Sound on Sound (January 2006)

In This Issue January 2006 In This Issue Click article title to open Reviews People Antares Avox From 4AD To Nine In...

324 downloads 1868 Views 3MB Size Report

This content was uploaded by our users and we assume good faith they have the permission to share this book. If you own the copyright to this book and it is wrongfully on our website, we offer a simple DMCA procedure to remove your content from our site. Start by pressing the button below!
Report copyright / DMCA form

DOWNLOAD PDF

In This Issue

January 2006 In This Issue Click article title to open Reviews

People

Antares Avox

From 4AD To Nine Inch Nails

Vocal Processing Plug-ins [Mac OSX/WinXP] With their new plug-in bundle, Antares promise to let you change the character of the human voice.

John Fryer

Buchla 200e Patchable Analogue & Digital Synthesizer

The likes of Depeche Mode, Cocteau Twins and Nine Inch Nails all owe a sonic debt to engineer/producer John Fryer, who explains his approach to production.

Futurism... PART 2: We conclude our look at synth pioneer Don Buchla's extraordinary new 200e modular synth.

CAD GXL Series Condenser Microphones This new set of Chinese-built mics offers good-quality vocal and instrument recording on a budget.

Paul White's Leader The year is 2016 and Editor In Chief Paul White pens another thought-provoking leader column...

Sounding Off: Monitors Mike Senior Monitors have so much to answer for...

Creamware Minimax ASB

Studio SOS

Modelled Analogue Synth Creamware's Minimax is certainly not the first digital emulation of a Minimoog to be released — nor even Creamware's first. But it bucks the recent trend for software recreations of vintage synths — by being hardware. Can it replace the real deal?

South The SOS team get busy at the London studio of a band called South, by transforming the sound of their troublesome monitoring room.

Digidesign Pro Tools v7 Recording Software [Mac/PC]

With new MIDI sequencing functionality among many other features, Pro Tools v7 is intended to be the ultimate audio and MIDI workstation. Will this release keep Digidesign on top?

DK Technologies MSD100C Stereo Audio Meter The baby of the MSD series of meters gets an overhaul.

Drawmer Three-Sum Band-splitting Processor

Technique

Anyone got time to make music? PC Notes We'd probably all prefer to keep our music PCs insulated from viruses, spyware, adware, phishing and the general nastiness of the Internet, but the way music software is developing makes this increasingly difficult. Find out more...

Avoiding The Blue Screen Of Death PC Musician If you've ever been confronted by the dreaded Blue Screen Of Death, suffered random reboots or faced the frustration of inexplicable PC crashes, read on for some preventative measures...

CLASSIC TRACKS: The Staple Singers I'll Take You There Producer: Al Bell; Engineers: Terry Manning, Jerry Masters For the Staple Singers' landmark 1972 Stax album, Terry

file:///F|/SoS/SoS%2001-2006/Contents.php.htm (1 of 4)12/19/2005 10:18:11 AM

In This Issue

Manning and producer Al Bell employed the talents of Memphis's finest musicians and two of the South's most famous studios. This intriguing new unit from Drawmer lets you split a stereo signal into three bands and then process each band with a different piece of outboard.

Guitar Technology Tips, Techniques & Gear Our new regular section for guitarists combines tips and techniques with a look at interesting new gear. This month, it's the Cornford Carrera recording amp and the Takamine Cool Tube acoustic preamp.

Korg D3200 Digital Multitracker Korg's newest workstation heavyweight boasts 32 recording tracks, a powerful 44:12:2 mixer, a programmable drum machine, and up to 11 simultaneous effects — all for under £1000. Read our test report...

M Audio Project Mix I/O Firewire Interface & Control Surface [PC/Mac]

Making A Living From Music For Picture Part 2 If you're ever going to make it in this game, you need a calling card, a way of impressing potential clients with your musical ability — you need a showreel. We explain what to do to create one...

Making The Most Of Digital Performer Plug-ins Digital Performer Notes & Techniques Can you honestly say you use all the features of DP's plugins? If not, prepare to be intrigued, as we dig deeper into Dynamics, Multimode Filter, Sonic Modulator and more, in search of the facilities you didn't know were there.

Mastering Your Album In Logic Logic Notes & Techniques We look at how to combine your hardware and the software-processing within Logic for mastering purposes.

Mix Rescue + Audio Files Can't get your mix sounding right? Let SOS sort it out! M Audio have packaged a fully featured control surface with motorised faders, an 18-input Firewire interface and eight mic preamps in one box — at a very competitive price.

Mackie Onyx 400F Firewire Audio Interface Mackie's new 10-in, 10-out breakout box includes and internal DSP mixer, MIDI connectivity, and four of their highspec Onyx preamps.

Native Instruments B4 II Modelled Drawbar Organ Software [PC/Mac] Since its launch in 2000, NI's B4 has been the software instrument of choice for those who want realistic tonewheel organ sounds from their computer. But NI are clearly convinced it can be better...

RPCX Blue

PCI Express: What Does It Mean For Mac Musicians? Apple Notes With the new single-processor, dual-core Power Macs shipping, we continue to investigate the impact of PCI Express on the Mac audio and music world.

Recording & Remixing With Ableton Live Ableton Live Notes & Techniques In the first instalment of this two-part article, we examine how you can use Live as a conventional digital audio workstation.

The Lost Art Of Sampling Part 6 We continue exploring what your sampler's synth engine can do to liven up bland self-sampled sounds, and explain the concepts of layering and multitimbrality.

VST 2.0 & AU Synth Plug-in [Mac/PC] Sound designer Rob Papen may be known to you as the Tomorrow's Musicians & What They'll Be name behind amazing sounds for the Access Virus or Alesis Playing Andromeda. Now he has moved into producing softwareControllers Of The Future based synths of his own. So how good is it? The New Interfaces for Musical Expression (NIME) Sample Libraries: On Test conference has been running for five years, and is a great place to see and discuss new ideas that may provide the Hot New Releases musical controllers of the future. SOS was in Vancouver to Funk University **** learn more... Koncept & Funktion ****


In This Issue

Chopped Guitars **** Downtempo Guitars Volume 2 **** Talkbox Guitar *****

Scan 3XS Dual-core Athlon 64 PC For Music

Using Sonar 5's Cyclone Loop Tool Sonar Notes & Techniques The Cyclone 'groove sampler' DXi, bundled free with Sonar, is a powerful tool that allows loops of all kinds to be dissected, manipulated and generally bent to your will...

Using The Logical Editor In Cubase SX & SL Cubase Notes & Techniques Cubase contains many powerful features for processing Dual-core CPUs promise a huge jump in performance at a MIDI data, such as the Logical Editor and Macros. This modest price, while RAID disk arrays can provide both faster month we look at how using them together can create and more secure storage. Scan Computers' Athlon-based some powerful solutions to potentially tedious problems. system features both technologies.

Tannoy Reveal 8D Active Midfield Monitors

Working With Video In Pro Tools Pro Tools Notes & Technique This Pro Tools workshop is the first in a series where we will explain how to use Digidesign's DAW to work to picture. First of all, we look at the decisions you need to make in setting up your system.

Updating their successful Reveal range, Tannoy have included new high-resolution tweeter technology, sophisticated room-correction EQ, and digital interfacing.

Tascam FW1804 Firewire Audio & MIDI Interface [PC/Mac] If you fancy Tascam's FW series of Firewire interfaces but don't need their control-surface features, the new FW1804 might be just what you are looking for.

Yamaha MG8/2FX Analogue Mixer Straightforward facilities, built-in effects processing, and an affordable price make this new analogue mixer a good choice for entry-level studio and live setups. Competition

WIN: East West/Quantum Leap Sample Libraries Sound Advice

Q. How can I achieve phase inversion? Q. Should I be using my mixer's group outputs or its direct outs for recording? Q. Should I opt for active or passive monitors? Q. What is 'aliasing' and what's the cause of it? Q. What should be the next step for my studio?


In This Issue

Q. Which microphone should I buy for recording vocals? Q. Why does my Mackie Control make strange noises in Cubase?


Antares Avox

In this article:

Deep Throat Hearing Double Bring The Noise Sybil Engineering Impressions

Antares Avox Vocal Processing Plug-ins [Mac OSX/WinXP] Published in SOS January 2006 Print article : Close window

Reviews : Software

Antares Avox £400 pros Easy to use. Flexible and creative vocal processing tools. Can create abstract as well as realistic vocal sounds.

cons

With their new plug-in bundle, Antares promise to let you change the character of the human voice. Paul White

Throat has to be used carefully in order to retain a natural vocal sound.

summary As a bundle, Avox offers the promise of creativity without tears and on the whole it lives up to that promise. Vocal modelling and resculpting still has a long way to go, but there are some genuinely useful tools here that are surprisingly straightforward in use.

information £399.50 including VAT. Unity Audio +44 (0)1440 785843. +44 (0)1440 785845. Click here to email

Antares have been a major player in vocal processing ever since they unleashed Auto-Tune upon an unsuspecting world, so anything new that they come out with is always worth a closer look. A recurring dream of engineers and producers is to be able to adjust the character of the human voice in a more or less natural way to make one singer sound more like another, and it's probably true to say that until this particular suite of Antares plug-ins appeared, TC-Helicon were the only big name doing anything significant in that particular market sector. Now Antares have joined the race, and though we're still a long way from the 'Dial E for Elvis' box, a lot of interesting vocal manipulation is now possible.

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

Supplied in VST and RTAS versions on Mac and PC, and Audio Units on Mac OS, Avox is protected using iLok; an iLok key is included in the package so you don't have to go out and buy one. Installation is straightforward, and you can run the program for 10 days prior to authorisation if for some reason you're not able to access the Internet to authorise your iLok key immediately.

Deep Throat

file:///F|/SoS/SoS%2001-2006/antaresavox.htm (1 of 6)12/19/2005 10:18:19 AM

Antares Avox

Avox is actually a suite of vocal effects, and appears as five separate plug-ins named Choir, Duo, Punch, Sybil and Throat. Since the latter is the most ambitious of the plug-ins, I'll begin with it. Throat is designed to change vocal character by processing a real vocal track through a virtual vocal tract, with controls to change the throat shape and size, breathiness and glottal waveform. Avox's modelled throat is divided into four sections, which can be changed in length and width to subtly or otherwise change the character of the sound being processed. A graphical display allows five points on the virtual throat to be dragged horizontally and vertically, with the original positions also being shown for comparison purposes. For Throat to do a good job, you need to tell it some basic facts about the source voice (or other sound) using a menu that lets you choose from soprano, alto/tenor, bass/baritone and instrument. The glottal waveform is the basic sound produced by the vocal cords prior to filtering by the vocal tract, and varies quite a lot between different singers. Here we have a menu of options that goes from hard to soft plus a pulse-width fader, and while this might be simplistic by comparison with a real voice, it still provides plenty of leeway for tonal variation. When a voice type is loaded, an appropriate glottal waveform is selected by default. Maybe version 2 will come with drag-and-drop vocal cord nodules for those Rod Stewart impressions? Next comes a control menu called Source Throat Precision, which relates to how accurately the algorithm tries to model the throat you've set up. As a rule, high precision settings work best for subtle vocal adjustments while lower precision settings can produce smoother results when creating more radical effects, and can also reduce whistling artifacts caused by extreme settings. The manual suggests that you always start with this parameter set to 'subtle' and then work up the list a step at a time until you find the bestsounding option. Breathiness Mix adds a synthesized breathiness to the sound, the character of which is drawn from the source sound and modified using the adjacent frequency slider. This needs to be used quite subtly to maintain a natural sound, and if you need a Darth-Vader-withasthma effect, there's more than enough range to achieve that. Once the faders are set and the menu options selected, you can start having fun in the Graphic Throat display by dragging the five boundary points in any


Antares Avox

direction; the grey band in the centre shows roughly where natural adjustments cease and creative abuse starts. The original throat plot is shown in blue, while the points you drag to new positions show up in red. The Length fader then scales these settings up or down globally. Should you stumble on some interesting dynamic effects while editing, you can automate the point positions in your sequencer to create, for example, the sound of somebody singing while being strangled! When audio is being processed, the display also shows the original and remodelled throat contours in real time, again in blue and red. A reset button puts the vocal tract back to its default start points, output gain provides up to 24dB of gain to compensate for level changes caused by the modelling process and Level Matching, when active, tries to make an automatic level compensation so the output is of similar loudness to the input. It is recommended that Level Matching be used only for comparison purposes and that it be switched off for serious use as it compromises the audio quality very slightly. Throat is a mono plug-in and requires a clean, monophonic source to work effectively as it needs to track the pitch of the incoming voice. A range of presets has been provided, but as every voice is different, these should be regarded as starting points only. My tests produced results that had an obvious parallel with the TC Voice Modeller plug-in for Powercore, insomuch as you have to keep the adjustments fairly small if you want to keep the sound natural. Go too far and you start to sound like a blackmailer's phone call or one of Doctor Who's enemies! More than a little processing also tends to lend the vocal sound a slightly lo-fi quality, and while it can be interesting to really push the sound into this realm for darker hip-hop vocal sounds, it is to be avoided for conventional vocal work. My own view is that used carefully, Throat can be used to create some really useful variations, but I don't think I'd often use it on a main vocal part unless I was after something obviously treated.

Hearing Double The Duo 'auto doubler' is an altogether simpler prospect, and sets out to emulate traditional double-tracking by processing a copy of the original vocal to change its timing, pitch and amount of vibrato. Again, it needs to be told the approximate vocal range of the input in order to be able to track the pitch effectively, so you have to pick from soprano, alto/tenor, bass/baritone or instrument. After that you simply have four sliders for controlling Timbre, Vibrato, Pitch and Timing, all of which introduce variations on the characteristics of the original voice. It should be noted that Vibrato acts on any vibrato present in the original voice, so if none is present, there will be no effect. Two further level sliders and pan pots allow the original and processed voices to be positioned in the mix. With timing set to maximum, you get a distinct slapback sound, whereas at lower settings the result is more akin to two singers trying to perform the same part. A degree of randomisation has been introduced to stop the process sounding to mechanically perfect.


Antares Avox

While I'm not convinced that Duo reproduces the same life and sparkle as a really good natural double-tracking job, the overall effect is still pretty realistic and the ability to tweak the character of the doubling voice slightly just adds to the illusion. The best results are achieved by panning the two voices to either side of centre and by not going overboard with the variation controls. While the plug-in can work in stereo mode, it processes only the left channel, so it is best to use it in mono-in, stereo-out mode.

Bring The Noise Choir takes the Duo concept one step further by allowing the user to add four, eight, 16 or 32 voices, all based on processes applied to one original part. The controls are similar to those of Duo but without the Timbre parameter: there are sliders for Vibrato, Pitch and Timing variation, with a further slider setting the stereo spread. The result is slightly artificial if scrutinised in isolation, but it gets far more realistic when you add concert hall reverb. For me the smaller four- and eight-voice choir sizes sounded the most realistic, so building up a large choir from several instances of the plug-in working on three or four original vocal parts would be a better option than trying to emulate a whole choir part based on one vocal track — and of course you'd have to do this anyway to build up harmony parts. Again only the left channel is processed if you use the plug-in in stereo mode, and the more voices you specify, the greater the CPU load. By combining compression and limiting in a single plug-in with a very simple control interface, Punch provides a simple way to get a vocal part to sit in the track. The Gain control adjusts the level of the signal feeding into the plug-in, while Impact sets the degree of compression/limiting applied to the signal. Ceiling sets the maximum level that the signal is allowed to reach, so in use, the controls are more like those of a limiter than a compressor. The effect of the processing can be seen in the dynamics of the output level meter but there's no dedicated gain-reduction meter, and I didn't feel comfortable without one. Though you should use your ears, I also like to be able to see how much the signal is being hammered! Punch has both mono and stereo versions and is certainly successful in creating that solid, up-front vocal sound without having to juggle a host of parameters, though if you apply too much Impact, breathing artifacts start to show up as they do with any over-enthusiastic application of compression. With the addition of a gain-reduction meter, this plug-in would have really won me over, but even without, you can nail the level of a vocal track in a mix without being a compression wizard.

Sybil Engineering While most of the plug-ins in Avox include some clever twists on what has gone file:///F|/SoS/SoS%2001-2006/antaresavox.htm (4 of 6)12/19/2005 10:18:19 AM

Antares Avox

before, Sybil is really a digital emulation of an old-school de-esser, where a highpass-filtered side-chain forces a compressor to pull down the overall signal level when 'S' and 'T' sounds are detected. A gain-reduction meter shows how much the level is being attenuated when processing is taking place, while a variable high-pass filter (which defaults to an 8kHz cutoff) lets you home in on the frequency above which sibilance occurs. Threshold sets the level at which detected sibilance triggers gain reduction while the compressor section has its own attack, release and ratio controls. Sybil is mono-only, and because it works like an old-school de-esser, it pulls down the level of everything when sibilance is detected, not just the highfrequency sibilant sounds. Used moderately it works fine and is easy to set up, thanks in part to the gainreduction meter, but if you overdo it, the processed sound takes on a lisping quality, which isn't particularly attractive. In this respect, Sybil isn't so much faulty as unsophisticated! I would imagine that most DAWs include a de-esser plug-in that's at least as effective as this one, so I can't quite see why Antares didn't go the extra mile and have the compressor act only on the upper half of the frequency spectrum or, better still, just on the frequency band containing the sibilance.

Impressions On the whole, Antares have come up with a suite of creative and easy-to-use plug-ins that make vocal manipulation very straightforward. My least favourite is the rather conventional (but still very functional) de-esser, as I think there are numerous better options available, but everything else does what it is claimed to do with an added dash of panache. Choir, when used to turn three or four singers into a choir, is most impressive, especially once you add a nice church or concert hall reverb, while Duo does a good job of turning one vocal into a plausible double-tracked part. It's still not as good as real double-tracking done well, but then not every vocalist can sing an accurate double-tracked part. These effects also work well on monophonic instruments. Punch doesn't do anything you can't achieve by other means but its simple interface should win it a lot of friends, as other than setting levels, you only have


Antares Avox

an amount slider to deal with and it does exactly what's required to pump up the vocal sound and to stabilise its level. If you can't get your vocal to sit in a track using this, it's probably deficient in some other way. It's a shame that more of the entry-level compression plug-ins don't follow this example as there's still a lot of confusion about how best to use a compressor. Throat is the most complex of the plug-ins and also the one likely to cause the most controversy over whether it sounds any good or not. Used with caution, it can add that missing edge or depth to voice without sounding unduly processed, and those into rap or hip-hop will probably appreciate some of its more aggressive settings. However, Antares have deliberately allowed this plug-in to go way beyond anything that might occur in nature, so sound designers for film may also find it useful. Some of the sounds you can get out of it are truly bizarre, yet you can still hear every word! If you need the sound of a snake with dentures or a cave troll singing while having a tracheotomy, this is your guy. For all that, Throat is still pretty easy and quick to use, as is the rest of this flexible and creative bundle. Published in SOS January 2006 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


Buchla 200e

In this article:

Buchla 200e

225e MIDI/USB Decoder Patchable Analogue & Digital Synthesizer 259e Complex Wave Published in SOS January 2006 Generator 292e Quad Dynamics Print article : Close window Manager Reviews : Modular Synth 281e Quad Function Generator The Manual 291e Triple Morphing Filter We conclude our look at synth pioneer Don 259e Waveforms extraordinary new 200e modular synth. 260e Pitch Class Generator Gordon Reid 266e Source Of Uncertainty 210e Control & Signal Following on from the first part of this Router review in last month's SOS, this month I'm 227e System Interface going to take a deeper look at each of the modules in the review 200e system, hook 249 (DArF) The Sound Of The 200e them together to see what sounds can be obtained, and then try to decide whether Pricing the 200e can justify its hefty price tag. Price & Prejudice Conclusions

Buchla 200e pros It has huge potential if you have the time to devote to it. It's very stylish — you'll love it or hate it. It's incredibly portable for such a 'big' system. For many of us, it's an entirely new way to approach analogue synthesis. The sound quality is excellent.

cons It can be very frustrating. It's not compatible with other modular systems. There's a steep

Buchla's

225e MIDI/USB Decoder As discussed last month, the 225e is the heart of the 200e, converting MIDI information to analogue control signals and Photos: Mark Ewing then supplying these by patch cable and buss to the dozens of destinations in the synth. The Preset Manager in the 225e is also capable of saving and recalling the values of most (but not all) of the knob and switch values in the system modules with an 'e' in their names. The method is a bit clunky, because you have to 'Remote Enable' the connection in each of the modules whose values are to be saved or loaded, and it's important to know which values are not stored, so that you can jot down their values manually. And of course, the 225e has no way of knowing which cables are inserted, so it can't provide a true patch memory system. If you fancy patching the 200e in a single, unchanging configuration, and using the 210e Control and Signal Router (see page 153) to control a limited number of CV sources and destinations in the same manner as an integrated synth with routing switches, you can get closer to saving and recalling sounds in their entirety, but only by sacrificing the flexibility of patching freely from anywhere to anywhere.

file:///F|/SoS/SoS%2001-2006/buchla200e.htm (1 of 14)12/19/2005 10:18:57 AM

Buchla 200e

learning curve. The manual's terrible.

summary The 200e is not a synth that will appeal to everyone. If your objective is to make sounds and play tunes with the minimum of fuss, the 200e is the wrong choice. But if you fancy expanding your horizons into deeply esoteric realms, a Buchla will develop your sonic palette beyond what is possible using more conventional models of synthesis.

Nevertheless, the Preset Manager is a big step forward from no memories at all. You can name and store up to 30 presets, and select them using the last/next buttons on the panel. You can also step through them by presenting timing pulses to the associated inputs. This could allow sounds to switch themselves to the next (or the previous) patch in a sequence! The final set of facilities in the 225e is called Global, and handles functions such as formatting memory cards, saving to them and recalling presets from them. Unfortunately, the cards seem to be proprietary. With USB memory so cheap and easily obtained, I am surprised that Buchla didn't adopt this approach.

information See the 'Pricing' box above. RL Music +44 (0) 118 947 2474. Click here to email www.rlmusic.co.uk

259e Complex Wave Generator The four 259e modules in the lower boat are the guts of the 200e, each including a Principal Oscillator and a Modulation Oscillator, or Mod Osc. Let's start with the Principal Oscillator... In the bottom right of the module, you'll find a coarse tuning knob calibrated from 27.5Hz (MIDI note A1) to 7040Hz (MIDI note A9), giving it a huge tuning range of eight octaves. This is echoed by the ±4 octave transpose range offered by each of the internal busses, which override the position of the tuning knob if you control the 259e via MIDI. Beneath the tuning knob, there's a 3.5mm (audio) FM input with an amplitude control knob, plus a CV input with a bi-polar amplitude knob. If you're playing conventional melodies using a MIDI keyboard, you need use these only for effects, because the 200e's internal busses take care of standard pitch control duties and, when controlled by one of these, a 259e tracks very well. The final control in this section is a tiny, unmarked knob for fine-tuning. Although the only waveform generated by the Principal is a digitally generated sine wave (the first of the waveforms shown in the box on page 154), this is passed down two signal paths ('green' and 'red') with eight waveshaping positions, whereupon it is either passed unmolested to the output (position 1) or warped into more complex shapes (positions 2 to 8). The amount of warp for each position is determined by the Warp knob, and the mix of the green and red channels is determined by the Morph knob. Both of these controls can be modulated by dedicated CV inputs, and the amount of Warp and Morph modulation can be determined independently by the adjoining bi-polar amplitude controls. Setting Morph to one extreme or the other and


Buchla 200e

sweeping Warp (as I did to create the waveforms shown on page 154) demonstrates that the sounds generated by the positions are very different from one another, and very different from the positions in the other channel, with the 'red' timbres typically having the less complex harmonic structures. When testing the 259e in this fashion, it soon became apparent that it sounds nothing like a conventional analogue oscillator. In stark contrast, it sounds like nothing so much as a wavetable synthesizer being swept though its more esoteric tables. It even generates a significant amount of aliasing if you play the more complex waves at high pitches. This is not surprising. As far as I can gather, the 259e uses waveshaping tables to distort the initial sine wave into all its other waveforms, with the Warp knob controlling the position in the tables. As you can imagine, the sonic complexity offered by two warpable, mixable waveforms is immense, but that's far from the whole story, because alongside the Principal, there's the Mod Oscillator. This has a low-frequency range from 0.25Hz to 64Hz, but also offers the same audio range as the Principal; eight octaves from A1 to A9. Like the Principal, this too has FM and CV inputs with amplitude control knobs, and it has a Pitch Track option that connects it to the appropriate buss (A, B, C or D, depending upon which 259e you're adjusting). This means that each 259e module in the system is a true audio-frequency dual-oscillator device. What's more, the Mod Oscillator generates its LFO waveforms — sawtooth, square and triangle waves — in the audio domain, so you could use it (at least in theory) to generate 'analogue' timbres that are hard to obtain from the Principal. However, there's a caveat; the Mod Osc aliases like crazy when used in this way (once again, see the box on page 154 for more on this). The Mod Osc can be directed internally to any combination of the Principal Oscillator's pitch, Warp and Morph, and can do so in either range, so you can create modulations ranging from gentle vibrato to outlandish screams of harmonic anguish, without a patch cord anywhere in sight. The amount of modulation is determined by the Modulation Index knob, and this can itself be modulated using the associated CV input and bi-polar amplitude control. The only facility I've yet to mention on the 259e is Sync. At the touch of a button, the Mod Osc can be hard- or soft-sync'ed to MIDI Clock or to the Principal Oscillator. The first of these is useful for reinitialising low-frequency modulation to keep tempo with MIDI-sequenced music. The second allows you to produce those instantly recognisable sync lead and bass patches. Finally, at the top of the module lie four outputs. These comprise two identical audio outputs for the Principal, plus a CV output and an audio output (which carry the same signal) for the Mod Osc. There's nothing stopping you from taking these outputs and feeding them back to the inputs on the same 259e to generate yet more radical (and usually cacophonous) sounds.

