Sound on Sound (November 2005)

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November 2005 In This Issue Click article title to open Reviews

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Ableton Live 5

Leader: Happy 20th Birthday SOS!

Loop-based Sequencer [Windows/Mac OS X]

Paul White As SOS starts its 21st year, Editor In Chief Paul White casts an eye back over the past 20 years and chats about this month's SOS DVD002.

Ableton Live is one of the software success stories of the last few years, with devoted users in fields as diverse as DJing, remixing, theatre sound and music production. Version 5 addresses requests from all these areas, whilst retaining the program's renowned ease of use.

Apple Soundtrack Pro Audio Sequencer & Editor [Mac OS X] Apple's loop-sequencing application has grown up, with the addition of sophisticated recording, editing and mixing facilities, a powerful waveform editor, and many of Logic's most sought-after effects.

Cakewalk Sonar Home Studio MIDI + Audio Sequencer [Windows] With Sonar Home Studio, Cakewalk have made the core features of their flagship sequencing application available at a bargain price.

Manu Katché Session Drumming & Music Technology He is one of the most famous drummers in the world, having played on more than 200 records. His CV reads like a Who's Who of English, American and French popular music, and even if he prefers not to lose himself in computers, he always takes a PowerBook with him, to write songs, at home or on tour. Meet Manu Katché, drummer extraordinaire.

Sounding Off: Stephen Bennett In a desert of noise, the personal MP3 player is an oasis of quiet...

Studio SOS Bella Saer

Crane Song Avocet Stereo Monitor Controller This new monitor controller sets new standards for operational flexibility and sonic fidelity.

Digidesign M Box 2

In another exciting installment of our studio 'makeover' series, the team rotate Bella Saer's entire setup through 90 degrees in search of a more effective working environment.

The Go! Team: Recording Thunder, USB Recording Interface [PC/Mac] Digidesign have given the most affordable interface in their Lightning, Strike Pro Tools range a makeover, with a new case, new preamps Gareth Parton and some additional features. East West Quantum Leap Symphonic Choirs Kompakt Instrument [Mac/PC] Ever fancied arranging 'Oops, I Did It Again' for a full choir? Well now you can, courtesy of Symphonic Choirs, the latest virtual-instrument sample library from East West and Quantum Leap.

Kawai MP8 Digital Stage Piano file:///F|/SoS/SoS%2011-2005/Contents.htm (1 of 4)10/19/2005 9:40:35 PM

The Go! Team's debut album, a glorious pile-up of mangled samples and lo-fi home recordings, is now attracting widespread acclaim — but its path to Mercury nomination and commercial success has been anything but smooth.

Technique

ReBirth & Reason

In This Issue

Kawai's last few digital pianos have been finely wrought things of beauty: solidly built keyboards with an amazingly realistic playing action and beautifully sampled piano timbres. But their latest claims to surpass all of those. Can it possibly be true?

Reason Notes ReBirth is dead. Long live ReBirth! We discuss the decision to discontinue this pioneering software and discover how Reason users can keep its spirit alive.

Korg OASYS: Part 1

News & Updates We round up a month of small, yet interesting, product releases from Apple, as well as looking at how the company is making it easy for Linux developers to port audio software to Mac OS X.

Workstation Synth

For over 15 years, Korg have produced the world's most successful workstation synths, and the OASYS is their new £5400 flagship, their attempt to take the concept to the next level. In the first instalment of our two-part in-depth test, we assess how they have fared...

Lexicon MX200 Multi-Effects Processor Lexicon's latest hardware reverb is designed to be as easy to control from your computer as a plug-in.

M&K MPS1611P Active Monitors

Apple Notes

CLASSIC TRACKS: Jimi Hendrix Experience All Along The Watchtower Eddie Kramer With his searing version of 'All Along The Watchtower', Jimi Hendrix set a standard for Dylan covers that has rarely been equalled. Eddie Kramer was behind the glass as the sessions moved from London to New York.

Digital Performer Notes News & Tips We've got news of some interesting new plug-ins this month, and the solution to one of those software compatibility problems that can drive you bananas...

Emu's Patchmix DSP Demystified... These new speakers from M&K let you switch between reflex and infinite-baffle designs on a whim.

Technique We demystify the powerful internal routing and mixing engine within Emu's recent popular range of computer audio interfaces.

Plug-in Folder

Logic Notes

Camel Space & DVR2 We test two hot new plug-ins: Camel Audio Camel Space — Formats: PC VST & Mac OS X AU TC Electronic DVR2 — Formats: Mac/PC Powercore

News & Tips More tips and news from the world of Apple Logic.

Rayzoon Jamstix Virtual Drummer PC VST Instrument

Managing iLok Plug-in Licences Technique There can't be many Pro Tools users left who have not had to invest in an iLok key in order to run their favourite plugins. But did you know that you can insure, transfer and even buy and sell iLok plug-in licences electronically? Find out more...

Mastering Reason 3 Mixes Virtual band members are now a regular feature in SOS Technique reviews, but how about a virtual drummer that actually will Reason 3 has several new tools for pumping up final mixes jam with you? Rayzoon claim Jamstix is able to do just that... and creating that 'finished product' sound.

Sample Libraries: On Test

PC Notes

Hot Releases Assessed We check out the latest sample libraries on the block: Reason Drum Kits ***** Beats Working In Cuba ***** The Giovani Edition *****

News & Updates This month we explore the relevance of User accounts and the vital contents of the Windows Documents and Settings folder.

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Pointing Devices For The PC Musician

In This Issue

SE Electronics Titan Multi-pattern Condenser Microphone

PC Musician If you've had enough of chasing a mouse around your desk, there are many other ways of controlling the onscreen pointer of your music software. We examine the options.

Pro Tools Notes

The capsule within this microphone has a diaphragm coated News & Tips with titanium, rather than the usual gold. We find out Once again, there's a bumper crop of news from Digi-world whether it makes any sonic difference... to report, including the long-awaited jump to version 7 of Pro Tools. SSL AWS900 Mixer & Control Surface

Sonar Notes

With their latest Superanalogue console, SSL bring the sonics and functionality of their flagship SL9000K to studios with smaller rooms and budgets. In the process, they've incorporated comprehensive control-surface facilities for driving computer DAWs remotely. But can the AWS900 really live up to its pedigree?

Steinberg Halion Player v3.1 Sample Library Player [PC/Mac] If you want the sample playback features of Steinberg's Halion sample player, but you don't need its detailed editing facilities, there's now an affordable answer to your needs and it includes the full Halion sound library, too...

Waves GTR Guitar Tool Rack Guitar Processing System [PC/Mac]

News & Tips When the version number's most significant digit increments, you know something big is going to happen — what's inside Version 5?

The Lost Art Of Sampling Part 4 If you want to artificially extend your instrument samples, or make entire backing tracks from one rhythmic snippet, you'll need to know about looping and time-stretching. And then there's keygrouping... We explain these fundamental sampling processes, and more.

Using Sonar's Dreamstation Soft Synth Technique This modest little soft synth has been bundled with Sonar since version 1.0 and is very kind to your CPU resources, yet many Sonar users are still unaware of just how much you can squeeze out of it. We present some evidence.

Using Hardware Controllers With Logic

Software guitar amp modelling has been around for a while, but GTR is Waves' first foray into the field — and as you'd expect, they have made a huge effort to get it right, even hiring guitar maker Paul Reed Smith to design a bespoke preamp.

Competition

Technique Not everyone can afford to invest in a dedicated high-spec hardware controller such as Logic Control. However, you can use even the most humble of MIDI devices to control your mixer, software synths, and plug-ins instead.

Using V-Racks In Digital Performer 4.6 Technique One of the most exciting and unexpected new features unveiled in Digital Performer 4.6 is V-Racks, and, like many of the best things in life, it's something you probably didn't realise you needed until it came along...

WIN Propellerhead Reason 3, Ableton Live 5, M Audio Key Rig, Drum & Bass Rig, Reason Strings, Drum Kits Refills + Working With Video In Cubase SX/SL ProSession sound libraries. Sound Advice

Q. Can I connect an AES output to an S/ PDIF input? Q. Did Mellotrons use tape loops or not? Q. How can I make the most of a small

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Technique Since its SX reincarnation three years ago, Cubase has once again become one of the most flexible tools for writing music to picture. Let's investigate...

In This Issue

studio space? Q. What is that 'robot voice' effect? Q. What is the best way to mic up a clarinet? Q. What is the difference between mono with one speaker and mono with two? Q. Why is the signal louder when it is panned to the centre?

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Ableton Live 5

In this article:

Ableton Live 5

All Things To All People Loop-based Sequencer Opening DAWs Published in SOS November 2005 Sound Content Plug-In Delay Compensation Print article : Close window Pocket Calculator Reviews : Software Instruments & Other Devices See You Later, Arpeggiator Simpler Updated Organisational Tools

Ableton Live 5 £299 pros Unique, powerful and simple compositional and recording tool providing a huge fun factor with serious results. Completely puts you in control of all the little bits of audio that make up your hard drive: every file is instantly usable. A big selection of sounds and sound-design Devices, with some interesting new sound-manipulation tools. More powerful Arrangement view with MIDI/keycontrollable locators. Reliable.

cons Too few to spoil the experience.

summary

[Windows/Mac OS X]

Ableton Live is one of the software success stories of the last few years, with devoted users in fields as diverse as DJing, remixing, theatre sound and music production. Version 5 addresses requests from all these areas, whilst retaining the program's renowned ease of use. Ingo Vauk

Version 5 of Ableton's Live loopsequencing environment is a serious, feature-packed update, and I can believe the Ableton team when they describe it as their largest development effort since the first version of the program. Ableton have integrated a lot of user requests in this release, and the good relations they maintain with their user base really seem to have paid off for both parties.

If you liked previous versions of Live you'll love this. Ableton have stayed true to their simple design and interface while adding high value to the software, expanding its capabilities in a very creative and usable way. I love it!

In terms of the user interface and graphic display of the software, Live 5 hasn't changed much since the last release, and if you're not familiar with the program I suggest you take a look at Sound On Sound's reviews of Live 4 (September 2004: www.soundonsound.com/sos/sep04/articles/live4.htm) and the original Live (February 2002: www.soundonsound.com/sos/feb02/articles/ableton.asp). There are a few enhancements, like the resizable mixer channels in Clip view, but overall, Live 5 doesn't look much different from its predecessors. To handle information the much greater complexity of the program, Ableton have introduced a couple of £299 including VAT. M Audio UK +44 (0)1923 new buttons to control the display, and a number of contextual menus that can the accessed via right mouse-clicking (Windows) or Ctrl-clicking (Mac). It is worth 204010. familiarising yourself with these menus early on, since they hide a lot of goodies, +44 (0)1923 204039. especially in the file-browser department and with regard to fine-tuning warp Click here to email file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (1 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

www.maudio.co.uk

markers.

www.ableton.com

All Things To All People While Ableton's aim with this release is to bring the software to the status of a serious DAW, they haven't forgotten their large contingent of DJ users and remixers, and some of the major improvements target these groups. With the introduction of support for the MP3 file format, Ableton have made the entire contents of everyone's digitised music libraries instantly accessible for what they lyrically call 'guaranteed mash-up gratification'. Not being much of a DJ myself, I had a go and must say that I'm a lot closer to beat-matching than I ever was, while combining Rachel Podger's Bach Sonatas and Partitas for solo violin with select NWA beats made for interesting listening (for a while...). There's also a new automatic track tempo-matching feature, which sets warp markers for imported audio files and is a huge time saver. The function works well, but in most cases it is worth following the manual's suggestions on how to help Live along to the best result. I found that often the general tempo of a recording was tracked quite accurately, but almost invariably the downbeat was missed, resulting in both a late trigger and an offset. Using the optional tools in the Clip Edit window (accessed by right-clicking or Ctrl-clicking on a warp marker) makes fine adjustments to the overall timing easy and painless, and spending about 30 seconds to prepare a track you want to 'mash up' doesn't seem too much to ask. A nice touch is the ability to batchprocess files by Ctrl-clicking a folder containing audio in the Live browser and using the Analyze Audio option (if you select a folder that only contains one audio file, this option doesn't appear, which had me fooled for a while — the logic, I guess, being that you're not really batch processing.) Using the batch processing I set up Live to anaylse my entire iTunes library overnight, and could get right into mash-up mode the next morning.

Right-clicking on a warp marker in the Clip Edit window allows you to fine-tune Live's automated settings.

The new, dedicated 'complex' warp mode should be mentioned here. This is a frequency-domain warp mode that is specifically designed to handle composite, mixed-down signals. It works well, though it is very processor-intensive, exerting about 10 times the CPU load of the other warp modes. The new Freeze Track feature comes in handy in this context, but more about that later.

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Ableton Live 5

Another feature that is targeted at remixers is the new Clip Transport, though this has more far-reaching implications for the use of the Session view in general. In Live 4 it was impossible to audition an audio or MIDI clip from somewhere in the middle, making it necessary to play the clip from the beginning in order to hear a certain section. My way around that used to be a quick transfer over to the arrangement view, but this also had its drawbacks, namely that it was less then instantaneous, and sometimes got in the way of an existing arrangement. In Live 5 you can now click anywhere in the waveform and the software will, true to Ableton's obsession with sync, jump there at the next global quantise value. In conjunction with the 'set' buttons introduced to define beginning and end points of a clip, on-the-fly editing of envelopes and events and positioning of long sections of audio or MIDI has become a lot quicker and more intuitive. Clip loops are now decoupled from the start and end points of the clips, which means that you can set up a section that runs continuously to a certain point and loops thereafter. Using the 'set' buttons you can also set this up on the fly. As with the vast majority of Live features, these functions can be mapped to MIDI controllers.

Opening DAWs As I mentioned earlier, Ableton are looking to become a serious contender in the market for digital audio workstation (DAW) recording software, and have thus introduced a number of features in the Live 5 release that should help them here. While it is undisputed that the simplicity, workflow, ease of operation and nondestructive nature of audio processing all make Live a very attractive tool to work on, there is an aspect to the system that needs specific attention when working in a pro audio, studio environment. Since Live is also a live performance tool, Ableton have built in a great deal of resilience and stability. The application hardly ever crashes, and you will need to push it very hard for the audio to stop completely. However, when you are edging towards that point of system overload both audio quality and timing accuracy begin to deteriorate. Both these artefacts are not acceptable in a professional studio situation, and other applications get around the problem by limiting the voice count and cutting audio that doesn't arrive at the output within spec. In a studio situation, where you have the opportunity to work around such problems, this is good practice in my opinion, since it gives the engineer peace of mind as far as the digital audio quality is concerned. The dynamic nature of Live obviously conflicts with such an approach, and personally I wouldn't want to trade all the great creative possibilities Live 5 offers in favour of deaf trust in the audio. However, it should be pointed out that with Live you have to use your ears to judge the audio: there can be degradation well before you see obvious signs of processor overload such as jittery graphics, and it can be subtle until you know what you're listening for. Ableton's head developer responded to my query on this question as follows: "There is no short answer to this. The behaviour of Live's audio processing under heavy load is a complex thing and heavily file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (3 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

dependent on the processor type and system architecture (multiple processors or cores). Of course we took care to give the most critical tasks the highest priority (recording over playback, playback over interaction and display)." It is always good practice to run a system with a degree of processing headroom and this applies even more here, since you are likely to have much more dynamic load variations on the processor than in more conventional applications. Having said that, Ableton have introduced a Clip Freeze function precisely in order to enable you to get more from your machine when running Live. Similar to other Set Live 5 going overnight, and it familiar DAW systems, Freeze generates a will analyse all the audio and 'bounced' audio file to play back processed MP3 files on your hard drive ready for instant use in the sounds off disk rather than running live via the future... CPU. Naturally the frozen processing cannot be manipulated in this state, but it is still possible to trigger clips in the usual way. This is great when you are working on a big arrangement and running out of processing power while you are still experimenting. However, I found that sometimes the frozen clips lacked the accuracy of material that was running live. Loop points sometimes seemed sluggish, and they also tended to shift slightly under moderately heavy load. One drawback of running frozen loops is that audio processors on random cycles, or cycles which are longer than the loop length — such as slow phasers, long delays or the new Beat Repeat device — lose most of their variations. There is a cure to this problem, which is to perform the arrangement in Clip view while the clips are frozen, but then carefully consolidate and bounce the arrangement one part at a time in Arrangement view. Unfortunately this method only applies to a studio situation, but I would have thought that on stage nobody would want to push the system to a point where a lot of frozen audio becomes necessary. Having said that, the way Live recovers files from the undo history after a crash works very reliably, and I haven't really lost anything major due to a crash — yet.

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Ableton Live 5

Sound Content Live 5 boasts a large new sound library, and the way Ableton have delivered it is different from previous versions, where most of the sounds came as sampled loops. Live 5 relies more on the sound-generating and processing power of the software to give the user the tools to write their own material, rather than chopping about pre-recorded sound bites. Simpler and Impulse both come with an extended library of sounds that is designed to give you musical starting points and sound-design inspiration in an open ended way. The new Device management system, which saves entire processing chains in one go, allows the user to port complex Device designs between Live Sets. The large collection of clips goes one step further and provides audio and MIDI clips together with associated Devices, so that you can treat a clip as a musical starting point by using the MIDI provided, or play the instrument that comes with the clip manually to create something new. The modular structure allows you to change effects chains that have loaded with a clip and save the result either in the Device format or as a new clip. To this the user adds his or her own variations and new material, so that almost any piece of audio sitting on your hard drive becomes a potential new sound configuration ready to use and recycle. Live 5 clips are organised into categories such as Bass, Beat, Percussive and Pad — some in MIDI and some in audio form — and there is a lot here to play with. The sound quality is of the high standard we have come to expect from Ableton libraries, and while everyone will still want to use sound sources external to Live, it certainly is possible to create entire tracks using only Live samplers, effects and Operator.

Plug-In Delay Compensation Two further new features in Live 5 help to improve overall timing. Automatic plugin delay compensation is provided for all Live plug-ins at both the track insert and return points, and a per-track time shift-delay can be found under each fader, as is the case on a number of digital mixing consoles and other DAWs. This is there to enable you to deal manually with other constant delay issues, such as delays incurred by sending to a hardware processor. The delay compensation apparently also constantly works to minimise latency for plug-in instruments. With this feature Live is catching up with the competition where delay compensation comes (if not always reliably and not always on group returns!) as standard. Ableton say that the delay compensation works reliably for third-party plug-ins that report the delay time correctly to Live, so it might still be a good idea to check for any unwanted timing errors on more obscure effects. Here the manual delay setting below the track faders will do the job. Incidentally, Sets created with earlier versions of Live will load with automatic delay compensation set to off, whereas anything from Live 5 onwards defaults to having it switched on. Another feature not unique to Live, but nevertheless useful, is the introduction of file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (5 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

arrangement locators, which can be set on the fly, named and recalled by mouse clicks, keystrokes or MIDI commands. Navigating though an arrangement this way makes it easy to mock up different structures and also helps when you are editing an arrangement in a more conventional DAW fashion. Also very welcome to a lot of users will be the addition of Mackie Control support. I'm sure that controlling Live from that family of control surfaces must be great fun, but unfortunately I didn't have one at my disposal to try it.

Pocket Calculator A few months back Ableton released Operator, an FM synth following in the Yamaha DX tradition, with four oscillators that can be combined into 10 different algorithms, from all serial to all parallel. Live 5 comes with a demo version, but for full functionality you will have to purchase a licence for 129 Euros. Frequency modulation (FM) synthesis is based on the fact that The signal flows from top to bottom in each you can generate harmonic Operator algorithm, so the envelopes of the sidebands by modulating the lowest oscillator in a serial configuration frequency of one oscillator (carrier) shape the overall amplitude and filter, while with that of another (modulator). those above determine the harmonic content Each time this happens the resulting of the sound. waveform becomes more complex, and thus more rich in harmonic overtones. This way it is possible to generate very complex waveforms from simple sources. The resulting, harmonically rich waves then get filtered in the traditional way in order to shape the sound. Ableton have managed to package this potentially challenging synthesis concept into a very simple and ergonomically friendly design. Every parameter is accessed from the front panel, and once you have understood the basic concept, it is easy to get right in there and create sounds from scratch as well as tweak the patches from the impressive factory library. Each oscillator is accessed on the left-hand side of the control panel where the pitch is set. 'Fixed' pitch can be used to add a 'clicky' percussive attack to sounds such as organs or a harmonically fixed overtone component for breathlike noises. When you click on an oscillator, the central panel switches to show the corresponding envelope. The envelope shape provides three rate parameters — attack, decay and release — and three levels, initial, peak and sustain. You can also set the oscillator's wave shape from a choice of 26, including noise, and a few other parameters such as phase, velocity sensitivity, velocity to envelope rate and key tracking are available. A neat feature, which turns Operator into an instant groove machine not unlike NI's Absynth, is the envelope loop function which repeats the envelope shape in a variety of ways, either beat- or temposync'ed or free-running. file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (6 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

Operator has seven envelopes in all, controlling the amlitude of each oscillator, the mod depth of each LFO, and overall pitch and filter frequency. All of these modulation sources make Operator a very A demo version of the Operator synth is flexible synth that spans the sound included with Live 5. spectrum from the traditional FM domain such as electric pianos, bells, brass and synth strings, via electronic percussion and drums, to modern rhythmical sound beds — instant dance tracks in fact. The library is impressive, and the hands-on design of the user interface makes it easily tweakable so that users will very quickly make it their own. The low load Operator puts onto the CPU make it an ideal sound source for Live 5, and combined with all Live's soundmanipulation plug-ins, it becomes a powerful sound design resource. But keep an eye on the voice count...

Instruments & Other Devices A number of improvements and new devices have been introduced to Live 5's armoury of instruments and effects. Since we reviewed Live 4 Albleton have launched Operator, a fully fledged FM synthesizer (see box). This instrument is not part of the Live package and must be purchased separately. However, all other Live plug-ins come free with the software, and in this release Ableton have added the intriguing Beat Repeat. Beat Repeat is, in essence, a delay unit with subsequent pitch transposer and filter, and an input feed that can be triggered in a variety of ways, but basically in sync with the track. If, for argument's sake, you feed a drum loop into it, it will pick individual beats or sections and repeat them — if you wish, with a pitchshifted feedback loop and subsequent filter. The resulting delay output can then be either mixed in together with the original signal in a dry/wet balance, used to replace the original in those sections it triggers, or heard on its own. Depending on the settings, you can achieve a variety of interesting effects, ranging from subtle rearrangements of a drum loop to mad harmonised effects not unlike those achieved with the MXR Pitch Transposer way back in the '80s. This is also very useful in the mash-up department, where whole tracks can be re-funked (try Cameo's 'Word Up'). Other remix applications would include vocal tracks, where you can achieve strange repeat and stutter phenomena if you're sick of the AutoTune yodel and want to try something a little different. Altogether, Beat Repeat is a very versatile and fun plug-in that yields many surprises. Other new plug-ins include a phaser and a flanger. I always find phase-related effects hard to judge in absolute terms, since they so depend on the source material. These two plug-ins apparently have been modelled on '70s guitar pedals (although Ableton are a little non-specific on what pedals these are), do a good job from the subtle to the not-so-subtle, and certainly enhance the sounddesign capabilities of Live. In an environment that is structured in such a modular fashion, phasing effects can always add colour to long processing chains. Nice. file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (7 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

Also new is a true stereo panner that provides you with four modulation waveforms (sine, triangle, sawtooth and random) that can be shaped towards a square wave in percentage degrees. The modulation of left and right signal can be in phase, which results in a tremolo to gating effect depending on depth, or phase-shifted by up to 360 degrees, with 180 degrees giving the maximum left/ right depth. There is also a 'spin' mode, which gives the left and right signal different rates of panning, with the maximum 50 percent being approximately a 3/4 ratio between the two. As usual better tried out than explained, this is a versatile plug-in to add interest and movement to your mixes. Designed to add warmth and fuzz, the new Saturator adds warmth and fuzz. Personally I find that kind of thing is best done in the analogue domain, but set the parameters carefully and make use of the Colour section and you can really add character to sounds, in a way that beats adding excessive EQ. Presets like 'Mid-Range Phattener' show the subtle side of this plug-in.

See You Later, Arpeggiator By public demand Ableton have also added an arpeggiator, and as you would expect by now, it can do more than your standard up and down patterns. The Live 5 arpeggiator comes with a list of 18 play orders (Styles), from the usual up/ down to chord triggers, converging and diverging arpeggios and more. Rate and gate settings are self-evident, but the Retrigger section is a little more interesting. Here you can limit the number of repeats the arpeggio runs New effect Devices in Live 5: from left, Beat for each trigger (from 1 to 16 and Repeat, Saturator, Flanger and Phaser. infinite) and set it up to retrigger either on a regular rhythmical sub-division, per note or not at all. You can achieve interesting pseudo-polyrhythmic effects by setting an odd number of steps like 7 or 9 and retriggering the result on a halfnote or whole-bar base. This way the arpeggio gets 'pulled back' into the overall rhythm while running on an unusual cycle. An unusual and interesting addition is the Transpose section, which depending on your setting adds notes of a predetermined interval into the arpeggio. These notes can be forced into a scale for obvious harmonic reasons, or set to run free at the fixed interval. The Steps fader in this section controls how many of these intervals get added to the original trigger. It sounds more complicated than it is, and you don't need to know exactly what's going on to create a nice-sounding sequence. Last but not least there is the Velocity section, where you can set a target velocity you want the arpeggio to reach over time. The arpeggiator will 'fade' from your trigger velocity up or down to the target velocity over the length of the file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (8 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

'decay' time (0.01ms to 60s). When the Retrigger mode is enabled in this section the velocity behaviour will follow the trigger setting selected in the Retrigger section of the Style settings. Using, for example, a synth sound with velocitysensitive filter settings, you can create dramatic slow build-ups with a low trigger velocity and a high destination setting over a long decay time. Alternatively, you can generate looped effects with the retrigger mode set to Beat. I could go into more detail, but suffice it to say that there are hours of fun to be had with this little beast. Ableton have managed to design this MIDI effect with relatively few parameters that all have a dramatic effect on the result, so when you tweak a control here you do hear the difference immediately.

Simpler Updated With the inclusion of independent envelopes for filter, pitch and amplitude, the Simpler sampler is still simple (just) but a little more powerful. Glide control and an improved LFO (the rate can be related to key position — build you own Theremin from a sine wave!) are welcome additions. This instrument needs to be seen in context with the other processing power of the software, since in combination with all the other plug-ins it becomes a surprisingly powerful sounddesign building block.

Organisational Tools Apart from all the previously described improvements and additions, the biggest news in my opinion is the rethink the Ableton developers have had on the filing structure for Live. I love the way everything is integrated now via the new Live Clip format. As mentioned above, saving a clip now means that all the relevant settings for the sound generator, any MIDI processing and audio plug-ins get carried with the clip file, regardless whether it happens to be an audio file, sampler or synth. Also, whole Live Sets now appear in the browser as folders containing all the clips they are made up of. So if you remember a track you have worked on and want to reproduce a certain sound or setting from that in a totally unrelated project, it is as easy as browsing to the old track and dragging the relevant clip into the new Live 5 Set. The beauty is that this works for previous versions of Live, so anything you have done up until now is available to you without any re-saving or other complication. I also like the fact that this is the same for audio or MIDI clips, which means that you don't have to set up different housekeeping systems and can just think of your sounds as sounds. Since this means that you are all of a sudden dealing with a lot more sounds (keep in mind the potentially large MP3 library you might have included) Live 5 also comes with an improved browser. All saving and loading of clips and Devices from within a Live Set can be done by drag and drop or simple save buttons, with Live creating the necessary tracks automatically on load. It is also file:///F|/SoS/SoS%2011-2005/abletonlive5.htm (9 of 11)10/19/2005 9:40:53 PM

Ableton Live 5

possible to drag entire Live Sets into an already open Set, which merges the two. One suggestion I would like implemented is to include the drag-and-drop facility for the Replace/Locate button in the clip display, since it can be a little tedious to have to browse from the root directory all the way to a sample location that might be open in the browser already. What this way of archiving sounds and ideas has that really sets it apart from any other file-management system I've come across is the way it so totally integrates with the operating system. Suddenly all your hard disks become library material — everything you work on can be combined with The new Arpeggiator MIDI anything else seamlessly, regardless of whether it is Device. in the audio, MIDI/sound-generating or mixprocessing domains. The integration of MP3 files with all my samples and sound processors even awoke my curiosity in the process of DJ mixing. Altogether the new browsing and filing system is a huge step forward, making Live 5 even more immediate and incredibly quick for accessing all your sounds. I should also mention Device Groups at this point; in essence, Live allows you to combine a number of plug-ins into a group, which can then be loaded and saved as one object. For example, suppose you have a chain consisting of an arpeggiator and a MIDI scale device plugged into Operator, through a ping-pong delay and compressor, to give you a patch that can be used as an instant trance generator. By shift-selecting all these plug-ins together and choosing the Group command from the Edit menu you can create a Device Group. Saving this just takes a mouse-click on the Save button of the Device Group, and once the Device then shows in the browser window you can rename it. What takes a little getting used to is that Ableton have opted for this format to save the settings as well as the instrument architecture, meaning that you cannot have multiple presets for a single Device Group — the Device Group itself is the preset. Once I got my head around that concept, though, I began to appreciate the simplicity: you never have worry about remembering how you created a sound again, just what it sounds like and where you archived it. Naming and archiving therefore becomes a key to a good library, and it is worth spending a little time on setting up a folder structure that will make it easy for you to manage your sounds. The fact that the browser is now searchable from within Live 5 helps a lot, and you can also create folders without having to leave the software. It just remains to say that in all of this I didn't experience any unpleasant crashes, and although it is quite obvious that a powerful audio program like Live 5 will suck up any processing power you allow it to, I was surprised by its moderate CPU consumption and reliability on my trusted G4 workhorse. Nevertheless, Live 5 is so much fun that it still made me crave a more powerful machine — and finally convinced me to get one! My main problem in writing this review was

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Ableton Live 5

simply to get to the point where I stopped playing with Live 5 and started writing the review. Several nights in a row I sat down to finally put pen to paper, only to end up playing and exploring until the early hours yet again, and it took serious strength of will to close it down and open my word processor! Published in SOS November 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

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Apple Soundtrack Pro

In this article:

Apple Soundtrack Pro

The Interface Say Hello, Waveform Editor Audio Sequencer & Editor Published in SOS November 2005 Mixing And Matching Minor Gripes Print article : Close window

Apple Soundtrack Pro £199

[Mac OS X]

Reviews : Software

pros Adds significant audio capability when working with Final Cut. Largely intuitive to use. Has access to some great Logic effects as well as introducing some new processes of its own. Very affordable. Includes a very powerful waveform editor and some great workflow tools, such as the Actions list.

Apple's loop-sequencing application has grown up, with the addition of sophisticated recording, editing and mixing facilities, a powerful waveform editor, and many of Logic's most sought-after effects. Paul White

The marketing copy for Apple's original Soundtrack claimed that it enabled anyone working in picture to build cons Being restricted to recording professional-quality audio soundtracks by combining the thousands of readyonly one track at a time precludes some music made, royalty-free Apple Loops that are applications for which the included with the program. Apple program might otherwise be describe the program as 'A royalty-free very well suited. orchestra at your command', but what it summary really offers is a royalty-free orchestra If you were to judge plus a royalty-free composer, both of whom will work for no wages, and I'm sure Soundtrack Pro as an audio sequencer, it could be seen to I'm not alone in feeling a little uncomfortable with this 'even the family pet can play it!' approach. To be fair, you can also record and import 'real' audio in the be lacking in certain key areas, but you have to form of AIFF or WAV files as well as bringing in Acid loops, but what it seems to remember that it is really a be saying to the musician busy working on TV projects is that we can cut out the picture soundtrack assembly middle man — and the middle man is you! tool and the fact that it gets so close to offering what the audio side of so many major sequencing programs offer is a testimony to its sophistication. Moreover, Apple don't force this sophistication on you — you can work at your own level of complexity without unwanted features getting in the way.

information

The new Soundtrack Pro, however, is an entirely different thermal inducing device full of aquatic lifeforms of the subphylum Vertebrata. Indeed, it is so much more sophisticated than the original Soundtrack that it has little in common other than the name. Yes, you can still do intuitive loop-based music compilation, but now you can also edit stereo and multitrack audio to single-sample accuracy, mix, use third-party plug-in AU effects and restore audio corrupted with clicks, hiss or hum. You can bring in movies and sync them to your music, while the time-bending tools built into the program let you change the tempo of the music to fit the video or sync to Final Cut 5's Scoring Markers.

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Apple Soundtrack Pro

£199; Final Cut Studio Suite £899. Prices include VAT. www.apple.com

As you'd expect, Soundtrack Pro is specifically designed to work alongside Final Cut Pro and other Apple software to facilitate the transferring of video and audio files between applications, and the multitrack capabilities of the program are ideally suited to layering up and mixing music, dialogue and effects for use in Final Cut Pro. If you think of the program as combining elements of programs like Logic, BIAS Peak LE and Garage Band in a way that is familiar to Final Cut users, you won't be far off the mark, though it also includes some very advanced workflow tools that you won't find in many other audio programs. You can even use Soundtrack Pro's waveform editor section as a stand-alone waveform editor for use with applications like Logic, Live or Reason. However, there's no MIDI or software instrument part to the program; Soundtrack Pro is more about editing audio and sync'ing it to picture. Soundtrack Pro is available as a stand-alone application but also comes as part of the Final Cut Studio Suite along with Final Cut Pro 5, DVD Studio Pro and Motion 2.

The Interface Essentially, Soundtrack Pro works in two distinct modes, one for handling multitrack projects and one for waveform editing (see box). In multitrack projects, you can record new audio and add or arrange audio clips (including Apple Loops) on a timeline, much as you can in Logic or Garage Band. The user interface is more Final Cut than Logic, with the same menus appearing in all windows and a toolbar at the top of the main window. However, the program does make extensive use of tabbed windows to jump between different project views and controls so you can access a lot of information without cluttering up your screen. Clicking a tab makes it active and brings it to the front, and the workspace can be customised to suit the project you are working: windows can be moved and resized, and the tabs can be reordered by dragging them. Similarly, you can decide which tools should be visible in the toolbar at the top of the screen. There's also a counterpart to Logic's screen sets feature. Soundtrack Pro can save and recall preset window layouts, which are named and then saved in the Layouts submenu from where they can be recalled at any time. Project controls are used to define the project's time signature, tempo, musical key, sample rate and time format. The musical key data controls the transposition of Apple Loops; non-looping (one-shot) files retain their length and pitch. To help prevent visual congestion, you can choose whether or not the video track, audio tracks, busses, outputs and master envelopes are visible. There's also a miniature global timeline view that provides an overview of the whole project and allows you to jump to different parts of the project. Fairly conventional transport controls start and stop playback or recording and set the position of the play cursor, and you can also set up a loop cycle when required. Soundtrack Pro also supports audio scrubbing, both in multitrack projects and in the waveform editor.

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Apple Soundtrack Pro

The Timeline itself is arranged into three horizontal sections for tracks, busses and outputs, where each track, buss and output has controls in its header at the left of the screen. Several different cursor tools are available, much as they are in Logic and other sequencer programs, for selecting and splitting audio clips and so on. Recordings are initiated on the Soundtrack Pro supports third-party effects selected channel after setting up the in Audio Units format, and allows you to audio source in the Recording dialogue combine them and apply them off-line as well window, but it seems you can arm only as in real time. one track at a time for recording. The whole track header background turns red in record-ready mode, so it's always very clear when a track is armed. A looprecording feature allows you to do multiple takes, then Ctrl-click between them to select the best parts of each. Soundtrack Pro allows you to add audio files in AIFF, WAV, MP3 and AAC (except protected AAC) format and you can also bring in Quicktime audio files. Audio files with any combination of 8kHz to 192kHz sample rate and 8- to 32-bit resolution can be imported. Though you can use the internal audio capabilities of your Mac for monitoring, the best quality and greatest flexibility is afforded by using an external interface box compatible with Mac OS 10.3 or above. If you want to record into Soundtrack Pro, an interface with a low-latency driver is recommended for obvious reasons. Soundtrack Pro supports a maximum sample rate of 96kHz at 24-bit for working, though it can import and convert all the formats listed earlier. Though Soundtrack Pro isn't designed to work with MIDI instruments, it does support MIDI sync via MIDI Clock or MTC at its MIDI input. Dropping a Quicktime movie onto the Quicktime video window or into the Video timeline places the audio component onto the first audio track, in stereo where the source is stereo. The header frame of the video is also shown in the Video track, but you can't string multiple video clips together here — even though the interface looks a little like Final Cut, it's for editing audio not video! A slider controls the video playback position directly, or you can type a time value into the value slider. When moving or resizing audio clips alongside video, you probably want them to synchronise with the video elements of the project, while when editing audio, you tend to do so by beats and bars. Here, you can set each track to be time-based or tempo-based as required, and Soundtrack Pro includes a Snap feature so that edits, markers and moves lock to the nearest Snap To position. Even envelopes and crossfades lock to Snap.

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Apple Soundtrack Pro

Say Hello, Waveform Editor Soundtrack Pro's Waveform Edit window can be opened by double-clicking in an audio region, or you can open audio files directly in the waveform editor without going through the multitrack view. Audio can be viewed and edited down to the individual sample level, either destructively or non-destructively. There are tools that allow audio files to be analysed for common problems — especially prevalent in video production — such as clicks, pops, hum and noise, and all the other common functions such as normalisation, fades and gain changes are also supported. At sample level, you can also redraw the audio data to get rid of digital spikes and other short-duration artifacts. Zoom tools allow you to home in on sections of interest, and interestingly, you can also toggle between a waveform view and an audio spectrum view. The waveform display also animates to enable you to visualise the results of your processing. Actions and Analysis buttons show either the Actions list or the Analysis Type with relevant parameters and analysis result listings. The Analysis functions include the ability to check for clipped audio, provided the file has not subsequently been reduced in level — the audio has to hit the end stops to be recognised as clipping. Some analysis types have parameters, such as threshold, that need to be adjusted prior to analysing the file. When Actions is active, you can add real-time effects via the Effects tab. One feature that will make Logic users jealous is that you can also 'flatten' the results to make them permanent, without having to bounce and re-import a new file. Any actions performed on the audio, such as normalising, are listed in the Actions window with a tick box alongside each action. Unchecking the box reverses that action up until the point where you decide to flatten the file and make your edits permanent. This is a particularly elegant feature and goes far beyond being a simple undo-redo history. For example, you can reorder actions by clicking and dragging, and you can undo an single action you did several steps ago without having to redo lots of later ones. Additionally, Actions can be temporarily bypassed rather than undone, so this feature really does make for a flexible workflow. What's more, you can save Action sequences as Apple Script droplets to facilitate drag-and-drop batch processing. Action droplet settings may be edited directly in the Apple Script editor so it's not difficult to make small changes to a multi-stage batch process that you've set up. Soundtrack Pro also ships with a number of 'Automator Actions' that take advantage of the new Automator features in the Tiger operating system. In effect, these make it possible to automate certain audio processing tasks (sample-rate conversion, audio trimming, denoising, normalising and so on) without having to write your own Apple Scripts. While the waveform editor might not have quite so much functionality as the most advanced dedicated audio editing packages, it comes pretty close and handles all the essentials such as sample-rate and stereo-to-mono conversion, fades in and out, normalising and gain change, reversing and time-stretching. It also has a fingerprint noise-reduction system that works reasonably well providing you don't give it anything hopelessly noisy to deal with. This isn't a conventional real-time plug-in: instead, you load in a section of noise, then loop the audio while adjusting the threshold control until you get the best result. When you've tweaked to your satisfaction, you can then apply the process to the whole file or to the selected region. While not quite state-of-the-art, it's infinitely better than the denoiser in Logic.

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Apple Soundtrack Pro

Mixing And Matching Track level and pan can be adjusted in the main page using controls in the track header section where you can also mute and solo tracks, add insert effects as plug-ins and send to a reverb buss. Automation is available via a pop-up menu and the now familiar folder content 'swivel arrowheads' allow the automation relevant to any given track to be displayed directly below it, again not unlike the way it is handled within Logic. Three automation modes are supported — Latch, Touch and Read — and automation data may be input directly from a mouse or hardware controller, or edited graphically using envelopes and breakpoints. Hardware control surfaces that impement the Mackie Control protocol are supported. When you come to mix a project, there's a dedicated Mixer page where each track, buss and output has a channel strip with level meter, level control, mute and solo buttons, slots for adding effects, pan controls and so on. Each track may be given an icon selected from a list, again rather as you can in Logic, though there is a great choice of colourful icons here that would make most Logic users jealous! A pop-up menu lets you select an output for the track or buss as well as letting you pick multiple physical output channels for the output audio channels. Conceptually, this isn't dissimilar to the Logic Track Mixer, though it looks a little simpler. Finished projects may be exported as audio files or as a Quicktime movie. There's also a Media and Effects Manager, which comprises five tabbed areas and is designed to allow you to find and preview audio, video and effects. A Browser window lets you trawl through your hard drives and folders in the usual way, and favourite items can be stored under the Favourites tab. The Search tab lets you locate audio files filtered by criteria and you can also search for specific text or Apple Loop tagging descriptors. Candidates may then be auditioned from within the Search Results list. The Bin tab contains media files added to the project and you can drag files directly from the Bin onto the timeline. Files not currently added to the project are dimmed, whereas off-line files appear with red text. Using an Effects tab, you can add plug-in effects and configure effects sends for the project and also call up information on effects already instantiated on tracks, busses, outputs or audio files. Soundtrack Pro now includes over 50 plug-in effects from the Apple range, including high-end reverb, delay, EQ and chorus plus more dramatic soundmangling processors like Sub Bass, Bitcrusher and Auto Filter. Many of the effects are ported directly from Logic and they still have their distinctive blue graphical interfaces. Even Logic's high-end Space Designer reverb is available here. As with Logic, third-party effects are supported in Audio Units format only — there's no VST or RTAS support.

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Apple Soundtrack Pro

I did encounter a couple of minor problems with Soundtrack Pro. For one thing, it seemed like whenever I did anything at all on screen, even adding track icons, the G5 processor fans revved up to take the strain. There were no performance issues associated with this, just a lot of huffing and puffing from the fans as though I was asking the CPU to do something arduous. Even dragging windows around the screen started it off, and it was most disconcerting. The program also seemed a little halfhearted about accepting my Logic Control as a hardware controller and 'unexpectedly quit' twice while I was setting this up. It seemed to recognise that a controller was connected but I had to tell it that it was a Logic Control, not a Mackie Control, and also tell it to which MIDI ports it was connected, even though it should have been able The Waveform Editor's Analysis functions to work that out for itself. After that it include a clipping detector. worked fine with no further crashes, though some functions don't seem to be supported from Logic Control, such as horizontal display zooming. In fact most of the Logic Control lights go out, leaving you with level and pan control, Select, Solo and Mute on the channels plus overall transport control with cursor jogging but no scrubbing. Track names are correctly displayed, so if you already have a Logic Control or a Mackie Control, it's well worth using it to drive Soundtrack Pro, but I wouldn't recommend buying one specifically for the job as many of the more sophisticated functions are unavailable. Logic users should also note that Soundtrack Pro's interface doesn't always work in the way you're familiar with, and I suspect this comes from making the program fit in with Final Cut working methodology. It may take a while longer before the best of both approaches becomes the new Apple standard. For example, in Logic you can double-click the stop button to return to the start of the file, but here there's a separate 'go to start' button; and to add points to an automation envelope, you have to double-click to create a new envelope point, and so on. Overall, though, this new Pro version is way more flexible and sophisticated than the original Soundtrack, and as such it offers far more than just a simple way to put media composers out of business! It comes loaded with a great set of movie sound effects as well as musical loops and can now handle serious multitrack mixing with the benefit of all Logic's best effects, including the excellent Space Designer. As a music recording program, Soundtrack Pro is very powerful providing you don't need to record MIDI parts, and gives the user access to effects that once cost more than the entire asking price for the program. It is pretty easy to learn, it offers practical waveform editing in a familiar way and it has full mix automation file:///F|/SoS/SoS%2011-2005/soundtrackpro.htm (6 of 7)10/19/2005 9:41:01 PM

Apple Soundtrack Pro

capability. I like the way you can configure the Workspace to suit your own style of working and the ability to create the equivalent of Logic's screen sets makes the program easy to navigate, even if you have only a single monitor. Realistically, you can do just about anything you might reasonably wish to do in a multitrack audio recording and mixing environment except recording multiple tracks at once, so if Logic seems heavy going and Garage Band is too simplistic, Soundtrack Pro might also be a good choice for musicians working on their own who don't work with MIDI instruments. Its real strength, though, is in bringing advanced audio mixing and editing into the sound for picture world. Published in SOS November 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

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Cakewalk Sonar Home Studio

In this article:

Piece Of Cake System Requirements The Pursuit Of XLence More Effective Better & Better

Cakewalk Sonar Home Studio MIDI + Audio Sequencer [Windows] Published in SOS November 2005 Print article : Close window

Reviews : Software

Cakewalk Sonar Home Studio £79 pros Inexpensive for a fully specified DAW. Easy to learn, easy to use. Stable. The XL version provides even more bang for the buck.

cons

With Sonar Home Studio, Cakewalk have made the core features of their flagship sequencing application available at a bargain price. Alan Tubbs

No Freeze function. Help could use a little help.

For the last couple of years, users of Cakewalk's entry-level Home Studio summary package have felt like second-class Sonar Home Studio 4 citizens, sitting at the back of the sonic provides outstanding value for bus with a Sonar 2 sound engine and a an entry-level professional screen stuck in the 1990s. It wasn't DAW. It doesn't skimp on the bad, but it was beginning to show its basics, and most of the missing features from the age. I used it mostly for MIDI work, higher-priced versions are recording external synths and functions most of us won't use rendering soft synths before mixing the often, if at all. audio in Vegas, which I was more information comfortable with. The new Sonar Home Studio now shares the cool grey look $ £79; XL version £109; Home Studio (SHS) remedies both of Sonar. The Track Inspector on the left upgrade from previous gives an in-depth look at the focused track, those complaints, using the Sonar 4 versions £39. while the Docking view (bottom right) sound engine and a clean blue/grey Edirol Europe +44 (0)20 expands over the entire right-hand side of user interface. There are other 8747 5949. the screen for audio and MIDI editing on a improvements and goodies, of course, +44 (0)20 8747 5948. large scale. Floating on top is an abbreviated but for musicians looking for a first look at the XL version's RXP REX player, www.edirol.co.uk loaded with a pre-sliced guitar file. DAW and those thinking of upgrading www.cakewalk.com or switching from a competitor's product, SHS streamlines Cakewalk's extensive (and sometimes confusing) Test Spec product line and offers a entry-level pro sequencer and audio recording package PC with 1700GHz AMD with support for sample rates up to 192kHz and the ability to use multiple Athlon CPU and 1GB RAM, processors, dual cores and Hyperthreading. The only major things missing are running Windows XP, with the high-end effects and synths, which can be added at one's leisure, and the Presonus Firepod and Mability to use more than 64 audio tracks. Audio Transit soundcards. Tested with Cakewalk

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Cakewalk Sonar Home Studio

Project 5 v2 plus various plugins.

Piece Of Cake SHS ships with a serial number, just like all other Cakewalk software, but unlike HS, it has to be registered on-line, too. However, this only took a minute to do and will hopefully forestall some of the hackers out there — it still beats the devil out of a dongle. Cakewalk's migration tool transfers settings from previous editions over to the new software, and all the songs I was working on in HS2 opened fine in SHS. The new program found both my audio interfaces, along with their attendant ins and outs, without difficulty. The new Sonar look of Home Studio is not only aesthetically pleasing, it is also easier to navigate, and if you find the the default colour scheme too muted, it can be altered. HS was very busy-looking, with several rows of Windows-style toolbars running along the top, and another above the track controls. The whole toolbar menu format is gone in SHS, to be replaced by a Master Control bar containing the play/record buttons, clock and most-used buttons from HS's toolbars. There are still track toolbars at the top of the tracks, but Cakewalk have also added a navigator button which opens up an overview at the top of the track headers and allows the mouse to drag the track view through your entire project. Another ergonomic addition is the Track Inspector. This gives more room for channel settings than previous versions of HS, where all the information System Requirements was squeezed into the track headers themselves. Windows 2000 or XP. There are other changes too: for multiple track 1 GHz or better processor. settings, the Console View is now only a mouseclick away in the Master Control bar, where it is 100MB hard disk space. easier to find than in HS's taskbar. The virtual 256MB RAM. mixer in the Console View gives all channels a Track Inspector-sized scribble strip, making it easy to tell at a glance what channel effects are inserted in the FX slots, as well as which tracks are sent to any auxiliary send. Unlike HS, SHS lets you add busses and assign them as needed: they show up in the centre Master section of the Console View. Once audio is routed to any buss, the buss can be used as an FX sub by inserting a track effect into it and routing back to the master, or into a sub out (hardware dependent, of course) for external processing, or into yet another buss. Every audio track (including busses) has a send section, and only the busses you assign a track to show up there, which helps keep things uncluttered if you have a lot of busing going on. Scribble strips allow you to name everything, which also helps to keep track of which is doing what to whom in a complex mix. The Master Control bar is not resizable, but it can be undocked and moved over the main screen or to a second monitor, if you like. And, of course, it always remains on top of the screen.

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Cakewalk Sonar Home Studio

These big controls are an improvement over the smaller taskbar buttons, and Cakewalk have made great strides in making SHS more of a single-screen environment. HS made use of deployable, resizable pages for editing audio and Acidised 'groove clips', MIDI info by piano roll, staff or event, and for adding synths or even lyrics to a project. These pages are now part of a docked and tabbed view which occupies the bottom right of the main screen, opposite the buss pane, in an otherwise empty area. The tabs for the Docking View don't appear until you have right-clicked on some audio or MIDI for editing, or have inserted a Staff, Event, synth or Lyric via the Windows View command. The tabbed views are too small for any real editing, as they occupy only the quadrant of the screen beside the busses, but a minimise/maximise button expands the edit view to the entire right side of the screen, replacing the track audio/MIDI content. This is usually enough room to edit without resizing the track headers themselves. I find this to be a more ergonomic layout than finding various pages scattered over two screens: if you need to edit some MIDI, right-click on them, Piano Roll jump to the top of the tabs, hit maximise, and you're ready to edit. Used in conjunction with the Navigator, it is easy as eating pie to edit song-length swathes of MIDI. Some of the changes from HS to SHS may force you to alter the way you work, though if you prefer the old ways, it is possible to right-click on individual Tab Views and disable each separately, returning the tab to a floating page. But the new look and feel are so easy to use I haven't gone back to the HS way of working.

The Pursuit Of XLence As well as the Reverb XL plug-in described in the main text, the XL version of Sonar Home Studio adds three extra synths to the mix. Dyad is a second Soundfont synth, but has considerably more control over the sound than the basic SFZ. You can load in two sounds and edit their pitch, filter and amplitude envelopes, filter cutoff, resonance and slope, and assign two LFOs. There is also RGC Audio's Square 1 virtual analogue synth, which is not as flexible as some of the other VA synths out there, but sounds excellent, with three oscillators and a good filter. Finally, there is the RXP1 REX file player, which can play back REX files as whole loops or slices. RXP will also load in WAV and other formats, but REX files are where the fun is. I had the lyrics for a blues song, but no music ideas to complete it, so I loaded up RXP with a blues guitar file that was included on the XL content disc. It was a very nasty guitar riff that I played from the keyboard, which turned my plinking into a John Lee Hooker lead. You can also use the button layout to play slices or the whole loop at different pitches. If you need any of the above items, each is worth the extra price by itself; together with Reverb XL, they represent a great deal.

More Effective Sonar Home Studio ships with the usual complement of effects: gates, limiters, compressors, parametric and graphic EQ and reverb are all there. These are workhorse effects — nothing too fancy, but they do what they should. SHS comes with Cakewalk's VST wrapper, so VST and VSTi plug-ins are available in file:///F|/SoS/SoS%2011-2005/cakewalkhomestudio.htm (3 of 5)10/19/2005 9:41:04 PM

Cakewalk Sonar Home Studio

addition to those in Cakewalk's native DX format. All my VST effects and instruments showed up and worked. If you have a large investment in some highend software, you might want to make sure it works with Sonar before buying, but there have been few complaints in the forums. Cakewalk do include one unusual effect with SHS: Powerstrip. Like the free FX Pad and their Spectral FX, this is a preset unit with different effects (or controls) assigned to sides of a X-Y control pad. You can grab and move the cursor or assign it to move in a circular pattern in time with your song. In the past, I had tended to use FX Pad rather than Spectral FX, since the effects were wilder, but Powerstrip, to my ears, sounds better than either of The Console View gives an overview for the others and has moved to the top of mixing and setting effects. The virtual mixing board is fairly straightforward. Reverb XL, my list. The SHS XL package adds included in the XL package, sounds better Reverb XL, a very tasty reverb. This than most stock reverbs, while Powerstrip doesn't have a natural sound like a can add motion to an effect. convolution reverb, but is more like a good modelling rackmount unit. It is a step above the stock reverbs included with most DAWs, offering a thick sound with a smooth tail. Besides effects, SHS comes with DXi synths. As well as Dreamstation, a basic polyphonic virtual analogue synth, there is a Soundfont (SF) playback synth by RGC Audio. Many people turn up their noses at Soundfonts, thinking of gaming boards, but SFZ provides an easy and cheap way to add sample playback to your project. Most SFs are pretty basic, but I used a bank I had found on the Internet to drive the middle section on one song. The bank had a bunch of effects and mechanical sounds, and I beat out an industrial rhythm from the keys, which I named 'the washerdrum'. Not bad for a free download. The other included soft synth is the Edirol Virtual Sound Canvas 1. This is the software version of Roland's once-ubiquitous Sound Canvas, a souped-up GM synth. The sounds are, for the most part, nothing to write home about, but it is good for temp tracks, or more if you find yourself in a pinch. There are the other improvements over Home Studio: Group edit now allows clips to be selected and have slip edits, fades and splits applied to each clip at the same time, nudging allows clips to be, well, nudged forward or backward by a pre-selected amount using keyboard commands, and the metronome finally allows audio clips to be used instead of MIDI notes. Other things have been left out, such as track Freeze. Sonar Home Studio allows you to put all the audio clips for a project into one folder, but Bounce to Track still doesn't allow you to name the new track, instead automatically assigning something like 'song 1, Master, Mix (2).wav'. If I'm going to have to name a file anyway, let me do it beforehand, not afterwards! And since there are only 64 audio tracks, you'll have

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Cakewalk Sonar Home Studio

to budget for the full version if you need to close-mic a full orchestra. Finally, the help doesn't include enough screenshots, and most of the included synth help files are pretty skimpy. There are on-line video tutorials, but when you need an answer while actually working, screenshots with the text search save time.

Better & Better On the SHS forum, some users back Cakewalk's claim that the Sonar 4 engine is more efficient; others claim they can't see any real improvement. I haven't noticed a great leap forward in CPU use, but I haven't had any crashes, either. With Home Studio 2 and previous HS incarnations I would sometimes get a program freeze, and I also ran into a bug whereby the MIDI icon stayed up in the taskbar after I used Alt+Del to close HS and wouldn't open correctly until I opened and immediately closed the program. It didn't happen often enough for me to track down and fix the problem, but it was still unwelcome. With Sonar Home Studio, I haven't had any crashes, freezes or any other surprises, and stability matters to me much more than any improvement in efficiency. Published in SOS November 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

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Crane Song Avocet

In this article:

Digital Interfacing The Remote Controller Signal Selection & Conditioning Input Gain Trim Options Using The Avocet

Crane Song Avocet Stereo Monitor Controller Published in SOS November 2005 Print article : Close window

Reviews : Accessory

Crane Song Avocet £1821

This new monitor controller sets new standards for fidelity.

pros

Exemplary sound thanks to operational flexibility and sonic discrete Class-A electronics and passive attenuators. Stunningly good internal DHugh Robjohns A converter. Extremely flexible facilities for console integration and This new high-end monitor controller external control. has been designed by Crane Song's Superb input-level offset mastermind, Dave Hill — someone mode. with a reputation for fastidious attention Independent volume for to detail and for demanding the highest main and headphone outputs, technical and sonic standards. While it dim, and talkback dim. Nice ergonomics throughout. is a widely held belief that passive monitoring controllers must be best, in Beautifully constructed.

cons You can't control the mono monitoring mode remotely. There are no individual speaker mutes. The truncated 16-bit mode is pointless.

summary An expensive but highly accomplished stereo monitoring controller with comprehensive facilities for stand-alone operation or full integration with a console. Exemplary analogue circuitry is complemented with a superb D-A stage, and controlled via an extremely ergonomic remote panel.

information

practice the inputs and outputs need buffering to remain reliable and consistent in the real world of unknown Photos: Mark Ewing impedances. Furthermore, it is extremely hard to derive mono from a stereo source passively without affecting the sound, so some electronics are inevitable in a versatile high-quality monitoring controller. However, these electronics clearly define the ultimate performance of the monitoring chain, so the Avocet's audio path employs just three all-discrete Class-A buffer amplifiers in each channel. Level control is another difficult area to address. Ganged potentiometers often track poorly and can produce image shifts, while active circuitry such as voltagecontrolled amplifiers, although affording very flexible control options, often introduce unacceptable levels of distortion. The ideal solution is usually passively switched attenuators, often arranged around an expensive rotary switch. However, the Avocet manages all of its level-control functions (and I/O selections) with sealed, high-quality relays and resistor chains. Clever logic control of these relays also allows some quite sophisticated additional features — regaining much of the flexibility of a VCA design without sacrificing audio purity.

£1821.25 including VAT. KMR Audio +44 (0)20 file:///F|/SoS/SoS%2011-2005/cranesongavocet.htm (1 of 7)10/19/2005 9:41:07 PM

Crane Song Avocet

8445 2446. +44 (0)20 8369 5529. Click here to email www.kmraudio.com www.cranesong.com

The bulk of the Avocet's electronics are housed in a 2U rackmounting case, which is controlled remotely from a desktop unit connected with a 25-foot cable. The front panel of the rack unit features only a stereo headphone socket and a large green vintage-style power lamp, while the rear panel is crammed full of XLRs. Inside the mainframe is a large circuit board covering the whole of the available floor area, with a long row of relays for the stepped volume attenuator across the front. Mounted above this all-analogue board is a smaller digital board, the role of which is described the 'Digital Interfacing' box. The Avocet is equipped with three stereo analogue inputs referred to as Analog 1, Analog 2, and Aux — all nominal +4dBu balanced XLRs with multi-turn trimmers to fine-tune the levels over a ±8dB range — and there are also three digital inputs feeding a high-grade D-A stage. Apparently, an internal link in the current Revision 4 version of the unit (the review model had older Revision 2 circuit boards) optionally provides the Analog 2 input with 14dB of additional gain to accommodate a nominal -10dBV semi-pro input, which is a useful facility. The rear panel also carries three sets of +4dBu balanced stereo analogue outputs on XLRs, a second stereo quarter-inch headphone socket, an IEC mains inlet (with integral fuse holder), and two D-Sub connectors. One of these (Remote) is used to hook up the remote control panel, while the other (Accessory) provides various auxiliary functions such as external switching of talkback, solo, dim, and mute, as well as generating buffered outputs for the talkback mic, headphone output, and unbalanced mono and stereo outputs (the last being intended for an external metering system). There is also provision here for an additional remote-control signal which will be used in future for linking Avocet units when monitoring in surround. Although missing from the review model, the Avocet is normally supplied with adjustable in-line XLR attenuators for use between the Avocet's outputs and the inputs to power amplifiers or active loudspeakers. They are intended to address the perennial problem that most amplifiers are far too sensitive for direct connection to nominal +4dBu line-level outputs. Rather than compromise the signal-noise performance of the Avocet's electronics, Crane Song supply adjustable attenuators which can be placed at the inputs of active speakers or power amps to reduce the signal level by up to 30dB.

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Crane Song Avocet

Digital Interfacing The Avocet is equipped with three 24-bit digital inputs. Two are straightforward AES inputs able to accommodate a full range of sample rates from 32kHz to 192kHz, and labelled Digi 2 and Digi 3. The third is referred to as the DAW input, and this has three input modes selected by a switch on the back panel. One accepts a normal 'single-wire' AES input via an XLR, while another accommodates a conventional coaxial S/PDIF input via a phono socket. The third mode invokes a second XLR socket to partner the first so that dual-wire highsample-rate AES signals can be connected. The digital inputs do not need to be synchronous — the internal DAC simply locks to the selected source — and changing digital sources is fast, without any nasty bangs or crashes. Interestingly, the Avocet's D-A design employs a similar strategy to that used in the Benchmark DAC1 (which I reviewed back in SOS July 2005), oversampling the selected input to 192kHz and passing it through a sample-rate converter to help remove jitter before D-A conversion clocked by a local crystal. Internal links allow the oversampling to be disabled or limited to 96kHz, if required, although the designer claims the 192kHz option sounds best. The selected digital input is received through a pair of Cirrus Logic CS8416 chips, before being routed to a Burr Brown SRC4492 sample rate converter, and then on to a Cirrus Logic CS4398 — the company's latest 192kHz flagship device. A local crystal is used to clock the output of the SRC and the D-A chip independently of the incoming digital signal's embedded clock. Among the impressive specifications and features of the D-A chip is the choice of two oversampling interpolation filter designs. This function has been made available through another link on the digital board, with the default being a 'slow roll-off' setting. A 'fast rolloff ' mode is also available, and may be more appropriate if source oversampling is not employed. One rather odd function of the D-A board is operated by a button on the Avocet remote control labelled 16-bit. This truncates the selected digital input to 16 bits and is apparently 'for checking what 16 bits sounds like'. I can't imagine any situation where one might want to do this — at least not without applying proper dithering as part of the process. Not a button that will see much use, I suspect!

The Remote Controller The remote controller is a chunky metal 'brick', roughly 45 x 200 x 190mm (hwd), the rear panel of which carries just a D-Sub connector, to link with the mainframe, and a female XLR connector. The latter is for a talkback mic (there being no phantom power available), and the mic preamp's gain is adjusted via a multi-turn trimmer adjacent to the XLR. The control buttons are laid out simply and logically in two long columns, with colour coding to help navigation. In the centre of the panel is a large green rotary encoder with a detented action and 24 green LEDs around the periphery to indicate the current level setting. The knob feels great to use — just the right size and weight to allow accurate click-by-click adjustments or a quick spin up or down with a flick of the wrist, causing the relays to chatter away in the main file:///F|/SoS/SoS%2011-2005/cranesongavocet.htm (3 of 7)10/19/2005 9:41:07 PM

Crane Song Avocet

frame. The level setting is very clear, and although marked in 2dB steps the LEDs resolve to 1dB level changes simply by lighting adjacent LEDs to show the midway positions. At the top of the panel is a highresolution stereo bar-graph meter, spanning a 46dB range and calibrated with 0dB at the top — although it A small trimmer on the rear panel of the seems a little odd to have a digitalremote control allows the talkback mic's gain style meter in an analogue controller. setting to be adjusted. An internal link configures the meter to show either the level of the selected input (default), or the output level from the Avocet. The right-hand column of buttons selects the input source, talkback function, and an odd 16-bit monitoring mode. Only one input can be selected at a time, and there is no summing or mixing ability. When the headphone control mode (of which more in a moment) is active, the bottom three buttons in this column serve to configure the headphone monitoring source. There are three options: a feed of the Aux analogue input (intended to pass through a cue mix from a console, for example); a fixed-level feed of whatever input is selected to feed the main monitors; or the same main-monitor feed but taken after the mono, phase, and level controls. The talkback signal is mixed in with the headphone outputs, regardless of what source mode is selected. If the solo mode is activated via the Accessory connector, then the currently selected input is overridden by either the Analog 1 or DAW inputs, according to the position of another internal jumper link. The idea of this feature is to allow complete integration between the Avocet and a mixing console, allowing soloed channels to appear on the monitoring automatically.

Signal Selection & Conditioning The left-hand column of buttons determines the output destination (only one of the three can be active at any time) and the input signal conditioning (phase, mono, dim, and mute), and activate the headphone control mode. With the headphone mode selected the rotary encoder sets the headphone volume, which is totally independent of the main monitoring volume. The mono monitoring mode can be configured with another internal link to send the derived mono signal just to the left speaker, or to both (the default is both). The Mute and Dim buttons are conveniently located at the bottom of the column where they are easy to reach, and the dim level can be adjusted by turning the volume control while in dim mode. The setting is then remembered, even after powering the unit down. The monitoring level when the talkback button is file:///F|/SoS/SoS%2011-2005/cranesongavocet.htm (4 of 7)10/19/2005 9:41:07 PM

Crane Song Avocet

pressed can also be set independently of the normal dim level, which is a superbly useful feature. Every button has an associated LED to indicate the current status, and the buttons all have a very positive mechanical click action. There are no obvious pops or splats on the monitored signal when operating any buttons — in fact the unit even turns on and off gracefully, without any nasty thumps. When powering up, all the settings are remembered, but the Mute button is sensibly always enabled.

