a start, this means switching off the confirmation messages which accompany every editing procedure Although the Edit Message confirmation and Locator-point change — they (above) can be useful occasionally for just add unnecessary key presses. checking editing settings, you'll find that it Head for the Utility menu's Global gets in the way if you do a lot of editing using the front-panel buttons. pages, and make sure Locator/ Scene Type is set to Quick and Edit Message is set to Off. Also change Edit Point Sw Type to Overwrite, so that you can set the In, Out, From, and To points with single key presses. (Incidentally, if you actually want to locate to any of these edit points, rather than overwriting them, hold down Shift.) Setting these three parameters correctly dramatically increases the speed at which you can carry out editing operations. There may be the odd occasion where you wish to see the Edit Message to confirm the exact timing of your editing points, in which case it's easiest to toggle the Edit Message parameter directly by holding Shift and pressing the Phrase/Region/Automix button. Unlike with the previous VS machines, you have a mouse at your disposal on the VS2480. This is especially useful for moving audio between tracks, and it's also great for quickly adjusting Phrase boundaries from the Waveform Display screen. However, I still find that I don't use the mouse for most of my edits, because using the buttons and data wheel is so much more reliable and repeatable. If, like me, you don't use the mouse for many of the editing operations, it's worth making full use of the dedicated Track Edit buttons under the LCD display. Not only do they bypass the Track menu, but they also provide a quick method of confirming your edits — for Phrase and Region Copy/Move commands (as well as for Phrase Trim/ Split) you can click the same Track Edit button again to confirm, rather than having to move your hand down to the Enter/Yes key every time. One thing to remember about the Track Edit buttons, though, is that you can only use them when they're lit up, and they only do that when a Region or Phrase is selected and the required editing points are set.

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There's a lot to be said for using delays at the mixing stage nowadays rather than reverbs, because delay can give you a sense of fullness and space without clouding the mix. One of the great things about the effects implementation in the VS2480 is that you can set up a true stereo delay — in other words a stereo aux buss feeding a stereo-in/stereo-out delay effect. The advantages of stereo delay are that any mono track can have its delay sound panned independently, and the delay sound for stereo tracks will share the same stereo spread as the dry signal.

There are only two true stereo-in/stereo-out delays in the VS8F2 algorithm set: Stereo Pitch Shift Delay (top) and Stereo Delay Chorus (bottom).

Here's how you set up this effect. First of all you need to link an odd/even pair of aux sends, as described above. Head to a spare effects-return channel and set its Assign parameter to the newly linked pair of aux sends before selecting Stereo Pitch Shifter Delay (one of only two true-stereo delays in the VS8F2 set) as its effects algorithm. Go into the editing pages for this algorithm and zero the Pitch, CrossFeedbackLvl, and DirLvl parameters, before setting up the remaining values to taste — just make sure that you set associated Lch and Rch parameters the same. When I'm using this patch as an alternative to reverb, I tend to zero the PreDly, set the FBDly according to the song's tempo, and then roll off high and low end using the effect algorithm's EQ block.

Quick Tips On the VS880, the Previous/Next buttons could be used to jump between Marker points, but on the VS2480 the default is that they jump between Phrase boundaries. If you want to change this back to the VS880 method, then go to the Global Utility pages and set the Previous/Next Sw parameter to Marker. If you link two mono mixer channels into a single stereo channel, their individual pan controls disappear and are replaced by a single communal control. Similarly, their individual fader levels are replaced by a communal stereo fader level. However, if you want to adjust the individual pan controls (say, to narrow the stereo image) or fader levels (to adjust a recorded level imbalance) you have to go to the Ch Edit View screen, select the pan or fader icon, and then press Enter. A small pop-up window



will then offer up the separate controls for tweaking. As a related point, if you're automating a pair of mixer channels which have been linked for stereo, you need to put the automation data on the odd-numbered channel, otherwise it won't have any effect.

Effect Chains & Effects-return Aux Sends The only drawback of this technique for creating a stereo delay is that the Stereo Pitch Shifter Delay algorithm only offers a maximum 500ms delay time. However, you can work around this problem by chaining two effects returns together, and this is quite straightforward to do using the Dir busses. Go to the first effectsreturn channel and set it up with a stereo delay as already described above, but leave the algorithm's EQ switched off and all FeedbackLvl parameters at zero. Switch off this channel's connection to the mix, assigning it instead to a pair of Dir busses. Now go to a second effects-return channel, select the appropriate Dir busses as its input, and duplicate the same effects patch as on the first channel. You'll now get a single delay through the two effects of up to 1000ms, to which you can add feedback by simply fading up whichever aux send on the second effects-return channel feeds the first effects-return channel. This technique for chaining effects can of course be put to use with any effects algorithms in much the same way, which can be handy for things like de-essing reverb sends and ducking delays.

A couple of useful effects setups. The upper screen here shows how you use the Stereo Pitch Shift Delay algorithm to imitate the traditional Harmonizer vocal chorus effect. The lower screen shows the speaker simulator block of the Guitar Amp Simulator algorithm — if used on its own with a Mic Setting value of three, it creates a nice stereo ambience.

The effects-return aux sends are handy in other situations as well. An obvious use for them is to add a little reverb to delays, helping to recess them behind the dry sound. Also, if you're unable to run a dedicated external reverb, you can increase the smoothness of the internal reverb sounds in the same way people often used to in the '80s: by layering different delay and reverb patches, and by feeding small amounts of signal between the different effects-return channels as well.



The Quickest Way To De-ess Using Automix Although de-essing a complete lead vocal using an automated channel EQ takes a fair bit of time to do, I consider it worth the effort nonetheless. I hope that the deessers which will become available on the VS8F3 card will render such an activity unnecessary, but at the moment none of the de-essers in the VS2480 do the job well enough, to my ears. If you have come to the same conclusion, then here's a way to make generating automated de-essing as painless as possible. First you have to make sure that you've got the lead-vocal channel selected, and that the assignable rotary controls are in the PRM Edit mode, so that you can access the EQ controls. In addition, set the frequency of the Hi-Mid EQ band to a useful value — maybe 6kHz to start with. Now set the lead-vocal channel to Automix write mode (hold the Automix button and press the Ch Edit button until the button goes red) and enter Automix recording mode (hold Automix and press Record). Return to the Home display and the press Scrub and then Wave Disp. You can now scroll through the vocal track using the data dial and shuttle wheel until you find the first case of excessive sibilance. Locate the transport to the beginning of the 's', turn the Hi-Mid EQ control a few clicks anticlockwise (each click will be a decibel of EQ cut), and then hold the Automix button and press Tap to save a snapshot. Move to the end of the 's', turn the Hi-Mid control the same number of clicks clockwise, and store another snapshot. You've now ducked the EQ band just for the patch of sibilance. You can proceed through the whole track in this way, and because you never have to stop the Automix recording mode, it's comparatively quick. If each time that you adjust the rotary control the display shifts to the Ch Edit View screen, you can get back to the waveform display by pressing the Ch Edit button — oddly, pressing the Wave Disp button won't work here. However, I prefer to switch off the automatic display switching before I start. The parameter you need is called PRM Knob Auto Disp, and you can find it in the second of the two Global Utility pages. Another thing to say is that you might decide after you've finished the de-essing automation that you want to try a different EQ frequency or bandwidth. My advice in this case is first to wipe all the frequency and bandwidth automation information for that EQ band for the entire track. You can do this by going into the Automix Edit screen, selecting EQ Hi-Mid Freq or EQ Hi-Mid Q as the Target information, and then using the Erase editing function. Once this is done, you can adjust the EQ frequency and bandwidth by ear while the track is playing, without automation data fighting you for the control of those parameters.