292e Quad Dynamics Manager file:///F|/SoS/SoS%2001-2006/buchla200e.htm (3 of 14)12/19/2005 10:18:57 AM

Buchla 200e

The 292e is a combined VCA/VCF module offering four devices called A, B, C and D in deference to their internal connections to the busses of the same names. There are three modes of operation — VCA-only, lowpass VCF only, and combined VCA/VCF — and the response for each device is determined by the CV inputs to the left of the module. Further control is provided by Velocity CV inputs, and the Remote Enable connects this to the MIDI velocity on busses A to D if desired. The only knob per channel is an Initial Gain control that passes signal unimpeded from input to output. You'll notice that there aren't any filter Cutoff Frequency and Resonance knobs. This isn't the only way in which the 200e fails to conform to the 'accepted' model of analogue synthesis, as we'll see.

281e Quad Function Generator Back in 1969, EMS confused UK synthesists by introducing a module that they called a 'Trapezoid'. This was a contour generator that provided Attack, On, Release, and Off stages or, when cycling, a range of lowfrequency waves shaped by the Attack and Release values. The 281e is simply four such trapezoid contour generators. Shaping a contour is achieved using just two knobs, Attack and Decay (which I would call Release). Both of these can also be adjusted using CVs. Three modes are provided — ASR, AR, and repeating AR, independently selectable for each trapezoid — and the output appears at each generator's blue socket. The red outputs provide timing pulses at the end of each contour or, when cycling, at the end of each cycle. But this is just the tip of the iceberg. The four contour generators — again called A, B, C and D — are arranged as two pairs that are capable of generating more complex contours. There are two architectures for doing this. Firstly, you can use the logic in the lower right-hand corner to mix the A&B (and C&D) trapezoids to create four-stage ADSR and five-stage AD1D2SR-type curves. It does this by allowing you to create a transient on (say) A, followed by an attenuated sustained section on B. Buchla's system (which he calls an 'Or') then determines the highest voltage at any given moment and presents this to the output (see the top two diagrams on the opposite page). Unfortunately, the attenuation levels for B and D are not stored in a 225e preset, so alas, you cannot store these contours as part of your patches. The second method for creating complex contours is called 'Quadrature Mode', and is


Buchla 200e

accessed by pressing the 'A-B' and/or 'C-D' buttons in the lower left of the module. In this mode, the linked pairs operate as follows (using 'A-B' as an example). Firstly, the A buss is triggered, and A enters its Attack phase. When this is completed, A remains at its maximum level, and B begins its Attack. When this is completed, everything is sustained (when in Sustain mode) or A begins its Decay (when in Transient mode) while B maintains its maximum level. B then begins its Decay. At the end of all of this, if A is in Cyclic mode, the entire process repeats ad infinitum.

Two 281e contours, one an AR transient, the other an ASR sustain.

Two dissimilar 281e ASR contours, A and B.

Summing the two contours while attenuating B by 50 percent.

Summing A and an offset version of B to create a new contour.

Buchla & Associates describe this algorithm as having the two contour generators 90 degrees out of phase with one another. That's not as daft as it seems; there are four stages, and B lags A by one stage. In the lower two diagrams opposite, you can see what happens in a simple case, and the contour that you obtain if you stack the CV outputs of A and B. Happily, the straightforward trapezoid contours, quadrature contours and Or contours are available simultaneously (with some logical restrictions), making each 281e enormously flexible once you get your head around what's going on. The 281e's fastest attack is quoted as a remarkable 1ms, and my tests verify this. I passed high-frequency noise through a VCA 'blipped' by one of the contour generators in Transient mode, with Attack and Decay set to 0. The VCA was open for file:///F|/SoS/SoS%2001-2006/buchla200e.htm (5 of 14)12/19/2005 10:18:57 AM

Buchla 200e

a total of around 190 milliseconds, with an almost instantaneous Attack. The Decay took much longer in total — it dropped back to zero in around 15ms, but took another 140ms to settle. I suspect that this profile is generated not by the contour generator, but is the response of the VCA in the 292e. Either way, this was surprising... so I duplicated the test, passing a high-frequency audio wave through one channel of a 292e in VCA-only mode. This verified the earlier result; patches shaped by a 281e and a 292e combine an extremely snappy attack with a much more sluggish release.

The Manual If there's one thing that annoys me about the 200e, it's the manual. At just 22 pages of loose-leaf text, this is — like the synth itself — densely packed, with a surprising amount of information in such a small space. But it offers no help to the novice, and even expert users will have to try to work out what's happening. You don't expect to have to take a voltmeter, oscilloscope or spectrum analyser to your $20,000 synthesizer just to find out how to use it, nor should you have to!

291e Triple Morphing Filter The 259e, 292e and 281e module together provide superlative waveform possibilities and flexible contouring, but very basic filtering, so it's no surprise to find that the 200e has another filter module. As its name suggests, the 291e (shown overleaf) contains three digitally controlled analogue band-pass filters, with control over centre frequency, amplitude and bandwidth, the last two of which imitate the resonance of traditional band-pass filters, but without self-oscillation. You can pass signals independently to the A, B and C filters (not to be confused with the busses of the same names) and treat the 291e as three (mostly) separate filters with independent outputs. You can also pass the same signal through all three filters using the All input, using the 291e as a three-band formant filter. If that description seems straightforward, the reality isn't. Notwithstanding the method of selecting the nodes to edit them, and of keeping track of which filter is doing what to which signal, the 291e is a fiendish module that some are going to love, and others will hate. This is because each filter offers eight snapshots that you can jump between (for sample & hold-type effects) or morph between (for dynamic filtering effects). Numerous ways of moving between these stages are provided, and the ability to sequence filter parameters is interesting. But if you take a step back and analyse what's happening, the amount of control that you have over each filter — frequency, width, amplitude, and step time — is little more than you can obtain by applying CVs to the centre frequency and resonance of a conventional band-pass filter. Even the various


Buchla 200e

morph modes — one-shot, looping and so on — are nothing more than you can obtain by applying appropriate CVs to the traditional filter. Perhaps in deference to this, each of the three filters in each 291e has CV inputs plus global modulation inputs (which are, strangely, on 3.5mm sockets) and these CVs can be directed to any combination of frequency modulation, bandwidth modulation and amplitude modulation. You can achieve interesting effects by combining the internal morphing with external control (and, in particular, voltage control of the morphing!) but I'd have to question how musically valid the results are. I have three more points to make about the 291e. Firstly, the maximum filter frequency is quoted as a little over 4kHz. You can't view this in the same way as a low-pass filter with a maximum cutoff of 4kHz, but it still places constraints on the range of effects that you can obtain. Secondly, the quantisation of the filter frequencies is clearly audible when you control them using the Freq knob, and the only way to fine-tune the filters is to apply static CVs. Thirdly, two aspects of the 291e escape me even now: how to use the 'expand input', and how to adjust the individual stage times for each filter. The manual states that you can do these things, but it doesn't tell you how! Making a pair of formant filters the primary sound shapers in the 200e is a bold move, and in some areas, it extends the synth's palette far beyond what traditional highpass and low-pass filters can achieve. But you can't live forever on the esoteric, and sometimes sausages and mash is preferable to the finest gourmet cuisine.

259e Waveforms

The waveforms reproduced in this box give you some idea of how far removed the sound generation in the 200e is from that of an analogue synthesizer. The first (top left) shows the unadulterated sine wave obtained from either the green or red channels of a 259e Principal Oscillator in position 1. The next (top right) shows the output from file:///F|/SoS/SoS%2001-2006/buchla200e.htm (7 of 14)12/19/2005 10:18:57 AM

Buchla 200e

the green channel, in position 4, with the Warp control set to 0. The considerably different third waveform (bottom left) shows the same output, this time with Warp set to 10. These last two were measured at the same audio frequency, although the perceived octave differed depending upon the strength of the harmonics present at different Warp settings. The final waveform trace (above) shows the sawtooth wave output by the Mod Osc. Though this seems like a perfect sawtooth, the square wave from the Mod Osc is very unlike an ideal square wave; it generates numerous additional enharmonic components, and sounds harsh and metallic as a result.

260e Pitch Class Generator The 260e (shown opposite) is perhaps the oddest module in the 'e' family. It comprises two 'Pitch Class Oscillators' that generate the same pure note (a digitally generated sine wave) in all the octaves of the audible spectrum. You can modulate the pitches of these using standard CV inputs and audio FM inputs, each with its associated Amplitude control. Below these lies an in-line five-band graphic EQ, and the output from this provides the raw material for the mis-named 'Escher's Barber Shoppe'. Mixing two metaphors — the rotating barber's pole that continually spirals upwards but never ascends, and MC Escher's visual paradox of the never-ending staircase — this takes the equalised pitch class signals and generates a number of audio paradoxes from them, the most famous of which is the Shepard Tone. This tone, which sounds like a rising (or falling) pitch that never actually climbs (or descends) has been used on numerous recordings, but it strikes me as strange that anyone would use an expensive module location (and, for that matter, an expensive module) to obtain it. Having said that, the 260e is elegant, and it offers numerous alternatives on the same theme, including continuous glide, chromatic glissando with up to 24 divisions per octave, and variable rate. The only restriction appears to be that you cannot use the FM or pitch CVs simultaneously with the barber pole effect. One other facility deserves mention; when set to either of the quantised modes, pulse outputs fire on each pitch step. You can use these pulses to control other modules (such as contour generators) which can then further modify the basic effects being generated. It's all very flexible. Unfortunately, the review 260e had a fault; it created a thump at the end (or start, depending upon how you look at it) of every cycle, rendering it useless for its intended function.

266e Source Of Uncertainty When I first saw the name of this module, I wondered why a noise generator could not simply be called a noise generator. But the 266e has four noise and sample & file:///F|/SoS/SoS%2001-2006/buchla200e.htm (8 of 14)12/19/2005 10:18:57 AM

Buchla 200e

hold sections, and far more control than is offered by any conventional noise generator, so its name is justified. The uppermost section is the simplest, simultaneously offering white, pink and red noise. Below this, the Fluctuating Random Voltages provide two channels (A and B) of unquantised, low-speed fluctuations, with the rate of change of each affected by dedicated knobs and CV inputs. Next come two channels (C and D) of Quantised Random Voltages. These require pulses to change state, and you can determine the number of states from 2 to 24 as well as the statistical distribution of the randomness obtained. To describe the voltage randomness in the simplest terms, a flat distribution means that all states are equally likely, so extreme voltages are just as likely as small ones. At the other end of the scale, a 'bell' curve means that the likeliness of a state is (roughly speaking) inversely proportional to its deviation from the centre, resulting in a 'tighter' sounding range. The time distribution seems to do a similar thing in the time domain, determining how likely a given state is by considering how long it has been since it last occurred. I should mention that the outputs at C and D are different from one another, even though the parameters controlling them are common. The fourth and final panel in the 266e is called 'Stored Random Voltages with voltage controlled probability distribution'. This is similar to the Quantised Random Voltages, but the voltages are not quantised, and you can determine the maximum spread of the output voltages, the distribution, and the degree of skew from favouring low voltages to favouring high ones. In short, far from being a footnote in the 200e family, the 266e is an excellent module that offers far more than you might think.

210e Control & Signal Router Up to this point, I've said little about patching the 200e, but this is a hugely important aspect of its operation, and the strongest weapon in its armoury is the 210e. This has two, independent sections (audio and CV) and allows you to make up to 80 connections using two 5x8 matrices. To use the CV section, you connect the CVs that you want to use to the eight inputs on the left of the module. Second, you connect the five outputs to the destinations of your choice. Having done so, you can make a connection between an input and an output (say, #5 input to #3 output) by stepping across and down the matrix and then turning the Level knob once you have reached the desired 'co-ordinate'. Not only is the signal routed as you wish, but you can scale it from minus infinity to unity gain. You might think this would save on patch spaghetti, but it doesn't. In fact, you need more cables than you would if you patched directly from each source to each


Buchla 200e

destination. However, it allows you to direct one CV to multiple destinations without the dreaded voltage droop that occurs when you stack banana plugs one upon the other. Better still, the routings and levels within the 210e can be stored as part of a 225e Preset so, with a bit of luck and lots of forward planning, you can make all the connections for a particular sound at the touch of a button. Buchla's marketing blurb states that if you route more than one input CV to a single output, the 210e acts as another logical 'Or', selecting the highest voltage and transmitting this to the destination. But my tests showed that the output was the sum of the scaled inputs, and this is far better, because it allows you to sum CVs without side-effects. The audio section works in the same way, except that the maximum gain for each connection is +10dB rather than unity. The fact that you can mix scaled audio signals means that the 210e not only acts as a powerful router, but as an even more powerful 'matrix' mixer. This allows you (for example) to direct one percentage of signal A and another of signal B to output 1, while at the same time directing a third percentage of A and a fourth percentage of signal B to output 2. Excellent stuff!

227e System Interface The final element in the audio path is the 227e (shown opposite), a mixer and output module that allows you to position your sounds in a quadraphonic soundfield. Each of the four primary channel inputs (1, 2, 3 and 4) can be mixed to the four outputs (A, B, C and D again!), each of which has dual 3.5mm outputs at the top of the module, as well as a quarter-inch output on the back of the boat. The 227e provides dynamic panning from left to right and front to back, as well as 'Swirl', which rotates the signal clockwise or anticlockwise, and allows you to determine the amount of channel separation so that you can control the amplitude of the effect. Master volume controls are provided for the front and rear pairs, as are two-channel EQs. These provide a maximum of ±15dB of gain at either extreme, and also allow you to tilt the overall spectral response by up to 12dB in favour of high frequencies, or up to 18dB in favour of low frequencies. A further four inputs (A to D again!) exist in a separate sub-mixer. You can direct these straight to the four primary outputs, whereupon 1 and A are summed, 2 and B are summed, and so on. You can also file:///F|/SoS/SoS%2001-2006/buchla200e.htm (10 of 14)12/19/2005 10:18:57 AM

Buchla 200e

direct the mixed output to the destination of your choice by patching. There are two further facilities: a stereo headphone output that allows you to monitor the front or rear pair, and a mic preamp with an XLR input mounted on the rear of the upper boat. This offers three gains — 10dB, 25dB, and 40dB — and in addition to line-level outputs, has an envelope follower that generates a standard CV.

249 (DArF) Buchla & Associates describe this module (shown overleaf) as 'two multi-segment function generators drawing from a parallel database', which is in itself enough to discourage purchasers. OK, so it's a complex and sophisticated module, but why not call a sequencer a sequencer? Put more simply, the 249 DArF (Dual Arbitrary Function Generator) provides two rows of 24 steps, each with two programmable pitch CVs and pulse outputs. However, despite the claims that the four CVs generated by the sequencer are internally connected to the four 259e oscillators, I can find no way to make the module in this system drive them without patch cords. This is irritating, because the pitch tracking of a 259e when responding to its CV inputs is less precise than when it's driven by the internal busses. Given the lack of the letter 'e' in the 249 name, I wondered whether a module from the original 1970s Series 200 had been installed by mistake, but there was no 249 back then, only a 248, and it looked nothing like this. It beats me. Overlooking this, you'll find that programming simple sequences using the centre section of the 249 is relatively straightforward. You can determine the pitches of each pair of CVs, determine the duration of each step, and create loops with a defined number of repetitions. You can ask any step in the sequence to glide from one value to the next, and to jump to any other step, either as an absolute value or relative to where you are in the sequence. You can even set a probability of a jump occurring, thus creating quasi-random sequences using the determined notes. The true complexity of the 249 starts to become apparent when you invoke the Stage Select, Status and Time Scaling panels to each side of the main section, the External Inputs at the bottom, and the more esoteric logic and timing functions. You can do things such as enable steps only when a pulse is present at an appropriate input, or only when the pulse is absent, or use the external CV inputs to determine the pitch and timing, or as multipliers for other pitch and timing values... and so on. Most confusing, perhaps, is the Stage Select, which provides numerous ways to force a sequence to a particular step. Of these, the one I found most intriguing was the X/Y file:///F|/SoS/SoS%2001-2006/buchla200e.htm (11 of 14)12/19/2005 10:18:57 AM

Buchla 200e

option, which allows you to apply one CV to move a sequence 'vertically' and another CV to move it 'horizontally'. In doing so, you can create all manner of cyclic, discontinuous sequences, some of which appear to be random whilst actually operating to well-defined rules. If you're wondering what all this is for, I found an interesting use in synthesizing the character of a picked guitar. I set all the notes in the sequence to those of a six-note chord and then used the X/Y inputs to fire the steps in different orders. By moving some of the notes onto different CV rows and outputting them to different oscillators, I could envelope the sounds in interesting ways and recreate the feel of strumming. I then extended the idea by using the output pulse at the end of a given number of repetitions to demand a new preset from the 225e Preset Manager, thus changing chords and the voicing of some or all of the destinations while the sequence was playing. The possibilities were enormous, but I'm not sure whether it was worth it, because the amount of work involved was horrendous. There's much, much more in the 249, and some users are going to love it. But I fear that it crosses the boundary from musical instrument to educational tool. You may feel differently, but I'm prepared to bet you're not going to sit down in front of the 249 and bash out a quick sequence the first time you use one.

The Sound Of The 200e If you think that $20,000 is going to buy you the equivalent of five Moog Voyagers (let alone 10 vintage Minimoogs, or 60 second-hand SH101s) you're in for a big disappointment. In fact, it won't buy you the equivalent of one of these, because that's not what Buchla synths do. Patching what you might consider to be a typical analogue lead synth sound on the 200e, a task which would take me a minute or so on a conventional analogue synth, took about an hour, because the 200e simply isn't designed to produce those kind of sounds. This tells you something important; if you want conventional synth sounds, buy a conventional synth! But on the other hand... when I took the patch cord out of the Mod Osc and stuffed it into the output of the Principle Oscillator alongside it, I immediately obtained gritty, harsh, PPG-esque timbres that would be impossible to obtain from a conventional analogue synth. Adjustment of the wave position and the amount of Warp generated all manner of excellent sounds, ranging from almost acoustic to almost percussion, to almost analogue, to almost something you've never quite heard before. Now the situation was reversed, and I was obtaining serendipitous sounds that — even if possible — would have taken forever to patch on a conventional modular synth. While I was doing this, I discovered that I preferred the results if I disconnected the signal passing through the morphing filter, connected another 259e (or two, or three) and experimented instead with dynamic control of the various Warps and Morphs. This made me realise something very important about the underlying philosophy of the 200e. The best way to approach it is to forget the conventional VCO/VCF/VCA model of analogue synthesis, and to start to think in terms of harmonic modulation, file:///F|/SoS/SoS%2001-2006/buchla200e.htm (12 of 14)12/19/2005 10:18:57 AM

Buchla 200e

waveshaping and mixing, rather than filtering. In the Buchla universe, the absence of resonant low-pass filters is not a problem (any more than it is, say, on a Synclavier II), and we can view the triple filters as powerful effectors rather than fundamental components of the signal path. Casting aside the acquired skills and preconceptions learned over 30 years of programming and playing modular analogue synths isn't easy when you're sitting in front of a modular analogue synth, but once I had accomplished it, the 200e and I finally started to make friends with one another.

Pricing As explained last month, there is no set price for the 200e, because it is a modular system. However, interested parties should be aware that the US price for the 200e system reviewed in SOS is a shade under 20 thousand dollars — US $19,850, to be precise. UK distributors RL Music do not quote sterling prices for the 200e, and so the exact cost fluctuates with the sterling/dollar exchange rate. A module-specific price list in dollars is available on their clear, detailed web site, www.rlmusic.co.uk. However, these prices do not include UK customs duty, which is payable on the system, nor the cost of transporting any Buchla modules or systems you purchase from California to the UK, nor the UK VAT at 17.5 percent on all of those costs. At the time of going to press (late November 2005), $19,850 is worth about £11,500, but don't forget, that excludes shipping, duty and UK VAT. Incidentally, in case we gave anyone the wrong impression last month, the prices quoted for the 201e6 and 201e18 cabinets (700 and 1400 dollars respectively, or about 400 and 800 pounds without shipping, UK duty, or VAT) were the prices for the empty cabinets, not for the cabinets filled with modules as in the SOS review system.

Price & Prejudice It's impossible to review something costing $20,000 without being aware of the huge amount of alternative equipment that this could buy. But is the 200e really that expensive? If you carefully consider what it might cost to purchase a modular synth with similar features from elsewhere, as I have done with a number of other modular manufacturers, the Buchla can almost seem cheap. But such comparisons are hard to make, not least because there are so many features in the 200e that have no close equivalents in any other manufacturers' systems. Furthermore, the sound and character of any alternative system will be totally different from that of the Buchla. There's also no sensible way to place a value on the amazing portability and convenience of the 200e, nor on the immediacy of alternatives from Analogue Systems, Doepfer, MOTM, or whomsoever. Given the feature-count in the 200e, it seems almost impertinent to ask if anything is missing, but it is a valid question. I'm not going to cry out for a classic Moog filter — that simply isn't part of the Buchla model — but the lack of inverters is a pain and, while the 266e is excellent, I think the synth would benefit from at least one genuine Sample & Hold (to be fair, you can force the 249 to act as a S&H, but that's an extremely expensive way to obtain a basic facility).


Buchla 200e

I feel that the 200e would also benefit hugely from a CV converter that produces precise 1V-per-octave pitch CV inputs and outputs for interfacing with other analogue synths, but since this is not on the horizon, the thing that I would add to the review configuration is another 210e Signal Router. The mixing and patching facilities of the one already installed proved to be very useful and, happily, the 225e will support two of them. Ultimately though, all this speculation is pointless, because there's nowhere left to squeeze anything in, unless you dispense with existing modules or purchase yet further Buchla boats and modules.

Conclusions The 200e is a highly unusual synth, born of one man's creative vision, and his unswerving refusal to embrace commerciality. As such, it commands great respect. But I suspect that that will be irrelevant to some potential owners who will view the 200e as a status symbol or a piece of technological art. Others will see it as an object of ultimate synth lust. I belong to neither camp, and I'm not afraid to stand up and say that the 200e is not the right instrument for me. I accept that even given the months I've been using it, I've had it for too limited a period to get to grips with it fully, and I know that it's still unfinished in one or two areas, but I find nevertheless that it stands between me and my musical ideas instead of enhancing my creativity. Maybe if I were less deterministic in my sound programming, less conventional in my composing, or if my ideas were triggered by new and interesting sounds, I would feel differently. But however you view the 200e, it's clear that it's a unique proposition: truly one of a kind. It's equally apparent it's going to stir up strong emotions for and against it. Here's my final thought. If you want the Buchla experience, you need a Buchla, and the only Buchla in production is the 200e. It's as simple as that. Published in SOS January 2006 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



CAD GXL Series

In this article:

CAD GXL Series

Construction & Specifications Condenser Microphones Options & Pricing Published in SOS January 2006 Performance

CAD GXL Series

Print article : Close window

Reviews : Microphone

pros Affordable. Comprehensive kit contents, including shockmount and pop shield. Good basic sound quality.

cons No cons as such, though there are lots of very similar mics out there vying for your money.