Input Gain Trim Options To me, one of the nicest functions is that input levels can be temporarily trimmed from the remote controller, but without messing up the precision gain structure dialled in using the rear-panel trimmers. This is essentially done by manipulating the logic data that controls the attenuator relays, so that the audio quality is not affected. The only downside is that you can't adjust the monitoring level beyond the physical attenuator range. For example, if you dial in a +10dB offset, and then crank the level up full, you are effectively asking the attenuator to provide 10dB above the maximum level, which isn't possible. In practice, though, this is very unlikely to become a concern. The gain trim mode is activated simply by pressing the selected input's button a second time, whereupon the gain can be trimmed by up to 10dB in either direction using the large green knob (the latest models provide 0.5dB increments, earlier units 1dB steps). Tap the source button a third time and the operation returns to normal, but with the new gain offset remembered — even if the power is turned off. This gain trim mode is excellent for allowing different sources to be compared accurately at the same perceived level — for example the original and compressed versions of a track.

Using The Avocet Connecting the Avocet is very intuitive and simple, as is the basic operation. Selecting sources and destinations, adjusting the monitoring level, and checking mono and phase compatibility are all very straightforward. The ability to set, and more importantly reset, the monitoring level within 1dB is excellent, and essential in a mastering environment. Likewise, the ability to set the headphone and dim levels independently, using the same control and in such an intuitive way, makes life very easy. The facilities to enable complete integration with a console are well thought out too, but don't compromise the stand-alone operation, or indeed the sound quality, in any way. There are only a couple of niggles. The inability to select mono to one or both speakers from the remote panel is disappointing. Clearly, David Hill recognises file:///F|/SoS/SoS%2011-2005/cranesongavocet.htm (5 of 7)10/19/2005 9:41:07 PM

Crane Song Avocet

the important distinction between these two monitoring modes, because the option is available via an internal link. Also disappointing is the lack of individual speaker muting. Often a problem can only be identified by listening to each channel separately. Bearing in mind the absence of separate speaker mutes, it is interesting to read in the Operator's Manual that there are plans afoot to The circuit boards feature more than a dozen enable the linking of multiple Avocets jumper connections which allow the user to to form a surround monitoring tweak the operation of the Avocet to their own personal tastes. controller. Being able to solo and mute individual channels is of paramount importance when working in surround, so it will be fascinating to see how this key functionality is provided. The most important thing about a monitoring controller, though, is how it sounds, and in this there is no doubt whatever. The Avocet sounds totally transparent and I could not detect any quality changes whatsoever when inserting the unit into my monitoring chain. The analogue signal path is exemplary in every way, demonstrated by the fact that the tonality and stereo imaging don't change at all, regardless of the volume setting. Auditioning digital sources was equally impressive, and it quickly became clear that the internal D-A stage easily matched the quality of my favourite new reference, the Benchmark DAC1. The performance of the Benchmark is truly stunning (once it has warmed up), so for the Avocet to match its resolution is praise indeed, and particularly noteworthy given that both D-As employ a very similar technological approach. The Avocet's headphone drivers use high-quality op amps and powerful LM3886 integrated power-amp chips rather than discrete Class-A circuitry, but even so I can't fault the performance at all. There is more than enough volume available, even for the most demanding of artists, and the design provides absolute clarity and neutrality, conveying accurate transients and powerful deep bass. The Avocet is as near to the perfect monitor controller as I have yet found. Yes, it lacks individual speaker muting, the mono check mode is not as versatile as I would like, and the truncated 16-bit mode is, to be blunt, utterly pointless. But the plus points far outweigh these irritations. Excellent ergonomics, stunning audio quality, precision level adjustments, console integration flexibility, and the potential of an upgrade path for surround work all make a very convincing case indeed for the Avocet. Published in SOS November 2005

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Crane Song Avocet

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

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

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Digidesign M Box 2

In this article:

On The Outside Up And Running Summing Up

Digidesign M Box 2 £335

Digidesign M Box 2 USB Recording Interface [PC/Mac] Published in SOS November 2005 Print article : Close window

Reviews : Recording Channel

pros Good sound quality, with decent preamps and a powerful headphone amp. Buss-powered, yet offers phantom power to mics. Now offers MIDI I/O, an output level control and the ability to record analogue and digital inputs simultaneously.

cons

Digidesign have given the most affordable interface in their Pro Tools range a makeover, with a new case, new preamps and some additional features. Sam Inglis

Still no option to disable input monitoring in software. Zero-latency monitoring still doesn't apply to digital inputs. Not entirely happy at low latencies in the review system. Lacks the M Box's insert points. Uninstalling Pro Tools on Windows can be a minefield.

summary If you're intending to buy a Firewire or USB 2 interface, Digidesign's revamp of the M Box is unlikely to change your mind. However, if you're content with USB 1.1, it's still one of the best options, and the improvements do make the M Box 2 superior to the original.

information £334.88 including VAT. Digidesign UK +44 (0) 1753 655999. +44 (0)1753 658501. Click here to email

In the three and a half years since the original M Box was introduced, it has become one of Digidesign's most successful products. A simple stereo-in, stereo-out USB audio interface, the M Box allowed anyone with a Mac or PC to call themselves a Pro Tools user without breaking the bank; and its clever circuitry meant that the Focusritedesigned mic preamps could derive phantom power parasitically from the host machine's USB port, making it a truly go-anywhere system. Recently, the M Box has faced increasingly stiff competition. There are plenty of affordable interfaces around which use the higher-bandwidth Firewire protocol, including those made by Digidesign's partner company M Audio, which are now compatible with Pro Tools. M Audio and other companies such as Edirol and Terratec also make USB devices that are much cheaper than the M Box. A change of plan from Digidesign seemed inevitable, and many people expected them to turn to USB 2 or Firewire to achieve a higher channel count, or perhaps just to drop the M Box altogether and extend their support for M Audio hardware to include USB interfaces such as the Fast Track Pro and Mobile Pre USB. (Or how about a PCMCIA card along the lines of Echo's Indigo IO? Please?) Instead, they've replaced the M Box with another USB 1.1 device which offers an almost identical feature set: the M Box 2.

www.digidesign.com

Test Spec

On The Outside file:///F|/SoS/SoS%2011-2005/mbox2.htm (1 of 5)10/19/2005 9:41:10 PM

Digidesign M Box 2

Pro Tools LE v6.8.1. Centrino laptop with 2.0GHz Pentium-M CPU and 2GB RAM, running Windows XP Service Pack 2. Tested with Steinberg Cubase SX v3.02.

The much-imitated look of the M Box has changed completely, although the new design is equally distinctive. The smart moulded plastic case can be stood on its end or arranged horizontally; in this case, the carrying handle attached to the front panel tilts the front of the M Box 2 up slightly, making it easier to see the settings. The handle can be swapped out with an alternative, solid piece of blue plastic if you prefer. According to the manual, this saves space and allows the unit to sit flat on a table top, but the one supplied with the review model was barely smaller than the handle, and still tilted the front panel upwards. If you really wanted to save space and lay the M Box 2 flat, you could just leave the handle off altogether, so the alternative faceplate seems pretty redundant. In terms of features, the M Box 2 has a great deal in common with its predecessor. Once again there are two analogue inputs, although since Digidesign's partnership with Focusrite is no more, the associated mic preamps are of a different design from the originals. Neutrik Combi connectors are no longer used, and the M Box's analogue insert points are also missing in action. Instead, each analogue input features separate XLR mic, quarter-inch balanced line and quarter-inch instrument DI sockets on the rear of the unit, with the controls on the front panel. These consist of a rotary gain control, a switch to engage a 20dB pad, and a button to switch the channel's source from Mic to DI. As on the original, 48V phantom power is available at both mic inputs (or neither — it is globally switched), and is derived from the host machine via the USB connection. Unlike its

On the 'if it ain't broke, don't fix it' principle, the M Box 2 also predecessor, features the same direct monitoring arrangement as its the M Box 2 allows you to predecessor. As well as being routed to the computer for recording via the A-D converters, the analogue input signals are record analogue and also sent directly to the outputs, and when you're overdubbing, digital inputs at a dial labelled Mix allows you to balance the levels of this the same time. monitor path against the previously recorded material coming back from the computer. By pressing the Mono button you can choose whether to hear the two directly monitored input signals panned hard left and right, or both panned centrally. The former is appropriate when you're recording in stereo, whilst the latter would be more natural when tracking two different mono sources. There is only one headphone socket this time around, on a front-panel quarter-inch jack. There are, however, three welcome improvements compared to the old M Box. The first is the addition of MIDI In and Out sockets on standard five-pin DIN connectors; the original was widely criticised for its lack of MIDI I/O, and although many of those with portable setups are now moving over to USB controller keyboards, its inclusion makes the M Box 2 a more complete recording product. The second addresses one of my major gripes against the original unit. The M Box had a headphone level control but no hardware control over the main output file:///F|/SoS/SoS%2011-2005/mbox2.htm (2 of 5)10/19/2005 9:41:10 PM

Digidesign M Box 2

level, which meant that you couldn't monitor on headphones and mute the main outputs. The M Box 2 has separate level controls for the phones and the main outputs, which is a real boon for those who might want to connect a pair of powered monitors, yet record in the same room. The third improvement is that the M Box 2 is actually a four-in, two-out device, because you can record the two channels of S/PDIF digital input simultaneously with the two analogue input channels. As with the original, however, there's no zero-latency monitoring on the S/PDIF input, which is a shame, and the digital output always mirrors the main analogue output.

Up And Running The M Box 2 ships with version 6.8.1 of Pro Tools LE. The unit itself acts as a dongle for Pro Tools, which won't run without it attached, but an iLok key is also included and used to authorise some of the bundled plug-ins. It's slightly annoying that you need to take up another USB port for the iLok; couldn't the M Box 2 itself have acted as a USB hub? I had a lot of trouble installing Pro Tools, mainly because I already had a different version of the program installed on my machine. LE spotted this and demanded that I use Control Panel's Add/Remove Programs tool to rid my computer of the other version. Once I'd done this, the LE installer got most of the way through its routine before crashing. It seemed that Add/ Buffer sizes between 128 and 2048 samples Remove Programs had not quite are supported. succeeded in eradicating the old version, and it took several hours of head-scratching, calls to Digidesign tech support, blue screens of death and Registry editing to get everything sorted. I say 'sorted', but even after I finally succeeded in installing LE, Windows didn't seem quite itself, sometimes doing strange things to keyboard input and the like. Once I'd got the software installed, however, I was able to get started straight away, and the plug-ins I'd installed in the old version of Pro Tools still seemed to work. Unlike many Firewire audio devices, the M Box 2 appears to be truly hotpluggable, although Windows sometimes took a while to notice that I'd attached it. The new MIDI ports worked as they should do, and with buffer sizes down to 128 samples available, software instruments felt nicely responsive from the keyboard. As on the original, the headphone amp offers plenty of power and a crisp, well-defined sound, whilst the main outputs (on balanced quarter-inch jacks) were also impressively clean.

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Digidesign M Box 2

Digidesign say that their new preamp design is the best available in any current buss-powered device, and having tried them out, I won't argue with that claim. The recordings I made with them were uniformly noise-free, detailed and solid. The M Box 2 had no problem phantom-powering two condenser mics simultaneously, and provided a decent amount of gain — I've found in the past that some budget Pro Tools LE is now a mature and wellspecified DAW, although the lack of any gear has trouble with the very low level facility to disable input monitoring can be output by my Rhodes piano, but here annoying when working with the M Box 2. there was power in reserve. I also had no problem recording the full complement of four inputs whilst playing back a stereo output. The limited bandwidth of the USB 1.1 protocol means that sample rates higher than 48kHz are not supported, but you can record and play back at 24-bit. As on the old M Box, the zero-latency monitoring works well enough, but Pro Tools LE still doesn't include an option to disable input monitoring in software. If you're overdubbing to previously recorded tracks while monitoring the input directly, you'll need to mute the track you're recording to, unless you want to hear the directly monitored input signal doubled with the delayed signal returning from Pro Tools. One consequence of this is that there's no way to drop in on existing tracks whilst monitoring with zero latency. Another is that you're constantly having to manually mute and unmute tracks in order to hear what you've recorded, which is a nuisance. Other software packages have handled this perfectly well for years, so it's disappointing that Digidesign still haven't implemented a better system in Pro Tools. If your system allows, of course, you could ignore the direct monitoring option and use a buffer size small enough to deliver an effective latency that's low enough not to be a problem; the minimum 128-sample buffer size equates to a combined record and playback latency of around 6ms at 44.1kHz. On my machine, however, I ran into problems with small buffer sizes whereby Pro Tools would sometimes stop recording or playback and warn me that 'The operating system held off interrupts for too long.' This never caused a crash, but it did cut me off halfway through a good take on occasion. At the default 512-sample buffer size, it happened infrequently; at smaller settings, it happened more often, and when I raised the buffer size to 1024 samples, it disappeared completely. I find it slightly odd that this should be occuring on a relatively new and high-spec Windows laptop, when Pro Tools M-Powered has never complained about running at 128 samples with an M Audio Firewire 1814. While I'm in 'minor gripe' mode, I'll mention that the I/O Setup window behaved a little oddly. I encountered similar problems when I reviewed PT M-Powered; with fewer inputs and outputs here, there is less to go wrong, but I found that it sometimes failed to create both stereo and mono paths for the S/PDIF inputs,

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Digidesign M Box 2

and that pressing the Default button didn't always rectify this. It's easy enough to create them manually, so it's hardly a fatal flaw, but it can lead to some confusion when you try to select an input, only to find that it doesn't show up in the list of available inputs. The M Box 2's alternate faceplates.

As well as the DAE engine used by Pro Tools, Digidesign also supply an ASIO driver for use with other applications. I tried this briefly in Cubase SX, and it appeared to work as expected, offering the same choice of buffer settings as in Pro Tools. If you didn't want to use the Pro Tools software at all, then the M Box 2 probably wouldn't be something you'd consider, as there are much cheaper USB interfaces available, but it's certainly useful to be able to run other applications in conjunction with PT.

Summing Up It's not hard to see why the general reaction to the M Box 2 has been one of surprise. It is so similar to its predecessor that some will doubtless accuse Digidesign of standing still. In their defence, they might say 'It wasn't broken, so we didn't fix it'. If you want Firewire connection and more inputs and outputs, Digi and M Audio already offer a range of alternatives, but the concept of the M Box still meets a demand, and the changes in this version do represent an improvement over the original. These advances are not momentous, and I can't imagine that existing M Box owners will see much point in upgrading to the new model, while those on the tightest budgets can find cheaper products with more features. In the final analysis, though, the M Box 2 still represents an affordable, practical and portable way to buy into the Pro Tools brand, and I expect that plenty of people will continue to do just that. Published in SOS November 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

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East West Quantum Leap Symphonic Choirs

In this article:

Building A Choir Installation Ooh, Aah Words Under Construction Lessons In Latin In Use Conclusions

East West Symphonic Choirs £500 pros Fabulous audio quality. Different mic positions allow control of the hall ambience, and offer support for surround output. Word Builder is capable of amazing results — with a little effort.

East West Quantum Leap Symphonic Choirs Kompakt Instrument [Mac/PC] Published in SOS November 2005 Print article : Close window

Reviews : Sound/Song Library

Ever fancied arranging 'Oops, I Did It Again' for a full choir? Well now you can, courtesy of Symphonic Choirs, the latest virtual-instrument sample library from East West and Quantum Leap. John Walden

While the sample world seems to be overflowing with quality orchestral Support for Word Builder as libraries, the same is not true for the a plug-in is limited to Cubase, obvious complement to the orchestra — Nuendo and Sonar at present. the choir. Admittedly, choral music is a Requires a well-specified somewhat more specialised genre but, computer (or multiple for film and media composers in computers) for most effective use. particular, the use of choral parts is a common requirement. Way back in summary 1997, Spectrasonics' five-disc Akai/ Symphonic Choirs raises the Emu Symphony of Voices library set bar for the sampled choir. While the Word Builder utility the standard for sampled choral is not perfect, with a little user sounds. It fully deserves its 'classic' effort, the results can be very status and can still be regularly heard in film and TV music. cons

realistic.

information £499.99 including VAT. Arbiter Music Technology +44 (0)208 207 7880. +44 (0)20 8953 4716. Click here to email www.arbitermt.co.uk

Of course, sample technology has moved on considerably since that time, and competing products have subsequently appeared. One of these is Quantum Leap's Voices Of The Apocalypse, a four-disc Gigasampler library that got a fivestar Sample Shop review from Mark Wherry back in October 2002. Again, VOTA has proved a popular library, and it also included one unique feature — the ability to 'build' words from the comprehensive sample set that consists of every consonant and vowel sound for both the male and female choirs.

www.soundsonline.com

Test Spec

Despite being capable of some excellent results, one of the criticisms levelled at VOTA was that the word-building process was rather clunky in operation. QL's new release, produced in collaboration with East West, is Symphonic Choirs,

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East West Quantum Leap Symphonic Choirs

PC REVIEW SYSTEM 3.2GHz Pentium 4 PC with 2GB of RAM running XP Pro (and SP2), with Echo Mia 24, Egosys Wami Rack 24 and Yamaha SW1000XG soundcards. Steinberg Cubase SX v3.1.0.

and, as well as being the largest choral library currently available (nine DVDROM disks containing over 38 Gigabytes of sample data), it comes with a dedicated application for turning text into sung phrases, the appropriately named Word Builder. This application has been written by Nuno Fonseca, who also developed the word-building utility for VOTA. As with other major releases from East West/Quantum Leap in recent months, Symphonic Choirs is available for both Mac and PC using Native Instrument's Kompakt as the playback front-end. So, providing you are not a member of a professional choir, is Symphonic Choirs a good thing?

Building A Choir The Symphonic Choirs package consists of three elements: NI's Kompakt, the extensive sample library and the Word Builder application. There is little to say about the first of these that hasn't already been covered in SOS. Just as it was when supplied with Colossus and RA, which I looked at in recent issues of SOS, Kompakt is a cut-down version of NI's flagship Kontakt software sampler, and the Symphonic Choirs samples set can only be used with the dedicated version supplied here, or the full version of one of NI's software samplers. It's worth mentioning that you need a DVD drive for the installation, plus of course that slimand-trim 38GB of hard disk space to put the library in. In terms of host computer specification, East West recommend that you own a minimum 3GHz Pentium III or Athlon-based PC running Windows XP, or a 1.8GHz Mac G5 running OS 10.2.6 or later, with a minimum of 2GB of RAM on either platform. The sample library itself can also be divided into three components; a series of Kontakt Multis for each of the five choir sections (Soprano, Alto, Tenor, Bass and Boys), individual Kompakt Instruments for the same five sections and Kompakt Instruments for three soloists (Soprano, Alto and Boy Soprano). In principle, the two sets of individual Instrument Programs are similar to more typical choir-based sample libraries in that they offer various performance combinations including vowel sounds, consonant sounds, vibrato, staccato and a selection of special effects such as whispers, falls and shouts. Many of the Programs include velocity and mod wheel control of the sound for extra expressive options, while the three soloists all have key-switched performance articulations available within their Programs, as you can see from the screenshot at the head of this article, which shows a soloist Program loaded. The blue keys contain the samples, while the brown keys select the key-switched performance articulations. The section Multis really come into play with third element of the overall package,

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East West Quantum Leap Symphonic Choirs

Word Builder. This sits between the normal MIDI data input (the melody of the line to be sung) and the Kompakt Multi. Each of the special Multis is arranged across five MIDI channels and contains a combination of the vowel and consonant samples. Given the user-specified phrase, Word Builder then splits the MIDI input across the five channels of the Kompakt Multi to sequence the various vowel and consonant sounds together to create the required phrase. As made very clear in both the advertising and documentation for Symphonic Choirs, it is worth noting that Word Builder can only be used with these specially designed Multis. The individual Instrument Programs — both of the choir sections and the soloists — cannot be used with Word Builder. As with the previous Symphonic Orchestra library, Professor Keith O Johnson supervised the recording process for Symphonic Choirs, using the same concert hall and recording arrangements employed for the orchestral library. All the recordings were made at 24-bit resolution using three mic positions. The 'C' (close) mic position reproduces the sound heard when standing directly in front of the singers, while the 'F' (full) position The stand-alone version of Word Builder, represents the sound from the very while similar to the plug-in version, does have the advantage of being resizable (the front of the stage and consequently stand-alone version cannot be resized). contains more of the concert hall's Resizing is particularly useful when you're natural reverberation. Finally, the working with the Time Editor. 'S' (surround) mic position represents the sound at the back of the hall (the manual suggests the front row of the balcony) and therefore contains the full acoustics of the hall. Each Multi and Instrument is therefore provided in three versions. This approach to the recordings provides three obvious advantages to the user. First, the output from the different mic positions can be used control the degree of ambience, either by using a particular mic position or by blending them as required. Second, output from the different mic positions might be used to create a surround sound mix — and this clearly has applications in a film music context. Third, with both having been recorded by the same team, in the same concert hall and using the same microphone configuration, blending the sounds of Symphonic Choirs with Symphonic Orchestra ought to produce a very coherent mix.

Installation After I'd made some sandwiches and poured myself a stiff drink or two, installation of Symphonic Choirs proceeded without any problems on my test

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East West Quantum Leap Symphonic Choirs

system — although given its size, it did proceed for quite some time! Registration of the Kompakt front-end involves a quick visit to the NI web site, but is painless enough. As with other Kompakt-based instruments reviewed recently in SOS, Symphonic Choirs is provided in all the common plug-in formats for both Mac and PC and as a stand-alone application. Word Builder is installed separately and is also available as a plug-in and a standalone application. However, the plug-in formats for Word Builder are VST-MA (Module Architecture, supported by Cubase and Nuendo from v2 upwards on PC and Mac) and MFX, for use with Sonar 4 under Windows XP. The implication is that users of other sequencers will, at present, need to use the stand-alone version of Word Builder (above).

Ooh, Aah While the potential star of the Symphonic Choirs show is the combination of Word Builder and the section Multis, the various Instrument Programs are also worthy of comment. In short, if you do need just some pad-like 'aahs', 'oohs' or 'mmms', then Symphonic Choirs is more than up to the task. For each of the four adult choir sections, individual Instruments are available for eight vowel-based sounds and 15 consonant based-sounds. These Instruments feature dynamic crossfading (changing the character of the sound) and four key-switched articulation options (normal, legato, staccato and slurred/sliding). The Instruments for the Boys section differ slightly, in that the vowel sounds only include two key-switched attacks; normal and legato — but in other respects they are the same as the adult sections. A small number of 'effects' Programs are also included for each section with performances such as unpitched whispered words, evolving vowels, slides, shouts and falls. Slightly fewer options are provided for the Boys section. In addition, four 'Full Chorus Church' Instruments — Three instances of Word Builder with provided in just the 'S' mic position — are the same phrase in English, Phonetics also provided. These are all based on and Votox formats — translation is done vowel sounds and include mod wheel automatically. control over vibrato depth. While they do not offer the detailed control of working with the individual sections, these single Instruments are excellent for a quick mock-ups or situations where you need a simple background choral 'wash'. Individual Kompakt Instruments are also provided for three soloists. Individual Instruments are provided for each of the Soprano, Alto and Boy soloists and each features a number of key-switches to control the syllable being sung and the style of the singing. These mainly feature vowel sounds sung with nonvibrato, vibrato, slur and other expressive options. The Boy soloist also features file:///F|/SoS/SoS%2011-2005/ewsymphonic.htm (4 of 9)10/19/2005 9:41:14 PM

East West Quantum Leap Symphonic Choirs

random Latin syllables and these could easily be strung together to construct a mock (or genuine!) Latin phrase. For convenience, the final element for each of the soloists is a pre-configured '5.1' Multi consisting of the 'C', 'F' and 'S' mic positions and ready for use in a surround configuration. While experimenting with these various component Programs I was, without exception, impressed by the quality of the sound. The sections are very full sounding and, when combining them to form a full Soprano, Alto, Tenor and Bass ('SATB') chorus, the results were The Learn function allows Word Builder to extremely good. The contrast between match the sung phrase to the timing of a the three mic positions is both obvious MIDI phrase. and useful. The 'C' position is very much 'in your face' and would be ideal if you wished to apply your own reverb. The 'F' positions have noticeably more ambience but are still very clear. As a result, I found myself starting with these versions before blending in either a little of the 'C' or 'S' sound as required. In contrast, the 'S' versions sound very lush and warm, and are extremely effective for constructing choral pads. Interestingly, no Tenor or Bass soloists are provided. The manual suggests that this is because these solo voices rarely appear in film scores. I don't think this is a major omission by any means, but it does reveal who East West have in mind as the potential purchasers of Symphonic Choirs. My only other comment would be that, compared to Spectrasonics' Symphony Of Voices, the palette of Symphonic Choirs is somewhat less diverse — there are no Gregorian chants, for example. Without doubt, though, the advances in sampling technology mean that Symphonic Choirs is putting the most impressive sampled choir sounds currently available into the hands of composers — even before we consider the potential of Word Builder...

Words Under Construction The most intriguing elements of Symphonic Choirs are Word Builder and the special Multis designed to work with it. The application interface uses the same styling as Kompakt. The Voice section has slight differences between the standalone and plug-in versions; in the former, the MIDI In/Out options can be set in this section, while both versions include useful MIDI connector icons that blink to indicate MIDI activity. In the plug-in version, though, because Word Builder is placed as an Insert within a MIDI track, the MIDI in/out routings are set within the host sequencer. The Type setting specifies the type of choral section to which this instance of Word Builder is connected, and this needs to match the type of Multi loaded in Kompakt. The phrase to be sung is entered in the Text Editor. This can be entered in file:///F|/SoS/SoS%2011-2005/ewsymphonic.htm (5 of 9)10/19/2005 9:41:14 PM

East West Quantum Leap Symphonic Choirs

standard English, Phonetics or Votox, a phonetic alphabet created specifically to work with Word Builder. East West recommend that regular users get to grips with it as, once mastered, it allegedly provides more precise control over pronunciation of words. Usefully, Word Builder includes a 100,000-word dictionary and can convert between the various formats. Using one of the phonetic alphabets, Latin phrases also seem to work fine — although if your Latin is a little rusty, see the 'Lessons in Latin' box on the right for some inspiration. The section that requires the most head scratching (initially, at least) is the Time Editor The options in the Letter dialogue (the lower right section in the screen below). box offer you detailed access to When a phrase is first entered, each syllable is each syllable within a word to allocated a certain default time length and these improve pronunciation. are displayed within the Time Editor for the selected word. Syllable lengths are not allocated in an arbitrary fashion; consonants are generally short, while vowels have longer sustains as would be the case for a real singer. By default, each word within the phrase is triggered by a single MIDI note (or chord) although, as described below, this behaviour can be changed. However, it is also worth noting that some syllables (which are triggered by the MIDI note off event) are only sung after the 'off' position on the time line. Therefore, to avoid the Program starting a new word before finishing the previous one, it is useful to leave a gap between the MIDI notes that trigger consecutive words. The default syllable lengths assigned in the Time Editor can be adjusted by dragging the coloured bars. Syllables can be overlapped; this creates a fade and is useful for smoothing out the pronunciation of certain words. However, in most cases, users will want the tempo of the sung phrase to be controlled by a sequence of MIDI notes. It is here that the Learn button (shown on the previous page) comes into play. This function allows Word Builder to learn the timing and length of each MIDI note within a phrase (either played via a MIDI keyboard or from a sequencer track) and to then adjust the lengths of the syllables within each word to match. On the whole, this works well enough but, for all but the most simple of phrases, I still found some minor manual editing was required to make the pronunciation clearer. Further phrase tweaking is possible in the Letter dialogue box (above), which allows the attack type and relative velocity of each syllable to be adjusted. Again, this can be useful for adding clarity to the pronunciation or to add emphasis to particular syllables within a sung phrase — although volume swells are probably more easily added via MIDI continuous controller #11 (Expression).

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East West Quantum Leap Symphonic Choirs

Lessons In Latin While it is difficult to resist getting Word Builder to conjure a selection of common expletives from the Symphonic Choirs Boys section, once this novelty has worn off, Latin phrases can still be good fun for scaring elderly relatives or small children. If, like me, your Latin is not so hot, you might want to check out this link, which was posted by a user on the East West forums: www.yuni.com/library/latin. html. This includes a huge list of short Latin phrases with English translations and, as well as the religious examples such as 'Agnus Dei' (Lamb of God) or 'Sanctum Sanctorum' (the holy of holies) there are also some classic one-liners. I can't vouch for the accuracy of the translations provided, but particular favourites of mine are 'Sane ego te vocavi. Forsitan capedictum tuum desit' ('I did call. Maybe your answering machine is broken') and 'Suntne vacci laeti?' ('Are your cows happy?') but there are countless others.

In Use Currently, the Word Builder plug-in only functions with Cubase, Nuendo or Sonar. Users of other sequencers will need to work with the stand-alone version. This has the same functionality, but is perhaps a touch more fiddly to set up, as it involves using a MIDI loopback connection of some sort. Fortunately, this configuration is described quite clearly within the documentation and on the East West web site. I did most of my own testing using the VST-MA plug-in via Cubase SX v3.1, and both Kompakt and Word Builder behaved 'as advertised'. I did, however, experiment with the stand-alone version too, and had no problems getting Word Builder to talk to Kompakt via a MIDI loopback utility. In use, one or two aspects of Word Builder are a little quirky. For example, in Cubase, key commands have to be disabled in order to type text into the Word Builder Text Editor — regular key command users will find this a bit irritating. A second oddity was the 'Hold Syllable On' function. This allows a MIDI controller to be assigned to enable a sung word to be held over several MIDI notes, effectively The use of multiple vowel sounds — as in stretching the word over a melodic the 'EE' sound used here — was a decent phrase. I could not make this function work-around when I couldn't get the 'Hold work under SX and, having searched Syllable On' function (shown at the top of the the East West on-line forums, it seems next page) to work under Cubase SX! that one or two other SX users have reported the same behaviour. However, the same forums produced a usable work-around. As it is usually a vowel sound that would be sung in this fashion, the vowel can simply be file:///F|/SoS/SoS%2011-2005/ewsymphonic.htm (7 of 9)10/19/2005 9:41:14 PM

East West Quantum Leap Symphonic Choirs

repeated as a separate word within the Text Editor for each note in the melody. Presumably East West and Quantum Leap will be working on these sorts of issues for an update to Word Builder. Two other practical aspects are worth mentioning. First, each instance of Word Builder can hold only a single phrase. A separate MIDI track, each with an instance of Word Builder is therefore required for each phrase. These can, of course, all be routed to the same instance of Kompakt so, for example, the phrases to be sung by the Tenor section only require a single instance of Kompakt to be available. Second, and perhaps more significantly, in order to use Word Builder with a full 'SATB' choir, four instances of Kompakt would need to be run, each with a suitable Multi loaded. By choosing the 'lightest' of the Multis (based on the 'C' mic position and with fewer velocity or key-switched layers), I was able to work quite happily on my test system, but attempting the same with some of the more expressive Multis soon The 'Hold Syllable On' option allows a generated 'memory low' messages. syllable to be held over several MIDI notes, While the 'track freeze' options within but I couldn't get this to function under Cubase SX. However, I found a work-around SX provide a suitable work-around, (see below left). professional composers using Word Builder and Symphonic Choirs on a regular basis would probably wish to follow the advice given in the printed manual — use a very well-specified machine or spread the workload across multiple computers. All this said, the combination of Word Builder and Symphonic Choirs is capable of some truly remarkable results. Is Word Builder perfect? No, it is not — it does take some time to learn how to use it and, even with experience, some manual tweaking of pronunciations and phrasing is required to add a sense of 'life' to a part. However, a useful comparison here is with Yamaha's Vocaloid (reviewed back in the March and December 2004 issues of SOS). While Vocaloid is also capable of amazing results, crafting an entire solo lead vocal is a major undertaking. With Symphonic Choirs, the task is less daunting, as the very nature of a choral performance makes the absolute clarity of the diction less critical. As a result, obtaining convincing results from Word Builder and Symphonic Choirs requires much less effort on behalf of the user, and I could imagine experienced users soon learning to make fairly light work of individual phrases.

Conclusions

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East West Quantum Leap Symphonic Choirs

Symphonic Choirs takes the 'virtual' choir to a new level. The sound quality of the sample library is simply magnificent; these sounds would add class to any musical production. Word Builder is both remarkable in its conception and fun to experiment with — but it can also be frustrating to the new user, so go in with your eyes open and, expect to invest some time, at least initially. I hope East West have the energy and resources to keep developing the potential of Word Builder, and I'm sure suggestions from users will provide useful input to this process. I can see this library appealing to hobbyist, educational and professional markets, although the two former groups ought to heed East West's recommendations about the need for a high-end system in order to keep the work-flow efficient. However, as film and TV music budgets are nearly always tight, Symphonic Choirs is undoubtedly going to be a solution that many a media composer will turn to. In that context, it is most certainly worth the asking price, and the computer resources required to get the most out of it are unlikely to be a major concern. As with East West's recent world and ethnic instrument library, Ra, I think you can expect to hear Symphonic Choirs coming to a film score near you soon. Highly recommended! Published in SOS November 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

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Kawai MP8

In this article:

Beauty & The Beast Features & Sounds Sound Comparisons Multitimbre Modes The AWA Grand Pro Keyboard Sound & Setup Modes XLR Outputs Conclusion Brief Model Comparison Chart

Kawai MP8 £1999 pros Arguably the best-sounding digital stage piano on the market. A wide range of additional sounds. 192-note polyphony. High-quality weighted keyboard action.

cons Weight of keyboard action takes some acclimatisation. Needs two people to lift it — worth remembering if you do plan to use it live... Master Volume slider bypassed when using XLR outputs. Significant timing latency when played via MIDI — although on a keyboard designed for live playing, this may not bother potential users much.

summary The Kawai MP range goes from strength to strength, with more of everything. The great piano sounds the company has become known for are enhanced by 192-note polyphony, 256 userprogrammable Setups, a wide range of supplementary

Kawai MP8 Digital Stage Piano Published in SOS November 2005 Print article : Close window

Reviews : Keyboard

Kawai's last few digital pianos have been finely wrought things of beauty: solidly built keyboards with an amazingly realistic playing action and beautifully sampled piano timbres. But their latest claims to surpass all of those. Can it possibly be true? Nick Magnus

The MP8 is the latest flagship addition to Kawai's acclaimed MP range of digital stage pianos, following on from the excellent MP9000 (reviewed in SOS January 2000 — read the review at www.soundonsound.com/sos/jan00/ articles/kawai.htm) and the MP9500 (SOS January 2003 — or see www. soundonsound.com/sos/feb03/articles/ kawaimp9500.asp).

Photos:Mike Cameron

Fans of the MP9000 proclaimed it to be 'near-perfect' — not least because of its superior AWA Grand weighted keyboard action. It also sounded rather splendid. The MP9500 received yet more accolades for its improved AWA Grand Pro action, and introduced a more flexible Setup mode, allowing up to four internal/ external sounds to be layered together, as opposed to the MP9000's two internal and two external layers. The MP8 brings yet more to the table — in short, the design philosophy is 'more of everything'.

Beauty & The Beast The MP8's new black livery and dark wooden end cheeks are striking and

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Kawai MP8

sounds and Kawai's top-ofthe-range AWA Grand Pro 88note keyboard action. Try it, and then try to resist it!

information £1999 including VAT. Kawai UK +44 (0)1480 474751. +44 (0)1480 474752. Click here to email www.kawaiuk.com

sensual, lending the instrument a sense of gravitas befitting a quality grand piano. The basic layout and appearance of the panel remains largely unchanged from previous models apart from some additional buttons at the right-hand end, and four red and four green LEDs accompanying the four sliders on the left — more on which later. The remaining front-panel controls, EQ and Effects sections have been carried through from the MP9000 and MP95000 — a detailed description of these can be found in that January 2000 MP9000 review. Rear-panel connections (see the final page of this review) are also much as before, with two differences. Firstly, the headphone socket is gone, having been relocated to the front (left) of the instrument. This is sure to please everybody who has ever complained about rear-mounted headphone sockets — myself included! Secondly, keeping in vogue with the current trend, a USB connector is provided. No specific editing software is bundled with the MP8, so in this case USB is provided simply as an alternative to using conventional MIDI connections. Windows XP and Mac OS X both include generic USB device drivers — Windows 2000/98SE users will need a suitable driver, which can be downloaded from www.kawai.co.jp/english/Download1.html. USB is not supported for Mac 9. xx, so a standard MIDI interface will be needed.

Features & Sounds Kawai have not just doubled, but tripled the polyphony previously offered on the MP9000 and MP9500. The MP8 can play a whopping 192 notes, allowing for the most pedal-heavy, cadenza-laden performances to be reproduced without missing a note. This is clearly of benefit when playing layered sounds — even with all four layers addressing internal sounds, a respectable 48 notes of polyphony is always available. The MP8 provides a sizeable set of 256 sounds, laid out in eight sound categories — Piano, Electric piano, Drawbar, Organ, Strings/Vocal, Brass/ Wind, Pad/Synth and Bass/Guitar. Each category has eight principal variations, and each of those variations has a further four sub-variations, accessed via the four new A,B,C & D buttons below the two rows of eight preset buttons.

The main volume slider, zone level faders, keyboard zoning controls, and four rotary controls at the left end of the MP8. The latter controls can be used to adjust effects, EQ, sound envelope parameters, or to send MIDI controller data, as on the original MP9000.

The acoustic piano sounds are as good — if not even better — than those on the previous MP models. Kawai's Harmonic Imaging system works extremely well, providing almost seamless dynamic variation and a consistent (although in places not perfect) tonal balance across the keyboard. file:///F|/SoS/SoS%2011-2005/kawaimp8.htm (2 of 8)10/19/2005 9:41:18 PM

Kawai MP8

The slightly plummy 'Concert Grand 2' and the Elton John-flavoured 'Studio Grand 2' get my vote as personal favourites, but all the variations are eminently useable, and are designed to suit a wide range of musical styles. The samples all sound as if they have been recorded using close-miking techniques — the tone is highly detailed, more so than if the microphones had been placed at a greater distance. This may not be to everyone's taste, and as such the MP8 arguably lends itself (in a DI'd recording situation) more to modern music styles. Of course, if you were giving a classical recital using the MP8 in a hall, the audience would provide its own distance! To further improve realism, there are two new additions to the pianos' edit parameters — String Resonance and Damper Effect. These give independent control over the reverberant 'thump' heard when notes are played with the sustain pedal down (Damper Effect) and the amount of sympathetic overtones created by other strings in a specific note's harmonic series (String Resonance). Both can be turned off if they prove distracting, which can be the case in headphones, but it seems less of a problem on speakers. These parameters replace the MP9000 and MP9500's Sympathetic Resonance, which on those instruments was provided as an EFX effect. There are also plenty of new and varied patches to be found, such as Steel Drums, Nylon Guitar, Scat Vocals, Bassoon, Pan Flute, Banjo and Pedal Steel to name just a few, and there are even four basic Drum kits lurking at the end of the Bass/Guitar category. Sonic highlights include the Rhodes soundalikes, which are fun to play and would certainly pass muster on stage or in the studio. The '60s E-Piano' is a tad too bright and 'clipped' to be a convincing Wurlitzer, but could easily pass as a Hohner Pianet N (think The Zombies 'She's Not There'). The FM-style electric pianos are pleasingly nostalgic, creating a suitably convincing 'LA' feel when layered with the acoustic pianos. Other sounds are less authentic (the woodwinds instantly bring to mind Roland's SH2000 synth, and the Clavinets come across as disappointingly 'fake' and synthetic) but this is offset by a number of very useable Drawbar Organs (featuring the Rotary EFX) and some rather comely String textures.

Sound Comparisons As mentioned elsewhere in this review, the MP8's piano sounds compare favourably with some of the high-octane software pianos on the market. Out of interest, I did a direct comparison with two pianos that I hold in high regard — the Roland SRX11 Complete Piano expansion board installed in my XV5080 (a version of which apparently graces Roland's new RD700SX), and Vintaudio's Yamaha C7 (close-miked version) running under NI Kontakt 2. The Vintaudio C7, incidentally, uses 4GB of samples, comprising six velocity layers with pedal up, down and release layers. The SRX11 has no pedal down or release layer, but all 88 notes have been sampled across four velocity layers. The MP8 appears to have been sampled at every fourth semitone, and has a release layer but no pedal down layer (although it does have the Damper Effect and String Resonance to compensate). The number of velocity layers on the MP8 is harder to determine — clearly Kawai's Harmonic Imaging method is very effective! I chose three existing projects to run in Sonar, each of which was piano-driven but with the emphasis varying between rock and orchestral. The results were very file:///F|/SoS/SoS%2011-2005/kawaimp8.htm (3 of 8)10/19/2005 9:41:18 PM

Kawai MP8

illuminating — all three pianos acquitted themselves admirably. However, what did become clear is that there is no such thing as the universally 'perfect' piano sound. Where one piano suited a certain piece, it sat less comfortably in another. The MP8 is arguably the most versatile of the three, having more tonal variations to choose from — however, within classical/orchestral passages the SRX11 was the most effective, possibly due to the fact that its samples are not as close miked as the MP8 or C7, hence it sounded more natural in a classical context. Both the MP8 and SRX11 responded in a very consistent and musical manner to the prerecorded performances' key velocities, whilst the C7 tended to over-react slightly at lower velocities. On a slightly less glowing note, I noticed the MP8 was prone to some MIDI timing discrepancies, which were especially noticeable when playing back a quantised piano sequence alongside other quantised (zero-latency) software instruments. The discrepancies varied from negligible to delays of anything up to 20ms, which may not sound like much but is distinctly audible, particularly within a simple arrangement. As these 'latencies' are not consistent, they cannot be compensated for by simply shifting the part (or recorded audio) forward in time — that just makes certain notes occur early rather than late. Something is clearly awry here, but admittedly, it's less of a problem on this keyboard, which is clearly designed to be played rather than triggered, than on, say, a multitimbral synth module. In these days of excellent software-based piano instruments and affordable (and much more compact) rack units, I can't believe there will be many people, or indeed anyone, who would buy the MP8 solely as a sound source for triggering over MIDI, so the problem, whilst certainly extant, is arguably limited by the MP8's very nature. Nevertheless, if they can, it would be good if Kawai could do something about this!

Multitimbre Modes Like the MP9000 and MP9500 before it, the MP8 can also function as a 16-part multitimbral instrument. The manual is extremely reticent about this, stating that you can do it but omitting to explain the procedure — which is bound to cause great frustration to MIDI technology novices! The procedure is actually straightforward, and of course requires the participation of a MIDI sequencer. Firstly, the MP8 must be set to Multitimbre mode in the System menu. Two such modes are offered, the difference being how the MP8 interprets MIDI program change commands. Mode 1 The display's a bit of an '80s throwback responds to bank and program changes by in terms of its size, but then there isn't number according to the list set out in the too much that needs to be accessed or viewed with it, so it's not too much of a MP8's manual. Mode 2 responds to problem. program and bank changes by name according to the GM standard — so if your sequencer is configured so that you can specify 'acoustic bass' as the sound you want, the relevant MP8 channel will select the nearest equivalent sound. Since

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Kawai MP8

the sounds for individual channels cannot be directly assigned from the MP8 panel, sending program change messages is the only means to select the parts' sounds. There are compromises to using Multitimbre mode — for example, the only effects available to each part are Reverb and Chorus. Unless I'm missing some glaringly 'obvious' hidden parameters, the other EFX effects appear not to function at all — and again, the manual makes no reference to this. As a result, the Drawbar presets (which normally use the EFX Rotary effect by default) don't sound as they should in this mode. This is curious, as the older MP9000 allowed EFX to be used on parts 1&2. Multitimbre mode also caused some strange things to happen when using the USB connection. My sequencer (Sonar 4) took a worryingly long time to start up, Sound/Setup selection was erratic (with buttons refusing to respond), and I lost USB input several times while in this mode, although the MP8 happily played back recorded data. None of these problems occurred using standard MIDI connections.

The AWA Grand Pro Keyboard First seen on the MP9500, the AWA Grand Pro keyboard constitutes Kawai's attempt to replicate the touch and feel of the company's own Concert Grand EX acoustic piano. The result is a fairly heavy action which will certainly delight some players, whilst conversely being initially quite hard work for those used to a lighter action. My initial impression was that it is distinctly heavier than many acoustic pianos I've played, but after a few hours of playing, you soon get used to the extra muscle power it demands. You also begin to appreciate the enhanced sense of dynamic control this keyboard brings — especially at the lower end of the velocity range. To help players adapt to the feel of the keyboard, the MP8 provides five preset velocity curves — the normal default 'linear' response plus 'light', 'light+', 'heavy' and 'heavy+' choices. Most people will probably be happy with the normal setting — however, if none of these curves feel comfortable, the MP8 very obligingly offers two user-definable curves. To define a User curve, simply play the keyboard in a manner that feels natural to you (the manual suggests turning the volume off so you won't be distracted by what you hear). When you've finished playing, the MP8 analyses your touch and scales the velocity response to match the gentlest and hardest notes you played. This seems to work quite well — the resulting curve feels quite comfortable to play, although you may subsequently find yourself tempted to redefine the curve according to each piece of music you're playing! One small criticism concerning the manufacturing finish is that the top corners of the keys (on the review model, at least) are rather sharp — so if you're planning on doing sweeping Hammond glissandi, make sure you have a first-aid kit to hand!

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Kawai MP8

On the MP9000 they were called Sound and Setup mode. In MP9500 parlance they were referred to as Single and Multi mode. Now the MP8 has reverted to calling them Sound and Setup... so how do these modes operate on the MP8? Basically, Sound mode provides access to 256 preset tones, arranged by category as described previously. Setup mode provides 256 user memories in which to store all your edited sounds and master-keyboard configurations. When first powered up, the MP8 presents itself in Sound mode; any of the 256 Presets selected in this initial state will call up a single sound. However, although the primary function of this mode is to select single sounds, its secondary function is to serve as your editing palette. From here, you can activate any or all of the four zones, select sounds for each zone, edit those sounds, apply effects, set zone key ranges, and so on. So rather than merely being an operational state Patch-selection and management controls from which you select single preset are to the right of the MP8's top panel, sounds, Sound mode is also a creative including the buttons for switching between starting point — a single preset 'Setup' Sound and Setup mode. in itself wherein all the MP8's parameters are freely editable. For example, you could make zones 1 to 4 active, layering four different internal sounds together, while zones 2 and 3 simultaneously transmit data to external MIDI devices on different channels, and the MP8 will remain in this state until you either change something or power off the MP8. Once you have created the sound and master keyboard configuration you want, you simply save it to any of the 256 Setup memory locations for later recall. If you wish to quickly return Sound mode to its original 'single sound' state, simply press the first two 'number 1' preset buttons simultaneously (marked 'Piano Only) and you're back to scratch. To make it easier to identify which zones are addressing internal sounds, external devices or both of these, each zone fader is now accompanied by a pair of status LEDs. If the red LED is lit, that zone is assigned to an internal sound. If the green LED is on, the zone is transmitting data from the MIDI output. If both LEDs are on, then that zone is assigned to an internal sound and the MIDI output. The large zone on/off LEDs below the faders normally glow red when a zone is active — however, if a zone has been set to less than the full 1-127 key range, it will glow green to indicate that zone operates over a restricted key range.

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Kawai MP8

XLR Outputs One further (but not visible) rearpanel change is to the XLR outputs; on previous models, these outputs rather perversely bypassed the Master Volume and internal EQ. On the MP8, the EQ settings do affect the XLR outputs, although the There are a couple of changes from the Master Volume slider remains MP9500's rear panel — firstly, the bypassed as before. This latter point headphone socket has sensibly been moved to the front, and MIDI interfacing is is less of a problem when playing now possible via the new USB port on the single sounds (you can simply raise right. Otherwise, the connectors — stereo or lower that sound's Zone fader), out on jacks and XLRs, five-pin MIDI trio, but becomes rather inconvenient expression pedal and footswitch jacks, when playing layered Setups, where and socket to attach the damper and soft the balance between the layers is pedal — are unchanged. critical — there is no single fader to control the level of the total sound. Admittedly, you can go into Edit mode and change the Setup's Master Volume setting there, but this wouldn't be an ideal solution if you needed to change level on stage in a hurry. Alternatively you could connect an expression pedal and use that to alter the volume, but I suspect most people would rather opt for the standard jack outputs so they can use the front-panel fader!

Conclusion The MP8 is a fine instrument, comparable with the better Giga-sized software pianos out there. In fact I'd wager that if you sat the MP8 next to a computer and told people they were playing some top-flight software piano from the Kawai's keyboard, most would accept it without question. Which leads me to wonder whether Kawai's next step might be to develop a truly comprehensive, gigabytesized piano sample set that would rival or improve upon the best of the software pianos. High-capacity 1GB and 2GB flash RAM cards are commonly available now that don't cost the earth — so why not take advantage of today's inexpensive memory technology and use it inside a stage piano? As a master keyboard, the MP8 may not offer the same level of control as the current crop of dedicated, purpose-built units — but it certainly provides more than you'd expect from your average digital piano. There's no arguing the quality of the MP8's keyboard, which lends a great deal of pleasure to playing those great piano sounds. Whether you feel the keyboard action is appropriate for all your playing tasks is very much an individual choice — I personally found the MP8's action slightly heavy for general non-piano applications. Rather like trying to trim your fingernails with garden shears, it's not necessarily the right tool for every job, leading me to prefer my trusty semi-weighted synth-action keyboard for tackling most non-piano parts.

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Kawai MP8

If the MP8 contained only the piano sounds, it would be a fantastic instrument. As it is, the wide range of additional sounds should prove a welcome bonus for players who major in piano but need a little extra firepower, especially in a live context. Anyone who has previously considered investing in one of Kawai's MP pianos should now be especially attracted by the huge polyphony of the MP8, and for those looking to incorporate it into a The F20 sustain and soft pedal unit supplied with the MP8. live MIDI rig, those 256 programmable Setup configurations make it an even more attractive proposition. Find one, play it, and try not to want it.

Brief Model Comparison Chart MP9000

MP9500

MP8

PRESET SOUNDS

16

64

256

SETUPS

64

64

256

POLYPHONY

32

64

192

KEYBOARD ACTION

Enhanced AWA Grand action

AWA Grand Pro action

AWA Grand Pro action

VOLUME/EQ TO XLR OUTS

No

No

EQ only

WEIGHT

34kg

32kg

32kg

HEADPHONE SOCKET

Rear-mounted

Rear-mounted

Front-mounted

Published in SOS November 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

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Korg OASYS: Part 1

In this article:

OASYS I & II Bringing OASYS To The Market OASYS III OASYS — Or POCKY? The Open Synthesizer? Big Screens, Little Screens HD1 EXs2 — Piano Lessens And The Sound? Next Month...

information OASYS88, £5399; OASYS76, £5149. Prices include VAT. Korg UK Brochure Line +44 (0)1908 857150. +44 (0)1908 857199. Click here to email www.korg.co.uk

Korg OASYS: Part 1 Workstation Synth Published in SOS November 2005 Print article : Close window

Reviews : Keyboard workstation

For over 15 years, Korg have produced the world's most successful workstation synths, and the OASYS is their new £5400 flagship, their attempt to take the concept to the next level. In the first instalment of this two-part review, we assess how they've fared... Gordon Reid

Tens of thousands...nay, hundreds of thousands of words have already been written about the OASYS, over 2000 of them in this magazine (see the preview in March's SOS at www.soundonsound.com/sos/ mar05/articles/ korgoasys.htm). Quite apart from Korg's humongous brochure, the Web offers numerous discussions, videos, and a host of support services that tell you everything you need to know about it. Except that they don't. You know that OASYS is expensive, has three modes of synthesis, is a powerful sampler, incorporates wave Photos: Richard Ecclestone sequencing and KARMA, has a big sequencer, built-in effects, and offers audio recording to make it as close to an all-in-one production system as is currently possible. But even if you know all that, do you know what it feels like to use one? Let me answer that: if you haven't used one, you don't. So, over the next two months, I'm going to try to tell you what it's like to program and make music on the OASYS88. What I'm not going to do, however, is tell you everything about it. Why? Because the manuals are well over two inches thick, that's why. If four issues of Sound On Sound are piled on each other, they're still not as thick as the OASYS's parameter guide! And do you know what...? These

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Korg OASYS: Part 1

huge tomes still don't tell you what it's like to use an OASYS! But now I'm getting ahead of myself. Let's take a step back and begin, as all good stories should, at the beginning...

OASYS I & II Back in June 1995, Korg asked me whether I would be willing to exhibit some vintage keyboards at an event they were planning to hold in the Science Museum in Kensington, London. They wanted to demonstrate the progress they had made since their earliest days, so I took a Minikorg 700, a Trident, a Polysix, and a PS3200, and sat back to enjoy whatever it was that the company had lined up for the evening. Soon, I wasn't leaning back, but leaning forward, intent on gleaning everything that I could from the presentation. The event was, of course, the dual launch of the Trinity and Prophecy, and it was riveting. But dramatic though the launches were, the presenter, Korg's Steve McNally, kept referring to two mythical products that appeared simultaneously to predate and supersede the Trinity and the Prophecy. Both had been developed at Korg R&D in California. The first, called Synth Kit, was a software environment within which the physical models in the Prophecy had been developed. Based on a Mac computer with additional DSP hardware, it was perhaps the first system capable of crunching the numbers needed to perform physical modelling in real-time. The other was a keyboard designed using Synth Kit, intended for thee and me, and which was mooted to cost £10,000. This huge sum of money was justified by the instrument's ability to run several synthesis engines simultaneously. There was even an example of it on show. It was big, it was blue, it was called OASYS, (the Open Architecture SYnthesis System) and it looked the business. But we couldn't touch it, we couldn't play it, and we most certainly couldn't hear it. The OASYS concept was an appealing one and, in 1995, quite radical. To quote Mr McNally, whom I subsequently interviewed, "OASYS is basically a computer, rather than a hardwired keyboard synthesizer. If you want to change how it works you can load a completely different synthesis system from a hard drive. It's multitimbral as well as polyphonic, and can also be multitimbral in the sense that different types of synthesis can be positioned under different areas of the keyboard. For example, you can have a single patch which, when you play softly at the top of the keyboard, gives you an FM sound layered with an analogue sound, but gives you a physical model of a file:///F|/SoS/SoS%2011-2005/korgoasys.htm (2 of 14)10/19/2005 9:41:30 PM

Korg OASYS: Part 1

saxophone when you play a little bit harder, and a PCM sample of a pipe organ when you bring in the ribbon controller... It's all completely controllable." Umm... no, it wasn't. The 'Blue Bomber' (as it was later called within Korg) never appeared. The reason was that it didn't work. Apart from a few limited demos, it had never worked, and it was never going to work. The technology available in 1995 simply wasn't up to the task. Four years later, the second incarnation of the OASYS appeared. The OASYS PCI card incorporated many of the ideas that had formed the bedrock of the OASYS ideal. It boasted a 12-channel mixer at the centre of the system, and this provided access to some superb physically modelled synth engines, more than 100 excellent effects, a PCM-based synthesis engine with multisample loading and manipulation, audio streaming from hard disk, 32 MIDI channels of control, plus 24-bit audio I/O. The difference was that, whereas the original OASYS had been planned as a synthesizer with a computer inside, the OASYS PCI was a synth on a card designed to go inside a computer. In some ways, OASYS PCI was a triumph. Its synthesis was superb and, when Korg released it, they announced that Synth Kit itself would soon be made available, thus allowing users to design their own models. Unfortunately, it had a fatal flaw. When used to the full, the polyphony dropped to as few as four or even three notes. So, while OASYS PCI was very close to McNally's description of the mythical OASYS keyboard, you couldn't do anything useful with it. Furthermore, while Synth Kit eventually appeared, the anticipated slew of third-party synth models and effects algorithms did not. OASYS PCI was not a success, and it passed through the world causing barely a ripple on the consciousness of music makers anywhere. But the promise of 1995 seems finally came to have come to fruition with the appearance of the third product to bear the OASYS name, the Open Architecture Synthesis Studio. The first production OASYS (serial number 000001) is in my studio as I type this, and it's the unit you can see in these pictures. As you can see, it's beautiful.

Bringing OASYS To The Market Korg faced a dilemma when developing the OASYS. Unlike previous workstations, whose capabilities and limitations are decided when their various VLSI chips are fabricated, Korg have stated that the OASYS specification is a work-in-progress, limited only by programming and marketing decisions. You want some more of this? Sure. Some more of that? No problem. This has led to an enormous wish list that encompasses the sound engines, effects, sequencing and recording capabilities. Korg couldn't incorporate every idea in the first release version of the OASYS's operating system, so they had to decide what was a priority, and what could be the subject of later upgrades. In the end, they decided that sound generation, sound quality, and the KARMA engine were the most important aspects of OASYS, so they concentrated on finishing the three synth engines that it boasts today. file:///F|/SoS/SoS%2011-2005/korgoasys.htm (3 of 14)10/19/2005 9:41:30 PM

Korg OASYS: Part 1

However, that's not to say that other aspects of the OASYS, such as the sequencer and recorder, are unfinished. I've had access to them for two months, and can attest that they are fully functional, and seem robust in operation. However, I think it's safe to say that we will see a great deal of development in these areas over the coming months.

OASYS III Every once in a while a keyboard appears that captures your imagination even before you start to play it — the ARP Quadra, the Roland Jupiter 8, and the Yamaha GX1 all spring to mind — and the OASYS is one of these. But unlike the colourful Quadra and Jupe, and the physically imposing Yamaha, the OASYS is slightly understated. When you sit in front of it, it says, "Yes, I'm bloody impressive, but in an unassuming sort of way. If you're a serious musician, I'm exactly what you need." Touching the OASYS confirms what your eyes are telling you. Take the keyboard as an example. Korg has pursued a policy for nearly two decades of building two families of synthesizers based on any given technology: a flagship line that incorporates expensive keyboards, and a more affordable line with low-cost keyboards. It's clear which one they've used in the OASYS; the keyboard is a pleasure to play. More imposing yet is the screen. Everybody must know by now that the OASYS is essentially a PC at heart, and that it is edited primarily via a 10.4-inch colour, touch-sensitive screen, yet this wasn't always the plan. Apparently, OASYS started life with a screen the same size as a Triton's. But, at one of the early development meetings, Tsutomu Katoh, the founder and President of Korg, entered the room, saw the drawings and said simply, "Too small... screen too small... make screen bigger". Soon afterward, an email was circulated to announce that the synth was to have the screen that it now boasts. So, what's it like to use a 10.4-inch colour, touch-sensitive screen on a synthesizer? The answer, in my best American accent, is "awesome, dude". As regular readers of Sound On Sound will know, I still use my Trinity Pro as both a major sound source and as my controller keyboard, and one of the most significant reasons for this is because the touchscreen works so well. Although small by modern standards, it is responsive, has no faulty pixels or lines across it, and it's angled by just the right amount to make it useable in almost all situations. On OASYS, it's simply excellent to have a large, bright, touch-sensitive The OASYS's colour touchscreen,

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Korg OASYS: Part 1

through which players access all the screen that you can place at the angle of parameters relating to the operation of your choosing. What's more, by avoiding the synth. Microsoft's Windows and Apple's Mac OS X, Korg have been able to write application software that is perfectly tailored for the screen, and I am convinced that it would not have been possible to equal its clarity within the standard operating systems. I do, however, have a reservation. Having owned a laptop whose hinge mechanism sheared on more than one occasion, I'm concerned about whether the clutch mechanisms supporting the OASYS screen are strong enough. The screen is supported only by its clutches, and that's a lot of weight to be bearing backward week in, week out. What's more, unlike the screen on a laptop, which suffers no poking or prodding, you're going to be stabbing at the OASYS screen from the day that you start to use it. I have discovered that the hinges are only guaranteed for 2000 movements, which may be fine in the studio, but which may well prove inadequate for long-term, regular live use. I'm surprised there's no support or brace, like the one that supports a Minimoog's control panel. I wonder whether Korg might consider this.

As for the rest of the controls, it's extremely rare for everything on a synthesizer or workstation to fall to hand so correctly. How many times have you felt that it would be nice if this switch were over there, or that fader were over here? I find few keyboards to be ergonomically satisfying, and the last with which I was really comfortable was the Trinity. Well... it's taken 10 years, but there's now another, and it's the OASYS. If you haven't yet used an OASYS, and are just looking at these pictures, it's important to realise that (with the exception of the value slider and up/down buttons) you edit it using the touchscreen and the switches to the right of the screen, and you control it using the performance controls and the knobs and switches to its left. This means that, for example, if you want to create a huge filter sweep in the middle of a song, you should first assign the cutoff frequency to a knob or fader, not try to control it directly from the screen. You can do the latter, of course, but why would you? It's the wrong way to go about it. Once you've got to grips with this basic philosophy, everything falls to hand: not just to the extent that editing sounds becomes a doddle, but putting together whole songs using synthesized sounds, samples, sequences and audio recorded to disk is remarkably straightforward. I would venture that, notwithstanding some criticisms (of which more later, and next month), there has never been a keyboard that is closer to being a complete and useable music production system.

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Korg OASYS: Part 1

OASYS — Or POCKY? I work in Japan for about 10 days each year, and I am a sucker for the snacks available from the all-night convenience stores spotted every hundred metres or so along every major street in Tokyo. There are cute little sandwiches, surprisingly good sushi and bento boxes — and irresistible chocolate biscuit sticks which go by the name of 'Pocky'. Well, it seems that Korg employee Steve McNally shares my taste for Pocky sticks, because (so the story goes) some years ago, he was nibbling away at a packet in a development meeting when Korg's president, Tsutomu Katoh, walked in and grabbed one of them. Apparently, Steve slapped his hand away and said something (probably through a mouthful of chocolate) that boiled down to, "Oi, get your own". Katoh growled, everybody laughed, and Katoh then left the room, only to reappear half an hour later with about 150 packs of the things, which he dumped on the table, before leaving again. Now fast-forward several years, to the rebirth of OASYS. As you may know (or not), Korg products are developed by individual country groups that work in specific areas of the music industry. For example, Korg Italy is responsible for domestically oriented instruments such as the PA1X, while Japan has traditionally developed workstations such as the Triton. Anyway, in 2001, Korg R&D (the Californian group responsible for Synth Kit, the original OASYS, and OASYS PCI) presented a proposal to Korg Japan. This suggested that the concept of an updateable, multi-synthesis, open-architecture keyboard was now feasible, and could be developed provided that they and Korg Japan worked in parallel to design and build it. It was a radical proposal, because no other major instrument had crossed geographic boundaries in that way, so the senior management asked for a Proof Of Concept for Korg (Japan) to be presented by the end of the Year. In short, a POCKY. I have picked up hints that — for a while, at least — this may have been an informal way of referring to the new project amongst chocoholic Korg employees. However, by the time the world heard of it, the product was once again named OASYS. In my view, that was a very good decision!

The Open Synthesizer? Although Korg like to think of the concept of open architecture synthesis as their own, the 10-year delay between the dream of OASYS and the reality has meant that the company has been beaten to the market not once, but several times, by very different systems that offer the same promise. One of these was the Roland VariOS, which Roland call an 'Open System Module'. This is a configurable module that — in principle — will allow you to load new synthesis engines, effects, and production tools as and when they become available. Unfortunately, VariOS was launched in 2003 with three modules (V-Producer, VariOS 303 and VariOS 8) and, two years later, these are still the only modules available. The blame for this lies entirely with Roland, who never released a third-party developers' kit. Not surprisingly, few people have chosen to investigate the potential of VariOS. file:///F|/SoS/SoS%2011-2005/korgoasys.htm (6 of 14)10/19/2005 9:41:30 PM

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More significantly, the past few years have seen the appearance of the keyboard/computer; most notably the Open Labs Neko, which was reviewed in Sound On Sound in December 2004 (see www.soundonsound.com/sos/jan05/ articles/openlabsneko.htm). Unashamedly a marriage between a hardware control surface in a keyboardshaped case and a PC running Windows (albeit with a shell operating system to hide the Windows core from the user), you might think that this offers everything that the OASYS does. An array of assignable sliders and rotary controls fall easily to hand to the left of the touchscreen.