Handy Effects Setups The Stereo Pitch Shifter Delay algorithm is also the place to look if you're after the classic stereo thickening effect which many engineers used to achieve using the old AMS sampling delay or the Eventide Harmonizer. You only need a mono send for this patch, although the effects-return channel still wants to have a stereo output, so leave the Mono Sw off. Zero all the effect algorithm's delay and feedback parameters, and check that the DirLvl is also at zero. Enter PreDly file:///H|/SOS%2002-05/Roland%20VS2480%20Masterclass.htm (9 of 12)9/22/2005 8:25:30 AM


values of 11ms and 13ms for the left and right channels respectively, and then use the pitch-shifting to shift the left and right channels a few cents flat and sharp respectively. How far you shift depends on the exact effect you like, but I usually stay within the ±5 cents range. The EQ can again be used for tonal fine-tuning. This patch is useful to subtly widen the stereo image of almost anything, and it's especially effective on vocals — just be careful not to overdo the levels unless you're covering 'Hard Habit To Break'... If you've got any single mono track which sounds a bit thin and uninspiring in your mix, a good way to beef it up is to put it through the GuitarAmpMdl algorithm — even a very clean patch can add a nice degree of extra body and resonance. Although you can insert this effect into the channel directly, this doesn't actually make the De-essing with a channel equaliser under best of the processing, because the Automix control: Once you've set up the Hioutput of the effect is actually stereo. Mid EQ band's frequency to something Instead, switch off the channel's sensible for de-essing (inset), you can switch assignment to the mix and send it to a on the Waveform Display and the Scrub function during Automix recording, isolating spare Dir buss, which can then be and ducking rogue patches of sibilance selected as the input for an effectscomparatively quickly using the data wheel return channel containing the ampand the PRM Edit Hi-Mid Gain encoder. modelling processing. Because the output of the return channel is stereo by default, you'll now also get the rather nice stereo ambience that is added as part of the speaker-modelling block — for the most pronounced result, choose the highest (most distant) value for the Mic Setting parameter.

Virtual Veteran In this short series I've demonstrated how careful consideration of your working methods is the key to getting the best from Roland's flagship recording workstation in your studio. I've also shown that many of the VS2480's functions (and quirks!) can be used creatively, and there's little doubt that such possibilities will multiply as Roland continue to expand the capabilities of this powerful machine.



VS2480 Web Resources www.discretedrums.com

This sample-library manufacturer offers multi-track drum libraries directly in VS2480 format. The libraries provide ready-to-go song sections which can be mixed and matched to create a suitable performance for your particular track. Because these are multi-track recordings, rather than just stereo loops, you can, of course, remix the drum balance as appropriate to your style of music, and they would also make great practice material for the novice at mixing drums. The library includes an audio CD in addition to the data CD-ROM, so that you can preview and select the sections you want to use before importing them into the VS2480. www.roland.com/support/en/

This is the location of Roland's new on-line manual library. There have been a number of manuals for the VS2480, and some are definitely better than others. When we first received the VS2480 for review here at SOS, the manuals seemed barely translated out of Japanese, but the manuals which were released with the VS2480 by Roland US were much better, and it's these which you can download. So if your manual contains gems such as "the parameter that set up ti with the AUX mixer because the surround function has materialized by using the bus changes writing for the use of surround," do yourself a favour and download the other version! If you're after a DVD manual, there's currently only a VS2400 title available through Roland in the UK — apparently there's been too little demand for them to ship them over from Roland US, who make them. However, some retailers in the US appear to stock the VS2480 DVD manual, so if you're desperate to get hold of it then you could probably get a copy shipped from there, although you may fall foul of customs charges. www.vsplanet.com

The VS Planet site seems to be going from strength to strength, even though it inevitably suffers from the problem of all large forums, namely that the useful information is spread very thinly over a lot of chat. However, given how complex the VS2480 is, this is probably still the best place to look if you're after the answer to a specific VS-related query.



www.rolandvs2480.com

This support site is officially linked with VS Planet, but makes it slightly easier to get to the useful stuff. There's also a great downloads section which includes track sheets, software update downloads, and a variety of miscellaneous utilities. www.soundonsound.com/sos/Mar02/articles/rolandvstips.asp

Here you can read the previous VS880/VS1680 tips article, from SOS March 2002. Although much has changed in the VS2480, there are still sections of this article which apply to it. www.soundonsound.com/sos/apr03/articles/qa0403.asp

A few early Roland VS2480 users encountered problems with their machine's preamps, and wrote in to SOS as a result. The following Q&A response clears things up. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Updating PC Hard Drives: The SOS Guide

In this article:

Drive Temperatures Before The Event Trial & Error The Silent Option Things To Save From Previous Installations Starting From Scratch Transferring Data Re-authorising Protected Software Attack Of The Clones The Clean-up

Updating PC Hard Drives: The SOS Guide PC Musician Published in SOS February 2005 Print article : Close window

Technique : PC Musician

You're planning to change your PC hard drives, so you just unplug the old ones, plug in the new ones, and off you go, right? Wrong: there's many a slip, and this month we aim to help you avoid all of them. Martin Walker

I suspect that most PC musicians will, at some time, install a new hard drive in their computer, either to expand their existing audio storage area or to replace an existing Windows drive with one that offers faster performance and more space. Physical installation of a drive is normally straightforward, simply requiring attachment using four bolts, and then connection of power and data cables between it and the motherboard. However, modern motherboards may offer various connection options for drives, and if yours offers PATA (Parallel Advanced Technology Attachment) and SATA (Serial ATA) connectors, you first have to choose the most suitable drive. Complications can then arise if you're replacing your Windows drive, as you may also have to change BIOS settings, especially if you're moving from PATA to SATA, and the various options can vary considerably depending on whether you're still running Windows 98/ME as opposed to Windows 2000/XP, or if you're attempting to run both in a multi-boot setup. The biggest challenge is if you want to replace an existing drive containing Windows and all your applications with a new model, and you don't want to have to spend countless hours reinstalling everything from scratch on the new and initially empty drive. This is exactly the scenario I faced this month — replacing two PATA drives, containing four Windows partitions and a further six data partitions, with two new SATA drives. Before retiring them I wanted all the PATA data transferred to the SATA models. I eventually got everything working perfectly, but not before experiencing disappearing drives, linked clones and nonfunctioning boot utilities (you'll find out what I mean in due course). Judging by the SOS Forums, many of you are attempting to do the same thing, so let's see what the best choices are, and find out how to avoid problems.