This new set of Chinese-built mics offers good-quality vocal and instrument recording on a budget. Paul White

CAD have carved out their own niche in the These CAD mics offer a cost- microphone world over the past few years with some very fine mid-priced studio mics. However, effective means for any project studio owner to get with these new Chinese-built models they seem to kitted out to make high-quality be trying to take a share of the entry-level market vocal and instrumental as well. The mics are available in a number of recordings. The kits are also good value and, although you different kits (complete with shockmount and pop shield) or individually. don't get any fancy camera

summary

cases with the kits, you do get shockmounts and pop shields as standard.

information See 'Options & Pricing' box. Unity Audio +44 (0)1440 785843. +44 (0)1440 785845. Click here to email www.unityaudio.co.uk www.cadmics.com

Construction & Specifications The smallest of the three mics under review here Photos: Mark Ewing is the GXL1200, a conventional end-fire 'stick' mic featuring a cardioid capsule a little over half an The GXL-series mics under review (clockwise from top left): inch in diameter, a FET preamp, and a GXL3000, GXL2200, and transformerless output stage. Its response is nominally flat from 30Hz to 20kHz, with very slight GXL1200. The shockmounts and pop shield are included dips in the 3-7kHz region and a modest presence with the GXL-series kits. peak at around 9kHz. The quoted -56dB sensitivity and equivalent noise figure of 14dBA are slightly better than average in this UK price range. Phantom power is required, but can be either 24V or 48V. The casing is nicely machined and the capsule itself may be unscrewed, suggesting that alternative capsules may become available. The GXL2200 is a side-address cardioid capacitor mic with a one-inch centre-

file:///F|/SoS/SoS%2001-2006/cad.htm (1 of 4)12/19/2005 10:19:16 AM

CAD GXL Series

terminated diaphragm, a FET preamp, and a transformerless output stage. The frequency response is from 30Hz-20kHz, with a few ripples below 200Hz or so and a 1dB presence peak mainly in the 12kHz region. Both the large-diaphragm mics require 48V phantom, and they also share a maximum SPL figure of 135dB and an equivalent input noise of 20dBA, the latter a little on the high side for modern mics, but still not a cause for concern in close-up studio applications. The GXL3000 is a large-diaphragm, side-entry mic designed primarily for studio vocals, though, as with most such mics, it also does a fair job on most instruments. Its one-inch centre-terminated capsule has a switchable polar pattern, giving cardioid, figure-of-eight, or omni variants. Unscrewing the end cap allows the body cover to slide off revealing the familiar 'everything fixed to two rails' Chinese internal structure — there's nothing unusual in here, but also no cause for concern as regards mechanical integrity. The metalwork seems to be acoustically quite well damped too, which can be important in keeping the tone as clean as possible, and the basket has a dual-layer structure to improve its effectiveness as a wind shield and electromagnetic screen. On paper the mic has a 20Hz-20kHz frequency range, with modest presence peaks at around 6kHz and 12kHz. Low cut can be engaged to compensate for the proximity bass boost when the mic is used close up in cardioid or figure-ofeight modes, and a 10dB pad is available for when you're recording very loud sources that might otherwise overload the mic preamp. Overall, the mic weighs 602g, which is substantial without being too challenging for the shockmount swivel joint.

Options & Pricing GXL1200, £60.76.

Small-diaphragm cardioid condenser mic. GXL2200, £72.62.

Large-diaphragm cardioid condenser mic. GXL3000, £169.94.

Large-diaphragm multi-pattern condenser mic. GXL2200 Studio Pack, £138.56.

One GXL1200 mic with mounting clip, one GXL2200 mic with shockmount, and an EPF15A pop shield. GXL2200 Stereo Studio Pack, £193.25.

A pair of GXL1200s with mounting clips, one GXL2200 with shockmount, and an EPF15A pop shield. GXL3000 Studio Pack, £222.66.

One GXL1200 with mounting clip, one GXL3000 with shockmount, and an EPF15A pop shield. GXL3000 Stereo Studio Pack, £276.21.


CAD GXL Series

A pair of GXL1200s with mounting clips, one GXL3000 with shockmount, and an EPF15A pop shield. All prices include VAT.

Performance Despite the noise figures of the two large-diaphragm mics, none of these mics generated any noticeable noise during normal use, and their tonal qualities also didn't disappoint. The GXL1200 has a fresh, open sound that works well with just about any instrument, though it also produces a nice vocal sound when used with a pop shield. It's good on acoustic guitar, but seems to be able to wring detail out of just about any instrument, especially percussion. As a vocal mic, I rather liked the no-frills GXL2200, as its more pronounced presence peak is high enough to create a sense of air and intimacy without making the upper mid-range sound harsh. It adds a nice gloss to vocals and also does a creditable job on acoustic guitar. Used fairly close, the proximity effect gives a plausible 'radio DJ' character to the low end of the voice. Although there is nothing much to differentiate this mic from dozens of similar Chinese-built models, it's certainly capable of making excellent-sounding recordings. The multi-pattern GXL3000 sounds pretty neutral, which is perhaps what you'd expect after seeing the nominally flat frequency plot with its subtle presence peaks. The omni mode sounds more open and natural than the cardioid and figure-of-eight modes, as is to be expected, but overall the sound is smooth and fairly classy with no obviously hyped characteristics, making this a good choice of mic if you need to use it as an all-rounder with different singers and different instruments. While these mics may differ little from other Chinese-built models, they perform very well within their price range, and the kits are a nice idea, as they provide everything you need to get going, other than mic stands and cables. The kits offer plenty of flexibility for vocal and instrument recording, though the GXL3000 kits are clearly more versatile because of the GXL3000's ability to switch patterns. The GXL3000 is also the most natural sounding of the mics, though the GXL2200 has a nicely flattering quality to it especially at the high end. I'd have no qualms about using any of these mics to handle a serious recording project, but as always I'd first ensure the mic suited the singer. Project studio owners today are very fortunate in being able to buy such credible microphones at such affordable prices, from a variety of manufacturers, and though you still get better quality when you pay out more money, the difference isn't always as large as you'd expect, and in many cases the recording conditions have a greater influence than the specific microphone used. Published in SOS January 2006


CAD GXL Series

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




In this article:


Box Clever Modelled Analogue Synth Triggering Published in SOS January 2006 The Sound & The Signal Path Print article : Close window Editor/Librarian Software Reviews : Keyboard Memories Are Made Of This Pretty Poly MIDI Effects Conclusions Creamware's Minimax

Creamware Minimax ASB £599 pros Minimoog sounds and layout. Extensive MIDI control. Exceeds the capabilities of the original Mini, offering memories, polyphony and effects.

is certainly not the first digital emulation of a Minimoog to be released — nor even Creamware's first. But it bucks the recent trend for software recreations of vintage synths — by being hardware! Can it replace the real deal? Paul Nagle

What can you write about the Minimoog that hasn't been written a cons hundred times before? Well for a start, Software needs a bit of a that it now has a hardware-based DSPpolish. driven dopplegänger. Awaiting No headphone socket. attention in my studio today is a small, External power supply. wood-framed desktop module with a Not rackmountable. naggingly familiar layout. Creamware's Minimax ASB ('Authentic Sound Box') summary is a digital reincarnation of the '70s For me, the Minimax is a monosynth, and loses the keyboard more-than-adequate Photos: Mark Ewing substitute for a real Mini — and left-hand controllers, but gains especially live. The MIDI MIDI capabilities, memories, effects implementation is a delight and even polyphony, while retaining many quirks of the original design — so and the addition of effects and polyphony a real bonus. A few there's no dedicated LFO, and no oscillator sync or pulse-width modulation, as found on some modified Minis. All this comes at a price considerably less than tweaks to the software will make it better still, but even that of a second-hand Minimoog — and I should know, as I sold mine a few as it is, the Minimax is years ago. If Creamware have nailed the sound, this module could find itself in reasonably priced and really great demand. And the company have been modelling the Mini for years, with the rather tempting. Miniscope and Minimax plug-ins for their Pulsar and SCOPE systems and later information for their Noah hardware synth, so you'd expect the Minimax to be a highly £599 including VAT. evolved emulation. Sonic8 +44 (0)8701 657456. +44 (0)8701 657458. Click here to email

file:///F|/SoS/SoS%2001-2006/minimaxasb.htm (1 of 8)12/19/2005 10:19:20 AM

Box Clever


www.sonic8.com www.creamware.de

Test Spec Minimax OS OS version reviewed: v00000100 (as stated in the Remote software when upgrading).

As you can see above, the Minimax's panel closely resembles that of the Minimoog, with sections labelled Controllers, Oscillator Bank, Mixer and Modifiers, although there's no headphone socket nor a A440 tone generator (the latter, at least, we shouldn't miss, thanks to digital stability). In use, the knobs are smooth, responsive and feel right. The minimal visible additions include two knobs that add velocity control over both envelope amounts, plus a Feedback switch to route the synth's output back through the filter section, permitting Minimoog-style overdrive. Underneath the main panel is the Configuration Strip, where more modern features are accessed. Here you assign the MIDI channel and access the effects, patch memories, and polyphonic mode. There's also a button that transforms the data-entry knob into a master volume. You'll note there is already a dedicated volume knob on the panel, but this is a separate, programmable parameter stored with each patch. The three-character LED display is basic but adequate (although there's no MIDI indicator LED), and all the lower selector buttons are of the LED-bearing, positive-clicking sort. The rear panel is very different to that of a real Mini — there are no voltage inputs, S-Trig connectors, or scaling pots for each oscillator (tweaking these was a regular hassle on my old Minimoog). The Minimax has MIDI In, Out and Thru, plus a USB socket. Don't be misled by the Minimax's twin audio inputs and outputs. The Minimoog had a completely monaural signal path, and so does the Minimax. Its oscillators, noise and any external signals pass through a single low-pass filter, so the two audio inputs are summed to mono internally, and the two outputs are only stereo in the sense that twin delay lines are hardwired to the right and left outputs. Finally, the power supply is external. I know these keep costs down, but as a live player, I hate them and as long as companies keep using them, I'll keep complaining!

Triggering The original Minimoog used single triggering with low-note priority, meaning that envelopes were not retriggered when you played legato. This gave rise to fluid solo playing styles, but could occasionally be a nuisance when you required articulation that was both fast and precise. The Minimax solves this dilemma by enabling single triggering as its default, but you can specify multiple triggering and last-note priority with the aid of the Remote editing software or via MIDI control changes. High-note priority is not available, though.

The Sound & The Signal Path The Minimoog, of course, had no dedicated LFO, although you could create the file:///F|/SoS/SoS%2001-2006/minimaxasb.htm (2 of 8)12/19/2005 10:19:20 AM


effect of one by setting Oscillator 3 to a 'Lo' setting. The Minimax works the same way, and equally faithfully, Creamware have varied one of oscillator 3's waveforms, replacing the mixed triangle/sawtooth with a reverse sawtooth (although this isn't shown on the front panel). The reverse sawtooth doesn't sound any different from the forward version, but it offers another, distinctive modulation source. A Modulation Mix knob fades smoothly between oscillator 3 and noise (white or pink) and switches determine the routing of these sources to the oscillators and filter, modulation amount being set by your MIDI controller's mod wheel. Ah, simplicity! Having had a quick look round, I was eager to make some noise. Predictably, I played my first note then reached straight for the Cutoff knob to be rewarded by a smooth, Moog-like sweep. The Resonance (or Emphasis, to use Minimoog terminology) sounded more realistic than the average modelled filter; it was possibly a fraction loud and shrill at its higher settings, but was definitely up there with the best of them, and self-oscillated nicely when cranked up to maximum. And when I lowered the cutoff and applied some pink noise modulation, I was impressed to hear the filter bubbling and warbling in a convincingly Minimoog-like way. Checking out the oscillators, I found the waveforms suitably authentic to the ear, from rich brassy sawtooth through to reedy pulse wave. When detuned, they exhibit the nice 'swimminess' of Minimoog oscillators, and authentically, As on the Minimoog's rear panel, there are they never quite lock together. I found audio outputs and inputs, but then come the the basic tones to be fairly bright, so modern additions — the traditional five-pin I'm guessing Creamware modelled one trio and USB connectors for MIDI. of the later Minimoogs. At high pitches, the waveforms are clean and free of artifacts, whilst lower down, there's a fullness sometimes lacking on virtual analogues. After finding the filter and resonance knobs completely smooth in operation, I was surprised to hear stepping on the oscillator frequency and patch volume controls, but most noticeably on the Amount Of Contour knob in the Filter section. Unsettled, I turned to the envelopes — and instantly realised something was wrong. The manual claims that these are 'modelled on the Original's behaviour', but they're not. Certainly, as on the Minimoog, the Minimax's envelopes only have three stages, with a switch determining whether release is off or is shared with the decay time. But if you play a note with a long decay/ release, and then shorten the time, instead of changing instantly, as on a Minimoog, the note still continues to its original length on the Minimax, the change only becoming apparent when you retrigger the note. All the envelope stages behave in the same way. Apart from this, I found the envelopes to be reassuringly snappy, capable of delivering the punchy basses and percussive sequences we all know and love.



There were a couple more noteworthy niggles. Despite sending loud signals to the audio inputs, I was unable to activate the yellow overload LED, and became convinced I had a unit with a hardware fault. Not so, Creamware assured me: this LED isn't yet enabled in software. Finally, I found that the Glide time was stepped, functioning more like glissando than the Minimoog's smooth portamento. When I reported this, Creamware claimed there was already a fix, and so there was — although it wasn't yet available from their web site! This process introduced me to Creamware's idiosyncratic way of dealing with OS updates (see the box overleaf).

Editor/Librarian Software A CD in the Minimax box contains the Remote software, which offers a visual display and editing environment for your patches, as well as giving complete access to all effects parameters and acting as a librarian and OS updater. Currently it is available for Windows XP only, although a Mac version is in beta testing. Prior to installation, you are asked to connect the Minimax to your PC via a USB port. Windows XP Service Pack 2 is recommended, so that the system will recognise the Minimax automatically — I had no problems with my XP-based PC. Once installed, XP 'sees' the Minimax as a generic USB Audio Device, although strangely, the USB cable doesn't carry audio at all, only MIDI! Incidentally, beware of sending notes to the MIDI In and the USB port at the same time; the Minimax gets very confused... If the driver install was painless, my initial efforts to run the editor were anything but; on my studio PC, the installer program crashed every time. Eventually, Creamware had to send me a version that didn't require installation — I just copied it into a directory and it worked. The editor has a clear, intuitive user interface divided into two halves: synth and effects. The synth graphic even has all the oscillator waveforms labelled correctly, unlike on the front panel! As you already have a knobby interface on the synth itself, I expect the main use for this program will be to tweak effects parameters and to store patches. You can even use it to wipe the Preset patch bank, and replace it with your own. Remote is also used to add OS upgrades. Unfortunately, you have to use this software to upgrade your OS, so if you have a Mac or don't have access to a recent PC, you may wish to delay your purchase until the OS is perfected. The OS update process is, frankly, more complex than is necessary. First you get the software running; this extracts the serial number from the Minimax. You then email this number to Creamware Support, and wait. When they email you back, it is (in theory, at least) with an activation code and with the OS itself, ready to be loaded into the software. This process leaves users of older computers and hardware sequencers out in the cold — why can't we just download a SysEx file from a web site and load it over MIDI in the usual way? file:///F|/SoS/SoS%2001-2006/minimaxasb.htm (4 of 8)12/19/2005 10:19:20 AM


Memories Are Made Of This Of course, the original Minimoog had no memories, but the Minimax offers 128 preset and 128 user patches. When selecting user patches, the Preset LED initially remains lit. To see the value of any control, you push the Preset button until its LED goes out, and then, when you turn a knob, the small row of yellow LEDs in the Match display at the bottom left indicates how far away you are from the stored value. When the middle LED lights, you're close, and when it flashes, you're there. LEDs to the right and left show you're moving away from the stored value. Values can be fine-tuned with the Up/Down buttons to the right of the display, or the data knob to the left. To return to patch selection mode, you hit the Preset button again. If this method doesn't appeal, you can hold down the User button for a couple of seconds. All the front-panel controls then become 'live' — my preferred method of working. I think programming a synth like this is more of a personal experience than, say, extracting pianos and violins from a modern workstation. Certainly the factory patches, mostly typical analogue sounds of the late '70s or early '80s, didn't do much for me, but there were sufficient polysynths, organs, leads and basses to serve as starting points. Soon, I had assembled a selection of warm, glide-based solo sounds, sweeping basses, subtle brass and all manner of thick, fluid, resonant leads. The Minimax handled like the most stable, flawless Minimoog I've ever played: there were no glitches or instabilities whatsoever. I particularly liked the filter overdrive; this is activated by switching on Feedback in the Mixer section, setting the external input to 'on' and cranking up its level. The result is a very creditable impersonation of the famous Minimoog trick of feeding the headphone output to its own audio input.

Pretty Poly Holding down the Up and Down value buttons together activates the Minimax's polyphonic mode, with the display showing 'on' when active and 'of' (!) when mono operation is restored. Curiously, though the manual claims that it is sixnote polyphonic, the Minimax can play 12 notes at once, although there is no attempt at offering multitimbrality. Polyphony requires that you approach the Minimax differently — patches created to sound fat and raspy in mono mode will sound excessive when you play chords, using the Feedback button to overdrive the signal gives pretty rough results, and at high volumes, nasty distortion results. Programming 'thinner' patches with subtle oscillator detune proved most effective, and I discovered a wealth of useable organs and strings, plus some stunning synth brass. Although I missed having pulse-width modulation, I thought the Minimax performed beautifully as a polysynth, with a presence and depth that few digital analogues can offer. However, voice stealing can occur unexpectedly when playing two-handed, since each voice appears to be allocated in the order it's played. This means that notes are preserved until released, regardless of new file:///F|/SoS/SoS%2001-2006/minimaxasb.htm (5 of 8)12/19/2005 10:19:20 AM


notes that may be played. This can be a good thing: if you program a patch with a long decay/release, you can hold a left-handed chord and then solo non-legato with your right, retriggering the same voice cleanly with each note, rather than creating a solo 'mush'. I rather liked this implementation, although Creamware tell me that polyphonic voice allocation is something they intend to return to in a future OS update.

MIDI The Minimax's MIDI implementation is extensive. Elsewhere in this review, I've noted the velocity control of filter and amplifier envelopes, with dedicated Amount knobs for each. This feature adds a sense of dynamics that's taken for granted on modern synths, but which can breathe fresh life into analogue timbres. In addition, there are several parameters that are not accessible via the ASB hardware. Of these, mod wheel depth and offset are probably the most valuable, since they can be employed to create patches with constant modulation effects. Aftertouch can be routed to filter cutoff, output level or to oscillator pitch, but, sadly, it can't be used to control modulation depth. Pitch-bend range is also programmable per patch. The best news is that every Minimax parameter has a MIDI Continuous Controller assigned to it — even those for which there are no physical controls. You can activate polyphonic mode, set every effect parameter, or change the envelope triggering response, all using simple MIDI CCs. As you'd hope, the knobs all transmit these, and there's even a dedicated Local Off switch to let you do so without altering the sound engine. This is great if you like controlling everything over MIDI from your sequencer.

Effects The onboard effects are basic but well judged, consisting of just a chorus/flanger and a two-channel delay line. The chorus/flanger allows control over feedback, phase, rate, depth and the wet/dry mix. In flange mode, with feedback set quite high, you get a welcome metallic bite that slices through a mix nicely, and the chorus fleshes out polyphonic string sections nicely. I've always felt that delay is the most valuable effect on analogue synths, so full marks to Creamware for putting not one but two on board. The maximum time available is about 1.5 seconds, and you are always required to set left and right delay times individually. There's no option to sync the effects to MIDI Clock, although you can independently set a tempo from 72 to 199bpm for each delay channel, and you can set the delay clock from 1/64th notes to single notes (including triplets). The manual fails to mention the 'Cross' feature, which alternates the feedback signals from one channel to the other, producing a pleasant stereo-widening effect. Finally, you can bypass the effects altogether with a dedicated switch — and once engaged, the bypass remains on when you select new patches.



There are five preset effect algorithms (referred to numerically due to display limitations), and you can only set the wet/dry mix and the three most important parameters for that preset. For example, you might be able to adjust left and right delay times and damping, but not feedback. However, this didn't bother me too much; I spent five minutes setting up a few knobs on my Novation Remote 25 to perform the full range of adjustments via MIDI CCs, including the effects bypass, and never gave it another thought!

Conclusions My initial impressions of the Minimax were less than positive. Given Creamware's past attempts at modelling the Moog sound, I was surprised to hear audible stepping on some controls, not to mention envelopes that didn't perform like a Minimoog's, and the non-operational overload LED. But as I started to play and program the Minimax, that impression changed. I was able to obtain sounds that closely matched my favourite old Mini patches, and whilst the Minimax sounds didn't have the instabilities you would find on a Minimoog (and therefore some of the magical nuances), I was happy to live with that, and welcomed the addition of velocity control, effects and polyphony. I should also mention that Creamware took on board all the points I raised, and promised to revisit the operation of the envelopes, to add smoothing to more of the controls, and to activate the overdrive LED, although these fixes had not yet been implemented when I submitted this review. I accept that the Minimax ASB will never compete side by side with a real Minimoog in terms of physical presence, and cannot offer the quirks and personality traits that come from an ageing set of analogue electronics. But the longer you work with the Minimax, the less these things seem to matter, and the more you come to value the convenience of a reliable, drift-free module that won't develop crackly pots or a duff keyboard. This was brought home to me when I asked a friend to bring his 1970 Minimoog around to put it alongside the Minimax. Significantly, he decided it was too fragile to risk the journey. The Minimax could be a perfect MIDI-capable solution for those who want to gig with their Mini, but dare not. Alternatively, if you've always wanted a Minimoog, but would prefer one in a reliable, affordable, hardware form, the Minimax could be what you've been waiting for. Published in SOS January 2006






Digidesign Pro Tools v7

In this article:

Time For An Upgrade RTAS Performance Improvements Ch, Ch, Changes Say Goodbye, HTDM RTAS/HTDM Compatibility Regional Variations Round And Round We Go Getting On The Right Track It's The Little Things Keeping It Real (-Time) Off-line MIDI Conclusion

Digidesign Pro Tools v7 Recording Software [Mac/PC] Published in SOS January 2006 Print article : Close window

Reviews : Software

With new MIDI sequencing functionality among many other features, Pro Tools 7 is intended to be the ultimate audio and MIDI workstation. Will this release keep Digidesign on top?

Digidesign Pro Tools 7 pros The improved RTAS support finally gives Pro Tools HD users a solid framework for running host-based plug-ins, and there's a significant performance increase in the new RTAS engine for systems with multiple processors. The new Region Groups are going to save a lot of time when editing a collection of audio or MIDI Regions. Instrument tracks help reduce the number of tracks required on the Edit window and Mixer when running software instruments in Pro Tools. There's some useful housekeeping in this release of Pro Tools, like the new menu structure, and many small improvements, such as a resizable I/O Setup window and the ability to drag Send assignment slots, that will improve the lives of Pro Tools users everywhere.

cons Users of single-processor systems might, in some

Mark Wherry

Digidesign's Pro Tools started out mainly as a hardware platform, with most users opting for alternative host software. Since the release of Pro Tools 5, however, the software element has become more important and is arguably now the core of the system, with Pro Tools' audio recording, editing and mixing tools complemented by comprehensive sequencing functionality. Pro Tools 7 arrives a whole three years after version 6, but Digidesign have released many significant upgrades in the interim. These added key features such as automatic delay compensaton, Track Punch, input monitor buttons, 23.976 frame rate support, MIDI step input, improved RTAS instrument support, enhanced tempo manipulation and MIDI functionality, AFL/PFL solo, better plug-in organisation, and the new EQ III plugin. During this time Digidesign have also released a new Icon family of control surfaces (which we'll review in SOS soon), aiming to give people an integrated alternative to traditional mixing consoles, and Venue, a live mixing console based on Pro Tools technology. In many ways, Pro Tools 7 is the most important release of the Pro Tools software to date. The question is, can Pro Tools really become the only audio and MIDI workstation you'll ever need, as Digidesign are hoping?

file:///F|/SoS/SoS%2001-2006/protools7.htm (1 of 15)12/19/2005 10:19:24 AM


circumstances, be able to run fewer RTAS plug-ins than in previous versions of Pro Tools. MIDI editing can sometimes feel a little clunky (although it's getting better) compared to other sequencers, and there's no way to view multiple Regions in the space of one piano-roll editor simultaneously with Controller data.

summary While there is no single killer feature in Pro Tools 7, Digidesign have instead focused on making the application better for every user, releasing what is overall a must-have upgrade.