Some have taken this line of thinking further. Surely, they say, you could emulate the OASYS's sound generators with a PCM-based software synth, a virtual analogue software synth, and a Hammond software synth? Add a software sampler, a competent MIDI sequencer with audio recording capabilities, and a bunch of VST effects, and you have the lot, at a fraction of the cost. But as you can probably tell, I don't think this argument stands up, and here's why. The heart of the OASYS is indeed a 2.8GHz Pentium 4 PC with 1GB of RAM. However, instead of running Windows, the boot system is a cut-down Linux core (Linux was derived from the Unix operating system, and is often used for critical applications where stability and reliability are of paramount importance). In the OASYS, there is just enough Linux present to make the computer boot and load Korg's proprietary application software. This confers a huge benefit. You have to understand that neither Windows nor Mac OS X was designed to be a real-time operating system. If, for example, Windows decides that it wants to check whether there is a modem present, or to update some of its internal parameters, it will interrupt whatever applications are running and do so. This is of no consequence if you're performing a spot of word processing or browsing the web, but if you're sending eight channels of 24-bit, 96kHz audio out of the PC in question, it can be disastrous. The only way around this is to disable as much of Windows as you can (which is one of the major elements of optimising PCs for audio applications) and to use software that has large enough buffers to bridge the gap when the processor goes off to gaze at Bill Gates' navel. So, what's the big deal about OASYS? It should now be obvious. If you ask a PC to play more voices, or use more effects, or output more audio tracks, there will come a point where the CPU can no longer cope with the demands being placed upon it by numerous disparate applications. Anyone who uses such a system knows what happens next. At that moment, it takes just the tiniest push to shove the whole thing over the edge, and you get a glitch, a freeze, or the 'blue screen of death'. Sometimes you can defer the point of no return by increasing the file:///F|/SoS/SoS%2011-2005/korgoasys.htm (7 of 14)10/19/2005 9:41:30 PM

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latency of the system, but that's often not acceptable in a musical environment. On the OASYS, you can continue to ask for voices, effects, samples, and streamed audio, and when you decide to add another channel of VA synthesis, or another Hammond (or whatever), the synth simply says, 'I can't do it all, mate...' and steals a voice from somewhere. Because the whole system is integrated, it can cope with excessive demands in a way that a Mac or PC cannot.

Controls on the right of the screen include numeric keypad and data-entry dial, Mode and Bank selection buttons, and the switches for starting and stopping the built-in sequencer and sampler (more on these next month).

Integration confers other benefits, the most significant of which is speed. Let's say that you're the arranger and keyboard player for a West End production, you do 'spots' on a TV show, and you play in a rock band for fun (if so, congratulations!). Each of these has different requirements, and the modern way to cope with them is to couple a portable PC offering sequencing and software-synthesis capabilities to a controller keyboard and a small selection of rackmount modules. It's not the most reliable of setups, and most players live in fear of glitches and crashes. On the OASYS, you hit a button or two, and everything is done; the appropriate synths and sounds are loaded, the sequencer is primed and ready to roll, and all the pre-recorded audio is sitting behind the internal mixer ready to be triggered. To be fair, this is not new — a handful of workstations offer similar facilities — but none promise to cope with everything so elegantly. Returning to the issues at hand, can we really say, therefore, that the OASYS is an open system? There is as yet no third-party developers' kit, and it cannot handle software that conforms to VST, MAS, DXi, or any of the other common 'open' application formats. Korg's stated view is that the OASYS is as open as (for example) Digital Performer or Pro Tools, which require developers to write to the MAS or Audiosuite standards respectively, but I fear that this is a tad disingenuous. There are thousands of DP systems and hundreds of thousands of Pro Tools systems in use worldwide, which makes it viable for small companies to develop for them. It's daft to imagine that Korg will sell many thousands of OASYSs, so it's much less likely that third parties will develop for it, whether it's 'open', or not. This places a heavy burden upon Korg to provide new synthesis engines and other upgrades, or they may find that the lessons of the Roland VariOS come back to haunt them.

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Big Screens, Little Screens Some people on Korg-oriented web forums have already started to speculate about using a large screen with the OASYS, and in principle, this is possible. If you look at the back panel, you'll find a blanking plate that covers a standard video output: remove this, and there's nothing stopping you from connecting an external monitor. Of course, disconnecting The multiple USB connectors on the back the touch-screen would make it panel are accompanied by this mysterious impossible to edit the OASYS blanking panel, behind which lurks a although, alongside the video standard video output connector. output, there are several USB2 ports, so it should also be possible to connect a mouse or trackpad (or both). For the moment, however, all this is moot, because the OASYS's operating system does not support it. However, when I asked Korg whether they intended to provide support for external screens and control devices, the answer I obtained was encouraging. They said, "If enough people want this, we'll look into it". Time will tell.

HD1 As things stand, there are three sound engines in the OASYS. The first is a PCMbased engine called HD1 which, contrary to some of the uninformed comments you may have read on the Web, is far more than just an expanded Triton. The other two engines are expansion instruments, or 'EXis': the AL1 analogue modelling synth, and the CX3 organ. Despite the fact that they come pre-installed from day one, Korg's use of the term EXi for the AL1 and CX3 demonstrates that they see HD1 as the heart of the OASYS. This makes sense; workstations are invariably based on S&S, no matter what other facilities they host. Currently, HD1 lives inside 1GB of RAM and, although there's a slot on the motherboard for a further Gigabyte, the OS is as yet unable to access it. Apparently, this upgrade is high on the list of Korg's priorities, and that's just as well, because the current arrangement imposes some unhealthy constraints on how you can use the system. When you switch on OASYS, it loads approximately 303MB of PCM samples into its memory. Korg refers to this as the 'ROM' but, of course, it's not. The samples are held on the internal 40GB hard disk, so it's possible for the company to update or replace them at any time it chooses. Next, the OASYS can load one of two choices of EXs (EXtended sample) memory. EXs1 is another selection of multisamples occupying 314MB, while EXs2 is a grand piano (see the box over file:///F|/SoS/SoS%2011-2005/korgoasys.htm (9 of 14)10/19/2005 9:41:30 PM

Korg OASYS: Part 1

the page) that takes up a whopping 504MB. Ah, you're ahead of me... If you try to load all three sets of samples, you need approximately 1.1GB of RAM, and the OASYS has only 1GB. This means that your options are (i) the 'ROM' alone, (ii) the ROM plus EXs1, or (iii) the ROM plus EXs2. Any unallocated RAM (up to a maximum of 500MB) is then made available to the sampler, which we'll discuss next month.

The main screen for viewing an HD1 patch, with the oscillator, LFO, filter and envelope configurations all on display.

While this system doesn't sound too bad, there's a problem. Firstly, if you want to switch between EXs1 and EXs2, you have to make your selection in the Global menu, switch the OASYS off and then back on again! Now, let's say that you've create a set of Programs that use the samples in EXs1. And let's say that you've placed them all in program Bank E so that you can find them quickly and easily. And let's say that you now want to load EXs2 to play the grand piano. Inevitably, every patch in Bank E is 'broken'. Likewise, you could program your piano sounds using the samples in EXs2, but these will go horribly silent if you invoke EXs1. Let's face it... that's clunky. At some point, someone within Korg must have decided it was unacceptable to have the piano sounds mutually exclusive from much of the rest of the OASYS, so the designers took EXs2, shortened the loops and used fewer samples to create a lesser piano that occupies about 133MB of the 314MB in EXs1. This, Korg claim, is a very respectable piano in its own right. In the version I am reviewing, 1.0.0, it loses one of the velocity splits (only the ff, f and mf samples have survived), and occupies just a single layer with no sympathetic resonance, so you would expect it to be less realistic than EXs2. Nevertheless, the difference is small. What's more, if you inspect the list of multisamples in EXs1 you'll find that the mp samples and the so-called 'Damper' sample are still available, so you can reconstruct something that sounds remarkably similar to EXs2, gain access to the other 200MB of EXs1 multisamples, and release 200MB for your own samples. This brings us to the issue of editing Programs in HD1. As I stated at the start of the review, I'm not going to describe the details of this, nor am I going to trot out the specification. But perhaps I can give you an idea of what it's like to create sounds on the OASYS. Put simply, it's great. You edit OASYS using the touchscreen to select things, and any combination of the Value slider, Up and Down buttons, or the data-entry wheel to adjust them. I found myself using the soft plastic end of a ballpoint pen to select items on screen, because this was more precise than my fingertips. Then, depending upon whether I was holding the pen as a 'righty' or a 'lefty', I could manipulate the slider and buttons to the left of the screen or the data-entry file:///F|/SoS/SoS%2011-2005/korgoasys.htm (10 of 14)10/19/2005 9:41:30 PM

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wheel to the right with the other hand. Very neat, and very comfortable. Sound design will be second nature to anyone who has programmed any previous Korg workstation. In short (but in a long sentence), two oscillators feed two filter sections that feed two amplifier sections that feed an insert effects section that feed the master The flexible AMS mixers from the OASYS's effects, the end result of which is sent HD1 synth engine, seen here in 'AxB' (left) to the outside world for your pleasure. and 'A+B' configurations. If that sounds remarkably like an M1, that's not surprising. It is, although the power of each section — oscillators, filters, envelopes, LFOs and so on — is hugely greater. So, while there may be a lot of synthesizer here, it's not mindnumbingly complex. You select pages (in effect, synthesis modules) using the two rows of tabs running along the bottom of the screen, and select parameters by touching... well, the parameters themselves. As you can see from the Program screen on the opposite page, the voice structure is surprisingly clear, and it's equally clear that a lot of thought went into the interface. For example, if you want to adjust the nature of Filter B in Filter1 of a dual-oscillator patch, you press the Filter tab, then the Filter1 tab that is thus revealed, and then the Filter B 'Type' control, selecting from the drop-down menu that appears. This may sound complex, but the beauty of the system is that, on a 10.4-inch touchscreen... it's a beautiful system! If I had to single out one aspect of HD1 that's worthy of special mention, it would be the AMS Mixers (shown in the screengrab above). The AMS (Alternate Modulation Source) concept first appeared on the Trinity, and in essence replaced the patch leads and CV inputs of a modular analogue synth, allowing you to specify which source modulated which destination, and by how much. On the OASYS, Korg have taken the concept a step further by adding two AMS Mixers in each voice (or, to put it another way, up to four AMS Mixers in each Program). These offer six modes: A+B, AxB, Offset, Smoothing, Shape, and Quantise. I particularly like 'AxB' mode, because this allows you to control the amplitude of one modulator using another, just as you would using a VCA on a modular analogue synth. Given that the outputs from the two AMS Mixers are themselves available as AMS sources throughout the voice structure, this suggests no end of programming possibilities. I would also like to compliment the OASYS on its patch-selection mechanism. Click on the Category area at the top of the Play screen and, rather than see an almost endless list of patches, you'll find a list of 17 categories (strings, brass, organs, lead synth, and so on) each with up to eight sub-categories. Click on one, then the other, and the thousands of Programs in the OASYS are narrowed down to exactly the type of sound you want, making selection painless. Bravo!

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EXs2 — Piano Lessens What constitutes a 'good' piano sound is a very personal matter, but few people would suggest that Korg have ever done other than lag behind the major players in the digital piano market. First, there was Kurzweil, and then there was Roland, and now... well, there's still Roland. In comparison with these, Korg's pianos have always sounded a little two-dimensional, and they have never achieved the same commercial success as other manufacturers' offerings. So, when Korg Japan were discussing what sounds to place in the OASYS, they decided that a superb piano was a top priority, and that it could occupy as much space as needed to get it right. The developers found that the required samples occupied more than 500MB, and thus was EXs2 born. So, how does it sound? Oh heck... I really don't want to say this, but my initial impressions were not great. At first, I couldn't easily determine what was bothering me, because the samples that comprise EXs2 are undoubtedly 'piano-y'. The recreations of the undamped strings at the top of the keyboard are good, the long decays when the damper pedal is pressed are superb, the stereo imaging is nice, and the sympathetic resonance adds a welcome layer of realism. Nonetheless, there are other areas in which the EXs2 pianos just don't sound realistic to me: for example, the lack of body in the middle octaves and the unnatural haste at which upper mid notes disappear to silence when the damper pedal is not pressed. Consequently, the EXs2 piano Programs didn't convince me, and I got a similar feeling from them as I get when playing one of Roland's early SAS (Structured Adaptive Synthesis) pianos, or even a Yamaha CP80... great instruments that you can use in the same way as a piano, but without them ever sounding like a perfect imitation of a piano. Faced with this, I removed all the effects from the Programs, inverted Korg's 'smiley face' EQ, and tweaked the curves and lengths of the envelopes a little. The sound improved considerably. Of course, the use of a conventional monitoring system ensured that I was not fooled into thinking that I was sitting at a nine-foot Bösendorfer (nor even my mum's Broadwood) but the modified Programs — although not up to the standard of the very best of the competition — now worked well. I am sure that all but the most demanding listeners would be happy to hear them in a mix and, with sympathetic playing and recording, many — myself included — could be fooled into thinking that they were listening to a 'real' piano. Of course, you may like Korg's Programs without having to go through these kinds of tweaks. As I said, piano sounds are very personal.

And The Sound? As you are probably aware, the OASYS offers extremely advanced effects, wave sequencing and KARMA to further spice up its sounds, and we'll discuss all of

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these next month. But I'm sure that you're dying to know how the HD1 Programs sound without all this fairy-dust being sprinkled across them because, unethical though the practice may be, the deficiencies of many a dull synthesizer have in the past been masked by built-in effects. In short, HD1 can sound superb. Traditionally, I have used four brands of synths for specific characters of sounds. Kurzweils have provided orchestral sounds and the warm, low-end timbres for which they're famous, Yamahas have provided precise, percussive sounds, Rolands have provided the pianos, ethnic instruments and pads that sit unobtrusively in a mix, and my Korgs have always been used for the wonderful washes and textures that they do better than anything else. The OASYS breaks the mould. Its orchestral sounds are a revelation: more authentic and with more character than any previous Korg. Its percussive sounds are brighter and more precise than any previous Korg. It is capable of ethnic sounds and pads that work better than similar sounds on any previous Korg. As for the spacey textures and effects... well, you know what I'm going to say. In part, this is because many of its PCMs are stereo, uncompressed and longer than those of other workstations. In part, the high internal processing speed means that the PCMs can be manipulated with high bandwidth and high precision, so they are clearer at high frequencies than ever before. In part, it's because its huge polyphony consigns note-stealing to history when using Programs. Furthermore, Korg's claims that the OASYS generates very little aliasing appear to be well-founded. Take something with a rich, highfrequency spectrum, and play it at the top end of the keyboard. Now apply an LFO to the pitch, and an octave of upward pitch-bend. On most synths, you would now be hearing all manner of unwanted frequencies polluting the sound. On OASYS, you hear little or none of this, and that's impressive. However... much less impressive are the obvious multisample points on many of the sounds. To demonstrate this to yourself, select patch INTA024: 'Pipe Organ Mixture' or INTA032: 'Clav1/Mure Sw1', and play middle 'C'. Now play the 'B' a semitone below it. As you will hear on both Programs, the timbres change considerably. If you want to shock yourself, isolate multisample 0072: 'Pipe Organ Mixture2' in the former Program. Alternatively, play A4 and A#4 using 0116: 'Tin Whistle' — the two notes sound as if they come from two different types of instrument. To be fair, generating large libraries of multisamples is difficult, and there are many examples of other manufacturers' instruments that suffer from the same

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problems. The silver lining here is that, because they are disk-based, the OASYS's sample sets can be replaced by improved versions. I hope that Korg will consider doing so. Another criticism concerns Korg's misleading claims that the oscillators (and I quote from the brochure) "support four-way layering, switching and crossfading". This is not true. While you can load up to four samples in a single oscillator, you can't layer them four deep; only two can sound at any given moment. On a more positive note, the mechanism for inserting samples and crossfading or layering them and the way that these operations are represented on the touchscreen are object lessons for other designers.

Next Month... While Korg can't claim to have invented wave sequencing (it's an extension of the wavetable synthesis made popular by PPG) or vector synthesis (this first came to prominence on the Prophet VS), it's fair to say that both these technologies reached maturity in the Korg Wavestation and its descendents. Happily, both these technologies are present in the OASYS, and in much improved forms. Similarly, the Korg M1 may not have been the first synth with splits and layers, but it cemented the concept of the Combi (in other words, a selection of sounds distributed across the keyboard as required) whereupon other manufacturers adopted it as the ideal way to create complex layered sounds. Again, this technology is present in the OASYS, and again, it's more advanced than before. However, I'm not going to say any more about these just yet because they are not used only by the HD1 engine, but also by the AL1 and CX3 engines. So this is where we'll pick up the story next month. Until then... Published in SOS November 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

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Lexicon MX200

In this article:

Lexicon MX200

Simple Operation Multi-Effects Processor Effect Algorithms Published in SOS November 2005 Reverb Options MX-Edit Software Print article : Close window The MX200 In Practice Reviews : Effects MX200 Audio Specifications Sounding Out

Lexicon MX200 £200 pros Good effect quality for the price range. Very easy to use. Can generate two effects at once with four routing options. Included software allows patches to be edited and automated within a Mac or PC sequencer and patch data is also remembered within the project file. Costs less than many reverb plug-ins.

cons Some limitations in the implementation of automation in Logic.

summary The way Lexicon have gone about making the MX200 appear as a DAW plug-in for control purposes is both simple and practical. You need to ensure you have enough spare I/O to handle the audio, but other than that it is very easy to use. DAW users who lack the CPU power for serious software reverb will find the MX200 particularly attractive.

information £199.99 including VAT. Harman Pro UK +44 (0) 1707 668222. +44 (0)1707 668010. Click here to email

Lexicon's latest hardware reverb is designed to be as easy to control from your computer as a plug-in. Paul White

For years now we've become used to software plug-ins that look like hardware, but the Lexicon MX200 turns this idea on its head by being a hardware effects box that tries to look like a plug-in. What this means in practice is that the front-panel controls Photos: Mike Cameron of the machine can be accessed via a plug-in window, and settings can be The MX200 hardware (below) can be controlled from your sequencer via its control saved as part of your DAW song as plug-in (above), here seen running under with any other plug-in. This happens Apple Logic Pro. courtesy of a direct USB connection that functions as a MIDI port, though the audio still has to be connected to your audio interface in the usual way — it would have been neat if the audio could also have gone via USB, but no luck there. The MX200 features the classic Lexicon reverb sound as well as providing additional effects (32 effect types in all), where two effects can run at the same time under one of four routing options. These may be accessed through very simple front-panel controls or using the MX-Edit editor/librarian software that works within VST or Audio Unit hosts. This way the MX200 is seen by the host software as a plug-in, even though it is really externally connected hardware. However, you can only open one instance of the plug-in at a time, of course, because there's only one MX200. In addition to being able to control the effect parameters in the same way you would with a plug-in, you can also automate the control settings and save or load patches. MX-Edit is included with the MX200 and runs on both Apple Mac or Windows XP systems.

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Lexicon MX200

www.harmanprouk.com www.lexicon.com

Test Spec Apple Mac G5 dual 2.5GHz with 4GB RAM, running Mac OS 10.3.9.

The 1U rack processor is powered from an included AC power adaptor. It has stereo balanced inputs and outputs on TRS jacks and can also handle S/PDIF I/ O. Alongside the USB link are conventional MIDI In and Out/Thru ports. When an external digital input is connected to the unit, the MX200 expects the external device to be the clock master.

Simple Operation To keep things simple, a front-panel LED matrix uses four columns of lights to show which two of the 32 available effects are active. There are separate control areas for the two effect processors with buttons for Effect Select, Tempo (for tapping in tempos directly), and Bypass. Three parameter control knobs adjust the three most important parameters for the currently selected effect, and further global controls adjust Input Level and Mix 1 and Mix 2 wet/dry adjustment, as well as allowing the storing or auditioning of patches. To help audition patches, there are five audio samples that can be fired off as source material. The large dial on the left selects effects programs, and it is pushed to load them. Lexicon have created 99 presets to get users off to a flying start, but there's room on board to save a further 99 user patches. As this is a budget unit, the display offers two-digit numbers only, with no opportunity to name patches, but this isn't unreasonable on a processor at this UK price point. The two effects engines can be used in four different routing configurations categorised as Dual Mono (two independent mono-in, mono-out processors), Cascade (stereo series), Dual Stereo (stereo parallel), or Mono Split (dual monoin, stereo-out processors with mixed outputs). Two banks of factory patches have been set up, one Series and one Parallel, where the Parallel bank is designed to be used with mixer sends and returns, providing a 100-percent wet signal at the MX200's output. Conversely, the Series-bank effects are set up to be best suited for use in insert points, where the wet/dry balance is set as required using the Mix 1 and Mix 2 controls. The default bank is the Series bank, so to switch to the Parallel bank you need to go through a short routine initiated by pressing the Store and Audition buttons together, as described in the manual.

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Lexicon MX200

Effect Algorithms Small Hall.

Reverse.

Arena.

Mod Delay.

Phaser.

Large Hall.

Vocal Hall.

Spring.

Reverse Delay.

Tremolo/Pan.

Small Plate.

Vocal Plate.

Compressor.

Rotary.

Large Plate

Drum Hall.

De-esser.

Vibrato.

Chorus.

Pitch-shift.

Flanger.

Detune.

Room.

Drum Plate.

Chamber.

Ambience.

Gated.

Studio.

Studio Delay. Digital Delay. Tape Delay. Pong Delay.

Reverb Options Clearly the Lexicon reverbs are stars of this particular show, with 16 variants on offer including some nice short plates, chambers, and room ambiences. Of the remaining 16 effects, there's a dynamics section designed by Dbx as well as all the common modulation/delay treatments and rotary speaker emulation. Pitchshifting is also catered for, along with reverse delay and de-essing, making this a real processing toolbox rather than a simple one-trick pony. Given the low cost, you may be wondering what the catch is, but other than a maximum working sample rate of 48kHz and the simplistic display, I can't see it yet if there is one! Further global settings enable the user to set the MIDI channel on which control data will be sent, or to disable MIDI completely. You can also set whether programs load as soon as they are selected or whether changes wait until the knob is pressed. Digital or analogue input selection is another global setting, though both analogue and digital outputs are always active. Another neat trick is that you can set the unit to send the dry signal out over coaxial S/PDIF while the processed signal comes via the analogue outs — a clever option for providing monitor reverb without actually recording it. And of course there's a factory reset that restores all of the original 'out of the box' settings. It is also possible to configure the MX200's outputs to run in mono or stereo.

MX-Edit Software Once installed, the MX-Edit software lets you access the factory and user patches. Communication can be via USB or MIDI, though USB is usually more convenient, and user patches from the MX200 can be retrieved by the software. New effects programs can be set up using the software via its graphical panel interface, and you can also create archives of all your user patches.

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Lexicon MX200

Separate VST and Audio Unit software allows the MX200 to function within a plug-in environment and this installs from CD along with MX-Edit. The audio routing through the MX200 needs to be set up via the DAW you happen to be using: Logic users can use the I/O plug-in, whereas Cakewalk users can add send and return effects to a buss. Steinberg's Cubase also has the option of connecting external effects. The Lexicon plug-in window is available within your plug-in list so you can open this in the channel that is set up to pass audio through the MX200. The session-recall and patch-saving routines then work as they would for any other plug-in, so you can guarantee repeatability provided that you make a note of the input gain setting on the MX200, as this isn't automated. The controls within the plug-in window operate in much the same way as in the MX-Edit software.

The MX200 In Practice Lexicon's software installed in one simple operation, and MX-Edit opened like any other application, its plug-in counterpart appearing in the host DAW's plug-in list. For my main Logic test, I inserted the MX200 plug-in into a vocal track and then used Logic's I/O plug-in to send audio to and from the MX200 hardware via another insert in the same track. I had no problem getting Logic to automate the MX200's parameters in more or less the same way as with any other plug-in, although it didn't matter whether I chose Latch or Touch modes for the automation, it always behaved as though it was in Latch mode. Other plug-ins inserted here behaved normally. Quite a lot of zipper noise was in evidence when changing some of the parameters (specifically pre-delay or anything else related to delay times) during playback, so clearly you need to do a few tests to see what you can easily automate and how quickly you can make changes before artefacts become audible. Reverb mix and decay time behave fine in this respect. That aside, the effects are all rather good, even though this is one of Lexicon's lower-cost processors. The reverbs may not have quite the same PCM91 or even MPX500 sparkle, but they still manage to sit well in a mix without clouding the sound, and they integrate well with the dry sound rather than sitting on top like a layer of fog. I particularly like the shorter plate and ambience treatments, though the algorithms on offer cover the full range of useful reverb types, from barely audible small-room acoustics to cathedrals. Even the spring emulation sounds really sweet on vocals.

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Lexicon MX200

The remaining effects don't disappoint either, with the reverse delay being one of my personal favourites. For the more conventionally minded, the delays, phasers, and chorus/flanger effects are right on the money. From my own viewpoint, I wouldn't use the dynamics processors for serious track processing, not because there's anything wrong with them but because an analogue unit (or a plug-in emulation of one) with good metering is so much easier to use. Having made that point, it can be extremely useful to place a compressor before a reverb (fed from a send) to pump up the reverb energy, and a de-esser can be very effective placed before a bright reverb to stop it over-emphasising sibilance in the original vocal sound. Operation is a real no-brainer, especially if you're using the software, as you can choose the effect algorithms for each effects engine by name from a pull-down menu, rather than having to step through the front-panel matrix. The legends beneath the three controls also change to reflect their function, which is clearly not the case on the hardware. In most cases, the controls stick to a predictable convention, such as reverb pre-delay, decay time, and timbre, but it's still easier when you can see what the controls do. The four routing options make this processor very flexible, and the mixed pair of mono-in, stereo-out processors is a very practical arrangement if you want to set up two send effects using only one stereo aux return. The parallel stereo mode is also good for creating spectacular spatial effects combining different reverbs on each engine, or using a reverse reverb or delay on one channel and a conventional treatment on the other.

MX200 Audio Specifications Audio inputs: quarter-inch TRS balanced (impedance 20k(omega)) or unbalanced (impedance 10k(omega)). Input level: +4dBu nominal, +20dBu maximum. Frequency response: 20Hz-20kHz ±1dB (reference 1kHz). Total harmonic distortion plus noise: less than 0.007% within the 20Hz-20kHz range. Audio outputs: quarter-inch TRS balanced or unbalanced. Output level: +20dBu maximum. Dynamic range: more than 107dBA. A-D conversion: 24-bit resolution, 48kHz sample rate. Audio processing: 24-bit resolution.

Sounding Out Aside from the small operational quirks I discovered when using the plug-in control panel within Logic, the system performed flawlessly. Being able to save tweaked effects settings within the sequencer made it almost as immediate as a file:///F|/SoS/SoS%2011-2005/lexiconmx200.htm (5 of 6)10/19/2005 9:41:33 PM

Lexicon MX200

software plug-in, but without the DSP load that good reverb invariably entails. You do have to be careful which parameters you automate and also how fast you change them, as delay-time adjustments sound very glitchy during the change, but most of the things you'd realistically like to do are possible with care. The main operational difference between the MX200 and a 'real' plug-in is that you can only use one instance of the MX200 at a time, but you can still set up two different reverbs in a parallel configuration and feed them from two sends in your virtual mixer. The sound quality of the unit is comparable with Lexicon's other entry-level boxes of recent years, which is to say exceptionally good, but still short of what a highend Lexicon box can deliver. You still get the characteristic density and shimmer when you need it, albeit without the same degree of finesse at the high-frequency end, and the additional effects cover most mixing eventualities. I've made no secret of my opinion that hardware needs to be able to function smoothly within a plug-in environment if it is to appeal to computer studio owners who take for granted that all effects settings will be saved with their sequencer projects. Lexicon have gone a long way towards getting the 'hardware as plug-in' paradigm right, and only the ability to stream audio via the USB connector would have made it better. The future of studio audio hardware is almost certainly that it will all connect to the central system via a high-bandwidth data hub of some kind — Yamaha have been working towards this scenario for many years with mLan — but for the present day Lexicon seem to have got it about right given that they need to maintain compatibility with just about any type of recording or live-sound system. Published in SOS November 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

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M&K MPS1611P

In this article:

Hardware Overview To Bung Or Not To Bung? In Use

M&K MPS1611P £1549

M&K MPS1611P Active Monitors Published in SOS November 2005 Print article : Close window

Reviews : Monitors

pros Tonally neutral but detailed presentation. Spacious stereo imaging. Flexible input options. Configurable bass response. Sealed or reflex cabinet modes.

cons

These new speakers from M&K let you switch between reflex and infinite-baffle designs on a whim. Hugh Robjohns

None.

summary A compact two-way nearfield active monitor with the unusual facility to operate either as a sealed cabinet or as a ported design, the latter offering greater SPL at the expense of some bass extension and smoothness. The MPS1611P maintains M&K's characteristic tonal and dynamic accuracy, and is an impressive addition to the product range.

information £1548.65 per pair including VAT. Stirling Trading +44 (0) 20 8963 4790. +44 (0)20 8963 4799. Click here to email

The American professional loudspeaker company Miller & Kreisel have built up a very solid reputation for their studio monitors. The unusuallooking multi-driver M&K MPS2510 flagships (reviewed in SOS July 2001) have become firm favourites in many small film and TV dubbing theatres, as well as in surround-sound remixing applications. The latest addition to the MPS range is a compact two-way nearfield monitor called the MPS1611P, and this one does things a Photos: Mike Cameron little differently from the norm too. The speaker is intended primarily for use in a 'satellite plus subwoofer' arrangement, although it can also be used as a standalone full-range speaker, albeit with a lower maximum SPL capability. The review was based on the MPS1611P monitors on their own, without a subwoofer.

www.stirlingtrading.com www.mkprofessional.com

Hardware Overview The speaker cabinet measures a neatly proportioned 320 x 213 x 308mm (hwd) and weighs a fraction over 9kg. The cabinet is ported to the rear, but is supplied with a removable plug (of which more in moment). The driver complement comprises a one-inch soft-dome ferrofluid-cooled tweeter and a 6.5-inch polypropylene woofer, with the system designed to provide 91dBSPL at one metre for a nominal-level input signal. The maximum SPL is not given in the

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M&K MPS1611P

specs, but like all M&K products, the 1611P is a very capable design and it's easily able to provide substantial levels when used in its intended nearfield role. The tweeter design incorporates an aluminium faceplate and bezel designed to help increase power handling and reduce dynamic compression effects at high volumes, while the rear of the tweeter is loaded with a proprietary 'transmission line' to prevent the sound projected from the rear of the tweeter being reflected back — something that would otherwise cause comb-filtering problems. The woofer has a generous magnetic motor design with full shielding. The rear panel of the speaker forms a chassis for the electronics — a 100W amplifier to drive the woofer and a 50W unit for the tweeter. Both are described as using 'mixed bipolar DMOS technology' and quoted with 0.02 percent total harmonic distortion (THD). The input signal can be accepted either via a balanced combi jack/XLR socket or an unbalanced RCA phono socket. There is no input selector — both appear to be active at all times — but there is an input sensitivity switch. This three-position toggle offers options of 200mV (-12dBu or -14dBV), 1.23V (+4dBu), or a variable level set by an adjacent knob. The last covers a range of ±6dB relative to the +4dBu reference level. These figures all relate to the balanced input; levels at the unbalanced input are halved (in other words, the options equate to 100mV (-20dBV), 615mV (-4dBV) and the variable ±6dB swing around -4dBV. In practice, these options make it easy to find a sensitivity setting that suits pretty much anything likely to feed the speakers.

Under the speaker's rear control panel is the single port — the plug can be removed to increase the overall output level of the speaker.

The remaining rear-panel controls are another pair of toggle switches that determine the low-frequency response of the speaker. The first switch is labelled Bass Response with Normal and Full Range options. In the first position the speaker has a claimed response of 45Hz-22kHz (±3dB), and in the second it extends down to 35Hz — although this low-frequency extension figure is based on the speaker being close to a rear wall. The last switch introduces an 80Hz high-pass filter, but this is only available when the Bass Response switch is set to Normal, and is intended for use when the MPS1611P is operating with an appropriate subwoofer. Below the large heat sink area on the rear panel is the IEC mains input connector, a fuse holder, and the main power switch. An LED next to the phono input socket provides an indication that the speaker is powered and working — there is also a recessed green LED on the front baffle above the tweeter to aid in aligning the speaker's listening axis.

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M&K MPS1611P

To Bung Or Not To Bung? The MPS1611P can be used either as a sealed-cabinet (sometimes called 'infinite baffle') design, or as a reflex-cabinet design which boasts a higher maximum SPL capability. As shipped, the speaker's rear port is closed off with a tight-fitting foam bung mounted between metal end plates, the front having a knob with which to insert and remove the bung. With the bung in place, the rear switch selector set to Full Range, and the speakers mounted in free space, the low-frequency response extends to 50Hz (6dB point) with a very gentle roll-off below that and minimal phase shift. In Normal mode the response rises to 80Hz at -3dB, and this is the standard alignment for use with an M&K subwoofer (such as the LFE4 or LFE5). If the subwoofer doesn't have bass-management facilities, then switching in the 80Hz filter introduces a steeper roll-off (-6dB at 80Hz). With the port bung removed the bass extension figures change minimally, but the character of the speaker changes dramatically, and it is capable of higher SPLs, as you might expect.

In Use The MPS1611P is a sweet-sounding speaker with a precise and neutral character, and it is claimed to be timbre-matched to other M&K monitors, although I was not able to verify that. Transients are replayed with crisp detail and natural dynamics, and with the port bung in place the speaker delivers a smooth and well-balanced bass response that manages to provide a surprising amount of information on low-frequency signals. With the bung removed the character of the bass presentation changes noticeably, but it remains a wellbehaved design overall. If the speaker was supplied only as a ported design then it would receive very favourable reviews, but as a sealed-cabinet design it gains a subtlety and smoothness in the bass region that really impresses. Personally, despite the lower maximum SPL, I much preferred the sound with the bung in place. Set up about 20cm from a rear wall, the speakers delivered a surprisingly powerful bass response, benefiting from the additional bass loading effect of the rear boundary. As long as the speakers are placed well away from the side walls they can deliver wide, stable stereo images with reasonable depth and spaciousness. I found accurate aiming of the listening axis towards the 'hot seat' was important, though, but that is made very easy thanks to the recessed LED on the front panel. With the aid of an assistant to move the speaker, the increasing apparent brightness of the LED makes it very obvious when the optimum angle and height have been found. Used farther from the rear wall, the bass response naturally becomes less powerful, but the smooth extension

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M&K MPS1611P

remains and still allows the detail, tuning, and relative balance of bass instruments to be heard clearly. Overall, these little speakers impressed me very much, delivering an accurate, tonally neutral sound with natural dynamics and transient details — even when delivering relatively high nearfield sound pressure levels. Although I was unable to try the MPS1611P with a suitable subwoofer during the review, experience would suggest that they will integrate very well, such is the controlled evenness of the bass response, in which case the combination would make a very competent and compact full-range nearfield monitoring system. Published in SOS November 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

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Plug-in Folder

In this article:

Camel Audio Camel Space TC Electronic DVR2

Plug-in Folder Camel Space & DVR2 Published in SOS November 2005 Print article : Close window

Reviews : Software

Camel Audio Camel Space Formats: PC VST & Mac OS X AU Camel Audio are a name that should be familiar to plug-in fans, having started out with the popular Camel Phat VST effect before unveiling the groundbreaking and critically lauded Cameleon 5000 additive synth. Lately, as well as updating Camel Phat to its third incarnation, they've given it a stablemate in the form of Camel Space, an interesting amalgam of a few studio staples along with some twists not found in your average effect plug-in. Like its brethren, one of Camel Space's biggest assets is its filter, which can provide 10 different modes of operation, comprising high-, low-, and band-pass algorithms in standard and 'fat' versions, along with the less common comb, notch, peaking and ring-mod types. As you might expect, 'fat' seems to denote more than just a steeper cutoff slope, instead affording the kind of grit and wallop that can make for much more pleasing filtering, not to mention the odd speaker-endangering resonant honk. It sounds great and provides a lot of versatility for broad and subtle applications, and there's a dedicated LFO to go with it. Camel Space also boasts auto-pan, enhancer, stereo delay, flanger and reverb modules that are all very straightforward and conventional in operation. The reverb is particularly simple, with just Mix and Size controls, yet sounds pretty smooth and non-metallic — particularly taking into account its very efficient performance in terms of CPU use. The enhancer is another highlight, capable of providing both top-end lift and soft saturation that can vary between a nice gentle warmth and some quite raucous distortion.

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Plug-in Folder

Whilst the various modules add up to a nice set of effects, the real heart of the beast is a step sequencer with which you can modulate the pan and filter cutoff, along with a simple one-button output attenuator named the 'trance gate'. With up to 128 steps in eight banks, the sequencer provides quite some scope for sculpting evolving rhythmic sounds, and there's control over the attack, sustain and decay characteristics as well as the ability to chain individual steps together. With different modulation destinations selectable on each of the eight banks it's an exceedingly capable tool and one that's not too fiddly in operation to put a damper on experimental fun. A run through the presets demonstrates well how powerful the sequenced filtering, panning and gating are in combination with the other modules. There are some great sounds on offer here, regardless of whether you want to make actual trance music, get seriously experimental, or just do 'How Soon Is Now' covers on the guitar. In fact, although only one of Camel Space's numerous categorised presets has the 'GTR' prefix, I found it excelled as a guitar effect. There's also a lot to help you get the most out of the plug-in's potential for real-time performance, including an X-Y pad for linking and manipulating any two rotary controls, and MIDI learn functionality for hardware control if your host application supports it. Plus, there's an invaluable randomise button that works intelligently to help create fresh — and always usable — patches when inspiration runs low. Overall, it's safe to say that Camel Audio have scored another hit, with a great plug-in that's as happy lending a little warmth and ambience as it is dealing allout swirling wackiness. Thanks to an attractive, well thought-out interface and some very convenient design features, Camel Space is a pleasure to use and would make a good addition to anyone's plug-in folder. What's more, its very nicely priced at 45 quid, so I'd definitely recommend grabbing the demo and giving it a try. Mike Bryant £45. www.camelaudio.com

TC Electronic DVR2 Formats: Mac & PC Powercore It seems that this year TC have launched a different Powercore reverb every few weeks, but to fair, they are all different and they are also of extremely high quality. The latest is DVR2, based on algorithms ported from the System 6000 processor and designed to generate what TC call 'generic reverb with true vintage flavour'. While room simulation reverbs are designed to create a specific sense of space, a generic reverb is more about adding reverb as an effect without suggesting any particular type of acoustic environment. DVR2 has clearly been designed to model as closely as possible the tonal attributes of the famed EMT 250 digital reverb processor. While EMT are best known for their plates, the all-digital EMT 250 was extremely well respected by file:///F|/SoS/SoS%2011-2005/pluginfolder.htm (2 of 4)10/19/2005 9:41:40 PM

Plug-in Folder

engineers and in some respects it was a convenient and modern replacement for their earlier plate units. It could thus be said that TC have attempted to emulate an emulation of a plate! DVR2 sounds plate-like without recreating the undesirable rumbling and ringing that is often evident in plate reverbs; it also includes internal chorus-like modulation drawn from the EMT 250, though what it definitely does have in common with plates is that there are no pronounced early reflections and the reverb density builds up quite quickly. This type of reverb works well for adding to material that has already been mixed or partially mixed or that contains a degree of acoustic ambience already. Typically it gives a flattering, larger-than-life sound with a nice gloss but without getting in the way, and it also works well on moving (or panned) sound sources. Of course what may be considered attributes in some situations are drawbacks in others, and a generic reverb such as this diminishes the stereo imaging of sources. Where the modulation is used, it can also cause problems with instruments that don't include any natural modulation, piano being the prime example. That's why there are different types of reverbs available for different jobs. To my ears, DVR2 sounds very much like a cross between an early Lexicon reverb and a vice-free plate, and gives a nice steamy sizzle to the sound being treated. TC's designers do seem to have worked hard to capture the authentic sound of the EMT 250 here, including electrical effects such as amplifier saturation and the original artifacts of the converters used at the that time. They've also worked to get the parameters to interact as they do in the real thing. DVR2 comes with a small but useful set of presets that can be easily tweaked by the user, and DVR2 presets can also be imported from the System 6000 platform. Compared with most software reverbs, the controls are very simple, affecting mainly the predelay, decay time, high cut and high/low damping of the reverb tail. A single slider controls the modulation and the high cut control also has a Q parameter for emphasising frequencies at and around the cutoff point. Two further buttons bring in emulations of the input transformer and low-end cut of an original EMT 250, while the Vintage Reset button claims to produce a sound that comes very close to that of a well maintained EMT 250. In high-resolution (non-vintage) mode, there's lower noise, wider bandwidth, and user control over the degree of modulation. My tests with this reverb largely supported TC's claims, insomuch as I wasn't immediately transported to a club, a concert hall or a cathedral, but the reverb provided sympathetic musical support to the material, particularly to vocals. It has the airy, larger-than-life quality that made the original EMT 250 so popular, and because the EMT 250 didn't set out to capture the less welcome aspects of real

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Plug-in Folder

plates, TC have not needed to model all the side-effects that engineers used to try desperately to EQ out of their plate reverb returns. As an addition to the existing TC Powercore reverbs, DVR2 works extremely well and provides a good alternative for material that would traditionally benefit from a plate reverb. Paul White £313 including VAT. TC Electronic UK +44 (0)800 917 8926. +44 (0)800 917 6510. Click here to email www.tcelectronic.com Published in SOS November 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

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Rayzoon Jamstix

In this article:

Recommended System Requirements Flavours Of Jam Jam Today Conclusions

Rayzoon Jamstix Virtual Drummer PC VST Instrument Published in SOS November 2005 Print article : Close window

Reviews : VST Instruments

Rayzoon Jamstix $99 pros Real-time response to MIDI or audio input has considerable creative potential. Once mastered, can produce some very convincing results.

Virtual band members are now a regular feature in SOS reviews, but how about a virtual drummer that actually will jam with you? Rayzoon claim Jamstix is able to do just that.....

cons Does take some time to learn. Current sounds and styles less diverse than some of the obvious competition.

John Walden

Pretty soon, SOS readers are going to have to carry out extensive auditions if they want to assemble a virtual band. summary The most recent candidate for the Uniquely, Jamstix allows you electronic drumstool is Jamstix from to put the phrases 'virtual band member' and 'interesting Rayzoon, which approaches the virtual and occasionally surprising drummer idea from an interesting playing' into the same angle, the unique feature being its sentence. Rock, blues and ability to jam 'live' with either a MIDI or funk fans should check out audio input, responding to what you the Jamstix demo for a costplay in the same way that a human effective route to some new drumming inspiration. drummer might. Rayzoon have (thankfully) failed to replicate some of information the other behaviours normally associated with real drummers — but just how $99 (approximately £57). good is a Jamstix jam session? Click here to email www.rayzoon.com

Test Spec Jamstix v1.2. PC with 3.2GHz Pentium 4 CPU, 2GB RAM, Echo Mia 24, Egosys Wami Rack 24 and Yamaha SW1000XG soundcards, running Windows XP Pro SP2. Tested with Steinberg

Jamstix is available for on-line purchase directly from the Rayzoon web site and although it has been tested with a wide variety of hosts including Cubase SX, Sonar, Ableton Live and Tracktion, Rayzoon advise potential purchasers to download the demo version prior to purchase. This has a number of limitations compared to the full version, such as random audio dropouts, but does allow for compatibility testing within a user's own system. You will need a broadband connection, as Jamstix is a large download. Installation proved straightforward on my test system, and Rayzoon supply a licence key via email once a purchase has been completed. Within a couple of minutes, I was up and running with Jamstix within Cubase SX.

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Rayzoon Jamstix

Cubase SX 3.0.1 and Groove Agent 2.0.

Jamstix includes a high-quality set of drum samples, featuring up to nine velocity layers and both close (mono) and ambient (stereo) mics that can be blended to taste. The range of drum kit styles is not extensive, perhaps suiting pop, rock and funk genres best. However, Jamstix provides a variety of ways to route its playing to alternative drum modules such as Steinberg's Groove Agent, Toontrack's DFH or FXpansion's BFD to trigger a different set of sounds. Jamstix also includes a large collection of drum patterns, intro, fills and endings to suit a range of styles. These do perhaps not cover such a wide musical territory as those found in Groove Agent but, as we'll see, a key feature of Jamstix is its ability to introduce a range of variations to those patterns much like a real drummer would do. Aside from the ability to jam, other features include a 'limb priority control' algorithm that means Jamstix will only attempt to play a combination of drums that is physically possible for a real drummer and an Arrangement screen that allows patterns to be sequenced on a bar-by-bar basis in much the same way as audio or MIDI loops might be within a sequencer application.

Recommended System Requirements Pentium III or Athlon 500MHz CPU (2GHz+ recommended). Windows XP or 2000 (Windows Me and 98SE are not supported but should work). 512MB RAM. 500MB free hard drive space. VSTi 2.0-compliant host.

Flavours Of Jam As shown in the various screenshots, the Jam, Rhythm, Arrangement, Mapping and Output tabs hide the large number of features hidden under the Jamstix hood. Three 'jam' modes are provided: Manual, Free and Keyword. In Manual mode, the Play Input button allows Jamstix to be played via a MIDI keyboard like any other drum module. Alternatively, any of the preset rhythm patterns can be loaded via the Rhythm tab and sequenced via the Arrangement tab; a maximum of 16 patterns can be loaded at any one time, although for reasons described below this is not as restrictive as it might sound. Intros, fills and endings can also be set within the Arrangement screen, so complete drum sequences can be built from scratch. Things get further off the beaten track in Free Jam mode. With Free Jam selected, Jamstix will, in essence, make up its own drum patterns on the fly and will adjust its playing to match aspects of your own playing, either in response to MIDI or audio input. Both can be live or from a pre-recorded track and a second AudioM8 plug-in is supplied. This is placed as an insert on the required audio

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Rayzoon Jamstix

track and routes the signal to Jamstix. With an audio input, Jamstix responds to the changes in volume (velocity) of the signal, playing more busily and louder at higher input volumes. With a MIDI input, the velocity response is similar but the rhythmic and harmonic content are also used to introduce some variations to the drumming. At the same time, the Style and Control settings can be adjusted to introduce all sorts of random variations much as a real drummer would. For example, Individual patterns can be customised within the degree of Funkiness, Complexity the Rhythm screen. and Accentuation can be adjusted within the Control section. The freshly created patterns are automatically placed within the Arrangement screen, so if you create either individual patterns or sequences that hit the spot, these can subsequently be fine-tuned and re-sequenced by switching to Manual mode. Key Word Jam allows the user to pre-select a number of the built-in rhythm patterns on the basis of key-word descriptions. These patterns are loaded into the Arrangement screen and can then be arranged in Manual mode; alternatively, Jamstix can arrange them automatically in response to a 'jam' against the audio or MIDI input as in Free mode. Again, the Control options allow the degree of variation around these patterns to be set. My initial 'jamming' felt a little clumsy: Jamstix varied its volume too much and its playing became both too busy and too loud at high input volumes. It took some time experimenting with the various controls to gradually achieve what I was after. There are aspects of the user interface and the manner in which Jamstix functions that are a little quirky, but it is certainly worth persevering: for rock, blues and funk styles, some of the results were truly excellent. The sensation of having a virtual drummer respond to the dynamics of your playing is quite remarkable, and Jamstix throws in some genuine surprises that are very unmachine-like.

Jam Today Jamstix also boasts plenty of other control options, some of which are worthy of particular mention. For example, the Arrangement screen is very straightforward to use, and right-clicking on a particular bar allows an intro, fill or ending to be inserted at a particular bar. Fills sit on top of the basic pattern, with the limb priority control ensuring things are kept within the bounds of possibility. The Toggle Locks button allows the arrangement to be locked once you have created your sequence. Within Cubase SX, arrangements can also be saved for later recall using the Save Bank option.

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Rayzoon Jamstix

One possible criticism of Jamstix is the rather narrow range of acoustic drum samples included; while their quality and velocity response is excellent, for dance, hip-hop and electronica styles, a wider palette (such as that included with Groove Agent) is really required. However, the Mapping screen includes a Sounds panel, from where an external drum module can be selected. In essence, that drum module is loaded into Jamstix and the Jamstix performance is then routed to the drum module before being returned to Jamstix for output. As shown in the screenshot above, I tried this with Groove Agent and it worked flawlessly. This offers instant access to any other drum sounds you already have available, which is very neat. The Output screen does pretty much what you would expect. Four outputs are provided, each with its own compression controls. If you are using Jamstix's own internal drum sounds, these can be routed to any of the four outputs within the Mapping screen, and the amount of ambience added from the stereo room mics can also be controlled from the Output screen. This is very effective in moving from a dry sound through to a more 'live' Bonhamlike sound.

The Arrangement screen does exactly what it says on the tin.

The Mapping screen allows the velocity response of Jamstix relative to the input velocity (either MIDI or audio) to be adjusted for each drum or globally.

Finally, it is worth mentioning that once you have your drum sequence completed, Jamstix can output it as a MIDI file. This is useful if you want absolute control on the level of individual hits, or you need to move to a different working environment. Jamstix's four outputs all feature compression and the amount of stereo mic room ambience can be adjusted to taste.

Conclusions Jamstix adopts a very different approach to virtual drumming from that offered by Groove Agent. The key feature of live response is unique as far as I'm aware, file:///F|/SoS/SoS%2011-2005/jamstix.htm (4 of 5)10/19/2005 9:41:45 PM

Rayzoon Jamstix

and is both good fun and very creative, although it takes some experimentation to get the best from it. Jamstix doesn't perhaps offer as diverse a range of drumming styles or sounds as GA2 and is most certainly more complex to learn, but it will occasionally surprise you by adding some distinctly human-like touches to its playing. Given the low price, for those with an interest in rock, blues and funk, Jamstix is well worth considering: download the demo and give it a through test run. Rayzoon have a virtual bass player, Jambace, in development and, on the basis of what they have achieved with Jamstix, I'll be very keen to see what that has to offer when it is released. Published in SOS November 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

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Sample Libraries: On Test

In this article:

Sample Libraries: On Test

Reason Drum Kits ***** Beats Working In Cuba ***** Hot Releases Assessed Published in SOS November 2005 The Giovani Edition *****

Star Fifth Beatles ***** George Martin

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Reviews : Sound/Song Library

**** Brian Epstein *** Stuart Sutcliffe ** Phil Spector

Reason Drum Kits *****

* Yoko Ono

REFILL On the evidence of this huge and detailed Refill, it seems clear that Propellerhead see their prime product, Reason, as much more than a loop-based dance tool. As you might guess, the focus is on the drum kit, as in kick, snare, toms, hi-hats, and cymbals. (Oh, and hand claps and finger snaps.) Names to drop here include Yamaha, Ludwig, Pearl, Zildjian, Meinl, and Paiste; if you need to know exact drums and head sizes, the printed manual reveals all. The kits were set up at Stockholm's Atlantis Studio. Classic mics recorded the drums to analogue tape, after which the lot was transferred to a Pro Tools HD system at 176.4kHz/24-bit resolution. The Refilling process has resulted in a well-recorded, well-edited, well-organised collection of NNXT sampler patches, both of full kits and of individual drums from which you can assemble custom kits. The set also features some simpler offerings for the Redrum drum machine, plus a smattering of RV7000 reverb and Scream 4 distortion patches. (There are no Combinator patches, as this library was released before Reason 3.) After the drums, the room, and the recording, the big issue here is multisampling, or Hypersampling as Propellerhead put it. NNXT's velocity-fired layering abilities are pushed into overdrive, and you soon learn not be surprised at seeing 14 samples assigned to a key. In addition, multi-mic setups have been faithfully reproduced, so you'll be able to mix close-miked main drums with overhead cymbal mics (complete with spill from the kit) and ambient mics that add the sound of the main room. Thus, the 14-layer sample just mentioned would turn up three times — once up close, once from the overheads, and once from the ambient mics — yet be triggered from one key. It's up to you to mix the result, via the NNXT's individual audio outs, which is a doddle courtesy of standardised routing and keymapping throughout (a handy little chart serves as an aide memoire).

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Sample Libraries: On Test

If all this sounds like it could become a little sample heavy, you'd be right — there are over 10000 samples on the installation DVD. And Propellerhead are ahead of you: stereo variants of kits and individual drums are joined by close-miked and 'no ambience' varieties, so you're not committed to maxing out your RAM and CPU load. The Redrum patches seem a bit token when compared to the NNXT offerings, but are welcome nonetheless. In addition, the whole lot is supplied as independent 16-bit and 24-bit Refills — the latter nearly 2GB in size — the contents of which are otherwise identical. It's impossible to pick out favourite kits or sounds — it's all good, if a little samey after an hour or two of auditioning. Even Propellerhead's grouping into pop, rock, and vintage soul/funk kits seems arbitrary. Familiarise yourself with what's on offer and choose the kits or drum sounds that suit your session. Then fix it in the mix! Derek Johnson Reason Refill DVD-ROM, £79 including VAT. M Audio UK +44 (0)1923 204010. +44 (0)1923 204039. www.maudio.co.uk www.propellerheads.se

Beats Working In Cuba ***** INTAKT INSTRUMENT This is another of Zero G's sample libraries based around an Intakt instrument front end — for information on this software's functionality, have a look back at the East West/Zero G instruments review back in SOS February 2005. As suggested by the title, this particular collection is a percussion and drum library, and the 'in Cuba' tag is a genuine one — all the recording was done in Estudios Abdala, a modern recording studio located near Havana, Cuba, and performed by established Cuban musicians. Indeed, full details of the recording process are provided in a DVD video that is included within the package. This is a really nice bonus and demonstrates the authenticity of the sounds and playing included within the collection. This is complemented further by a detailed HTML-based document which, given the size of the library, is very helpful for the new user trying to find their way around the contents. The sample content is provided on two DVD-ROMs and occupies approximately 8GB of hard drive space. The samples are dominated by multitracked loops but there are also a large number of individual drum hits supplied for those users who want to build their own loops or augment those provided. The library covers what are considered the dominant rhythm types found in Cuban music: Cha Cha Cha, Danzon, Son Montuno, Bolero, Pilon, Son Traditional, Guajira Son, Guaracha, Mambo, Songo, Timba, Conga Habanera (Carnival), and Conga Moderna. Some of the styles contain additional variations, so they can be played in a 'modern' (often with a drum kit) or 'traditional' (without a drum kit) format. The file:///F|/SoS/SoS%2011-2005/sampleshop.htm (2 of 5)10/19/2005 9:41:50 PM

Sample Libraries: On Test

single hits aside, most of the loops were recorded as an ensemble — to allow all the performing musicians to get into the groove — but with multiple microphones set up. As a result, in addition to pre-mixed stereo versions, the collection includes the outputs from each close mic and a number of 'room' mics that could be used to add ambience or to create a surround-sound configuration. A minimal amount of EQ or compression was used in the recording stage, so users can add their own to taste. When first starting to explore this library, the sheer number of Intakt patches is quite daunting. However, the contents are organised in a consistent and logical fashion. At the top level, all the material is categorised by rhythm style (Cha Cha Cha, Bolero, and so on), and within each of these categories are between 20 and 50 further subfolders. Each of these contains all the samples and loops for an individual groove arranged as a number of different Intakt patches. The first of these is always based upon a stereo mix, the full stereo loop being generally mapped to C1 using Intakt's Time Machine mode, so that it will sync to the tempo of the host sequencer. Other samples are mapped onto keys above this, and these generally include some combination of the main individual elements of the loop (for example the congas or timbales), the 'room' microphones, or, for those styles where it is appropriate, the 'modern' and 'traditional' alternatives. This arrangement works well, because, as well as giving access to the full loop, it is also possible to break it down and construct variations on it using these key elements. The bulk of the remaining Instruments within these subfolders contain each individual percussion element from the particular groove presented in Intakt's Beat Machine mode, with slices mapped to a series of keys. In use, the considerable lengths to which Zero G have gone in the recording process really do pay off. While the audio quality is very good, most importantly so are the grooves. I particularly liked those in the Bolero section, which had a lovely slow, lazy (even sexy) feel. In contrast, the more up-tempo Mambo loops had plenty of flair and a much 'hotter' vibe. I also enjoyed the Son Montuno and Congo Habanera grooves, although in truth there was little not to like — these types of rhythms are just made to dance to, and if this collection doesn't make you want to dance around your studio then, frankly, you should take up stampcollecting or train-spotting! On the downside, I would have really liked an additional single Intakt patch for each of the major musical styles, containing all of the stereo loops for that particular style mapped across the keyboard in Time Machine mode. This would be ideal for quick auditioning (much quicker than having to load each loop, in turn, into Intakt) or for occasions when mixing and matching stereo loops is all that is required for the project in hand. In addition, for the Time Machine-based loops I think the majority of users would prefer to see Intakt's Loop and Legato modes engaged by default (it is in just a few cases, but far from all). When this is done loops keeps playing as long as a key continues to be pressed, and any

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Sample Libraries: On Test

loops triggered play back in sync with each other. The user can change these settings, but it is a bit frustrating to have to do it on every occasion. Both of these issues could easily be addressed by an update to the Intakt patch files, so maybe Zero-G should consider offering this as an update. These minor niggles aside, there is no doubting either the quality or authenticity of the material itself. Beats Working In Cuba was obviously quite a significant undertaking for its producers and that, and the quantity of sample material, explain the fairly hefty price tag. As a result, and given that the size of the collection owes much to the multitrack format, this is probably less likely to appeal to those producers just looking for a few Hispanic loops to spice up their tracks. However, those specialising in Cuban music styles and who want their samples to be as close to the genuine article as possible will probably find this an essential purchase. John Walden Intakt Instrument (including VST, DXi, Audio Units, RTAS, and stand-alone versions), £249.99 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.zero-g.co.uk

The Giovani Edition ***** MULTI-FORMAT Bela D Media built their reputation on the excellent Diva solo soprano library, and this new library once again aims for the unique, offering a library purely of children's voices. Girls' and boys' choirs are recorded in 24-bit on separate DVDROMs, each having Ensemble sections of sixteen voices and Chamber sections of four voices. Both sections boast sustain, staccato, pads, and so on, and they're packed with controllable articulations such as glottals, using release triggers and crossfades. There are also phrases — from theological 'Agnus Dei' to surreal 'Mary, Mary quite contrary'! Finally you get a series of patches designed to take advantage of the bundled Vocal Control software. Hooked up to a sequencer, this utility allows for precise manipulation of phrases. A table allows you to construct up to four chained phrases, adjusting the time spent on each syllable and the point at which one phrase switches to the next. Whilst not as comprehensive in terms of phonemes as East West/Quantum Leap's Symphonic Choirs, it's a far more accessible and user-friendly approach that is surprisingly effective. A built-in Legato tool strings notes together and slurs up and down as required in a phrase. So how does it sound? In a word, natural. It's not clinical, it's not hyper-accurate, but it is, above all, moving. Considered use of the close and ambient patches can have shivers running down your spine with a single note. Never have I heard as file:///F|/SoS/SoS%2011-2005/sampleshop.htm (4 of 5)10/19/2005 9:41:50 PM

Sample Libraries: On Test

much emotion from a choir that I wasn't conducting live myself! Whether soaring above an orchestra or blending into the background, the voices exude pathos. The playable range is restricted, as befits the vocal capabilities of the girls and boys in question, and there's no oversampling or pitch-shifting at all — the intention was to construct a pure library, not in terms of perfect pitch or harmony, but rather in terms of the naturalness of the sound. Some users might frown upon hearing one or two singers lagging behind the rest in some of the more poignant phrases, but as the session conductor was heard to quip, 'If there is a choir who can sing at perfect zero all the time, firstly give me their number; and secondly you can't afford to sample them!' However, far from being detrimental, this adds to the natural feel of the library. All this realism will be off-putting for those that want absolute perfection in every note, but if all music were perfect every session muso would already have been replaced by a synthesizer. To pitch-correct every note of a choir would be rather like strictly quantising a jazz drummer — it's just not the done thing! Pardon the paradox, but it's the impurities that make it pure. I can't be the only one thinking this way, as the first production run sold out in four days! My biggest gripe with The Giovani Edition is that it sounds so good you can't help but cry out for more — namely adult voices. Thankfully, Bela D Media are hard at work on a sister library featuring adult choirs. The two together will be a powerful combination, and now that my appetite has been whetted, I can't wait! Hilgrove Kenrick Gigastudio 3 or Kontakt 2-DVD-ROM set, £229 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.beladmedia.com Published in SOS November 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

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SE Electronics Titan

In this article:

Titan Trials Verdict

SE Electronics Titan Multi-pattern Condenser Microphone Published in SOS November 2005

SE Electronics Titan £999

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Reviews : Software

pros Flattering, musical sound. Good transient response without harshness. Distinctive appearance should impress the clients! High-quality packaging and shockmount.

The capsule within this microphone has a diaphragm coated with titanium, rather than the usual gold. We find out whether it makes a difference.

cons Because the mic seems to pick up more room ambience in cardioid mode than the other models I compared it with, more care may need to be taken over the local acoustic environment when recording.

Paul White

Most large-diaphragm condenser microphones doing the rounds today are based around a gold-coated mylar diaphragm, where the gold is either evaporated or sputtered onto mylar film prior to capsule assembly. This is also true of most of the SE summary I still don't know how much of Electronics microphone range, though their new and distinctively styled multi-pattern Titan is something of this mic's sound is down to the use of titanium, but it a departure from this tradition, because it uses certainly has a subtly titanium rather than gold for the conductive flattering musicality of its own. diaphragm coating. information £999 including VAT. Sonic Distribution +44 (0) 1582 470260. +44 (0)1582 470269. Click here to email www.sonicdistribution.com www.seelectronics.com

Apparently, titanium is much more dimensionally stable than gold when subjected to temperature change. Vocal mics in particular have to exist in a fairly harsh environment, where warm, damp air is Photos: Mike Cameron being breathed over them all the time, so if the gold expands at a different rate to the mylar beneath it, then it's reasonable to assume that the mic's character will change. Another issue is that gold-coated diaphragms can wrinkle over time due to the unevenly matched expansion coefficients of gold and mylar, whereas the SE Electronics designers expect titanium to give better longevity in this respect. Beyond such matters of stability, I can't see why the choice of metal should affect the tonality of the microphone in any significant way, as its primary function is to act as a charge-carrying electrode, though the designers believe that it is a significant contributory factor. The bulk of the moving mass is made up of the

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SE Electronics Titan

mylar —the metallic coating is typically only a few atoms thick, whereas mylar membranes tend to be between three and six microns thick — so I would have thought that the mechanics would have been dominated by the physical properties of the mylar rather than its conductive coating. The casework and shockmount design are certainly eye catching, but other than the new diaphragm material the rest of the engineering design is unashamedly traditional. Centre termination has been used for the dual-diaphragm, multipattern capsule, while the onboard preamp is based around tried-and-tested Class-A FET circuitry with a transformerless balanced output stage. The mic needs only standard 48V phantom power, and there are horizontal toggle switches below the basket to bring in a 10dB pad and a low-frequency rolloff filter. A further switch selects cardioid, omni, or figure-of-eight polar patterns. According to the manual, the microphone has a particularly good transient response, so in addition to its obvious application as a vocal microphone it should also be well suited to recording percussive instruments. On paper, the Titan's frequency response covers 20Hz-20kHz, but checking the frequency plot reveals a gently rising response with a presence peak up at around 10kHz, which no doubt helps the microphone resolve transient detail. The sensitivity of the microphone is 40mV/Pa ±2dB (reference 1V/Pa at 1kHz), which puts it in about the same ball park as other studio capacitor microphones, but the 18dBA EIN, while typical of many competing products, is not nearly as quiet as the best. The maximum SPL for 0.5 percent THD at 1kHz is 128dB, which again isn't particularly high by modern standards, but should be adequate for anything other than close-miking exceptionally loud sources such as kick drums and some brass instruments. The mic's circuitry is mounted on two small circuit boards, with surfacemount components appearing alongside discrete FETs and transistors to increase the board density. The capsule, which is around one inch in diameter, is perched on an isolation mounting inside the dual-layer, spherical steel basket, while the body itself is made from machined brass. I feel the basket could have been made a bit tougher, but provided that you take reasonable care with the mic, it should stand up fine to normal usage and it could be argued that anything heavier might have interfered with the sound. A further machined locking ring secures the microphone to the included shockmount, and the whole kit is shipped in a foam-lined camera case, complete with an XLR cable.

Titan Trials I set up the Titan in its shockmount and tried it first on vocals. Its resistance to popping seemed rather better than that of most capacitor mics, no doubt due to

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SE Electronics Titan

the larger than usual basket. Because there are very few physical obstructions close to the capsule, the sound is also rather less constricted and more threedimensional than I'm used to hearing from a large-diaphragm microphone, especially when in cardioid mode, which is the most challenging in this respect. Having said that, a separate pop shield is still a good idea, as you can make the mic pop if you try. When working fairly close to the microphone, the proximity effect adds a degree of warmth without making the sound boomy or over-coloured, though the low-cut switch can be brought in if necessary to counter this. While the low end is warm and solid, the high end is open and airy, but in a very smooth way. Overall, the sound is flattering and intimate, but not at all hard or harsh. My tests with acoustic guitar were interesting, because right away I felt the mic was picking up rather more of the room ambience in cardioid mode than the other cardioid mics I was using for comparison. In the right room, this conjures up a nice sense of space, but where the room is less than ideal it may be necessary to take a little more care than usual setting up acoustic absorbers around the recording area. I can only assume from this that the cardioid pattern is on the wide side of normal, and though this means that the angle of capture is wider, it also tends to produce a more open sound with better rear rejection than a tightly focussed cardioid. The tonal change between cardioid and figure-of-eight is subtle, and moving to omni opens the sound up more, involving the room ambience to a greater extent. Finally I tested the mic on some hand percussion and found that it captured the transient character of the sound in a very convincing and natural way. As with acoustic guitar, you get the impression of accuracy, though in reality there is some subtle flattery going on. What the Titan doesn't do is add an artificial edge to transient sounds as some mics seem to do — the result is always very musical and spacious.