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Drive Temperatures Did you know that many modern hard drives let you monitor their temperature? No, neither did I until recently, when Glenn Garrett of Quiet PC provided me with a link to a very useful utility that lets you display the current temperatures on your Taskbar. DTemp is a tiny 94kb free download (http://private.peterlink.ru/tochinov) that, when run under Windows 9x, 2000, NT or XP, interrogates the Smart Monitoring features of your hard drive(s), and can show info such as capacity, buffer size, supported and current transfer modes, various of the more esoteric features that it supports, and details of current parameters including error rates and temperature. You don't have to enable each drive's Smart Monitoring mode in the BIOS to do this, either (most builders disable Smart Monitoring mode, as it imposes a tiny extra overhead during data transfers). I was particularly interested in temperature monitoring because my two new Seagate Barracuda SATA hard drives run hotter than the previous PATA versions, and I was unsure whether or not to fit them into my existing Silent Drive sleeves for fear of overheating. Many musicians have abandoned their existing sleeves because of this worry, but using DTemp I could adopt a more scientific approach and monitor the stabilised temperatures of my drives before making a final decision. According to the spec sheet for my ST380013AS drives, they can run with an operating temperature of up to 60 degrees Centigrade (you can generally find safe operating temperatures on drive manufacturers' web sites). In free space (outside my PC's case) they both settled after some hours at a steady 42 degrees Centigrade, and once mounted inside the sleeves this temperature didn't rise beyond 43 degrees. At first I was surprised by this, as I expected their temperature to rise considerably once confined, but this has not happened. However, I've mounted the sleeves at the bottom front of my PC's case, where they sit in the incoming cool airflow from the two front intake fans, which probably explains things. I'm certainly not complaining, as I don't now have to invest in new drive silencers.

Before The Event I hadn't planned to buy new hard drives, but had to respond quickly when I experienced problems with my existing Windows drive. Twice this month I booted up my PC and encountered 'bad sector' faults. These are flagged when Windows has difficulty reading files on a drive, and it suggests that you run the Check Disk utility to scan the disk's entire working surface in search of further problem areas. If any bad sectors are found the data stored in them can then be rescued, as far as possible, and then the sectors are permanently marked as bad so that Windows can work around them. If you suffer a single bad-sector problem out of the blue, it may be because you accidentally knocked your PC with your foot or the vacuum cleaner while it was switched on and jarred the drive's read/write heads. However, two instances within a short period suggests that the drive may be about to fail, and you should



immediately back up all its data and strongly consider buying a replacement. Even if Windows doesn't routinely report any bad sectors, it's worth periodically running Check Disk to check your drives. You can do this by right-clicking on them in Explorer, choosing the Properties option, and then clicking Errorchecking on the Tools page. Make sure the box labelled 'Scan for and attempt recovery of bad sectors' is ticked. Since hard drives are comparatively cheap, but the data stored on them is generally far more valuable, I decided to buy a replacement drive. I took advantage of the situation to switch from two Parallel-connected Seagate Barracuda ATA drives to a pair of the faster Serial ATA (SATA) models. These are by far the most popular drives with specialist music retailers. While the PATA models provide about 40Mb/second sustained transfer rates for both read and write, the newer SATA models can manage 56Mb/ second, which means that Windows and your applications will load more quickly, and — probably more important for the PC musician — you'll be able to stream more audio tracks and samples if you need them.

Periodically checking your hard drive partitions for bad sector errors, as shown here, can tip you off about problems that point to the drive being possibly about to fail.

Like many others, my Asus P4P800 Deluxe motherboard provides connections for up to four PATA drives, using the familiar ribbon cables, and two SATA drives, and it came bundled with a pair of the thinner SATA cables, so all I needed was the drives themselves. I took advantage of this to order a pair of the slightly cheaper OEM (Original Equipment Manufacturer) drives with no cables, manuals or bundled software extras. In the old days of PATA-only motherboards, it was comparatively simple to decide where to connect your hard drives, since there were generally only two IDE sockets, each supporting up to two drives as Master or Slave. Your boot drive was always connected as Primary Master, your audio drive normally as Secondary Master, and any CD-ROM or DVD drives in the Primary or Secondary Slave positions. Although there were some variations on this theme, there weren't that many options to try out. But many modern motherboards with SATA support are capable of running six or more hard drives, and sometimes even RAID (Redundant Array of Inexpensive Drives) arrays (the ICH5R South Bridge chip on mine supports up to four Ultra DMA 100 IDE drives, plus two SATA drives with a RAID 0 function), and there are, therefore, many more permutations on offer. I still maintain that for musicians who don't want to run more than 48 simultaneous tracks at 24-



bit/96kHz or video tracks alongside audio, RAID is unnecessary, so if you don't fit into those categories you can generally leave this option disabled in the BIOS. The most important settings in most BIOS menus when you're installing a new drive are found on the IDE Configuration page. Over the last year or so I've noticed various SOS Forum posters asking for help in deciding on the most suitable settings for their systems, but this entirely depends on what OS you're running and what combination of drives you have installed. However, all should become clear if I describe the options I had on my PC. As is the case with most motherboards, there was an 'Onboard IDE Operate Mode' with Compatible or Enhanced Mode options. Depending on this setting, a second parameter appeared. In Compatible Mode it was labelled 'IDE Port Setting', while in Enhanced Mode it was labelled 'Enhanced Mode Support On'. Both settings led to three further options — no wonder people get confused! If, like me, you upgraded to a SATA-capable motherboard but initially carried on using some existing PATA drives because their performance was perfectly adequate, you can simply accept the default 'Enhanced Mode' setting. Only if a SATA drive is plugged in will the various other choices come into play.