Time For An Upgrade Version 7 is the first paid upgrade of the Pro Tools software you can actually buy and download on-line. Another change for version 7 of Pro Tools HD is that the application now requires an iLok authorisation instead of a serial number, regardless of the Digidesign hardware configuration you're running. This stops larger facilities (or even groups of friends) buying one copy of the upgrade and installing on it on a number of different systems, which seems fair enough, and all Pro Tools HD systems already come with an HD Bundle iLok anyway. Pro Tools LE retains the existing serial-number authorisation process, which is good because not every LE user has an iLok — or any free USB ports! The MPowered version for M Audio hardware does require an iLok authorisation, but as it's so new, Digidesign are making the version 7 update a free download for existing users.

information Basic upgrade for Pro Tools 6 users £105 (HD), £45 (LE); free for Pro Tools M-Powered. See Digidesign web site for options. Digidesign UK +44 (0) 1753 655999. +44 (0)1753 658501. Click here to email

If you're reading this before December 20th, it's worth investigating Digi's Upgrade Plus option, which costs $70 more for the HD version and $24 more for the LE version, but allows you to choose two extra plug-ins from a very tempting selection. Those ordering upgrades can order upgrade documentation for an additional $15 plus postage, or a full set of documentation for $30. While these manuals are available for free on-line as PDFs, I like being able to have a printed copy on my desk, especially as Digidesign's documentation is generally pretty well written.

www.digidesign.com

Test Spec Dual 2.7GHz Power Mac G5 with 2.5GB RAM, Pro Tools HD2 and a 192I/O Digital interface, running Mac OS 10.4.3 and Pro Tools HD 7.0. 12-inch Powerbook with a 1.5GHz G4 processor, 512MB RAM, and an M Box 2, running Mac OS 10.4.2 and Pro Tools LE 7.0.

To start with, I installed the 7.0 HD update on my dual-2.7GHz Power Mac G5, which was already running the latest customer service release of 6.9.3. I let the installer run over my existing installation and, other than a new splash screen, the process is identical to installing previous versions of Pro Tools.

RTAS Performance Improvements As mentioned in the main text, with HTDM plug-ins no longer supported in Pro Tools 7, Digidesign have dramatically improved the performance of RTAS plug-ins for both HD and LE users. The engine for running RTAS plug-ins is now multithreaded and fully supports multiple processor cores (for both dual-processor and dual-core systems) and technologies such as Intel's Hyperthreading. To configure this, there's a new option in the Playback Engine Setup window called RTAS Processors where you can set the number of processors to use for RTAS plugins. In Digidesign's tests, the company quote getting a 83 percent improvement when running D-Verb, 60 percent with NI's Battery 2, and 151 percent with EQ III. In order to test this, I did some experiments with three different PT7-optimised RTAS plug-ins from URS (S Series EQ, Fulltec EQ, and the 1970-CLS dynamics plug-in) on my dual-2.7GHz Power Mac G5 with processor usage set to 85 percent in Pro Tools and a buffer size of 512 samples. In Pro Tools HD 6.9.3, I



managed to run 95, 65 and 16 instances respectively of these three plug-ins. In each case, Activity Monitor reported Pro Tools using between 121 and 129 percent of the system, with the User usage between 56 and 72 percent, so despite Pro Tools overloading, it was clear it wasn't taking full advantage of the available resources. PT7 could definitely handle more plug-ins, though with the ones I'd chosen for the test I didn't see the dramatic improvements quoted by Digidesign, achieving 118 instances of S Series EQ, 64 Fulltec EQs and 23 1970CLS plug-ins. Unfortunately, there aren't any improvements for single-processor users, and in fact, Digidesign warn that with certain PT7-optimised RTAS plug-ins you'll actually see a slight drop in performance when compared with earlier versions. I didn't find this to be as severe as it sounds, though: for example, on my Powerbook I could get 34 four-band EQ III instances running in Pro Tools 6.8.1, and exactly the same number in PT7.

Ch, Ch, Changes The first important point to note is that Digidesign have introduced a new '.ptf' file format for version 7 Pro Tools Sessions which isn't backwardly compatible with the previous format. One advantage of this new version 7 format is that it consolidates Mac and Window compatibility into one consistent format, so the PC/ Mac compatibility tick box is no longer there. PT7 will still open older Sessions, and there's an option to save Sessions in a 'version 5.1-6.9'-compatible format if you need to. There are a couple of changes to the mixer in PT7. Fader range is now fixed at the +12dB level which was optional in v6.9, and there's such as support for up to 10 sends per audio channel (split into two blocks of five, so you don't have to waste screen space for sends you're not going to use). There's also support for up to 160 inputs and outputs for audio tracks at 96kHz, a configuration which requires one Core and four Accel cards to support 10 192I/O interfaces — each card supports two interfaces and each interface supports 16 channels of input and output. This follows on from v6.9's support for 160 Auxiliary Inputs, which is vital for large Sessions and especially important to help Icon systems compete with large-format digital consoles that offer a large number of physical inputs and outputs.

The System Usage window now shows the used TDM Voices for Pro Tools HD systems, which is useful to keep track of resources when running a large number of audio tracks and RTAS plug-ins.

Once you have a Session open in PT7, you'll see that, well, it doesn't look too much different from version 6! In terms of the user interface, the appearance is identical to previous versions, and the only major change is a reworking of the file:///F|/SoS/SoS%2001-2006/protools7.htm (3 of 15)12/19/2005 10:19:24 AM


menu organisation. Like many users, I feel a certain amount of dread when the menus are reorganised in an application I use on a regular basis, but in this case Digidesign have actually made it easier to find commands in Pro Tools. For example, rather than having dedicated menus for MIDI and Movie operations, along with track-related commands in the File menu, there are now three menus labelled Track, Region and Event. The Track menu contains commands for creating new tracks, deleting and grouping, along with other track-related functions like Split Mono and Make Inactive. Similarly, the Region menu contains Region-related commands, and the Event menu is where you'll find most of the commands for manipulating MIDI and Tempo Events that were stored in the MIDI menu in previous versions. The Window menu has also been simplified, and there's a new separate View menu that's used to configure the appearance of Pro Tools' windows. To add a movie to a Pro Tools Session, you now use the File / Import / Quicktime Movie command, and the features for configuring how the movie relates to timecode are now in the Setup menu, as is the old Session Setup window. Overall, this is a big improvement, and many of contextual menus have also been cleaned up in a similar way. While the visual change in PT7 is pretty minimal, once you start working with the application, you'll soon notice enhanced functionality in almost every area. Even a basic operation such as the Track / Duplicate command now has options for choosing how many copies of the selected tracks should be made and which settings should be duplicated to the newly create tracks. The Show/Hide List from previous versions of Pro Tools is now known as the Track List, with extra options for organising the List in track-type order, by groups, name, and so on. As before, the order of this List is always reflected in the order of the tracks on the Edit window, so choosing a different Sort By option also affects the Edit window. One other user-interface improvement is the inclusion of 'tool tips', so if you hover your mouse cursor over a control for a moment, a handy tip will appear telling you what that particular control is, or does. A further numerical improvement is that PT7 Sessions now support up to 999 memory locations, up from 200.

Say Goodbye, HTDM Perhaps the biggest architectural change in PT7 is that HTDM support has been discontinued. The HTDM format was originally introduced back in Pro Tools 5 days as a way of integrating host-based plug-ins into the TDM mixer. This was necessary because Digidesign's other real-time host-based plug-in format, RTAS, was more limited in the functionality that could be provided. However, HTDM was never a perfect solution: stability was often a big problem, and as host processors got more powerful, it became possible to run more HTDM instances than would be supported by the HTDM Stream Manager, the program that ran on the Pro Tools DSP cards to stream audio to and from the TDM mixer to HTDM plug-ins.



In PT7, HTDM has been replaced with a new RTAS implementation that is more powerful both in terms of sheer performance with the number of instances you can run (see box above for more information), and the functionality of how you work with these types of plug-ins in Pro Tools. RTAS plug-ins can now be used both before and after TDM plug-ins on Auxiliary Input tracks and audio tracks, and it's now possible to add RTAS plugins to Master Fader Tracks in the TDM version as well, which is a really welcome improvement. RTAS plug-ins You can create multitrack Region Groups also gain full side-chain capabilities in that incorporate Regions across multiple tracks of different types. TDM, and can also be used in the full mono-in/stereo-out configurations. Pro Tools HD users pay one slight penalty for the more flexible RTAS configuration, though, which is that instead of using a DSP chip to run something like the HTDM Stream Manager, RTAS plug-ins now use the same Voices that audio tracks use to transfer audio to and from the TDM Mixer. Every time audio needs to be sent to or from the TDM world, one voice per channel of the plug-in is used (one voice for a mono plug-in, two voices for a stereo plug-in and so on). So if you have a stereo Auxiliary Input track that takes its audio input from a Pro Tools interface and you use an RTAS plug-in, you need two voices to get the audio from the interface to the host processor, and then when the output of that audio gets routed back to an physical output or buss in the TDM world, you need another two voices. This means that one stereo RTAS plug-in on a stereo Auxiliary Input track requires four voices. If you have two RTAS plug-ins in series on the same track, the output of the first RTAS plug-in gets fed into the second without any trips to the TDM world; but if you put a TDM plug-in in between two RTAS plug-ins on a track, now eight voices are required to run the RTAS plug-ins because the output of the first RTAS plug-in has to go back to the TDM hardware for processing by the TDM plug-in and be sent back again to the host processor for processing by the second RTAS plug-in. In practice, this is more straightforward than you might think, and unless you run really large Sessions with a large number of audio tracks, you shouldn't find yourself running out of voices with RTAS plug-ins. One nice touch is that the System Usage window now includes a new section telling you the number of TDM Voices currently allocated, which makes it much easier to keep track of your voice usage. As a final note, unfortunately you still can't monitor RTAS plugins in real-time on an audio track, even with the new RTAS architecture, but you can at least fudge something by using a combination of Auxiliary Inputs, audio tracks and busses.



RTAS/HTDM Compatibility The ability for RTAS plug-ins to be used anywhere on the Mixer is also quite important for opening previous Sessions that used HTDM plug-ins, because when you open such a Session in PT7, the application will attempt to open an RTAS version instead of the HTDM plug-in and convert the settings, which is really handy. Not every HTDM plug-in will convert correctly to an RTAS version, but at the time of writing developers are already releasing versions of their plug-ins optimised for compatibility with PT7. For example, when I installed PT7 with my existing Waves installation, none of the HTDM instances would convert to RTAS instances. However, after upgrading to a PT7-optimised Waveshell, the same Session loaded fine with RTAS instances of plug-ins that were previously HTDM. Many existing plug-ins will run just fine, but some will require new versions. For example, trying to use the TDM version of Sony's Oxford EQ produced an alert telling me 'The plug-in could not be made active because it is not supported in this version of Pro Tools. Please contact the manufacturer for upgrade information.' Interestingly, the Pro Tools 7's Region Groups make it easy to RTAS version worked fine. This turn this... doesn't seem to be a serious issue, though, as developers seem to be releasing compatible versions, and I'm pretty sure most plug-ins will be PT7-compatible by the time you're reading this — before I submitted this article, for example, Sony had already published a Pro Tools 7-compatible upgrade for the Oxford EQ. If in doubt, check Digidesign's 'Plug-in and Software Compatibility with Pro Tools 7' document (www.digidesign. com/developers/plugin_info/grid) and the web sites of plug-in manufacturers. Another area where focusing on RTAS and discontinuing HTDM aids compatibility is for LE and M-Powered users, since these lower-cost systems never supported HTDM in the first place. One last issue concerning compatibility is for third-party host applications. Neither Apple's Logic Pro or MOTU's Digital Performer is compatible with the version 7 DAE for HD hardware, so don't upgrade if you need to use these applications — or create separate 6.9 and 7.0 boot drives if you need to alternate between the latest version of the Pro Tools software and a third-party application.

Regional Variations



One area of PT7 that's particularly handy is a collection of new features for dealing with Regions. Instead of having separate Region Lists for audio and MIDI Regions, as in the previous versions, PT7 has just one Region List that details both. Even in areas such as post-production, more and more people these days work by loading sound effects into software samplers within Pro Tools and triggering via MIDI instead of using audio Regions all the time. This means that having a consolidated Region List with its associated commands (such as Find, with its new Find History operations) that doesn't discriminate between MIDI and audio Regions is definitely a good move. If you're worried about MIDI Regions cluttering up your audio Region List, the default option is to sort the Region List by type, so audio Regions are always displayed together at the top of the Region List. But if this isn't enough, you can always choose not to display MIDI Regions in the List from the Region List menu. Pro Tools 7 also introduces a new type of Region called a Region Group. As the name suggests, this is basically a Region that consists of a group of other Regions. For example, say you've completed a particular tricky set of edits on several adjacent audio Regions. Previously you might have locked these to make sure the edits ...into this. didn't get changed accidentally; but what if you later wanted to move the whole lot to a new place, keeping the edits intact? It was possible, of course, but there was also a chance of making a mistake. Region Groups make this kind of work easy by grouping together a collection of Regions: simply select the Regions you want to group and select Region / Group — the newly created Region Group replaces the original collection of Regions and you can tell it's a Region Group by an icon in the bottom left of the Region. Region Groups show up on the Region List just like any other Region (except with a different icon) and you can manipulate them in the same way you would any other Region — even cutting a Region Group into other Region Group Regions — and all the original internal edits are preserved. And by Consolidating a Region Group you can create a new audio file (and so audio Region) that is a fixed, composite version. I had to cut together a soundtrack album from a collection of cues recently and this one feature would have saved so much time! In fact, if there's one feature in PT7 that might justify the whole price of the upgrade, this could be it for a large number of users — and it gets better. In addition to being able to group Regions on a single track, it's also possible to group Regions on different tracks, and of different types, into one Region Group. This works in exactly the same way as for grouping Regions on one track — simply select the Regions you want to group and choose Region / Group — and while it's possible to group Regions that aren't on adjacent tracks, it tends to work better if they are.



For example, I tried creating a simple multitrack Region Group from a stereo audio Region and a MIDI Region on the track below, and this worked fine: I could slide the Region around and everything on both tracks was kept together. Once you move a track that contains Regions that are part of a multitrack Region Group, it's still linked in terms of editing (as is the case with ordinary track Groups), but I found some of the handling a little peculiar at this point. In my example, if I moved the MIDI track one track down in the Track List on the Edit window, when I moved one of the separated Regions, the other Region moved with it, as you'd expect. However, if I moved one of the Regions a second time, a copy of just the MIDI Region was left behind in its old position. I think this could be a bug as I can't see any good reason why it would work this way, and it keeps happening even if you bring the separated Regions together again — the only way around it is to ungroup and regroup the Region Group. Other than this one issue, multitrack Region Groups are great, and I can see them coming in really handy when you're working with multitrack recordings of drums, or anything recorded with multiple microphones. You could also use multitrack Region Groups as an arranging tool, creating different Groups for different sections of a song, and perhaps the icing on the cake is that it's possible to export Region Groups from one Session into other Sessions.

Round And Round We Go One other neat Region-related feature in PT7, especially when combined with Pro Tools' new ability to import and work with REX and Acid files, is Region Loops. This enables you to select a Region and loop it over a given period of time. Previously you could use Pro Tools' editing commands to copy a Region the required number of times, but the new Region Loop commands are much more flexible. When you have a Region you want to loop, select it and choose Region / Loop. A window will appear where you can specify whether the Region should loop a certain number of times, to fill a certain amount of space, or simply until it reaches the next Region on that track or the end of the Session. You can enable crossfading for looped audio Regions; the Settings button opens the familiar Fade window where you can configure the crossfade to be used. There is now one consolidated Region List for audio, MIDI and Group Regions, and the newly organised Region menu enables you to sort the List to make it more manageable.

A looped Region functions like one single Region after it's been created, and a loop icon will be displayed in the bottom right corner of every complete iteration of the original Region within the file:///F|/SoS/SoS%2001-2006/protools7.htm (8 of 15)12/19/2005 10:19:24 AM


looped Region. One particularly nice thing is that you can use the Trim tool to adjust the length of the looped Region, and this will extend out the number of times the Region loops, just like in Acid, or Live, or other loop-based music programs. Another point to be aware of is that MIDI Controller data on MIDI Regions will get looped, but automation data on audio Regions won't, which is exactly right, I think: you can always copy automation data separately if you want it to loop as well, but it's perhaps more common to have a section of audio looping while modulating other parameters, such as volume, or insert effects. Looping is quite a useful technique for trying out ideas quickly, but once you have your ideas in place, you may want to turn the looped Regions into individual Regions so you edit them individually. To do this, select the looped Region and choose Region / Unloop, and Pro Tools will ask you if you want to Flatten the looped Regions and create an individual Region for every loop iteration. When you're working with MIDI Regions, it's important to Flatten a loop before making alterations to individual notes: editing notes without Flattening the loop creates additional Regions and the track becomes rather messy, rather quickly. There is one exception to this, though, which is when you have the new Mirrored MIDI Editing mode active. When this mode is active, editing one of the notes in one Region will adjust that note in all instances of that MIDI Region in the Session, and hence within the loop, which is pretty useful. I mentioned that Pro Tools now supports REX and Acid-format files at the start of this section, and although I didn't have any Acid files to hand, the REX-file import worked very well indeed. REX files can be dragged from the Finder/Windows Explorer or Pro Tools' Browser window into the Edit window or Region List, although before you do this, it's important to make sure you're dragging the REX file on to a tick-based track so the REX file remains tempo-independent. A REX Region appears on the Edit window as a Grouped audio Region, although you can select a REX Region and choose Region / Ungroup if you want to access the slices directly.

Getting On The Right Track While Pro Tools started life as an audio recording application, Digidesign's goal these days is clearly to make it into an integrated audio and MIDI production tool that won't need to be complemented by a third-party MIDI sequencer. Pro Tools 6.7 already introduced quite a few new MIDI features, especially with its improvements to tempo editing and manipulation, and version 7 takes Pro Tools sequencing functionality another step further. Thanks to virtual instruments there are now plenty of people who sequence without using any external MIDI gear that makes a noise, so a great new feature in PT7 is the introduction of Instrument tracks. To use a virtual instrument in previous versions of Pro Tools, you needed both a MIDI track and an Auxiliary Input track that would host the Instrument plug-in as an insert. The new file:///F|/SoS/SoS%2001-2006/protools7.htm (9 of 15)12/19/2005 10:19:24 AM


Instrument tracks basically consolidate MIDI and audio track functionality into one track, so you have a single track that accepts a MIDI input and provides an audio output. Instrument tracks appear on the Edit window as if they were MIDI tracks, defaulting to the Notes View that provides piano-roll-style editing. The MIDI Input is specified in a new Instruments View, available on both Region Looping makes it easy to work with the Edit and Mix windows, where there's also a MIDI Output selector that audio and MIDI loops in Pro Tools, as shown by this looped Region Group that started life defaults to send MIDI data to an as an imported REX file. Looped Regions Instrument plug-in that's added, as are identified with a 'looped' icon in the before, as an insert in the first insert bottom-right corner of a Region. slot. By holding down the Ctrl key when assigning a new output, you can specify multiple MIDI outputs so you can trigger both a virtual synth and an external MIDI box from the same track. The Instrument View also contains a MIDI meter that shows MIDI data being played on that track or being recorded, but doesn't indicate MIDI Thru data, which I found a little curious. On the subject of MIDI Thru, there's a neat new option in the Preferences for MIDI Thru called 'Follows First Selected MIDI Track', which basically means that if you have an Instrument track selected, incoming MIDI data will be routed to that track without you having to record-enable the track. If you have multiple tracks selected, the first track in the selection receives the MIDI Thru data, and a side-effect of this is that even if you have an audio track selected, MIDI Thru will still be routed to the first Instrument track in the Edit window. Therefore, making your first Instrument track a piano, for example, ensures that you always have something sensible to play. Since the Instrument track is a hybrid MIDI and audio track, the Instrument View also contains MIDI volume and pan settings, along with a separate MIDI Mute button that mutes playback data, but not MIDI Thru data. This is actually quite nice as it means you can control the internal MIDI volume of a virtual instrument separately from its audio output, for example, which is useful if you've recorded a large amount of Controller 7 (MIDI volume) data but just want to trim the overall audio output for the final mix. And these MIDI volume, pan and mute parameters are available as separate automation parameters from the audio volume, pan and mute parameters. While you can use the automation features on MIDI and Instrument tracks to record MIDI Controllers, it's perhaps a shame that the MIDI Volume, Pan and Mute controls on the Instrument view (and on MIDI tracks) don't visually respond to MIDI Controllers. For example, if I'm playing a synth in MIDI Thru mode and I'm using the Controller 7, I can hear the synth change volume, but the MIDI volume parameter on that track in Pro Tools doesn't reflect this. Instead it displays the last volume automation event played back, or the last value chosen



by the user when dragging the on-screen MIDI volume control. In addition to separate MIDI controls for some parameters, Instrument tracks, like other audio-based tracks, can accept an audio input in addition to a MIDI input. This is, of course, really useful for vocoders and other audio plug-ins that can accept a MIDI input to modulate an audio signal.

It's The Little Things One of the things I like about Pro Tools is that Digidesign are usually pretty good at responding to user requirements, and the Pro Tools 7 upgrade includes many small enhancements that are nonetheless very worthwhile. For example, I know many engineers who have been frustrated at not being able to resize the I/O Setup window in previous versions — and if you have a system with 160 inputs and outputs, you'll soon see why this is annoying! So, for many, the fact you can now do so will be a big, big deal! Another feature that frustrated me in previous versions was not being able to drag and copy Send slot assignments in the same way you could drag and copy insert plug-ins, but in PT7 this is possible. And speaking of dragging, as with REX files, you can now drag and drop MIDI and audio files from the Finder or Windows Explorer directly into Pro Tools, and when adding multiple Regions to the Edit window you can set whether they should be stacked from top to bottom (on different tracks) or left to right (on the same track). You can also drag and drop audio files from Digibase directly to a plug-in window, for plug-ins that support this behaviour — the new version 7-compatible release of Digidesign's own audio loopmanipulation plug-in Synchronic already does. A new option on the Edit window is Link Track and Edit Selection (next to Link Timeline and Edit Selection). This means if you select a Region in the Edit window, the track on which that Region is located will be automatically selected.

Keeping It Real (-Time) In previous versions of Pro Tools, all processing on MIDI data happened off-line, and involved manipulating the actual Regions on tracks. PT7 introduces real-time MIDI processing via the new Real-Time Properties window and display section in the Edit window's Track List, the combination of which allows you to process all Regions on a track with a set of global parameters, or set specific parameters that affect only one Region. If you've ever used other sequencers, you should be able to draw parallels to these new features in Pro Tools from concepts like Cubase's Inspector and Event Infoline and Logic's Track and Region Parameter boxes. When you select View / Edit Window / Real-Time Properties a new View Section is displayed against every track, whereby MIDI and Instrument tracks display five new buttons for Quantise, Duration, Delay, Velocity and Transpose. Each of these buttons is an on/off toggle; to enable real-time quantising on a track, you click the Qua button and two additional parameters (resolution and swing file:///F|/SoS/SoS%2001-2006/protools7.htm (11 of 15)12/19/2005 10:19:24 AM


percentage) appear to configure the way notes are quantised. The resolution menu allows you to choose the note lengths from a whole note to a 64th note, and there are additional flags for dotted and triplet values. A really nice touch is the inclusion of various 'groove' quantise settings in the style of other, well known sequencers, which you can also access from the resolution menu, including Logic (the infamous 8A, 8B, and so on), Cubase, Feel Injector and the Akai MPC series. The Duration parameter lets you change the length of notes in a variety of different ways. You can simply add or subtract to a length in musical time (which you can specify either in ticks, or in whole notes to 64th notes of regular, dotted or triplet values), or you can adjust a legato parameter, either to widen the distance between notes or introduce overlap, also in musical time. Finally, you can scale the lengths via a percentage value as well. Delay makes it possible to delay or advance MIDI Events in either ticks or milliseconds, Pro Tools 7 introduces a new track type independently of any offsets you might called Instrument tracks, consolidating an already have specified for a Track or audio-based track and a MIDI track into one. globally in the MIDI Track Offsets Notice how MIDI controls are provided in the Instrument View section on both the Edit and window. Velocity enables you to either Mixer windows, in additional to the standard add or subtract a value from all audio controls on the channel. velocities, or scale the velocities by a percentage. And, finally, Transpose enables you to either transpose note pitches by octaves and semitones, or set all notes to the same pitch with a useful Transpose To command. There's also a separate Real-Time Properties window, which allows you to set up these five properties for the selected track in more detail than is possible from the Edit window View, with additional options for the Quantise, Duration and Velocity parameters. When you enable Real-Time Properties on the Edit window for a track, a 'T' appears in each MIDI Region on that track, indicating that those Regions are affected by Real-Time Properties set on a track level. However, it's also possible to set these properties individually for specific Regions, by choosing Region instead of Track from the Apply pop-up menu in the Real-Time Properties dialogue. The letter 'R' will then appear in the MIDI Region instead of a 'T'. Disabling Region-specific Properties again won't set the Region back to using the track's Properties until you click the Clear Region Properties button in the RealTime Properties window. If you want to make the Real-Time Properties permanent (which is to say not realtime any more), there's a handy Write to Track/Region button in the Real-Time file:///F|/SoS/SoS%2001-2006/protools7.htm (12 of 15)12/19/2005 10:19:24 AM


Properties window that will apply them to the selected track or Region. Clicking Write to Region will process just the selected Region, while clicking Write to Track will process every Region on a track with the track-level Properties. If you have a track that contains Regions with both track-level and Region-specific properties, Pro Tools will display a window asking if you want to process only the Regions assigned to track-level properties, or if you want to set all Regions to track properties and process everything.