Verdict The Titan comes towards the top of the SE Electronics price range and is aimed at professional users and very serious project studios. I can't say for sure that the mic would sound any different if a conventional gold coating had been used in place of titanium, but, whatever the technical reasons, this microphone has a distinct and very musical character.

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SE Electronics Titan

As a vocal mic, the Titan should suit a wide range of singers and styles, as its presence peak is above that part of the spectrum responsible for harshness, while its subtle flattery really makes for a classy and intimate sound. What struck me most on a subjective level was the spacious and involving sound in cardioid mode, though the silky top end is equally impressive in its ability to open up the sound and enhance detail without adding hardness or sounding in any way brash. Overall, the Titan certainly doesn't sound like a 'me too' microphone, and I'm sure it will take its place alongside the Gemini as a serious contender that people choose because of its distinctive sonic character. Published in SOS November 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

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SSL AWS900

In this article:

What Exactly Is The AWS900? Superanalogue Circuitry Channel Facilities Cues & Auxes Meters & Routing Options Channel Fader & Rotary Encoder Master Section Assignable Dynamics Processors Monitoring DAW Control On Session With The AWS900

SSL AWS900 Mixer & Control Surface Published in SOS November 2005 Print article : Close window

Reviews : Mixer

With their latest Superanalogue console, SSL bring the sonics and functionality of their flagship SL9000K to studios with smaller rooms and budgets. In the process, they've incorporated comprehensive controlsurface facilities for driving computer DAWs remotely. But can the AWS900 really live up to its pedigree?

SSL AWS900 £58163 pros Superanalogue signal paths. Switchable E/G-series EQ. Flexible dynamics facilities. Extremely versatile signal routing. HUI-format interfacing maximises DAW compatibility. Superb DAW integration ergonomics. Comprehensive monitoring and talkback.

cons Only two channel dynamics processors. No channel surround panning facilities. AWSomation and Total Recall are cost options.

summary A fully featured SSL console in looks, sound, and facilities, but at a price that makes it affordable for mid-range studios, tracking rooms, and post houses. The superbly engineered DAW interface and integrated automation and recall facilities are the

Hugh Robjohns

The acronym SSL is likely to be recognised immediately by everyone reading this review. The Oxfordshirebased manufacturer Solid State Logic redefined the art of the large-format analogue mixing console some thirty years ago when it launched the original SL4000 desk. This console became a must-have for almost every high-end recording studio around the world, and it has maintained that position — albeit Photos: Mark Ewing with various updates and improvements over the years — virtually right up to the present day. While the SL9000 series Superanalogue console took over as the flagship product nearly a decade ago, the popularity of the venerable SL4000 has continued, and I noticed a brand-new SL4000 being readied for delivery during my recent factory visit. SSL recently passed through a rocky patch in the company's long history, but with new management and funding now in place the future is looking bright once again, with four main product lines. The flagship product remains the SL9000K Superanalogue console for music recording and production, and is accompanied by a new generation of digital consoles aimed primarily at the broadcast and postproduction sectors: the C100, C200, and brand-new C300.

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SSL AWS900

icing on an already very tasty cake.

information AWS900 console, £58162.50; Total Recall snapshot recall system, £2937.50; AWSomation automation system, £3525. Prices include VAT. KMR Audio +44 (0)20 8445 2446. +44 (0)20 8369 5529. Click here to email

The other two product lines are closely related to each other, and are both derived directly from the SL9000: the XLogic rackmount equipment, including the Superanalogue Channel which I reviewed back in SOS February 2004; and the AWS900 console, the subject of this review. In order to make the console commercially viable, a way had to be found of reducing the production costs while retaining the Superanalogue sound quality — and that meant moving to surfacemount technology and automated circuit board production. In the process of redeveloping the SL9000 circuitry for the AWS900, it became apparent that surfacemount circuit boards would also lend themselves nicely to rackmounting applications, which led to the development of the XLogic series.

www.kmraudio.com www.solid-state-logic. com

What Exactly Is The AWS900? The clearest way to look at the AWS900 is as a Superanalogue console in miniature, combined with a bespoke HUI-compatible control-surface interface. It was designed as a high-quality tracking console for use with Pro Tools DAW rigs and the like, but also serves as a superb analogue mixdown desk. It features twenty-four highly specified input channels with switchable E/G-series EQ, in-line functionality, and motorised faders. Unlike the SL9000K desks, which incorporate dynamics in every channel, the AWS is equipped with just two dynamics processors, but these can be assigned to any two channels or outputs. The classic G-series buss compressor is also included, along with very comprehensive 5.1 surround monitoring facilities. Bar-graph metering is built in for all the inputs and outputs, while the two main outputs are displayed on largeformat VU meters. In addition to the 24 input channels, there are four stereo effects returns and a pair of main buss direct (cascade) inputs, allowing up to 34 input sources at mixdown. Full snapshot recall and fader/mute automation of the console are available as cost options. The very elegantly designed and fully integrated hardware control element of the console surface is compatible with any system that supports multiple HUIcompatible surfaces, allowing the console to appear as a continuous 24-fader surface to the DAW. A similar approach is used by Yamaha with their DM2000, DM1000, and 02R96 consoles, and the necessary multi-port interfacing is currently supported by Pro Tools, Logic, Pyramix, SADiE, and Soundscape. At the present time Nuendo and some other DAWs only support a single HUIcompatible interface, and the AWS900 can only provide eight control faders with these systems. The desk and DAW are linked via multiple MIDI connections, a decision which initially seemed odd — why not use a fast USB interface? However, it was explained that MIDI is an established interface format to which the majority of DAW manufacturers conform (thus maximising compatibility), and using multiple interfaces allows data transfer rates much faster than actually required by the desk. MIDI can also be used over greater distances than USB (up to 15m instead of 5m), which is very useful when the DAW computer is located in a machine

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SSL AWS900

room away from the console. The motorised Alps channel faders and the associated assignable rotary encoders can be switched globally between controlling the console's analogue signal paths and various DAW parameters, allowing a kind of virtual in-line working practice to be set up, building up a monitor mix on the channel faders by remotely controlling the DAW's internal mixer channels. The AWS900 looks like a 'proper' SSL in every way. It works and sounds like an SSL, and the bespoke DAW interface is extremely ergonomic and very cleverly engineered so that controlling the DAW's signal-processing facilities becomes a natural extension of the console, with the same immediacy and control resolution. Of course, none of this comes cheap in the absolute sense, but it is certainly a lot more cost effective than a fully featured traditional SSL console, and represents an ideal solution for many 'tapeless' studios and smaller post houses. In fact, some of the leading studios around the world have already reequipped tracking rooms with AWS900 consoles.

Superanalogue Circuitry The AWS900 console is a single-box product which can be either built into the studio furniture or supported with an optional floor stand. The internal mains power supply has heat sinks running across the rear of the meterbridge, so there are no noisy fans. Borrowing an idea from long-established telecommunications technology, the incoming mains is converted to a high-voltage (400V) low-current supply for internal distribution, with local conversion and regulation on each circuit board. Although unusual in the professional audio world, this approach is reliable, efficient, and extremely cost effective — although it does mean that service engineers have to be as careful working on live AWS900 boards as anyone working with traditional valve equipment! With the exception of dual engineer's headphone sockets, all of the console's I/O is arranged along the back panel. Channel inputs and outputs are provided on XLR or TRS sockets, as are most of the main outputs, while the monitoring section, buss outputs, and cue/effects sends are accessed via 25-pin D-Subs in the familiar Tascam format. This enables the console to be quickly and easily installed, and also allows it to be used without a patchbay, should this be necessary. The audio circuitry is derived directly from the SL9000K series, but redesigned to use mainly surfacemount components. SSL claim that the sound quality has not been compromised in any way during this redesign, and in some cases it has actually brought extra benefits thanks to the easier introduction of ground planes and the shorter circuit track paths. It uses the same DC-servo technique throughout to avoid coupling capacitors in the main audio path, and provides the same wide bandwidth and uncoloured signal quality as the flagship console. The specifications claim a response that is 3dB down at 135kHz and 4dB down at file:///F|/SoS/SoS%2011-2005/sslaws900.htm (3 of 12)10/19/2005 9:42:01 PM

SSL AWS900

200kHz, so 'wide bandwidth' is definitely an appropriate phrase here! The desk also boasts headroom of +27dBu, distortion of 0.002 percent at +24dBu, and noise below -90dBu (providing more than a 110dB dynamic range when feeding professional A-D converters).

Channel Facilities Each channel has three inputs: mic, line, and instrument — the first two on XLR and the last via a quarter-inch socket. The mic/line selection can be controlled globally from the console's master section, or overridden locally with a Flip switch. The mic input is essentially the same as that used in the SL9000K-series consoles and XLogic channel strip, and provides continuously variable gain from +15dB to +75dB. A 20dB pad accommodates loud sources, and phantom power and polarity are switchable on each channel. The line input has a separate ±20dB gain control, and the FET-based instrument input — something no other SSL console can offer — provides a very high impedance to suit electric guitars and the like. The EQ section is, again, derived from the K-series console's four-band equaliser, including a separate variable high-pass filter. The top and bottom bands are shelving types, independently switchable to a bell shape, while the two middle bands are fully parametric. The EQ section also features switchable characteristics corresponding to the original E-series and G-series designs. The default mode provides the E-series response, featuring standard 6dB/octave filter slopes, while the G-series mode provides steeper slopes with an inverted gain region immediately before the turnover frequency. For example, dialling in a highfrequency boost to give 'air' also results in a modest level dip over the frequency region below the turnover frequency — a combination that often works very well and certainly gives a different flavour to more conventional EQ designs. Located in the centre of the EQ panel section are three buttons which switch the entire EQ section in or out, switch the insert point in and out (with separate balanced TRS sockets for send and return), and reorder the signal path so that the insert point is before or after the equaliser.

Cues & Auxes The AWS900 has flexible aux and cue-send facilities, derived once again from the SL9000K consoles. Each channel can access one of two stereo cue busses, with level and pan control and pre- or postfade selection. The send is turned on and off by pressing the level knob. There are also two aux send controls — labelled FX1 and FX2 — each being post-fade and switchable between two effect busses, giving up to four aux-buss outputs in total. Again, each send is turned on and off by pressing file:///F|/SoS/SoS%2011-2005/sslaws900.htm (4 of 12)10/19/2005 9:42:01 PM

SSL AWS900

the level knob. All these cue and aux sends can also be switched into an EFX mode, in which the relevant send output is isolated from its normal buss and routed instead to the Track Buss routing matrix (of which more in a moment) in place of the normal channel output signal (the Cue signal is routed in mono when switched to the EFX mode). Obviously, only one send control per channel can access this EFX mode at a time, but the idea — borrowed from the SL9000 console — is to allow a lot of flexibility in signal routing and output destinations while using a small control set. As an example, you could hook up the eight Track Bus outputs to eight headphone amplifiers, and thus use the system to generate an additional eight separate mono headphone feeds. Equally, you could use the Track outputs to feed additional effects processors, extending the number of effect sends. The channel's Direct output socket normally carries a post-fade signal, although it can be switched pre-fade using a button at the top of the sends section of the channel strip. However, if the EFX mode is activated on any of the Cue or FX sends, the corresponding signal is routed to the channel direct output instead. This whole EFX scheme may appear a little complicated at first sight, but it allows for some extremely versatile signal routing possibilities. The last of the channel strip's analogue controls is the main stereo routing section, which provides a pan pot and buttons to access the independent stereo Record and Mix busses. The Record buss is intended to feed the DAW while overdubbing, with the Mix buss allowing simultaneous and independent audition of the complete mix — another example of the flexibility of the desk's design.

Meters & Routing Options The meterbridge above the main channel strips carries a pair of bar-graph meters and indicators for each channel, along with various routing buttons. The console's eight Track Busses can be selected individually and are normally fed with a post-fade channel signal. However, additional buttons allow pairs of Track Busses to be fed with a post-pan (stereo) signal, or with a mono pre-fade signal. If the EFX mode is active, then the Track Busses are fed from the appropriate Cue/FX control as previously described. The bar-graph meters show either the analogue channel signal or the corresponding DAW channel, depending on the mode of the desk. Normally, only the left meter is active, but when a stereo DAW channel is being controlled, both left and right channels are displayed. Above the meters are a pair of tally lights which illuminate when the relevant DAW track is armed to record, or is currently being addressed by the plug-in editor. Four more buttons here allow one of the two assignable dynamics processors located in the master section of the console to be inserted in the channel, with

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options to insert the dynamics pre-EQ (after the channel input) or post-EQ (at the channel output). Combined with the insert point routing facility incorporated within the EQ section, these switches enable the channel signal path through the EQ, insert, and dynamics blocks to be configured in six different ways.

Channel Fader & Rotary Encoder The remaining portion of the channel strip comprises an encoder knob, the motorised fader, two four-character electronic scribble strips, and separate Solo and Cut buttons for the analogue channel and DAW channel. There is also a Select button to access the appropriate DAW channels for track arming, plug-in editing, and so on. An Auto button near the top of the fader is used to activate the various console and DAW automation modes, with adjacent red and green LEDs to show the current read and write status. The electronic scribble strips are used to show DAW track or channel names, the assigned encoder control function, DAW I/O allocations, fader and automation trim levels, and various other useful things, depending on the console mode. The default mode is for the fader to control the analogue channel level, while the rotary encoder controls the corresponding DAW channel's virtual fader. However, a Master Flip control in the master section swaps these controls, allowing a monitor mix to be built up on the faders while the console channel 'record' levels are set on the rotary encoders — the latter then being akin to the small fader on a traditional in-line console. It probably won't come as a surprise to learn that audio is not routed through the fader or encoder. Instead, both generate high-resolution control signals that are routed either to the DAW or to the channel MDAC (Multiplying D-A Converter) which essentially uses switched resistors to provide the required degree of signal attenuation. The MDAC technology was adopted because it provides better sonic performance than traditional VCAs for the console signal path. The fader and rotary-encoder data is transmitted to the DAW as a pair of Continuous Controller messages, allowing excellent 14-bit resolution, and every button sends separate control messages to indicate when it is both pressed and released. Clearly, this means there is a lot of data passing between console and DAW, but there is no perceptible control or tally-light lag at all. Everything works exactly as you would expect it to, as one perfectly integrated system with no difference in response between using a control to address a console signal path file:///F|/SoS/SoS%2011-2005/sslaws900.htm (6 of 12)10/19/2005 9:42:01 PM

SSL AWS900

or a DAW channel.

Master Section The master section of the console occupies the right-hand side and cannot be relocated to the centre of the desk, but this is not a problem, since the entire console can be reached easily from a central position, and hence there is no need to leave the monitoring sweet spot while working. The master section incorporates all the traditional analogue console facilities: main fader, buss compressor, FX and Cue send masters, monitoring controls, and so on. But there is also a full set of transport controls (with auto-locator), a comprehensive array of dedicated DAW control keys, and a colour TFT screen, of which more in a moment. Above the display screen is a panel section which includes a line-up oscillator providing one of six selectable tone frequencies or pink noise (with both preset and variable level options), routable to the Track, Record, and Mix busses. There is also a built-in talkback mic with separate output-level controls for Slate, Foldback, and Direct output, plus an input-level control for the studio Listen mic, which is processed with SSL's infamous Listen Mic Compressor. An external talkback mic can be used instead of the internal mic, if required. A quartet of LEDs indicate the presence of the critical power-supply voltages. To the right of the screen is a section devoted to the four stereo returns and the two foldback outputs. The stereo returns are usefully equipped with Width, Balance, and Level controls, in addition to separate left and right cuts, AFL, and independent routing to the Record and Mix busses. There are also facilities to send these return signals to the two Foldback outputs, with an independent Studio level control, making it easy to set up reverb in the headphones when recording vocal takes. Each foldback output is provided with Level control, cut and AFL facilities, and six source-selection buttons, allowing contributions from the two stereo Cue busses, the Record and Mix busses, the four stereo returns, and the control-room monitor outputs — all of which makes it easy to arrange zero-latency headphone monitoring for performers. The engineer's headphone outputs can be fed with the control-room stereo monitoring signal or either of the foldback signals.

Assignable Dynamics Processors Above and to the right of the TFT display are the dynamics facilities: a pair of mono multi-function processors and the stereo buss compressor. The two mono units can be linked for stereo operation and allocated to any pair of input channels (using the appropriate buttons in each channel's routing section) or to the Record or Mix busses using a clever matrix arrangement. file:///F|/SoS/SoS%2011-2005/sslaws900.htm (7 of 12)10/19/2005 9:42:01 PM

SSL AWS900

The mono dynamics processors provide both compressor and expander/gate facilities, sharing a common gain-reduction element. The compressor section follows SSL's usual practice of providing variable Ratio, Threshold, and Release controls with fully automatic gain make-up. By default the side-chain is RMS-sensing with a soft-knee transition, but the Peak button switches the response to peak-sensing and hard-knee, which is generally more appropriate for transient-rich signals. The attack time is normally programme dependent (varying between 3ms and 30ms), but a Fast Attack button fixes the attack time at 3ms if necessary. A column of yellow and red LEDs shows the compressor's gain reduction, while an adjacent column of green LEDs shows the gain reduction applied by the gate section. The gate has four rotary controls — Range, Threshold, Release, and Hold — plus another Fast Attack button and an Exp switch to change the mode to a 1:2 expander instead of a hard gate. The gate can be triggered by an external key signal (input via a rear-panel connector and shared by both dynamics channels).

Underneath each channel's multi-function meter are a matrix of routing buttons, below which are the switches which are used to assign the two mono dynamics sections for individual channel processing.

The stereo buss compressor is derived from the G-series quad compressor, and uses the same dual-VCA feed-forward topology. It can be switched into either the Record or Mix busses (but not both at the same time), and is equipped with a full set of controls: Threshold, Ratio, Attack, Release, and Make Up Gain, plus a bypass switch and a traditional white-on-black moving-coil gain-reduction meter. The main buss matrix allows any of six separate functions to be introduced into either the Record or Mix busses. Both stereo busses have their own balanced insert points which can be switched in or out using this matrix, and unusually the insert return can either replace or be summed with the main buss signal — a facility that allows an external mixer output to be combined with the main busses, or for the Record buss output to be summed back into the Mix buss, for example. The two dynamics processors and/or the buss compressor can also be inserted using the matrix, and the master stereo fader can be assigned to control either buss. By default the master fader operates with unity gain at the top of its travel (which makes fade-ins and fade-outs a lot easier), but an additional 10dB of gain can be introduced if required, so that it works like the channel faders, with gain in hand. In the top right-hand corner of the master section are rotary level controls and

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SSL AWS900

associated AFL buttons for the eight Track Busses, and a section concerned with the master outputs for the two stereo Cue busses and four mono FX busses. The level controls all have centre detents which correspond to unity gain, with an additional 10dB of gain available if required. Each output has its own AFL button, and a set of bar-graph meters above these controls shows the output levels for the Cue, FX, and Foldback busses.

Monitoring The monitoring section is surprisingly well equipped, with comprehensive facilities to drive two sets of 5.1 monitors and two sets of stereo monitors, complete with automatic downmix to stereo or mono. There's also individualspeaker level calibration, bass management, and encode/decode processor inserts. Further clever facilities make it easy to integrate a visiting engineer's preferred front monitors while retaining the 'house' rear channels. External monitor inputs are provided for four 5.1 sources and a further four stereo sources — all of which can be mixed together and/or routed to the main monitors, studio foldback, or headphones. The first six Track Busses are also available to the surround monitoring (to accommodate a 5.1 stem) and there is even a dedicated 5.1 output facility to feed a multi-channel metering system like DK Audio's MSD range. The Solid State Logic web site already has The monitoring facilities are further an extensive library of FAQs and detailed enhanced with an elaborate solo tutorials, and there are even things like track system, again derived from the sheets to download. SL9000K console. Destructive in-place solo, stereo AFL, and mono PFL modes are all available, with a separate PFL output to feed a dedicated PFL speaker if required. There is also a useful in-front solo mode, which is essentially stereo AFL, but with a user-configurable level of the entire mix added back in, allowing the soloed channels to be heard more clearly, but in context.

The master volume control has an associated numeric display to show the relative level either of the entire selected monitoring system, or of individual channels in calibration mode, or of the current dim levels when configuring the Solo modes. A bank of six buttons is used to assign the volume knob to adjust the default levels of various monitoring signals including the AFL, PFL, and soloin-place feeds. Three large white buttons switch the monitoring between the alternate 5.1 system, and the two stereo 'mini' systems. Additional arrays of buttons serve to mute or solo the individual monitoring speakers; select 5.1, 5file:///F|/SoS/SoS%2011-2005/sslaws900.htm (9 of 12)10/19/2005 9:42:01 PM

SSL AWS900

into-2 downmix, or mono modes; and select the monitoring source, with an option to mix the selections together if required. Its all very intuitive and flexible, and just what you would expect of a thoroughbred console from the SSL stable.

DAW Control As already mentioned, the multiple-HUI aspect of the desk enables the 24channel fader, mute, solo, encoder, and channel-selection controls to be assigned to operate the desk's analogue path, the corresponding DAW channels, or both, in banks of eight. The faders assigned to the DAW can be scrolled individually or in banks to access any number of DAW tracks. However, it is in the master section that the majority of DAW facilities are to be found. The basic transport functions are supplemented with auto-locate features, including Return To Zero, End, Loop, Pre/ Post-roll, Punch In/Out, and so on. There is also a numeric keypad, a jog/shuttle wheel, and a set of navigation cursor buttons. Particularly thoughtful additions are the Undo, Save, Shift, Alt, Control, Option, Escape, and Enter buttons, which allow access to all the context-sensitive alternate functions which proliferate in every DAW's menu architecture. This renders the keyboard all but redundant for normal operations. The small colour TFT screen has a relatively low resolution, but provides clear graphical and textual information relating to both the console automation and the recall facilities, as well as various DAW functions including effects plug-ins. The question many will ask is, can the display be output to an external screen, as on the big SSL consoles? The answer is no; the display resolution is much too low, and there really is no need because its wide viewing angle allows it to be seen clearly from whichever end of the console you happen to be sitting. The various menus and graphical displays are controlled using an array of soft keys and four rotary encoders below the screen. The encoders (with a press switch action) and associated soft keys are used mainly to control graphical plugin parameters, while two more buttons scroll forwards or backwards through any additional parameter options or menu pages. Two rows of eight buttons below these access the various console and DAW global options and commands, as shown with neat labels running across the bottom of the screen. This arrangement is very clear and intuitive, although some familiarity is required to manage these functions at speed. Finally, to the right of the jog/shuttle wheel are an array of talkback buttons which fall easily to hand, and there is even a button to switch on the studio's red light — file:///F|/SoS/SoS%2011-2005/sslaws900.htm (10 of 12)10/19/2005 9:42:01 PM

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how thoughtful!

On Session With The AWS900 The sonic and ergonomic pedigree of this console was immediately evident from the outset. It is very easy to find your way around the channel strips, and the flexibility of signal routing allows the desk to do far more complex things than you would have thought possible. The mic preamps and EQ are classic Superanalogue designs and do exactly what you need them to do. The monitoring section is surprisingly comprehensive (and future-proof too with its 5.1 support), and all the tools are provided for easy communication with the talent. After only a few minutes familiarisation I felt completely at home with the analogue aspects of the console, and controlling the primary DAW functions — arming tracks, recording, overdubbing, adjusting channels, tweaking plug-in settings — came just as easily. Overall this desk is Intuitive, ergonomic, and professional. However, there are a few aspects that restrict the ultimate flexibility of the console and highlight the fact that it was conceived mainly as a tracking desk, before analogue mixing again became more fashionable. The first clue is the absence of any groups or VCAs, although you can link faders. Perhaps a more obvious clue is the lack of dynamics on every channel — a pair of assignable dynamics processors may well be sufficient when tracking, but a lot more are usually required for mixdown. Of course, you could use DAW plug-ins, but that defeats some of the advantages of an analogue mixdown. Fortunately, SSL have addressed this issue by announcing the XRack range, the first unit of which will provide up to eight dynamics processors (the same as those in the console) which can be patched into the required channel insert points. The rack also incorporates Total Recall facilities which integrate fully with the console's system. The second element is the lack of surround panning facilities in the channels, making surround mixdowns rather tricky. Again, there are various workarounds, but a lot of reliance must inherently be placed on the DAW's facilities. Fortunately, again, the monitoring section is sufficiently flexible to allow several 5:1 stems to be combined from the internal Track Busses and external inputs. Having said all that, these compromises are not likely to be particularly significant to most users, and are easily outweighed by the numerous highlights of the console — a Superanalogue mixer with 'big desk' sound, features, and flexibility, for a remarkably modest UK price. You can really see where the money has been spent on this console, and it constitutes a truly professional high-end console for the mid-market studio or post house. Throw in the perfectly executed DAW interface and console integration and this has to be my product of the year!

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SSL AWS900

Published in SOS November 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

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Steinberg Halion Player

In this article:

What's In The Box Halion & Halion Player Compared How's It Look? In The Library Room For Growth End Game

Steinberg Halion Player £70 pros Great value. Easy to use. Sounds great.

cons Custom sample loading and program creation facilities are somewhat cursory, and you aren't allowed much in the way of sound modifications. Not even simple effects for custom programs.

Steinberg Halion Player v3.1 Sample Library Player [PC/Mac] Published in SOS November 2005 Print article : Close window

Reviews : Software

If you want the sampleplayback features of Steinberg's Halion sample player, but you don't need its detailed editing facilities, there's now an affordable answer to your needs — and it includes the full Halion sound library, too... Derek Johnson

With their Halion software sampler, Steinberg released a product deservedly renowned for its power, summary sound and flexibility. The engine at the Inevitably, packing the heart of the software also forms the entirety of Halion's engine into backbone for a number of third party a £70 program results in the sample library plug in instruments, loss of some features, but at including Steinberg's own Halion String this price, who can Edition (now with added '2'). This huge reasonably argue? Given the job Steinberg wanted Halion and detailed library is self contained, Player to do, this bundle not requiring the full version of Halion to succeeds eminently. run. information £69.99 including VAT. Ar biter Music Technology +44 (0)20 8207 7880. +44 (0)20 8953 4716. Click here to email www.arbitermt.co.uk www.steinberg.net

Test Spec

Somewhere along the line, Steinberg must have asked the question: 'What if there are people out there who want the library bundled with Halion, but don't need the full range of editing facilities?' The answer is Halion Player, subject of this review. HP (as I'll be referring to it in this article) is equipped with the simplified interface adopted by String Edition and the increasing number of sample library instruments that use Halion Player as a virtual instrument 'wrapper' for their sounds. However, HP differs from these in that it can load other sound sets, while the libraries that use HP as a 'wrapper' are 'fixed'. Best of all, though, HP only costs 70 quid!

PC REVIEW SYSTEM 3.06GHz Pentium 4 PC with

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Steinberg Halion Player

1.25GB of RAM running Windows XP. Steinberg Cubase SX v2.2. Cakewalk Sonar v3.1. Steinberg Halion Player version reviewed: v3.1.

What's In The Box In a nutshell, then, Halion Player places a streamlined front end onto the Halion engine and comes bundled with the full 2.5GB Halion library. It can play back any libraries or Programs created with Halion, including, as I've just mentioned, the libraries of any third party plug in based on the same technology. All the core technology of Halion is included, such as the 32bit audio engine, disk streaming, effects and the ability to output audio in surround. So if you load Programs created with Halion into HP, they'll sound exactly the same, right down to effects processing and surround routing — but you can't access these functions for editing from HP's streamlined interface. The full package's sophisticated interface and plentiful editing and audiomanipulation tools are missing, too. It's fair to say that HP is aimed at novices or entrylevel users who want to get underway without spending too much money. The version we're examining is numbered 3.1 — this is actually the first release of HP, but its anomalous version number reflects the latest release of the full version of Halion. Whether you're a Mac OS X or Windows XP user, Halion Player will work for you, although the minimum system requirements are quite high, as you would expect for a modern software sampler, and Mac use is only advised if you're running Mac OS 10.3.3 or later. Steinberg recommend a minimum processor speed of 800MHz for PCs (Pentium or Athlon), and 867MHz for Macs (G4 or higher), with a minimum 384MB of RAM on both platforms. You'll need 2.5GB of hard disk space for your installation, and, as with all recent Steinberg products, a key. These come free with Cubase and Nuendo, but if you're not already a user of either of those programs, you'll have to buy a key (they cost £20), as they're not included in the HP package. Assuming you have the hardware to do the job, HP supports VST, DXi2 and Audio Units plug in standards, and that means practically any serious MIDI + Audio sequencer can host the new plug in. The RTAS standard adopted by Digidesign's Pro Tools family is not supported, but Rewire compatibility (in conjunction with stand alone operation) lets HP run happily alongside the latest versions of this range of software. Be aware that surround output isn't available in stand alone mode: all audio is mixed to stereo. For the purposes of this review, I ran HP on my PC under both Cubase SX and Sonar, as you can see from the picture at the head of this review. Although Player lacks Halion's deeper editing options, a user's own samples can still be loaded, to form the basis of new, albeit simple, Programs. To this end, the software can import WAV and AIFF audio files, sliced up REX/REX2 loops produced by Propellerhead's Recycle, and instrument files from Steinberg's own LM4 virtual drum module. HP can also load ZGR files produced by Zero X's PC only Beat Quantizer (a slicing and audio quantising tool that offers a different approach to Recycle). The software is distributed on a single DVD (although demo CDs in the package

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let you try out Steinberg's Cubase SX sequencing package and the full version of Halion). There are no Quick Start or Installation guides, let alone a full printed manual, but there is a booklet that discusses Steinberg key activation, which seems a bit backtofront to me — especially as Halion Player, as already mentioned, doesn't even come with a key! Installation was easy, although there were a few oddities. A Read Me file on the main DVD points at the 'Documentation Folder' as the location of the user manual, but this was absent from my disc. A manual (complete with installation instructions) appears on your hard drive as part of the install, but this is rather after the fact!

Halion & Halion Player Compared There are many obvious differences between Steinberg's flagship sampler and its entrylevel offspring. There's a short chart below that summarises the most important variations. A longer version of this comparison can be found on Steinberg's site at: ftp://ftp.steinberg.net/Download/VSTi_HALion_Player/ HALion_3.1_HALion_Player_Comparison.pdf. FEATURES

HALION PLAYER

HALION

Maximum audio outs per instance:

256

256

Maximum instruments per instance:

16

256

MIDI channels:

16

128

Access to all sound and modulation parameters at once:

No

Yes

Graphical envelope editing:

No

Yes

Control of effects and internal routing:

No

Yes

Waldorf filters:

Yes (one)

Yes

Creation of velocity splits and crossfades:

No

Yes

Graphical sample mapping editor:

No

Yes

Keyboard mapping editor:

Yes (but use is restricted — see main text)

Yes

Graphical loop editor:

No

Yes

Usable file import formats:

WAV, AIFF, REX/REX2, SDII (Mac * only), LM4, ZGR

* As for Halion Player, plus Akai, Emu, Roland, Kurzweil, Gigasampler, Kontakt, EXS24, Sound Font 2, ISO and Nero Disc Image

How's It Look? Physically, with Halion Player what you see is what you get, and the layout is similar to that of the third party sample libraries that have licensed Steinberg's playback technology. Life centres around one large ish window (seen at the head of this review) which displays 16 Program slots, a strip of real time control knobs (labelled Sound Edit) and an on screen keyboard. file:///F|/SoS/SoS%2011-2005/steinberghalion.htm (3 of 8)10/19/2005 9:42:13 PM

Steinberg Halion Player

The Program slots can accommodate Halion Programs — such as those included in HP's included library — or your own samples, more about which shortly. The supplied library is organised by category, and a dedicated category selector (shown overleaf) is part of the Program slot, which speeds up searching for Programs. In addition, each slot has its own MIDI channel, making Halion Player up to 16part multitimbral. Simple mixing controls are provided for each slot — Level, Pan and basic output routing. In its default state, the signal routing is fixed at four stereo output pairs, four mono outs and one of four varieties of 5.1 surround buss. But HP can handle up to 256 audio outputs, although these are inaccessible unless a thirdparty bank of Programs that exploits this capability is loaded. Finally, a button cryptically labelled 'Ins.' selects a program slot for playback by the onscreen keyboard and for editing via the Sound Edit section's Q controls (Q stands for Quick). The eight knobs in the Q section don't have fixed assignments; their function is determined by the Program currently loaded. Q controls are also a feature of the latest version of Halion, although in the full version, the knobs can be manually assigned to almost any parameter. The idea is simply to provide you with quick access to whichever sonic elements you use the The Program selector menu, showing some most, and the assignments are saved of the various Program categories included with a Program. If you load such a in the supplied library. Program into Halion Player, these assignments are faithfully reflected on screen. Even if effects parameters have been linked to Q controls, the links remain in Halion Player and behave faithfully. This is rather frustrating, as there is no other way to access the hidden effects from HP (although effects can be muted if desired). Equally, no other synthesis or sound bending parameters can be accessed beyond those assigned in an external program. Should your Program not feature Q control assignments, and in those instances where you've created a Program from scratch with your own samples, the Sound Edit Q controls default to a set of useful but immutable assignments: filter cutoff frequency and resonance, filter envelope amount, amplitude envelope attack, decay, sustain and release, and amplitude envelope amount. Incidentally, the filter type enabled in these circumstances is a 24dBperoctave Waldorf lowpass filter. And summarising the filter EG in one knob is an interesting, and not ineffective, choice, especially if you automate it — all eight knobs can be assigned to the external MIDI controller messages of your choice. The middle section is filled out with a global level control, global tuning offset and a polyphony readout (HP has a maximum of 256note polyphony), and at the bottom is a mouseable keyboard. The latter is zoomable, and scrollable across the whole MIDI note range. Centrepiece of the HP interface, though, is the file:///F|/SoS/SoS%2011-2005/steinberghalion.htm (4 of 8)10/19/2005 9:42:13 PM

Steinberg Halion Player

futuristiclooking Options ball in the centre section, which can't fail to catch your eye. Halion Player's is smaller and less impressive than the one in Halion, but that turns out to be appropriate — aside from flashing mysteriously when you scroll across it, the ball doesn't actually do much, although clicking on it opens a Player Options window (above), where some basic operational functions are defined. Chief amongst these are MIDI controller assignments to the Q control knobs, but you've also the chance to define the software's voice buffer and RAM preload amount. HP streams samples off disk, but you can control how much is streamed into RAM at a time. Two little buttons also govern audio and resampling quality. Between them, you can balance number of simultaneous voices versus playback quality, and determine how well aliasing artifacts are suppressed. There are no other operating screens, though a couple of handy pop up windows can be invoked by right clicking on your mouse (Control clicking on the Mac version). One summarises various load, save and clear options and the other lets you see what sample or samples are assigned to a given key. This is especially useful when trying to figure out what samples have been layered in a particularly complex Program.

In The Library Of course, one of Halion Player's main reasons for existence is to provide affordable access to the Halion library, which comes in four banks that appear as one collection in the Program browser. It's a curiously patchy affair, although it's certainly large given the price you pay to access it in HP — for £70, I can't really complain. And what is provided is very good and often excellent. Guitars, basses, drums and synths (various flavours of pads, comps and leads) are certainly provided in force, and there's a good smattering of acoustic and electric pianos. (I took to the 'Grand Piano 1 XXL' program immediately — it's a great ad for Steinberg's The Grand, from which it's taken). Ensemble strings are also well represented, but solo strings are nonexistent, the brass complement is rather cursory and as for woodwind, there's but a single Program, 'Tenor Sax Soft'. I would have thought that an entrylevel package aimed at newcomers might try to cover more bases, even if they were covered thinly. The drum kits are especially good — one of them takes advantage of Halion's rather confusingly named 'round robin' facility to create more realistic drum rolls by rotating between several different samples for each successive hit, thereby reducing the 'machinegun' effect you get when you repeatedly trigger the same sample several times in quick succession. I also liked the detailed sixstring basses and the surround synth pads, and the raw Minimoog oscillator set was great, with appropriate Q Control assignments. A good number of patches are thoughtfully supplied in multiple forms: Eco (nomy), Mid and XXL. If your system can handle it, you should go for the XXL Programs, which have the most samples and layers and so on. Mid quality file:///F|/SoS/SoS%2011-2005/steinberghalion.htm (5 of 8)10/19/2005 9:42:13 PM

Steinberg Halion Player

programs reduce the number of samples by about 40 percent and Eco by about 65 percent, compared to an XXL program. In many cases, the Eco and Mid options still sound good, but could be used in situations where the part you're using them on isn't too prominent. And I guess you could use Eco programs whilst working on a track and then change to XXLs during a mix or when bouncing to disk, or if you take your session to a studio with a more powerful computer.

The Player Options window customises some basic playback and operational functions — note the window with the list of MIDI controllers.

This is all good stuff — I just wish there was more in the library. And how strange that space has been taken up with a handful of loops, as HP doesn't have the option to change the tempo of sampled drum loops. Instead, the collection is divided into fast, medium and slow tempo sets (no tempo values are given in beats per minute), which is of limited use to serious programmers — although some of the loops do sound pretty good.

Room For Growth While it might seem initially that there will be some redundancy if you buy a thirdparty Halionbased library due to the inclusion of another Halion Player, this isn't the case. Whilst the Player supplied with thirdparty content may have been jazzed up graphically by the thirdparty library developer, it will only ever be able to play that library. The ability of HP to load programs from any library means that it's possible to mix and match favourite Programs from any banks that you own. And as I hinted earlier, Halion Player can also accommodate your own samples, within certain limits. On the plus side, HP can load a wide range of sample types (though not quite as wide as the full version of Halion — see the comparison table on the second page of this article). The limits appear as soon as you try to do anything too fancy. Loading samples into the currently selected Program slot is simply a matter of using the Program browser and choosing the Load Sample function. You can replace what's in the current slot or merge the new sample with it. A pop up Mapping window (shown below) lets you determine how a sample is treated once it's loaded into a slot. You choose a start (root) key and then decide how the sample will be played back. If you choose the Chromatic option, one sample will be played across the whole keyboard, centred on the start key. In addition, the program can automatically set the start key if it's included as part of the sample name. So, if you add something like 'C3' or '60' (ie. a MIDI note number) to the file name, Halion Player will detect it during import and map it accordingly. Alternatively, if the file has root key data embedded in it, the software can be told

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Steinberg Halion Player

to detect this. Finally, there's an option for assigning samples to just white or black keys. What's missing is a way to assign key ranges to samples. A loaded sample is either playable across the whole MIDI range (the Chromatic option) or by one key only. Thus, it's possible to create, say, a drum kit or a simple one note per key multisample, but nothing with more subtlety. Similarly, although it's possible to assign multiple samples to one key (or the entire note range), there's no way to assign velocity ranges to each layer. And although a sample that contains a loop point as part of its data structure will loop on playback within HP, it's not possible to 'turn on' a loop for any other kind of sample. Of course, no effects or detailed editing can be undertaken.

The Mapping window, used when importing your own samples.

On a positive note, once loaded, your sample set can be saved with its own Program name (set the name in the Program slot window first), with Sound Edit knobs to tweak as you wish. Additionally, preset Programs can be renamed and saved. A Bank of 16 Programs can also be saved, comprising your own or factory material (this option, by the way, is necessary when using Halion Player in Rewire or stand alone mode, since Program assignments won't be saved from within the host as they are when running HP as a plug in). REX and REX2 Recycle files are recognised by HP; each slice is mapped to adjacent keys starting from the note of your choice. Recycle can produce a MIDI file that plays back the slices to maintain the original 'groove' of the file, though you can trigger them back any way you like. In many situations, this will be redundant, since most MIDI + Audio software hosts can already load REX files directly, but if you create the right kind of REX file, one that's made up of individual drum hits, you'll be able to create a simple drum kit really quickly.

End Game Even non beginners could find a home for HP; anyone with multiple computers that uses Halion for custom sample manipulation and Program creation could put Player on a slave computer to give them extra polyphony, MIDI channels and audio streams, saving themselves the expense of another full copy of Halion. When all is said and done, paying under 70 quid for a capable sample playback front end and the Halion library (even with its curious omissions) is a pretty fine deal. As a beginner's introduction to sample playback, HP works admirably, and even remains viable if you add extra libraries. And as beginners turn into file:///F|/SoS/SoS%2011-2005/steinberghalion.htm (7 of 8)10/19/2005 9:42:13 PM

Steinberg Halion Player

seasoned users, and wish to trade up to the full version of Halion, a favourable upgrade path is in place. In short, at this price, HP has got to be worth checking out. Published in SOS November 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

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Waves GTR Guitar Tool Rack

In this article:

Hardware Requirements Amps, Speakers And Mics Stomp Effects Pedal Power WPGI: The Hardware Plugin? Playing Around Verdict

Waves GTR £1200/£1800 pros Very easy to use. Comes with guitar-optimised DI box. Simple but very musicalsounding stompbox emulations. Can create the whole range of contemporary guitar tones with a high degree of authenticity.

Waves GTR Guitar Tool Rack Guitar Processing System [PC/Mac] Published in SOS November 2005 Print article : Close window

Reviews : Software

Software guitar amp modelling has been around for a while, but GTR is Waves' first foray into the field — and as you'd expect, they've made a huge effort to get it right, even hiring guitar maker Paul Reed Smith to design a bespoke preamp. Paul White

Waves are one of the best established plug-in companies around and have an enviable cons reputation for well thought-out, good-sounding The info screen that opens processors. They've so far steered clear of the before the plug-in is irritating. software instrument market, preferring to Amp and stomp pedalboard settings are stored separately. concentrate on sound processing, but their new XLR mic level output on the Guitar Tool Rack sees them entering the guitar amp modelling arena at a very serious level. It review model DI box produced noisy results. consists of a software package augmented by a No way to delay one side of hardware WPGI preamp that might best be a dual-cabinet setup to described as a guitar-optimised DI box; Waves emulate mic distance. sought the help of guitar design guru Paul Reed summary Smith to make this behave exactly like the input Rather than try to provide stage of a typical guitar amplifier. Apparently Paul every possible feature going, was also involved in the voicing of the amplifier Waves have opted to develop models. a simple, good-sounding system with enough flexibility to keep most players happy. Guitarists will inevitably disagree on whether or not it is a complete substitute for a miked amplifier but it is certainly capable of excellent results if you're prepared to step beyond the presets and fine-tune the sounds for yourself.

In addition to amp and speaker modelling, Guitar Tool Rack also offers a comprehensive set of 23 guitar stomp effects plus a tuner that can be used chromatically or set to custom guitar tunings. The pedals are called up separately from the amp model and are not complete plug-ins in their own right, but can be opened up in virtual 'floor unit' plug-ins that each hold two, four or six virtual pedals. Doing it this way means that any of the different stomp effects can be set up before or after the amplifier model as is most appropriate. For example, gates, distortion and fuzz tend to come before amps while delay and reverb are usually

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Waves GTR Guitar Tool Rack

information Native version £1200; TDM version £1800. Prices include VAT. Sonic Distribution +44 (0) 1582 470260. +44 (0)1582 470269. Click here to email www.sonicdistribution.com www.waves.com

Test Spec GTR1 v1.0. Apple G5 dual 2.7GHz Mac with 4 GB RAM, running Mac OS 10.4.1. Tested with Apple Logic Pro 7.1.

applied afterwards. Most of the effects are familiar 'staples' such as EQ, compressor and gate, delay, reverb and various modulation effects such as phaser, flanger, chorus and wah. There are also several types of distortion effects and an octaver as well an interesting reverse delay with additional twists, such as pitch-shifting, called Lay'D. MIDI control from an external device is simplified by virtue of a straightforward MIDI learn function and full parameter automation can be handled by sequencers that support plug-in automation. In Logic, the parameter names also show up correctly in my Logic Control. Naturally, these stomp effects can be used in any audio track, though for guitar use, Waves understandably recommend you use them with your Waves Amp plug-in. Waves Amp is designed to emulate a range of classic amp sounds, but these are arranged by degree of dirtiness rather than by cryptic clues as to the actual amps they were modelled on. They can be further modified using familiar amp drive and bass, middle, treble and presence tone controls, as well as a choice of modelled speaker cabinet types, microphones and on or off-axis mic positions. The amps range from clean vintage to modern high-gain with all the usual stages inbetween, and each is designed to give the same dynamic response as a real amplifier. Inevitably there are presets to get you started, but equally inevitably, you get the best results if you work at the settings and come up with your own tones. Once you have a suitable amp sound set up, you can then place your choice of stompboxes both before and after the amplifier; the software is intelligent enough to handle the mono/stero designation depending on what stomp effect is in which part of the signal chain.

Hardware Requirements Because guitar modelling needs to be applied in real time in order for the player to hear the results, a low-latency system is essential, but most DAWs based on modern computers should be able to to achieve this. Pro Tools TDM users obviously benefit from very low latency, but most native systems these days can also go low enough: for me, a 128-sample buffer size at a 44.1kHz sampling rate proved quite acceptable. GTR works on both Mac (OS 10.3.8 and above) and PC (Windows XP) platforms and requires a minimum of a 1.25GHz G4 or a 1.7GHz Pentium 4. Pro Tools users need to be running version 6.7 or higher. Note that GTR works with Pro Tools HD and HD Accel hardware at sample rates up to 96kHz but Pro Tools Mix systems are not supported at all. The plug-ins are also available in the native RTAS and off-line Audiosuite formats for use in Pro Tools. TDM hardware supports running any plug-in on up to a single DSP element but a single plug-in cannot run on multiple DSPs. This creates some inherent limitations where not all the plug-ins will be able to run under TDM in all situations. Digi HD Core and HD Process types running at up to 48kHz will allow all the stomp plug-ins to be available but the Waves Amp plug-in will run only in mono; for stereo you need to open two instances and route accordingly. The full manual

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Waves GTR Guitar Tool Rack

is available on the Waves web site and gives more details of the limitations when working under TDM. As a VST or AU plug-in, the system runs at up to 48kHz and no other restrictions apply other than those imposed by maxing out your CPU.

Amps, Speakers And Mics When you first load the Waves Amp plug-in, one minor irritation is that an info screen loads first and you have to click on this to get to the 'real' plug-in window. The same is true of the virtual stomp pedalboards, which load as separate plugins. In the studio, it is not uncommon to record the same amp using two different microphones or even to use two different amps at the same time. To help emulate this, the Waves Amp plug-in comes in several variants. Mono Amp feeds a single cabinet/microphone filter while Mono Dual Cab lets you feed the amp section through two different cabinet/mic setups which are then summed back to mono. Mono-to-Stereo is similar but keeps the two cabs in stereo, while Stereo Amp is actually a dual-mono configuration, accepting a stereo input and outputting in stereo: the left input goes through the amp model to cabinet 1 while the right input goes directly to cabinet 2. If your DAW allows, you can also route to two instances of the plug-in at the same time to achieve the effect of recording two different amplifiers. The six mic emulations include dynamic, ribbon and capacitor models with a choice of on or off-axis placements, which provides a useful range of natural-sounding tonalities. A total of seven amp models are included. Direct combines light EQ with mild dynamics processing and can be used with any cabinet and mic combination, or with none at all. Clean is based on a tube amp with relatively little distortion, while Edgy sounds like a '60s tube amplifier that can be either The Waves Amp plug-in includes seven clean or quite dirty depending on how different amp models, plus cabinet much gain is used. This is quite emulations that can be placed in mono or responsive to picking pressure or to stereo configurations. the guitar volume control setting. Drive gives a bluesy growl that can be escalated to a saturated rock sound, but it can still play pretty clean at low drive levels. Crunch is perhaps the definitive solo setting, and also gives a pretty convincing grunge sound. Odd harmonics are exaggerated at higher drive settings and even at low drive levels, the sound never cleans up enough to be described as clean. Hot is suitable for power chords and more aggressive soloing while Modern Lead most definitely goes to 11 for ultra-high-gain soloing and power chords. Regardless of which amp is called up, there's a Drive control as well as bass, middle, treble and presence, but these act differently according to the amplifier file:///F|/SoS/SoS%2011-2005/wavesGTR.htm (3 of 8)10/19/2005 9:42:21 PM

Waves GTR Guitar Tool Rack

type. Interestingly, the tone controls (other than presence, which adds boost only) can either deliver cut or boost, with flat being the mid position. A neat touch is that the drive control is arranged so that the amp output level doesn't change when more drive gain is applied, making it easy to fine-tune your sound without constantly changing the output level. The amp plug-in can be bypassed completely for a clean DI sound, or you can bypass just the amp and leave the cabinet and mic models running. Each cab has its own volume control and in the stereo configurations, each also has a pan control. Where dual cabinets are being used, each has a phase-invert switch which affects the sound when the two cabinets are combined, but there's no way to add a variable delay to just one of the cabinets to simulate the effect of varying the microphone distance of one mic in a dual-mic setup, which I feel is a missed opportunity. There's also a master output gain control that can be used to get as much level as possible out of the plug-in without clipping. Metering is available in the centre, with a red clip indicator.

Stomp Effects No guitar effects collection would be complete without its distortion and overdrive pedals and this one is no exception. The Distortion stomp is hotter then Overdrive and comes in mono and stereo configurations with drive, level, tone and contour controls, the latter acting as kind of 'fat' filter. Fuzz is reminiscent of the old '60s fuzzboxes (though to my ears not quite as raspy) and has controls for sustain, level and tone so it couldn't be simpler. Less familiar is Buzz, which features a resonant peak at the high-pass cutoff frequency for a kind of distortion/wah morph. The Metal stomp is distinguished by a dual-distortion engine for extra searing overdrive and includes low, mid and high tone controls with a variablefrequency mid-range bell filter. Moving on to the modulation arena we find a Flanger with a wide range of adjustment and a neat LFO sync option. Vibrolo is a combination of tremolo (amplitude modulation) and vibrato (frequency modulation), again with a sync'able LFO. There are also separate Vibrato and Tremolo stomps as well as a panner and a phaser, again sync'able. Chorus (mono or stereo) is also in this section along with the slightly more subtle Doubler, which creates a slightly delayed, pitch-modulated signal to emulate double tracking.

GTR effects and processors are loaded into the Waves Stomp 'virtual pedalboard' plugins.

Octaver emulates the old octave divider pedals, and only works correctly on single-note parts. There's control over gain and panning of each octave and you can create both single- and double-octave drops at the same time, mixed with the dry sound. Another familiar face is the Wah-wah pedal, which can either work automatically from the input signal envelope or be controlled from a MIDI pedal or fader. The Wah-wah stomp has five controls for sensitivity, speed, range, mode file:///F|/SoS/SoS%2011-2005/wavesGTR.htm (4 of 8)10/19/2005 9:42:21 PM

Waves GTR Guitar Tool Rack

and wah frequency. Pitcher is a more conventional pitch-shifter for creating automatic 'parallel' harmonies and pitch-shifting in mono or stereo. Shifts of up to one octave in each direction are available with both semitone step and fine-tune controls. The other guitarists' favourites, delay and reverb, are not neglected either, and again Waves have chosen to keep the controls simple and pedal-like. The Delay stomp offers up to two seconds of delay time and is designed to give a warm, analogue sound. This can generate anything from simple mono delays to complex ping-pong stereo effects and the delay time can be sync'ed to tempo with variable high-cut filtering of the delayed sound. The clevery titled Lay-D combines conventional delays with reverse delays and pitch-shifting to provide the nearest thing to psychedelia offered by this package. You really have to hear it to appreciate what it can do but it's well weird! The Reverb stomp is a room emulator type of reverb but much simpler than you'd expect from a DAW plug-in. It includes pre-delay, decay time and tone but there's nothing else to twiddle. Similarly, the spring reverb emulation mimics the kind of spring you'd expect to find in a guitar combo, except it doesn't go sproing when you kick it and it allows you to add pre-delay! On top of these fairly obvious effects are the more subtle but no less important compressor and gate, which are combined into one very simple Gate/Comp stomp unit. These two stalwarts are also available separately in slightly more sophisticated forms alongside a three-band mono/stereo EQ stomp with bass, treble, mid boost and mid cut. If this isn't enough, you can also call up a six-band graphic EQ with a low-cut filter built in to suppress hum and rumble where each band has a ±12dB cut/boost range. The final plug-in isn't an effect at all but a very sensitive guitar tuner that can be chromatic (with an auto mode) or set to different open tunings.

Pedal Power The 23 stompbox effects (see box for details) can be loaded into the Stomp 2, Stomp 4 or Stomp 6 'pedalboard' plug-ins depending on how many of them you wish to use. These pedalboards come in mono, mono-to-stereo and stereo variants, and appear in the host program's plug-in processor list. Stomps can be dragged around to change their order, and as with the amp plug-in, there are normal save and load menus as well as presets. However, it should be noted that as the amp and pedalboards are separate plug-ins, their setups are saved separately too — so if you have a killer sound that uses some pre- effects and some post- effects as well as an amp model, you have to recall three lots of plugin patches rather than just one, as would usually be the case for other plug-ins that combine the amp and effects in the same plug-in. In Logic Pro you can simply save the whole channel strip along with its plug-ins as a user preset, but with other DAWs you'll have to make your own arrangements. Each pedalboard has controls for input gain, Setup A/B, Previous Setup and Next Setup as well as output meters and a clip indicator. Setup A/B (which you can automate) allows you to change between two settings virtually instantaneously providing the same stompboxes are used and in the same order — only the settings change. Individual stompboxes can be bypassed. file:///F|/SoS/SoS%2011-2005/wavesGTR.htm (5 of 8)10/19/2005 9:42:21 PM

Waves GTR Guitar Tool Rack

Automation can be catered for in the host sequencer; each pedalboard has a bypass and up to seven parameters per slot. As the stompboxes have similar control configurations (typically three to five knobs) to their hardware counterparts, this is generally more than sufficient. Where relevant, delay times and LFO rates can be sync'ed to multiples of the song tempo. Most of the stompboxes have mono, mono-to-stereo and stereo versions except where the function dictates — for example, the panner needs to be stereo-out to make sense. MIDI control is also available by assigning the stomp controls to MIDI controllers enabling third-party MIDI pedalboards and similar devices to be used. MIDI assignments get saved along with your presets and a learn mode simplifies this process, allowing the parameter to be controlled to be chosen from a dropdown menu. Though you could use the plug-ins with any high-impedance audio interface and a guitar, the included hardware (see box) is optimised for guitar and certainly does a very good job of preserving the tone of your guitar while keeping noise to a bare minimum if you use the jack or XLR line output — though I don't know why the XLR output is so noisy when set to mic level, as many other DI boxes work fine this way. I found the four-LED metering a little vague but if you doublecheck the levels in your DAW software there's no problem.

WPGI: The Hardware Plug-in? Both unbalanced jack and balanced XLR outputs are available from the included WPGI preamp box, and in my tests there was a definite improvement in tone and feel when using this box when compared to using the high-impedance DI input on my audio interface. However, for some reason, the balanced output from the box was quite noisy at the mic level setting, although the unbalanced jack or the XLR output fed into a line input behaved fine. If this is a technical limitation, then it would perhaps have been better to leave off the mic out setting altogether as it's bound to trip up some users. A four-LED meter lets you set the input gain reasonably accurately but there's no phantom powering option, which is a shame — you need to use two 9V PP3 style batteries or an optional 12V power adaptor. To activate the unit, it must be switched on and a jack plugged into the instrument input. A warning LED shows when the battery is getting low. There's also a ground-lift switch, which you don't need to use when recording guitar, but it can be useful in avoiding ground-loop hum if you're using the system with a mains-powered sound source such as a synth.

Playing Around To my ears, the amp model default settings sound very American with a fairly hard attack, bags of definition and a bright edge to the sound, but once you start to play around with different cabs and mics, pretty much the whole range of accepted guitar sounds becomes available. Using the Overdrive stomp with the file:///F|/SoS/SoS%2011-2005/wavesGTR.htm (6 of 8)10/19/2005 9:42:21 PM

Waves GTR Guitar Tool Rack

tone turned down slightly also coaxes a much more British rock sound out of the Drive and Crunch amps. Further tailoring can be done by inserting EQ stomps before and after the amp model. I felt the models responded well to playing intensity, cleaning up nicely when the guitar volume control is turned down. Some of the heavier sounds also have a nice meaty chug to the low end, but I never felt the models made the guitar feel quite as 'springy' as when playing through a really good tube amp. In this respect, GTR feels similar to the best hardware guitar modelling solutions, though arguably is better than most when it comes to producing a believable clean or almost clean sound. With the exception of the rather weird but very seductive Lay-D reverse echo/ pitch-mangling pedal, the stomps are all fairly conventional and very well behaved apart from the octave divider, which I found to be almost unusable with my Strat as it tended to gargle and warble, no matter how cleanly I played my single-note lines. I've used hardware octave pedals and they're far more forgiving than this emulation. The delays, reverbs and modulation effects are, however, first-rate and behave very much like their pedal counterparts. I also liked the Compressor stomp as it adds a useful degree of sustain and energy in a very musical way. Compressor gain reduction is indicated by a simple meter in the centre of the pedal.

Verdict Though TDM users may face some limitations as to which stomps can be used at the same time due to DSP loading issues, VST and AU users should have no problem putting together just about any reasonable combination of amps and stomps. I though the amp sounds were really very good after some user tweaking, though the lack of a save feature that can store both an amp and its associated plug-ins at the same time is frustrating. I also felt that Waves As well as a comprehensive range of missed out on a trick by not providing stompbox emulations, the GTR plug-ins some way to create separate chains of include a well-specified guitar tuner. stomps to process, or at least delay, the outputs from the two speaker cabinets independently — but they're not alone in this omission. To my knowledge, only Native Instruments' Guitar Rig allows this sophistication of stereo routing. The hardware front end certainly improves the results that can be achieved, if you use the line output. There's still some residual noise when using heavily overdriven amps or effects, but this is to be expected and can be addressed

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Waves GTR Guitar Tool Rack

reasonably effectively by using a Gate stomp prior to the amplifier. However, nobody yet seems to have cottoned on to the fact that dynamic noise filters (a high-cut filter where the cutoff frequency is controlled by the input signal envelope) work exceptionally well with guitars and are far more effective than gates at hiding hiss-type noise. Overall, the GTR system does an exceptionally good job at modelling the sounds of real-world miked guitar amplifiers and the lack of emulations of specific amplifiers doesn't seem to restrict its flexibility at all. It doesn't always feel quite as springy as a real amp but a carefully set-up compressor stomp before the amp model can help redress this. Some pre- EQ built into the amp model rather than inserted as a stomp plug-in might have made fine-tuning the sounds a little faster, but after an hour or two playing with the package, I found it pretty easy to coax just about any guitar sound out of it, from twangy country to dense, searing modern metal. More importantly, it's also good at those tricky 'on-the-edge' sounds so loved by blues and vintage rock players. The standard of the effects is also generally very high indeed (other than that unruly octaver) and having the ability to sync certain effects to tempo makes them far more flexible in this respect than their hardware cousins. Waves' aim was to create a guitar plug-in that would satisfy the majority of professional players, and it will be interesting to see how many albums end up being recorded using GTR rather than a conventional miked amplifier. Published in SOS November 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

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Q. Can I connect an AES output to an S/PDIF input?

Q. Can I connect an AES output to an S/PDIF input? Published in SOS November 2005 Print article : Close window

Sound Advice

I own a Focusrite Liquid Channel (which is great!) and would like to connect it to my MOTU 828's digital input. However, on the digital side of things the Liquid Channel only has an AES-EBU input and output, and the MOTU 828 only has S/PDIF and ADAT inputs and outputs. How can I connect them together? Bernhard Wagner Technical Editor Hugh Robjohns replies: The proper way to connect AES to S/PDIF is to use a dedicated digital format converter, of which there are plenty around (although some older designs only pass 16 bits rather than the full 24, so check before buying). The M Audio CO3 and the Behringer Ultramatch and Ultramatch Pro probably represent the most affordable options. However, although it is a rather makeshift solution (a 'bodge', to use the technical term), if you only need the signal to travel a fairly short distance — say, no more than two metres — you can get an AES output to feed an S/PDIF input with just a simple XLR-to-phono cable. Wire pin 2 of the XLR to the tip of the RCA phono jack, and pin 3 of the XLR to the sleeve of the phono. Pin 1 of the XLR should remain wired to the cable screen at the XLR end, but leave it disconnected and insulated at the phono end. Strictly speaking, the Channel Status and other subcode data is formatted differently between AES and S/ PDIF, but very little equipment bothers to send or read the full subcode data set anyway, so it is rarely a problem. Published in SOS November 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

file:///F|/SoS/SoS%2011-2005/qa1106_5.htm10/19/2005 9:42:45 PM

Q. Did Mellotrons use tape loops or not?

Q. Did Mellotrons use tape loops or not? Published in SOS November 2005 Print article : Close window

Sound Advice

I've read some conflicting opinions about the Mellotron. Some say that it used tape loops but others are saying that playback stops after eight seconds. Surely if tape loops are used, the sound would sustain indefinitely. Can you put the record straight? Warren James SOS contributor Steve Howell replies: Yes, Warren — there is some confusion regarding the workings of the mighty Mellotron, so let's sort it out now. The Mellotron (or the Chamberlin, from which the Mellotron is derived) does not use tape loops. Each key has a length of tape below it that lasts approximately eight seconds. In effect, each key is a crude tape machine and when a key is pressed, a pinch roller moves down onto a constantly rotating capstan that runs the length of the keyboard. The tape is then dragged forwards and over a tape head and the sound recorded on that tape is played back (see diagrams below). When the tape has come to an end (often stopping somewhat gracelessly!) that's it — no more sound will be made by that key until you release it, whereupon a spring pulls the tape back, ready for use again. Mechanically, the Mellotron was a nightmare — 35 tape heads to keep aligned, 35 pinch rollers to keep clean, and 35 pressure pads to adjust so that the tape made proper contact with the heads (not too much pressure, not too little). It was necessary to keep an eye on those return springs, too (all 35 of them!), because if they malfunctioned, the tape wouldn't be returned to the actual start and subsequent playback would start some way into the sound, resulting in the characteristic 'clicky' Mellotron effect heard on so many records. However, this had another side effect; if the tape didn't return to the start, you wouldn't even have eight seconds to play with! This limitation regarding sustain was the biggest frustration for most Mellotron players and it spawned a unique 'crawling spider' playing

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Q. Did Mellotrons use tape loops or not?

technique, whereby sounds would be artificially sustained by playing different inversions of chords before the tapes ran out. One of the Mellotron's most famous players was Rick Wakeman. As well as playing Mellotron as a session player on David Bowie's 'Space Oddity', he was, of course, to rise to fame as the goldencaped keyboard player of Yes, boasting two Mellotrons in his massive keyboard rig. Such was his frustration with the instrument that he funded a project with one Dave Biro to create the Birotron, which did use tape loops that rotated continuously. The basic principle was much the same as the Mellotron, except that continuous loops were used for indefinite sustain. However, this project never really got off the ground and only 35 Birotrons were made. Wakeman had four (of which two were stolen) and the instrument was used on Yes's Tormato album. Published in SOS November 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

file:///F|/SoS/SoS%2011-2005/qa1106_2.htm (2 of 2)10/19/2005 9:42:50 PM

Q. How can I make the most of a small studio space?

Q. How can I make the most of a small studio space? Published in SOS November 2005 Print article : Close window

Sound Advice

I'm just about to set up a studio in a room of my house and I was wondering if you have any tips on where to put everything. I have quite a few keyboards, sound and effects modules, guitars and a computer workstation, and I also record vocals, so I need to provide for all those things. The room I have is quite small and I know it'll never be perfect in terms of acoustics, but can you offer any advice on general layout? Tony Mendelle SOS contributor Tom Flint replies: If your workspace doesn't feel right, you'll probably find it difficult to commit yourself to doing any work. Get it right, on the other hand, and hopefully inspirational recordings will follow! The way I see it, a typical home studio has three important spots where most of the activity takes place, which we'll call the engineering seat, the performance seat and the listening seat, and it's a good idea to start by thinking about where these might be best situated. The engineering seat is where you'll be when working at a computer, hardware multitracker or stand-alone sequencer. This is where the majority of editing, mixing and programming takes place, and is therefore somewhere you are likely to be for long periods of time. The performance seat is the place where music is created and played, perhaps using a workstation keyboard, a guitar or a simple MIDI controller. It can even be the same place as the engineer's seat if you use a desk with a sliding keyboard shelf. The last significant spot is the listening seat. Even the smallest studio needs one, in my opinion. It is where you can relax with a cup of tea, listen to your music and ponder. Without one, your studio will merely be a place of work and you'll be tempted to leave it whenever you feel jaded. Naturally, this position should be angled towards the studio's speakers, but you won't be doing any really critical listening from here so you needn't worry too much. For efficient working you'll want to have the most frequently used bits of kit within reach from the key working positions. If, for example, you are using hardware multi-effects processors patched into the send/return loop of a workstation, you'll want to be able to easily reach the effects parameter controls during mixing. Many pro studios tend to have racks of gear under the desk, but I avoid this arrangement at all cost. If, like me, you find touching your toes a struggle, you'll be much more comfortable having everything up at eye level. Not only does this leave leg room below, it also makes the programming of effects units or rack synths a more appealing prospect, allowing you to get the most from your equipment without having to move from your playing or mixing position. The space under the desk need not be wasted though. I use mine to store boxes and unused bits of kit, and it is also home to my two guitar amps, which I don't need daily.