Trial & Error I found that the most helpful way to proceed was to connect both of my new SATA drives, providing me in total with four hard drives and a CD-ROM drive, and then use a trial and error approach by selecting each of the six IDE options in turn, saving the BIOS configuration changes each time and exiting the BIOS, then immediately pressing the Delete key to re-enter the BIOS with the new settings in place (your PC may use another key, such as F1 or F10, to enter the BIOS). With each IDE setting a new combination or arrangement of my five drives appeared. The general advice is to choose Compatible Mode if you're still using a 'legacy' operating system such as MS-DOS, Windows 98, ME, or NT 4.0. This will restrict you to a maximum of four simultaneous drives in various combinations, but does still allow you to benefit from the speed advantages of the latest SATA drives. With 'PATA Ports Only' the two new SATA drives were totally ignored in favour of the PATA Primary and Secondary Master and Slave drives. I can't see a lot of use for this option unless you temporarily want to access four PATA drives and ignore your new SATA ones. After all, this is exactly the same configuration as leaving the default Enhanced Mode on with no SATA drives plugged in. The other two options are more useful: 'Primary PATA + SATA' left me my Primary Master and Slave plus the two new SATA drives, and 'Secondary PATA + SATA' moved the two new SATA drives to Primary Master and Slave positions, leaving my original Secondary Master PATA drive (and any Secondary Slave drive such as a CD-ROM or DVD drive) in their original positions. In practical terms there's little difference between these two. You can access two PATA file:///H|/SOS%2002-05/Updating%20PC%20Hard%20Drives%20%20The%20SOS%20Guide.htm (4 of 12)9/22/2005 8:25:36 AM


drives in each case, either connected as a pair to the Primary IDE ribbon connector or the Secondary IDE ribbon connector. Use whichever is more convenient. For those running Windows 2000 or XP, the Onboard IDE 'Enhanced Mode' lets you access more than four drives. With the default 'SATA' option for the 'Enhanced Mode Support On' parameter (which, on my motherboard, seemed to do exactly the same as the 'PATA + SATA' option), my original Primary and Secondary Master and Slave drives were left as before, with the new SATA drives appearing as Third Master and Fourth Master. This is the classic configuration for most modern PCs. Unless you really do need a third or fourth hard drive for streaming samples alongside your System and Audio drives, the best configuration is probably SATA drives for system and audio duties, with any CD-ROM and/or DVD drives connected as Primary and Secondary Master on the old PATA ribbon connectors. This gives all four devices Master status. On my motherboard the final Enhanced Mode 'PATA' option moved the SATA drives to Primary and Secondary Master positions, shunting the PATA drives to Third and Fourth Master and Slave. Again, since the vast majority of BIOS designs let you specify a 'boot sequence' for the available drives, so you can decide which one becomes the System drive with Windows on it, this doesn't seem to offer anything different from the default 'SATA' setting.

The Silent Option



While many modern drives are extremely quiet compared with their predecessors, and any remaining noise may be sufficiently silenced by lining your PC's case with acoustic material, movements of the drive's read/write heads can still sometimes be obtrusive in the studio. So once you've decided on the best connection and BIOS options for your needs, you may also want to consider some sort of silent mount for your new drive. Your motherboard may support various

The most famous is Molex's Silent combinations of SATA and PATA drives in Drive sleeve (about £22) which totally configurations like this. Only Enhanced encloses the drive. Its lead cut-outs modes support more than four drives. actually only fit IDE cables, but it takes just a few minutes with a Stanley knife to modify these holes to suit SATA drive cables. Of more concern to many musicians is that some modern drives may run too hot inside it, although I didn't have any problems with my SATA drives and my existing Silent Drive sleeves (see 'Drive Temperatures' box). Other, more recently introduced, solutions include Zalman's Heatpipe Cooler ZM2HC2 (about £25). As its name suggests, this is primarily a cooler, although it does incorporate rubber-damped mountings that greatly reduce the amount of vibrational noise from the drive that is transmitted into the PCs case and amplified. If vibrational noise is your main concern, rubber cradle mounts such as Noise Control's Hard Drive Cage (£25) or the Silentmaxx Rubber HD Isolator kit (£23) may fit the bill. For the budget-conscious there are rubber hard-drive mounts (around £10 for a set of four). The latest deluxe hard-drive enclosure must be the Silentmaxx. Although expensive, at about £38, this 100 percent aluminium enclosure offers maximum heat dissipation and a huge reduction in airborne noise, and it also comes with rubber mounts to reduce vibrational noise. All the above solutions fit into a standard 5.25-inch drive bay, and if you can place them in the incoming cool airflow somewhere near the bottom front of your PC's case it will further aid in drive cooling. Some musicians now swear by drive caddies. A caddy is a two-part device that lets you plug different drives into your PC and reboot it. Having one for your Internet usage and another for music-only duties is the ultimate approach to keeping viruses, spyware and other nasties at bay, and placing your drive in the caddy also provides a useful reduction in acoustic noise. I discussed caddies in depth in SOS June 2004, as part of my 'Ready For The Road' feature. You can buy all these products from suppliers including Chillblast (www.chillblast. co.uk), The Cooling Shop (www.thecoolingshop.com), KoolnQuiet (www. koolnquiet.co.uk), Kustom PCs (www.kustompcs.co.uk), and the one that started it all: Quiet PC (www.quietpc.com).



Things To Save From Previous Installations If you're starting afresh with a clean Windows installation, don't abandon the contents of your old Windows partition before you've saved any data you still need. If you have an imaging utility, such as Norton's Ghost, the safest approach is to save an image of your old Windows partition in compressed form onto a data partition on the new drive, then you can selectively restore specific items from this, as required, to your new Windows partition. The most vital items will be any audio files or synth/plug-in presets you've created. By default, DirectX plug-in presets may be saved somewhere in the plug-in folder structure, while VST plug-in presets may end up in the 'vstplugins' folder. Make sure you drag any you find into a folder on your new drive, or (like me) always save all your presets onto a separate data partition so you know where they all are and can back them up more easily. One thing that's easy to forget is your old Internet Explorer Favourites and Links collection. This may have grown into its present form over several years and could take a long time to recreate. All you need to do to save it for posterity is copy the contents of the 'C:\Documents and Settings\General\Favorites' folder somewhere safe to a data partition that will still be visible on your new hard drives, and then drag it into the identically named folder on your new Windows partition. Transferring email or other software address books is another time-saver, although some ISPs (Internet Service Providers) do store the contents of your Address Book for you, so when you install its software and log on it may be intact. Don't leave it to chance, though!

Starting From Scratch At this stage, those who are adding an extra data drive can simply reboot their PC, have the drive recognised by Windows XP as new hardware, and have any required drivers for it installed automatically. If the drive is larger than 137GB it will require 48-bit LBA (Logical Block Addressing) to access data beyond the 137GB boundary. Windows XP with Service Pack 1 or later, plus a compatible BIOS, will be needed. Most modern BIOS setups auto-detect new drives and their capabilities, and will enable LBA mode for you when appropriate, but if your PC is more than a couple of years old you may either need a BIOS update or to be content with a maximum of 137GB from your new drive. Also, if you've installed Intel's Application Accelerator drivers for your Intel motherboard (see SOS April 2003 for more details) you may also need to update them to provide 48-bit LBA support. Once the drivers have been installed, you can use Windows XP's Initialise and Convert Disk Wizard, which you can find in the Disk Management section of Computer Management, launched by clicking on its shortcut in the Administrative Tools applet of Control Panel. There's an extensive help file to guide you through



the process. You end up with a new drive that's completely 'unallocated', and you can then right-click on it and select 'New Partition' to either create one huge partition or split the drive into several partitions, for Windows and its applications, Data, Backups and so on. This aluminium Silentmaxx enclosure should If you've decided to have a good completely remove drive noise from your PC, clearout and start from scratch with a for the ultimate in hard-drive silencing. freshly installed Windows XP on your new drive, followed by your applications, there's a possible complication if you have a large drive. If your Windows XP CD-ROM is one incorporating SP1 or SP2 you shouldn't have any problems, but if the CD-ROM is the first release, without LBA support, you may find that your drive isn't recognised properly. The answer to this conundrum is to install XP in a smaller partition, upgrade to SP1 afterwards, and then use a thirdparty utility such as Partition Magic to increase the partition size to your desired capacity. For more answers and FAQs concerning 48-bit LBA, and particularly the issues arising with Windows 2000, 98, and ME, the definitive web site to visit is www.48bitlba.com.