Off-line MIDI The off-line MIDI Operations window has also been enhanced. The Select Notes and Split Notes pages have been combined, generating more options for selecting notes based on velocity, duration and relative position. There's also a new option when splitting notes that enables you set whether the split data gets cut or copied to the clipboard, a new track, or a new track per MIDI note pitch. This puts notes of each different pitch onto a different track, which is useful for 'exploding' MIDI drum parts. The quantisation options have also been consolidated, occupying two pages in the MIDI Operations window, rather than three as before, and including a Randomise option. The Change Duration window has also been improved and includes the The new Real-Time Properties Edit View consolidated Set/Add/Subtract/Scale section and window enable you to set pop-up menu from the Real-Time various properties to process MIDI data in real time on MIDI and Instrument tracks. Properties window, and the Legato Here you can see that the first and third MIDI Gap/Overlap options. There's also a Regions are affected by the track settings Remove Overlap command, and (as indicated by the 'T' icon in the upper rightTransform Sustain Pedal to Duration, hand corner) while the middle Region is which uses any sustain pedal data to affected by the properties in the window (as indicated by the 'R' icon). adjust the note lengths. Finally, the Transpose page also includes the new options from the Real-Time Properties window as well. Some of the smaller MIDI improvements include a new Remove Duplicate Notes command, and new options for importing and exporting MIDI files. When importing MIDI files, you now have three separate options as to whether existing Instrument tracks, MIDI tracks and MIDI Regions are removed, in addition to the existing toggle for importing the tempo map from the MIDI file. When exporting MIDI files you can now set a SMPTE start time for the exported MIDI file and whether the Real-Time Properties are applied. And, on the subject of MIDI files, they are now supported in Digibase as well. A nice touch in the piano-roll MIDI editor is that you can now click on the minipiano-keyboard with various modifier keys to make pitch-based selections and file:///F|/SoS/SoS%2001-2006/protools7.htm (13 of 15)12/19/2005 10:19:24 AM


audition notes. This should be familiar to users who have worked with other sequencers like Cubase. For me, though, the thing I miss the most when working with Pro Tools as a sequencer is the fact you can't open Regions from multiple tracks in an editor simultaneously. I like the way Pro Tools is mainly a two-window application, for the most part, so I wouldn't want to advocate a new window, but if there was some way of expanding the Edit window to show one large piano-roll-style editor displaying several Regions across different tracks, I think more people would feel comfortable in sequencing with Pro Tools. It's important to see multiple Regions superimposed on top of each other for harmonic reasons, and a dedicated graphical MIDI editor window would also help when you want to see and edit MIDI Controllers and MIDI notes simultaneously without switching to different views.

Conclusion Pro Tools 7 is a pretty impressive piece of software, and it's an interesting update in that although there are many great new features, it doesn't feel like there's a single really dramatic, life-changing feature. But I don't mean this as a criticism because, instead, it seems Digidesign have just focussed on making the application simply better — every new feature, whether big or small, leaves you thinking 'Oh great, that's really going to help me.' And while the menu structure has changed, it's going to be fairly easy for existing Pro Tools users to adjust: despite the new features, PT7 actually feels less cluttered than version 6! In terms of stability, I might put my neck out and say that Pro Tools 7 seems like the most stable initial release of a Pro Tools version to date, and this is a good thing. In two weeks' use (albeit only on Mac OS X) the application only crashed twice, and once was when I was trying to push the RTAS processing to the limit for the performance comparisons. But that's not to say PT7 crashes when you run a large number of RTAS plug-ins, because after I had to Force Quit Pro Tools due to the 'spinning beachball of doom', I reloaded the application, performed the same test and everything was fine. The other time was when I was dragging a converted REX Region Group from the Region List to an empty Edit window so that Pro Tools would create a new audio track (which is now possible in version 7), which is a little more worrying, although I wasn't able to reproduce that problem either. For audio recording, editing and mixing, Pro Tools is hard to beat. It's ironic that these days Pro Tools is regarded as an expensive option, when 10 years ago, Digidesign were hailed as the company to bring affordable hard-disk recording to the masses. And while a Pro Tools HD system is indeed more expensive than purely host-based options, the quality and guaranteed performance it offers are hard to argue with. However, you could contend that the gap between LE and HD hardware has become too great, and maybe a product along the lines of the old Pro Tools Project card that just took care of I/O and left the DSP processing to file:///F|/SoS/SoS%2001-2006/protools7.htm (14 of 15)12/19/2005 10:19:24 AM


RTAS plug-ins on host machines would be a good way of bridging the gap. I'm not sure if I'd want to sequence in Pro Tools myself, mostly because of the lack of a really powerful graphical MIDI editor; but depending on your sequencing requirements, Pro Tools 7 offers the best functionality yet for this task. A friend of mine recently was working with a couple of composers who sequenced an entire film score in Pro Tools 6.9, so with the new features in version 7 the potential is definitely there, and I look forward to seeing more developments in this area in the future. If you're a Pro Tools user already, the version 7 upgrade should definitely be considered a must-have. In terms of an audio environment, I still think Pro Tools is the standard to beat: it might not have every feature offered by competing products, but it works and is dependable. Published in SOS January 2006 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



DK Technologies MSD100C

DK Technologies MSD100C £1410 pros Excellent colour display. Instant access to three meter presets. Eleven preset options stored internally. Completely configurable using the supplied PC utility. Stereo analogue and digital inputs.

DK Technologies MSD100C Stereo Audio Meter Published in SOS January 2006 Print article : Close window

Reviews : Accessory

The baby of the MSD series of meters gets an overhaul.

cons It may seem an expensive luxury — until you use it! Spectrum analyser display no longer available.

summary This meter from DK Technologies has gained the same bright colour display as the other MSD meters, and benefits from a new PC configuration utility.

information £1410 including VAT. DK Technologies UK +44 (0)870 241 4118. +44 (0)870 241 4119. Click here to email www.dk-technologies. com

Hugh Robjohns

Danish company DK Technologies have introduced another updated version of their MSD100 stereo meter. The black-and-white display of the previous model, reviewed back in SOS April 2002, has been replaced with a colour LCD panel, and the internals have been completely redesigned. Sadly the spectrum analyser function has got lost in the process, so this is only available now on the MSD200C and MSD600 models. A rear-panel 25-pin D-Sub accepts balanced stereo analogue signals, stereo digital signals (AES3 or S/PDIF) at up to 96kHz sample rates, and 12-15V DC power from an external switched-mode line lump. A standard VGA output allows the metering display to be repeated on a larger screen, and an RS232 serial port can connect to a PC for configuration purposes. The bright, clear screen is divided into three parts: a vertical phase meter on the left, a goniometer (vectorscope) display in the centre, and bar-graph metering to the right. Three buttons below the screen recall one of three programmable display presets. Each preset's bar-graph type can be changed simply by doubleclicking the appropriate button and then selecting the required scale from a list of 11 options including four analogue PPM scales (NBC, ABC, Type I Nordic, and Type II BBC), four digital input scales (NBC, ABC, BBC, and DMU), and three further combined options.

file:///F|/SoS/SoS%2001-2006/dktechmsd100c.htm (1 of 2)12/19/2005 10:19:27 AM

DK Technologies MSD100C

If this isn't sufficiently flexible, there is a Windows-based PC utility on an included CD-ROM which, communicating via the RS232 interface, can alter virtually any aspect of the 11 stored meter configurations: signal source and display colour for all three meter sections; bar-graph meter scales, including the option to create bespoke scales; and phase meter integration time. Having modified the 11 presets, three can again be allocated to the meter's front-panel buttons for immediate access. I've been a big fan of MSD meters for many years now, and find them utterly indispensable. The bar-graph metering is extremely accurate, and the ability to change scales can be surprisingly useful. The goniometer display takes all the guesswork out of panning and stereo mixing, and the phase meter provides reassurance that all is as it should be. While many of the MSD's metering facilities are provided as standard in most DAW systems now, if you work regularly with hardware mixers and recorders the MSD100C would be a wise investment that will remain in daily use long after you have repeatedly upgraded the rest of your equipment! Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


file:///F|/SoS/SoS%2001-2006/dktechmsd100c.htm (2 of 2)12/19/2005 10:19:27 AM

Drawmer Three-Sum

In this article:

Analogue Hardware For Multi-band Processing Split & Mix Studio Tests Three-Summary

Drawmer Three-Sum Band-splitting Processor Published in SOS January 2006 Print article : Close window

Reviews : Processor

Drawmer Three-Sum £546 pros Conceptually simple. Effective limiter built in. Excellent audio performance. Surprisingly affordable.

This intriguing new unit from Drawmer lets you split a stereo signal into three bands and then process each band with a different piece of outboard.

cons No per-band level controls.

Paul White

summary A great idea for anyone who has a desire to combine more conventional analogue equipment for multi-band processing.

information £546.38 including VAT. Drawmer Distribution +44 (0)1924 378669. +44 (0)1924 290460. Click here to email www.drawmer.com

Low-cost plug-ins and multi-band hardware boxes, such as the TC Electronic Finalizer and Drawmer DC2476, have made more people aware of the possibilities of multi-band Photos: Mark Ewing signal processing, especially in mastering applications. However, many of the big-name mastering engineers still prefer to work with analogue equipment, albeit very high-end, expensive analogue equipment. What's more, they like to be able to pick which equaliser or compressor to use depending on the material they are working on, and it isn't unusual for mastering engineers to have their own custom switching and mixing systems built to enable them to do this easily. For those on a lower budget who still prefer to work in the analogue domain and who demand a very high technical specification, Drawmer have come up with an ingenious piece of equipment that enables those old analogue processors you locked away in the cupboard to be given a new lease of life.

Analogue Hardware For Multi-band Processing What Drawmer have done here is build a processor that can split a stereo audio signal into three frequency bands, providing individual connection points for each band so that the user can insert their own stereo processing devices into the signal path. This enables the user to patch in different external devices for processing each frequency band. Although compression is the most obvious choice here, there are some other intriguing possibilities opened up by such a

file:///F|/SoS/SoS%2001-2006/drawmerthreesum.htm (1 of 6)12/19/2005 10:19:32 AM

Drawmer Three-Sum

device, not least that of using controllable distortion to add energy and punch to specific parts of the mix spectrum. This last point was really brought home to me when I bought a Drawmer DC2476 Masterflow three-band digital processor, which has, amongst more familiar processes, adjustable tube saturation emulation for each of its three frequency bands. The difference this makes in enabling me to warm up the bottom end of mixes or to add breath to the top end is far greater than I ever expected, and it's a feature I now employ to some degree on most of the pop and rock mastering jobs I get involved with. They really ought to make that one process available as a plug-in!

While the theory of multi-band processing is too wide to go into here in any great detail, it is worth pointing out the main benefits, specifically as they apply to compression. Conventional compressors turn down the level of the whole signal, no matter which part of the frequency spectrum is responsible for the level peak that triggered the compressor. In a pop mix, this often results in the kick drum triggering the compressor and causing high-frequency sounds such as hi-hats to be dropped in level whether they need it or not. By processing each band separately, peaks at the bass end can be controlled without disrupting what's going on higher up the spectrum. Overall this allows more compression to be applied without unduly affecting the subjective transparency of the mix. Additionally, there are benefits to being able to use different amounts of compression in the different bands — for example, you can increase the density of the low end by using a higher compression ratio or a different compressor threshold, without altering more subtle settings for the mid- and high-frequency bands.

Split & Mix Conceptually, the Three-Sum is pretty straightforward. It's a 1U box that uses precision crossover-style circuitry to split the incoming audio into three bands. The three sets of stereo signals are then routed to rear-panel balanced XLR connectors, and the returns from the externally connected stereo devices are brought back into the Three-Sum and added back together to give a full-range stereo signal. A precision two-stage brick-wall limiter is included in the summing section to prevent overloads caused by combining the three sets of processed signals, and there's a pair of meters that can be switched to monitor the input or output levels. Apparently, the limiter deals with high frequencies separately, so as to retain transparency during limiting.


Drawmer Three-Sum

The first thing the input signal encounters when it enters the unit via the rear-panel XLRs is a simple Level Trim control with ±10dB of gain range. From there, the signal is split into three bands, where the lower split frequency is continually variable between 18Hz and 1.6kHz, and the higher split frequency between 530Hz and 42kHz. A fairly gentle filter slope is used to help avoid artefacts at the crossover The limiter has been designed to operate transparently, but will also give a classic points, specifically level 'humping' when compression is being used. Each analogue pumping effect if driven hard. band has a Normal/Mute switch, which is useful if you need to scrutinise just one or two bands, and each band can also have its external connection points bypassed to remove the effects of the externally connected processor. A further Gain control is placed after the point where the returned signals are summed, and this feeds directly into the limiter, which can be set to operate at levels from 0dBu to +16dBu. A four-LED meter shows the gain reduction caused by the limiter's action, and two moving-coil meters with VU characteristics can be switched between input or output levels. Because many digital recording systems require a very high input level to reach digital full scale (typically around +16dBu), the meter scale can be switched from reading 0VU full scale to +10VU. In fact the only obvious omission is a level control for each band, so if you're patching in your own processors, they'll need to have their own level controls to enable you to balance the contributions from each band. Compressors invariably have a make-up gain control of some kind, so this shouldn't be a problem. However, it would have been better from an ergonomic point of view to be able to fine-tune the band levels from the front panel. On the rear panel are the IEC power inlet, balanced XLRs for the stereo inputs and outputs, and balanced XLRs for the stereo sends and returns from each band. There are no jack alternatives, which would have made life easier for me, but in a serious mastering situation XLRs would probably be the connectors of choice. That's a total of 16 XLRs, so no wonder there's no room for jacks! A mains voltage selector is available inside the case, which means you won't reset it accidentally.

A 115V/230V voltage selector switch is only provided inside the casing so that it cannot be accidentally switched to the wrong setting during use.

For any mastering engineer to take a product like this seriously, it needs to have an impeccable technical specification, and Drawmer have been careful to keep


Drawmer Three-Sum

the audio bandwidth wide enough to keep the tweakers happy, but without it being so wide as to behave as an impromptu radio receiver! The response is flat to within 1dB from 17Hz to 28kHz, while the 3dB down points are at 10Hz and 47kHz. THD + Noise is better than -85dB, with crosstalk lower than 63dB at 10kHz. Although the crosstalk figure may not look particularly impressive, it is rather better than that which many competing multi-band devices offer, and in any event some analogue buffs cite a degree of crosstalk as one of the factors responsible for the 'analogue' sound.

Studio Tests The first thing to decide, after thinking about what processes you want to apply, is where to set the crossover points. Different pieces of music demand different solutions, but in most instances I start by keeping the mid-range fairly wide and open by setting the low crossover point to around 150-250Hz and the high crossover point at 2-5kHz. You can learn a lot by muting the various bands to see what part of the musical spectrum is being affected. You may also be able to manage without processing all three bands in all cases, though for mastering you may well want to. A possible approach is to leave the mid-band alone, compressing the bass end to pump up the energy a bit, and overdriving a tube processor slightly to brighten up the top end. In this respect, adding subtle distortion only to the high end works a bit like a harmonic enhancer. My first test involved using a Drawmer DL441 quad compressor patched in to handle the low and high bands only. Each compressor pair was linked to ensure consistent tracking. I used a hard-knee low ratio on the bass end and a soft-knee low ratio at the high end, adjusting the thresholds to get just a few decibels of gain reduction in each band. The mid-band of the Three-Sum remained bypassed at this time, and I used the output gain controls on the compressors to balance the high and low ends against the middle. The result was an extremely crisp mix with bags of transient detail, loads of low punch, and a generally nicely produced sound. Using just these two units alone allows the user to polish mixes in a hugely effective and classy-sounding way.

Adding a further compressor to work on the mid-band can enable the overall level to come up slightly, but where compression has been used on individual tracks at the mixing stage this may not be necessary. It's also worthy of note that the limiter works extremely well for catching transients, and that its effect is extremely benign when showing between one and three decibels of gain reduction. If you push harder into limiting by turning up the output gain, you get the classic analogue pumping effect — this can actually be used quite creatively file:///F|/SoS/SoS%2001-2006/drawmerthreesum.htm (4 of 6)12/19/2005 10:19:32 AM

Drawmer Three-Sum

on some rock mixes or drum submixes, but for conventional use it's best to set the limiter so that only one LED flashes on the meter, and that only infrequently. If you need to bully more loudness out of a mix without making it sound overprocessed, there are more effective plug-ins that can do that. Adding distorting devices to the high end can help create the impression of detail from a dull mix, but a lot depends on the device you're using. A dual-channel tube processor of some kind that can be driven into harmonic distortion is probably the best bet, and it will sound more subtle than using guitar distortion devices. However, for deliberately aggressive musical styles, more overt distortion mechanisms might just do it for you. Whatever you use, it is important that the left and right channels can be accurately matched, otherwise your stereo image will suffer.

Three-Summary Used with care, the Three-Sum can be very effective, but I also appreciate that it may not be a solution for everybody. In my view, it's more likely to appeal to mastering engineers and high-end project-studio owners who know their signal processors in some depth, rather than the casual user looking for the magic 'fairly dust' button. However, add in a Drawmer DL441 and you get pretty close to instant magic! With the benefit of the 20/20 vision of hindsight, Drawmer might have made this box more attractive by building variable analogue tube emulation into each band, because as it stands you can't do anything with the Three-Sum until you plumb in some external equipment. However, they obviously also had to meet a viable cost point, and it does provide a simple and effective means of combining other analogue equipment to create a bespoke multi-band processing setup at a UK project-studio price.

The Three-Sum has been designed so that it can generate the high-level signals demanded by many professional audio interfaces and A-D converters. However, when driving the unit this hard, the VU meter becomes pinned to the end stop, so Drawmer have provided a switch beside the meter which reduces the VU reading by 10dB when needed.

The quality of the results available with this system will be dictated mainly by the skill of the user and the quality of the connected equipment, but the whole can often be greater than the sum of its parts. My tests confirmed that analogue multi-band compression still has a lot to offer, and using the Three-Sum to achieve it is a lot less costly than buying a


Drawmer Three-Sum

dedicated multi-band analogue mastering compressor. Of course there are some advanced features missing, but apparently Drawmer's engineers started by sketching out a unit that had everything, before pruning it down to the essentials to reduce cost and minimise the signal path. For example, some mastering engineers would have liked switchable filter slopes, per-band level controls, and an overall bypass switch, while I'm pining for my multi-band tube emulation, but in reality Drawmer have delivered the essentials without compromising on audio quality. There are also applications beyond those of mastering, as this processing can also be used beneficially to polish up vocals, drums, bass guitars, and other sources, though I think you'd have to be pretty dedicated to patch up something like this just to work on a bass-guitar track! The Three-Sum may not be destined to be Drawmer's biggest-selling product, but it offers a genuinely useful facility for those users who do a lot of multi-band analogue processing, especially those project studio owners with analogue compressors who'd like to get more involved in doing their own mastering and who find digital plug-ins and processors too cold sounding. Published in SOS January 2006 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



Guitar Technology

In this article:

Cornford Carrera TECHNIQUE Pedalsnake G2 Takamine Cool Tube

Guitar Technology Tips, Techniques & Gear Published in SOS January 2006 Print article : Close window

Reviews : Guitar Amplifier

Cornford Carrera Recording amp with switchable choice of output tube type Cornford Amplification, the UK-based maker of high-quality guitar amplifiers have launched a companion to their acclaimed Harlequin 6-Watt, class-A recording amplifier. Their new Carrera model adds a number of features, including a mid-range EQ control, a front-panel send/return effects loop, standby switch and a spring reverb. Most interesting of all, however, is the inclusion of two different types of output tube, one eight-pin type (such as the 6L6 and EL34) and one nine-pin (6V6, EL84), with the user able to switch between them to get different sound characteristics. The Carrera ships with an EL84 (the same as the Harlequin's single output tube) and a 6L6 on board, offering 5 Watts and 8 Watts respectively, but you can swap these for, say, a 6V6 (like a Fender Deluxe) and an EL34 (typical of most Marshalls), greatly enhancing the amp's versatility in the studio. Although there have been previous amp designs that can accommodate a range of different output valve types (notably the THD models), there's nothing like being able to do it at the flick of a switch to encourage creative experimentation. You can also swap any of the Carrera's three 12AX7 preamp valves for a lower gain 12AT7 or 12AU7 to further fine-tune the amp's gain structure to your own requirements. In action, the amp sounds every bit as sweet as the Harlequin, but with more scope to lift or de-emphasise mid-range without sacrificing overall drive level. Switching to a different output valve type really does change the vocabulary of the amp, with a 6L6 in particular giving a warmer and bigger sound, especially on cleaner or right-on-the-edge sounds. On all valve types, the overall amp voicing remains just on the dark side, like the Harlequin, making it ideally suited to closefile:///F|/SoS/SoS%2001-2006/guitartechnology.htm (1 of 6)12/19/2005 10:19:36 AM

Guitar Technology

miking. Brighter amps may be better at filling a room, but they tend to sound brittle and harsh when close-miked. The Carerra's 'browner' tones are ideal for the robust directness of an SM57 right against the cone, whilst still rewarding subtler treatments like a ribbon mic or a condenser backed off a couple of feet. At £999, the Carrera complements the Harlequin in the Cornford range rather than replacing it, and should certainly be on the audition shortlist for any recording guitarist into premium tube amps. Dave Lockwood SUMMARY: A very successful realisation of the high-quality, low-power guitar recording amplifier concept, with the added twist of switchable output tube types. Nothing else exactly like it on the market; a THD UniValve with a suitable speaker would get you in the same sonic territory, but you'd have to swap over the tube type manually when you wanted a change of character. www.cornfordamps.com

TECHNIQUE Successfully combining mics and pickups in acoustic guitar recording Recording satisfying acoustic guitar sounds in the home-studio environment can be difficult, particularly if the part will be prominent in the mix. Most home studios are not acoustically well balanced or quiet enough to allow optimum studio miking techniques to be used. Pickup systems of all varieties each have their own compromises too. Combining a pickup and a mic signal recorded to separate tracks offers one way forward, but often doesn't sound as satisfying as it should because the pickup tends to remain dominant in the mix until it is reduced to the point where it is almost inaudible. The key to making this work is time alignment of the two signals, which is easily achieved if you are working with a software sequencer (or a reasonably sophisticated hardware digital recorder). Normally, the pickup is heard as the dominant source because its signal always 'speaks' first — the electrical signal from the transducer under the bridge is practically instantaneous, whereas the microphone signal has to travel the distance between the guitar and the mic, and other elements of the sound take even longer as the bridge drives the top into vibration and stimulates the resonance of the body enclosure. file:///F|/SoS/SoS%2001-2006/guitartechnology.htm (2 of 6)12/19/2005 10:19:36 AM

Guitar Technology

Sound propagates in air at a speed of roughly one foot per millisecond, so if your mic is a foot away from your guitar, delaying your pickup signal by one millisecond would be a good place to start (44.1 samples if you are working at 44.1kHz sampling rate). It is only a starting point, however, for the dissimilarity between the two signals means that there is no exactly 'right' time-alignment point. The different shapes of the waveforms make it quite difficult to decide by eye when the two tracks are optimally aligned. The best solution, I find, is simply the one that The mic signal (top waveform) can clearly be seen to lag behind the pickup (lower trace). sounds best, starting from the nominal '1ms per foot' mic distance compensation and then nudging by ear until the attack of the pickup no longer dominates, but its track continues to add sustain and body to the notes.