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Q. How can I make the most of a small studio space?

If you are recording your own vocals and use hardware preamps, it is also vitally important to have them close to hand when standing near the mic, so that adjustments can be made while singing. In my setup, for example, I have a 10U rack of gear seated on top of my desk so that all my effects and preamps are between three and five feet high, which is ideal for checking settings and making adjustments. If the preamps are feeding a multitracker or computer workstation, it is also useful to be able to see the input meters on the display screen from the vocal position, at least while levels are being established. Once again, if everything is to hand, excessive levels can be curbed by simply reaching for the preamp output level control. It's often possible to use footpedals to remotely adjust transport controls when performing, but it is well worth arranging things so that recording devices can be manually operated too.

When space is limited, you need to think carefully about how to lay out your studio.

Having the right desk surface can make all the difference, and often the most effective option is to build your own so that it fits the room, your equipment and your own physical stature. I built my desktop at a height that allows me to comfortably play my master keyboard either when I'm standing up or when seated on a stool. In fact, I have everything, including my multitracker, placed on the same desk so that it can be operated from both positions. You'll save money by building your own desk, too. My desk, which is nine feet long and two feet deep, has a frame built from thick planed pine and an MDF top, and it probably cost less than £40 in materials. Putting up some shelves will give you somewhere to put boxes of leads and adaptors, old copies of SOS and other general studio clutter. The side benefit is that by placing them carefully they can also be used as a crude, but often effective, form of acoustic treatment. You might also consider hanging your guitars on the wall to free up floor space and mounting monitors on wall brackets to free up the desk. One of the most limiting factors governing your studio layout will be the interconnectivity of the gear. Every studio requires power, audio I/ A comfy chair or two can make your studio a O and, in most cases, a network of MIDI connections. For mains much nicer place to be. power, I like to wire up my own plug-boards so that I can cut the cable to the required length and run it around the edge of the room, well away from doorways. In fact, it's a good idea to make sure that there aren't going to be any cables running in front of doorways, cupboards and so on, otherwise sooner or later they'll get hooked on a fast-moving foot and will do some damage to you or your equipment. Furthermore, a patchwork of wires makes vacuuming difficult. Being able to clean a studio effectively is an important consideration, because dust can seriously damage audio equipment such as mics or faders. MIDI leads are best kept as short as possible, because the data stream doesn't include any error correction and errors become more likely the longer the cable is, so it's worth planning the MIDI layout early on. If MIDI Merge or Thru boxes are used in the system, these can be carefully placed to act as hubs for the network. It's possible to buy both audio and MIDI leads in a variety of colours and this is well worth doing, because it makes it far easier to distinguish each cable from the rest. Otherwise it's a good idea to label each lead using stickers wrapped around either end. Labels can be used for mains plugs too.

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Q. How can I make the most of a small studio space?

Before building your studio, it's also worth remembering that bad posture and poor ergonomics can cause repetitive strain injury (RSI) and other uncomfortable conditions. For more on the subject, check out the feature SOS ran in the January 2002 issue (www.soundonsound.com/sos/Jan02/articles/studioergonomics.asp). Published in SOS November 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

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Q. What is that 'robot voice' effect?

Q. What is that 'robot voice' effect? Published in SOS November 2005 Print article : Close window

Sound Advice

In the recent TV ad campaign for Marks & Spencer, they use the Electric Light Orchestra track 'Mr Blue Sky'. There's a distinctive robotic vocal sound in it that I am curious about. How was it made? I was thinking at first that it was something like Auto-Tune (as on the annoying Cher single!) but the ELO record was made years before that. Or is it a remix? (I'm not old enough to remember the original!) Danny Finn SOS contributor Steve Howell replies: The effect is created using a device known as a vocoder, which is short for voice encoder, though it was also briefly known as a 'voder'. Like so many things in this business, the vocoder dates back many decades and, again like so many things in this business, is derived from telephonic communications technology! It was originally developed by Homer Dudley of Bell Labs in the '40s as a means to compress audio for transmission down copper telephone lines. Later, one Werner Meyer-Eppler of Bonn University saw the potential for the vocoder in the then-emerging genre of electronic music. Basically, a vocoder has two inputs: a modulator and a carrier. The modulator is usually fed by a microphone, typically with sung or spoken words, and the carrier will take a bright, sustained synth sound. Chords are played into the carrier input and words are spoken (or sung) into the modulator. The spoken/sung words are electronically imposed on the carrier signal, to create the effect of the synth speaking or singing. So how does this magic work? Modern software vocoders like NI Vokator The carrier signal is split into different frequencies, using very tight are far more sophisticated than their band-pass filters (not unlike those in a graphic equaliser), and each hardware forebears. of these has a voltage-controlled amplifier or, more recently, a digitally controlled amp. The modulator input is similarly split into different frequencies and on the output of each of the modulators' band-pass filters is an envelope follower that opens and closes the corresponding amplifier on the carrier input (see diagram, left). Thus, if you were to say 'ooaaah' into the modulator, the lower filters on the modulator would activate and open the lower filters on the

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Q. What is that 'robot voice' effect?

carrier's signal; as the modulating signal moved into 'aaaah', the modulator's higher filters would be activated, in turn opening the carrier's upper filters and creating the illusion of vocals. The number of filter bands the vocoder has is crucial. In the early days of analogue vocoders, for technical reasons (and reasons of cost) they typically only had around 10 bands, making speech somewhat unintelligible. More recent developments using modern DSP allows vocoder designers to include almost any number of filters, meaning that intelligibility is greatly improved, although they still sound like vocoders. Of course, these filters only really deal with the vowel components of a sound; to cater for sibilants and fricatives, such as 's', 'b' and 'p', noise generators are sometimes used, which are triggered when the modulator detects them. While they help, they are still not convincing. Vocoders were grossly over-used to the point of cliché in the '70s ('Mr Blue Sky' being a prime example!) and they subsequently fell from grace. However, they can be responsible for some stunning sounds, and one only has to listen to Herbie Hancock's use of his Sennheiser vocoder in his brief foray into dance/disco music in the '70s and '80s to confirm this. Feeding the vocoder with an impeccably phrased Minimoog, Hancock created perfectly realistic and fluid lead vocals but with a curious robotic quality. He multitracked these to create harmonies and backing vocals to stunning effect. More recently, the vocoder has become much more than just a 'speaking synth' effect, and people routinely now feed drum loops into the modulator to rhythmically chop the carrier signal. Prominent vocoder manufacturers of old were EMS (whose products are still on sale), Moog (who I believe based their design on the vocoder Wendy Carlos contructed out of discrete modules on her giant Moog modular), Sennheiser, Korg and, of course, Roland, with their famous Vocoder Plus. More recently, there has been a veritable glut of software vocoders on the market, some free, some shareware and some payware. Notable examples are Akai's DC Vocoder and Native Instruments' Vokator, both of which offer outstanding intelligibility and flexibility. Published in SOS November 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

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Q. What is the best way to mic up a clarinet?

Q. What is the best way to mic up a clarinet? Published in SOS November 2005 Print article : Close window

Sound Advice

I'm in need of some help and advice. I was working with a group at the weekend who have a wind player in their line-up. She alternates between flute and clarinet. I had no trouble picking up the flute, but when it came to the clarinet I just could not get a signal that would cut through the mix, despite using a second mic. The mics I was using were a JTS smalldiaphragm condenser (cheap but usually effective) and a Shure SM58 pointed towards the bell of the instrument, but slightly off-axis. I have spoken to another engineer who has also worked with the group and he confirms that he also had problems with the clarinet. Any suggestions? SOS Forum Post Technical Editor Hugh Robjohns replies: The clarinet has the widest dynamic range of any orchestral instrument, so it's no wonder you had problems! First of all, don't try placing the microphone anywhere near the bell. Most of the sound of the clarinet comes from the unstopped finger holes, so if you focus on the bell you'll be missing 70 percent of the instrument's harmonic range! Instead, you need to place a mic where it can 'see' the entire body of the clarinet. Try placing the small-diaphragm condenser mic about a foot away and aiming more or less at the lower hand position. You can experiment with the height, angle and distance of the mic until you find a balance you're happy with, but it may also be worth having a quiet word with the player to explain the problem, in order to extract a more appropriate performance in terms of dynamics and consistent positioning relative to the mic.

Different mixers employ different panning laws.

In a live situation, miking may be more difficult because of spill, feedback or over-exuberant performers, amongst other things. If so, you might find it easier to use a clip-on instrument mic. There are lots on the market that will hold a small mic in a reasonable position and keep it there regardless of the gyrations of the player.

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Q. What is the best way to mic up a clarinet?

Published in SOS November 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

file:///F|/SoS/SoS%2011-2005/qa1106_7.htm (2 of 2)10/19/2005 9:43:07 PM

Q. What is the difference between mono with one speaker and mono with two?

Q. What is the difference between mono with one speaker and mono with two? Published in SOS November 2005 Print article : Close window

Sound Advice

I read recently that when top engineers check their mixes in mono, they don't just hit a mono switch, but instead route the mix through a single speaker to hear it in true mono. What's the difference between the two? SOS Forum Post Technical Editor Hugh Robjohns replies: It's important to check the derived mono signal from a stereo mix to ensure that nothing unexpected or unacceptable will be heard by anyone listening in mono, as could be the case in poor FM radio reception areas, on portable radios, in clubs, on the Internet and so on. Mono compatibility, as it's called, is very important for commercial releases — the artist, producer and record company want the record to sound as good as possible in these less-than-ideal circumstances. In addition to simply checking the finished product, mixing in mono, or regularly switching the monitoring to mono while mixing, is very useful and a good habit to get into. Summing to mono removes any misleading phasing between the left and right signals that can make a stereo mix sound artificially 'big'. The crucial difference between auditioning the summed mono signal on a single speaker, as compared to a 'phantom' mono image between two speakers, relates to the perceived balance of the bass end of the frequency spectrum. When you listen to a mono signal on two speakers, you hear a false or 'phantom' image which seems to float midway between the speakers, but because both speakers are contributing to the sound, the impression is of a slightly over-inflated level of bass. Listening to mono via one speaker — the way everyone else will hear it — reveals the material in its true form!

A single speaker in a sealed enclosure is the classic means of monitoring in mono.

Checking the derived mono is always best done in the monitoring section of the mixer or with a dedicated monitor controller. Although a mono signal can be derived in the output sections of a mixer (real or virtual), this is potentially dangerous — if you should forget to cancel the mono mixing, you'll end up with a very mono final mix. It does happen, believe me! Sadly, very few monitor controllers outside of broadcast desks and related equipment provide facilities to check mono on a single speaker. Most provide a phantom mono image, which is fine for checking imaging accuracy and phasing file:///F|/SoS/SoS%2011-2005/qa1106_4.htm (1 of 2)10/19/2005 9:43:11 PM

Q. What is the difference between mono with one speaker and mono with two?

issues, but no good for checking the mono balance. Published in SOS November 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

file:///F|/SoS/SoS%2011-2005/qa1106_4.htm (2 of 2)10/19/2005 9:43:11 PM

Q. Why is the signal louder when it is panned to the centre?

Q. Why is the signal louder when it is panned to the centre? Published in SOS November 2005 Print article : Close window

Sound Advice

When I plug my guitar into my 16-track and send the same signal to two channels, if I pan both channels to the middle it sounds louder than if I pan one all the way left and one all the way right. Surely it should sound the same — if they are both in the middle, the signal is coming through both speakers, and if one is panned left and one right, it's still coming through both speakers. Can you explain what's going on? SOS Forum Post Technical Editor Hugh Robjohns replies: Panning laws vary between products, depending on whether they are designed to maintain constant voltage, constant power, or a compromise between the two. The compromise version is probably the most common these days, with pan pots designed to provide something like a 4.5dB attenuation when at the centre. Constant power gives 3dB of centre attenuation, while constant voltage gives 6dB. If you pan identical signals fully left and right, you have full-level signals in each output channel. However, if you pan the signal to the centre, the left and right outputs will be attenuated by (in the case of the common 'compromise' panning law) 4.5dB. But because you have panned both input channels to the centre, each output channel is receiving two lots of signal, each 4.5dB lower than the level of a single channel panned fully left or right. Since your two signals are identical, they will sum together and the level will rise by 6dB. So we go up 6dB from -4.5dB and find that each output channel is now carrying a summed mix of +1.5dB. Hence, each output channel is now carrying a signal that is 1.5dB higher than it was when you panned the channels individually left and right, so it will sound slightly louder. For the record, if the desk employed the constant power law, with 3dB central attenuation, the two channels panned centrally would produce an output of +3dB in each channel, while a desk with the constant voltage law would produce an output that was exactly the same level as the fully panned channels (in terms of signal voltage, at least). The constant power panning law is used where you want a panned signal to remain at more or less the same perceived volume regardless of where you pan it. However, this panning law looks wrong on the desk meters, which only show a constant level if you use the constant voltage law! Hence the halfway-house compromise law, which tries to satisfy the demands of both situations reasonably well. Published in SOS November 2005 file:///F|/SoS/SoS%2011-2005/qa1106_6.htm (1 of 2)10/19/2005 9:43:15 PM

Q. Why is the signal louder when it is panned to the centre?

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

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

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Leader: Happy 20th Birthday SOS!

Leader: Happy 20th Birthday SOS! Paul White Published in SOS November 2005 Print article : Close window

People : Industry/Music Biz

Welcome to this birthday issue of Sound On Sound, which enters its 21st year this month. I've often commented on how much music technology has changed over this period, but other aspects of our lives have also altered considerably. These days, SOS is colour throughout, but in 1985, most music magazines used it only for the cover and special features (although SOS always had more colour pages than any other music mag). Back then, the magazine text was edited on BBC Model B computers, and pages were physically pasted together with wax! The role of computers in music in 1985 was barely out of the experimental stage, MIDI sequencing and recording was new and radical, and samplers cost around the price of a small apartment. If you had an eighttrack recorder you were seriously into recording, and if you had 16 tracks, you were probably a pro. Some years later, I remember saying that I thought it would be a long time before digital reverbs became available for less than £100, or hard drives became as cheap as two-inch tape. As it happens, I was right — it did take quite a few years — but now hard drive space is a fraction of the cost of two-inch tape, and cheap digital effects processors abound. It wasn't so long ago that £1000 didn't seem like an unreasonable amount to spend on a mediocre digital reverb — there was no alternative if you wanted your recordings to sound in any way professional. I recall, when hard disk editing came onto the scene, being very pleased at getting a trade deal to buy a huge 650MB hard drive for only £2500 because I'd finally be able to load an entire album onto my multi-thousand-pound Digidesign Sound Tools card and interface! Who would have dreamed in 1985 that nearly all of that expensive hardware would one day be swept away by the computer revolution, and become available at commodity prices? But that's progress — all sorts of things have changed since SOS first appeared in the shops in 1985 including, perhaps most noticeably, the Internet, which now permeates almost every aspect of our lives. The SOS web site, with its massive archive of articles, is now a major international resource and attracts a vast number of visitors, many of them music-technology students. And that's another thing; back when we started there were no music-technology students, other than those on the BBC training course. You couldn't even find books on the subject. file:///F|/SoS/SoS%2011-2005/leader.htm (1 of 2)10/19/2005 9:43:40 PM

Leader: Happy 20th Birthday SOS!

These days recording systems are almost invariably digital, and that applies to video as well as audio setups — something that has made it possible for us to produce our own DVD material and edit it in house. It involves a lot of work, and it's an on-going learning experience, but as with life, the journey is also the reward. I've really enjoyed getting involved with producing DVD content, though it can be very time-consuming — what you watch in five minutes could take several weeks to produce and edit. And to set the record straight, the Studio SOS visits that we film have no more pre-planning than a phone call. If these features have a fly-on-the-wall look about them, it's because that's exactly what they are, and in some home studios, the walls are rather too close for comfort once you also get a couple of cameras in there... There's a lot of interesting and informative material on our second DVD, including the entire November 1985 issue of Sound On Sound (80 pages!) in PDF format. International readers will be seeing the disc for the first time — so please let us know what you think. As always, we're interested in your feedback, both on the DVD and on the direction of the magazine. Now, here's to the next 20 years — who knows what might have happened by then? Paul White Editor In Chief Published in SOS November 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

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Manu Katché

In this article:

Uniting Nations Sketchpad In The Neighbourhood Drum Sound Facing The Future A Tune A Day Manu Katché: The DVD

Manu Katché Session Drumming & Music Technology Published in SOS November 2005 Print article : Close window

People : Artists/Engineers/Producers/Programmers

He's one of the most famous drummers in the world, having played on more than 200 records. His CV reads like a Who's Who of English, American and French popular music, and even if he prefers not to lose himself in computers, he always takes a Powerbook with him, to write songs, at home or on tour. Meet Manu Katché, drummer extraordinaire. Franck Ernould

"Does it make sense to spend several weeks in a recording studio, working on the same album, when you could do it in days? Yes. Is this time spent thinking, going deeper into the music and the performance? Yes. Is this time useful to get to know the artist you're working with better? Yes. Is it time for pure pleasure, to enjoy being there? Yes. Of course, studio hours are very expensive, but all these factors are human. For a musician, it is of utmost importance to know, as closely as possible, the artist's code, the artist's world. Being together almost 24/24, not only in the recording studio, allows us to define the artist's personality more precisely. We joke, we chat, we laugh together, we talk about more Photo: Jean-Baptiste Mondino or less serious subjects, but in the end, we get into each other's minds; we have a better understanding of why the song exists, why the artist wrote it, and how we can contribute to enhance it. When you have plenty of time, you can try a lot of things: you take a lick here, play it there, everything goes better, you can go back if you're not happy... It's a fact: when you listen, a few years later, to an album made this way, it often doesn't seem old or old-fashioned, because the music has been thought, worked and reworked, conceptualised, deepened. There are good reasons not to hurry!"

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The man who speaks has spent most of his life in a recording studio or on tour. He has developed an immediately identifiable drum sound and style, and he has played with the planet's greatest singers and producers. His name: Manu Katché. Born in 1958 near Paris, France, Manu Katché studied percussion at the SaintMaur Conservatoire, then earned a Grand Prix at the Paris Conservatoire; he also played drums, and with a love of jazz, he soon became a busy session drummer. He was perfectly capable of playing in 'standard' ways, but for other projects, he developed his own style, incorporating high-pitched sounds, distinctive syncopation, tiny splash cymbals used as punctuation, tom hits where nobody else plays them, and playing between beats rather than on them. Manu was soon working with the best French singers, and even producing records with his friends Kamil Rustam (guitar) and Jean-Yves d'Angelo (keyboards). The Michel Jonasz album Unis Vers L'Uni (1985) included some number one hits and earned the trio a Victoire de la Musique, the French equivalent of a Grammy. After this, Manu's phone never stopped ringing, and he eventually joined Peter Gabriel's team for the So album and tour. Sting invited him to play drums on his 1987 Nothing Like The Sun album, and since then, he's been the drummer of choice for both artists, on stage and in the studio. Unlike some drummers, he's never been an enemy of technology: to be able to play back, live in concert, the intricate drum programming on songs like 'Don't Give Up' and 'In Your Eyes', he used a Roland Octapad and an Akai MPC, and triggered pre-produced sounds with contact mics placed on his toms' shells. Since then, Manu has played with, among others, Dire Straits, Tears For Fears, the Christians, Joan Armatrading, Simple Minds, Rick Wright (Pink Floyd) and Joni Mitchell, as well as many French artists such as Laurent Voulzy, Francis Cabrel, Véronique Sanson, Stephan Eicher and Michel Petrucciani. Last year, he could even be seen on French TV, as a jury member for musical reality show La Nouvelle Star.

Uniting Nations Manu is in a unique position to compare recording styles as they differ across the English Channel. "Beyond the stories about people spending two days just to get a snare or bass drum sound — a situation I never experienced myself — English and American artists know how important it is to spend time together, to get to know each other better and to trust each other. I was raised the French way: when you are in the recording studio, everything must go quickly, there's no money, so you must record four songs a day! So, I developed the ability to understand at once how the song's built: I listen to it once, then I go and I know immediately what to play. The first time I worked with Peter [Gabriel], I installed my drums in the studio, the sound engineer placed his microphones, I played a song, we listened to it... and we had a break! We had tea, we told each other stories, then we played again, the we had dinner — and same thing the following days. In England or in the US, when you record music, you take your time. No wonder resort studios are so busy!

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"Sometimes, it's good to record quickly, to favour spontaneity. If you have too much time, you can get lost: you have to find the right attitude. On Peter's last album, Up, we could record five, six or seven long takes in succession — 15 minutes long or more — for the same song. We let things go, we didn't try to be concise, we lacked focus. Peter never trashed anything, he kept it all on the computer, and took some bits here, some bits there... That's a risk with Pro Tools: it's so easy to recombine things, you end up keeping not-so-good material and playing around with it. That's not my favourite production style. I preferred So, it was much more direct. 'Sledgehammer' was recorded in three takes, just before we all went to the airport!"

Sketchpad Manu doesn't have a home studio, as he doesn't want to delve into technology, but he owns a portable setup: an Apple laptop running Logic Pro, a Digidesign M Box interface, a pair of Genelec monitors, a microphone and the necessary cables, all in a suitcase. "That's my musical sketchpad. I take it with me everywhere I go, and in 10 seconds, I'm ready to record. I use a limited sound palette, I don't want to get lost in the plug-in preset lists. I chose some fairly basic sounds I like: piano, bass synth, and simple loops. I don't lose time with loops: I take two bars and that's it. On tour, if I want to work on songs with, say, Dominic Miller [Sting's guitarist], he comes into my hotel room, he listens to my synth gimmick, he has an idea, I record him, and the whole song progresses this way. So I get a better idea of a song's potential, and even if the sounds are not final, they're pretty close. When the music goes, it's essential not to stop the creative flow, to stay simple, not to try something when it's not needed." On some occasions, 'temp' sounds even end up on the final mix. Manu has written music for several movies, including When Stars Meet The Sea, a movie made in Madagascar by local director Raymond Rajaonarivelo. "It tells the story of a young boy looking for his father on Madagascar, with references to voodoo, marabouts, and so on. I had the idea to use the duduk as the main instrument. I knew a Katché at his drum kit. virtuoso duduk player, Levon Minassian. It's a nightmare of a doublereed wind instrument — very old, male and female at the same time, difficult to play and unsample-able. On the demos I used African sounds — some kora samples and other ethnic sounds. When we got into the studio, I transferred my Logic 5 projects onto tape, then Levon came and played over what I already had. The director was there, and was so pleased with the result we ended up using a lot of 'temp' sounds direct from my sketchpad. We redid some guitars, basses and drums, but in the end, working this way allowed us to save time and money.

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"I know there are some extraordinary tools, like [Spectrasonics] Stylus, which we used on Sting's most recent record, Sacred Love. I remember that on a song called 'My Beautiful Smile', not included in the international versions of the record, but only in the Japanese version, the producer had programmed the drums with Stylus, and the result was pretty amazing. I recorded my drum parts on the song, then we compared them, and it was really difficult to choose. In the end, we kept Stylus in the intro, then my real drums come in. I told myself that the era of TR808s and RX7s was definitely over!" On the other side, even the best tool in the world will become boring if used by someone who doesn't know much about music, or lacks experience, or simply has nothing to say. "There are no more accidents, no more risks taken on records, like Janis Joplin did, for example. In music, an imperfection can become a quality: that's precisely what makes good musicians! Industrial, risk-free music produces songs whose lifespan is extremely limited. Listen to the first Police recordings: on some songs the tempo gets faster and faster, that's audible, but nobody cares about that, it didn't keep the songs from being number one! That was Nigel Gray producing — Hugh Padgham would never have left that, he liked to sample with the [AMS] RMX16 to get quantised and aligned beats. Trevor Horn was inflexible about timing too, but if you compare his productions to what we have these days, it was very audacious, miles away from today's 'on the grid' sound."

In The Neighbourhood For his first instrumental solo album, released this September, Manu Katché enjoyed working with a famous producer: Manfred Eicher. Boss of the ECM label since 1969, Eicher has released more than 900 records, and he combines the executive and the artistic aspects of producing. "Manfred has this rare ability: he has a musical intuition to tell him how to combine musicians from different countries, with very different personalities and styles, even having classical performers playing with jazzmen. Nobody else would have had the same ideas, and the results often sound fabulous! He even had the idea of having me play a duet with American pianist Keith Jarrett, but the project was cancelled when Keith fell very ill. "Manfred chooses the musicians, books the studio, is there during the sessions. He leaves total freedom, many takes, he's open-minded, but he knows perfectly well where he wants to go. He tells us 'This is the good take,' we work together on it, he edits, he mixes, he always listens to what we tell him, but in the end, he's the only decision-maker. On my Neighborhood album, I played with Jan Garbarek [saxophones], Tomasz Stanko [trumpet], Marcin Wasilewski [piano] and Slavomir Kurkiewicz [double bass]. We had two days to record, in Rainbow Studio in Oslo, which is owned and operated by veteran sound engineer Jan Erik Kongshaug. Manfred mixed the album with Jan Erik in one day. That's the rule: most ECM file:///F|/SoS/SoS%2011-2005/manukatche.htm (4 of 9)10/19/2005 9:43:44 PM

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records were made this way." Some will be surprised by Neighborhood: it doesn't sound like a solo drummer's album! Manu doesn't hit his drums hard, or play tons of drum rolls at lightning speed, or show any displaced virtuosity. Every sound is subtle, and the production is intimate: "I don't consider myself a 'drum monster'. I don't think I've made the instrument develop like Steve Gadd or Peter Erskine did. They have a real musical touch, they changed the idea of drumming itself — it was considered as a percussion instrument, but they made it melodic."

Drum Sound Since we're talking to one of the most famous drummers in the world, let's take his advice on recording the instrument. When you hear a nice drum sound on a record, where does this come from? The drums? The room? The microphones? The drummer? "There's no rule, actually. The final drum sound may come from a good recording, or may be built at the mixing stage. Let's talk about the drum sound on 'Slave To The Rhythm', a Grace Jones song Trevor Horn produced in 1985. Trevor told me the drummer could not keep the beat, he kept on changing. After a certain number of takes, Trevor chose two bars and looped them. It's the playing which does the trick, there's a hell of a groove. Trevor enhanced the sound at the mixing stage, but even then, it's the way the drummer played that makes the song groove. "It all depends on what you want to do with the drum sound. Even today, some sound engineers prefer to damp the drums, like when I began, at the end of the '70s. That was the disco era — Kleenexes on the snare drum and every tom, blankets in the bass drum, low studio ceilings, no acoustic resonance, no ambient mic, overheads over the cymbals. Like the rhythm machine sounds we had! Hugh Padgham came a few years later, and on Peter's third album, he created Phil Collins' drum sound. He opened a door there, and a few years later, drum sounds had become very live and ambient." It's the drummer who makes the sound in the beginning, and if he doesn't work with the sound engineer, nothing good can happen. "I worked once with one of the greatest producers in the world: Glyn Johns. He produced a Joe Satriani album I played on in 1995. He recorded my drums with only three tube microphones: one in the bass drum, one overhead above the snare drum, miking the charley, the left cymbal and the alto and mid tom, and At the desk in Ferber Studios, Paris. one microphone above the floor tom, just under the ride cymbal. The sound he got was amazing! But I had to be very careful: with such a setup, the sound engineer is trapped. What he gets on tape is what the drummer played, he can't file:///F|/SoS/SoS%2011-2005/manukatche.htm (5 of 9)10/19/2005 9:43:44 PM

Manu Katché

change the balance. "With these three microphones, I was back in the '60s, in the best jazz recordings: the drummer has to balance his playing, be subtle, and not smash everything because he suddenly feels like it, as it would overload the microphones. Drumming is a real job! Imagine you have to record a pianist who plays all the bass very loud with his left hand: if you let him play like that, you won't be able to use the tracks, there will always be too much bass. A real musician has to balance his playing, to learn to be measured, to give sound to the microphone. Glyn told me, right at the beginning of these sessions: 'You're a top drummer, you have to master your playing, to tune your drums. There's no reason the sound engineer should have to play with your sound once you've left the studio!' That's pretty logical, and very true! "In fact, there are two ways to play: in the studio and on stage. In the studio, you have to have discipline, to get control. The microphones are near you, the studio is silent, the slightest sound is audible, so you have to find the right balance. On stage, the conditions are all different, even for the public, and you can free yourself. I remember an interesting experience at a NAMM Show. Yamaha had organised a Drums Night, and all the drummers they endorsed played on stage. Same drums, same microphones, set up the same way, same sound engineer, same sound system. The only thing that changed was the drummer himself. They sent me the DVD they shot during the night, and believe it or not, there were incredible differences from one set to the other — you could tell immediately if it was Erskine or Jordan playing. When the drummer changes, the drums sound different."

Facing The Future At 47, Manu Katché admits he's an 'old generation' studio musician. He witnessed the shift from analogue to digital and from tape to computer, in the 'pro' domain and in home studios. For him, there's a risk of confusion: "Today, with a laptop, the world of musical creation seems wide open. All these easy-to-use music programs lead people to think anyone can become a producer instantly, and that, as long as you know how to use the tools, you are a master in the domain, and you will produce masterworks. I don't believe this at all, it's a lie. People forget that music cannot be restricted to technical issues to do with computer programs. All this powerful and affordable software and hardware is good news for the music industry, for the home studio sellers and for computer makers, but it's dangerous. It's so easy to get lost in all the menus, the possibilities, the configurations, the setups... I respect people using these modern technologies, but they're alone in front of their computers, they lose perspective, and what they produce is not interesting any more." So, everything's negative about music and computers? "Not at all! What people like Moby or MoJo create, for example, is very original, pretty sophisticated and well done. You can still find things to criticise, but the fact is these people have been discovered and made it thanks to their personal use of technology: perhaps 15 years ago, they wouldn't have been so successful. The computer and the tools made them develop an artistic sense, and allowed them to create a world they

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Manu Katché

wouldn't have even dreamt of if they had needed real musicians and expensive studio time to build it. I admire this young generation: some of them never get lost, even if they're alone in front of their computers. Now, there's not organic music any more, there's electronic music, and some tunes they make are really incredible. "It's a question of generation. These young people were born with the Atari, and they use the computer in a 'primary' way. This is really a compliment: they master the digital tools, they know instinctively how to use them, they go directly where they want to, they use the machine only for what they aim. That's amazing! We, the old generation, we learnt music in a different way. We knew analogue multitrackers and consoles; we had to adapt ourselves to digital computer audio workstations, but we kept the reflexes we had from the old days: 'OK, I have this sound, what if I try this effect? No... this one? Finally, what about EQing it?', and the possibilities were not endless. Today, using this kind of approach is the ideal way to get lost among the endless possibilities of a computer software's menus and submenus. "Another point is that in the recording studio, there were always many people working together at the same time: musicians, recording engineer, producer, arranger... There was lots of talking, a direct contact between human beings, negotiations and compromises had to be made, and in the end, all this was beneficial to the music. All these relationships disappear when you're in front of the screen of your computer. You lose perspective so easily, and nobody's there to tell you you're going in the wrong direction, or to give you good ideas."

A Tune A Day Manu Katché is also emphatic about the importance of tuning drums. "This doesn't mean a note per drum, even if some percussion instruments, timpani for example, produce recognisable notes, with a precise pitch. Nor does it mean that you will have a fifth interval between, say, the floor tom and the mid tom. The notion of drum tuning is wider: it's about creating 'harmony' between the different sounds. For example, I tend to like high-pitched snare sounds, with plenty of harmonics. So my toms must sound the same way, but with more depth. I won't use plastic transparent skins, like the Diplomat, but the white Ambassador ones, which have much more resonance and harmonics. I tune the snare first, then the other drums. And even if the drum sounds don't have recognisable pitch, I'll make the snare and toms higher or lower according to the song's key." Manu even tunes his drums according to the artist's personality: "If I play with Peter Gabriel, I tune my drums differently. If it's for my own music, then I'll unstretch the skins and the rattling chains, leave the snare to resonate a little. With Sting, I'll dampen the snare a little more. With Peter, its sound would be rounder. And in a drum kit, every sound is relative to the others: that's why it takes time to tune drums — and that's without even mentioning the mechanical problems we always have to fix... It's very important to give the drummer time to install his drums in the studio, and to feel how it sounds." Manu has some other advice to give to recording drummers: "Always replace the drum skins before a recording session! In France, this is not a tradition, but it's file:///F|/SoS/SoS%2011-2005/manukatche.htm (7 of 9)10/19/2005 9:43:44 PM

Manu Katché

very important. When I am in the studio, I replace the skins every day. It's like using new balls when you play tennis: with old balls, you can't play well. If the drum skin is 'tired', no matter how you play, the sound's gone. Same thing with the cymbals, ride or crash: playing them hard, temperature differences and travel, like on tour in rock music, sucks their sound out, and when they're too old, they lose harmonics and release, and it's not so bright any more. That's true with modern cymbals, not with old ones, which were made differently." With all his knowledge, does Manu tell the sound engineer which microphones to use, and where to place them? "Never. The only thing I ask for is 'No noise gate during the recording.' Sometimes, the engineer comes to me and tells me, for example, that my snare drum sound has too many harmonics. I answer 'That's normal, it's my sound, and don't be afraid, you may think, when you listen to it alone, it's too high, but once this sound will blend with the others instruments in the mix, it will go in place.' Usually, I don't know the room, the microphones are different, the listening conditions are different, so I re-tune my drums, and I never record anything final before having listened to a proper balance of what we played, with the others. I trust the sound engineer, if he knows his job he knows what to do and respects the musicians' desires. I have my own sound, my own style, and people call me for that: if they want a Californian sound, they don't call me! So I don't need to ask the sound engineer for that."

Manu Katché: The DVD It's no surprise that Manu Katché has been asked repeatedly to record sounds and loops for a sample CD-ROM. He has always declined. In 2006, however, he will release a more ambitious project: a DVD video in French, English and Japanese. "There will be a training session, a drummers' game with several levels and pattern puzzles; once you find the solution, you see and hear me playing the patterns. I filmed studio sessions, masterclasses and concerts, there are multi-angle video sequences, there are interviews with artists I have played with [including Sting, Eicher, Gabriel, Jonasz and Voulzy], and sequences shot during the recording of my 1991 solo album. We even filmed a recording session, from beginning to end, with the Neighborhood group: we enter the studio in the morning, we set ourselves up, we record, we discuss... and in the end, we go away with a CD with the rough mix of the song, a Burt Bacharach cover." When we interviewed him, Manu was right in the middle of recording sessions at the Ferber recording studio in Paris, with Pascal Danaé (voices, guitars) and Laurent Vernerey (bass): it's a personal project (the three men have played in residence for weeks in Tahiti, but have never recorded together), not signed by a label yet, and put together by French sound engineer and producer Jean Lamoot. "That's a personal album, so I feel free to get a different sound. It's like the DVD: I play like I feel, with my instinct, without thinking of a musical format. We aren't doing it to sell it by the million, but to interest people, to make them feel like playing. It's important to remember that music is something you play — it has a game-like dimension. Let's never forget that."

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Manu Katché

Published in SOS November 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

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Sounding Off:

In this article:

About The Author

Sounding Off: Stephen Bennett Published in SOS November 2005 Print article : Close window

People : Sounding Off

In a desert of noise, the MP3 player is an oasis of quiet... Stephen Bennett

When I was a teenager, there always seemed to be time to listen to music. I'd built a turntable in kit form and filched the interior of an old Radiogram and six or eight speakers to create a 'hi-fi'. If I remember rightly, I had the speakers in various cupboards and controlled the level by opening and closing the doors as I could never get the volume control to work properly between off and LOUD. With this primitive system, I'd listen to music, uninterrupted, for hours on end and came to intimately know and love various albums and artists that still move me today. Every event in my young life was accompanied by music — listening to 'Telstar' by the Tornadoes while making Thunderbirds out of Lego, my first kiss to the sound of 'These Boots Are Made For Walking' by Nancy Sinatra, holidays in Wales listening to Egg and Genesis, homework done to the clatter of Yes, ELP or the soothing sounds of Bread and spending time with my first proper girlfriend in a tower block while the Sunday Top 40 counted down in the background. And there always seemed to be time to listen, carefully, to whole albums. I can't imagine when I found the time to eat or socialise — it just seemed to be endless winter evening in my room amongst my albums. It was all low technology, not the greatest sound, but somehow none of that mattered. These days the soundtrack to my life is work, sirens, drunken passers-by, cats vomiting, US Air Force flybys, the dishwasher, the distorted crap music drifting

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About The Author Stephen Bennett is a musician, writer and film-maker. He lives in Norwich with two cats, four Apple Macs and a Swede. His ambition is to become the best duduk player in East Anglia.

Sounding Off:

If you would like to air out of neighbours' windows on sunny days and people your views in this mowing the lawn and hoovering. I've no desire to column, please send listen to those Industrial albums I bought in the '80s — your submissions to all I need is to open my window to get the same effect soundingoff@ as Einsturzende Neubauten. These days I have a soundonsound.com or decent Naim hi-fi, in a nice room, with well-placed to the postal address listed in the front of the speakers and a comfy chair. So I put on a record and magazine. sit down to listen. And what do I hear accompanying the beautiful beginning of the latest Sigur Ros LP or the Elbow album? Mopeds. I hate these things with a passion. You can hear them coming from about a mile away. Over the next 15 minutes the sound slowly builds up like a demented wasp, closer and closer until, as they pass right outside the house, the windows rattle with the roar of their 'customised' exhausts as the be-tracksuited rider passes, leaning into the wind, at 25mph. Then another 15 minutes go by until the noise dissipates completely and I realise I've missed a whole side of the record. Again. So I put on another album. As the needle hits the vinyl and the piping trumpet of Miles Davis emerges from the speakers, I hear the buzz outside begin again and the whole pantomime repeats itself. Double-glazing doesn't help, as mopeds seem to emit a noise on a frequency that transmits itself on an atomic level through any man-made structure. I have considered putting my hi-fi in a nuclear bomb shelter but I'm not convinced even that would keep out moped noise, though it would stop my cats having to listen to the Art Bears.

Then I bought an iPod — and before any Apple haters turn away, this applies to any MP3 player. I've never been a fan of earlier Walkmen as I've always balked at the idea of carrying bundles of my precious CDs around with me — and the sound that emanates from cassettes is just too horrible for me to even consider listening to. I originally bought the iPod to audition my own mixes and to download plays and comedies from Radio 4 and BBC7. But of course I soon started transferring some of my CD collection onto the machine. Sure the quality isn't perfect but, as when I was a teenager, somehow that doesn't seem to matter. The iPod has allowed me to listen, uninterrupted, to whole albums in one go. Even 80-minute albums, previously beyond my dreams to even attempt to listen to the whole way through in one go without interruption in the living room, can be tackled on the iPod. I take my iPod on walks, I listen in bed, in the bath, in the car and while working. I even started to buy CD versions of my vinyl albums, something I thought I would never do, just so I can hear to them on the iPod. The expensive hi-fi is now gathering dust while my listening has undergone a renaissance. I've rediscovered old friends of my youth and opened many new musical doors. Perhaps a new career awaits me as a personal music player evangelist? 'Buy one now and remember why you love music in the first place!' The soundtrack of my life has become a musical one again and for that, I am truly thankful. Published in SOS November 2005

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Sounding Off:

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

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

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Studio SOS

In this article:

A Radical Rebuild Monitoring & Acoustics Tweaks Setting Up A Sequencer Project Template Bella's Comments Guitar Amp Modelling Tips

Studio SOS Bella Saer Published in SOS November 2005 Print article : Close window

People : Studio SOS

The team rotate Bella Saer's entire setup through 90 degrees in search of a more effective working environment. Paul White

This month's Studio SOS comes from sunny Brixton, where Bella Saer is a singer, songwriter, and composer for TV. Her music-to-picture work comprises commercials and, more recently, documentaries. She has written songs and sung in bands since she was a teenager and has also sung classically, including as a choral scholar at Trinity College, Cambridge. She plans to release her first album Although Bella's studio setup was not particularly complex, the way it was next year and is working on several configured made the working area feel other collaborative projects. Bella cluttered. spent a couple of years as a reporter and producer for the BBC and ITN before deciding to devote more time to music. In an upstairs bedroom of her house, Bella has an Apple Mac G5 running Logic Pro, a setup which she uses for both songwriting and composing to picture. When we arrived, she had the studio set up on a desk in an alcove next to the fireplace, leaving her space for recording vocals and live instruments at the other end of the room, where she'd erected a couple of used office dividers that she'd acquired to provide some acoustic isolation. The floor underneath her desk was a sea of cables and daisy-chained mains connectors and she had a Spirit mixer wired as a monitor level control so that she could add monitor reverb via an Alesis Picoverb. A Focusrite Platinum Voicemaster was patched into one of the line inputs of her MOTU 828 MkII audio interface for vocal and instrument recording via a Rode NTK tube microphone, and an Emu Proteus 2000 provided file:///F|/SoS/SoS%2011-2005/studiosos.htm (1 of 7)10/19/2005 9:43:49 PM

Studio SOS

additional MIDI sounds, though Bella said she didn't use the Proteus as much as her software sampler instruments. Bella uses both an electric piano and an M Audio Oxygen8 MIDI keyboard, and she had both of these plugged into an Emagic MT4 MIDI interface which also fed the Proteus 2000. As well as being cluttered, the studio environment was not acoustically ideal, and Bella's Tannoy Reveal active monitors, although on good stands, were partially obscured by her computer monitors. Also, their proximity to an untreated side wall made the stereo image somewhat unbalanced. Bella was keen to try a different layout, but wasn't sure what was the best way to go. She also wanted any acoustic treatment to be removable, as she wasn't sure how long it would be before she moved house. Although she wasn't entirely sure about the idea at first (she had only just re-configured the system with a sound-engineer friend) I suggested we dismantle the entire setup and then rebuild it facing down the length of the room. This would mean putting the desk beneath the window with one speaker on either side and with the piano to Bella's left. If we could find a way to dispense with the mixer, we'd also have plenty of room on the desk for her Oxygen8 keyboard, computer keyboard, and mouse mat.

A Radical Rebuild

Paul and Hugh decided to completely reorganise Bella's setup, and they started off by disconnecting all the wiring and moving the main studio desk to the room's shorter wall. Bella then applied herself to some dusting while Paul headed under the desk to deal with the mains wiring, gaffer-taping distribution boards to the desk to keep things tidy.

Having talked Bella into this radical rebuild (and having made a good start on her chocolate Hobnob stash!) we set to work, sorting the cables by type and removing the cable ties that had been used to keep them coiled (most were much longer than necessary). While we were doing this, Bella went shopping for an antistatic fluffy duster, some nylon cord, and some picture-rail hooks, as Hugh and I decided these would make a convenient way to mount the Auralex foam panels necessary for acoustic treatment, without making the fixing permanent. She also bought an extra mains distribution board, as we felt she had too few spare sockets.

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Studio SOS

Before reassembling the system, I crawled under the desk and taped a couple of power distribution boards to the desk's metal frame to keep them off the floor. These were surge protected and all supplied from a further six-way distribution board plugged into a nearby wall socket, so we created in effect a 'star' mains system fanning out from one point, which is less likely to lead to ground-loop hum problems. Before deciding on how exactly to configure the system, I created a Logic default template song with a Platinumverb reverb plug-in set up so that Bella could try using this to add monitoring reverb while recording. The MOTU 828 MkII runs happily at buffer sizes of 256 or even 128 samples, so there's no real need to set up hardware monitoring given that the latency is negligible. Bella decided this sounded fine, so we were able to simplify the system by losing the mixer and the Picoverb. The volume control on the front of the MOTU 828 MkII was then used to control the monitor speaker level, and the phones outlet on the same unit was used for monitoring while recording. However, we had moved the interface further from the live area, so Bella needed to buy a headphone extension cable. We arranged the desk with the computer screen behind the line of the monitors and angled the monitors to converge at a point just behind Bella's normal listening position. This is a good compromise if both the engineer and a client wish to hear the mix accurately. All the USB devices (other than the monitor-screen controls) were fed from a USB hub, and the Logic dongle was plugged into the Mac keyboard. A separate desk, which had been used to house Bella's mixer, her old G4, and a printer, was dismantled and the computer system moved to the lounge to give us more space.

Monitoring & Acoustics Tweaks

Bella couldn't fix any acoustic treatment permanently, so Paul suggested threading string through pieces of acoustic foam and then hanging them from the picture rail on hooks.

Once we were satisfied everything was working, we set about trying to improve both the monitoring and recording acoustics. To help tame the low end, we placed a couple of Real Traps bass traps across the front corners, standing them on the floor to avoid fixing problems. More would have been ideal, but two definitely helped. Four-inch-thick Auralex panels were then hung on the side wall (also using

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picture-rail hooks to hang the left panel on the wardrobe doors) with two more hung at the top of the rear wall and one placed above the fireplace to damp down reflections from that area. Checking the system with Hugh's test disc confirmed that the imaging was much better and the low end was pretty even, a fact no doubt assisted by the fact that the Tannoy Reveals don't have a hugely extended bass end anyway. The speakers were positioned on their stands using blobs of Blu-Tac, which is as good a way of doing the job as any, so we didn't need to make any changes there. Bella mentioned that she'd been unable to get her Airport wireless Internet connection working on that machine, despite the fact that she'd bought the machine supposedly set up to be 'Airport ready'. I offered to have a quick look in case it was anything obvious and discovered that, although an Airport card had been fitted, the antenna cable was still coiled up and secured with a cable tie. Once I'd plugged the antennae cable into the card — which is a bit fiddly — it all worked fine. With G5s, you also have to remember to fit the included T-shaped antenna to the socket on the back of the machine (which Bella had done), otherwise the signal can't penetrate the all-metal casing. Some Real Traps Mini Traps

Because we already had office screens available, it were placed in the corners wasn't too difficult to set up a reasonably dead behind the speakers to help with low-frequency absorption. recording zone for vocals and acoustic guitars, but we did rotate the rug to make it easier to roll up, so that she could record guitars over the wooden floor to get a more lively sound. I also mentioned the Frontier Designs Tranzport wireless transport controller that I use in my own studio, as this makes it very easy for musicians working on their own to control the basic essentials of recording without having to keep walking across the room to the computer. Office screens have a fairly thin fabric/padding layer and so are only effective at high frequencies. Bella had hung thick duvets over two partitions, so we kept these as they improved absorption across the vocal range. We suggested Bella work with her back to the corner where the two panels meet, as this would make the best use of the absorption and also minimise reflections from elsewhere in the room, as she wouldn't be directly facing any flat surfaces. Our only concern was that the panels weren't quite high enough, so we improvised a solution by curving our remaining foam panel and then wedging it on top of the office panels above the corner, which put it directly behind the singing position. We made a couple of quick test recordings to confirm that this worked OK and were rewarded by a nicely neutral vocal sound that had very little room tone superimposed upon it. Arranging longer legs or supports for the office partitions would also be a good solution.

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Setting Up A Sequencer Project Template The final task was to help Bella set up a template project in Logic that would allow her to start work without having to set up lots of things each time. As she was using a single monitor, I arranged two Screensets, one showing the main Arrange page and one the Environment audio mixer, so these could be called up directly using The final studio layout was much less numeric keys. Bella also asked for a cluttered, partly because the reorganisation third Screenset to open the Score of the studio routing had rendered a variety window. On the Arrange page, I set up of Bella's hardware redundant. 12 audio tracks, 12 virtual instrument tracks, and 16 MIDI tracks to address the Proteus 2000, as most projects would require fewer than that. A stereo Aux object was then used to bring the stereo output of the Proteus 2000 into the mix, and I also set up an Audio object with the Proteus 2000 as its source so that Bella could record it as hard audio if she needed to. I set up the 'comfort' reverb on a Buss Audio object using Platinumverb (as it is quite CPU efficient), and set up a total of eight further busses giving Bella the scope to create more sophisticated routing without having to add more Environment objects. Bella said she'd always found the Proteus hard to use, as patch selection via Logic was very confusing, so I suggested we do an Internet search for a Proteus 2000 patch-name list. This took around 15 seconds and lead us to the site of wellknown Logic guru Pete Thomas. Thanks, Pete — it worked brilliantly and made Bella very happy! Now, Bella can call up the patch needed for each part according to bank and program names. We simply copied and pasted Pete Thomas's Proteus Multi Instrument object into our new default Environment page so that it would always be available at the start of each new project. I also took this opportunity to sort out the I/O Labels section in the Audio Configuration window so that audio inputs and outputs could be selected using meaningful names rather than raw numbers.

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Bella's Comments "I'm really pleased with the transformation. The studio's new layout means it's now a spacious and serene place to work. Paul and Hugh created a lot of space by getting rid of a whole table full of kit — my mixer, my other Mac, my external monitoring reverb unit, and other stuff that had accumulated over time. I was a little apprehensive about letting them rewire everything, but the result of their rationalisation is a really neat setup — by the time they'd finished there was a pile of unwanted cables in the corner of the room. "The Auralex foam had a dramatic effect on the room's acoustics and I was glad it was the nice blue colour as opposed to the lurid purple that they also stock. I'd only recently plugged in my new G5 and I hadn't got round to setting up a default Logic Song, so Paul's help there was greatly appreciated. Loading up the Proteus bank names from Pete Thomas's site was a stroke of genius — well done, Paul! I had stopped using my Proteus in favour of software sampler instruments because I got bored with loading up Multis all the time on the actual Proteus. Now the Proteus is much simpler to use, so I'll pay it a lot more attention as it has some good sounds. I particularly liked the small touches, like the way my mains distribution boards were gaffer-taped to the mixer stand that supports my work surface. "Thank you Paul & Hugh for coming to my studio and providing a day's top-quality entertainment. And thank you Andy for fixing my fridge!"

Guitar Amp Modelling Tips As Bella also plays a bit of guitar, we experimented with Logic's Guitar Amp Pro plug-in to see if it suited her picking style. The first two channels of the MOTU interface have high-impedance inputs suitable for guitars and basses. As Guitar Amp Pro doesn't include effects other than basic spring reverb and tremolo, anything fancier has to be added by placing plug-in effects after it. Logic's various modulation effects really appealed to Bella, particularly the Mod Delay, Chorus, and Rotary effects. She was also quite intrigued as to the possibilities afforded by the tempo-synced Tremolo/Panner. We took the opportunity to go through vocal compression with her, in order to provide some tips on achieving the best settings. Using Logic's own compressor and starting with the default settings, we reset the ratio to around 5:1, switched from Peak to RMS mode (Peak is more appropriate for percussive sounds), and then adjusted the Threshold control until the absolute maximum amount of gain reduction on loud vocal notes was around 8dB. This thickened the sound and ironed out level changes quite effectively without obvious side-effects, so we left file:///F|/SoS/SoS%2011-2005/studiosos.htm (6 of 7)10/19/2005 9:43:49 PM

Studio SOS

the Attack and Release settings at their default values. Even if you use one of the included presets, you still have to adjust the Threshold to get the desired amount of gain reduction, as all compressors are dependent on the recorded level and dynamics of the signal being treated. Bella's usual strategy had been to apply a little compression while recording vocals using the Focusrite Platinum Voicemaster, adding a little more in software if necessary, and we had no reason to suggest that she change this perfectly sound approach. By the end of the day, Bella's studio was looking much tidier, and both Hugh and I had spent a while Bella's vocal recording area, vacuuming and dusting so that the assembled complete with Paul and system was free of dust. Furthermore, Andy Hugh's improvements. Brookes, (who had escaped from the SOS graphics production area to act as one of our camera men for the day) had also adjusted Bella's fridge-freezer to stop it dripping on the floor. Given more time he would probably have boxed in the pipework around her boiler as well! We all agreed the monitoring environment was better, the ergonomics of the system better, and the system generally more streamlined and easier to use. Certainly the studio was no longer cluttered by excessive Hobnobs by the time we left! Bella also had her Airport system working and so no longer needed to trail an Ethernet cable out of the window and into the adjacent room every time she needed a software update! We tested this new-found freedom by downloading the Spectrasonics CD Joiner patch needed to make fresh installations of their virtual instruments under Mac OS X — an evening job we left for Bella... On this month's SOS DVD002 you can see for yourself how Studio SOS transformed Bella's recording setup, plus you can listen in on Paul and Hugh discussing the various recording challenges with her on the day, and also check out some of her own music. Published in SOS November 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

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The Go! Team: Recording Thunder, Lightning, Strike

In this article:

Straight To Video Going Further Cleaning Up The Act Outboard Magic Remote Mixing It's Supposed To Be Like That!

The Go! Team: Recording Thunder, Lightning, Strike Gareth Parton Published in SOS November 2005 Print article : Close window

People : Artists/Engineers/Producers/Programmers

The Go! Team's debut album, a glorious pile-up of mangled samples and lo-fi home recordings, is now attracting widespread acclaim — but its path to Mercury nomination and commercial success has been anything but smooth. Tom Doyle

Recorded in a garage in Swansea, mixed almost entirely in mono and offering a headspinning brew of Northern soul, electro, cheerleader chants, Charlie Brown-styled piano and abrasive, Sonic Youth-inspired loops, the Go! Team's debut album Thunder, Lightning, Strike is one of the most unusual and innovative records of recent times. Essentially the work of Photos: Richard Ecclestone Brighton-based Ian Parton, formerly a Gareth Parton at Fortress Studios, where maker of documentaries for the Thunder, Lightning, Strike was mixed. Discovery Channel, the Mercury Prizenominated album began as a series of sample-collage experiments fitted in between overseas filming trips. The record's co-producer, Ian's elder brother Gareth Parton, has been engineering at various studios including Strongroom, Livingston and the Church since the early '90s and was brought in to help with the project's pre-production, long before the operation moved to Fortress Studios in North London for mixing. "It's Ian's original concept and on the record it's pretty much him playing everything apart from the samples," Gareth explains. "He's been mucking around with it for years and years. The original demos he did were with an Atari 1040 running Cubase and an S1000 sampler in about 2000. But ever since I've been file:///F|/SoS/SoS%2011-2005/goteam.htm (1 of 10)10/19/2005 9:43:52 PM

The Go! Team: Recording Thunder, Lightning, Strike

working in studios he's been popping in doing stuff similar to the Go! Team." Very much a one-man operation in the initial stages, the preliminary Go! Team sessions found Ian Parton bravely attempting to engineer and record everything himself at the Partons' parents home in Wales. "We'd set up the desk in their garage and have tie lines going up to the kitchen where he'd record the drums," says Gareth. "He'd be pressing Record and then running up and down the stairs. Frustrating way to do it, working on his own, but he had to do that for a while to get the original demos done. I would go down and help him set up — mic the kit up and then leave him to it because I'd be working on something else and he'd be down there for a week." In keeping with the retro-futuristic, lo-fi concept of Thunder, Lightning, Strike, the original tracking equipment was devoutly old-school. "Ian had an Otari half-inch eight-track and an old Soundtracs desk. He'd stripe timecode and have the samples running — bits he'd nicked from charity-shop records or whatever. The drums would go down live and then be bounced to one track and he'd use the other seven tracks to fill with whatever he could."

The control room at Fortress Studios is based around a Neve desk with Flying Faders automation.

Were the demos fairly loose-sounding then? "Yes! Too right. But that's part of it. It's supposed to sound spontaneous and fun, like a band jamming in a room. It's not supposed to necessarily sound like these are samples and these are live instruments and it's all glued together. It's an absolute mish-mash." Gareth admits that there is a specific philosophy behind the sound of the Go! Team album — a concept album in all but name. "It's not just lo-fi and cobbled together, it's deliberately sounding like that. The vocal samples are quite often taken from VHS, the rappy cheerleader stuff is from straight-to-video films. And 'cause they sound fucked, you'd kind of have to make everything else sound a bit fucked around it. The mono thing is intentional as well. If you've got a bunch of samples and you start panning them around, it starts sounding quite fake. The Northern soul samples in there, there's not a huge amount of bass on those records, so that was the kind of angle we were going for."

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The Go! Team: Recording Thunder, Lightning, Strike

Straight To Video Most of the work in preparing the 'clean' version of Thunder, Lightning, Strike involved recreating samples as closely as possible, but there were two tracks where new vocal lines were recorded. For the catchy proto-rap rhymes of 'Bottle Rocket', the Go! Team's live MC Ninja was brought in to perform the track. "She changed the words and it's got the same kind of feel, but it's mixed slightly more as a single. The original vocal was extremely quiet on there, you couldn't make out the lyrics at all. But that was deliberate as well." Then, as luck would have it, one of Memphis Industries' co-founders was visiting the Stoke Newington Festival when he chanced upon a troupe of teenage girls from South London performing Go! Team-like Double Dutch chants. Within weeks, they'd been spirited away to Fortress Studios to record the new vocals for 'Huddle Formation'. "We got them down the studio and they were ace," Gareth enthuses. "The new 'cheerleader' vocals had to sound like they were taken off VHS, so I did just that. Recorded them onto a VCR at high level and I even added a very high-pitched 16KHz whistle at low level to make them sound like they'd been taped off the telly in 1984."

Going Further When the Go! Team demos attracted the attention of North London-based boutique indie label Memphis Industries, Ian Parton suddenly found himself with a budget — albeit a limited one — to finish the album. Gareth admits that it took some time for his brother (who he reckons would "happily master everything to cassette if he could") to accept his advice and take the leap into the digital domain. "Now he's got Pro Tools with a Digi 001 and a G4 Mac running OS 9. He didn't want to go Digi straight away because of the sound of it, basically. But I convinced him that as long as you get the analogue process in there to start with — slam it to tape, get it through the desk and crank it up — then transfer it to Pro Tools, you can do all your editing. Which was a Godsend really, 'cause he's a good drummer, but when you're playing on top of loops, it's getting a balance between being totally tight-ass, 'cause that's not what this is about, and being... acceptable. It wouldn't necessarily be going into Beat Detective or anything like that and completely going 100 percent tight-ass on it. It would be getting a bunch of takes, getting the greatest hits and thinking 'That's the best fill there,' or 'That's a fantastic feel for the verse,' or whatever." Gareth sees his role in the project as being something of a sweeper-upper. "He comes to me with what he's done and I try to make it sound a little bit posher." Was Ian happy with Pro Tools in the end then? "Oh yeah. And I don't think he's used his S1000 since. Now he just tends to use [Serato's] Pitch 'n Time to stretch stuff. You can kind of do it in real time a bit better with an S1000, but it's trial and error sitting there with a guitar tuned to the note you want it to go to. He's got a lot of patience to do what he does on that side of things."

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The Go! Team: Recording Thunder, Lightning, Strike

Once the proper album sessions had begun, in general, the original methods stuck, including the Partons' strippeddown kit-miking technique. "We tend to mainly use an ambient mic for the drums, a Rode valve — budget stuff because it was on a budget — a 58 for the close snare mic that didn't really get used that much in the mix and then a D112 on the kick. What I like to do, The live room at Fortress. which we used on quite a lot of the stuff, was place a PZM on the floor underneath the snare, so you're getting the under-snare sound and the splat off the front of the kick, but you're not getting much of the top kit, so if you want to distort the fuck out of that, you can. You're getting the crunch of the kick and the snare without the room ambience, which we tended to distort anyway. It was just a cheap Pearl kit, nothing very posh at all. Almost stubbornly, deliberately not posh." While the range of instrumentation on Thunder, Lightning, Strike is fairly eclectic — including banjo, recorder and pub piano (as recorded, quite literally on 'Hold Yr Terror Close', in a Brighton bar) — it's perhaps Ian Parton's idiosyncratic selection of samples that really defines the sound of the album. Gareth is cagey when it comes to revealing their sources, mainly because some are still in the process of being cleared. "Most of them are stuff you'd never ever pick up on. It's snippets of charity-shop records that were deleted. People come up to me and go 'Oh yeah, that's the sample from Bob Dylan,' or something. But it's not. There's a few higher-profile things, but they tend to be cover versions of higher profile acts, like the strings from a cover version of... something or other."

Cleaning Up The Act In fact, there are actually two versions of Thunder, Lightning, Strike: the original illegal, sample-strewn version and the recently released, legally 'clean' version, featuring cleared samples or approximate recreations of them. "On the original version, all the vocals were samples," Gareth reveals, carefully. "But because it went out on a small indie and we thought it wouldn't do anything, none of the samples were cleared. It's not a good thing to do because in the end it's a big pain in the arse. The label didn't really have any choice, they weren't in a position to do what the Avalanches did, to take a year to clear all the samples. It was a case of 'Let's just put the bloody thing out and see what happens.' "The next thing, it started getting a bit of attention and we hadn't made any file:///F|/SoS/SoS%2011-2005/goteam.htm (4 of 10)10/19/2005 9:43:52 PM

The Go! Team: Recording Thunder, Lightning, Strike

contingency plans for having to recreate it without any of the samples. If I'd have realised at the time, I'd have stuck down mixes without the samples on." Considering that samples can cost an average of £6000 to clear and each of the tracks on the Go! Team album features multiple samples, the potential sums involved might have easily bankrupted Ian Parton and Memphis Industries or seen the record being pulled from the shops. Luckily, the album's early success attracted the attention of Sony/BMG in the UK and Columbia in the US who agreed to buy into the record by way of sample clearance funds. "That's kind of saved the record's ass really," Gareth admits. If the original album cost less than 10 grand to make, the producer reckons, then the cost of the remake was easily 10 times that amount. Most of the recordings made

"Some of the samples have been cleared and we've at Fortress were tracked to an had to rewrite the ones we couldn't get publishing Otari reel-to-reel recorder before being transferred to on. Some of them we're paying publishing on and Pro Tools. we're recreating the samples. Which is a real shame, y'know, when you've lived with a record and it's your baby. We had a musicologist come down to the studio and we sat here going through all the samples and he would go 'Nah... nah... can't do that... gotta change that.' By the end of the day, Ian was pale and feeling rather ill. At that stage we'd started recreating the record and the guy was going 'No, that's too close.' But I think what we've come up with in the end is really good — I don't think we've taken away from the spirit of it or the sound of it." In actual fact, the most unpredictable part of the remaking process came from the Partons employing sample-recreation companies to copy certain key lifts. The results, as Gareth recalls with a smile, varied wildly. "There's a few different companies out there who profess to be samplerecreation companies, so we farmed some of the ones out that we thought we couldn't do ourselves. And I'm not gonna name names, but we had a bunch come back and they were shocking. We'd sit here, almost in tears, laughing at what they'd done. These dodgy pub singer versions of [the hook from 'Ladyflash'] 'We came here to rock the microphone'! We thought 'Shit, who are these guys?' And it turned out that most of the stuff they do is ringtones. "We panicked at that stage 'cause we had to make the whole bloody record. But then we found these guys who'd done stuff for Lemon Jelly and Fatboy Slim and they were a lot better. Still, some of them we had to touch up with our own horns and stuff. I don't know what it is with people trying to record horn sections these days, but it always ends up sounding like keyboards. If you're recording in a small room close-miked, multitracking the same trumpet, then it starts sounding really thin and keyboardy, so we had to get some extra horn players in to beef it file:///F|/SoS/SoS%2011-2005/goteam.htm (5 of 10)10/19/2005 9:43:52 PM

The Go! Team: Recording Thunder, Lightning, Strike

up. "I also ended up getting string players in and doing some of the strings in here, just a viola and a violin, and multitracked it, real top-line stuff. In 'Everyone's A VIP' there's a really high-pitched string line that's time-stretched to buggery and distorted. So I had to varispeed the track in Pro Tools and pitch it right down, 'cause they could never get those really piercing high notes in reality."

Outboard Magic Despite being a fan of Pro Tools, Gareth admits that, for the Go! Team sessions at least, he tended to avoid its plug-ins. "Maybe Auto-Tune on a few things. But because Fortress has got a decent bunch of outboard compressors, I tend to go for those. Usually the [Urei] 1176s or the old Dbx 160s and then there's the Alan Smart SSL copy which we tend to use across the mix buss."