Transferring Data At this point some people will be inclined to rip out their old drives to replace them with the new ones, but if you want to save yourself a huge amount of time you shouldn't be so hasty. Since I'd already got several perfectly good Windows partitions on both of my old drives, I decided to follow my own advice, originally featured in SOS March 2003, and clone them on to the new drives, to save me having to install everything all over again. So I left my two PATA drives temporarily in place and plugged in the two new SATA drives, balancing them on cardboard boxes outside the PC's case, with the side panel off. I then booted as usual from my old PATA drive into Windows XP SP2, and let it detect the two new SATA drives and install any drivers it needed to use them. My old data partitions were the easiest to deal with, since I could copy their contents directly, using Windows Explorer, into their partitions on the new drive. Then I used Partition Magic to copy all three of my existing Windows XP partitions across to the new SATA drive I eventually intended to boot from. I then had to check that each partition had the correct information in its Boot.ini file. Regular readers will remember from the March 2003 feature that this file points to the partition in which XP is installed, so a clone will only work first time if its Boot. ini file points to exactly the same partition (first, second, or third) on the new drive as on the old. If not, it will need editing to boot correctly.



Last time around I hadn't cottoned on to the possibilities of Partition Magic's Browse function. With a little fiddling you can use this to modify any file on a currently hidden partition — Browse to its root folder, use the right-click option to copy the Boot.ini file, then browse to a visible partition and paste it somewhere handy. Now you can load the file into Notepad and modify it to point to the new partition, then use the same cut/paste operations to place the modified file back into the hidden partition.

Re-authorising Protected Software Moving the entire contents of an old Windows partition to a new drive means that you'll certainly save yourself a day or two of reinstalling Windows and all its applications. Any software protected by dongles will almost certainly continue to work, as will any that has Pace protection (as used by Waves, amongst others). But the chances are that at least some of your protected software that uses challenge/response authorisation will 'throw a wobbly' and demand re-authorising. Strictly speaking, some applications only allow a single authorisation, or perhaps a couple (one for your desktop and one for a laptop, say), but given that so many of us upgrade hard drives and PCs, developers have to allow us some leeway. Here's a list of the products that needed authorising again on my music partition, and how the developer responded in each case. Since I still maintain an Internet-free music partition I also explain how I dealt with re-authorisation requiring Internet access. * AAS Tassman 4.0 and Lounge Lizard 2.0: Required re-entry of serial number and then issued a new challenge. I copied the HTML pages generated when these new After cloning a Windows partition onto a new challenges appeared to my SDRAM drive, a few of your applications may require card, plugged this into my Internetre-authorising. This is the relevant page for enabled laptop, then ran the pages Waves products. in turn and clicked on their Submit buttons. New responses appeared within seconds, which I could transfer back to my music partition using the SDRAM card and then copy and paste into the authorisation dialogue. Spectrasonics Atmosphere & Trilogy: Generated a new challenge that I could copy and paste onto my laptop and request a new response. Since I'd already used up my two initial authorisations, the submit page asked me the reason for my additional authorisations, but accepted 'Hard Disk Upgrade' as a valid reason and provided me with new ones. Steinberg Groove Agent: This was the only application I had to reinstall from scratch, but it simply required a standard Windows uninstall via the 'Add or Remove Programs' applet, and then installation from the original CD-ROM, which only took a few minutes. Steinberg Wavelab 5.0: Came up with an error message stating that it wasn't properly installed, but I was able to run its existing Uninstall utility and choose its 'Repair Wavelab' option to reinstall it with exactly the same components as before from the original CDROM, and without disturbing the subsequent 5.01a update.



TL Audio EQ1 plug-in: Simply required the original CD-ROM to be placed in the drive so that the plug-in could be re-opened successfully. Waves plug-ins: These generated a new challenge, since they were no longer on the drive that was previously authorised, although if you have an iLok key (for plugin versions 4.0 and above) you will be immune to drive changes. I copied this challenge to Notepad and saved it as a text file, logged on via my laptop to my Waves Account using my ID and password, used the 'Request Disk Re-authorisation' option and got a new response shortly afterwards.

Attack Of The Clones So far so good — I could now reboot into the BIOS setup and navigate to its Boot page, where I could change the Boot Device Priority to point to my new SATA drive, save the new setting and then reboot, so that Windows XP now booted from the clone on the new drive rather than from the original drive. However, for the first time ever I experienced the problem that's been plaguing some other musicians who've tried to clone a Windows XP partition — the clone seems to work, but is actually still linked to its originating partition, and won't run without it. The giveaway is that if you examine your drives in Explorer after booting into the new cloned partition you'll still see the old drive designated as the C: system drive, while the clone has some other drive letter (mine ended up as F:), even though your PC now boots from the clone and uses the bulk of its files. It didn't matter whether I created the clone by copying the existing XP partition across, via Partition Magic, or using a previously saved image file made with Drive Image — the problem was still there. Fortunately, I remembered participating in a very in-depth discussion about XP partitioning on the SOS Forums, and was able to track down the solution. It's actually quite simple once you know how. First, note down the drive letter of the clone's partition as shown in Explorer, and then open the registry editor, by choosing the 'Run' command from the Start menu and typing in 'Regedit'. Now locate the entries under the heading 'HKEY_LOCAL_MACHINE \SYSTEM\ MountedDevices\'. Near the bottom of this list you'll find a set of DosDevices: delete the existing \DosDevices\C: entry, and then After cloning a Windows partition it's just as rename the one for your clone (mine well to clear out any unwanted files, using a utility such as Iomatic's Disk Medic. was \DosDevices\F:) to \DosDevices \C:. When you reboot, your clone will finally be standing on its own two feet (so to speak) and you can safely disconnect the old drive. Strangely, the other two XP partitions I cloned onto my new SATA drive