Pedalsnake G2 Configurable guitar rig multicore system Where do you turn when you need a multicore that will handle two MIDI lines, a 9V DC power line and a voltage control line? Suddenly needing all of the above for my on-stage electric guitar rig, and finding the idea of four more separate cables running out to my already crowded footswitch position distinctly unappealing (to say nothing of the additional setup and breakdown time), I started to look at making up a custom multicore until I realised that the perfect solution was already available. Pedalsnake is a multicore cabling system for guitar pedalboard users, with a patented internal shielding system that allows audio signals to share a snake with DC power and switching lines (not normally a good idea!). The multicore is terminated with the necessary variety of connectors for each end, with jacks for guitar and line signals, male and female 2.1mm barrel connectors for power and so on, allowing all your pedal power supplies to remain at the back of the stage with your amp.

file:///F|/SoS/SoS%2001-2006/guitartechnology.htm (3 of 6)12/19/2005 10:19:36 AM

Guitar Technology

In the Pedalsnake CS range, the connectors are all hard-wired, either in off-theshelf or custom configurations, but the company's G2 range allows the user to attach a selection of tails to set the function of each multicore line. Each of the colour-coded lines of the G2 base snake terminates in a standard MIDI connector, whilst the tails all have a female 5-pin DIN with a short cable leading to the appropriate connector for your application. In my case, I terminated one line with a quarter-inch jack at each end for my voltage control pedal, another line with a male and female 2.1mm connector for power, and left two lines unterminated as my MIDI send and return. If I were thinking of running sensitive, guitar-level audio down it, I would probably opt for the CS range, with hard-wiring and all-metal connectors, but the G2 range is the one that offers complete reconfigurability and a perfect solution to my particular problem and many others, both on stage and in the studio. Pedalsnake systems start at around £53. Dave Lockwood SUMMARY: A unique problem solver. The reconfigurability offered by swappable tails adds a great deal of flexibility and ensures that your investment in the basic cable can never be rendered useless by minor changes to your setup. www.pedalsnake.com www.madisonandfifth.co.uk

Takamine Cool Tube Tube-based onboard preamp for electro-acoustic guitars For years now, acoustic guitarists have been choosing tube-based outboard preamps to warm up the signals from piezo-based pickups. The piezo's overfast attack and non-linear output can be partially mitigated by the tubes' tendency to gently compress (within suitably designed circuits). Now Japanese electro-acoustic specialists Takamine have designed a tube preamp that mounts within the guitar itself! The CTP-1 Cool Tube preamp addresses the obvious concern about excess heat within the guitar body affecting the wood by running the tube at just a couple of degrees above ambient temperature, using a supply of just three Volts for the heater circuit. In fact, the 12AU7 tube doesn't visibly 'light up' at all, leading some people to wonder whether it's actually doing anything! Running a tube at very low voltage is perfectly valid provided you are not using it as a gain stage. Tubes work by causing free electrons stirred up by heating the oxide coating of the cathode to jump across to the positively charged anode or 'plate', with the current flow being regulated by a negative control voltage, or 'bias', on the 'grid', located in between the two. Applying a small input signal to the grid therefore modulates file:///F|/SoS/SoS%2001-2006/guitartechnology.htm (4 of 6)12/19/2005 10:19:36 AM

Guitar Technology

the flow of electrons to produce a larger version of the signal at the plate... hence, gain! In a low-voltage circuit, tubes are most likely to be used as harmonic distortion generators, with a low plate voltage used to deliberately increase nonlinearity, accompanied by a solid-state stage to provide gain. Takamine's Cool Tube preamp allows you to blend the desired amount of 'tube effect' into the output, but the subjective audible effect is more warmth and gentle compression than distortion. It actually works really well! Takamine's patented Palethetic pickup system, using an array of multiple piezo transducers built into the bridgeplate during manufacture, is already a half-decent-sounding solution, but the sound is still rather brittle, with unnatural dynamics. Dialling in even a little of the Cool Tube effect, the slightly sterile sound of the pickup system alone becomes far more three-dimensional and complete, almost like adding additional transducers to sense more of the body vibration. I'd hesitate to say that it is significantly more microphone-like, but what it certainly is is much nicer, both to listen to and to play. Low-level detail becomes more audible and sustain is increased, as if using a good compressor. There's a useful degree of variation available too; I found I used about 60 percent Cool Tube in a recording context, for maximum warmth, but only around 30 percent on stage, where I wanted to both retain more edge and minimise feedback. The CTP-1 conforms to Takamine's Sound Choice modular preamp format found on all their guitars since 1989 and features a builtin tuner as well as the standard three-band EQ with semi-parametric mid-range. It is powered by four AA batteries, giving around 24 hours of playing time — enough to get you through the gig, but clearly not ideal on a long-term basis. An alternate power source switch on the preamp module suggests that external powering will be possible, but no details are currently available. The tube itself is a standard type that should last as long as the guitar in a circuit of this kind. Many attempts have been made in recent years to overcome the inherent limitations of all acoustic guitar pickup systems, ranging from alternative transducer technologies to on-board DSP modelling. Takamine's Cool Tube approach is undeniably one of the more successful ones. The CTP-1 module costs £209 in the UK and is fitted as standard on Supernatural and Nashville models. Dave Lockwood SUMMARY: Battery life remains an issue until a phantom powering solution is available, but the Cool Tube circuitry really does significantly improve the sound of file:///F|/SoS/SoS%2001-2006/guitartechnology.htm (5 of 6)12/19/2005 10:19:36 AM

Guitar Technology

Takamine's integrated pickup system. An external tube preamp will get you close, but there is something sonically very nice about the way Takamine have implemented this. www.takamine.com www.takamine.co.uk Published in SOS January 2006



file:///F|/SoS/SoS%2001-2006/guitartechnology.htm (6 of 6)12/19/2005 10:19:36 AM

Korg D3200

In this article:

Korg D3200

Screen Test Digital Multitracker Inputs & Outputs Published in SOS January 2006 Keeping Up Appearances Print article : Close window Fresh Ideas Reviews : Multitrack Recorder The Mixer Internal Drum Machine Smart Editing Built-in Effects Backups & USB Korg's new workstation heavyweight boasts 32 recording Performance In tracks, a powerful 44:12:2 mixer, a programmable drum Practice machine, and up to 11 simultaneous effects — for under Multitrack Monster

£1000!

Korg D3200 £999 pros Loads of features for the price. Intuitive hardware controls and a welldesigned operating system. Independent phantom power for each XLR input. Exceptional digital editing options.

cons Track count halved if 24-bit recording is selected. Short faders. EQ controls limited on some mixer channels. Internal drum machine could be more programmable. Gates and compressors can't be triggered by other tracks. Screen size not appropriate for the software. The supplied effects and processing are

Tom Flint

Although the prodigious 32-track playback of Korg's new D3200 is enough to turn heads on its own, the company haven't rested on their laurels. They've also included a a 44channel, 12-buss mixer, a respectable set of digital editing facilities, powerful multi-effects processing, a programmable drum machine, and MIDI control/synchronisation. The machine is no slouch on the hardware side either, with a decent array of I/O facilities and an intriguingly knobular user interface.

Photos: Mark Ewing

The 32 playback channels are assigned to 16 hardware faders in two banks which you can switch between at the press of a button. There's also a dedicated fader for the drum machine and the Master fader which controls the overall mix level. Each track, including the stereo Master track, has seven virtual tracks for storing alternative takes or ideas. Up to 100 songs can be stored on any single drive, although there's only really space for about 20 medium-length 32-track compositions on the 40GB drive provided. Still, there's no real need to pack the drive with songs when they can be backed up to CD-RW media using the onboard burner or filed away on a PC or Mac hard drive via the rear-panel USB connector. The D3200 is capable of both 16-bit and 24-bit recording at either 44.1kHz or 48kHz sample rates, although recording at 24-bit resolution halves the track count whichever sample rate is selected. The manual states that up to 16 tracks can be recorded at once, but, given that there are only 12 analogue inputs, you'd need to be recording the

file:///F|/SoS/SoS%2001-2006/korgd3200.htm (1 of 11)12/19/2005 10:19:40 AM

Korg D3200

spread pretty thinly over 32 tracks.

summary Although the D3200 is not perfect, it's still pretty powerful and offers some innovative features despite its comparatively low cost.

S/PDIF digital input and the internal drum machine's outputs at the same time! I'm not quite sure why you'd want to record the drum machine, though, as it's easily synchronised and has a dedicated track of its own. In 24-bit mode simultaneous recording is reduced to 12.

For synchronisation purposes, the multitracker can send and receive MMC and MTC information, and can act as a MIDI Clock master. It will also respond to MIDI Continuous Controller, Program Change, and Note On/Off messages; so it is amply capable of talking to external effects modules and sequencers. The onboard information automation system is able to record fader moves, panning, channel on/off commands, £999 Including effect-send moves, and expression data. All automation is stored in an event list and VAT. can be edited in a number of ways. The mixer also offers snapshot-based automation, Korg UK Brochure Line +44 (0) and there's another editable list showing where each snapshot happens. 1908 857150. +44 (0)1908 857199. Click here to email www.korg.co.uk

Screen Test

www.korg.co.jp

The screen is arguably the most important single feature on a multitrack recorder, yet I've noticed that many of the products competing at the same level as the D3200 are a disappointment in this department. Could it be that the screen is the one component that pushes up production costs more than any other? Of the competing products, the Yamaha AW1600 has the best combination of screen size and display design, even though its resolution is low and it doesn't quite have the scope of its forebear, the AW2816.

Test Spec Korg D3200 OS v1.0.1. 2.66GHz Pentium 4 PC with 256MB RAM running Windows XP Home.

The Zoom MRS1608 has a much cruder screen by comparison, but the designers have proved that a really basic display can be functional if it is logical enough, with large easyto-read characters. The Boss BR1200CD's screen is a little more detailed than the Zoom's, but it still feels a little small for the job at hand. Korg's high-resolution display is by far the most sophisticated in its class, and much of the graphical interface seems to have been taken directly from the well-designed D32XD. Nevertheless, the whole thing is still compromised because of the screen's size and contrast. I haven't had the chance to use Tascam's 2488 (priced the same as the D3200 in the UK), but I suspect that its relatively small backlit LED screen, measuring approximately 2.5 inches square, is also a little inadequate for the mixing and monitoring demands of a 24-track machine.

Inputs & Outputs Most high-end digital multitrackers are designed with lots of output options, so that a commercial project can potentially be sent track-by-track into a larger desk for mixing. The more modestly priced products, such as the D3200, assume that the buyer will want to do everything in one box, so the emphasis is on getting multiple audio signals in, and completed stereo mixes out of the other end. Bearing that in mind, it's no surprise that the input side of the D3200 is the most impressive. A row of XLR sockets allows as many as eight microphones to be connected at once, each XLR having its own 48V phantom-power switch and


Korg D3200

associated status LED. Individual phantom switching is rare on multitrackers in this price range, but it's a definite plus point, as it allows condenser and dynamic mics to be safely mixed in any combination, and removes any necessity for external preamps. Each input channel has its own switchable 26dB pad and level Trim knob, complete with a Peak LED to indicate if signals are being clipped. In line with the XLRs are eight quarter-inch jack sockets that offer an alternative format for inputting audio — no combi jack/XLR sockets here! Next to these are four more jack inputs, taking the total number of analogue inputs that can be used simultaneously to 12. The only remaining analogue input is a high-impedance jack Although the D3200's screen is a quite small in relation to the amount of information it socket labelled Guitar In, designed to allow provides, it can at least be tilted to a variety any bass or electric guitar to be connected of angles to increase visibility. directly without the use of an external amplifier. Helpfully, a copy of this signal is always being sent to the internal tuner, which pops up on screen instantly if the dedicated Tuner button is pressed, and can be calibrated to a reference of your choice. The rear panel of the machine is where the digital connections are located. A pair of optical S/PDIF sockets and a USB connector make it possible to get digital signals in and out of the machine, therefore avoiding the A-D/D-A conversion process. Next to these are the MIDI In and Out sockets, and two footpedal inputs. Certain transport functions can be remotely controlled using a footswitch plugged into the first of the two pedal inputs, while the second exists so that expression pedals can be used to modulate filter effects such as wah-wah. The remaining connectors form a collection of mostly unbalanced jack output sockets situated to the left of the screen, which deal mainly with monitoring requirements. The Master outputs are conceived as a direct feed for external stereo recorders and are designed to take balanced or unbalanced jacks. The neighbouring Monitor outputs are a little more flexible, having a level attenuator and a mute switch all to themselves. There's also an independent headphone output with its own level control. To the left of all these sockets is a pair of Aux outputs that one would typically use to route signals to external multi-effects processors.


Korg D3200

Keeping Up Appearances Although the D3200 is a powerful piece of kit, Korg couldn't resist pulling a few tricks to enhance the product's appeal. A fine example of this are the wooden end cheeks, stained to look like mahogany in order to give the impression of a high-end product. They do have a pleasant tactile quality, but I suspect Korg didn't add them for comfort. The designers have also gone to the trouble of printing several collections of dots on the front panel, which from a distance look very much like the venting grills commonly found on valve desks. Closer inspection reveals that they are just paint!

Fresh Ideas From certain perspectives the D3200 appears to be a cut-down version of the D32XD multitracker (reviewed in SOS December 2003). The former machine was more than twice the price in the UK, being aimed at the semi-pro market, but there are definite similarities in the software interface and hardware design. However, although the D3200 has fewer professional features, it introduces several innovative ideas Korg have not used on a digital studio before. For a start, Korg have obviously had a rethink about how best to offer hands-on control of the various parameters that are displayed on screen. Instead of adding a row of three or four software-assignable parameter knobs along the side of the screen, as many competing products do, Korg have prominently placed a matrix of 16 knobs at the foot of the display. The advantage of having so many Knobs becomes clear pretty quickly. For example, when a channel EQ page is displayed the knobs offer control of all 12 parameters; when in the Pan page they provide instant adjustment of 16 channels at once; and when the Send button is pressed, they become assigned to the effect send level of each track. They're also used for adjusting effects parameters for whatever algorithm is being viewed, and for controlling the individual levels of each of the drum machine's sounds. Korg's other main innovation is a new tool for navigating through screen menus and selecting the options buried within the software pages. They've still retained the page menu tabs at the foot of each window, but instead of positioning a corresponding row of function keys under the screen, selection is done using a joystick, which Korg have proudly trademarked ClickPoint. Just like a mouse, or a laptop's trackpad, ClickPoint controls the movement of an onscreen pointer, and when the stick is pressed like a button it activates whatever menu option is under the arrow at that moment. Those who have not used a laptop may find controlling the pointer a bit fiddly, particularly file:///F|/SoS/SoS%2001-2006/korgd3200.htm (4 of 11)12/19/2005 10:19:40 AM

Korg D3200

as the on-screen graphics are very small. However, there are still four cursor buttons surrounding the joystick providing a more conventional method of navigation across the page, as well as another two buttons labelled Tab Page, which take you back and fourth through the menus at the foot of the window.

The Mixer This D3200 has a surprisingly well-featured mixer section which is only really let down by a lack of channel dynamics, a short travel on the faders (45mm), and the omission of channel delay. Having just two dedicated auxiliary outputs could also be an issue for those who like to use a lot of outboard effects. The main Channel View screen (top) shows the status of all processing and level parameters, as well as providing comprehensive metering facilities. The Channel Routing screen (middle) complements this by giving a graphical representation of current configuration settings, while another dedicated screen provides a detailed view the equaliser curve.

With so many tracks to cope with, it's good to see that there are fader and mute grouping systems, without which controlling 32 channels with two hands would be a nightmare. Channels can be freely assigned to any one of four fader groups and four mute groups, making it possible, for instance, to attenuate an array of backing vocals or drums just by moving one fader. Inputs can be sent directly to recording channels or can be submixed to the output buss for mixing or monitoring purposes. This makes it possible to have 32 tracks of audio playing back from the recorder while a further 12 input sources are being added to the mix. Each mixer channel can be viewed in two ways via the Ch View button. The first window that appears shows the status of all elements in the signal path: EQ, effect inserts, sends, phase reverse, panning, metering, and fader/mute grouping. The second window provides a very informative schematic, illustrating how everything is interconnected, and, as far as I can tell, all of this is exactly as it was on the D32XD. The rest of the hardware mixer-related buttons all lead to a set of menus where particular mixer functions can be edited globally for all tracks and channels. These are labelled Send, EQ, Pan, and Effects, although there's also a Mixer button leading to the settings that are not adjusted quite so often — it is here, for example, where the onboard fader and pan automation is reached. Although there are no channel dynamics, EQ is provided and offers four sweeping filter bands, each with ±15dB of gain and a Q control for all of the first 24 tracks. The bands file:///F|/SoS/SoS%2001-2006/korgd3200.htm (5 of 11)12/19/2005 10:19:40 AM

Korg D3200

are labelled Low, LowMid, HighMid, and High and all have a sweep range of 21Hz20.1kHz, with Q values from 0.1 to 10. The Low and High bands also become shelving filters if the Q parameter is turned fully to the right. Monitoring the EQ can either be done directly through the EQ pages, or from within the Ch View screens, and in both cases Korg have used a graphical representation of the curve to show the results of any adjustments. Tracks 25-32 and those of the input submixer have only two EQ bands and no Q control.

Internal Drum Machine

Drums are not the easiest things to record, so it's no wonder then that the D3200 and other similar products include rhythm machines in their feature set. Korg's Session Drums drum machine has its own fader and mixer channel and, when activated, makes use of the four locate buttons for transport control and pattern triggering. A variety of kits are used to create patterns, and patterns can be chained together to form a complete arrangement using a dedicated Pattern Map sequencer. Programming the drum machine is a simple matter of selecting the most appropriate pattern and then changing its feel using the knob matrix to alter Accent, Human, and Shuffle parameters. Each drum and percussion instrument is also assigned its own volume knob, so its relative balance is adjustable, but further tweaking of the tuning and panning can be achieved in another editing window. The drums even have a global two-band EQ and a routing page to determine where they are inserted into the mixer's signal path. The only real drawback with the facility is that you are forced to use preset patterns and fills as the basis of a composition, because there is no editing of individual drum events, which is a pity. The manual seems to suggest using the drums as a guide for a real drum performance, or in addition to real drums. Korg tell me that they sell lots of multitrackers to guitarists, so perhaps the idea is for guitarists to record to a guide pattern and then call in the drummer later on. Nevertheless, seeing as so much can be adjusted to taste, surely a method of creating custom patterns would have been a worthy inclusion?

Smart Editing No multitracker's editing toolbox would be complete without erase, delete, insert, compress/expand, copy, optimise, and track-swapping options, and Korg have included all the above dutifully, but amongst the D3200's 12 editing tools are some file:///F|/SoS/SoS%2001-2006/korgd3200.htm (6 of 11)12/19/2005 10:19:40 AM

Korg D3200

valuable options that are not quite so commonly included. For a start there is a Reverse processor, enabling any section of audio to be selected for treatment. Many computer editors offer reversal as an option, and countless psychedelic records were made by playing sections of tape backwards, yet I only recall seeing this facility on Roland multitrackers before now, so it is great to see Korg are getting in on the act. Fade in/out is also rarely seen amongst the digital editing tools of a hardware multitracker, which is a shame, because, although fades can be done using automation, affecting the audio at source is often a much neater way to work. Korg's Fade editor is good, in that it offers a variety of curve shapes to choose from. Another unusual multitracker option is the Noise Reduction facility. This editor is clever enough to be able to analyse a specified area of recorded background noise, learn its characteristics, and then remove it from the rest of the track. Korg have even added something called Erase Punch Noise that homes in on plosive noises and reduces their impact. One other feature worth a mention is the very nicely designed Waveform dialogue box. Most competing products do have waveform displays, but this one has a stereo button that selects the waveforms of two adjacent channels so that left- and right-channel recordings can be seen side by side. There's also a really useful Search Zero button, allowing you to find the nearest zero-crossing point, where an edit will tend to be least noticeable. It's certainly worth playing around with some of the tools, because there are 16 levels of undo/redo to fall back on if you make a mistake.

Built-in Effects The D3200 is well equipped with effects, although you do have to plough through a menu or two before you can start adjusting the algorithms. The main Effect button takes you to a set of menus that relates to all aspects of effect assignment. By default, no effects are selected, so it's a case of deciding whether you want to create an Insert, Master, or Final effect patch, and then calling up the Select Effect Category page. Here there are two lists relating to mono and stereo algorithms, and these are subdivided into Reverb & Delay, Modulation & Pitch, Dynamics & Filter, SFX, and so on. An extra Multi option only appears in the mono list, and this contains the guitar effects chains. The maximum number of algorithms that can be used simultaneously is 11, including eight Insert, two Master, and one Final effect. The two Master effects are basically on a send/return loop, and Korg expect these to be used for global reverbs and special treatments. The Final effects are for mastering and are inserted into the path of the stereo output buss. The remaining Insert algorithms can be used on the input signals while recording, or in the path of a mixer channel during playback. Thankfully, channel file:///F|/SoS/SoS%2001-2006/korgd3200.htm (7 of 11)12/19/2005 10:19:40 AM

Korg D3200

EQ hasn't been classed as an effect, but dynamics processors such as compressors, limiters, and gates all have to be sourced from the effects. Having just eight insert effects for 32 channels doesn't seem a lot if you like to use a lot of compression, and yet there are further limitations to consider. For example, you only get eight when exclusively using what Korg call Size 1 effects. Most of the mono reverbs and delays are Size 1, but stereo reverbs and compressors are Size 2 and there's even a whopping Size 4 multiband limiter.

Korg's intriguing ClickPoint device resembles a trackball on first inspection, but actually works a bit like a miniature joystick, springing back to its central position when released. Once the pointer is positioned over the item you want to select, pressing the top of ClickPoint selects it.

The Multi guitar effect chain counts as all eight Insert effects, as it provides a chain of individual effect blocks that can be switched in and out of service as required. Of particular interest to guitarists will be the amplifier and cabinet models, each with controls for drive, volume, bass, middle, treble, presence, and a noise gate. Featured amps include the Vox AC15 and AC30, various unspecified tweed and boutique models, plus some blues, rock, and metal types. There are a similar variety of cabinets, including stacks, small combos and twins. After an amp type is selected, a new window appears showing some graphical representations of the effects in the chain. Clicking on one calls its parameters to the window below where they can be immediately adjusted using the knob matrix. The configuration makes programming fast and easy, and the Store, Rename, and On/Off buttons elsewhere on the page bring all the relevant functions together neatly. Incidentally, Korg have used the same modelling algorithms on numerous other products, so there are no surprises, but it has to be said that they act as a pretty good replacement for the real thing in many recording situations. As there is a guitar DI input and a tuner, everything necessary for guitar recording is here other than the guitar itself.

Backups & USB Before you can back up any data, either to CD or via USB, the composition in question has to be saved to a part of the internal hard disk called the PC Drive. I found that for a 611MB song file I'd been working on this process took about six minutes. Once a task is complete everything becomes very easy indeed. Getting the multitracker connected to my computer was an almost instantaneous process, and from there it was merely a matter of opening the project file with my PC and copying it across. This time the 611MB file took only something like 20 seconds to copy over, which is very fast. To save the same file to a CD-R the D3200 has to first generate a disc image file, after which the machine automatically begins burning the backup CD-R.


Korg D3200

Performance In Practice Overall, the D3200 is quite easy to operate, and is certainly one of the most userfriendly products currently on the market. Potentially complicated processes like building a drum-machine performance prove to be relatively easy when compared to the methods used by some of the competition. It's also pretty easy to automate dropins, add marker points, and operate mixer features such as track solo. On a more general level, the machine works very quickly, showing no obvious processing delays in any of its operational modes. It starts up rapidly and shuts down even faster, saving the current setup as it goes. The recorder does have a tendency to make a high-frequency whine when it's first switched on, although it soon settles down and generally the recorder is very quiet. It seems that the days when noise was a serious problem in multitracker design are long gone. As far as the sound is concerned, the recording quality is very good. The dynamics processors are effective and usefully programmed, and the effects sound reasonable. You'd probably want to use a dedicated reverb if you had a TC Electronic or Lexicon processor to hand, but the onboard algorithms are enough to be getting on with.

Detailed editing of audio and automation data is facilitated by a waveform view (left) and detailed automation editor (right).

Although the machine does have a lot of on-screen controls, Korg have got the balance between software and hardware control about right. There are hardware file:///F|/SoS/SoS%2001-2006/korgd3200.htm (9 of 11)12/19/2005 10:19:40 AM

Korg D3200

buttons to take you to all the main operation pages, and once you're there the relevant software options are gathered together logically. The software itself, coupled with the implementation of ClickPoint, really makes you feel like you're working in a Windows OS environment — it has that kind of menu-driven look and design. Korg's knob matrix is also success, and a great idea. In fact the only operational problem is the tiny screen, measuring just a couple of inches across, which severely compromises the usability of features like the matrix. Granted, the resolution is very high, so the graphical elements are clear, but they're still tiny! The size problem is most apparent when the channel and metering pages are selected — there just isn't space for all the information to be included at a sensible size. Having a five-position tilting screen is a nice touch, but that too is undermined by the screen's contrast, which is not great no matter how it is adjusted. Even at an extreme setting the blacks look grey, making it harder to detect what parameter is highlighted for adjustment.