Fortress's outboard was heavily used in preference to plug-in effects and processing. Left, from top: ADR stereo compressor, GML EQ, Al Smart stereo compressor, Thermionic Culture Culture Vulture distortion unit, Trident Audio compressor/limiter, Dbx 120XP bass enhancer, Drawmer DS201 gates (x2). Right, from top: Amek dual compressor/limiters (x2), Antares AVP1 vocal processor, Drawmer 1960 voice channel, Valley People Dynamite compressor, Dbx 902 de-essers (x2) and 160X compressors (x2).

Outboard compression aside, the producer is also a huge fan of tape compression. "There's quite a lot of tape processing going on throughout everything. If we're recording new instruments, we tend to record onto the Otari MTR90 in sync with the Pro Tools and then transfer it over. On this stuff I tend to run the tape really hot — lining the machine up to +9 or whatever we can get out of it and making sure the output level isn't too high, so when it comes back into Pro Tools it's not taking the roof off it. We're purely using it just for the effect of it. "When we put the final mixes down, we also go through the Massenburg GML EQ, just for picking things out and making it sound even nastier, believe it or not! All along the way, especially when I was doing the recreation mixes and sending them off to Ian, I thought I'd already given it enough distortion and he'd go 'Can we distort it a bit more?' And he'd spot if I was trying to make anything slightly stereo and say 'Can you bring that in a bit?' I'd try to sneak it in every now and then. I go along with his philosophy, but it's not the way I tend to mix everything because that would be a bit limiting. But on this particular stuff, I think it works really well." file:///F|/SoS/SoS%2011-2005/goteam.htm (6 of 10)10/19/2005 9:43:52 PM

The Go! Team: Recording Thunder, Lightning, Strike

Remote Mixing If the process of recreating the legally cleaned-up version of Thunder, Lightning, Strike was a convoluted affair, then the mix stage of the album's second incarnation provided further complications: Ian Parton was off on tour with the live line-up of the Go! Team promoting the original version of the album, at the same time as the new one was being mixed. "He was in Japan or Australia or America, so I'd spend most of my time on the computer in the office here — from the sample recreation guys emailing their recreations over to me, me slotting them into the rebuilt mix without the illegal samples, dirtying everything up, then having to do an MP3 and email it over to Ian in Japan at, y'know, two o'clock in the morning. He'd be sitting there listening bleary-eyed, going 'Yeah, it's good, but turn the piano up a little bit,' or something, basically directing me from the other side of the world. So then I'd make the changes, MP3 him another one overnight and then he'd have a listen to it in the morning. So it was kind of remote mixing. "At the beginning of the session, I thought 'Wow, this is fantastic, we can do this,' and then you think 'Oh shit, this is really annoying,' 'cause you'd have to work out what time it was on the other side of the world and there's a delay in uploading it and downloading it, so I'd be sitting around for ages just waiting for responses. Quite frustrating." Since all of the original mixes had been Most of the basic tracks for Thunder, done at Fortress, Gareth decided that it Lightning, Strike were recorded at the was best to work in the same room on Partons' Swansea family home, using a Soundtracs desk and an eight-track Otari the same Neve V1 for the new reel-to-reel. versions. "I'm used to working in this room, I've worked down here for years and I'm really comfortable with the Genelec [1030A] monitors. I've got a pair at home as well and a lot of people think they make everything sound a bit too nice, but I think alongside NS10s, they're kind of workable. If you're listening to loud, distorted music all day, I can't do it on NS10s. Familiarity is the thing about NS10s — you've grown up with them and you're supposed to know how much bottom end you can get away with on them. Well, roughly anyway. It's a bit of a dark art down the bottom on those things. "With the Neve, I love Flying Faders. It beats anything. You have to be a computer programmer in order to automate on an SSL. Flying Faders is just grab it, it does it. Because of the way that I work, I don't tend to do a lot of automation within Pro Tools because if you're doing fader rides before you hit the

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The Go! Team: Recording Thunder, Lightning, Strike

compressors on the desk, then that kind of defeats the object of it. So it makes sense to use Flying Faders automation and it's great anyway. But the desk is preVR, so there's no recall on it. We couldn't just press a button and get the recall back, we had to sit there A-Bing the original mixes. That became, like, urgh. You sort of start mixing it from scratch and then you have to A-B all the time to make sure we're going along the same route." To achieve the same distorted crunch of the original mixes, Gareth again used Thermionic Culture's Culture Vulture and hired in the same Neve mic preamps as he had for the first sessions. "Can't remember what model they are, but they're the greeny bluey ones with the chunky knobs," he laughs. "I'd patch the auxiliary into the mic pre and crank it right up and patch it back up a fader. It distorts a different way from a distortion box and it can give it that bit of edge, a slight fuzziness to it and it's sort of smoother."

Fortress's Yamaha NS10 and Genelec 1030A monitors: Gareth Parton says he prefers the latter for most mixing tasks.

If at the end of all of this, the mixing of the second version sounds like it was a bit of an engineering nightmare, then Gareth concedes with a grin that it was. "The problem across the board with Ian's stuff is it's layers and layers of samples. So there can be four bass guitars going off at the same time or four drum kits going off at the same time. To control that is the biggest nightmare. You end up filtering the bass off on just about every sample, which thins it out, but that's part of the reason it sounds the way it does. So that was one of the big problems — sorting out the layers of stuff and making sure we had all the right things coming through. We tend to use filters on the desk quite a lot rather than go straight for the EQs. "The other problem was that some of the stuff actually survived from the very first demo sessions that we did, so the drums for instance on 'Junior Kickstart' survived from the very first time he put it down. There's one track of drums that we've got bounced together with an out-of-time tambourine and there's fuck-all we can do about it, because there's something about that drum take, some really nice fills in there with a great sound, and we tried to redo it and it didn't sound as good. "Also, when that was transferred over, the Otari eight-track has a tendency to drift with the tempo, so the tuning across the whole thing was a bit strange. The samples tended to be taken off vinyl, as well, and sometimes they're between pitching, they're not concert, so everything else has to be tuned around it. So there's a little bit of Auto-Tuning going on, say on the new live trumpets where the player wasn't quite getting the right inbetweeniness and we had to do a bit of tinkering. And on the bass some of the time as well — the intonation on the guitar was a bit wonky. Cheap instruments! Hopefully you can't hear it."

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The Go! Team: Recording Thunder, Lightning, Strike

The mixes were put to an Otari MTR12 half-inch machine and sent straight from the repro head to Pro Tools, although in line with the lo-fi audio aesthetic (and particularly with the low budget constraints of the original sessions), the Partons kept on repeatedly re-using the same reels of tape. Gareth smiles and says "I'm sure it sounds better when it's thinner anyway."

The Go! Team's six-strong live line-up.

It's Supposed To Be Like That! Live, the Go! Team are a 12-legged groove machine featuring Ian Parton on guitar, his two friends Sam and Jamie on guitar and bass, plus two girl drummers, Silke (from Germany) and Chi (from Japan), alongside rapper Ninja. While professing that he was "too old and fat" to get involved in the live side of things, Gareth laid down backing track mixes alongside the album versions. "I think Ian's tinkered around with them a little bit and had to pitch them, because of course the versions that we had weren't in concert pitch, and obviously you can't be retuning guitars all the time." For all its retro values, Thunder, Lightning, Strike is clearly an album that could only have been made in recent times. "You couldn't make it any other time than now, no," Gareth nods. "Not without splicing up a lot of tape and spending about 12 years doing it." Nevertheless, he admits that some people miss the point of its intentionally degraded sonics. The producer admits he's had a few strange emails in the wake of the album's release... "They say 'You can't hear the vocals, it's distorted, it's mono.' But it's supposed to be like that! I get people writing going 'You need to get a better pair of monitors 'cause it sounds shite.' I feel like writing back saying 'Uh... I've actually got Genelecs, mate..." Published in SOS November 2005

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The Go! Team: Recording Thunder, Lightning, Strike

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

file:///F|/SoS/SoS%2011-2005/goteam.htm (10 of 10)10/19/2005 9:43:52 PM

ReBirth & Reason

In this article:

Rebirth of Rebirth That's Not All Folks... Reason On The Up

ReBirth & Reason Reason Notes Published in SOS November 2005 Print article : Close window

Technique : Reason Notes

Rebirth is dead. Long live Rebirth! We discuss the decision to discontinue this pioneering software and discover how Reason users can keep its spirit alive. Derek Johnson

It has been said that Propellerhead, for their second product ever, really wanted to release something like Reason. But computers weren't up to the job in 1997, so Rebirth RB338 was their follow-up to Recycle instead. This accurate representation of the classic acid studio — virtual Roland TR808 drum machine and dual TB303 Basslines — was an instant hit, growing a virtual TR909 beatbox with v2.0. Rebirth, a very early entry into the soft instrument stakes, had quite an eventful life, particularly when inquisitive users discovered how to replace the software's graphics, and even its drum samples. Rather than take umbrage — though they must have been surprised — Propellerhead legitimised this 'mod' culture, releasing an application with v2.0 that allowed anybody to do their own mods. The 2000 release of Rebirth's successor, Reason, could have been Rebirth's death knell. But Propellerhead thoughtfully built a conduit for the software — the Rebirth Input Machine — into their new super studio. Development of Rebirth effectively ceased with v2.0.1, and although some users might like to see the classic software rebuilt for modern times (especially for Mac OS X), Propellerhead have decided that their resources are better deployed on current and future projects. With regret, therefore, Rebirth has been discontinued.

Rebirth of Rebirth

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ReBirth & Reason

But all is not lost: discontinued does not mean dead and buried, as it might at other software houses. First, take a trip over to www.rebirthmuseum.com. Here, you'll be able to gen up on the whole story, and download v2.0.1 of the software for free! For Windows users, that effectively means an extra four modules for their Reason setup, since RB338 still runs on that platform. Any Mac users still on Mac OS 9 can also gain this free Rewire-compatible mini-studio. The download takes the form of a rather large disk image, using the Bit Torrent distribution method. Burn that image to a disk yourself, and you're ready to go. (Mac users can run from the disk image on their desktop.) This is unsupported software, but the museum offers loads of docs, FAQs and a live message board. The museum also has a lot of 'mods' to download, so you can sample some of that late '90s hacking action. There are apparently some people who have Propellerhead salute their breakthrough never stopped using Rebirth; they're to software with the Rebirth Mod Refill. be applauded, but they're also to be tempted with a very favourable (but time limited) upgrade path to the latest version of Reason. It might be time to bite, people! More Rebirth history can be found at the SOS web site (www.soundonsound. com): reviews of v1 and v2 appeared in our August 1997 and November 1998 issues respectively, both available on-line.

That's Not All Folks... If you can't be bothered with running Rebirth, or are glad to be shot of OS9 but still have a soft spot for Rebirth sounds, all is not lost. Propellerhead have 'Refilled' all Rebirth's drum sounds — and they've included a selection of recreations of kits from the 'mod' community. And the 100MB Rebirth RB338 Mod Refill is free! Go to www.propellerheads.se. Now. The Combinator device has been used to group together Redrums, NNXTs, Spiders and other devices to recreate the sound and feel of the various TR808 and TR909 kits. Standard layouts have been used, and Combinator programmer knobs and buttons have been used sensibly and consistently to provide instant access to important parameters. Needless to say, most of the mod kits don't sound anything at all like an 808 or 909, but they've been recreated as accurately as possible, to behave as if they file:///F|/SoS/SoS%2011-2005/reasonnotes.htm (2 of 3)10/19/2005 9:44:13 PM

ReBirth & Reason

were loaded into Rebirth. And this isn't just a patch collection, as all the samples are, obviously, available for your use too. Be aware, though, that this collection follows Rebirth's sample-naming conventions. No matter what the sample is that's meant to go in a particular slot, it still has to have the same name as the sample it replaces. So, for example, a sample meant for the TR808 bass-drum slot in Rebirth will still be called TR808BD, even in this Refill. That's the case even if the replacement is a loop, synth sound or wibbly noise. There's only one annoying thing about this collection: every Combis has been supplied with a 'startup sound' that chimes every time it's loaded. This gets old pretty quickly. It can be disabled but will only stay disabled in future if you re-save the Combi. However, that's a pretty minor quibble for a fabulous collection that costs precisely zero pounds. There's even a 13-page PDF manual, in the full Propellerhead house style, that goes into lots of detail about how the Combis have been programmed and offers backgrounders on the modders whose work you're enjoying.

Reason On The Up You might recall that last month's overview of Propellerhead's beta-testing process mentioned that the recent v3.0.3 update was about to be superseded. We can now confirm that version 3.0.4, which fixes various small but significant issues, is ready for download. It's also worth noting that Reload, Propellerhead's free (to registered users) Akai S1000/S3000 sample conversion utility, has been upgraded. There were, apparently, some 'compatibility issues' with Reason 3, and these have been addressed in Reload v1.0.1. Published in SOS November 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

file:///F|/SoS/SoS%2011-2005/reasonnotes.htm (3 of 3)10/19/2005 9:44:13 PM

Apple Notes

In this article:

Apple Notes

Nano, Nano News & Updates Are You Being Xserved? I Just Called To Say iTunes Published in SOS November 2005 You... Print article : Close window .Mac To The Max Technique : Apple Notes X Marks The Spot For Linux Audio Apps

We round up a month of small, yet interesting, product releases from Apple, as well as looking at how the company is making it easy for Linux developers to port audio software to Mac OS X. Mark Wherry

While Apple withdrew from presenting the usual keynote for the recent Paris Mac Expo, where last year they announced the iMac G5, there's been no lack of interesting Apple product news over the last month. Apple CEO Steve Jobs and Senior Vice President of Worldwide Product Marketing Phil Schiller held a media conference at the Paris Expo and, among the topics discussed, Jobs confirmed that the company was on track to have Intel-based Macs available in June 2006. Jobs also spoke about the iPod and Apple's digital music initiative, discussing many of Apple's product announcements of the preceding couple of weeks.

Nano, Nano Towards the end of August, news reports indicated that Apple planned to buy as much as 40 percent of Samsung's NAND flash-memory output for the second half of 2005, to use in a new flash-based iPod Mini. However, this new flashbased iPod turned out to be the iPod Nano, which replaces the now discontinued Mini, and is possibly the most seductive iPod Apple has ever released. If you haven't seen one already, you're not going to believe how small the Nano actually is. Measuring just 3.5 by 1.6 inches and only 0.27 inches deep, it really is a marvel of design, weighing just 1.5 ounces and featuring a clear 1.5-inch colour LCD screen that can also be used to view photos. The iPod Nano is pretty much functionally identical to the iPod, featuring the same click wheel for control; the big difference is that the Nano only supports USB, while the iPod can be connected via Firewire or USB. The Nano is charged through the USB port, although a separate charger is available, and takes three

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Apple Notes

hours for a full charge or 1.5 hours for a fast charge to 80percent capacity. Apple say that the battery lasts for 14 hours of music playback, or four hours of slideshows with music, and in the case of the Nanos I've seen this is fairly accurate. The iPod Nano also seems to be pretty durable, and if you didn't see the review on Ars Technica (http:// arstechnica.com/ reviews/ hardware/nano.ars/3), it's able to withstand drops from various heights and being run over by a car! In terms of expansion, the iPod Nano features the same 30pin connector as the iPod and iPod Mini, making it compatible with a wide range of third-party accessories. Apple also supply a curious 'Dock Adaptor' that actually has Courtesy of Apple no use whatsoever right now. According to Apple, some The iPod Nano. iPod accessories are designed around specific iPod Small but perfectly dimensions, but since iPods vary in size (and this formed, as they presumes that Apple intend to offer other different-sized say... iPods in the future), third-party manufacturers have to offer different models for different iPods. In the future, accessories will be designed to accommodate Dock Adaptors, so that a single model is compatible with all iPods. The iPod Nano is available in 2GB or 4GB capacities for £139 and £179 respectively, in black or white. At the time of writing, in the USA the 4GB black model seems to the most sought-after . But I have to offer this warning: to see one is to want one. I guess the remaining question, if Apple's marketing and naming strategy holds true, is when we're likely to see a Mac Nano!

Are You Being Xserved? September brought an update to Apple's Xserve RAID storage system, which now supports up to 14 500GB ATA Apple Drive Modules (ADM) for a total of 7TB of storage (previously the highest capacity of each ADM was 400GB). Xserve RAID is available in three standard configurations, with either 2TB storage across four 250GB ADMs for £4349, 3.5TB on seven 500GB ADMs for £6149, or a fully expanded 14 500GB ADM system for £9399. Stay tuned for a future Apple Notes column where we'll be taking a look at the Xserve RAID's suitability for running music and audio applications such as Logic and Pro Tools.

I Just Called To Say iTunes You... The iPod Nano was announced on September 7th at another special music event organised by Apple, where Steve Jobs also publicised the release of the anticipated iTunes-compatible mobile phone. The Motorola ROKR is the result of file:///F|/SoS/SoS%2011-2005/applenotes.htm (2 of 5)10/19/2005 9:44:16 PM

Apple Notes

a collaboration between Apple, Motorola and Cingular (the last being the largest mobile phone carrier in the US). Despite this US-centric alliance, the ROKR should also be available in the UK by the time you read this column. The ROKR features a colour display with a user interface similar to the iPod, allowing you to view album art, and has built-in stereo speakers and stereo headphones that also function as a headset and microphone for hands-free calls. The phone connects to the computer via USB and is functionally similar to the iPod Shuffle, in that you can either fill it manually or have iTunes automatically load your iTunes 5 features a redesigned interface. favourites. Motorola claim that the The ability to organise playlists into folders ROKR can store around 100 songs will be welcomed by many users. (based on four minutes per song at 128kbs AAC encoding), and automatically pauses the music when you receive a call. iTunes 5 was also released at Apple's special music event and features a redesigned user interface, a new search facility, the ability to create folders for your playlists, and Smart Shuffle, which lets you adjust the 'random' way in which iTunes shuffles your music. After installing iTunes 5 on my 2.7GHz G5, I was initially dismayed to see that the application could cause a kernel panic and bring down the whole system when I tried to rip an audio CD. Fortunately, this issue was resolved after I installed the subsequent 5.0.1 update: Apple's release notes mentioned that "iTunes 5.0.1 features several stability improvements over iTunes 5." No kidding!

.Mac To The Max As mentioned at the start of this column, there was no Apple keynote speech at this year's Paris Mac Expo, but the company did use the show to announce what it terms as "major enhancements" to the .Mac service. Dot-Mac (which is how you pronounce it) started life as the free iTools service (remember when a mac.com address was free?). The name change and the £68.99 per year subscription free was introduced, along with some new services, after the July 2002 Macworld New York show. Users now get 1GB of storage as standard (up from 250MB) for email and iDisk usage, although it's still possible to purchase additional storage space if you need it, for an annual charge of £34.99 ($49) per gigabyte. Apple also introduced version 3 of the .Mac Backup software, which has been redesigned and now makes it easy to back up iLife content, purchased music and your home folder, while the new .Mac Groups is a new way to create online communities and share data. For day-to-day Mac life, .Mac is file:///F|/SoS/SoS%2011-2005/applenotes.htm (3 of 5)10/19/2005 9:44:16 PM

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worth the yearly subscription to me personally, if only because I can have multiple Macs (and a web-based interface) synchronised to the same email account, along with calendars and so on. Despite the 'general use' nature of .Mac, there are a couple of features that will be of interest to musicians, most notably the fact that you can download the Garage Band Jam Packs for no additional charge from your iDisk (once you're a paying member), assuming that you have enough bandwidth. And in the .Mac Learning Center there's a growing collection of tutorials for Apple's iLife and pro applications such as Garage Band 2.

X Marks The Spot For Linux Audio Apps As Mac OS X is built on the Berkeley Software Distribution (BSD) of UNIX, under the bonnet Apple's operating system shares much functional similarity with other UNIX-like operating systems, such as Linux. It's this similarity that makes it relatively easy to port Linux software to Mac OS X, and a good example of this is the Jack audio connection kit (www.jackosx.com) discussed in October 2004's Apple Notes by Daniel James (see www.soundonsound.com/sos/Oct04/articles/ applenotes.htm), which was originally developed for Linux and allows you to route audio between multiple applications running simultaneously. Despite the similarity at the lower levels of the operating system, writing user interfaces for Mac OS X is very different from writing user interfaces for any other operating systems and requires a great deal more work to port the code. User interfaces on UNIX-like operating systems are usually built using the X Window System (X11), a framework that defines how windows appear and user input is handled. The actual implementation of the user interface is handled by a so-called window manager, of which there are many in the Linux world, such as Kwin, used by the K Desktop Environment (KDE) that's supplied with many popular Linux distributions. Aqua itself is another example of a window manager.

Thanks to Apple's X11 window manager, it's easier for developers to port applications from other UNIX-based or influenced operating systems that use the X Window System for their user interface. Here you can see a build of the popular open-source Linux digital audio workstation Ardour running on Mac OS X.

In order to encourage programmers of UNIX and UNIX-like operating systems to port their applications to Mac OS X, Apple made available a public beta of an X Window System implementation at around the time of Jaguar. This effectively allows X-compatible applications to be recompiled for Mac OS X with fewer tweaks than would be necessary if one were writing a native Aqua user interface. (Since the public beta, Apple's X11 has been supplied with OS X, but is not usually installed by default. To install X11 under Tiger, for example, run the Optional Installs application, located on the root of your Tiger DVD, navigate file:///F|/SoS/SoS%2011-2005/applenotes.htm (4 of 5)10/19/2005 9:44:16 PM

Apple Notes

through the first pages of the installer and licence agreement, select X11 when you get the Custom Install window, and click Install. The X11 application itself will be placed in the Applications/Utilities folder.) A positive upshot of this move, especially in the world of open-source software, is that it makes it relatively easy for developers to port more Linux applications to Mac OS X. An interesting example of this is Ardour, which is perhaps the most prominent digital audio workstation in the world of Linux audio software and is written by Paul Davis, the main author behind Jack. The use of Jack here is important because Ardour itself doesn't need to know anything about Core Audio, since Ardour uses Jack for audio I/O the same way as it would on Linux. Like X11, this really does facilitate more collaboration between developers of Linux and Mac OS X audio software. In order to run Ardour, you need to install X11 and Jack (see the previously mentioned Apple Notes column or Jack's web site for more information), and you can download the latest Ardour beta from www.ardour.org/download.php. As mentioned on Ardour's web site, you may need to run Jack Pilot (a tool included with Jack) before running Ardour, to start the Jack server. Published in SOS November 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

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CLASSIC TRACKS: Jimi Hendrix Experience All Along The Watchtower

In this article:

Ahead In England Ready For A Change You Keep Me Hanging On Shy Guy

CLASSIC TRACKS: Jimi Hendrix Experience All Along The Watchtower Eddie Kramer Published in SOS November 2005 Print article : Close window

Technique : Recording/Mixing

With his searing version of 'All Along The Watchtower', Jimi Hendrix set a standard for Dylan covers that has rarely been equalled. Eddie Kramer was behind the glass as the sessions moved from London to New York. Richard Buskin

By 1968, Eddie Kramer had already earned his stripes as an engineer. A native of South Africa who had studied classical piano, cello and violin before developing twin interests in jazz and engineering, he'd relocated to England at age 19 and worked at Pye, his own KPS Studios, Regent Sound and Olympic through the mid-'60s, where he'd accumulated recording credits that included the Kinks, the Beatles, the Rolling Stones, the Small Faces, Traffic and, most notably, Jimi Hendrix. Kramer was, after all, the main man behind the board for most of the guitar virtuoso's brief time in the spotlight. It was at Olympic that Kramer had previously sat alongside producer Chas Chandler and engineered Jimi's stunning 1967 debut, Are You Experienced?, which spawned the British hit singles 'Hey Joe', 'Purple Haze' and 'The Wind Cries Mary', as well as its follow-up, Axis: Bold As Love. And it was also at Olympic that, in early 1968, sessions commenced for Hendrix's third and arguably best effort, Electric Ladyland. Although not as consistent as its predecessors in terms of material and musical arrangements, this hallucinogensoaked double album nevertheless featured the 25-year-old guitarist/vocalist/ composer at the top of his game, showing off his breathtaking skills on classic cuts such as 'Voodoo Chile', 'Crosstown Traffic' and '1983... (A Merman I Should file:///F|/SoS/SoS%2011-2005/classictracks.htm (1 of 9)10/19/2005 9:44:19 PM

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Turn To Be)', while Eddie Kramer helped realise Jimi's sonic vision by pushing the technological envelope to its absolute limits. "The fact that Jimi really had a great idea of what he wanted to hear enabled us to create sounds for him and enhance what he was giving us," Kramer says. "That really helped make Electric Ladyland unique." Along with the aforementioned 'Crosstown Traffic', the tracks recorded at Olympic included one with which Hendrix would forever be associated even though he didn't write it: an exceptional interpretation of Bob Dylan's 'All Along The Watchtower'. The song's poetic, allegorical lyrics about social change had intrigued Hendrix when he first listened to Dylan's John Wesley Harding album, and in his version they would be infused Photos: Kramer Archives with four blistering and distinctive guitar solos, Jimi Hendrix at the Datamix aligning the song far more with its dramatic context console at the Record Plant, New York, during the than Dylan's much sparer rendering had. Other recording of the Electric artists would cover the song, including the Grateful Ladyland album. Dead, Eric Clapton, Neil Young, U2 and the Dave Matthews Band, but Hendrix's intensely evocative version remains the most famous, and his arrangement was the one that Dylan himself subsequently adhered to.

Ahead In England At Olympic the setup comprised a Dick Swettenham-designed Helios desk and Ampex four-track tape machine housed within what Eddie Kramer describes as "a wonderful, spectacular-sounding room. The modular console, with its great EQ, great reverb and great compression, was absolutely marvellous — we were ahead in terms of console design in England. In fact, that desk was so advanced, the American engineers would come over and say 'Wow, what an incredible board,' to which we'd say 'Yeah, but you've got eight-track.' We were so jealous of what was going on in America when we were stuck with this bloody four-track format where we had to go four to four to four all the time, even though in essence this really trained us to make decisions then and there about the stereo bounce — we had to be very accurate with our mixes, and that was a different kind of training because it forced you to be creative as the sound was being printed. That held me in good stead later on." While Hendrix's amp was screened off in the studio and miked with Neumann U67s, Mitch Mitchell's kit was positioned on a riser within a roofed, open-sided booth to give it depth and miked with a combination of U67s and AKG C12s.

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"Initially there was no bass," says Kramer. "Jimi just played a six-string acoustic guitar while Traffic's Dave Mason played 12-string and Mitch was on drums. That's how Jimi wanted to cut it, and as a result the track had a marvellous, light feel thanks to the acoustic guitars that were driving it. Jimi not only loved the lyrics but also the chord sequences of 'All Along The Watchtower', and he just gave them a terrific bed to do a nice solo. He also showed Mitch how to turn the beat around on the intro, but Dave Mason couldn't get it together and he was up to about take 20 when [Rolling Stone] Brian Jones walked in. Actually, let me correct that: he staggered in. He was completely out of his brain. Poor Brian, he was a good mate of Jimi's and we all loved him. Jimi could never say no to his mates, and Brian was Eddie Kramer in New York, 1968. so sweet. He came in and said 'Oh, let me play,' and he got on the piano, it was take 21, and we could just hear 'clang, clang, clang, clang, clang...' It was all bloody horrible and out of time, and Jimi said 'Uh, I don't think so.' Brian was gone after two takes. He practically fell on the floor in the control room... Dear Brian. "It actually took about 27 takes to get the track going because Dave Mason couldn't get it together, but eventually he did and that was all that mattered. Jimi was driving the train. He always drove the train, whatever he was doing, and he had a magical ability, bar none, to take other people's material and make it his own. In fact, he also played bass on this track. When he said that's what he wanted to do, Noel [Redding] pissed off to the pub. He didn't want to know. "Recording was always a learning process for Jimi, so each take would be different, and for 'All Along The Watchtower' there was no real rehearsal. Jimi just wanted to record the song. He loved Bob Dylan and he always carried his songbook with him. In this case, he was fascinated by the colour of the lyrics and the tone of the lyrics, and of course the chord sequences were wonderful, too. It was a very special song." When asked about the techniques used to record Hendrix's guitar, Eddie Kramer's response is concise and to the point. "I'd stick a bloody mic in front of it and hope for the best," he jokes. "Nah, generally speaking it was either a 67 or [a Beyer] M160 or a combination of both, which I still use today. It might be slightly different, of course, but the basic principle's the same — a ribbon and a condenser, along with compression and EQ and reverb. All that stuff was always added during recording."

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Watchtower' had been tracked, the Electric Ladyland sessions switched location to the US. Hendrix wanted to move back there, and when Record Plant owner Gary Kellgren invited Eddie Kramer to come work at his newly opened studio in New York City, the engineer jumped at the opportunity. He was, after all, ready for a change. "In England, if you were successful, people would do their utmost to cut you down," he says. "Nobody appreciated you — 'Oh, you flash bastard. How dare you make £95 a week when I'm making £50.' I couldn't stand that attitude and to this day it makes me very angry. I didn't feel at all appreciated." Experience drummer Mitch Mitchell behind

Consequently, April 17, 1968, was the his kit at the Record Plant. date of Eddie Kramer's arrival on American soil, and amid all the upheaval the Electric Ladyland project provided him with a strong sense of continuity despite the sudden change of environment and transition from fourtrack to 12-track. What's more, it also included the Record Plant's very first sessions, interspersed with Hendrix's frequent assignments on the road.

"We would get a week in the studio and then he'd be off, and then we'd get another week or 10 days and again he'd be off," Kramer recalls. "There I was, an emigré, establishing myself, getting into sessions, figuring out what America was all about. It was quite a challenge, and in the beginning it wasn't easy, but once I got used to the vibe I was flying. I loved it. The culture was so completely different — 'If you do a good job, you're gonna get paid, brother,' — and so was the technology for me, going from four to 12-track and bypassing eight-track completely. Hello. What a wake-up call. "Instead of the Helios I was now sitting down at a Datamix board that was completely foreign to me, in a room that didn't sound the way that I was used to hearing things, and using a 12-track, one-inch Scully machine that was a horrible piece of crap. Good God, it was so noisy, it was horrendous. We scrapped it after the first couple of months and went straight from 12 to 16-track. When the songs we'd recorded at Olympic had been transferred from four-track half-inch to 12track one-inch, Jimi had said 'Wow, man, now I've got eight more tracks to fuck around with. Cool!' and of course they all got filled up. But then we scrapped the bloody machine after transferring all of the 12-track tapes to 16-track. "So, for instance, 'All Along The Watchtower', which had started four-track in England, was transferred onto the Scully and then the first Ampex 16-track, which was actually not a bad machine. In fact, listen to any of the tracks on that album and the lack of hiss is so apparent. That's down to the way we hit the tape — we hit it very hard and got all the necessary compression that came with it. Also, what with all the intensity that was going on, there was a fair amount of file:///F|/SoS/SoS%2011-2005/classictracks.htm (4 of 9)10/19/2005 9:44:19 PM

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signal on the tape, and that really helped. It was quite a journey. "Eventually, I got used to the Record Plant room, which was just a fairly live rectangular box with a couple of panels on the wall, made of pegboard with a little bit of fibreglass behind them. The control-room window was so thin, and the wall itself was so thin — we had these four huge 15-inch Tannoys mounted above the window, and when Jimi would blast we could hear him through the wall. There was virtually no soundproofing. The room was constructed out of breeze block, there were two thin curtains, and that was it. Very primitive. This was the antithesis of what I'd been used to in England, but nevertheless what we got out of it was magic. The board was fairly flexible: not a great-sounding console, but we made it work."

You Keep Me Hanging On Chas Chandler, the former Animals bass player who had discovered Jimi in 1966 and since guided his career, was initially at the helm in New York as manager and co-producer to continue the work that had commenced in London. However, as has been well documented, he grew quickly tired of all the hangers-on attending the sessions, and decided that he wanted to opt out. "Once Jimi got to the States I think his whole attitude changed towards what he was looking for musically," Eddie Kramer remarks. "I don't know whether this was aimed at having Chas leave — one could never tell — but there was definitely tension between them. Chas liked to run his sessions in a very strict, formal manner, without wasting time. He'd say 'We're here to work. The hangers-on must leave,' and it was left to me or Chas to tell the hangers-on to take a hike. Well, I guess that caused some friction and eventually Chas couldn't really take it any more. He hated wasting time, and if Jimi wanted to do 30 takes it would drive him nuts. "Chas was the boss. Not musically, per se, but the boss in terms of not allowing any wasted time, and I think the restrictions he placed on Jimi for the first two albums were really good. I don't necessarily agree with what happened on the Electric Ladyland sessions, but without Chas there would have been no huge superstar. To start with, Chas recognised Jimi's talent, and then he was able to corral that raw talent and develop it and encourage it. He would sit with Jimi every night, helping him to write lyrics and helping him with the song structures, encouraging him to write. However, during that third album the sessions took their own course, and Jimi, with his strong vision, just allowed things to happen in a very casual way. "Now, having said that, I also feel very strongly that he had a master plan, and as chaotic as it may have seemed to an outside observer it was actually quite well thought out. The classic example of this was 'Voodoo Chile', which was really created as a jam but a very, very calculated jam. I mean, after Chas left [the project], Jimi had wonderful aid and assistance from a quite unlikely source: the Scene club which was, fortunately for him, around the corner from the Record Plant. Having booked the session for seven o'clock, we'd be sitting there, tapping our fingers on the desk and twiddling our thumbs, wondering when he was going to show up. After he'd done this a few times we all knew this was Jimi's way of working. He'd be over at the Scene at 10 and show up at the studio at 12 or one, dragging behind him an entourage that included musicians whom he had sussed file:///F|/SoS/SoS%2011-2005/classictracks.htm (5 of 9)10/19/2005 9:44:19 PM

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out as being the key players to try out that evening. "You see, he had a specific plan in mind. There was a certain sound he was looking for, and he'd eyeball the musicians very carefully to make sure that they were going to be compatible with what he wanted to do. Generally speaking, he got the cream of the crop, because at that time, 1968, there were some phenomenal musicians around, and 'Voodoo Chile' was a classic example of Jimi figuring 'OK, I'm gonna get these guys in to play this particular song'. He'd bring them in at midnight or whenever, and everything would be ready: the amps, the mics, the headphones — I'd tested everything. Then he'd show them the song, and there'd be one run-through and one take, maybe two. Bam! It was done. So, to the outside observer there were the hangers-on and the whole rigmarole with onlookers, and sometimes that made it a bit challenging to work, but it never detracted from Jimi's goal. Chas may have commented 'Oh, he's playing to the gallery,' but it didn't seem to bother Jimi. In fact, it probably encouraged him to play more to the gallery, because maybe that was the vibe he was looking for. "Chas was absolutely essential to Jimi's development as a writer and as a performer, as well as in terms of putting the band together, producing the records and giving Jimi the necessary discipline to come up with the goods. But with the Electric Ladyland album his role diminished as soon as the sessions moved to the States. You could tell there was a sea change in Jimi's behaviour, in his attitude and so on. I think the album as a whole has a journeyish feeling to it — and I'm not referring to the band, for Chrissakes. It rambles a bit, but it rambles with a purpose. And I love how there are so many different moods. Of all of Jimi's albums, it's the one that has the most moodiness — to some people it represents the most fun that you could have on a record. I mean, it was very daring to make a double album with all that experimentation. That was making a statement in 1968. And although it was looser than his previous records, it had a purpose, it had a focus. The purpose was 'Let's be loose!' "I think this was Jimi expressing himself for the first time, completely unfettered. He could basically do anything he wanted — it was his album, from soup to nuts, and while it bears the stamp of Chas on some of the songs it very much shows Jimi's freedom in the creative process; the freedom to do a 14-minute opus like '1983...' Taking a chance to do something like that was very '60s."

Shy Guy Hendrix recorded all of his vocals for the album at the Record Plant, and as usual a Beyer M160 was the mic of choice while a three-sided screen provided him with the desired privacy. "He'd always face the other way," says Kramer. "He hated to be looked at. He was very shy about his vocals. The truth was, he had a great style and I loved his vocals, but he hated them. He was so embarrassed by them. 'Oh man, was that OK?' 'Yeah man, it's cool.' 'No, I've got to do another one.' 'OK.' Jimi was not a great vocalist in the classic sense, but his vocal style suited what he did to the nth degree. I mean, it was very emotional and very personal, and I can't think of anybody else doing what he did. He was eminently capable, and the singing was an integral part of what he was doing, because he would often take a guitar solo and sing the melody line in unison with that solo — which is an old jazz trick — and it was wonderful."

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So, for that matter, was the constantly evolving 'All Along The Watchtower', recorded in consecutive layers: acoustic guitars, drums, bass and electric lead taped at Olympic, before the vocal and percussion overdubs took place in New York. "What's amazing to me is how the original four-track punches through so much at the end," Kramer remarks. "Jimi's loping bass line doesn't have any top end to it at all — it's just round and lovely — and you can hear him moving around on the bass as if it were a guitar. You see, he could play pretty much anything — the piano, a bit of drums. It was feel over technique in that department, whereas with stringed instruments like bass and guitar he was magnificent.

Noel Redding at the Record Plant during the Electric Ladyland sessions; the bass on 'All Along The Watchtower' was played by Hendrix himself.

"When it came to the mix it was a case of Jimi and I doing it together and just making it sound as commercial as we possibly could. At least, that's what I was going for with the judicious use of compression and EQ and reverb. That's what we had at our disposal and I think it worked — that reverb on the offbeat is just one example." He can say that again. It was Eddie Kramer who, with the few effects then at everyone's disposal, made highly innovative use of flanging, chorusing, looping and backwards tapes to realise the sonic vision inside Jimi Hendrix's head. As a result, the studio itself became yet another of the artist's instruments, plied like his guitars to mould sounds in his unique, groundbreaking style. Yet, while 16track really opened up the doors in this regard, there is something to be said for limited resources inspiring creative thinking. "When you recorded four-track you really had to have your act together," Kramer asserts, "and during the mix you knew you just couldn't screw it up, because this was the final one. In the transfer from the first four down to the two tracks of the next four-track machine, that first pass had to be absolutely spot-on. Of course, you could go back and remix it, but you wanted to avoid that if all possible. So, you just had to have it all there, and thank God we were taught correctly by great mentors: people like Bob Auger at Pye and Keith Grant at Olympic. They were really great teachers. You just rehearsed a song and got it right, got the EQ, got the compression, got all the bloody bits and pieces in there. And you knew in the final analysis that when you added the extra two tracks and bounced back to the first four tracks and then maybe went back one more time, all of those stages along the road had to be absolutely spot-on. "When I listen to those tapes now I'm still amazed that we actually got what we got. In the moment we didn't give a damn about convention or anything, we just did it, and tape was running most of the time. Occasionally we'd edit together takes, but not so much with Jimi. Mixes we would splice together, and there'd be

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lots of panning and loads of cups of tea, but his performances were pretty damned complete." So, does Eddie Kramer consider Electric Ladyland to be an artistic high point for Jimi Hendrix? "His whole When Jimi Hendrix decided to build his own career was an artistic high point," comes the reply. "I think he really loved Electric Lady studio in New York, Eddie Kramer designed it and became the first the record, although he was pissed at head of engineering there. the fact that we were not allowed to go to the mastering. We gave it to the crew over at Columbia and they completely screwed it up, because they didn't know what to do with the phase content and all of that. It was subsequently remastered at Warners." And once again in 1997, by Kramer himself, for the reissue on MCA. "I've completely screwed it up now," quips Kramer, who relocated to LA last year and recently worked on the restoration of the original Woodstock movie, to be released in 5.1. "No, seriously, I'm very proud of it. It sounds really good, the best it's ever sounded. When I listen to the original tapes I love hearing the conversations between myself and Jimi. They're always a lot of fun. He was a funny bloke in the studio, he was hilarious. He was always cracking jokes." Following its release as a single in September 1968, 'All Along The Watchtower' climbed to number 20 on the Billboard Hot 100, providing Jimi Hendrix with his only American Top 40 hit. The song went all the way to number five in the UK.

Eddie Kramer in 2002.

"Hendrix is the Robert Johnson of the '60s, and really the first cat to ever totally play electric guitar," wrote Tony Glover in his Rolling Stone review of Electric Ladyland. "Hendrix, psychedelic superspade??? Or just a damn good musician/ producer? Depends on whether you want to believe the image or your ears. (And if you wanna flow, dig this on earphones, and watch the guitar swoop back and forth through your head.) Hendrix is amazing, and I hope he gets to the moon first. If he keeps up the way he's going here, he will." Published in SOS November 2005

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

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

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Digital Performer Notes

In this article:

Digital Performer Notes

The Curse Of The Restless News & Tips Parameters Antares Get Vocal With New Published in SOS November 2005 Toolkit Print article : Close window Can We Fix It?

Technique : Digital Performer Notes

We've got news of some interesting new plug-ins this month, and the solution to one of those software compatibility problems that can drive you bananas... Robin Bigwood

The partnership of Digital Performer and Reason, linked via Rewire, is a powerful one, and might be all you need to produce your masterworks. There's an annoying flaw in the setup, however, that can wreak havoc if you're not expecting it, and is irritating even if you are. It's this: adjusting parameters on a Reason device can sometimes cause parameters on other devices to change without warning. There you are tweaking your Subtractor and meanwhile an NNXT is going haywire. What's happening?

The Curse Of The Restless Parameters When you adjust a parameter on a device in Reason, it produces MIDI continuous controller data that is transmitted over the Rewire link to DP. This is a good thing, as it means you can move Reason knobs and sliders and have the MIDI data associated with that recorded in DP. You can effectively 'automate' Reason device parameters without having to mess around with Reason's own automation scheme side by side with your DP sequence. However, the Reason/DP combo doesn't do this quite as intelligently as you'd expect — DP can't identify which Reason device is the source of the CC data, so an entire Reason rack becomes a kind of 'monster' continuous controller data source. If you have a record-enabled MIDI track in DP driving a Reason device, switching to Reason and adjusting a parameter on any device in the rack will cause the CC data associated with that to be 'reflected' back to the device that the DP MIDI track is driving. The situation apparently gets more chaotic the more devices you have in your Reason rack. Fortunately, there are solutions.

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Digital Performer Notes

Antares Get Vocal With New Toolkit Antares, traditionally one of the most DP-friendly third-party developers, seem to have experienced something of a renaissance recently. There's a new logo, a new website and an interesting new plug-in bundle called AVOX — the Antares Vocal Toolkit. Although Antares' days of plug-in development in DP's native MAS format are gone, the Audio Unit versions they produce work beautifully, at least in the latest versions of DP, so there's no problem with compatibility and reliability, and the five AVOX plugins bear this out.

Throat, from Antares' new AVOX plug-in bundle. At around the £400 mark, AVOX isn't cheap, but it does things that would be hard to achieve by other methods.

Possibly the most unusual is Throat, which takes a vocal or instrument input and actually remodels it as if it were being produced by another human throat. You can dial in subtle character changes, or make otherwise great singers sound ridiculous, and surprisingly realistic breathiness can be added. It even made my vocals sound OK. Duo and Choir add one or more 'phantom' vocalists to a single vocal line, for instant double-tracking or multitracking effects. Punch looks like a compressor, but is no doubt rather more complicated, and adds bite and impact to vocals. Finally, Sybil is a smart de-esser.

Can We Fix It? First, and simplest, this behaviour will never happen when there are no recordenabled tracks in DP, so remembering to disable recording before tweaking Reason devices could restore sanity. This isn't bombproof, though — sometimes you'll need to tweak one Reason synth while playing another live, and that will cause problems. Another idea is to turn off DP's 'MIDI Patch Thru in Background' feature (in the MIDI Solo and Patch Thru pane of the Preferences window), but this means that you can never play (live) any Reason device from DP while tweaking its parameters on screen. Possibly the best solution is to turn on Multirecord in DP's Studio menu. If you then choose your master keyboard as the MIDI input for your record-enabled tracks, DP will never 'hear' the unwanted Reason CCs. You can then play the Reason device when DP is in the background and tweak other devices' parameters without conflict. Success! Or is it?

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Digital Performer Notes

If there's a downside, it's that you can't record your master keyboard and Reasongenerated CCs at the same time, but I doubt many people do that anyway. In any case, you could achieve it in another way, by simply creating another MIDI track in DP, setting its MIDI input to the Reason device whose CCs you want to record, and recording MIDI notes on one track and CCs on the other. You could later merge these, if desired, or keep them separate and live with having a 'notes' track and a 'controller' track for one device. Published in SOS November 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

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Emu's Patchmix DSP Demystified...

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Emu's Patchmix DSP Demystified...

Basic Mixer Features Technique What's So Different About Published in SOS November 2005 Patchmix DSP? Problem-solving Tips Print article : Close window Sessions & Sample Rates Technique : Recording/Mixing Recording Options On-line Help DSP Effects Routing Patchmix DSP Tips More Complex Signal Paths We demystify the powerful

internal routing and mixing engine within Emu's recent popular range of computer audio interfaces. Martin Walker

The new range of Emu soundcards (1820M, 1820, 1212M, 0404, and by the time you read this the 1616M as well) has been selling like hot cakes, but many PC musicians find various aspects of their associated Patchmix DSP utility difficult to comprehend. It's not that the mixer is difficult to use; more that it offers so many possible options that it can be difficult for the newcomer to perform even basic tasks. The Patchmix DSP mixer design is based on a conventional analogue mixing console, with the input channels on the left-hand side, main and monitor mix output channels on the right. These various input and output channels have inserts, aux sends, pan and level controls, mute and solo buttons, and so on, which generally makes perfect sense to most musicians who have been brought up with such mixers. However, there's a new breed of musicians who have only really experienced the virtual world of the software studio, and it's hardly surprising that they find this layout unfamiliar. So in this workshop I'll be covering the basic principles behind Patchmix and then explaining some of the more arcane operations.

Basic Mixer Features Let's start by quickly recapping on the basic functions available. Emu have made

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Emu's Patchmix DSP Demystified...

the Patchmix DSP mixer totally configurable, so you can have as many input channels as you need, up to a maximum of sixteen in stereo — you can add extra channels at any time. I normally opt for the simplest arrangement for the task in hand in order to keep the mixer smaller on screen so that it doesn't get in the way of other windows. A label showing the channel type appears at the top of the strip, and you can drag and drop the input channels into any order in your mixer design. Some people get confused by the fact that every channel strip is labelled as Input, whether it's being used to listen to a physical signal source plugged into one of the Emu input sockets, or carrying a playback output signal from your audio application. However, in both cases you're sending these signals through the mixer either to monitor via one or more of the physical output sockets, or to route them elsewhere using sends. For each mixer channel you create using the New Strip function you must choose one of the available input options from a selection of Physical Sources comprising whatever input sockets your particular Emu card has on offer, such as the mono mic/line inputs, stereo line inputs, S/PDIF input, or ADAT inputs. Alternatively, you can select one of the Host Sources — either one of the sixteen stereo ASIO Output Sources (carrying stereo playback signals from your audio application), or the single stereo Windows Source for playback when you're using an application with Direct Sound or MME-WDM drivers and when playing back Windows sounds. Below each channel strip's label is a set of six Insert slots, into which you can insert a variety of different objects (more on these later on). Next are the pan controls, followed by two aux sends, which can be used to add variable amounts of Emu's DSP effects, such as reverb or chorus, to all the channels globally. The bottom section of the channel strip comprises a strip level fader with associated text box so that you can directly enter a suitable level in decibels. The Mute button removes the output of that channel from the mix, while Solo allows you to hear it in isolation. Finally, at the bottom there's a 'scribble strip' where you can enter a descriptive name for the channel.

What's So Different About Patchmix DSP? Some audio interfaces simply hard-wire the various driver playback channels to the physical output sockets (for example, ASIO 1/2 playback comes out of Out 1/2, ASIO 3/4 from Out 3/4, and so on), and hard-wire the various physical input sockets to the various driver inputs (In 1/2 to ASIO In 1/2, and so on) for recording. This is the way a lot of musicians want to work anyway. Those interfaces offering direct or 'zero latency' monitoring will also provide the option of routing physical input sockets direct to physical output sockets, so you can monitor the incoming signals during recording without any software delays. The next step up in versatility is providing a dedicated hardware monitor mixer on the interface, so that you can mix together various live input signals and playback signals from your sequencer, and either use this combined stereo signal to feed a file:///F|/SoS/SoS%2011-2005/emupatchmix.htm (2 of 9)10/19/2005 9:44:25 PM

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physical output — a monitor mix on headphones for performers during the recording phase, for instance — or send it to a driver input so you can record lots of signals onto one stereo track of your sequencer. This is the approach used by manufacturers such as Edirol, M Audio, and Terratec. In these cases, however, you can still completely bypass and ignore the monitor mixer by choosing the direct routing options for both physical inputs and outputs, as before. Emu have taken rather a different graphical approach, and at first it appears that every signal must be routed through the entirety of the Patchmix DSP mixer. The secret is that, while much of the mixer is indeed dedicated to creating a monitor mix that appears on both the Main and Monitor outputs, you can also use the Insert slots in each channel to route input signals directly to your sequencer, or output signals directly to physical output sockets, as with any other audio interface. If this is all you want to do, a useful mental trick is simply to ignore all the Patchmix DSP controls below the Insert slots, as well as the mixer's entire output section (see screenshot). It may also be useful to activate the Mute buttons for these channels, so you remember that you're not listening to them through the monitor mixer but by directly routing them elsewhere.

Problem-solving Tips One of the first difficulties experienced by many users is simply getting a sound out of the mixer when attempting to play back WAV files or soft-synth sounds. The easiest emergency advice is to load a New Session (configuration of input and output channels and routing), using the Save As option that pops up to preserve any of your own existing settings for another time, and then to choose a template from the thirty or so on offer. This will reset everything to suitable defaults, and in many cases your sound will return immediately. If you're using the 1820 or 1820M, though, many of the templates assume that you'll be using the Audiodock outputs 4L and 4R as Monitor outputs, so make sure these are what your amp/speakers are plugged into as well. If you're using the MME-WDM drivers with an application like Steinberg Wavelab, your output signal will be passing through the input channel strip marked Wave L/R, while ASIO applications will probably default to ASIO Out 1/2. At this point, let me introduce you to the single most helpful hint of all. To help sort out Patchmix DSP routing problems, always put a Peak Meter in the topmost slot of each mixer channel by rightclicking on the slot and selecting 'Insert Peak Meter'. This will show you where you have active signals, even if you can't hear them due to some routing problem. The easiest output routing is through the monitor mixer, the controls for which are on the righthand side of the mixer window: master controls for the main and monitor outputs, read-outs of current sample rate and clock source, plus a 'TV window' above that which displays extra contextfile:///F|/SoS/SoS%2011-2005/emupatchmix.htm (3 of 9)10/19/2005 9:44:25 PM

Emu's Patchmix DSP Demystified...

sensitive controls. You'll probably already see the Main Mix output meter twitching in time with your signal (if not, you've either pulled down the channel fader, activated its Mute button, or have some other Insert blocking the signal). Now all you need to do is make sure the output of the mixer is routed to the output socket that you're listening on.

To play back stereo sounds from ASIO and MME-WDM applications, this is all you need in the Patchmix DSP mixer. Notice how the mixer output has been routed to the physical output socket named Dock Out 4 so that you hear the signals from both channels simultaneously).

Click on the Outputs button above the 'TV window', and then on the box marked Physical — this will display vertical strips of red and green boxes on the left, and a list of possible physical output sockets on the right, depending on the Emu card you're using. To connect the Main or Monitor output to your choice of output socket just click on the red or green box adjacent to it — you can send either of these signals to as many simultaneous physical outputs as you wish. The monitor output carries an identical signal, but with separate volume, balance, and mute controls, which is a handy when you're wanting to work on headphones. If you want a more direct signal path without all the extra options of the monitor mixer, just activate the Mute button of the channel in question, right-click on an empty Insert slot, and select Insert Send (Output To ASIO/WAVE Or Physical Out). When the New Send Insert window appears, choose the desired physical output from the drop-down list (for example, PCI Card S/PDIF L/R, PCI Card ADAT Out 3/4, or Dock Out 3L/3R). This is the approach to use when (for instance) you want to send eight ASIO channels from Cubase SX to individual ADAT outputs — just create one stereo send for each of the four ASIO channel pairs.

Sessions & Sample Rates You can save your own different configurations of the Patchmix DSP mixer (known as Sessions), for use with different applications and in different scenarios. For example, I've created some that incorporate the settings needed to transfer digitally to and from my portable DAT recorder. Depending on the interface model, Emu also provide a set of about thirty default Sessions covering multitrack recording and playback, ADAT transfer, guitar tuning, and a special one for use with the Emulator X Studio sampler if you have that as well. One important thing to remember is that the Patchmix DSP mixer offers DSP effects that only run at either 44.1kHz or 48kHz sample rates. There are also several restrictions on the digital ADAT and S/PDIF connections beyond 48kHz. Because of these various fundamental internal changes, it's impossible to let software applications switch sample rates for you automatically behind the scenes. So, although with most audio interfaces you can load and play WAV files at almost any sample rate, the interface automatically accommodating you by changing its current rate behind the scenes, the Emu range (along with various others including Edirol's UA and FA ranges) require you to set the appropriate sample rate manually. If you try to play back or record at a different rate from the one currently set in the hardware, you'll either get an error message or just silence.

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To set your preferred sample rate directly from the Patchmix DSP utility by choosing one of the New Sessions on offer — when you click on the New Sessions button on the mixer toolbar a dialogue screen will be launched with three main tabbed pages. If you choose one of the supplied sessions on the 44k/48k page you can decide which of the two sample rates you want on the following page, while the sessions on the 96k and 192k pages are fixed at these sample rates, and provide default sessions with no effects processing. The eagle-eyed among you may have noticed that 22kHz, 88.2kHz, and 176kHz haven't been mentioned. This is because the Emu range doesn't currently support these sample rates at all. Although this may prove frustrating to some users, there is one advantage in not supporting 22kHz. While ASIO drivers won't convert between sample rates behind the scenes, MME-WDM ones normally will, and if you play a 22kHz system audio file, your sequencer project may end up having its sample rate changed until the next boot. With Emu's approach, the 22kHz file will simply be ignored, leaving your sequencer playback intact.

Recording Options Next, let's say you want to record a stereo signal through a couple of the line input sockets and therefore want to route its signal to your audio application. If your chosen line input pair (Dock In 1L/1R for instance) isn't already displayed in the Patchmix DSP mixer as a stereo channel strip you first need to use the Append New Strip option to add it to the mixer. Then you have to route it to the audio application. Right-click on one of its Insert slots, select the Insert Send (Output To ASIO/WAVE Or Physical Out) option, and then choose where you want to send it to, for example HOST ASIO In 1/2 for any application using ASIO drivers (such as Cubase or Sonar), or HOST WAVE L/R if you're sending it to an application using MME-WDM drivers. If at any time you want to refer back to this routing, just click on the Insert in question, and the connection you made will appear in the 'TV window'. With the current Patchmix DSP version, each of the mono Dock Mic/Line inputs of the 1820/1820M models can only be routed to a stereo destination. So, although you could for file:///F|/SoS/SoS%2011-2005/emupatchmix.htm (5 of 9)10/19/2005 9:44:25 PM

Here's a setup that will convert two mono mic inputs into a single stereo input in your

Emu's Patchmix DSP Demystified...

instance route mono input Mic/Line A to the ASIO sequencer. 1/2 inputs of your sequencer and pan it hard left so it only appears on the ASIO 1 input as a mono source, you still wouldn't be able to use ASIO 2 for a second mono input channel, since the ASIO 1/2 pair is now greyed out as an option. This can be frustrating if you're wanting to record with stereo mic pairs, or if you'd like to use the two inputs as extra stereo line inputs. However, there's a way round this limitation, allowing you to send the two channel strips into your sequencer as a single stereo input. First, create the two channel strips for Mic/Line A and Mic/Line B, but making sure that you tick the Aux Send Pre-fader option in the New Mixer Strip dialogue. Next, pan the first strip hard left, and the second hard right, and set each of the Aux 1 send controls to 0dB — this results in the two mono signals being sent to the stereo Aux 1 bus. Now mute the two channel strips (the pre-fader option we used earlier prevents the Aux 1 signals disappearing at this point), and move to the Aux Effects section of the mixer. Set the master Aux 1 Send to 0dB, but the master Aux 1 Return to its minimum -132dB setting. The final step is to right-click on the first master Aux 1 Insert , and select Insert Send (Output To ASIO/WAVE Or Physical Out). Choose a stereo HOST ASIO destination, and our two mono channel-strip signals will end up in the sequencer as a stereo pair. Once you're sure this routing is working, save the Session for later use.

On-line Help You'll find lots of posts about Emu soundcards and the Emulator X soft sampler in the SOS Web Forum, but there are other resources that cater more specifically for Emu-owning musicians. The most popular are the unofficial Emu forums at www. productionforums.com/emu/default. asp, which cover all of Emu's interfaces and software, as well as dealing with integrating these products into all the major sequencers. You'll also find that Emu staff regularly put in an appearance with 'official' answers, despite the fact that these forums are not run by Emu themselves. If your Emu soundcard is one of the ones bundled with Emu's Emulator X sampler, I've also discovered two more web sites with useful information. Emus On Acid (www.emusonacid.co.uk/forum) is primarily a forum dedicated to Emu hardware samplers, but does host one for the Emulator X as well, as do the Studio-central Community Forums (http://studio-central.com/phpbb).

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DSP Effects Routing The bundled effects are roughly equivalent in quality to a mid-range hardware unit, and run internally at 32-bit resolution and 44.1kHz or 48kHz sample rates. As with an analogue mixing desk, the two aux sends come after all the Inserts, and are therefore ideal for adding effects for monitoring purposes (such as reverb during vocal recordings) when you want the recordings themselves totally dry to retain more flexibility during mixdown. This is where the functions of the Patchmix DSP mixer start to outstrip most of the competition, since you're not using any of your PC's native processing power to run the effects — you just left-click on the FX button to launch the Effects Palette window, and then drag and drop the desired effect across to an empty Insert slot in the appropriate channel. With earlier versions of the Patchmix DSP mixer it Using this channel was confusing trying to add different DSP effects to strip setup you multiple playback channels — the process involved could record a creating a number of stereo output buses in your guitar both clean chosen multitrack audio application, routing each one and with external hardware effects to a different ASIO output, creating the appropriate onto separate ASIO channel strips in the Emu mixer, adding the sequencer tracks effects as just described, and then routing the treated simultaneously. signals to physical outputs or back into the sequencer application. Thankfully version 1.6 made life far easier, and you can now access the various Emu DSP effects directly from within your chosen sequencer application as ASIO plug-ins. Just choose the plug-in marked Emu Power FX, then drag and drop one of the available DSP effects into an Insert slot in the Emu Power FX plug-in window. Some musicians have apparently experienced Emu Power FX settings disappearing when reloading Cubase songs, and a few have suffered crackling and synchronisation problems with some host applications. If this turns out to be a problem for you, then remember that there are also plenty of ways that you can use the DSP effects within the Patchmix DSP mixer. For instance, why not add compression, distortion, or a complete chain of effects, to a guitar input in real time? The channel Inserts operate on the signal from the top to the bottom slot, so bear this in mind when setting up more complex routing. Just drag and drop the desired effects to Insert slots on the input channel you're using — remember that you can re-order them by dragging them around as well. For example, if an input channel has (from top to bottom) peak meter, send, and EQ Inserts, you'll get a clean signal sent to your audio application, but hear the EQ'd version when monitoring through the DSP mixer. Conversely, a guitar sent through an input channel with peak meter, distortion, and send Inserts will record the distorted signal as well as monitoring it.

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Patchmix DSP Tips If you think your playback signals are quieter than normal, you've probably loaded a Session containing -10dBV instead of +4dBu output levels — you can change them by clicking the Session Settings button and selecting the I/O page, where you'll find all the current settings of both inputs and outputs, including the configuration of the PCI card's optical input and output (ADAT or S/PDIF). Don't worry if you can see signal activity on a peak meter inserted in the 1820/M Dock In 3L/3R inputs when you have nothing plugged into these jack inputs (or hear lots of hiss coming from this stereo channel through the monitor mixer). This is because the high-gain RIAA-equalised turntable (phono) preamp shares these input channels, and remains active until you plug something into the Line 3 L/R jack sockets. To remove this hiss from the monitor mixer output, just click on the channel's Mute button. Similarly, to minimise any contribution from the mic preamps when you're not using them, click on their channel Mute buttons. This will ensure that background noise levels remain as low as possible on the main mix output. If you create a useful mixer setup, don't forget to save it for posterity with a descriptive name so you can use it again later on. This will save both time and frustration. The Insert Test-tone/Signal Generator includes a sine-wave oscillator that I've found really handy for tuning instruments. Just temporarily insert the oscillator on any mixer channel not playing back the instrument in question and type in the desired frequency — such as 440Hz for a concert 'A'. Then you can tune the instrument to the test tone. Emu's WDM drivers currently only support a single stereo input and output, which some musicians find incredibly restricting. However, the vast majority of professional audio applications now support ASIO-format drivers, and these do support full multitrack operation. Only those few multitrack applications that don't offer ASIO driver support (such as Adobe's Audition, previously called Cool Edit Pro) will therefore be restricted to stereo-only operation, but hopefully this will change in the future.

More Complex Signal Paths If you want to record the mixed-down stereo output from your MIDI+Audio sequencer, plus the combined stereo mix from a set of hardware MIDI synths being triggered from MIDI tracks, it's easy using the Patchmix DSP functions. Just create your combined mix using the various input strips, and then add an Insert Send (Output To ASIO/WAVE Or Physical Out) to one of the Main Inserts to return this combined mix to a stereo input pair on your sequencer. You can even send signals from one application to another through the mixer. For instance, if you choose the stereo MME-WDM E-DSP Wave output option in an application like Wavelab, its signal will appear in the Patchmix DSP mixer on the Wave L/R stereo playback input strip. Then, if you add an Insert Send (Output To ASIO/WAVE Or Physical Out) to one of the Host ASIO In options, this

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stereo signal can be recorded directly into Cubase/Nuendo or any other ASIOcompatible application. You may instead want to patch external hardware effects into some sequencer tracks, and again Emu have made this comparatively easy — you just route the stereo signal in question to an ASIO output in your multitrack sequencer application, make sure this ASIO output appears as a channel strip in the Patchmix DSP mixer, and then choose the Insert Send/Return (Physical Output And Input) option in one of its Insert slots. Choose the physical output and input sockets to which you want to attach your external effects (including S/PDIF if your hardware effects box has digital I/O and you need a cleaner signal path that doesn't pass through A-D and D-A converters). Another example of a complex setup that's easy to create is where a guitarist wants to record a guitar with Pod-style modelling effects, but also record it dry (in other words with no added effects) in case the sound needs to change later on. To do this on the 1820M, just plug your guitar into the Line A or B inputs, add an Insert Send (Output To ASIO/WAVE Or Physical Out) near the top of the Insert chain to any available 'Host ASIO In' to send this dry signal to your sequencer, then add an Insert Send/Return (Physical Output And Input) below it, and patch the Pod input and outputs into the chosen send and return sockets on the Audiodock I/O box. You'll be able to hear the treated (wet) signal through the Main or Monitor outputs of the Emu Patchmix DSP, but both wet and dry signals can now be simultaneously recorded in your sequencer onto separate stereo tracks. Once you've got your head round the routing options I've discussed here, you should be able to work out plenty more to suit your individual requirements — just remember to have a peak meter inserted into each mixer channel, and then you shouldn't lose your signals ever again. Published in SOS November 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

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Logic Notes

In this article:

Logic Tips Networking Noises

Current Versions Mac OS X: Apple Logic Pro v7.1

Logic Notes News & Tips Published in SOS November 2005 Print article : Close window

Technique : Logic Notes

Mac OS 9L Emagic Logic Pro v6.4.2 PC: Emagic Logic Audio Platinum 5.5.1

More tips and news from the world of Apple Logic.

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 logicnotes@soundonsound. com, 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.

Stephen Bennett

Although Logic is a fully featured piece of software, there will still come a time when you'll probably require some thirdparty software to cover the areas where Logic is lacking. Apple acknowledge this, of course, by including the Waveburner CDauthoring package with version 7. There's also a free beta version of Sound Diver, the patch librarian/editor program, from the old Emagic site at www.emagic.de — although this does Redmatica's EXS24 Sample Manager. need an XS key to run, precluding Logic Express users from taking advantage. There are also a few separate third-party programs that are designed to integrate with Logic. One of these is the EXS24 Sample Manager, available from Redmatica (www.redmatica.com). It's a stand-alone software program designed to assist in keeping track of all your samples and EXS24 Instruments. It can check EXS24 Instrument dependencies and see whether any samples have gone astray or have been duplicated. The software creates a searchable database of samples and has several disk-consolidation features that help speed up loading of samples within Logic itself — if Project Manager isn't enough for you, you may want to check it out. It costs 40 or 80 Euro depending on the feature set, and there is a demo available so you can try before you buy. Apple have announced that the replacement of defective old-style 'blue' XS keys won't be possible after the end of 2005. If you're a Logic 7 user with a blue XS key, you'll be able to replace a broken key with a new white one. However, if

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Logic Notes

you're still on Logic 6 or earlier and you break your key, you'll have to upgrade to version 7 — which comes with a white key. In the past, Apple's policy on the replacement of defective keys has been unpredictable. Some users have had a free replacement after sending in the old broken key, while some have had to pay a nominal fee. As those Logic users with a blue key will be well out of any warranty period by the end of 2005, the announcement shouldn't affect too many people. There's more on this at www.apple.com.