remained totally independent, requiring no further Registry tweaks, but using this cloning technique to move all three existing XP partitions across to the new bootable drive saved me countless hours of reinstalling Windows and all its applications in each one, and I didn't have to contact Microsoft by phone or Internet to reauthorise any of them either. Once all the data had been copied across or cloned, I could finally power down and unplug my two old drives, pull them out of their Silent Drive sleeves, and bolt the new ones into the sleeves instead. However, while you're shuffling drives, a few words of warning. During the process I found a few instances where unplugging my old PATA Primary Master drive resulted in my PC not being able to boot up. This was because, despite designating one of my new SATA drives as First Boot Device, the BIOS was still looking for the previous Primary Master boot drive. The only way to continue was to plug it back in so that I could enter the BIOS to disable the auto-detect function for the Primary IDE Master drive. So before you completely remove your old drives, just unplug their power cables temporarily to see if everything boots successfully without them. I still prefer to use the Boot Magic utility to choose between my various Windows partitions at boot-up, rather than using Windows' own boot menu, so I installed Boot Magic onto my first new XP partition, used its configuration utility to choose the appropriate partition options for my new boot menu, and then rebooted. Unfortunately, no boot menu appeared, and it took me some time to track down the cause of the problem. It transpired that Boot Magic always modifies the boot sector of 'Drive 1' to add its menu options, rather than whichever drive you choose to boot from in the BIOS, and in my PC 'Drive 1' was still one of my old PATA drives, which was being used to transfer various other data partitions. As soon as I physically unplugged the PATA drive, the SATA boot drive became Drive 1 and the Boot Magic menu could be successfully saved to it.

The Clean-up After this huge changeover, for each of my new Windows XP partitions I used the steps described in PC Notes June 2004 to make any hidden and no longer used devices visible in Device Manager — most of the 'greyed out' entries due to unconnected hardware (and therefore suitable for deletion) were to be found in the Storage volumes section and labelled 'Generic volume', because of many changes in partitions, along with two entries in the Disk Drives section for the two PATA hard drives I'd now unplugged. I had a few applications with challenge/response copy protection to re-authorise (see 'Re-authorising Protected Software' box), and a little tidying up to do. I used Iomatic's Disk Medic (www.iomatic.com) to ferret out loads of temporary, obsolete, backup, and empty (0-byte) files, plus invalid Start Menu links. It found at least 40Mb of these on each of my Windows partitions, and it also offered to clear various Internet Browser caches, histories and cookies, and wipe the recent document histories of 96 different applications. At just $19.95, Disk Medic has file:///H|/SOS%2002-05/Updating%20PC%20Hard%20Drives%20%20The%20SOS%20Guide.htm (11 of 12)9/22/2005 8:25:36 AM


proved to be a very handy utility. I then further cleaned up the Registry in each of my Windows partitions, using Microsoft's Reg Clean and Iomatic's Registry Medic, then defragmented each of them. So there you are. Installing a new drive can be comparatively easy, but sometimes there can be quite a few complications along the way. I hope this feature will help you avoid most of them! Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q. Can I feed a 24-bit signal into a 16-bit device?

Q. Can I feed a 24-bit signal into a 16-bit device? Published in SOS February 2005 Print article : Close window

Sound Advice

I want to record eight tracks at once into my Akai DPS12 multitracker, which has six analogue inputs and a stereo S/PDIF input. All the sources will be analogue, either from mics or DI boxes. As I need to get a portable mixer/preamp box for this project as well, I thought a Soundcraft Mseries mixer would be perfect, as it has an S/PDIF output in addition to direct outs and would easily allow me to put down the required number of tracks. However, the S/PDIF output is fixed at 24-bit on the Soundcraft, and the Akai's S/PDIF input is fixed at 16-bit. The digital outputs on the other options I've looked at, such as the Focusrite Octopre, are also fixed at 24-bit, which seems to be the standard these days. So now I'm looking at getting a completely analogue mixer and a separate A-D converter to convert two of the outputs. Can you see another solution? SOS Forum Post Technical Editor Hugh Robjohns replies: Feeding a 24-bit signal to a 16-bit device will result in the truncation of the bottom eight bits — the device receiving the signal simply ignores the bottom eight bits. This effectively raises the noise floor by more than 40dB but, more importantly, will increase the amount of distortion considerably, especially for low-level signals. This will tend to make things sound 'hard' and 'grainy', but just how noticeable and/or objectionable this will be depends on the kind of music you're recording and the level of quality you expect. Ideally, you should find a way of re-dithering the 24-bit signal to 16bit before outputting it from one machine to the other. Pretty much all fully digital sound desks and computer DAWs have facilities for this, but few modern A-D converters or digital signal processing devices have the capability — most operate with a fixed 24-bit resolution, as you say. It's worth noting that the S/PDIF output on the Soundcraft M-series mixers will only carry the main stereo output signal, so if you want to use it to get two separate channels of audio into the Akai, you won't be able to use the mixer for anything else at the same time.

The Akai DPS12 multitrack recorder has a fixed 16-bit S/PDIF input.

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Q. Can I feed a 24-bit signal into a 16-bit device?

As an alternative to the Soundcraft M-series mixer, you could look out for a second-hand digital desk, like a Soundcraft 328 perhaps, or a Yamaha 01V. All modern digital desks will offer switchable output resolution. If you decide to go with an analogue mixer and buy a separate stereo A-D converter, find one that offers switchable output resolution, so that when you upgrade the recorder you won't have to upgrade the A-D as well. SOS contributor Tom Flint adds: A few years ago, when an Akai DPS16 was my main recording device, I encountered the same dilemma. In my case, I wanted to record 10 tracks simultaneously, using the DPS16's eight analogue inputs plus its stereo S/PDIF in. I first began looking for a stand-alone A-D converter which would allow me to feed the Akai with the stereo analogue output from one of my sound modules, but most dedicated converters were very pricey. Then I noticed that the Lexicon MPX100 multi-effects processor had an S/PDIF output and a Bypass button, which made it possible to use it to convert any stereo analogue input into digital, with or without using the internal effects. I picked on the MPX100 purely because it offered a reasonable specification at a very low retail price. I wasn't actually in the market for an effects box, but having some Lexicon algorithms available was a bonus. In terms of compatibility, the MPX100's S/PDIF output is 20-bit, which matches neither the 16- nor 24-bit modes offered by the Akai DPS16. Ideally, some sort of dithering was required, but my intention was to feed the Lexicon with the signal from a drum machine or sound module, and not for anything really sensitive, like vocals. I managed to borrow an MPX100 to test how much difference the lack of dither would make, and found it not to be a problem, so I went ahead and made the purchase. I certainly don't think it would be worth spending large sums of money on additional equipment just to get around the bit-rate discrepancy, particularly for the sake of the DPS12, which is a budget product, and could easily be replaced with a machine that records up to eight tracks or more. Matching the bit rate wasn't the only consideration, though. In order to use the digital input, the Akai had to be slaved to the incoming digital signal, and therefore be locked to the Lexicon's internal clock rather than its own. This was a cause for concern, because it meant that the Akai's A-D performance could be compromised by a poor pulse, just as it could be improved if fed a more stable clock. I don't remember if I made any investigation into the relative quality of the clocks, but my own recording tests didn't lead me to think that the MPX was doing a poor clocking job in any way. Sample rate was also a consideration because although the MPX has a fixed rate of 44.1kHz, the DPS16 can operate in 32, 44.1, 48 and 96kHz modes. Nevertheless, I eventually decided that I'd rather save disk space by running the Akai at 44.1 than try to obtain higher quality from a mid-priced machine that probably isn't quite good enough to demonstrate the benefits of 96kHz anyway. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q. How can I get my multisampled synths to sound more even?