Multitrack Monster The D3200 is a pretty impressive product, albeit with a few significant flaws which threaten to undermine the rest of the designers' good work. Korg might have got away with the small screen if it was on a much simpler machine, but for a 44-channel mixer it is way too small, and no matter how you adjust the contrast knob it never seems to have enough definition. The Irony is that the software itself has been carefully designed to be as user-friendly as possible — as demonstrated by the extensive use of pictures to illustrate many of the functions. I'm sure it all looked great on the software developers' computer monitors, and must have suited the D32XD (for which it was originally intended), but here eye strain is a definite possibility! Incorporating a larger screen, or a socket for connecting a monitor, might have added to the cost, but it would have made a dramatic difference. It would have been nice to have had dynamics on every channel, as on Yamaha's AW machines, particularly as that would have freed up the effects processors for other things. It's also a real shame that the compressors and gates don't have a sidechain key option — professional engineers use triggering all the time, so this is a missed opportunity to widen the product's potential. The short-throw faders are another negative aspect of the machine — they make small Guitarists will find the onboard tuner handy when recording directly into the D3200. adjustments difficult, especially if something needs to move by nothing more than one decibel. It is possible to change level in tenths of a decibel on screen by turning the data wheel, but grabbing a fader is more intuitive. The halved track count in 24-bit mode, the restricted EQ on some mixer channels, and the lack of a pattern editor in the file:///F|/SoS/SoS%2001-2006/korgd3200.htm (10 of 11)12/19/2005 10:19:40 AM

Korg D3200

drum machine must also count against this unit. The D3200 does score highly in other areas though. The USB and CD-RW facilities are very well integrated into the operating software, the editing options and the array of signal inputs are impressive, and there seems to be a hardware button or software page for everything you want. ClickPoint is a nice alternative to the touchscreens Korg have tended to use in the past, and the knob matrix works effectively for controlling a whole range of different parameters. I can't think of another multitracker with anything quite as good. There's no denying that a lot of effort has gone into the design of this product, but I feel that a few too many cuts were made to break the £1000 barrier in the UK. Having 32 tracks is a big selling point, but I would have settled for a 24-track machine with a bigger screen, longer faders, and more dynamics processors. Nevertheless, if you have good eyesight, you should still be able to get some great results with this machine. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



M Audio Project Mix I/O

In this article:

Built To Last? 1814 Overtures Project Mix & Cubase MIDI Mode Project Mix & Ableton Live Project Mix & Pro Tools What Else Is Out There? Final Analysis

M-Audio Project Mix I/O £760/£950 pros Superb value at its introductory price. Eight mic preamps, and 18 (eight analogue plus eight ADAT or stereo S/PDIF) audio inputs. Nearly complete Mackie Control emulation. Faders and buttons can be configured to send any MIDI CC or note value.

cons After January 15th, it will only be available as a £950 bundle with Ableton Live. Not expandable at present. Documentation could be better. The utility used to configure the Project Mix's MIDI mode is a bit crude in its current form.

summary The Project Mix combines an 18-input audio interface, an eight-channel preamp and the near-equivalent of a Mackie Control — and if you move fast, it's a bargain at £760.

information £759.99 until January 15th; then £949.99 with bundled Ableton Live. Prices include VAT. M-Audio +44 (0)1923

M Audio Project Mix I/O Firewire Interface & Control Surface [PC/Mac] Published in SOS January 2006 Print article : Close window

Reviews : Computer Recording System

M-Audio have packaged a fully featured control surface with motorised faders, an 18-input Firewire interface and eight mic preamps in one box — at a very competitive price. Sam Inglis

When the Project Mix I/O arrived at the SOS office, we all thought 'Blimey! It looks just like the Digi 002!' Digidesign and M-Audio are now part of the same empire, so it seemed logical that they might have pooled their resources to create new hardware, and both products are variants on the same basic concept. Like the 002, the new unit combines a multi-channel Firewire audio interface with a control surface based around touch-sensitive, motorised faders, and can be used with Pro Tools or any other major DAW.

Photos: Mark Ewing

However, M-Audio say that the Project Mix has been developed independently of Digidesign, and on closer inspection, it turns out to be quite different from the 002. In many ways, in fact, it turns out to be better. The Project Mix features word clock I/O, it has eight mic preamps rather than four, it has a shuttle wheel and master fader in addition to the eight channel faders, and there are two headphone outputs rather than one. True, its rotary encoders lack the LED 'ring' displays featured on the 002, and it can't be used as a stand-alone mixer — but on balance, the Project Mix is definitely the more feature-rich of the two. Another difference is that it uses the Mackie HUI protocol to communicate with Pro Tools, rather than the system Digi themselves developed for the 002 and

file:///F|/SoS/SoS%2001-2006/maudiopmio.htm (1 of 9)12/19/2005 10:19:45 AM


204010. +44 (0)1923 204039. Click here to email www.maudio.co.uk www.m-audio.com

Command 8. The Project Mix also supports the Mackie Control and Logic Control protocols which are implemented by most of the other major sequencers, and is clearly intended as a universal controller and audio interface. The Mackie Control and some similar products can be expanded by the addition of 'sidecar' units that provide extra fader banks. M-Audio say that they have no current plans to offer such an expansion for the Project Mix, which is a pity.

Test Spec M-Audio Firewire driver version 5.10.0.5035x25. Project Mix firmware version 10.19.05a. Control utility version 1.00.5. Intel Centrino laptop with 2.0GHz Pentium-M CPU and 2GB RAM, running Windows XP SP2.

The review unit didn't come with any music software at all, and until January 15th, this is how the Project Mix will be sold. After that, however, it will only be available as a bundle with the full version of Ableton Live, with the price rising from £760 to £950. To my mind, this is bizarre. Some Project Mix buyers will already own Live, and the rest would surely prefer to choose their own recording software — after all, one of the Project Mix's big pluses is that it works with everything. And if it has to be bundled with something, why Live? Pro Tools MPowered, an M-Audio product, would be much more appropriate for the multitrack recording jobs that are the Project Mix's bread and butter.

Tested with Steinberg Cubase SX v3.0.2, Ableton Live v5.0, NI B4 II, Digidesign Pro Tools M-Powered v7.0.

Built To Last? Given its low price, you might expect M-Audio to have cut some corners in the construction of the Project Mix, but it's actually very solid. At 20 inches wide by 18 deep, the case is quite large, and feels very substantial, with rigid moulded plastic surrounds and a metal surface that doesn't flex under pressure. The motorised faders, perhaps the most important components, are reliably touchsensitive, acceptably smooth and quiet, and are full-length 100mm devices rather than the 60mm efforts some companies inflict on us. Some manufacturers of cheap control surfaces save money by not including a display, but the Project Mix's two-line LCD is clear and bright. There are no dedicated meters, but in most applications the LCD can be used to display channel levels. Cost-cutting rears its head only in a few details: the plastic gain knobs and rotary controllers look cheap, the laptop-style external power supply doesn't inspire confidence, and the shuttle wheel wobbles a bit. But hey, at least it's got a shuttle wheel, unlike many budget controllers. Six-pin to six-pin and six-pin to four-pin Firewire cables of decent length are included. Installing the drivers from CD was straightforward, and although it generated a couple of error messages while attempting to uninstall my existing M-Audio Firewire drivers, worked first time. The Project Mix comes with a very brief, foldout Quick Start Guide and a PDF manual. The latter is not exactly comprehensive, and although it tells you the function of each control, you will need to consult your sequencer's own documentation to learn how it works with a control surface.

1814 Overtures



In terms of its capabilities as an audio interface, the Project Mix I/O is almost identical to another M-Audio product, the Firewire 1814: so similar, in fact, that I won't go into details here, but will refer anyone who's interested to Martin Walker's review of that unit in SOS October 2004 (www.soundonsound.com/sos/ oct04/articles/maudio1814.htm). As on the 1814 there's an assignable level controller, which can be set to adjust various important signals such as the level of the main stereo output, and an A/B switch which can either be used to turn direct monitoring on and off, or to switch the source of the first headphone output. Apart from the analogue gain controls and Mic/Line switches, the Project Mix's front panel offers only very limited control over the its audio interface parameters, most of which must be adjusted in software from with the Control Panel utility. Where the Project Mix improves over the 1814 is in its analogue circuitry: there are eight rather than two mic preamps, there's a high-impedance jack socket for connecting electric guitars directly, and all eight line inputs are on balanced jacks. I noticed no difference at all between the sound of the 1814 and the Project Mix. The preamps are clean and quiet, with plenty of gain, the guitar DI socket worked as expected, and the headphone outputs put out a decent level. A vintage Neve desk it ain't, but I think it's fair to say that most of those who will buy a Project Mix are unlikely to find themselves limited by the quality of its preamps or converters. In any case, you can always attach third-party preamps and converters thanks to the ADAT I/O.

Project Mix & Cubase At launch, the Project Mix had three control modes, for Cubase SX, Logic and Pro Tools, and a firmware update shortly after release added support for Digital Performer, Sonar and Ableton Live. However, Pro Tools users had to wait until the launch of version 7 to access the Project Mix's audio interfacing. PT7 is reviewed elsewhere in this issue, but as the M-Powered version wasn't available until late in the review period, I began my tests with Cubase SX instead. As far as Cubase is concerned, the Project Mix I/O has nearly the same functionality as the Mackie Control it emulates. That is, it's of limited use for setting up projects or editing audio and MIDI, but it's very useful for tracking and mixing. Things that you can do easily include setting levels and pan positions, locating the Project Cursor, nudging, looping, scrubbing, setting Markers and in/out points, soloing and muting channels, global vertical and horizontal zooming, track arming and recording, setting FX send levels, making real-time parameter changes to software instruments and effects, selecting Events on the Arrange page and toggling between the Arrange and Mixer pages. Things that you can also do, but with more difficulty,



include selecting, inserting and editing audio and MIDI effects and VST Instruments. A list of what you can't do at all would include creating, naming and deleting tracks, choosing input and output routings, accessing off-line audio or MIDI processes, opening or using Editor windows, zooming individual tracks, and changing tracks' automation status (this is possible on Mackie Control, but not here).

As is the case with most control surfaces, many of the buttons take on different functions in different applications, but the labelling seems to be based primarily on Pro Tools. For instance, the five Aux buttons allow you to set the levels of sends 1 to 5 in Pro Tools, but work differently in Cubase.

Unless the Flip button is engaged, the faders always control (and reflect) the positions of the faders of the eight channels in the current bank, while buttons adjacent to them shift this selection up or down by a single channel and in banks of eight. Touching a fader automatically selects that track within Cubase, which is sometimes what you want, but not always: there are times when you want to be able to adjust the volume of other tracks without having to remember to return the focus to the one you're recording on. The rotary encoders default to panning duties for the eight channels in the current bank, but five Aux buttons switch them to controlling the eight FX sends for the selected channel, selecting that channel's insert effects and switching them off or on, doing the same for the first eight slots in the VST Instrument rack and Master effects, and controlling the EQ on the selected channel. There are no dedicated Page up and down buttons, but in all of these modes, holding Alt and pressing the Bank Select buttons allows you to access deeper-level parameters, such as controls for the plug-ins you've selected. When I first switched the Project Mix on, the shuttle wheel worked backwards. However, hitting the Setup button accesses various housekeeping functions, one of which is Jog Wheel Calibration, and after I'd visited this, everything was normal. There are things I don't like about the way Cubase deals with the shuttle wheel, but these are Steinberg's fault rather than M-Audio's. The most annoying is that if you have Snap to Grid enabled in Cubase, but you stop the transport with the Project Cursor between grid lines, the shuttle wheel will retain that offset from the nearest grid line rather than actually snapping. One or two of the Project Mix I/O's buttons have no function within Cubase, but in general, the mapping of available controls onto functions is sensible and logical. Although there are separate Alt and Shift buttons, most commonly used functions can be accessed without them, and they're reasonably well placed at the top between the rotaries and the Aux buttons. The Project Mix does lack some Mackie Control features that are supported in Cubase, including the Fader Groups buttons, the user-definable Function keys, the Solo Defeat button, the LED song position display, the channel activity and Rude Solo LEDs and the ability to deactivate the fader motors; the ones I missed the most were the automation Read and Write buttons and the Undo and Redo keys. None of these is vital, however: the latter functions are easily accessed from the QWERTY keyboard, and thanks to the touch-sensitive faders, you can put Cubase's automation into Touch mode and leave Read and Write permanently enabled for



all tracks. On the plus side, the Project Mix I/O's labelling is sometimes more logical than that of the Mackie Control, and of course there's no need to connect it to a separate MIDI port on your computer — all the control data is handled via the Firewire connection. As with most control surfaces, some of the tasks that are possible with the Project Mix I/O in Cubase are so much easier from the computer keyboard that you'd have to be very determined to do them any other way. For instance, you can use one of the rotaries to select which VST Instrument is installed in which slot in the rack, but you can't select that VST Instrument as an output for your chosen MIDI track, and this is a setting that needs to be made every time you select an Instrument. With VST plug-ins that report their parameter names back to the host, you can have a lot of fun making real-time parameter changes: however, not all plug-ins are well-behaved in this respect, and the likes of Sampletank quickly sent me back to my QWERTY keyboard. Also, for some reason, not all the VST plug-ins on my computer showed up as available for selection from the Project Mix I/O, although I could choose them with the mouse from the insert slots on screen.

MIDI Mode As well as having preset modes dedicated to specific applications such as Cubase and Logic, the Project Mix also acts as a fully configurable MIDI controller. Hit the MIDI button on the control panel and launch the Control utility from the Start menu, and you will be greeted with an on-screen replica of the Project Mix. Hovering the mouse pointer over any button, fader or rotary controller will tell you what MIDI data it is set up to output, and on what channel. Clicking with the mouse then allows you to change this assignment. The faders, rotaries and shuttle wheel can output any MIDI Continuous Controller value (but not NRPNs), while the buttons can send any MIDI Note On at any velocity value. When you've set the on-screen version up to your satisfaction, you can then dump its settings into the hardware Project Mix, which should remember them. The Control utility can also be used to update the Project Mix's firmware. This could be a fantastically useful facility for those who want to control a piece of software that is not directly supported by the Project Mix. To test it, I used the Project Mix's Control utility to assign the preset drawbar CC values in Native Instruments' B4 II to the faders: it was quicker and easier than using B4's MIDI Learn feature, and the results worked perfectly. However, there are areas where the Control utility could be improved. The dialogue boxes where you enter CC or note values are wrongly named: for instance, clicking on a fader tells you that you're entering CC values for a Mute button. More importantly, there's no way to save or load Project Mix setups within the utility, and nor can you store more than one user setup within the Project Mix itself. It would be nice to be able to set up a



number of different templates for different applications, but as things stand, this is not possible. It would also be handy if there was a way to switch all the parameters to a different MIDI channel in one go — opening 60-odd dialogue boxes in order to do this takes forever. Hopefully, M-Audio will continue to develop this utility and add some of these features in a later version.

Project Mix & Ableton Live Support for Mackie Control was added in the latest version 5 of Ableton Live, and provided you have recent enough firmware, you can put the Project Mix into Live mode by holding down Aux 5 as you switch on. All you have to do then is visit the MIDI/Sync section of Live's preferences and tell it that a Mackie Control is attached. Live mode wasn't mentioned in the version of the Project Mix I/O manual I saw, but with the aid of Live's own documentation it's not too difficult to work out which button does what. The Window button toggles between Session and Arrangement views, and once again, the basic Pan mode sees faders and rotaries used to control track level and pan, with the record arm, select, solo and mute buttons doing what you'd expect. And just as for Cubase, you can change the focus to access settings that are specific to a channel, such as effects sends. Unlike Cubase, however, Live's Mackie Control implementation depends heavily on a feature that M-Audio have left out of the Project Mix: the ability to use the rotary encoders as momentary buttons. For instance, pressing the Plug-in button shows the Devices that are active on the selected track, and according to the Live manual, you should be able to select a Device for editing by pressing the corresponding rotary encoder. Since the Project Mix's rotaries only respond to twisting, and Live wasn't mentioned in the documentation, I thought that this wasn't possible. Just as we went to press, though, M-Audio's technical people told me that you can select Devices by holding down the Alt key and pressing the channel's Sel key. It was too late for me to test this arrangement, but it seems logical, and should make the Project Mix a good companion for Live. I still think Live is an odd choice to bundle with the unit, though!

Project Mix & Pro Tools Version 7 of Pro Tools M-Powered finally arrived towards the end of the review period, so I was keen to try it out with the Project Mix. To set Pro Tools up to be controlled from the Project Mix, you simply tell it that an eight-fader HUI is attached. As you might expect, M-Audio have put a lot of thought into the way that the Project Mix integrates with Pro Tools, and the way the buttons are labelled suggests that the designers worked with this application in mind above all. You could also argue that Pro Tools is intrinsically more suited to hardware control than some other software, thanks to its simple two-window interface. Either way, you can certainly do everything that's possible in Cubase, and more. It is, for instance, straightforward to choose input and output routings for tracks in Pro Tools, while the In and Out buttons allow you to make selections on the fly.



The Project Mix's shuttle wheel does nothing in Pro Tools unless either the Zoom or Scrub buttons are pressed. When the Zoom button is unlit, the arrow keys have the same navigation The Project Mix I/O improves on M-Audio's Firewire 1814 interface by offering eight mic functions as the 'P', semicolon, 'L' and preamps and balanced line inputs. apostrophe keys in Command Focus mode. With the Zoom button lit, they control global horizontal zoom, and vertical waveform zoom for all audio tracks (it would be much more useful if they controlled track height, preferably on just the selected tracks, but there you go). Pressing the Zoom button again makes it flash, whereupon you can use the left and right arrow keys in conjunction with the shuttle wheel to select Regions or horizontal sections of a Session. The Scrub key, meanwhile, cycles Pro Tools through its two scrubbing modes. Despite the slightly wobbly feel of the shuttle wheel, I found it perfectly usable for locating edit points. It took me a while to get my head round the way that plug-in editing from the Project Mix works in Pro Tools, but it is possible. Once you have worked it out, it's slightly more practical than in other applications, though in my view it's still easier to use the mouse. Again, some third-party plug-ins either have too many parameters or don't report their names properly, but at least Digi's own Digirack processors are well-behaved. Oddly, there's only room to display four plug-in slots on the Project Mix's screen at once, and you have to visit a separate page to bring up the fifth. It's also a shame that inserting and selecting plug-ins from the Project Mix doesn't automatically bring up their editing windows in Pro Tools. The main point of a control surface is to help out during mixing, and in this respect the Project Mix works very well in conjunction with Pro Tools. It's easy to automate track levels, send levels and pan for mono tracks, and quite easy to automate plug-in parameters, although you have to select them for automation first within the plug-in window. Again, the fact that you can't change tracks' automation status from the control surface isn't too much of a problem: you can simply put all tracks into Latch mode for an initial pass and then switch them to Touch mode for fine-tuning. With Pro Tools 7 being a major new version, there are a few areas that suggest a game of catch-up between M-Audio and Digidesign's development teams. For instance, pressing Alt plus the left arrow key is supposed to return the cursor to the Session start, but sometimes doesn't work. Elsewhere, the Project Mix's graphical display of pan positions went awry occasionally, and I also managed to crash PT7 on one occasion by hitting Scrub while Zoom mode was active, although I couldn't repeat this reliably. I didn't encounter any serious problems, though, and I'm sure that any small inconsistencies will soon be cleared up by service updates from Digidesign.

What Else Is Out There? file:///F|/SoS/SoS%2001-2006/maudiopmio.htm (7 of 9)12/19/2005 10:19:45 AM


When it was first announced, M-Audio set a recommended retail price of £759.99 for the Project Mix I/O. This later turned into an 'introductory offer', but if you can get it at this price, it represents excellent value. It's lower than the current street price of the Mackie Control Universal — and although the Project Mix lacks one or two of the Mackie Control's features, the Mackie Control doesn't have eight mic preamps and an 18-input Firewire audio interface! To get close to the Project Mix's functionality you would need to add something like a Presonus Firepod, taking the total cost well over £1200, and even then you wouldn't get ADAT or word clock I/O, nor the advantages of having your audio interface, monitoring controls and fader surface integrated in one box. There are, of course, other one-box products in this vein, and the Project Mix compares well to most of them, too. The Digi 002/Pro Tools LE bundle has fewer features and still costs over £1500 at current street prices. Yamaha's 01X is another possible alternative: it boasts built-in DSP mixing capabilities and comes with a decent software bundle, but lacks ADAT I/O, has 60mm rather than 100mm faders and has only two XLR mic inputs. It's now available at under £800 on the street, which is also good value, but perhaps suggests that the mLAN protocol hasn't taken off in quite the way Yamaha intended. The most direct competitor is probably Tascam's FW1884, which was launched two years ago at £1299 but is, again, available at under £800 in some shops. Like the Project Mix I/O, the FW1884 combines an 18-input Firewire audio interface with eight mic preamps and a nine-fader control surface based on the Mackie Control and HUI protocols. In some respects, the 1884 has the edge: it features insert points on the analogue inputs, eight analogue line outputs rather than four (making it suitable for 5.1 surround monitoring without extra hardware), and four MIDI Ins and Outs to the Project Mix's one. It can also be used as a stand-alone mixer, and can be expanded with a sidecar called the FE8, each of which adds an extra eight faders. However, it lacks any kind of display — which is important if you want to use a control surface for more than track levels and panning — and implements a slightly different set of Mackie Control features. There is no way to edit plug-in parameters from the 1884, but unlike the Project Mix, it has dedicated buttons to set automation modes, plus editing features such as cut and paste, and undo. Oh, and of course the 1884's audio interface won't work with Pro Tools.

Final Analysis In the past, I haven't found small control surfaces for digital audio workstations useful enough to justify their expense. It's not that products such as Mackie Control and Digi 002 are overpriced for what you get — after all, something with that many buttons, pots, faders and displays is never going to be cheap to make. It's more that even with all those controls, you never get enough functionality to leave the QWERTY keyboard behind, and the intuitiveness of hands-on mixing is eroded by endless bank switching and shifted key combinations. file:///F|/SoS/SoS%2001-2006/maudiopmio.htm (8 of 9)12/19/2005 10:19:45 AM


However, products like the Project Mix make Mackie Control-style mixing so affordable that even the diehard sceptic will have to think again. At its £760 introductory price, it would be hard to find a competing product with just the preamps and audio interfacing, let alone a fully featured control surface. You could think of the Project Mix as a nice audio interface with a free hardware controller! Even if you only ever used the transport buttons, this would be good value, and if you're one of those people who hates mixing by mouse, it's a bargain. It will still be pretty competitive even when the price reverts to £950 on January 15th, but I think there are rival manufacturers who will be very relieved! Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Mackie Onyx 400F

In this article:

Features Overview Rear-panel Socketry Bundled Software Onyx Opinion Verdict

Mackie Onyx 400F Firewire Audio Interface Published in SOS January 2006 Print article : Close window


Mackie Onyx 400F £704 pros Extremely good audio quality. Easy to use. Full version of Tracktion 2 included.

Mackie's new 10-in, 10-out breakout box includes and internal DSP mixer, MIDI connectivity, and four of their high-spec Onyx preamps.

cons No ADAT expansion, but you can already daisy-chain two 400Fs under Mac OS 10.4 if need more I/O. Currently you can't mix all your DAW streams within the integral DSP mixer.

Paul White

When Mackie introduced their Onyx mixer range, the mixer's newly designed mic preamps attracted a lot of favourable comments. These summary preamps appear to have been Mackie have succeeded in designed to subtly flatter the sound in Photos: Mark Ewing combining their excellent the same way that many of the popular Onyx preamps with a flexible but simple DSP routing/mixing vintage preamps do, while offering a wide dynamic range (123dB) and very low distortion (0.0007 percent THD). system in a stylish Firewire When I reviewed the Onyx mixer, I was particularly impressed by the sense of audio interface. clarity and detail these preamps presented, and they come very close in information performance (both subjectively and technically) to some of the extremely £703.83 including VAT. expensive and esoteric boutique mic preamps currently available. The Onyx Mackie UK +44 (0)1268 mixer also had a Firewire option allowing it to be used as an audio interface to 571212. computer music systems, so it wasn't entirely unexpected when Mackie +44 (0)1268 570809. announced a stand-alone Firewire interface based around their Onyx preamps. Click here to email www.mackie.com

Test Spec Dual 2.5GHz Apple Mac G5, with 4GB of RAM, running Mac OS X Tiger. Mackie Onyx 400F firmware v1.3.1. Mackie Console configuration software v1.02.