Logic Tips You probably realise by now that you can copy plug-in effects between tracks and insert slots using the hand tool in the Track Mixer window. What you may not realise is that you can also drag plug-ins in the Arrange page's channel strip by dragging while holding down the Apple key — of course, you can only drag plugins between insert slots here, as there's only one channel shown! If you have a lot of tracks in your Arrange window, the Track Mixer can become unwieldy, forcing you into a lot of scrolling to find the tracks you want. If you create an empty Folder in the Arrange page, open it and create copies of the Tracks which the backing vocals are on, for example, double-clicking this Folder will show just these Tracks in the Track Mixer. You can repeat this for groups of drum tracks, guitars, and so on. You must make sure the Track Mixer's Link mode is on for this to work, though.

Networking Noises I was setting up a simple Logic system for a friend recently, which comprised an M Audio Ozone MIDI controller/USB audio interface and an Apple iBook. Installation went fine, but I did come across a problem when we tried to record some electric guitar. There was a loud buzz, which changed in intensity depending on whether the strings were touched or not. Unplugging the iBook from the mains and running on battery power helped reduce the buzz. The next hour was spent in fruitless re-plugging of mains leads as I tried, without even the aid of a Hobnob, to find the source of the problem. Finally, I tried turning off the Linksys wireless network router (which was in the same room) and the buzz disappeared. I know that badly shielded guitars (yes, Stratocaster, I'm talking to you) are susceptible to interference from TVs, but I hadn't previously considered wireless networking as being a potential problem. So if you're plagued with strange interferences you could try switching off wireless networking first, before you get down and dirty with the cables and mains sockets! Published in SOS November 2005

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Logic Notes

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

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

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Managing iLok Plug-in Licences

In this article:

Managing iLok Plug-in Licences

Transferring Licences Technique Buying An iLok Published in SOS November 2005 Zero Down Time Don't Shoot The Messenger! Print article : Close window

Technique : Pro Tools Notes

There can't be many Pro Tools users left who haven't had to invest in an iLok key in order to run their favourite plug-ins. But did you know that you can insure, transfer and even buy and sell iLok plug-in licences electronically? Mike Thornton

Plug-ins are one of the elements that make Pro Tools so versatile, and since Digidesign opened their TDM, RTAS and Audiosuite protocols up to third-party developers, we now have an enormous range to choose from. However, plug-ins have always been susceptible to unauthorised copying, and software manufacturers and developers have felt the need to protect their investments by copy-protecting them. First it was those dreaded floppy key disks that became unreadable just at the wrong time — and when manufacturers recognised this and gave you a spare floppy key disk, they found that users installed another copy of the software on a second system and used the spare key for that! Next we had, and still have, authorisations that are computer- or drive-specific, which mean that plug-ins are only available to use on a specific computer, or you have to carry around a separate hard drive just to make your plug-in authorisations portable. What was needed was a more portable key that would enable the user to work with their plug-ins on whichever machine they were in front of, whether then owned it or not. This has become more and more crucial as the Pro Tools Session format replaced the two-inch tape as arguably the universal format for moving recording sessions about the world. Recognising this requirement, PACE Anti-Piracy went back to the drawing board and developed the iLok. An iLok is a

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Managing iLok Plug-in Licences

small key-shaped 'dongle' which you can plug into any USB port on a computer. The plug-in software checks for its presence, and if the iLok is detected and contains the correct authorisation, permits you to use that plug-in on that system. Take the iLok away and the software is no longer authorised and won't function. This makes the iLok a very valuable item as, in effect, it is worth the combined value of all the software plug-ins that are authorised on it — and each iLok can store up to 100 authorisations. It is also small, which makes it very easy to lose or steal. To assist software manufacturers and developers, PACE set up a 'central clearing house' and management web site called iLok.com (below). This site enables the manufacturers and developers (who PACE call Vendors) to 'deposit' licences (sometimes called 'assets') into users' accounts. The user is then able to download these onto their iLok. The web site also enables users to 'synchronise' their iLoks with ilok.com for users to upload licence details from the licence cards that some Vendors supply, for example, with retail products. PACE have also introduced a number of additional licence-management features aimed at helping us the end user to manage our iLok authorisations (for free) and protect our valuable iLoks (for a fee!) and it is these features we are going to look at in this workshop.

Transferring Licences You can transfer licences between different iLoks. This service is free for any iLoks you own within one account on iLok.com. However, you can also transfer licences from one iLok to another on different accounts for a fee of $25 per licence, which enables you to buy plug-in licences 'second-hand'. "Moving licences from iLok to iLok within a single account is free," says Andrew Kirk, Vice President of PACE Anti-Piracy. "This was the most demanded feature from user groups. A lot of people have two or more iLoks that they've collected over the years, going back to their Mix systems. Now they can combine those multiple licences onto one iLok, or organise their plug-ins on different iLoks as needed." * To transfer within an account (free) To transfer licences between your own iLoks, first make sure your iLoks are plugged in to USB ports on your computer and then log into your account on iLok. com and choose the Transfer Licenses option. You will then need to synchronise your iLoks to make sure the data held on iLok.com matches what is on your iLoks. This will also tell you which licences are on which iLok. Make sure you don't remove your iLoks from the computer at this point. Then choose the source iLok and which licences you wish to move, select the destination iLok from the

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Managing iLok Plug-in Licences

simple step-by-step instructions and its done. I used this recently to rearrange my iLok licences so that for certain types of work away from my studio I only need to take one iLok with me now. Be aware, though, that you can run into problems synchronising and The iLok.com web site allows you to audit transferring licences if you're your plug-in licences and transfer them connecting to iLok.com from behind a between multiple iLok keys. proxy server or firewall, which will be the case for many corporate and educational users. If you do run into the dreaded "unexpected authorization error 411", your only option is to find another machine that's connected to the Internet in a different way. * To transfer between different accounts ($25) Transferring licences between different accounts (also known as transfer of ownership) is a very similar process to transferring within your own account and is great if you want to buy and sell any plug-ins from other users. It is handled securely by iLok.com, but be aware that this type of transfer is not free and iLok. com will charge you $25 per licence to transfer ownership. So, for example, if you bought an upgrade with one the Massive Pack deals and you found you already owned some of the plug-ins in the bundle, you could sell the duplicates to a friend — if the Vendor makes it possible. Andrew Kirk explains: "Some vendors [software publishers] allow for iLok-to-iLok transfer within an account, and some don't; however, iLok.com will tell you if they don't. Also, some companies don't allow any transfer of licences at all. This is based upon each software publisher's business policies." So it is worth checking with a software publisher before you try to buy or sell a plug-in, in case they do not allow transfers and you end up buying a plug-in you can't transfer onto your own iLok! It is up to the individual software publishers to determine what transfers they will permit under the terms of their licence to you as the end user, but of course you all read and are fully conversant with the terms and conditions of each software licence... after all, nobody would consider just clicking OK to the 'Have you read the term and conditions' tick box without studying them in detail, would they? Another factor to be aware of is that although iLok.com will email the Vendor to inform them of the licence transfer, the software manufacturer won't necessarily provide support for transferred plug-ins: as before, it is the choice of the software manufacturer, and PACE are bound by the individual Vendors' policies.

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You can buy blank iLoks direct from PACE via the Buy section of iLok.com, or from Digidesign and their dealers. Additionally, Sonic Distribution are official distributors in Europe, and you can buy iLoks from their site at www.sonicdistribution.com. As well as iLoks themselves, PACE also sell a short USB extension cable they call a Dongle Buddy. Like any other hardware device, the iLok is susceptible to damage, and PACE found that a lot of the iLoks that had failed had been broken by for example, moving the computer too close to the wall or dropping something onto the keyboard. By taking the tension off the iLok, the Dongle Buddy helps to protect it from accidental damage. It has to be said, though, that the vast majority of iLoks work without any problem, and if you suspect your iLok is starting to fail, you can be pre-emptive and buy a new one before transferring the licences across for free, with no down time!

Zero Down Time PACE also offer a Zero Down Time package that is designed to get you back up as soon as possible after an iLok loss or failure. You should note that it is something you must sign up for before you lose your iLok, and it is a paid upgrade service available from iLok.com. With ZDT you can be back up within 15 to 20 minutes, but without it, it could be several weeks before you finally get sorted. ZDT is a kind of insurance policy and costs $30 per iLok per year or until the service is used. To use it, log into your account on the iLok.com site and follow the RMA (Return Merchandise Authorisation) procedure. The site will automatically deposit two-week temporary licences for all your covered plug-ins into your iLok account. You then use the site to download those temporary licences in the normal way onto a spare iLok and you are back in business. Now, as instructed, send the damaged iLok to PACE immediately. Once they have analysed it and validated it, they will put replacement full unlimited licences into your account for you to download before the temporary licences expire. As this process can take up to two weeks it is vitally important to get the damaged iLok off to them ASAP! Otherwise you will end up without cover, as the temporary licences will have run out and PACE won't have validated the damaged iLok to enable them to supply you with replacement licences. The damaged iLok has to go back to PACE in the USA, so it can take some time for them file:///F|/SoS/SoS%2011-2005/ptworkshop.htm (4 of 7)10/19/2005 9:44:32 PM

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to receive and analyse it, and PACE are not legally permitted by the Vendors to replace the licences without having done this. If your iLok is found to be damaged beyond repair or recognition so that PACE are unable to validate what licences are held on it, you will need to claim from your insurance company for the loss. Before you can transfer any plug-in licences, your iLoks must be 'synchronised'.

If you aren't covered by ZDT then you can still use the iLok RMA procedure and send them the broken iLok, but you will be without your plug-ins until PACE have received your damaged iLok and validated it. That could be up to two weeks or so, as ZDT iLoks will take priority! Also note that PACE will only replace the iLok free of charge if it just stopped working and was found to be defective under their warranty terms. If was damaged by force (and this includes accidental damage) then you will need to buy a new iLok too. If you are covered by ZDT and your iLok gets lost or stolen then PACE will still provide you with a set of two-week temporary licences. However they won't be able to supply you with replacement full licences as there is no iLok to analyse to prove what was really on it. Under these circumstances you will need to contact each software manufacturer and plead your case with each one, as they have different policies on replacing lost licences. Obviously you will also need a new iLok, too, but that will probably be the least of your worries. PACE and iLok.com operate a seven-day period of grace from the day when you take out ZDT to the day when they will provide temporary licences. This is to prevent you trying to subscribe to the service 'after the event'. However PACE haven't missed a trick, and if you are caught without ZDT you can bypass the seven-day grace period by paying an expediency fee of $100! There are a small number of software developers who are not fully involved with ZDT: Waves, DUY Research, Grey Matter Response Inc and Audio Ease. Even if you have ZDT, you will still need to deal directly with these manufacturers if you lose or damage an iLok containing an authorisation for one of their plug-ins. Consequently, you should consider having these manufacturers' plug-in licences on a different iLok, as if a problem arises, you may need to send the appropriate iLok to the appropriate Vendor rather than to PACE. These Vendors all have the ability to provide temporary licences to you but you will need to deal with each one separately. Waves state in their FAQ that they won't reissue replacement licences for a lost or stolen iLok and that you should insure against its loss: don't forget to arrange cover for the value of all the software authorised on it and not just the value of the iLok itself! As for defective or damaged iLoks, Waves advise you to follow the iLok.com RMA procedure. Then contact Waves Tech Support (using the Tech Support link in your Waves account), and send them your email correspondence file:///F|/SoS/SoS%2011-2005/ptworkshop.htm (5 of 7)10/19/2005 9:44:32 PM

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with PACE to confirm the return of the defective iLok key. Make sure to include the old iLok key serial number and the RMA number in the email, and once PACE have validated the damaged iLok, Waves will 'reset your account'.

Don't Shoot The Messenger! PACE have had a rough ride in some quarters over the years, but in my opinion it is mostly not their fault. If we where all nice honourable people, software manufacturers wouldn't need to use copy-protection products like the iLok system to protect their business and investment through developing all these creative and useful plug-ins. I accept that PACE have a duty to provide a reliable product so that we, the software developers' customers, are able to use our legally purchased plug-ins without problems and hindrance and I believe PACE have worked hard to develop a reliable system, as subcontractors to the software developers, to deal with end users like us through the iLok.com service. "We must be doing something right," insists Andrew Kirk of PACE. "When software publishers offer options between hard disk challenge/response and iLok, the end users often choose iLok for the portability and convenience." The other criticism levelled at PACE and iLok is that it's unfair to charge us for services like ZDT: after all, the argument goes, we have legitimately bought our plug-ins, and if we have a problem it should be sorted out as part of 'customer support'. Well it can and it does. If you are not covered by ZDT then, as with most products, all it costs you is to ship the damaged item back to the manufacturer or their agent (in this case PACE) and they will fix the problem and send you back a fully functioning device. The charges for that service would reflect whether the item was in warranty or not. What PACE and iLok are offering with ZDT is an insurance policy, similar in many ways to the sort of policy we take out to cover the possible loss of all our credit cards. In that case, all we have to do is phone one number and that company will contact all the different card companies on our behalf and get the lost cards blocked. We don't tend complain about paying for that sort of service, and for professional users, $30 per iLok per year is peanuts compared with two weeks' lost work and goodwill because you don't have any plug-ins on your system to edit and mix with. Published in SOS November 2005

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

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

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Mastering Reason 3 Mixes

In this article:

Mastering Reason 3 Mixes

Why Mastering? Technique The Equaliser Mastering For The Masses Published in SOS November 2005 Stereo Image Print article : Close window What's The Frequency, Technique : Reason Notes Kenneth? Some EQ Guidelines Dynamics The Other Suite Of Four Mastering Tools

Reason 3 has several new tools for pumping up final mixes and creating that 'finished product' sound. Simon Price

Part of the appeal of Reason is the idea that you can create tracks from start to finish in one integrated environment. However, for a while the program struggled to compete sonically with the final mixes possible in packages featuring bundled 'mastering' plugins. Reason 3 addressed this shortcoming with the introduction of several high-quality processors, meaning that now you really can go from scratch to a polished final track in Reason.

Why Mastering?

Reason 3's MClass Mastering Suite is a Combinator patch containing all four of the new mastering processors.

Last month, we looked at how to improve your mixes in Reason, and mastering is, in some ways, an extension of this topic. However, mastering is also a specific discipline that exists for two purposes: tweaking the final stereo mix for creative reasons, and optimising the output to suit the final delivery medium. These days, the stereo mix will either be destined for a CD or MP3/AAC delivery format, both of which will usually be created from a 16-bit, 44.1kHz stereo audio file, so mastering aims to contain your mix within the limits of this format. In the days when vinyl was the main music format, mastering was particularly aimed at the technical limitations of the format, such as stopping the needle from jumping out of the groove! Nowadays, when most musicians talk about mastering they tend to be thinking about working on the sound of the mix, rather than its technical aspects. file:///F|/SoS/SoS%2011-2005/reasontech.htm (1 of 7)10/19/2005 9:44:37 PM

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Mastering involves listening to and altering several attributes of the final stereo mix signal. The frequency composition of the mix is scrutinised, and EQ may be used. The spatial (stereo) characteristics of the mix may be altered, to create space or improve definition, and for technical reasons. Finally, level and dynamic range are massaged to make the most of the technical limitations of the final medium, to suit the type of music and to create a consistent, polished-sounding product. Each of Reason's mastering processors takes on one of these aspects of the mix. The devices we're going to look at are the MClass Equaliser, Stereo Imager, Compressor and Maximiser. These devices can be used in the rack like any others, but in the context of mastering they should be inserted between the main mixer's stereo outputs and the Hardware Interface device (see screen, right, for the rear-panel cabling). If you wish to use all the devices, you can choose to create the MClass Mastering Suite (see screen at the start of this article), which is actually a Combinator patch that includes all the MClass effects, cabled and ready to go. The Mastering Suite also lets you play with some preset mastering patches to get an idea of what can be achieved.

The Equaliser The MClass Equaliser (see screen below) is a much smoother processor than the older PEQ2 EQ device, and is ideally suited to making adjustments to the final mix. The device offers low and high shelf EQs, two fully sweepable parametric EQs and a low-cut filter. Each EQ stage can be switched in or out with the small red button next to its name. The first thing to say is that you may as well leave the low-cut filter switched in. It simply rolls off everything below 30Hz, which is almost certainly only going to be nasty stuff that you don't want to retain from samples. By far the most useful EQ modules are the two parametrics, as you can configure exactly how they work. They're your main tools for identifying and solving frequency-related problems in your mix and they have the following main controls: The Frequency control, which centres the frequency that the EQ is boosting or cutting. The Q control, which determines the

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Effects devices that you intend to use for mastering (such as this MClass Maximiser) must be connected between the main mixer and the Hardware Interface outputs.

Mastering Reason 3 Mixes

range of the effect, from a wide area to a reasonably narrow peak/notch. The Gain control, which provides +/-18dB of boost/cut — way more than you'll ever need. The next issue is what to do with them. First, have a good listen to the mix you're working with and try to identify anything that is niggling you. This could be annoying boominess, boxiness, muddiness, resonance, sharpness, harshness, or lack of clarity or presence. These are all words often used to describe problems in the frequency content of a music mix, and they tend to be associated with certain areas of the frequency spectrum. If you've never done this type of listening before, now's the time to learn how — it's a lot easier than you might have been led to believe. When you've identified once which frequency band is associated with a particular problem, it will always stand out to you in the future and you'll have an idea what to do about it. Listen to one of your songs and enable one of the parametric EQs. You might already be able to hear some areas of the spectrum that poke out, especially if you're comparing with a commercial CD (see 'The Other Suite Of Four Mastering Tools' box). Whack the Gain control up on your enabled EQ. Sweep the Frequency control slowly up and down until the offending area sounds much worse. What you are trying to do is find where a problem is and exaggerate it, which makes it really obvious and gives you confidence that you really were hearing a problem.

The MClass Equaliser's parametric EQs are perfect for tracking down and smoothing out frequency problems in your mix.

Now turn the Q knob so that you're only boosting the 'nasty bit' and not its surrounding areas, and pull the Gain control down, so that you're cutting slightly at this frequency. If your mix is fairly good in the first place, you should only have to cut by 2-3dB to smooth things out. If you're cutting areas by much more than this — say, 6dB — you should consider going back to an earlier stage of your mix and trying to sort out why there is an annoying peak or clutter at this frequency. Normally, EQ is used to cut frequencies, but you can boost an area if necessary. (See 'What's The Frequency, Kenneth?' box for some useful EQ starting points.) Now that we've basically covered the use of the parametric modules, what about those shelves? The high shelf is pretty useful for giving a little boost to mixes that lack high-end life and sparkle, but the low shelf needs to be used very cautiously. It can be employed to roll off very low-end bass, although you're probably better off using the more precise parametrics. The problem with the shelf is that unless

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you have Q at minimum, there is a frequency boost around the cutoff frequency when you're cutting. This is pretty normal for an analogue-emulating shelf EQ, but this bump tends to boost the nasty boomy and boxy frequencies between 200Hz and 500Hz. The last thing to say about the MClass Equaliser is that it doesn't have an output gain control to counteract any overall level change caused by your treatments. This means that changing the EQ could alter the response of the Compressor or Maximiser that comes afterwards in the Mastering Suite. Adjust the input gain on the Compressor to counter this.

Mastering For The Masses You're probably waiting for the bit where I say that all these mastering tools are no substitute for a man in a professional mastering studio with 10-grand speakers, 'golden ears' and a beard. Certainly, if you're releasing a commercial CD this kind of highly skilled post-production is a valuable investment. But there are great reasons for all producers and musicians to learn to include mastering in their studio work. For a start, lots of music is now released non-commercially or independently, via the web or podcasts, and these tracks will not get the benefit of pro mastering. Secondly, many engineers will run their mixes through a maximiser to reference how their mix will sound when it's been mastered. Most importantly, if you are mastering with the full mix available to you, you have many more options than a mastering engineer working on a two-track mix. You can take what you learn from listening and tweaking at the mastering stage and go back and make changes to the mix, rather than just processing the stereo signal.

Stereo Image The MClass Stereo Imager (below) lets you tweak the stereo width of your mixes in quite a sophisticated way. Width can be narrowed or widened separately in two different frequency bands. A crossover control sets where the high- and lowfrequency portions of the mix are split, and you can solo each portion for reference. Typically, this processor is used to ensure that the low-frequency parts of the mix are close to mono, which keeps the mix tight and is also technically advisable for playback on stereo speakers, and for vinyl. The higher frequencies can be left at their original width, or even widened, if desired. Don't overdo widening, as this can just sound weird and chaotic. The Stereo Imager splits the mix into two frequency bands that have no crossover, a trick that can be used for other purposes than those intended. For example, check out the Mastering Suite preset patch 'Dual Band Compressor', which sends the high and low components separately to two compressors.

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What's The Frequency, Kenneth? Some EQ Guidelines Low, bassy boominess usually occurs around 100-250Hz. Try cutting here, and also compressing the instrument(s) responsible. This will make the low-frequency sound sources punchier and more defined. A boxy, 'roomy' mix will benefit from a cut at around 350-600Hz. This can make a mix sound very tight, but if overdone may make it too 'dry'. Muddiness can be addressed by reducing peaks anywhere between 600Hz and 1kHz. Cutting at around 1kHz can reduce sharpness, and make mixes sound more powerful, but can also be overdone. If this happens, the mix may sound artificially processed, like the effect created by the 'loudness' button on a dodgy hi-fi. Shrill and piercing sounds tend to live at around the 2.5kHz mark, so a cut here can smooth a mix. Conversely, if your mix is a bit flat and limp, a boost here can liven things up. Mixes that lack presence, shine or the sought-after 'air' might need a subtle boost between 3kHz and 6kHz.

Dynamics Compression, limiting and maximisation are all provided for in the MClass Mastering Suite. A decent mix will have made good use of compression before the mastering stage, but a little extra on top from the Compressor can help bind the mix together. Try using a medium threshold and a ratio of 1.5:1 or less. As I said last month, the MClass Compressor is probably more useful as a generalpurpose tool in Reason than as a mastering processor (see screen below). The MClass Maximiser (bottom of page) is where most of the magic happens, and where you can get the results that most people probably associate with mastering. The Maximiser does three things (which we'll look at in more detail in a moment): it boosts the overall average level of the mix; it prevents the signal from clipping (hitting 0dB); and it adds some subtle 'analogue-style' distortion. Maximisation is achieved in the Limiter section. The technique behind maximisation is limiting peaks in the mix so that you can push up the quieter sections, increasing overall loudness. This doesn't mean turning up quiet sections of a song and turning down the loud bits. A maximising

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The ingenious Imager lets you pull the lowfrequency components of your sounds into the centre of the stereo image, tightening up the mix.

Although this device is better as a generalpurpose compressor, a little touch of MClass compression brings the mix together and adds a bit of punch at the mastering stage.

The MClass Maximiser: this peak limiter and soft clipping processor can make the difference beween a decent-sounding mix and a polished, defined, pumped-up floorfiller. Or it can make your bad mix bad but louder.

Mastering Reason 3 Mixes

limiter compresses the transient peaks within the signal from moment to moment. This means that sections of the song that were already peaking at maximum can be turned up. The brain doesn't notice if fast peaks are limited, so if you don't go overboard the mix can sound a lot louder, while peaking at the same level and without sounding more compressed. The screens on the right show two waveforms of the same short section of music. The first waveform is the finished mix, but with no maximisation. The second waveform shows the mix with heavy maximisation. Both waveforms peak at -0.2dB but the second sounds a lot louder. Applied at this extreme level, maximisation can be heard: the mix will sound as though it's constantly flat out, straining at the edges and almost breaking up. This is the kind of limiting used on most commercial music radio, or TV commercials. More subtle maximisation, however, gives a louder mix with no perceived loss of dynamic range. Looking at the Maximiser (see previous page), the signal flows from left to right. The Input Gain control effectively pushes the limiter harder and harder, so this is the main control for setting how loud you wish to make your mix. However, there is an Output Gain control, which can then push the peaklimited signal further into the Soft Clip section. Any boost you give the signal at the Output stage is basically clipping the signal (going over 0dB), but the Soft Clip stage is transparent enough that you can squeeze a few extra dBs through, if you really want them. The Attack and Release controls really need experimenting with for each song. Often, the slower you can get away with setting these, the more transparent the results. Set them too The same piece of mixed music before (top) fast and the mix will pump and sound and after heavy maximisation. Both peak at compressed. Set them slow and the the same level, but the second sounds much sound is smoother, especially if you've louder. managed to achieve a pretty loud mix in the first place. If sharp peaks (such as drum hits) suddenly poke through, you'll need to use a faster response speed. The MClass Maximiser can stop your mix from ever clipping in two different ways. Firstly, if you set the Attack Speed to Fast, and enable Look Ahead mode, the mix will be prevented from clipping (provided you don't turn up the Master Output above 0dB). Look Ahead mode imposes a 4ms delay on the signal, which the Maximiser uses to get a head start on any sudden transients. This stops any sharp peaks from 'surprising' the limiter and getting through. The second way in which the device imposes a so-called 'output ceiling' is by using the Soft Clip stage. If you switch this in and set the Soft Clip Amount knob at minimum, the signal will not 'break the rules' by hitting 0dB, but will be able to 'flatline' if pushed file:///F|/SoS/SoS%2011-2005/reasontech.htm (6 of 7)10/19/2005 9:44:37 PM

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hard, which amounts to the same result sonically. However, if you bring up the Soft Clip Amount, the top few dBs of the signal will be rounded off, more as if you were overdriving an analogue device. Unless you really push the mix, this will sound like a very subtle warming distortion.

The Other Suite Of Four Mastering Tools Reason provides four MClass mastering processors, but the other 'big four' are ears, reference CDs, speakers and headphones. The best way to create a mix you'll be proud of, and learn a huge amount at the same time, is to stop and listen to some CDs that have a sound you'd like to get close to. Listen via the speakers you're using with Reason, and also with headphones. Try to listen from a technical point of view, especially for things that you might find surprising. How loud is the mix, and is it louder when there are more parts playing than just one or two? Can you actually hear any compression working in the mix? How much low end is there, and how does the overall frequency content compare to your mix? What else is different, and what might you try to make your mix sound similar? Published in SOS November 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

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PC Notes

In this article:

Automatic Logging On Multi-user Accounts PC Snippets User Account Tweaks

PC Notes News & Updates Published in SOS November 2005 Print article : Close window

Technique : PC Notes

This month we explore the relevance of user accounts and the vital contents of the Windows Documents and Settings folder. Martin Walker

I don't know about you, but I've never personally needed to create multiple accounts or use passwords on my PCs — I just have the single account that was created when I first installed Windows XP. This has Administrator status, so that I have total system control over my PC. However, if you wish, you can create other accounts with more limited status, so that other people can use the PC but not accidentally trash it. By default there's also a Guest account with even fewer powers, so that your friends can go on-line to receive emails, and so on, without you worrying about security. I discussed the pros and cons of multi-user accounts for the musician back in the PC Notes column of SOS March 2003, and came to the conclusion that while they're helpful in many instances, they're not suitable for creating several setups optimised for different uses, such as Games, Music and Internet. This is because although each account has its own Start menu, colour scheme, O/S tweaks and so on, the applications themselves are installed in a vast Program Files folder, references to them are stored in one big Registry, and any virus contracted from the Internet can immediately infect the entire system. For me, the beauty of a multi-boot PC is that each Windows installation is housed in its own self-contained partition, with its own slimmed-down set of installed applications and more compact registry. Even if the worst happens and your Internet-enabled partition falls over, the likelihood of your music partition being affected is minimal, and zero if you use a boot-manager utility such as Bootstar, that makes Windows partitions invisible when they're not active, or have multiple drive caddies so that your music drive isn't even plugged in when you're accessing the Internet.

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PC Notes

Automatic Logging On If, like me and many other musicians, you're the sole user of your PC and you don't employ passwords and multi-user accounts, you don't have to log on manually each time you boot up your PC. Following the steps in this checklist will ensure that Windows takes you straight to the desktop every time you boot: The Welcome Screen must be switched on: use the same dialogue box as the Fast User Switching option (see main text). The Guest account must be switched off in the User Accounts applet, and here must be no other accounts set up. Your user account must have no password.

Multi-user Accounts Having said all the above, I'm sure that some of you have PCs with multi-user accounts that are used by all the family, and in this scenario separate accounts do offer some extra security for your music applications. If your PC does have multiple users, one related 'music tweak' worth implementing is disabling Fast User Switching: select User Accounts in Control Panel, click on 'Change the way users log on or off', un-tick the box marked 'Use Fast User Switching', then click the Apply Options button. While being able to switch rapidly from one account to another without having to close any programs can save time, it can consume lots of resources that are better made available to audio applications.

While you may only have a single user account (as shown here), its settings and, more importantly, the contents of its associated folder, may be very important to you as a musician.

Every time a user logs on to a Windows XP computer, Windows automatically creates a profile for that user, unless one already exists, and these profiles are stored in the 'Documents and Settings' folder. Each profile contains that user's documents, plus any application settings specific to that user, such as their desktop arrangement, Internet favourites list and cookies, Start Menu contents and arrangement. Personally, I've never stored any of my documents, music or other multimedia data in the Windows-designated 'My' folders ('My Movies', 'My Music', and so on). While such folders are obviously convenient for those whose hard drives

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PC Notes

comprise one large partition, and especially when lots of people use the same PC, I'd much rather have all my data on another partition, or preferably another drive, where it can not only be accessed by each of my different Windows installs, but also be more easily backed up to CD or DVD for safety.

PC Snippets Intel now seem to be concentrating on releasing more efficient processor chips in 2006, and although laptops are obviously the prime target market for such devices, designers are also increasingly looking at ways to make more compact desktop and server PCs that run cooler and quieter — always good news for the musician. January 2006 should see the release of the first Napa notebooks from a variety of manufacturers, all of which incorporate a new Yonah dual-core CPU and are expected to be up to 20 percent smaller than existing designs. However, these will rapidly be superseded by the Merom 64-bit-capable dual-core processor chip later in the same year, with similar efficiency but better performance.

With 670 categorised entries, the Ultimate List Of Free VST Plug-ins certainly seems to live up to its name.

The clever Maxivista utility (www. maxivista.com) that lets you use your laptop or other PC as a second monitor is now up to version 2.0. It uses a special driver that tricks your main PC into believing that two monitors are connected and then sends a compressed video image to the second PC via a wireless LAN or Ethernet connection. Version 2 lets you control up to four computers from a single keyboard and mouse. When you move the mouse position to a particular monitor, both the mouse and your keyboard are then in control of that PC. The WIndows clipboard is also shared, so you can copy/paste data between machines. A free trial version is available, and the full version currently only costs $29.95. There are now so many freeware plug-in effects available that it can sometimes be difficult to track down what you're looking for. Enter Dodo Bird, whose web site, at www. sadglad.com, not only sells his music CDs, advertises his gigs and offers his intriguing and free Super Eel VST plug-in, but also maintains 'The Ultimate List Of Free VST Plug-ins'. The total currently stands at 670 entries, spread across 27 categories, all with a thumbnail pic and direct link to the developer's own web sites. Makes you feel as though all your birthdays have arrived at once!

User Account Tweaks The Start menu includes the common program groups that are installed by Windows and appear for all users, such as Programs, Accessories, Multimedia and so on. I always find it difficult to find specific applications on other people's PCs, because they tend to accept the install defaults and end up with one big list, whereas all my PCs have additional folders, with names such as Audio Editors, file:///F|/SoS/SoS%2011-2005/pcnotes.htm (3 of 5)10/19/2005 9:44:42 PM

PC Notes

Audio Utilities, Plug-ins and Soft Synths, which makes it much quicker and easier to find whatever I'm looking for. However, while it's easy to drag and drop existing shortcuts to rearrange them on your Start Menu, to add new folders you need to know where they are stored inside the 'Documents and Settings' folder. Inside this folder, most users will probably find a selection of sub-folders, called Administrator, All Users, Default User, Local Service and Network Service, plus one with whatever user name you typed in when first installing Windows, and any other account names you've created for other users. Only accounts with Administrator status can change common program groups, while — as its name suggests — All Users contains all the settings common to each account. Every new account you create is initially a copy of Default User, while Local Service and Network Service both provide extra security, by running various Windows Services with non-Administrator status. Most people will find that the vast majority of their personal data only appears in the All Users and 'own name' account, and if you're the sole user of your PC it makes little practical difference which one is used. When you install software, you may be given the option to 'Install shortcuts for current user only' or 'Make shortcuts available to all users'. This option defines in which of the two folders mentioned above they are saved. Similarly, when you uninstall any application whose info has been placed in the All Users folder, you may get a warning message something like: 'This will affect all users. OK to proceed?' Although you can navigate to the current user's account files by hand via Windows Explorer, here's a really quick shortcut: using the Start Menu's 'Run' command (most quickly accessed using the Windows-R shortcut), simply type in '.' (a single period character) and press return, and the current user's folders will appear inside a new Explorer window so that you can explore or modify them. You can now click on the Start Menu folder, for example, and add new folders to the Start Menu using the normal right-click New Folder option, or click on 'Send To' and add new shortcuts to your Send To options using 'New Shortcut' (as an example, I've added a shortcut to Notepad, so that I can explore the contents of any file in Explorer). Even if you've never stored a single document in your account's 'My Documents' folder, you'll find a set of tiny files, your desktop 'themes', stored here by default, and you should back these up for safety. If you've spent a long time choosing colours, font sizes and so on, in order to cram the maximum number of audio tracks on to your sequencer screen and make it as legible as possible, it would be pretty annoying to have to go through the process again should you ever need to reinstall Windows. Your Internet Favorites (sic) are also stored here, but possibly the most important folder is the one named Application Data. Some applications use this folder to store all your personal presets and user library data. For instance, on my PC the 'C:\Documents and Settings\General\Application Data' folder contains the entire

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PC Notes

settings collection for Cubase SX, complete with all patch names, presets for the MIDI effects, Logical Editor, Micro Tunings and so on. This is where you should place any patch-name scripts that you create or download (a common cause of confusion for Cubase users). And if you create any new presets that fall into any of these categories, this is where they will be stored. Other vital files in the Application Data folder that I've found on my PC are the current database files for my AAS Tassman, Lounge Lizard and String Studio synths, containing not only the factory-installed presets but every one that I've created myself and those I've downloaded from the AAS web site over the years. NI's Reaktor 5 also stores its User Content here by default, although I've amended its Preference settings to point to another drive. My Emu 1820M interface also stores the current Patchmix DSP Session here, so that it always appears exactly as I last used it, while my word processor stores its default document template here. To sum up, whether you're the sole user of your PC or have lots of different account users, the Documents and Settings folder may contain a host of vital files that took you a lot of time to create but are not saved elsewhere. It's important to back these up by copying the files onto another partition, drive or CD R/DVD-R disk, so that if you ever have to change your hard drive or reinstall Windows after a re-format, you won't lose them. Published in SOS November 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

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Pointing Devices For The PC Musician

In this article:

Mouse Features Ergonomics & RSI Mouse Resolution & Scanning Rate Cordless Options Going For A Spin Keyboard Shortcuts Keep Using The Tablets More Exotic Options Final Thoughts MIDI Controller Vs Pointing Device

Pointing Devices For The PC Musician PC Musician Published in SOS November 2005 Print article : Close window

Technique : PC Musician

If you've had enough of chasing a mouse around your desk, there are many other ways of controlling the onscreen pointer of your music software. We examine the options. Martin Walker

The graphical 'point and click' interface of modern computers is now second nature to most of us, and of course the humble mouse is its primary tool for both pointing and clicking. Many people buy a PC system and carry on using whatever mouse was bundled with it, but others (particularly game players) may explore other mouse designs and variants in search of the perfect rodent experience, for reasons of accuracy and speed of response, or to gain access to more buttons, wheels and other luxuries. Ultimately, most people are looking for the most comfortable pointing device, or the easiest to use, or perhaps the most versatile, but there may be other considerations for the musician. Sometimes you simply don't have enough room on your studio desk to roll a regular mouse about, or want to use a device that can be bolted to your live rig so that it can't bounce about and fall down the back of your rack mid-gig. Fortunately, there are plenty of other pointing devices to explore, including trackballs, trackpads, graphics tablets and joysticks, plus more exotic options such as touchscreens and 3D motion sensors.

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Mouse Features It seems hard to believe that the original computer mouse was invented as far back as 1963 (by Douglas Englebart, at the Stanford Research Institute). However, we have Apple to thank for turning it into a low-cost device for use with their Lisa and Macintosh computers in the mid '80s. Like all early mice, Apple's first design used a rolling ball that was prone to picking up debris, requiring occasional cleaning to keep its performance up to scratch. The majority of modern mouse designs now use optical technology, with an LED that shines down at an angle onto the surface beneath it, to make any slight irregularities more obvious, and then a sensor that picks up the changing reflected signal as you move the mouse. The optical mouse requires no cleaning, and also works on a much more versatile selection of surfaces — including trouser legs and carpet! Many users also find that they last considerably longer than the mechanical variety. I've been using a couple of Microsoft basic Wheel Mouse Optical models for years with no problems at all. The majority of modern PC mice also incorporate a scroll wheel, which doubles as a third button, as well as letting you scroll vertically through documents. Some of the latest mice now incorporate 'tilting' scroll wheels that allow you to scroll both horizontally and vertically. Most modern browsers and word processors work with scroll wheels, scrolling by a user-defined amount each 'click'. This amount can be adjusted using the Mouse applet in Control Panel, from a single line to jumps of multiple pages. This setting is generally ignored by music applications, which instead use the scroll wheel for a variety of functions that are often not well documented. The easiest way to find out how your music app supports scrolling is to try using the scroll wheel in your sequencer by itself and in conjunction with the Shift, Control and Alt keys, to see what happens. Common functions are horizontal or vertical scrolling through an audio waveform or set of audio tracks, zooming in/ out and data entry. The functions offered may be context-sensitive — the scroll wheel may do different things depending what the mouse pointer is hovering over at the time — and you may get a higher scroll-wheel resolution when holding down the Shift key.

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Ergonomics & RSI Although some musicians are naturally adventurous, or just find the humble mouse frustrating to use, many more investigate alternative pointing devices at the onset of RSI (Repetitive Strain Injury). As its name suggests, RSI can result from any repetitive task, but it has been described as the greatest professional hazard of the modern computer professional, due to the incredible number of tiny wrist and finger movements that are necessary when one is operating both mice and computer keyboards. You should see your doctor if you experience any cramps, loss of sensation, tingling, swelling or inflammation in either your fingers or the muscles and tendons moving them. To minimise the chances of developing wrist pains, you should reduce your typing speed, take regular breaks and remove your hand from the mouse when not actively using it, rather than leaving it poised for hours on end in the same position. If you want to stick with a mouse but find a more comfortable option, manufacturers are very keen to help, offering a huge range of ergonomic shapes designed to fit your hand more comfortably or place it in a more natural position, plus gel wrist-rests to provide extra support. Also, make sure your posture is good (don't slump in your chair when making computer-based music, and try to keep your working surfaces at the most suitable height). If you want more general information about ergonomics in the studio, read our January 2002 feature on the subject, which is available on-line at www.soundonsound.com/sos/jan02/articles/ studioergonomics.asp.

Mouse Resolution & Scanning Rate Speaking of resolution, many of the exotic mice sold to games players offer a higher resolution than standard models, which means that you need to move the mouse a shorter distance to achieve the same movement on the screen. For instance, a typical mouse offering 400dpi or 800dpi (dots per inch) can move up to 400/800 dots on your screen for each inch you physically move the device, but gaming mice are available with resolutions of 1600 and even 2400dpi, to provide lightning-fast screen movements with tiny mouse moves. Resolution figures are not widely quoted, except in the case of high-end models, but if your PC system has been bundled with an anonymous mouse it's well worth investigating any bog-standard 'name' mouse from companies such as Microsoft and Logitech, as these may well provide a more responsive experience. However, unless you're going to be playing 3D games that need 'fasttwitch' acceleration, it's simply not worth buying a mouse with a resolution of over 800dpi. Even game players find these take some getting used to, and they often switch to a lower resolution for more precise mid-game activity. The sensitivity of your mouse will also depend on how often its position is updated by its Windows drivers. The 'refresh rate' is normally somewhere between 40Hz and 125Hz, and if you want to find out how often yours is being file:///F|/SoS/SoS%2011-2005/pcmusician.htm (3 of 10)10/19/2005 9:44:51 PM

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scanned, download Oliver Tscherwitschke's freeware mouse-rate checker utility at http://tscherwitschke.de/ download/mouserate.zip. Some exotic gaming mice have special drivers that boost the refresh rate to 200Hz, for increased sensitivity when playing 3D games, but again I suspect that would be counter-productive for the musician. In the case of a USB mouse, it might even cause problems if you had a USB MIDI or audio interface on the same port. The important thing is to adjust the 'ballistics' of your pointing device for the most comfortable pointing experience, using the various options provided by Windows for pointer, wheel, and button options in the Mouse applet in Control Panel. I find it especially important to activate the 'Enhance pointer precision' tick box, to provide fast moves over large distances, but more control when you're making small movements.

Cordless Options If you've only got a small amount of desktop space, or you need a change due to the symptoms of RSI, a trackball like the ones shown here (Kensington's Orbit Elite and their Expert Mouse Pro), or the Logitech Marble Mouse (shown on the page opposite) could certainly help.

For many musicians, a wireless mouse is a must. Team it with a wireless keyboard and you'll be able to control your sequencer from any part of the studio, simply by picking them up and carrying them about with you — and no tripping over cables. The wireless keyboard/mouse combo is ideal for the guitarist/ vocalist who needs to record a bit further away from a noisy computer, or for the musician who regularly moves between mixer and keyboard. Yet another advantage for the studio owner is security. You can keep your wireless mouse with you or locked away, to prevent clients or other visitors being able to open personal files easily. The only downsides are that the internal batteries can be a pain if you forget to recharge them regularly (although some modern designs can last up to three months on a single AA cell), they can make the mouse somewhat heavier than its corded cousins, and some game players grumble about a small time lag between moving a wireless mouse and seeing the corresponding movement on screen. However, like most other aspects of the pointing experience, such issues tend to be highly personal, and what suits one person perfectly might prove

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totally unusable to another.

Going For A Spin If you can't live with your mouse, why not buy a dead one? The trackball is essentially a mechanical mouse lying on its back so that you spin the ball directly. One instant benefit of this approach is that it takes up a lot less space, because you don't need a mouse mat or mousing 'area' on your desk. Since the trackball doesn't have to be mobile, it also means that you can buy models that you can fix permanently in place, which is perfect for the gigging musician or studio owner who doesn't want his mouse to 'walk' when their back is turned. Depending on the trackball design, the ball may have inertia, so that it continues moving after you release your finger, and the size of the ball can vary from a small marble to a golf ball, so it may take some time to find the perfect trackball for you. Models such as Logitech's Marble Mouse feel like an advanced mouse and offer a more For the ultimate in mobile convenience 'precision' experience for just £15, around the studio, why not try a wireless miniwhile the same company's Cordless keyboard with built-in trackball? Optical Trackman provides the total freedom of a large finger-operated trackball that requires no desktop space at all. Larger fixed trackballs are available from a variety of companies, but Kensington have arguably set the industry standard for the last 15 years. Their Orbit Optical Trackball has a blue-illuminated 1.75-inch ball for about £30, while their Expert Pro provides 10 buttons and a scroll-wheel for about £80. Both come with Mouseworks software that provides extra features and customisations. If you require a larger ball, for the ultimate in control, the BIGtrack has one three inches in diameter, and two huge buttons. It can help those with severe RSI or arthritis. Trackballs do require a different technique that may take some time to master, but those who persevere find it's well worth the effort. And anyone looking for the ultimate in convenience can buy a keyboard that incorporates a tiny trackball, such as the one shown on the left.

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Keyboard Shortcuts While using a mouse, trackball or tablet will always be the most flexible way to control every single function of a sequencer, you can generally improve your productivity a great deal by using keyboard shortcuts wherever possible, and this can also help reduce the onset of RSI. Most music applications provide copious shortcuts and some even let you redefine the default settings to your own preferences. If you don't know many shortcuts, try learning a new one on each session rather than reaching for the mouse, or even better, buy a dedicated computer keyboard from Logickeyboard (www.logickeyboard.com) with your particular music application's shortcuts already printed on the keys, or a set of self-adhesive Editor's Keys (www.editorskeys.com).

Keep Using The Tablets Nearly all laptops provide an integral touch-sensitive trackpad instead of a mouse, with two buttons beneath it to provide the left-click and right-click functions. I've met very few musicians who get on very well with these trackpads, even if they were using more advanced driver software that defines a strip down the right-hand side as a dedicated scroll-wheel area and detects double-clicks on the pad as left-mouse button clicks. The main limitation is that trackpads are so small: it's difficult to get all the way across the screen without lifting the finger off the trackpad at least once, yet if you do manage to increase the pointer speed sufficiently to manage this feat it ends up too sensitive to make accurate small movements. However, a larger trackpad or tablet is another matter, and many musicians have been won over by products such as Wacom's Pen Tablets (www.wacom. com), which use a special pressuresensitive stylus and a tablet that plugs into a USB port. This will obviously feel far more natural for drawing purposes than a mouse, but tablets also work well with most sequencers once you get used to them, being faster to use, more comfortable during long sessions and perfect for drawing in automation curves! A tablet and pen also has another distinct advantage over a mouse or trackball, in that you don't have to slide from one area of your screen to

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Wacom's Graphire 3 Classic graphics tablet has been used very successfully by many musicians, who find it faster and more precise with sequencers than a mouse, as well as more comfortable.

Pointing Devices For The PC Musician

another — instead, you can jump directly from point to point using pen (or absolute) mode, which experienced users find a lot easier and more precise, although most tablets also provide an alternative mouse (or relative) mode if you run into problems with a particular software application. Sadly, some music applications fall into this category, including Cakewalk's Sonar/Project 5 and Ableton's Live, amongst others. While most pointing operations work fine in absolute mode, dragging and dropping faders to a different position often results in them ending up at the top or bottom of their travel. There are also plenty of cheap tablets available from just over £20, but the quality of Wacom products and the versatility of their software drivers mean that they come highly recommended by a huge number of professional users. Graphic designers may find the larger (and therefore more expensive) models, such as the Intuos A4 or A5 more appropriate, especially as they offer 1024 pressure-sensitivity levels to control such software features as brush width or opaqueness, respond to pen 'tilting', and have a row of programmable buttons along the top edge of the tablet surface itself. However, for audio sequencing use only, these functions are largely irrelevant, so the 512 sensitivity levels of the cheaper Graphire series should be more than adequate. The Graphire 3 Classic A6 model with an active tablet area of about 5x4 inches can be bought for around £65.

More Exotic Options While mice, trackballs and tablets are the most useful pointing devices for the majority, some musicians have investigated other options. A welldesigned analogue joystick can place your hand, wrist, and arm in more restful and natural positions, with your thumb on the main button and your fingers curled round the stick (don't confuse these with the cheap digital joysticks, AKA joypads and gamepads, that simply provide up/down/left/right and 'fire' buttons). Some of the cheaper analogue joysticks sold for flight simulator games may not be precise enough to be used as a mouse replacement (although they can make great X/Y MIDI controllers), but two models can be recommended. These are 3M's Ergonomic Mouse EM500 (also known as the Anir Vertical Mouse), available in three models to suit different hand sizes; and (if you want the ultimate in freedom) file:///F|/SoS/SoS%2011-2005/pcmusician.htm (7 of 10)10/19/2005 9:44:51 PM

If you want to experience the ultimate in pointing freedom, you could try Gyration's Ultra GT mouse, incorporating gyroscopic technology.

Pointing Devices For The PC Musician

Gyration's Cordless Optical Air Mouse (www.gyration.com), at around £50. The latter allows you to use gyroscopic technology to operate the mouse on your desktop or in the air, which is great for presentations (and possibly for posing). However, for the pixelperfect positioning required to operate the knobs, faders and buttons of many sequencers and soft-synths you might find some cordless trackballs more suitable, including the unusuallyshaped Perific Wireless Dual Mouse (www.perific.com), which can either be used as a conventional mouse on the desktop or (once removed from its shell) worn on the hand and used in mid-air

The Perific Mouse looks odd at first, but can be used on the desktop as normal, or gripped in a variety of ways in your hand for a more mobile experience.

For many musicians, the ultimate interface would be a touch screen, so that they could interact directly with the image in front of them, draw in waveforms or automation data, and edit video with perfect precision. A classic example is Wacom's Cintiq, available as a 15.1-inch version with XGA 1024 x 768-pixel resolution TFT screen, an 18-inch version with SXGA 1280 x 1024-pixel resolution and a 21-inch version with UXGA 1600 x 1200-pixel resolution, all models having both DVI-I and VGA connectors that can be plugged into any graphics card. The screens are actually graphics tablets, operated via a pressuresensitive pen just like the Intuos and Graphire models discussed earlier, and can either be used on a stand, at various angles from vertical to nearly horizontal, or removed from the stand and used on your lap. As you might expect, their main disadvantage is much higher cost. At around £1150, even the smallest Cintiq 15X won't prove affordable for many musicians, while the 21-inch version is now around £2150.

Final Thoughts If you're about to buy an expensive pointing device, it's worth asking on a few music forums before buying, to see whether anyone else has already tried that particular model. Bear in mind that while many of the devices mentioned here can be plugged in and used with the standard Windows mouse drivers, some of the fancier devices need special drivers that may sometimes compromise sequencer performance, either by increasing refresh rates and disturbing the smooth running of other USB devices, or sometimes by directly conflicting with other drivers. Moreover, extra functions such as multimedia buttons and the like may not be recognised by much music software, although it may be possible to remap them using utilities such as the Girder remote-control utility (discussed in connection with more general remote controls back in SOS September 2003).

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This utility has proved so successful that it has moved from being freeware to being a boxed product, now available through www.promixis.com. Most of the devices described here are freely available from on-line retailers and shops, but in case of difficulty some specialist UK companies offer a much wider range of ergonomic solutions than most, including Keytools Touchscreens like Wacom's Cintiq, shown Ergonomics (www.keytoolshere, provide direct interaction with the image on screen — appealing, but expensive. ergonomics.co.uk), Barry Bennett Ltd (www.barrybennett.co.uk), the Keyboard Company (www.keyboardco.com), and Osmond Ergonomic Workplace Solutions (www.ergonomics.co.uk). Whichever pointing device you buy, make sure you tweak it to suit your personal preferences and achieve the maximum amount of control over your sequencing application. After all, the whole point of having a more refined pointer is so that you can forget about it and concentrate on being musically creative!

MIDI Controller Vs Pointing Device Some musicians get so fed up with using a mouse with their sequencer application that they buy a sophisticated MIDI controller with dedicated transport controls and banks of faders, knobs, switches and buttons. Having one of these will almost certainly make your sequencing experience a faster and more intuitive one but, as most users find, a dedicated controller doesn't completely replace the regular pointing device. Controllers are perfect for setting up mix levels, pan positions, automation moves and transport controls (and of course you can move multiple controls simultaneously if required), but the average sequencer has so many different windows, each with dozens and sometimes hundreds of individual controls, that even if you buy a unit dedicated to a specific sequencer it's unlikely that it can manage more than a good subset of the total number of functions. For the remaining ones the pointing device is still king, so you still have to find handy positions for the controller, keyboard and pointing device in your studio. One way in which manufacturers could help us out of this dilemma would be to introduce a MIDI controller with built-in mini-keyboard and trackball, so that you could place this combined controller device in front of your monitor screen and have all your sequencer controls in one place, rather than trying to juggle a separate controller, keyboard and mouse. They needn't involve any extra circuitry either — just a standard keyboard and trackball connected to ports so you can plug them directly into the usual sockets on your computer. Published in SOS November 2005

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

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

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Pro Tools Notes

In this article:

We Have Ignition Can It Be Magic? Service With A Smile Spend, Spend, Spend

Pro Tools Notes News & Tips Published in SOS November 2005 Print article : Close window

Technique : Pro Tools Notes

Once again, there's a bumper crop of news from Digiworld to report, including the long-awaited jump to version 7 of Pro Tools. Mike Thornton

There are two major news items from Digidesign this month. The first is the new M Box 2, which is reviewed in this issue. As described in that review, the interface now features MIDI I/O — apparently the number one request from existing M Box users — and lets you use two analogue inputs and two channels of S/PDIF digital input at the same time to achieve four simultaneous channels of audio input. Second, just before we went to press, Digidesign gave the world a sneak preview of Pro Tools 7, which should be the current version by the time you read this. Although their press release didn't go into much detail, it seems clear that some fairly major changes have been made. Perhaps the most fundamental is a new track type: Instrument tracks are, as the name suggests, designed "for improved integration with virtual instruments and MIDI sound modules", although it's not yet clear how this will work. There will also be real-time MIDI processing features, presumably similar to those found in Cubase SX and Sonar, while the ability to "group any combination of audio and/or MIDI regions together to quickly build alternate arrangements" suggests a feature along the lines of Cubase's Play Order Track. REX and Acid file formats are now supported, there is Tool Tips help, and the M-Powered version now supports a wider range of M Audio interfaces. See Digidesign's web site for more details.

We Have Ignition Digidesign have also announced the Pro Tools Ignition Pack, a new collection of free tools included with the M Box 2 and every Digidesign Pro Tools HD and LE

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system. The Ignition Pack includes updated versions of programs included in the previous free software bundle, such as Ableton Live Lite 4 Digidesign Edition, Propellerhead Reason Adapted 3 and IK Multimedia's Sampletank 2 SE, Amplitube LE and T-Racks EQ. Two new software packages have been added: Fxpansion's BFD Lite, which provides a drum studio with virtual acoustic drum kits and individual samples, and Celemony's Melodyne Uno Essential, which enables you to re-tune mono audio tracks and edit the timing of melodic lines. In addition there are range of extra 'bits', like a CD from Bunker 8 Digital Labs that has a collection of REX files, and the Pro Tools Method One instructional DVD, which covers Pro Tools system essentials from setting up Sessions and recording audio to editing MIDI, working with loops and plug-ins, automating mixes, and more. You can also promote your music around the world with a free one-year membership to Broadjam.com, an easy-to-use web service for independent musicians.

Can It Be Magic? I'm looking forward to trying out DUY's Magic EQ. This is an OS Xonly TDM plug-in that can 'rip' the EQ curve from one piece of music and apply it intelligently to another. DUY claim to have developed three new technologies with impressivesounding acronyms — OFIR (Optimal Frequency Impulse Renderer), Adaptative Spectral Matching (ASM) and Historical Audio Statistics (HAS) — and I'll be intrigued to see what Magic EQ can do.

Service With A Smile There is also another range of 'cs' updates for Pro Tools TDM and LE software. The main fix in this set of updates is that Digidesign have responded to customer pressure to restore the keyboard shortcuts for fine-tuning automation breakpoints to the pre-version 6.9 behaviour. For more information on this, go the Download section of the Digidesign web site, click on the link for a page showing details of all the 'cs' updates, and select the correct one for your system. Especially on the Mac side, there are a lot of different versions of the software depending whether you use TDM or LE and whether you are running Panther or Tiger, so take care selecting which update to download. Digidesign have also announced upgrades to a couple of the Digirack plug-ins. The version 6.9cs3 update of EQ III, for Pro Tools TDM, Pro Tools LE, Pro Tools M-Powered and Avid systems on Windows XP and Mac OS X, fixes a problem where the one-band plug-in would not hold its Q setting when used on mono or multi-channel tracks. The Mod Delay II 6.7cs1 plug-in update for Pro Tools TDM, Pro Tools LE, Pro Tools M-Powered and Avid systems on Windows XP and Mac

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Pro Tools Notes

OS X with Pro Tools 6.7 and higher fixes two problems. First, when tempo sync was enabled, the Mod Delay II plug-in would not follow tempo changes if the meter values were set to anything other than quarter-note resolution. Second, Mod Delay II would also set its delay time two times faster than the set tempo when tempo sync mode was used in eighth-note resolution.

Spend, Spend, Spend In the 'acquisitions and mergers' department, Digidesign have bought Wizoo Sound Design, the developers of virtual instruments, sample libraries, and realtime effects led by keyboardist, sound designer and author Peter Gorges, Steinberg co-founder Manfred Ruerup, and legendary film composer Hans Zimmer. Presumably, the intention is to integrate Wizoo's convolution and sample-playback technologies into the Pro Tools environment. Existing Wizoobranded products such as Wizooverb, Darbuka and Latigo will continue to be distributed worldwide through Digidesign's partners M Audio, while all future products will be branded and distributed by Digidesign. Published in SOS November 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

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Sonar Notes

In this article:

64 Bits Better? REX Support Speaking Of VST... V-Voice

Sonar Notes News & Tips Published in SOS November 2005 Print article : Close window

Technique : Sonar Notes

When the version number's most significant digit increments, you know something big is going to happen... Craig Anderton

Sonar's 'update odometer' has just flipped over from 4.0.3 to 5.0. Whatever happened to those '.1' and '.2' updates? Let's just say that what I've seen at a special preview of Sonar 5 justifies the jump. There's a lot to investigate — more than I can fit in one Sonar Notes — so I'll cover what I can and carry on next month.

64 Bits Better? Sonar 5 retains the Producer/Studio version division, and either package includes two versions of the program on the same distribution media, one with native 32-bit and one with native 64-bit operation. Based on what I've seen floating around the web, there are a lot of misconceptions about what this means, so first, let's address bits. Both new Sonar versions use a 64-bit data path from input to output, and all calculations are done in 64-bit floating point. Sonar isn't the first program to use a true 64-bit data path; iZotope's Ozone plug-in, for example, implemented this some time ago. However, it does not require a 64-bit operating system to work. It simply means that audio, regardless of its original resolution, is calculated with 64-bit floating point precision. This applies to all branches of the signal path — tracks, busses, effects and so on — not just the summing buss. For those who feel that 'something funny' happens when digital signals are summed, this should settle the issue. It's true that the lower the resolution, the greater the odds of round-off errors accumulating. This was very audible with 16bit systems, and some would argue that 32-bit floating-point and 48-bit fixed

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calculations also degraded the sound, albeit subtly. However, it's hard to imagine that argument being made for 64-bit floating-point systems. We're talking huge amounts of resolution, where any round-off errors are more of a theoretical than a practical reality. But if both the 32- and 64-bit versions use 64-bit maths, why bother upgrading to Windows x64 and a 64-bit capable processor? The answer lies in V-Voice puts Roland's Variphrase technology the ability to address 128GB of into a plug-in. physical RAM, and some performance gains. As mentioned in a previous Sonar Notes, the extra RAM gives advantages regarding streaming huge samples and recording into RAM, while the ability to use longer word lengths yields performance increases of 10 to 30 percent. However, there's a major stumbling block with a true 64-bit operating system: the plug-ins and other system elements must all be 64-bit as well. Or must they? Actually, Sonar 5 employs a clever technology called Bit Bridge, which allows the hosting of 32-bit VST plug-ins in a 64-bit environment. Yes, even "the most obscure independently developed shareware programs" (a quote from the Sonar press information) can run on your 64-bit computer. That's a pretty good thing, because the VST spec has not yet caught up with 64 bits. On the other hand, DX and DXi, being part of the Microsoft OS, are inherently ready for 64-bit operation. Incidentally, most new computers have 64-bit capable chips; they just haven't had anything to flex their muscles on yet. With the official release of Windows x64 and Sonar 5, the latest processors can be used to full advantage. The fact that you don't have to give up your older plug-ins is great. But note that Cakewalk have already started porting their plug-ins over to 64-bit operation, starting with the Sonitus line.

REX Support Sonar 5 offers REX file support, and its REX file player makes it easy to drag the accompanying MIDI sequence into the track view so that you can modify the sequence, transpose it, copy it and so on. We'll take a look at the new version's improvements to the MIDI side of the program next month.

Speaking Of VST... Sonar 5 now offers native VST support — no wrapper required. The VST file:///F|/SoS/SoS%2011-2005/sonarnotes.htm (2 of 3)10/19/2005 9:44:58 PM

Sonar Notes

configuration menu still looks very much like the old VST-DX adaptor, but what's going on 'under the hood' is quite different. Also, unlike other programs that take forever to boot up as they scan the VST plug-ins folder, Sonar 5 apparently has some kind of routine that re-scans only if it detects a change in the status of installed plug-ins. Very sensible.

V-Voice Another big Sonar 5 feature is the incorporation of Roland's Variphrase audio technology into a plug-in called V-Voice. This plug-in renders audio malleable in many ways. You can detect and then modify the pitch of entire sections of audio or selected notes, constrain audio to particular notes, vary the formant independently of pitch and amplitude, add vibrato (either manually or via an LFO) and create dynamics where none previously existed. You can also stretch pieces of audio temporally, in a similar way to Live's 'elastic audio' feature, by clicking on a section and dragging it left to shorten or right to lengthen. Best of all, the sound quality is remarkably free of nasty artifacts, even with significant amounts of stretching. It's primarily intended as an effect for voice and pitch correction, but V-Voice also works well with drums, lead guitar, bass and other monophonic instruments. It's suitable, too, for creative sound design with polyphonic instruments, and even program material. This is something you have to hear to believe, as it opens up both realistic and off-the-wall sonic options. I bet some people will upgrade just for this feature. Published in SOS November 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

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The Lost Art Of Sampling

In this article:

The Lost Art Of Sampling

Going Round In Part 4 Circles Published in SOS November 2005 Short Loops Tricks Of The Print article : Close window Trade Technique : Sampling Looped Loops & Stretched Time Throwing Out The Baby? The Stretching If you want to artificially extend your instrument samples, Solution or make entire backing tracks from one rhythmic snippet, Of Keys & Maps you'll need to know about looping and time-stretching. And Next Month

then there's keygrouping... This month, we explain these fundamental sampling processes, and more. Steve Howell

As promised, I'm going to look at sample looping this month. However, one problem associated with tackling this topic is that looping means different things to different people. As explained in the second part of this series, it originally meant finding a way to artificially extend a sample's length by playing a portion of it over and over again, back in the early days of sampling, when the hideously expensive nature of sampling RAM made taking full-length samples of instruments impossible. However, from the mid-'80s on, looping came to mean something rather different. By looping a short recording of percussion or drums so that it could be repeated to play over and over again, samplists learned how to easily create backing tracks, thus giving rise to the term 'drum loops' — short, pre-trimmed, repeatable sections of rhythm that could be used to build rhythm tracks fast. By the early '90s, vocals were being looped in the same way, and then people began sampling whole sections of existing tracks and building new compositions around them, particularly in the worlds of rap and R&B production. I'll say more about this later on this month, but first, let's look at instrument sample looping.

Going Round In Circles Conceptually, this type of looping is very simple — you just find a reasonably constant portion of the sound in your sampler, put loop points around it, and then when you play it

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back, the sound will play back in an infinitely sustainable form, forever repeating the constant-sounding section between the loop points until you take your finger off the trigger key. Like most things, of course, it's a lot harder in practice! Apart from the simplest electronically generated waveforms, most instrument sounds vary considerably in amplitude and harmonic/timbral content over the course of a note. On a quick Here's a very simple repeating waveform. It listen, therefore, while many sounds might should be easy to loop, but picking bad loop appear to have a reasonably constant-sounding start and end points (top) results in a looping sustain portion (ensemble strings, for example), waveform that clicks or glitches (above). if you analyse the sounds closely, you will usually find some complex sonic 'movement' going on during that sustain. You can see this for yourself very simply by looking at the amplitude changes in pretty much any sound in a software waveform editor (or on-screen on your hardware sampler) — finding sections that The same waveform looped properly, with its remain even close to constant in amplitude over loop points placed on zero crossings (top), loops without glitching (above). any period of time is very hard. As I mentioned last month while explaining the process of trimming samples, if you edit a sound such that there is a very great, instantaneous change in a sample's amplitude from before the edit point to after it, you'll hear a glitch or click in the edited sound, and this is equally true when looping sounds. And even if you can get the amplitudes to match at your loop points, there will often be harmonic changes happening over the course of a section that disrupt your attempts to make inaudible loops. Before you give up on the whole process as unworkable, take heart. While these difficulties do make sample looping something of a trial-and-error process, it's also true that the more you try to do it, the more you get used to what to look and listen for. With experience, you can often nail a decent loop almost immediately with fairly coarse loop settings, which can then be further refined with careful nudging of the loop start and/or end points. And successful looping isn't all down to experience, either — there are a few cunning techniques that can help...