Q. How can I get my multisampled synths to sound more even? Published in SOS February 2005 Print article : Close window

Sound Advice

I have recently bought an Akai S5000 sampler and I am attempting to create samples of some synths I have. Getting the samples in is fine, and so is sample editing. I understand the concept of multisampling and I am taking a sample every octave, but when I map them out across the keyboard, I can hear very noticable tonal changes between the keygroups. How can I overcome this? Andy Blakey Sound library developer and SOS contributor Steve Howell replies: You're on the right track with multisampling, but one sample per octave is probably not enough in this case. Even with multisampling, samples are still subject to 'munchkinisation' — that is, a change in tonal quality as a result of speeding up and slowing down the original sample when you play different notes/pitches. You don't say which note you are sampling for each octave or your keygroup low/high notes, but let's say you're sampling the F of every octave with a keygroup range of Cn-Bn. What's happening is that at, say, B2, your F2 sample is transposed up six semitones and so is sounding a bit bright and thin and, well, 'speeded up', but at C3 right next to it, your F3 sample is transposed down five semitones and is sounding a bit dull and, well, slowed down! Juxtaposed next to each other , as you play across keygroups, the tonal jump may well sound abrupt. So what's the solution? Of course, the most obvious one is more samples per octave, so that none of them is transposed too much to exhibit these 'munchkinisation' effects — maybe the C and G of every octave with keygroup mapping from A#-E# and E-B respectively so that at any time, the samples are only transposed by two or three semitones up or down. Of course, even more samples would be better — sampling You can produce more even-sounding every minor third (C, E#, F# and A of every octave) would mean that multisample patches by using the Akai samples are only ever transposed by plus or minus one semitone. S5000's Keygroup Crossfade function. However, this brings its own problems, not least of which is that there are twice as many (or more) samples to edit, trim, normalise, loop and so on. So what's to be done for a quick fix? Firstly, I'd head for the S5000's Keygroup Crossfade function. By overlapping the keygroup ranges and enabling crossfading, at the crossover point of the keygroups you would hear both samples, with each one file:///H|/SOS%2002-05/Q.%20How%20can%20I%20get%2...tisampled%20synths%20to%20sound%20more%20even.htm (1 of 2)9/22/2005 8:26:13 AM

Q. How can I get my multisampled synths to sound more even?

merging into the other — this can actually work quite effectively for smoothing out the transition between adjacent keygroups. The only problem you can run into is chorusing and/or phasing artefacts at the crossover point if the adjacent samples aren't quite in tune. That said, a judicious tweak of each keygroup's tuning can temper this. Another technique you can try is to experiment with the upper and lower extremes of the keygroups. One's first assumption is to have the sample somewhere in the middle of a keygroup (for example, samples on F, keygroup ranges of Cn-Bn). However, depending on the nature of the sound, this might not always be appropriate. For example, you might have a sound that transposes up well but not down. In this case, try a keygroup range of, say, E#n-Dn+1 so that the sample isn't transposed down as far but extends further up. The converse could also be true. Of course, the only real solution is to record as many samples as you can across the instrument's range (ultimately, one for every note!) and dispense with the hassles of looping by simply recording much longer samples than you will ever need to sustain. When I started making sound libraries back in the '80s, entire pianos and orchestral string sections had to fit into 2MB of memory and be stored on 1.4MB floppies. These days, these limitations simply don't apply with the latest raft of hardware and software samplers that feature either large amounts of RAM and/or disk streaming. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q. Should I always pan my vocals to the centre of the mix?

Q. Should I always pan my vocals to the centre of the mix? Published in SOS February 2005 Print article : Close window

Sound Advice

I'm about to do some mixing and I'd just like to find out if there are any right or wrong ways to pan my audio. I'm sure I've read somewhere that vocals should always be panned centrally, as well as the bass, while it's OK to pan other things like guitars and the spread of the drum kit. Jeremy Rodgers SOS contributor Tom Flint replies: It does seem to be standard practice to pan lead vocals, and often the bass, centrally, and some people believe that this configuration is a must. Nevertheless, there is nothing to stop you breaking such 'rules', if you have good reason to do so. Although the mixing conventions you've mentioned are now well established, when stereo recordings first became commonplace in the 1960s, no-one was really sure how best to use the format. You only have to listen to a few compilations of songs from that era to hear some pretty interesting panning! Today, most people have become accustomed to thinking of the output of a stereo speaker pair as a single, even spread of sound, with the most prominent elements of the mix, such as lead vocals, appearing in equal amounts in each speaker. This is easily achieved now that we often have every element of a recording on a separate track, and the freedom to pan each track anywhere in the stereo field — but in the early days of stereo, recordings were often made to three or four tracks, and panning options consisted of hard left or hard right. Some engineers thus treated stereo as providing a second speaker which they could use to showcase their lead instrument or singer. This would, therefore, often be panned totally over to one speaker so that it would stand out, leaving the majority of the other instruments and backing vocals in the other speaker. Many songs mixed in that way actually still sound very natural.

It's up to you whether you choose to follow the accepted panning conventions, or try something a little less predictable.

It's perhaps worth noting that those records which featured extreme panning usually still had a little vocal appearing in the opposite speaker, and some instrument spill beneath the vocal, providing a unifying ambience. Sometimes the spill would have been a result of everything being recorded in one room, although the practice of playing a recording into an echo chamber, recording the result, and using it in the mix, was also common.

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Q. Should I always pan my vocals to the centre of the mix?

One of the reasons that the modern stereo mixing convention of placing the vocal in the middle prevailed was that it's the best way to ensure that the recording will work on the majority of sound systems. If, for example, a listener has a dodgy speaker, or has placed one behind the sofa, a panned vocal could be lost or compromised. Therefore having the most important thing in both channels ensures that the core of the song always remains intact. Bass guitar, and usually the bass and snare drums in a kit, are also placed fairly centrally these days, so that the punchy sounds which tend to drive a track along are evenly spread, giving a recording a unifying balance in both speakers. Panning low bass sounds centrally also allows a vinyl record to be bassier without causing the needle to jump. Other instruments are less unsettling when panned to extreme positions, and tend to be scattered across the stereo spread to increase the perceived width of the sound picture. In recent years, surround sound has forced engineers to choose between a number of different mixing strategies, each one suggesting a different way of spreading a mix across the array of speakers. When mixing a band, for example, in 5.1, some engineers like to have all the instruments spread out across the front speakers so that the listener feels as if they are sitting in the audience in front of the stage. The rear speakers are then used for ambience effects, emulating the reflections the listener might hear from a concert hall's walls. More experimental mix engineers have tried placing instruments, and even separate notes and beats from instruments, all around the room, arguing that a recording can be a unique listening experience, not necessarily a realistic reproduction of a live event. Even though there are strong arguments supporting both these examples of mixing approaches, in my opinion, it is likely that over time surround mixing will become more and more standardised, and the resulting recordings progressively more conservative, just like the development of stereo mixing. At the end of the day, it's up to you whether you follow the accepted panning conventions, or elect to go for something a little less predictable. Either way, the success of your mix may well hinge on something other than panning, like the relative levels of the individual parts of your mix. Published in SOS February 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q. What kind of mic setup should I use to record a choir?