Short Loops In the early days of sampling, when conserving sample memory and storage space was essential, it was common to store just the attack portion of a sample, and then set a very short loop of just a few milliseconds to convey the rest of the sound. These days, in modern samplers with generous memory complements, loops like this really are a thing of the past, but hardware manufacturers still use the technique in their synths and workstations. A typical example might be a bass guitar sample — the 'pluck' and initial decay would be heard, and then this would settle down into a static loop. This creates a totally unnatural sound, of course, but the original instrument's envelope and tonal movement can mostly be restored artificially using the sampler's envelope and filter functions (see diagram, right). It might sound horribly primitive and crude, but a sound like this can work surprisingly well when programmed by someone with skill, especially when the sound is used in a full

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arrangement. You might never know that it uses 92 Kilobytes of memory, not 92MB!

Economical sound design using very short loops. The top diagram shows the naturally decaying envelope of a sound. In the middle diagram, a very short loop has been set up to create the sustain portion of the sound, which works, but sounds very unnatural. However, the natural envelope of the sound could then be restored using the sampler's onboard amplitudeenveloping facilities, as in the lower diagram, thereby creating something close to the original sound, while taking up far less sample RAM.

Tricks Of The Trade Firstly, and as with trimming, explained last month, it's usually a good idea to set loop points on zero-crossing points, because there won't then be any amplitude changes at the edit points (see the diagrams on the first page of this article). This, at least, takes care of amplitude-related loop clicks. Many software wave editors and the editing facilities in some hardware samplers include a 'zero crossings only' function (although it's often known under another name). With this enabled, when you try to move your loop points, they will 'snap' to zero-crossing points only. And if your editor doesn't have such a function, you can still try to find zero crossings in the waveform by eye. Crossfading (sometimes known as crossfade looping) is another handy looping trick, although sometimes it creates its own problems. Once again, the concept is simple — instead of 'hard cutting' from the end of the loop section to the start at a fixed loop/edit point, you create a crossfade section instead, where the end part of the loop fades out, while the start portion of the loop fades in at the same time. In some cases, this can 'smooth over' a difficult loop point very nicely (the diagram at the top of this page should make this clearer). Some samplers also offer other possibilities such as forwards/ backwards looping. In theory, this is an ideal solution to looping awkward, evolving sound sources such as (say) ensemble strings or voices. When you play the sample it plays back normally, then plays the loop section, then plays the loop section backwards to its loop start point, then forwards again through the loop section, then backwards again... and so on (the diagram on the right should explain this more clearly). However, in practice, I have to say that I've never really found this option to be much help in looping sounds. In theory, you'd think it a perfect solution, but I rarely found it so. Some people seem to like it, though, so maybe it'll work for you — if you have access to such a function, it's certainly worth experimenting with. While all of these techniques can take care of differences in waveform amplitude across loop points, as mentioned earlier, the tiniest tonal or harmonic changes over the course of the sound can create audible problems when looping, even if the loop you make is free of clicks. If the pitch of (say) a string or guitar sound varies even slightly across your loop

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point, you'll hear a disconcerting pitch change as the sound loops, and even if you put in a crossfade, you'll hear a flanging or chorus-type sound as the two sections at different pitches overlap. Much worse is the problem of audible repetition. If a portion of a sound is cycling round and round as it is being sustained, our ears can pick up on this (even subconsciously) very quickly, and the effect can be as irritating and obvious as a clicky loop, and/or interfere with the rhythm of any piece of music you try to use the looped sound in! The top diagram shows a sample with bad loop points, which, when looped, results in the glitching, clicky waveform in the middle. However, applying a loop crossfade could 'smooth out' these clicks, as in the lower diagram.

For this reason, it is best to make loop sections as long as you can, to minimise the impression that the sample consists of an intro portion and then the same few milliseconds of sound repeated ad inifinitum (even though that's exactly what people used to do in the days when sample RAM was expensive — see the box on the previous page). These days, I usually take samples of around five seconds, and typically make loops of around four seconds — this way, the cycling is less repetitive and less obvious to the ear. In addition to using these techniques, you may find that your sampler or editing software has some functions that can make looping much easier. I've already mentioned zerocrossing finding functions, but there are others. Many devices or audio-editing software packages allow you to listen to your loop section cycling as you adjust the loop's start and end points, so you can hear how the loop will sound and can tweak it until it improves. Some hardware samplers will also display the waveform 'join' at the loop point, making it easier to match the two points up visually, and others have a 'Loop Lock' function which allows you to slide the entire loop around the sample in search of a better position. Some (most notably, those hardware samplers made by Ensoniq) allowed you to modulate loop position in real time. In the world of software, there are even dedicated programs designed to help create smooth-sounding loops, such as Antares' Infinity for Mac OS 9 and Beatcreator's Zero X Seamless Looper for Windows. If you're wondering why I didn't just mention these earlier and spare you all of the stuff about manual crossfading and zero crossings, there are good reasons. In much the same way as having a copy of Auto-Tune doesn't make you a vocalist, having a looping editor doesn't make you an experienced sound designer! Even if you make use of such applications, a basic knowledge of how to manually set loops is beneficial, so that you can set some rough loop points in advance according to your judgement and then let the applications' clever looping and crossfading algorithms work from your starting points. Simply setting arbitrary loop points and throwing the loop at the software is unlikely to yield good results. What's more, these software tools aren't always equally effective at dealing with all types of sounds (some are less good at dealing with sounds containing a lot of harmonic movement, for example), and so it can help to have

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some idea of what to do when the software doesn't quite give you the result you were hoping for. When I'm looping sounds, I'll sometimes set rough loop points using my experience, put the sound through a software looping tool, and then tweak the loop points again manually afterwards. It's good to have some idea of manual looping techniques, so that you're not simply stuck with what your looping software gives you.

Looped Loops & Stretched Time Once you have an understanding of good instrument looping and basic sample-trimming techniques (as described last month), then creating rhythmic loops — the other kind of looping mentioned at the start of this month's article — is a piece of cake. Here's a simple example, based around looping a small four-bar section of drumming to create a continuous rhythmic backing track. First, don't sample four bars — sample five! Then, in your sample editor, set the sample start to the first beat of bar one (taking care to put it on a zero-crossing point, as described last month, to avoid clicks). Now set the sample end to the first beat of the start of bar five (again, taking care to set it on a zero-crossing point). You now have a loop that, if repeated, will give you a rhythmic backing as long as you want it to be. If you know the tempo of the drumming you've looped, you can load the loop into a sequencer, put the sequencer at the same tempo as the loop, put triggers in to fire off the loop every four bars (thus ensuring the loop stays locked to the rest of the arrangement) and add other sync'ed material on top at the same tempo. If you don't know the tempo of the audio you've looped, your sequencer may have a 'tap tempo' function that will work it out for you if you tap along to the beat of the loop.

Looping forwards and backwards. The upper trace shows a sample with loop points set, and the lower trace shows the results of playing the sample back with a forwards/ backwards loop.

This is the basis for loop-based composition, but of course, your loop doesn't have to be of drums — anything with a rhythmic element will do, as Fatboy Slim (to name just one fan of this technique) has been fond of proving to people with looped vocals, looped piano, and so on. Now, that's fine if you are working with one sampled loop at a certain tempo — everything you add on top is simply recorded at the same tempo — but what happens if you want to alter the tempo of the piece as it grows, or you want to add another sampled loop at a different tempo? As soon as you alter the tempo of your backing track, the loop will no longer play back in time with it. You could slow down or speed up the playback of the loop, of course, but unless it's a totally unpitched loop you're working to, such as drums or percussion, the pitch of the loop would normally change too, which could mean that it will no longer fit musically with the rest of your backing (and even with unpitched drum and percussion sounds, if you change the speed by very much, the sounds soon become noticably slowed down or sped up). This was a problem for samplists in the '80s, until the problem was partially solved by the invention of time-stretching... file:///F|/SoS/SoS%2011-2005/lostscience.htm (5 of 9)10/19/2005 9:45:02 PM

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Time-stretching is a process which allows you to lengthen or shorten a sound without changing its pitch. Conceptually, it's not difficult — to make a sound longer, duplicate samples are inserted throughout the sound, and to make it shorter, samples are removed. Once again, in practice, it's not so easy, which is why its invention provided only a partial solution! To insert or remove samples, small crossfades have to be applied where the data is modified so that the transition is smooth, and certain decisions have to made (in software) with regard to where the samples are inserted/ removed. If those decisions are wrong, inappropriate data might be inserted, the best example being when stretching a drum phrase — if the percussive transient of one of the drums is duplicated/inserted, you will get a flam sound (in other words, it could sound as though the drum has been struck twice). Similarly, if the percussive transient is removed, the sound will be corrupted, with a loss of attack.

The effects of time-stretching. This is a sample before stretching.

After time-stretching by 200 percent. Samples have been duplicated and inserted, with crossfades to smooth things out.

Complex sounds which consist of many frequencies and harmonics also tend to timestretch badly, as what works well for one set of frequencies usually doesn't for another. Consequently, if you try to stretch a sample of a Time-compression in action — here the complete mix, you might find that the low sample is half its original length. To achieve this, samples are carefully removed (with frequencies stretch smoothly, while the mid and crossfading once again to smooth the high frequencies exhibit problems with flam-like transitions). sounds in percussive transients and other unnatural-sounding artefacts. Conversely, sometimes the high frequencies will sound acceptable while the bass end sounds 'wobbly', especially on sustained bass notes. This has been typically overcome in the past with a well-known scientific technique known as 'compromise'; sampler manufacturers frequently offered several different time-stretch 'presets' optimised for certain types of sound sources (percussive samples, sustained sounds, and so on). It's also possible to partially overcome these problems by splitting the sound into different frequency ranges and processing them separately, but that in itself can cause phase and other problems when the processed sections are recombined. A similar dodge was used to time-stretch stereo signals, by splitting the signal into Left and Right components, and processing them separately, but again, the phase problems this caused were often severe. Later attempts at stereo time-stretching analysed a mono sum of the the two channels and applied an overall stretch process to the left and right sides separately, which at least meant that both sides were processed similarly, but this was hardly foolproof either, as it suffered from the same problems besetting the first attempts to stretch any complete mix or harmonically complex sound. Of course, time-stretching recordings of individual instruments (or isolated recordings from a multitrack), particularly

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if they were harmonically simple ones, didn't suffer many of these problems, but that wasn't what many people wanted to do.

Throwing Out The Baby? In the top diagram, you can see a sample looping smoothly thanks to crossfade looping. For creative sound design purposes, you could discard the attack portion completely, and keep just the looping section, as in the middle diagram. You can then add back an envelope using your sampler's sound-shaping facilities, as shown in the lower diagram. This could create an ethereal sustaining pad sound with a slow fade-in, quite unlike the original sampled sound.

As explained last month, when you have created a successful loop, it's a good economy measure to discard all sample data after the loop's end — you're not ever going to hear it, and the data is just wasting memory and storage space. But in some cases, you might actually want to try removing the stuff before the loop start as well! This might fly in the face of conventional thinking, where the attack of a sound is the most important element that helps us identify instruments, but if you're not bothered about realism, and you're just trying to design an interesting sound, you can create some great washy pads by lopping off the attack section of the sound, and creating a new, artificial attack with your sampler's built-in envelope generators (see the diagram right for more on how this is done).

The Stretching Solution The solution arrived in 1994 from then-new Swedish software house Propellerhead, in the form of their Recycle software. Most sampler owners these days use this software, or at least files in its REX format, without being aware of how it works. It's one of those ideas that's so brilliantly simple that it can't fail to be a success — and Recycle was most certainly that. Basically, the software takes a sample (typically a drum phrase with percussive transients) and slices it up into separate beats/hits, making each hit a separate sample. Once the 'hits' comprising the loop are separate, it is possible to vary the distance between each of them, and hence change the tempo without changing pitch — if you want the loop to play back more slowly, you space the beats out more widely, and if you want it to play faster, you squeeze them more closely together. That would have been clever enough for most people, but Propellerhead added a further cunning twist — they also very cleverly generate a standard MIDI song file of the pattern/phrase that's been sliced. This MIDI song file can then be imported into a sequencer and its tempo can be varied there to match the tempo of the song being created. As regular users of REX files know, this makes it possible to take almost any sampled pattern and use it at any tempo in your music. Furthermore, you can rearrange the beats in your sequencer to create your own file:///F|/SoS/SoS%2011-2005/lostscience.htm (7 of 9)10/19/2005 9:45:02 PM

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patterns and variations, and, of course, they can be quantised. Not surprisingly, the concept underlying Recycle is now available in almost every sampler, software or hardware, even if it works in slightly different ways, and machines or programs that don't have the ability built in usually have a REX-file input option at least. The idea is inspired, simple and effective and allows another variation on the concept of 'looping'.

Of Keys & Maps Once you've taken some samples, edited, looped and tweaked them to your satisfaction, you need to assign them to MIDI notes so that they can be played back from a controller or a sequencer. Again, this should be simple to understand — if you've taken a sample of an organ playing a 'C#', you'll want to map that to the MIDI note corresponding to 'C#' in the correct octave. However, it gets more complicated when you've taken samples of a less-than-complete set of notes to save memory (as explained in this series two months ago). You'll then want some of the samples, on receipt of the appropriate MIDI notes, to be played back untransposed, while others are shifted a few semitones to provide the complete range of notes across an entire keyboard. There's also the issue of what to do with drum sounds (or the loops you've carefully edited to provide backings, as described on the previous page). Typically, you'll want each MIDI note to trigger a different drum sound or loop, so that you have a spread of drums or loops available to you across the span of a MIDI keyboard, rather than (somewhat less usefully) a single drum sound or loop that pitches up and down over the range of the keyboard.

Mapping single samples of G notes in every octave across a full MIDI keyboard, as you might do it in a modern software sampler or hardware sampler's editing package. First, the G-note samples are dragged and dropped on every 'G'.

The range of the G1 sample is extended down to C1 (usually by simply dragging 'handles' at the edge of the G1 Keygroup icon).

Extending the G1 sample up to B1 in the same way.

Extending the G2 sample down to C2.

Extending the G2 sample up to B2.

So that a sampler knows what to do when it The keyboard after all the samples' key receives a given trigger note, it can hold in its spans have been adjusted. memory a 'map' of instructions telling it which samples to trigger on receipt of which notes, whether to transpose them before outputting them, and so on. As with many sampling features I've discussed in this series, these instructions are set up differently from sampler to sampler, and they're also called different things on different samplers (some call them 'zones', others 'groups', for example). I've always preferred Akai's nomenclature, Keygroups, because it seemed logical to me — a Keygroup refers, after all, to a group of keys — so that's the term I'll be using here.

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Setting up Keygroups on hardware samplers, with their tiny displays, used to be a horribly fiddly process, but in these days of software samplers and software editors for hardware, it's much less of a problem, as you nearly always have access to a graphical overview of the sample mapping. Moreover, whereas in the old days you had to select the samples you wanted to assign to Keygroups from a cumbersome, scrolling list of all your samples, these days you can drag and drop samples directly on to a graphical representation of the keyboard, and define the spans over which you want the samples to be stretched by simply dragging handles on the various Keygroup graphics (see the diagram above for a representation of how this could be done).

Next Month In next month's instalment, I'll conclude looking at Keygroups and mapping (as well as explaining drum mapping), and start looking at some of the great things you can do using the built-in synth engine hidden away in modern samplers. Published in SOS November 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

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Using Sonar's Dreamstation Soft Synth

In this article:

The Architecture Two Ways To Save Programming The Dreamstation Inserting The Dreamstation It's A Percussion Synth! Tuned White Noise Hard Sync Dreamstation Quick Tips String Synthesis FM Synthesis The Secret Weapon: Processing Further Information Methods Of Automation

Using Sonar's Dreamstation Soft Synth Technique Published in SOS November 2005 Print article : Close window

Technique : Sonar Notes

This modest little soft synth has been bundled with Sonar since version 1.0 and is very kind to your CPU resources, yet many Sonar users are still unaware of just how much you can squeeze out of it. We present some evidence. Craig Anderton

The Dreamstation DXi2, a simple soft synth that was bundled with the original version of Sonar and keeps reappearing in subsequent versions, doesn't get a lot of respect. Compared to other soft synths, its sound is... well, let's just say it's not a Minimoog. But it does far more than most people realise, and with the right tweaking it can make some very satisfying noises — as we'll find out.

The Architecture The Dreamstation is a 'virtual analogue' synthesizer with three multi-waveform oscillators and the following features: Oscillator 2 can work with Oscillator 1 to provide hard sync, ring modulation, and frequency modulation (FM). Oscillator 1 offers pulse width modulation, but the others do not. The amplifier has a dedicated ADSR (Attack, Decay, Sustain, Release) envelope, and a Gain control that crossfades between velocity control only and file:///F|/SoS/SoS%2011-2005/sonartech.htm (1 of 9)10/19/2005 9:45:06 PM

Using Sonar's Dreamstation Soft Synth

envelope control only. The filter (which also has a dedicated ADSR envelope) offers five modes: 12dB/octave low-pass, 24dB/octave low-pass, band-pass, high-pass and formant. However, in this instance, Gain controls the envelope level only and does not affect velocity. Unfortunately, there's no way to tie velocity to the filter cutoff. A User Envelope can control FM amount, Oscillator 1 pulse width, Oscillator 1 frequency, Oscillator 2 frequency, Oscillator 2 level or Filter cutoff. Bass patch.

The LFO has sine, square, sawtooth or random output and can feed Oscillator 1, Oscillator 2, Filter, or Pulse Width (one at a time). There are separate Depth and Rate controls. An additional Vibrato section offers Delay, Depth, and Rate. Other controls include Portamento time, Distortion, Output Level, Pan, Tune, and Polyphony (up to 16 voices). An additional option, Keyoff, when set to non-zero values, turns a note on for a specific period of time and is intended for percussive sounds.

Two Ways To Save As you check out the following patches, I hope you'll want to save them. There are two ways to do this: into a folder, where you'll probably save all your patches; or into the list of presets, which is intended to be more of a 'greatest hits' collection of sounds. To save to a folder of presets, click on the Save button, then navigate to the desired folder. If you go Programs / Files / Cakewalk / Shared DXI / Audio Simulation, you'll find the Dreamstation DXi2 folder, which is as good a place as any to save your sounds. Files are saved with a .DSI suffix. To save into the presets list, just type the patch name into the presets field and click on the floppydisk button. To delete a preset, click on the X button.

Programming The Dreamstation Each of the following patches highlights a particular aspect of the Dreamstation. As you program the patch parameter values, remember to turn on any elements that are needed. For example, if there are parameter settings for the User file:///F|/SoS/SoS%2011-2005/sonartech.htm (2 of 9)10/19/2005 9:45:06 PM

Using Sonar's Dreamstation Soft Synth

Envelope, make sure the User Envelope is turned on. To help make things clearer, sections that aren't used are greyed out in the accompanying screen shots. Also note that when references are given to calling up patches, these are the ones included with the Dreamstation. You call them up from the preset field. Let's begin by building a powerful bass patch, just to prove that it's possible. This fat sound uses all three oscillators and the filter. 1. Call up patch 03-Sawtooth. Play a few keys; not very impressive, is it? 2. Click on Oscillator 2's Off button to turn it on, and select its Sawtooth waveform. 3. Adjust Oscillator 2's fine-tuning to -10 percent and you'll hear that distinctive 'beating oscillators' sound. Turn the Oscillator 2 volume control to 80 percent. 'Analogue' snare patch. 4. Now turn on Oscillator 3, select Sawtooth for the waveform and set the fine-tuning to +10 percent. Turn its volume control to 80 percent. 5. Let's rough up the sound a bit. Click on Oscillator 2's Ring button. 6. The Amplifier Gain control is interesting. When it is set to 0 (counterclockwise), MIDI velocity affects gain and the envelope has no effect. When it's fully clockwise, MIDI velocity has no effect and the envelope controls amplitude. Settings in between give a combination of envelope and amplitude control. As we're aiming for a bass sound and we want a reasonably constant level, set gain to 50 percent. 7. For a fairly punchy effect, set the Amplifier's A parameter to 0, D to 80 percent, R to 20 percent and S to 80 percent, so that there's some level if you decide to hold a key down. 8. Now to get rid of that buzz and give the sound some depth. Turn the filter On. I find that adding a bit of resonance really helps, even if you don't want a resonant sound, so start out with a 15 percent setting. 9. Click on LP2 for a 24dB/octave filter response (LP1 is 12dB/octave). 10. Set Cutoff to 35 percent, A to 0, D to 60 percent, S to 20 percent, R to 20 percent, and Gain for the desired amount of high-end kick. Try 42 percent for now. 11. That's better, but we could give the sound more animation with a very subtle vibrato. Set Delay to 0, Depth to 5 percent and Rate to 30 percent. 12. Finally, if you're going to be playing a single-note bass it might be as well to set Polyphony to 01. Sounds pretty good! For a dirtier sound, turn up the Distortion control in the Misc section.

Inserting The Dreamstation When you go Insert / DXi Synth / Dreamstation DXi2, it doesn't matter whether you choose 'First Synth Output' or 'All Synth Outputs'. The Dreamstation is not a device with multiple outs, so you'll end up with a stereo output no matter what.

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Using Sonar's Dreamstation Soft Synth

It's A Percussion Synth! The Dreamstation makes a fine 'analogue' percussion synth. This electro-snare sound uses FM and modulates Oscillator 2 with a super-fast transient created by the User Envelope. Load the preset Effect 04-Wind and make the following settings: Oscillator 2: Waveform, Sine; Tune, 7 percent; Fine Tune, 0 percent; Kbd Track, Off; FM, 100 percent; Sync and Ring, Off; Volume, 100 percent. Amplifier: A, S, Gain, 0; D and R, 23 percent.

Electro kick-drum patch.

User Envelope: Dest, 02; A, D, S, R, 0; Gain, 20 percent. Turn off Oscillator 3, Filter and LFO. Tuned white noise patch.

Set all Vibrato and Misc controls to 0. Distortion makes a huge improvement in the sound of the following hard electrokick drum. Keyboard tracking is turned off so that you'll hear the same pitch no matter where you play on the keyboard. Start with preset 01-Sine. Oscillator 1: Tune, -37; Kbd Trk, Off. Amplifier: A and S, 0; D, 18 percent; R, 8 percent; Gain, 37 percent. User Envelope: Destination, 01; A, S, R, 0; D, 2 percent; Gain, 50 percent. Misc: Set Distortion at 55 percent.

Tuned White Noise This ghostly, evocative sound makes good use of white noise but also shows how the 12dB/octave low-pass filter mode can come very close to self-oscillation. Start by calling up the preset Effect 04-Wind. Amplifier: A, S, Gain, 0; D and R, 66 percent.

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Using Sonar's Dreamstation Soft Synth

Filter: Cutoff, 2 percent; Resonance, 100 percent; Mode, LP1; Kbd Track, On. Set all Envelope parameters to 0. Turn the LFO off. You could alternatively choose BP (band-pass) filtering, for a broader, less resonant sound. Also note that you'll need to play this patch in the upper range of the keyboard.

Hard Sync The Dreamstation does decent hard-sync sounds (i.e Oscillator 1's frequency is continuously reset by Oscillator 2's pitch) in the lower keyboard ranges, but it's important to choose the right waveforms and other options. This hard sync patch is fairly complex, but it does a good job of creating sync effects. Start by calling up preset 03-Sawtooth and edit the parameters as follows: Oscillator 1: Waveform, Pulse; Tune, 80 percent; Pulse width, 35 percent. Oscillator 2: Waveform, Sawtooth; Sync, On; Volume, 0 percent.

Hard sync patch.

Oscillator 3: Waveform, Sawtooth; Fine, +10 percent. Amplifier: Adjust for the desired dynamics. Try A and R, 0 percent; D, S, Gain, 60 percent.

String synth patch.

Filter: Adjust to taste, starting with LP2, On; Cutoff, 43 percent; D and S, 60 percent; A and R, 0 percent; Gain, 45 percent. User Envelope: Dest, 01; A, S, R, 0 percent; D, 50 percent; Gain, 20 percent. For the best results, add some tempo-sync'd delay, and wiggle those pitch-bend and mod wheels. You might also want to add some gentle high-frequency roll-off with EQ.

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Using Sonar's Dreamstation Soft Synth

Dreamstation Quick Tips The Dreamstation was designed during the days when CPUs weren't as powerful as they are now, so it's possible to enable and disable oscillators and filters to save CPU power. But even with everything turned on, by today's standards the Dreamstation is very CPU-friendly. You can thus stack multiple instances without worrying too much about CPU load. Here's a few things to bear in mind when you start to use the Dreamstation in earnest: You have to turn on Oscillator 2 before you can turn on Oscillator 3. If Oscillators 2 and 3 are on and you turn off Oscillator 2, Oscillator 3 will turn off too. Keyboard tracking cannot be turned off for Oscillator 3. If you look in Sonar's online help, you won't find anything under 'Dreamstation', nor will you find context-sensitive help if you hit F1. Instead, right-click anywhere on the Dreamstation itself to call up its own Help menu. Click on 'Clr', toward the upper right, to wipe all existing settings and load a default preset.

String Synthesis The Dreamstation can do an excellent imitation of old string synthesizers. This patch demonstrates how modulating pulse-width with two different modulation sources, then adding vibrato to all three oscillators, produces a big sound with lots of movement. This is another involved patch: the only unused sections are the User Envelope and Misc controls. Start with Preset 05-Pulse. Oscillator 1: Waveform, Pulse: Tune, 0 percent; Kbd Trk, On; Pulse width, 26 percent; PWM, 13 percent. Oscillator 2: Waveform, Sawtooth; Tune, 0 percent; Fine, +7percent; Kbd Trk, On; FM, 0 percent; Sync and Ring, Off; Volume, 100 percent. Oscillator 3: Waveform, Sawtooth; Tune, 0 percent; Fine, -7 percent; Volume, 100 percent. Amplifier: A, D, R, 65 percent; S, 75 percent; Gain, 41 percent.

FM brass patch.

Filter: Cutoff, 66 percent; Resonance, 8 percent; Mode, BP; Kbd Trk, Off; A, 35 percent; D, 0 percent; S, 100 percent; R, 65 percent; Gain, 0 percent. LFO: Dest, PW; Wave, Sine; Depth and Rate, 50 percent.

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Using Sonar's Dreamstation Soft Synth

Vibrato: Delay, 5 percent; Depth, 6 percent; Rate, 37 percent. Set all Misc controls to 0 percent. Polyphony: At least eight voices so that decays don't steal notes. A little post-processing with reverb won't hurt; the Sonitus reverb's 'Cathedral' preset is a good choice.

FM Synthesis This shows how the Dreamstation's FM synthesis can produce brass-like 'filtered' attacks, but without the use of conventional filtering. It won't put normal FM synths out to pasture, but it's another facet of the Dreamstation. Click the Clr button to create a default patch. Osc 2 Amplifier: A, 47 percent; D, 0 percent; S, 100 percent; R, 20 percent; Gain, 100 percent. User Envelope: Destination, FM; A, D, S, 62 percent; R, 20 percent; Gain, 32 percent.

The value drop-down menu not only lets you select an envelope parameter, but also provides a useful reference for which MIDI controllers match particular Dreamstation parameters. For example, Controller 22 controls the Oscillator 2 FM Amount parameter.

Vibrato: Delay, 33 percent; Depth, 3 percent; Rate, 37 percent. Misc: Portamento, 1 percent. Polyphony: Greater than 01. For expressiveness, try increasing Oscillator 2's FM Level parameter. You can also produce very different sounds by varying the Amplifier and User Envelope Attack controls.

The Secret Weapon: Processing Now that you have some sounds, you'll find that following them with a processor

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Using Sonar's Dreamstation Soft Synth

or two can add another dimension and still not drain too much CPU power. I like the Amp Simulation effect for a really nasty, industrial sound, while modulation (chorusing, say) adds space. As an example, with the 'String Synthesizer' patch, insert the Sonitus Modulator into the Dreamstation audio path, and call up the 'Slow Ensemble' preset. Set the Mix parameter to about 35 percent and the result wiill be a more spacious, spread-out sound. EQ can also help. In the case of bass patches, I usually roll off about 4dB of highs with a high shelf starting at about 5-6 kHz, cut mids by a couple of dBs at 500Hz and add some low-frequency boost with a low shelf at about 150Hz.

Further Information Check out the Dreamstation tip in Sonar Notes January 2004 (available at www. soundonsound.com). This describes layering two instances, one set for no velocity to provide a constant sound, the other set for considerable amounts of velocity so that playing harder brings in another sound. The end result is a much more dynamic effect. Also see 'Tweaking the Dreamstation' in Sonar Notes September 2004, which covers using the FM function and the Tape Sim processor.

Methods Of Automation The Dreamstation offers more automation options than most DXi devices. Envelopes Right-click on the MIDI track driving the Dreamstation and go Envelopes / Create Track Envelope / MIDI. The MIDI Envelope dialogue box appears, with three drop-down menus (see screen above). For 'Type', choose 'Control' and you'll be presented with a list of all automatable parameters, along with their matching controller numbers. External MIDI Controllers (This includes fader boxes and so on). Put the MIDI track driving the Dreamstation into Record mode, then move the hardware knob or slider whose controller number corresponds to the Dreamstation parameter you want to control (or move several at once, for that matter). You can view the list of automatable parameters and their controller numbers via the 'Envelopes' procedure described above. Front-panel Control Motions As when using external MIDI controllers, put the MIDI track in record mode and file:///F|/SoS/SoS%2011-2005/sonartech.htm (8 of 9)10/19/2005 9:45:06 PM

Using Sonar's Dreamstation Soft Synth

simply operate the Dreamstation controls. These motions will be recorded as MIDI continuous controller values. When you create a MIDI continuous controller message (as with the last two methods mentioned above), don't forget that these can be converted into envelopes using Sonar's 'Convert MIDI to Shapes' function. To do this, in the Piano Roll view use the Select tool to drag a rectangle around the controller data you want to convert into a track envelope. Then go Edit / Convert MIDI to Shapes. A 'Convert MIDI to Shapes' dialogue box appears. Enter the type of controller, controller number and channel of the selected data, then click 'OK'. This creates an envelope and simultaneously deletes the MIDI controller data. Published in SOS November 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

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Using Hardware Controllers With Logic

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Using Hardware Controllers With Logic

Automation Quick Access Hands-on Control Of Multiple Technique Published in SOS November 2005 Parameters Using Contour Designs Print article : Close window Shuttle Pro Technique : Logic Notes Using Hyper Edit With Automation Automation & Hyper Draw

Not everyone can afford to invest in a dedicated highspec hardware controller such as Logic Control. However, you can use even the most humble of MIDI devices to control your mixer, software synths, and plug-ins instead. Stephen Bennett

Although I do most of my automation in Logic using the mouse and Key Commands, there are times when only the twiddle of a knob (or a slider) will do. While I'm happy to adjust levels, pan sends, and effect plug-in parameters with the mouse, it's always easier to control the virtual knobs on a virtual instrument with their real-world counterparts on some kind of controller. This is particularly true of virtual analogue synthesizers, where I'll probably want to automate the synthesizers' parameters live as the track progresses. Doing this in real time and with real knobs is the best way to achieve the feel (and occasional serendipitous accident) that you get if you use a 'proper' analogue synthesizer. Version 7 of Logic has made some great strides in making this control easier to achieve, even without dedicated control hardware such as Logic Control.

Automation Quick Access I use a Alesis Photon XP controller while automating mixes, but the techniques described in this article are applicable to any controller keyboard or MIDI fader box. Occasionally, I'll want to use a fader to input volume or pan automation into Logic. It can be quite efficient to use a hardware fader to create rough automation lines around vocals, dynamic volume curves in sustained string or trumpet parts, or fades at the end of a Song. I usually use my master keyboard's modulation wheel for this, or one of the sliders on my Roland XP50, though any control that outputs MIDI controller data can be used. The way you set it up is via Logic's Automation Quick Access feature, which is found in the Track Automation

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Using Hardware Controllers With Logic

submenu of the main Options menu. The first thing to do is choose which automation parameter you'll be adjusting from the Arrange page. Make sure Track Automation is switched on in the Arrange page's View menu, and choose an automatable parameter from the Track Instrument — in this case it's Volume. If you are selecting Automation Quick Access for the first time, Logic will ask you if you want assign the controller. Click on Assign and the Automation Quick Access window will open. If you move the hardware control you want to use up and down, Logic will tell you that it has assigned it to your chosen automation parameter. Click on Done and close the window. You can now use the control to adjust Volume automation. If the hardware controller you are using is usually assigned to something else (say modulation) you can turn Automation Quick Access on and off using the Toggle Automation Quick Access Key Command.

Hands-on Control Of Multiple Parameters While setting up a single hardware controller rapidly like this can be useful when automating Volume or Pan parameters, when you want to use a hardware controller with a virtual instrument you'll probably need several controllers to be assigned for use at the same time. My Photon XP has 10 controller knobs, for example, so here's how I set it up to control multiple parameters in Logic's own ES1 soft synth. First insert ES1 on an Instrument Track and open the Controller Assignments window from either the Track Automation submenu of the Options menu or the Control Surfaces submenu of the Logic Pro Preferences menu. Create a new Zone from the '+' button underneath the Zone column, naming it something useful, and click on the Learn Mode button. Use the mouse to move a parameter on ES1, in this case Cutoff, and then move the hardware control you wish to control it with. You should see the assignment listed in the Control/ Parameter column. Repeat this process until all the ES1 parameters you wish to control are assigned to the required hardware controllers. Now, if you set ES1's automation mode to Latch, you can record the movements of the hardware controllers, and thus ES1's parameters. If you didn't press Record, you can use the Capture Last Take Key Command to record the performance after the event. Of course, you can always record the notes first and then the parameter manipulations on a second pass. You can continue to assign your hardware controller box to other virtual instruments or plug-ins in just the same way, creating each set of assignments in a new Zone. I used this controller assignment system recently at a live gig, using Apple's AULooper plug-in to replace a hardware echo unit I'd previously been using. I had a microphone plugged into the Photon via a small mixing desk, and file:///F|/SoS/SoS%2011-2005/logictech.htm (2 of 4)10/19/2005 9:45:10 PM

Using Hardware Controllers With Logic

AULooper inserted into an Input Audio object through which I'd routed the microphone input. With a buffer size of 128 samples I could easily play the flute live through AULooper and set up complex loops and rhythms all adjustable by the Photon's knobs.

Using Contour Designs Shuttle Pro Like many Logic users, I also work with Apple's Final Cut Pro video-editing software. To use this more effectively I've bought Contour Design's Shuttle Pro multi-button jog-wheel control surface. As this works by emulating the Mac's keyboard (thus allowing it to control Final Cut Pro via keyboard shortcuts) I thought I'd try using it with Logic in a serious mixing session. Reassigning my Logic Key Commands allowed me to use the scrub wheel to move rapidly in a horizontal direction around Logic's Arrange and editing windows. I could also Play, Stop, and change automation modes directly from the Shuttle Pro. It's a different kind of hardware control to the standard MIDI-based system, but I've found it useful during both automation and recording, so you should consider giving it a try if it might suit the way you work.

Using Hyper Edit With Automation While real-time knob twiddling and on-screen mouse dragging are rapid ways to automate in Logic, I often need precise control of the position and value of automation events. For this, I like to use Logic's Hyper Edit window. Logic's automation works with Fader messages, which are similar to MIDI controllers, but are internal to Logic. Each on-screen parameter of an effect plug-in or virtual instrument has a specific Fader message associated with it, and you can generate these outside of Logic's automation system using the Hyper Edit window. The Hyper Edit window is especially useful for generating events based on a grid. Here's an example. Suppose you wanted to change the Colour parameter of the Phaser plug-in every eighth note of every bar in an exact stepped way. You could do this using the Arrange window's automation facilities, but it'd be difficult to get it exactly right with the mouse. Using Hyper Edit, you can draw these fader events on a preset grid. To do this you create or select a region, open the Hyper Edit window, and Create Hyper Set from the Hyper menu. After setting the Status field in the Parameters to Fader, you can then enter the Channel value (which represents the slot the plug-in is inserted on) and the '-1-' value (which defines which plug-in parameter you want to control). In the screenshot I've set up the Fader events to control Phaser's Colour parameter (a '-1-' value of three).

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After inserting Fader messages using the Hyper Edit window's tools, you can further edit the Fader message data in the Event List. This allows you to perform all the usual fine editing procedures, including quantisation. So the question you'll probably be asking is 'How do you know which Fader message values control which plug-in parameters?' Well, you can insert your plug-in on an Audio object in the Controlling the Phaser plug-in using Fader Environment and cable it to a Monitor messages created in the Hyper Edit window. object, whereupon adjusting the plugin's on-screen parameters should display the Fader messages. However, this isn't working for me since I installed Logic version 7.x. The good news, though, is that you can also determine the Fader message values used by a plug-in for each of its values by recording the automation data in the Arrange window in the usual way and then using the Automation Event Edit Key Command to open the Automation Event list, from which you can read off the message values.

Automation & Hyper Draw I have a lot of older Logic Songs which use the earlier Hyper Draw system for automation. Hyper Draw is a MIDI-based system, where curves are drawn directly onto the regions — hence it is also often referred to as region-based automation. You can convert the older Hyper Draw data to the newer track-based automation by selecting the regions containing Hyper Draw data and using the Move Current Region Data To Track Automation function from the main Options menu's Track Automation submenu. There's also a global Move All Region Control Data To Track Automation menu item. If you are using a system like Korg's Legacy hardware MS20 controller, which outputs MIDI controller data, you can record your knob-twiddling then convert it into track-based automation using the menu items listed above. This will save you having to define a controller set within Logic especially for the MS20 using the controller assignment functions. Published in SOS November 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

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Using V-Racks In Digital Performer 4.6

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Rack Attack The Future For V-Racks? Getting Started The Alternative To V-Racks Advanced Racking Loading & Saving Master Fader Mania V-Racks: The Downside

Using V-Racks In Digital Performer 4.6 Technique Published in SOS November 2005 Print article : Close window

Technique : Digital Performer Notes

One of the most exciting and unexpected new features unveiled in Digital Performer 4.6 is V-Racks, and, like many of the best things in life, it's something you probably didn't realise you needed until it came along... Robin Bigwood

You can work with DP for many years and not realise it, but a Digital Performer project can incorporate multiple sequences. This capability isn't of much importance if you only ever use the program as a glorified multitrack recorder, or if you stick to simple, short projects, but being able to work with multiple sequences can be A typical V-Rack setup, containing a MOTU useful in lots of ways. For example, if Symphonic Instrument, Altiverb 5 and some you're composing music for film or post-EQ on an Aux track, and a Master Fader track equipped with two Waves stage and need to develop several processors. When hosted by a V-Rack, all different versions of the same cue these goodies become available to every simultaneously, you can do so by single sequence in your project. keeping each in its own sequence. That way, they're all easily accessible, and they can each have their own timeline and separate editing windows. As another example, for location recording work, and the subsequent editing, you can more easily organise your sessions by recording into separate sequences. This ensures that you're less likely to get muddled when dealing with hundreds (or even thousands) of soundbites. To give yet another example, mastering an album's worth of music can work well when you give each album track its own sequence. All the audio files are kept together within one project folder, but you can deal with each as a completely separate entity. Finally, project development, and even structural experimentation, can be made easier. Individual sequences can be used to 'freeze' a project in a particular state of development, or you can file:///F|/SoS/SoS%2011-2005/performertech.htm (1 of 8)10/19/2005 9:45:16 PM

Using V-Racks In Digital Performer 4.6

use a sequence as a 'building block' in a larger musical structure by utilising DP's Song window. Incidentally, this way of looking at sequences accounts for why they are also sometimes referred to as Chunks in DP. The fundamental drawback associated with some of these approaches comes to light when you have multiple sequences all using their own audio plug-ins and virtual instruments. Switching between sequences can cause severe delays as samples and patches are loaded, and you can even run into annoying instabilities and crashes as multiple plug-ins instantiate simultaneously. Maintaining multiple instances of instruments and plug-ins in different sequences can also sap your processor and memory resources. Trying to build songs from sequences that just drive external MIDI devices is all very well — that was how DP was originally designed — but when there are heavyweight plug-ins and software instruments involved, forget it! You'll also quickly discover how patient you are if you have many related sequences all incorporating the same complement of effects, because if you discover you want to make an effects change in all of them, you end up having to dial it in many times.

The 'Add V-Rack' command can be found almost everywhere you'd go to create a new sequence, but clicking the Mixing Board's 'V' button is quicker still if you don't already have a V-Rack in your project.

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Using V-Racks In Digital Performer 4.6

Rack Attack V-Racks are MOTU's response to these not-inconsiderable problems, and they're a valuable new tool in a DP user's kit. They work literally like 'racks' of effects processors or virtual instruments (or both), running in DP alongside your usual sequences, and available to them for various useful purposes. While they don't allow you to do anything you couldn't previously achieve in other ways, they make life much easier in some situations, particularly if you're involved in certain kinds of composing work or in mastering, or if you're a heavy user of soft synths and samplers. V-Racks exist as a new type of sequence (or Chunk) that can run in parallel with your 'normal' sequences, hosting instruments, plug-ins and master faders that the sequences can then utilise. If you switch to a new sequence, the V-Rack remains in place and active. Using this new facility, film composers (for example) can still have their multiple versions of a cue in different sequences, but would be able to set up their virtual instruments in a single V-Rack. There's no longer a need to set up the same instruments for each sequence, saving time, effort and computer resources. Similarly, the 'song structure' approach is once more feasible in DP, because each component sequence need only have simple MIDI and audio tracks, with all virtual instruments and effects processing 'shopped out' to the V-Rack. Mastering tasks can also benefit, for similar reasons, and there are yet more possibilities, such as setting up a V-Rack to handle signal routing to and from external hardware processors, keeping your actual working sequence much cleaner and easier to deal with. The more you work with V-Racks, the more uses you find for them.

The Future For V-Racks? When I first saw the little 'V' button in the Mixing Board, during a demo of forthcoming features in DP that took place earlier this year, before v4.6 was released, I suspected it had something to do with a distributed processing scheme, along the lines of Logic's node system. I wasn't disappointed to find out that it hadn't, but I'm still wondering if V-Racks have a future that encompasses some sort of distributed processing. The fact that V-Rack tracks aren't bound to timelines or disk-based audio recording and playback would seem to make them ideal candidates for 'contracting out' to additional CPUs, perhaps via Firewire or Ethernet. This is pure speculation, but I've got my fingers crossed...

Getting Started V-Racks are created by choosing 'Add V-Rack' from almost anywhere you'd normally go to create a new sequence in DP. This includes the following menus (see the screens above):

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Using V-Racks In Digital Performer 4.6

The Sequences sub-menu in the Project menu. The Sequence menu in the top left of the Tracks Overview. The mini-menu in the Sequence Editor or the Control Panel. Perhaps easiest of all, though, is to click the new 'V' title-bar button in the Mixing Board. If your project doesn't already have a V-Rack, one will be The 'V' button in DP 4.6's effects windows created for you, and if it does the switches them between viewing plug-ins in Mixing Board switches its layout to your sequence and in a V-Rack. It doesn't display it. A fundamental fact about Vtransfer a sequence's plug-in to a V-Rack, or Racks is that the Mixing Board is about vice versa. the only place you can actually interact with them, so this is a good technique to learn! If you're exploring V-Racks for the first time, try the following, which walks through all the main operational steps: 1. Start by creating a new DP project, so that you've got a blank canvas to work with. 2. Create a new sequence, so that there are two in total. You can do this using the Sequence menu in the Tracks Overview. Using the same menu, you can also rename the sequences, perhaps to 'A' and 'B'. 3. Now create your V-Rack. Hit shift-M to open the Mixing Board, and click the 'V' button in the title bar. 4. You'll see that, by default, a single Instrument track is created, so you might as well place a soft synth or sampler on this by choosing it in the uppermost insert slot, before making sure that the track has a valid audio output. 5. Now create an Aux track by choosing Add Track / Aux Track from the Project menu. Place an audio plug-in on this (maybe a reverb or delay), set its input to be a buss or buss pair, and the output as your main output pair. 6. Create a Master Fader track, set up a limiter (such as the Masterworks Limiter) on it, and configure its output as your main output pair. 7. You can now switch back to your 'normal' sequence by clicking the Mixing Board's 'V' again. You can no longer see your V-Rack, but it remains active. Now, if you look in a MIDI output pop-up in the Tracks Overview, for example, you'll see your V-Rack's virtual instrument in the list. Similarly, any audio directed

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to the buss or buss pair you chose for your VRack's Aux track will end up passing through your 'rack' reverb or delay. And all tracks routed to your main output pair will automatically take a trip through your V-Rack's Masterworks Limiter. Here's the final test: switch to your other sequence and you'll notice that all the same options are available. The V-Rack remains in place, nothing has to reload, and you're completely free to continue working with no delays. As well as being able to view your V-Rack using the 'V' button in the Mixing Board, you can also access V-Rack plug-ins using the 'V' button in plug-in The Mixing Board's hidden pop-up menu, accessed by windows. Initially, it might look as though clicking Apple-clicking the right-hand this button would somehow 'transfer' a plug-in from side of its title bar, is a sequence to a V-Rack but that's not actually what fantastically useful for it does. Instead, it simply switches a single effect switching between multiple Vwindow from 'sequence view' to 'V-Rack view', and Racks. you can then use the Track and Insert pop-up menus to directly access any plug-in either in the sequence or the V-Rack. This is a great facility that allows you to control important V-Rack instruments or plugins without always having to visit the Mixing Board first.

The Alternative To V-Racks Although V-Racks have plenty to offer in streamlining and speeding workflow, they're not a panacea for all ills. Indeed, in many projects you'll have no need for them at all. Two of their key benefits — simplifying signal routing and offering loading and saving options for plug-in and instrument setups — can be recreated in other ways. For example, DP's Track Grouping and Mixing Board Layout features can go a very long way towards keeping the DP environment streamlined and easy to work with. As for building up banks of plug-in chains and settings, DP has been able to do this for years, courtesy of the Clippings feature. As is often the way with DP, MOTU give users multiple ways to achieve very similar outcomes, and the smart approach is always to know about all of them, and mix and match.

Advanced Racking We've already looked at pretty much everything a V-Rack has to offer — namely, Instrument tracks, Aux tracks and Master tracks — and the way in which it remains quite independent from the other sequences in your project. There are a few extra techniques that can help you get the most from your V-Racks, however...

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Moving tracks to V-Racks: First, you're free to move entire Instrument, Aux or Master tracks from normal sequence tracks to a V-Rack at any time — so it's easy to update your old sequences, for example. All you need to do is select the track you want to move (making sure it's actually in the sequence that is currently playing back), then choose 'Move Selected Tracks to V-Rack' from the mini-menu in the Sequence Editor or Control Panel, or the Sequence menu in the Tracks Overview. DP puts up a dialogue box asking which V-Rack the track should be added to (if you have more than one), or if a new V-Rack should be created for it. Incidentally, once a track's in a V-Rack it's not so easy to transfer back to your sequence. Your only option, unfortunately, is to delete it (using the mini-menu at the bottom of its 'channel' in the Mixing Board), and then set it up again from scratch. Multiple V-Racks: As moving sequence tracks to a V-Rack suggests, it's quite possible to run multiple V-Racks in a Project, and the way in which you set them up is entirely up to you. You could have a soft-sampler V-Rack, an 'effects rack' V-Rack and an 'external routing' V-Rack, for example, all helping to keep very complex routing manageable. You can rename your V-Racks by opening the Chunks window (via the Project menu, or Shift + C) and option-clicking on their names. To quickly view or edit them in the Mixing Board you can doubleclick their names in the Chunks window, or (even better, I think) try this nifty shortcut: Apple-click the Mixing Board's title bar at its far right. A 'hidden' pop-up menu appears (see above) and you can directly choose to view any sequence or V-Rack in your project.

Loading & Saving If you've spent lots of time perfecting a VRack that you know you'll want to use again — perhaps you've created a favourite mastering chain or a big soft-sampler orchestral setup — there's a way that you can recall it in other projects even though MOTU don't provide a dedicated function for doing this. 1. First, save the project that contains your favourite V-Rack, and then close it. 2. Then create a new project, which you might call 'My V-Racks', for example. 3. In this otherwise empty project choose the Load... item from the File menu, and then navigate to your first project — the one that contains your V-Rack — in the file browser file:///F|/SoS/SoS%2011-2005/performertech.htm (6 of 8)10/19/2005 9:45:16 PM

It's easy to load up entire V-Racks (or indeed conventional sequences) from other projects using the File menu's Load... command. With a bit of forward planning, you can maintain all your favourite V-Racks in a single project.

Using V-Racks In Digital Performer 4.6

that appears. Open the actual project file and you'll be presented with a dialogue box (see the screen on the right) that allows you to load chunks individually. 4. Select your V-Rack (or Racks, by shift clicking items in the list), and make sure the 'Load Chunks' and 'Data' options are selected to the left of the list. Nothing else in the dialogue box needs to be selected. 5. Finally, click OK. Your V-Rack will be loaded into the new project. You can then save and close it. Using this technique you can maintain your 'My V-Racks' project as a repository for all your favourite V-Racks, and whenever you need them just load them from it, in exactly the same way as described above.

Master Fader Mania When you run multiple sequences and V-Racks in a complex project, there's the very real possibility of ending up, inadvertently or not, with multiple Master Faders for the same hardware outputs. This is not a helpful thing and normally DP won't allow you to do it. In fact, with the advent of V-Racks DP still won't allow you to do it, but Master Fader conflicts are handled differently. The golden rule is this: no matter how many similarly-assigned Master Faders you might think you have, only one is active. Exactly which one is governed by the relative position of Chunks in the list in the The Chunks window displays all the Chunks window. The Master Fader sequences and V-Racks in your project, but belonging to the sequence or Vtheir position in the list also determines how Rack that's in the topmost position any Master Fader conflicts are resolved. of the list is the one that's active, and this is why you're allowed to drag V-Racks around in the list. (Note that you can't drag V-Racks into a Song window as you can normal sequences.) To illustrate, here's a typical scenario. Imagine you have a number of sequences, each containing a separate piece of music for a film cue. You've also got a VRack set up with a Master Fader that's handling some subtle final compression and limiting for all of them. Out of the blue, you're required to change the sound of just one of your sequences, applying aggressive limiting and distortion. Rather than mess up the Master Fader settings in your V-Rack just to treat this one sequence, all you need to do is create a Master Fader within the sequence itself, apply your limiting and distortion there, and in the Chunks window drag the sequence to a position in the list that's above your V-Rack. Now, when your 'distortion' sequence is played its own Master Fader is used, and the Master Fader on the V-Rack is ignored.

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Using V-Racks In Digital Performer 4.6

What? There's a downside? Well, unfortunately the way in which V-Racks work has a knock-on effect on two key Digital Performer features, and it's worth understanding this before you find out the hard way. First, because V-Rack tracks don't have a timeline or any editing windows, it's not possible to automate them using DP's track-based automation. There's just nowhere for the data to 'go', and no way for you to edit it even if there were! This is not as bad as it sounds: it's much more common to automate synth parameters using MIDI continuous controller messages anyway, and those are certainly not affected. But if you're into heavily automating audio effect plug-ins you won't be able to do this if they're in a V-Rack. Similarly, it's not possible to automate V-Rack fader or pan movements, unless you wanted to set up a really complicated system using continuous controller mapping and one of DP's consoles, which I wouldn't recommend. Another 'gotcha' waiting to happen relates to DP's freeze tracks feature. Basically, freezing can't accommodate V-Rack tracks if you have them routed directly to your hardware outputs. However, it is possible if you take a slightly different approach to setting up your V-Rack's signal routing. What you do is route every Aux and Instrument track in the V-Rack via DP's busses back into Aux tracks in your 'normal' sequence. Then, when you need to freeze a track in the sequence you're working on, you just select it, along with its 'aux return', from the V-Rack. The rest of the time the V-Rack works as normal. This method detracts from the essential 'tidiness' of the V-Rack concept somewhat, because you end up with extra Aux tracks in your sequences, but it's certainly worth bearing in mind, especially if track freezing is an important part of your workflow and something you rely on. Published in SOS November 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

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Working With Video In Cubase SX/SL

In this article:

Inside Or Out? Understanding Cubase's Video Player Device Running Video In Cubase All You Need Is Cache

Working With Video In Cubase SX/SL Technique Published in SOS November 2005 Print article : Close window

Technique : Cubase Notes

Since its SX reincarnation three years ago, Cubase has once again become one of the most flexible tools for writing music to picture. Let's investigate... Mark Wherry

There are two things you need to master when working with video in Cubase: first, the mechanics of actually getting video to play in time with your Cubase Project and working with timecode, and second, building tempo maps that allow you to accurately place musical moments at specific time locations in the video. In this month's Cubase workshop, we're going to look at the first of these areas — but stay tuned, as we'll be turning our attention to learning how to build tempo maps specifically for writing to picture in a future issue.

Here you can see a Video and an Audio Event on the Project window, with Cubase's built-in Video Device Player window open. The parameters available in the Event Infoline are for the selected Video Event, and you'll notice that they're pretty much identical to the parameters you would expect to see for an Audio Event.

Inside Or Out? There are two ways you can run video alongside a Cubase Project. The first is to use Cubase's built-in video player and make your video file (or files) part of your Cubase Project. The second is to synchronise an external video device via MIDI Time Code (MTC) so that when Cubase's transport is running, the external device chases and runs in sync. There are many advantages to running video inside Cubase: not requiring any extra equipment is an obvious one, but another is that the synchronisation between Cubase's transport and the video is instant,

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so the picture starts playing immediately with the Project, and the appropriate frame at the current position of the Project Cursor is always shown. Despite these advantages, though, running picture in Cubase isn't always the best solution. As anyone involved in film or TV will know, it's rare to only ever receive one cut of video. A far more common situation is that you receive multiple cuts, on a weekly (or sometimes daily) basis, that have to be re-integrated into your Project. Let's say you're working on a 30-minute TV show that comprises 20 different music cues, and each of these music cues is a separate Cubase Project, each containing video. Every time you get a new cut of video, you'll have to update the video in 20 different Cubase Projects: if you're working on a feature film that might have 50 to 100 cues across multiple reels of video, this situation becomes really frustrating, really quickly. For this reason, a large majority of working media composers tend to run video outside of their sequencers, and even if your main sequencer is Cubase it's pretty common to see a separate system, locked via MTC, being used to run video. The big advantage of an external video machine is therefore that you can reload the video once and every Cubase Project will lock to whatever the external video machine is playing. The disadvantage is that it usually takes a second for the video machine to lock in with Cubase, and getting the video to display the exact frame at the current position of the Project Cursor isn't always as easy as you might think. So if you're working with one or two video files that won't be updated on a regular basis, the best approach is probably to run video within Cubase. However, if you're working with multiple video files that will have frequent updates, the extra expense of a separate computer to run video is probably justified by the time you'll save. While this article covers using video in Cubase, in the near future we'll be looking at tips for running a separate video system alongside Cubase.

Understanding Cubase's Video Player Device Cubase's built-in Video Player Device is fairly straightforward, but there are some differences to be aware of, depending on what type of movie file you're working with, what codec the video is compressed with, and whether you're running a Mac or Windows computer with or without external video peripherals. To configure the built-in Video Player Device, select Devices / Device Setup to open the Device Setup window, then choose Video Player within the Video folder, to see the available options in the panel to the right of the window. At the top is a pop-up menu labelled Playback Method, where you select which video engine is used to play back the video. By default, Mac users only have one option, 'Quicktime Video', while Windows users have this option along with 'Direct Show Video' and 'Video for Windows'. The Video Properties group offers various parameters for configuring the chosen Playback Method. The Quicktime Video Playback Engine is pretty flexible and supports Quicktime, AVI and MPEG video formats that have been compressed with either the Cinepak, DV, Indeo, Motion JPEG or MPEG video codecs. If you have any

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additional video hardware in your computer that supports Quicktime, you should be able to choose to output the video via this hardware from the Ouputs pop-up menu. Mac users are fortunate here because Quicktime for Mac supports DV output via a Firewire port (choose Firewire from Outputs and an appropriate output format from the Formats pop-up menu), which allows them to use a device such as Canopus' ADVC110 (www.canopus. com) to convert DV-encoded video from the Firewire port to S-Video or composite. This means that it's possible to output video from Cubase to a TV, for example. You do need to ensure that the video you load in Cubase is in DV format. If it isn't, Quicktime Pro can export the movie in this format.

Notice how the settings differ slightly for the Mac and Windows version of the 'Quicktime Video' Playback Method in the Device Setup window.

If you do end up using an additional video output device in conjunction with Cubase's Video Player Device, there's a useful Frame Offset parameter in the Quicktime Video Playback Method's Video Properties group that lets you specify by how many frames to play the video ahead of the actual Project time during playback. For example, setting this value to three means that the video will always play three frames ahead of the Project time during playback, which allows you to compensate for any latency introduced by the video hardware. So if you were using the ADVC110 you'd want Frame Offset to be set between five and six frames to allow for the digital-to-analogue video conversion. The 'Direct Show Video' Playback Method for Windows supports the same codecs and file formats as the Quicktime player, with the exception of support for Quicktime files themselves, but with the addition of support for Windows Media Video (WMV) files. So, for example, if you need to play back Quicktime files, you'll need to have the 'Quicktime Video' Playback Method selected; but if you have a WMV file, you'll need the 'Direct Show Video' Playback Method instead. The 'Video for Windows' Playback Method only supports AVI files and the Cinepak and Indeo codecs, although Motion JPEG can also be supported, depending on what video hardware you're using. The only Video Properties available for the 'Direct Show Video' and 'Video for Windows' Playback Methods are the choice of three possible sizes for the Video Player window. While the 'Quicktime Video' Playback Method doesn't offer these choices, it is instead possible to resize the Player window to any size you like by dragging the bottom-right corner as you would any other window.

Running Video In Cubase When you're running video in Cubase, you can be grateful that it has possibly the

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best internal video handling of any sequencer or digital audio workstation available, thanks to functionality that was originally introduced in Nuendo 1.0 way back in 2000. Basically, an audio file is referenced in a Cubase Project as an Audio Clip, and when you use an Audio Clip in the Project window you create an Audio Event, which can be dragged around, resized, and so on. A video file is therefore referenced as a Video Clip, and a Video Clip appears in a special Video Track in Cubase as a Video Event, which can be dragged around, resized and handled just as if it was an Audio Event. To add a video file to a Cubase Project, select Pool / Open Pool Window, click the Import button at the top-centre of the Pool window (or Video Clips are organised in the Pool's Video Folder. Notice how the Info column details select Pool / Import Medium) and several useful parameters, such as the frame choose the video file to be added. A rate of the video. Video Clip will be automatically added to the Pool to represent the video file. If your video file contains audio content, such as dialogue or temp music, you'll need to extract this data as a separate audio file, as Cubase has no way of playing audio contained within a movie file. To do this, select the Video Clip in the Pool and choose Pool / Extract Audio from Video File. A new Audio Clip is added to the Audio Folder in the Pool. At this point it's worth noticing the Info column in the Pool, which for a Video Clip tells you the frame rate of the video (in addition to its length and resolution), and for an Audio Clip shows what sample rate, bit depth and channel format are used, along with the length of the file. It's important that the frame rate of the Cubase Project matches that of the video, so that the timeline of the Project exactly matches the timeline of the video. To set the Project's frame rate, open the Project Setup window by selecting Project / Project Setup (or pressing Shift +S), choose the appropriate frame rate from the Frame Rate pop-up menu, and click OK. The next step is to create a Video track in the Project window (Project / Add Track / Video) and an Audio track (Project / Add Track / Audio) of the appropriate format (such as stereo or mono, as indicated by the information in the Pool). Now, drag the Video Clip from the Pool onto the Video track so that it's positioned at the start of the Project; and do exactly the same with the Audio Clip, so that it lines up precisely underneath the Video Event. In order to keep the video and its audio together, it's a good idea to group these two Events, by selecting them both and choosing Edit / Group (or pressing Ctrl/Command+G). Now, any edit operations you apply to either Event will be applied to the other Event. You can later ungroup these, if you want to, by selecting both Events and choosing Edit / Ungroup (or pressing Ctrl/Command+U). Once you have Video Events in a Video track, you can open Cubase's video window where the playback of the video will be displayed by selecting Devices / Video or pressing F8. See the 'Understanding Cubase's Video Player Device' box (below) for more about the various playback options.

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Working With Video In Cubase SX/SL

At this point, it's important to make sure that the video is placed (or 'spotted') to the correct timecode location in your Cubase project. Hopefully, your video will contain a timecode burn-in (literally, timecode that's visible, or 'burnt', into the image you see) from the editorial department that gave you the video; if it doesn't, you may need to ask them what the first frame of picture should be. If there is indeed a burn-in on your video, you need to make sure that the timecode in Cubase precisely matches the timecode on the video.

The Event Display-Video page in the Preferences window lets you disable the drawing of thumbnails on Video Events to save your computer's resources, and also to set the size of the video cache memory used to store the thumbnails, saving on disk access.

If the timecode starts running at the start of the video, all you need to do is take a note of what the SMPTE time is on the video. This is easy if the Video Event is at the start of the Project, as you can just set the Project Cursor to zero and look at the Video Player window. Next, open the Project Setup window again and set the Start value to the timecode value at the start of your video, before clicking OK. Now, wherever you place the Project Cursor, the timecode visible on the video should be exactly the same as the timecode readout in Cubase. If you can't see a timecode readout in Cubase, set the Secondary Display Format to Timecode by clicking the Select Secondary Time Format button (which is located just above the Record button on the Transport Panel) and selecting Timecode from the pop-up menu. Usually, it's best to leave the Primary Display Format set to Bars+Beats, since that's the time format you'll need to work in once you start writing some music. The only slight problem you might encounter is if the timecode on the video doesn't start running at the very start of the video — in this case you'll need to crop the Video Event so that the start of the Event is the first frame of video you can reference in terms of timecode. This is pretty easy, as you can change the start point of the Video Event just as you would that of an Audio Event, by dragging the Event's start handle, or numerically by selecting the Event and changing the Start time in the Event Infoline. If you're going to adjust the start of the Event in the latter way, you might want to swap the Primary and Secondary Display Formats temporarily, so that you can adjust the start of the video in terms of frames rather than bars and beats. Do this by clicking the Exchange Time Formats button (which is an icon just above the Stop button on the Transport Panel and looks like a thin line with two arrows).

All You Need Is Cache

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Working With Video In Cubase SX/SL

By default you'll notice that the Video Event displays a series of frames from the video, to give an approximate idea of what's going on in the picture at any given position along the timeline. The option to see (or not see) these so-called 'thumbnails' is located in the Event Display-Video page of the Preferences window, and you'll notice that this page also contains an option for setting the size of the Video Cache. The Video Cache is used to store the thumbnails generated from the video file in memory, so that Cubase doesn't have to read from the disk every time the thumbnails need to be redrawn. If you find yourself zooming in and out of the Project window horizontally quite frequently, this causes the thumbnails to be constantly redrawn, so it's worth increasing the cache size if you notice either your computer slowing down or the process of redrawing the thumbnails on the Video Event being sluggish. Cubase SX/SL 3.1 adds a further new trick for improving performance when working with thumbnails, by offering the option to create a thumbnail cache file on disk as well. The way this works is that if Cubase needs to redraw a thumbnail that doesn't exist in the memory cache and would take too much processing to generate from the The Project Setup window enables you to set video file, it uses a lower-resolution a SMPTE timecode position for the start of thumbnail stored in the cache file the Project, which makes it possible to precisely synchronise the playback of the instead. Once the processor is free to video in Cubase. The frame rate of the video redraw the thumbnail from the video you're working with is also set in this window. file, Cubase will do this, and the thumbnails will be redrawn at a higher resolution. To create a thumbnail cache file, select the Video Clip in the Pool and choose Pool / Generate Thumbnail Cache. At this point I have a small confession to make: there is actually a quicker way of performing the tasks I've just described. Selecting 'File / Import / Video' displays a file selector with two built-in options that are enabled by default: 'Extract Audio' and 'Generate Thumbnail Cache'. If these options are enabled, Cubase will automatically extract the audio, generate the thumbnail cache, add the appropriate Audio and Video tracks and the Audio and Video Events. The reason for explaining the whole of the process is more for those situations where you might need to do something out of order, or work with multiple video files in the same Project. This latter point is actually really important, because Cubase SX is one of the only sequencers on the market that can deal with multiple video files in the same Project, since the Video track, like any Audio track, can contain multiple Events from different files. This is incredibly handy, because TV shows and films are often broken down into acts and reels, and Cubase allows you to load these in a single Project if you wish, without having to use a separate application to merge them together. And that's just about all there's space for this month. Coming soon, synchronising

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external video playback, and a look at how to write music to picture using Cubase's tempo map features. Published in SOS November 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

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