Q. What kind of mic setup should I use to record a choir? Published in SOS February 2005 Print article : Close window

Sound Advice

I have been planning to record an upcoming choir concert with a centred ORTF stereo pair of Rode NT5 cardioids. It's a 30-person choir and the location is a 900-year-old Danish stone church. The problem is, the choir director won't let me put a mic stand in the middle of the centre aisle because it is narrow, the church will be crowded and she has to go back and forth to play the organ in the back of the church for several numbers. However, most songs will be a cappella. There's nothing suitable to hang the mics from, and I only have ordinary K&M mic stands, with small booms. Since the choir is in the front, and the organ is in the back, I figure a good solution is use to a spaced pair of omni condensers, such as AT3032s, on stands on each side of the aisle, a couple of metres back from the choir (about the third pew). As an alternative, I did find a short gooseneck section to put at the end of the boom and attach to an ORTF stereo bar. If I can tilt the boom a bit out toward the middle I will still be able to bend the gooseneck up so the bar is level. This is my first choir recording, and I haven't bought the AT3032s yet, so I'd really appreciate any advice or suggestions, on mic placement (height and distance from choir) and choice of mics. SOS Forum Post Technical Editor Hugh Robjohns replies: A stone church is likely to be very reverberant so I would think spaced omnis would produce a very 'swimmy' sound, and your only way of balancing the choir against the organ would involve moving the mics up or down the church, which doesn't sound like it will be a popular or easy thing to do. I'd stick with the original idea of ORTF NT5s for the choir, as their cardioid polar patterns will tend to reject the direct organ sound coming from behind them and thus give you more control over the choir-to-organ balance. I would then rig a second mic or pair of mics specifically to capture the organ. Cardioids facing the organ would tend to reject the direct choir sound and thus give more control again, although a fairly close pair of

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omnidirectional mics would give a big organ sound and lots of spacious reverb which might work well. In all likelihood, the organ mics will have to be on very high stands to get a good balance of all the pipe sounds. As for mounting the choir mics, I'd be very wary of using a gooseneck as they tend to droop, and the last thing you need halfway through a recording is for the mics to end up pointing downwards at the end of a sagging gooseneck! You don't necessarily have to use a single stand in the centre of the aisle for ORTF miking — you could use two separate stands, one either side, with their boom arms stretched over the top like an arch. Mount one mic on each stand and position as required. There is no law to say the two ORTF mics have to be mounted on the same bit of metal. As long as you are careful to get their relative spacing and angles right (see the diagram on the facing page) you'll get exactly the same results from two mics placed on separate stands. If your existing stands aren't big enough to accommodate this (or if the idea of two stands doesn't appeal), then it might be better to invest in a single big stand with a long boom instead. Place the stand safely to the side of the aisle, and extend the boom arm over to place the mics in the centre above everyone's heads. When using any stand with an outstretched boom arm, always make sure that one of the stand's legs is extended directly below the boom arm, to ensure that there is no risk of toppling. I use heavy-duty K&M tall stands with long boom arms for this kind of job, as they are very stable and thus safer, as well as more versatile to use. The legs of these stands do splay rather wide wide, though, and although this is what makes them very stable it can present a trip hazard. I use stripy black and yellow 'hazard' tape to make the legs of the stands more visible, and it's also a good idea to try to tape or rope off the area around the stand whenever possible.

The Rode NT5's cardioid pattern will reject sound coming from the rear, which can be useful in live recording situations.

With either of these 'up and over' approaches, just make sure that you tighten the boom arm clamps well. If there is any possible danger that a collapsing mic stand might hit someone, I usually also use a length of chain or rope attached to the counter weight end of the boom arm and fixed to the base of the stand to ensure that the boom arm can't swing down and hit anyone accidentally. In terms of positioning for the ORTF pair for the choir, I'd suggest starting at about three metres back and the same height if you can, and then adjust as necessary. With small-diaphragm condensers it won't make a huge difference whether the mics are perfectly horizontal or pointing slightly down, although the associated tonal changes would be far more obvious with large-diaphragm condensers. Remember that the audience will soak up some of the reverb, so don't worry if the rehearsal sounds slightly (and I do mean only slightly) too wet. Of course, it is fairly easy to add additional reverb later if the recording ends up a little too dry, but the reverse is impossible, so it's best to err on the side of miking too close rather than too far away. Published in SOS February 2005






Q. Will connecting a high-quality D-A converter improve the sound of my CD player?

Q. Will connecting a high-quality D-A converter improve the sound of my CD player? Published in SOS February 2005 Print article : Close window

Sound Advice

Will it make a big difference to my sound if I connect an inexpensive CD player with a digital out to a high-quality D-A converter, such as an Apogee Mini DAC or a Benchmark DAC1, in order to bypass the low-quality converters built into the CD player? Some people argue that it will, because going out of the digital optical out will bypass the CD player's cheaper converters. Some argue that it won't, because even though you're going out of the digital optical out, you are still passing through the low-quality converters. In fact, connecting the player to a really good D-A converter might be even worse, because you would be able to hear the inferior converters at work. Also, another issue to consider would be that of clocking. An inexpensive CD player might have a low-quality clocking system as well and might not be able to act as a slave and run off the better clock in the D-A converter. Who is right? I was thinking about picking up a high-quality D-A converter and using it in conjunction with my RCA CD player. Sam L Editor In Chief Paul White replies: Using better converters on a cheapo CD player should improve the sound quality as you are bypassing the converters in the CD player. Digital outputs don't have digital-to-analogue converters in line with them by definition.

A high-end D-A converter like that included in the Apogee Rosetta 200 will do a better job than the converters in a domestic CD player.

The big 'however' is that the CD player may have a poor transport that generates a high error rate and there may also be significant levels of clock jitter, which will degrade the signal, specifically affecting the noise floor and stereo imaging. Nevertheless, I'd be very surprised if the better converters didn't bring about some improvement.

Being practical, however, unless you already have the converters or are buying them primarily for some other purpose, it probably isn't worth buying anything too fancy to hang on the end of an indifferent CD player. A pragmatically chosen mid-price converter might sound just as good, or you could consider a professional-

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Q. Will connecting a high-quality D-A converter improve the sound of my CD player?

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


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