Features Overview Housed in a conventional 1U rack case and requiring only Firewire 400 connectivity to hook up to a computer, the Onyx 400F comprises four Onyx mic/ line preamp channels augmented by four further line-only channels, where the mic/line channels also benefit from analogue insert points on TRS jacks. Nextgeneration AKM 24-bit/192kHz A-D and D-A converters are used to maintain the audio quality of which the Onyx preamps are capable. There are eight balanced

file:///F|/SoS/SoS%2001-2006/mackieonyx400f.htm (1 of 5)12/19/2005 10:19:54 AM

Mackie Onyx 400F

line outputs, S/PDIF stereo digital I/O (coaxial), and word-clock I/O on conventional BNC connectors. Unlike some competing products, though, there's no ADAT I/O, which rules out adding more I/O channels without adding another interface. (Multiple interfaces can be used together on Mac OS X 10.4, and Mackie are working on this capability for Windows). According to the spec, special DSP tweaks enable particularly fast audio transfer, and hence lower latency. Both Mac OS and Windows are supported at 24-bit resolution only and at up to 192kHz sample rate. Both the analogue and digital I/O can be used together for a maximum of ten simultaneous inputs (mixed to five stereo pairs) and all 10 outputs from the DAW are routed directly to the 10 physical output jacks of the 400F when the integral DSP mixer is switched off. The DSP mixer just alluded to is a 10 x 10 DSP device that can route any input directly to any output with negligible latency, and which can also be used to set up five different custom stereo mixes, making it ideal for monitoring. This mixer has the ability to save and recall settings, and uses 64-bit floating-point processing to maintain the best possible signal resolution. In effect, the use of the floating-point maths provides more mix headroom, which is particularly important when summing signals. However, it must be noted that this mixer can only mix the 10 inputs (eight line and two S/PDIF) plus the stereo DAW output — it can't be used to sum ten separate outputs from the DAW mix, which is perhaps missing a trick if the mixing engine is really that good. The mixer can also currently only output as stereo pairs. When overdubbing, the 400F's dual headphone outputs provide both the engineer and performer with their own headphone feeds (both based on the control-room mix or the output 7+8 mix) with independent volume control. Those who do not need the extra capabilities of the DSP mixer can switch it off in the software, whereupon the 400F becomes a straight 10-in, 10-out audio interface, with all 10 inputs feeding ASIO or Core Audio streams 1-10, and DAW outputs 110 feeding the 10 output jacks on the back of the 400F.

Rear-panel Socketry In addition to its I/O capabilities, the Onyx 400F also functions as a monitor level controller and a single-port MIDI interface. The two independently adjustable headphone outlets are on the front panel, along with instrument input jacks for channels one and two. All the other connections are on the rear panel, where rear-panel combi jack/XLR sockets handle the mic/line channels and conventional balanced quarter-inch jacks take care of the line inputs and line outputs, including a dedicated stereo control-room output. This keeps the unit very tidy, but it does mean that if you want to switch between mic and line inputs on the first four channels, you need to be able to access the rear of the unit. Mains power comes in via the usual IEC socket, and there are two six-pin Firewire ports to facilitate device chaining.


Mackie Onyx 400F

The front panel is simply set out, with clean, stylish lines, and using green LED indicators to the left to show the clock source, Firewire status, and MIDI I/O activity. A single knob adjusts the control-room output level, with two further knobs to adjust the phones levels. Right of centre are the four mic/line input channels, the first two with switchable high-impedance instrument inputs on unbalanced jacks, and all four channels have simple four-LED level metering at 40dB, -20dB, -10dB, and Overload. A single 48V button applies phantom power to all four mic inputs when active.

Bundled Software Bundled with the Onyx 400F is a full version of Mackie's own Tracktion 2 Audio + MIDI sequencer (reviewed in SOS August 2005), which combines a practical level of flexibility with ease of use. Although the boxed retail version is bundled with extra plug-ins, the bundled version is otherwise identical as far as its sequencer section is concerned. Tracktion 2 is available for both Mac OS and Windows, though some Mac users may prefer to use Garage Band (bundled with all new Apple Macs), which is also relatively straightforward to use. The minimum PC computer specifications for running the Onyx 400F and the Tracktion 2 software are Microsoft Windows XP (SP1) or later running on a machine with a Pentium 4, Celeron, or Athlon XP processor. A minimum of 256MB of RAM is specified, though at least twice that would be desirable. Mac users must be running Mac OS 10.3.9 on a G4 or more powerful machine, again with at least 256MB of RAM. The control settings for the Onyx 400F are handled via further software that comes with the unit. In addition to setting the sample rate and digital synchronisation source, you can set up the DSP mixer which makes it possible to create up to five stereo pairs of low-latency mixes at the outputs, based on the line and S/PDIF inputs and independent of the DAW software. The stereo DAW mix can also be added to the monitor mix. If not needed, the DSP mixer can be switched off, and the last setting is remembered even when the unit is not connected to a computer.

Because the 400F remembers the latest state of the DSP mixer when the console is off and the computer is removed, it can act as a stand-alone mixer or four-channel mic preamp. Suggested applications in this mode include using the 400F as an eight-channel line-level and two-channel S/PDIF rackmount keyboard file:///F|/SoS/SoS%2001-2006/mackieonyx400f.htm (3 of 5)12/19/2005 10:19:54 AM

Mackie Onyx 400F

mixer at a gig — all 10 inputs could be mixed down to stereo and routed out to a stage DI for the front-of-house mix, and this mix could also provide a personal monitor feed adjusted by front-panel Control Room level pot. While it is possible to use the 400F as a four-channel stand-alone Onyx mic preamp, all four mics could optionally be mixed down to stereo for direct-tostereo recording, so there's another stand-alone application. In the project studio, the user could leave his or her guitar preamps, keyboards, and mics patched in and mixed to stereo, making it possible to play or rehearse without having to switch on the computer. The tabs across the top of the configuration software's main window select which output pair you're dealing with, and a simple level-pan mixer (with solos and mutes) appears below for setting up the mix for that particular output. It really is that easy — you just have to remember to switch off software monitoring in your DAW if you want to set up low-latency direct monitoring. And talking of easy, installation on a Mac was simply a matter up unzipping the installation archive and then selecting Onyx 400F in the driver setup section of the DAW software (Apple Logic Pro in my case). On Windows machines, ASIO, WDM, and GSIF are currently supported, but users are advised to check the Mackie web site for the latest driver and software versions. The first eight numbered inputs and outputs are the analogue connections, with the S/PDIF pair showing up as 9+10. A setup window enables the user to select the sample rate and clock source, and also lets you switch the headphones to mirror outputs 1+2 or 7+8.

Onyx Opinion Listening tests confirmed the Onyx preamps to be as clean and flattering as I remembered them, and undoubtedly the quality of the preamps is what will sell this unit against the competition. The LED metering reports overloads with a decent safety margin before digital clipping actually occurs, so to get a more exact idea of how much gain you need, it's best to consult the input metering in your DAW software. Arranging low-latency monitoring and custom mixes is extremely easy, but I can't help thinking that the ability to set up alternative monitor mixes might have been more useful if they could have been sent to the individual headphone outputs, rather than just to the main line outputs. The only other real limitation is that of having no ADAT expansion slot for adding more I/O, as there are several very attractive preamp racks around now that use the ADAT interfacing standard. However, Mac users can employ Tiger's built-in Aggregate Devices to combine a couple of 400Fs, providing 20-in, 20-out capability with eight mic preamps. Although Mackie are working on updating the driver to allow daisy-chaining in Windows, they can't yet give an indication of when this will be ready. I would have liked to be able to use the DSP mixer for combining all 10 DAW signals, but maybe this will come in a future upgrade. The MIDI ports worked with absolutely no fuss, and having insert points on the file:///F|/SoS/SoS%2001-2006/mackieonyx400f.htm (4 of 5)12/19/2005 10:19:54 AM

Mackie Onyx 400F

rear panel means you can hook up the unit to a patchbay if necessary, enabling hardware compressors, equalisers, and so on to be used while recording. I still can't see the need for 192kHz sampling unless you're using converters that cost more than your car, but at least it is supported for those people who think they need it. Offering only 24-bit recording isn't a problem, as modern DAW software invariably works best at 24-bit resolution and generally offers dithering down to 16 bits for audio CD burning. Where dithering isn't available, most DAWs will happily accept a 24-bit signal and truncate it to 16 bits.

Verdict I'm aware that this is turning out to be a rather short review, but that's because everything about the Onyx 400F is extremely straightforward, even the 'no brainer' software installation. I really like the Onyx preamps, which are probably worth the ticket price on their own, and the Firewire side of things works just as flawlessly as it did on the Onyx mixer. I think Mackie have got the configuration software about right, as it does most of what most people need to do with absolutely no fuss, and though you can't set up custom headphone mixes directly, you can set up a custom mix on outputs 7+8 and then mirror that. Even so, it might have been more useful to be able to directly route different custom mixes to the two headphone outlets. With the DSP mixer off, Onyx 400F is a true 10-in, 10-out interface, and though many people now seem to mix directly to stereo within their DAW software, you can if you wish switch off the DSP mixer, route 10 tracks from your DAW to the 10 physical outputs of the box, and then patch those into the mixer of your choice. I have no hesitation at all in recommending the Onyx 400F in terms of audio quality and ease of use. Published in SOS January 2006 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



Native Instruments B4 II

In this article:


Installation & Compatibility Modelled Drawbar Organ New Look, New Toys Published in SOS January 2006 Teething Troubles Better By Tube Print article : Close window Cabinet Office Reviews : Software Expert, Preset & Setup Views Conclusions

Native Instruments B4 II £150 pros Excellent tube amplifier and speaker-cabinet models. Sounds even more authentic than before. Comprehensive MIDI control with recallable controller maps.

Software [PC/Mac]

Since its launch in 2000, NI's B4 has been the software instrument of choice for those who want realistic tonewheel organ sounds from their computer. But NI are clearly convinced it can be better... Nick Magnus

cons Some questionable issues regarding CPU usage.

Since 1980, when Korg released their original CX3, and made what was summary arguably the first serious attempt to NI have raised the bar yet replicate the sound of a Hammond again. B4 II is capable of organ in a portable keyboard, there recreating a range of have been scores of ersatz tonewheel Hammond sounds which were organs, some good and others, well, quite elusive on the previous not so good. With the advent of version. This, and its customisable compatibility software instruments, 'virtual' with a wide range of MIDI tonewheel emulators began to appear, hardware controllers, make and it was NI's B4, first reviewed in B4 II a must-have for anyone The main, 'Manual' view in B4 II. SOS back in November 2000, that looking for a serious and went on to become something of an eminently affordable plug-in alternative to the real thing. industry standard. B4 hasn't been without its competitors — Emagic (now Apple) soon followed up with EVB3 (see SOS February 2003), and since then, we've information seen the likes of USB's Charlie, a sample-based VST instrument (reviewed in £149.99 including VAT. SOS September 2004). Despite these, though, NI's B4 has remained the Arbiter Music Technology +44 (0)20 8207 benchmark for Hammond plug-ins, gaining favour not only with studio-based musicians but with live performers, too. 7880. +44 (0)20 8953 4716. Click here to email www.arbitermt.co.uk www.nativeinstruments.de

There is, however, always room for improvement. Whereas Emagic/Apple's EVB3 offers four different 'Leslie' cabinets and a choice of single or dual rotary speakers, B4 has no choice of cabinet simulations and no ambience simulations, and one frequently voiced complaint is that the volume level rather frustratingly tails off from around middle 'C' downwards. What's more, B4's functionality has

file:///F|/SoS/SoS%2001-2006/nib4ii.htm (1 of 7)12/19/2005 10:20:01 AM


Test Spec 2.4GHz Pentium 4 PC with 1GB of RAM running Windows XP. Cakewalk Sonar 5.

not been updated since its first release, five years ago! However, rather than making changes to B4, NI have opted to create a new generation of the plug-in. Significant enhancements and additions abound in this version — a choice of cabinet emulations, dual rotary speakers, and a new modelled tube amp are just a few of the goodies on offer. So does it sound even more authentic than before?

NI B4 version reviewed: v2.0.0.007.

Installation & Compatibility B4 II will run as a stand-alone instrument or as a VST, Audio Units, RTAS or DXi plug-in. Minimum system requirements are Windows XP with a 700MHz Pentium or 1.3GHz Athlon XP processor and 256MB of RAM, or a 733MHz Mac G4 with OS 10.3 and 256MB of RAM. Once installed, you have 30 days of full functionality before the software has to be registered to continue working. You do this using NI's Registration Tool method; this generates a System ID based on your computer's hardware, which you email to Native Instruments. They in turn email an Authorisation Key back to you, which, when entered into the Registration Tool, activates B4 II permanently. In the event that you have no Internet connection available at all, registration can also be done by snail mail — but not, apparently, by telephone.

New Look, New Toys In contrast to B4's two screens, or views, B4 II presents five different views. Constantly visible at the top of all five views is a panel from which you can select Presets, store sounds and select any of the five views. The main view, 'Manual' (shown above) displays the entire instrument with both manuals, pedalboard, three groups of drawbars and the performance controls from a real Hammond. Changes which are immediately obvious are the addition of a Rotator Brake/Run toggle switch below the keyboard (B4's Rotator and Drive on/ off switches are gone) and a control box housing Reverb level and Drive (tube distortion) amount knobs. The Rotator Brake/Run switch allows the Rotator to be stopped whilst still allowing the sound to be coloured by the Rotator's crossover network. The Rotator can still be bypassed entirely, but via a different editing view, as will be seen shortly (see page 76). The controls for Percussion and Chorus/ Vibrato are present as before.


'Organ' view gives better access to the drawbars than Manual view, and provides quick access to the modelled 'Tube Amplifier' and speaker cabinet options.


The upper and lower manuals and pedalboard are accessible via their own MIDI channels as on the previous version, and keyboard splits can be set up, providing 'zoning' of the sounds of both manuals and the pedalboard from one MIDI channel if desired. Keysplits can be assigned directly from the Manual view by holding down the right mouse button (the Control key on a Mac) and clicking on an upper manual key, and then making your choice from the resultant dropdown menu. The manuals and pedalboard can also be independently transposed by ±1 octave by a similar click-and-right-mouse-button (or Control button on the Mac) manoeuvre. Keysplits and transposition settings are global to B4 II — in other words, they're not storable on a per-preset basis. The Organ view (above) is where many of B4 II's new features are to be found, and is divided into four sections. The lower section duplicates the drawbars and performance controls of the Manual view, but displays them larger, for which my aging eyes are truly thankful. The upper section is split into three panels: Organ/ Pedal bass, Tube Amplifier and Cabinets/Microphones. The Organ section provides control over Key Click amount, fully variable response to keyboard velocity and Leakage amount. Leakage is a characteristic of older Hammonds, where the crosstalk 'whine' between the tonewheel outputs can be heard, varying in timbre according to which drawbars are in use, and which notes are playing. When applied carefully, this can add a great deal of realism. The bass pedals now have a String/Organ option, with a Sustain (release time) amount. When in String mode, notes decay to silence like a string bass when pedals are released. When in Organ mode, increasing the Sustain parameter acts like a 'hold' feature — the notes are held at full level according to the Sustain time amount — which is very useful for keeping the bass notes flowing if you're not too nimble on the pedals. Rather cunningly, this hold/decay works monophonically (although the pedals still play polyphonically), in that new notes curtail the previous decaying/held note to avoid unpleasant bass-note 'smudging'.

Teething Troubles A couple of problems came up with B4 II during the review period. Under certain conditions, B4 II's CPU usage ran exceptionally high when using it alongside other plug-ins in Cakewalk's Sonar 5 (my sequencer of choice). When B4 II is the only instrument present, the CPU meter runs at around 12 percent. However, in one particular arrangement, which included two instances of Kontakt 2 and various effects plug-ins, B4 II gobbled up nearly 40 percent of the available CPU, causing it to max out. Replacing it with the original B4 reduced the CPU to normal levels. B4 II's track audio meter (in Sonar) also displayed irrational (but inaudible) overloads of +18dB — but curiously only when it wasn't playing! Finally, in the stand-alone version, I noticed that moving the Drive knob fully anticlockwise sometimes caused audio to cut out completely.



Better By Tube NI have completely redesigned the built-in tube amplifier emulation for B4 II, and claim that the new model has been modelled on the original Leslie tube amplifier. The character of the distortion, especially at high drive levels, is considerably more authentic than in B4, being far less fizzy and imparting a satisfying growl. This makes for a more solid, focused sound that works just as well with single notes as with chords. At lower drive levels, it doesn't entirely capture the soft, sexy 'purr' of the real thing to my ears, erring a little on the crackly side. Nevertheless, this can be greatly improved by the choice of virtual speaker cabinet (see the 'Cabinet Office' box below). The new Tube Amp also offers extremely effective tone controls that work at just the right frequencies, especially when emphasising the bass end of things — an area in which B4 was rather lacking. An additional benefit of the Tube Amp is the way in which it compresses the volume level as you drive it harder, much like a real tube amp. This, in conjunction with the tone controls and The Expert screen (see overleaf) is similar to various other niceties, goes a long way the Organ view, but provides this panel of more detailed controls for the organ towards addressing B4's problem of percussion, reverb, and Leslie rotors. the drop in volume in the lower key ranges. Now it's a doddle to recreate aggressive, spitting bass tones (think 'History Repeating' by the Propellerheads) that were frustratingly elusive on the original B4. Speaking of volume consistency, B4 II now features loudness robbing, another Hammond 'feature' whereby the sound's volume is 'compressed' whenever the same tonewheel is being played simultaneously by multiple keys. All this leads to a much better sound balance overall. As well as the various speaker cabinet simulations that can be chosen in Organ view, four of the virtual microphones' parameters are located here. Firstly, the relative levels of the dry tonewheels and Rotator output can be balanced. As hinted earlier, turning this fully anticlockwise is how you would completely bypass the Rotator (as opposed to hitting the Brake), and the option to balance the two is there for those wanting to recreate the effect of playing both through a Leslie and a Hammond's own stationary speaker. The treble and bass rotors' levels can be balanced, as well as the treble and bass microphones' relative pan positions. The Air parameter is the most interesting here, increasing the level of early reflections in the sound. This creates a convincing sense of distance from the speaker cabinet without adding any significantly measurable reverb.



Cabinet Office B4 II offers 13 different speaker cabinet simulations. The first four of these are variations on Leslie 122 and 147 cabinets, each one being either fully enclosed or with the 'back' removed to reveal the rotating speakers. With the exception of the final simulation, Direct (a model of a DI box), the remainder are modelled after classic guitar speaker cabinets. Although NI have avoided using brand names, the illustrative icons make the intended references clear. 'Citrus', for instance, is sure to be an Orange 4x12 guitar cabinet, 'AC Box' must be an AC30, and 'Jazz' looks very much like a Roland Jazz Chorus amp. The most interesting one here is, unusually, 'Bass VT' — undoubtedly Ampeg's famous bass guitar cabinet. This has a particularly beefy sound with a lot of high frequencies, and is excellent for emphasising the rotary effect's swirling of the upper drawbars. These cabinet simulations are the real key to modelling specific Hammond sounds that are associated with certain genres of music. Each cabinet confers signature frequency characteristics to the sound, in some cases dramatically so. For example, there is a clear distinction between the Open 122 Rotary model, which delivers a bright, aggressive rock tone, and the Closed 147 Rotary model, which produces a mellower, jazzy timbre but with a shiny-sounding top end. Such subtleties of tone could be quite difficult to convey with the original B4. For example, I have tried to duplicate the classic Al Cooper 'Like A Rolling Stone' sound — a full, flutey sound but with silvery, swirling upper harmonics. I never quite managed to nail it on B4, but the registration 800036030 through B4 II's Rotary 147 Closed cabinet with the 'Air' parameter at 50 percent and drawbar Leakage set to around 30 percent hit it dead on. Interestingly, using a guitar cabinet in conjunction with the rotary speaker balance at 100 percent is an interesting concept. For this to work in real life, you would have to send the rotary speakers' microphone outputs to a matched pair of amps feeding two identical, acoustically isolated speakers and mike these up all over again! However, the inclusion of guitar cabinets is not as daft as it sounds — Jon Lord's signature overdriven Hammond sound in Deep Purple evolved through such experimentation, namely driving Marshall stacks directly from the output of his C3 — often without the Leslie. When using the additional Vox Continental or Farfisa tonewheel sets, the guitar cabinets come into their own, as this is representative of how they might originally have been amplified. The combination of DI Box and rotary effect, on the other hand, is a rather more intriguing fantasy concept! The provision of these cabinet simulations in B4 II is well judged, and broadens the sound palette to a remarkable degree.

Expert, Preset & Setup Views The Expert view (shown on the previous page) delves deeper into the finer sound-editing details. Percussion volume, decay and harmonics can be finetuned here; similarly the Vibrato mix and depth can be adjusted. In the Rotary section, new additions are individual Spread (stereo width) controls for the treble and bass rotors, and the choice of single- or dual-rotary speakers. With Dual selected, the two Rotators spin in opposite directions and at slightly different speeds, for an even fuller, wider sound.



Two Reverb types have been added to B4 II, Studio and Spring. These are surprisingly flexible — in addition to what you'd expect, some very bizarre effects can be coaxed from them, as certain Presets demonstrate. The Reverb takes its signal from the Tube Amp, and this can then be routed before or after the cabinet/ rotator, or any mix in between. Some delightfully spooky Doctor Who-style effects can be obtained by routing the Reverb through the Rotator, then sweeping the Spring reverb's Size parameter while playing — a trick that's possible if you assign the Size parameter to a MIDI controller. Tonewheel sets can be changed too. You could optionally purchase extra sets for B4, including the sounds of Vox Continental and Farfisa Compact organs, an Indian harmonium, and a selection of variously aged Hammonds ranging from pristine condition to 'trashed beyond repair'. These extra sets are now included as part of B4 II. Preset view is the place to keep all The Setup screen. your sounds in some sort of order. All the usual Preset management options are here — sounds can be saved and loaded individually or in complete banks, and they can be moved to different locations in the Preset list by simply dragging and dropping. Deleting Presets is similarly simple, and the Preset list may be compacted to move the empty spaces to the end of the list. And very helpfully, the Preset manager has Undo/Redo buttons with up to 30 history levels, so if you regret deleting that cheesy 'Days Of Our Lives' organ 10 minutes ago, the chances are you will be able to restore it. Three handy Audition buttons are provided, each preloaded with short ditties in Jazz, Rock and Blues styles, which is great when you want to appraise Presets away from a keyboard. There's also a built-in MIDI file player, which could be useful for importing your own preprepared audition pieces. Setup view (above) manages the various global settings for B4 II, including Transpose, Keysplits and MIDI channels for the manuals and pedalboard. Various controller options are also provided to ensure maximum compatibility with external MIDI hardware devices. External audio input can be enabled/ disabled (in stand-alone mode) when using B4 II as an effects unit — this is automatically enabled when B4 II is inserted as an effects plug-in within a host DAW. The DXi version of the effects plug-in is now fully automatable in Sonar (this was not the case with the original B4, and is warmly welcomed by this Sonar user!). One particularly powerful feature is the comprehensive MIDI controller assignment map. To make a MIDI controller assignment, you simply highlight the desired parameter name in the list, click the Learn button and move a fader or knob on your MIDI hardware. The list then updates to reflect your choice. Once you have made your assignments, they can be saved as a Controller Map file. NI have thoughtfully provided a number of pre-made hardware Controller Maps, including templates for their own B4D hardware controller, plus various Novation,



Korg, Kurzweil, Nord and Behringer devices among others. Helpfully, text files accompany these maps (in the B4 II Controllers folder) with information on the assignments for each map. There is even a template for the Korg OASYS — NI are clearly confident that OASYS owners will prefer B4 II to the tonewheel organ engine built into their own illustrious keyboard!

Conclusions I'm hugely impressed by NI's attention to detail for this version of B4. The new features are well chosen and tastefully implemented, in particular the new tube amp and speaker cabinet models, which make possible the recreation of stunningly realistic Hammond sounds of virtually any style or era. I think the Rotator is one of the best Leslie simulations out there, and the implementation of full, customisable MIDI control makes B4 II compatible with just about any piece of MIDI hardware you wish to use. If you thought the original B4 hit the mark, be prepared to experience deep joy all over again. Published in SOS January 2006 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



RPCX Blue

In this article:

RPCX Blue

Overview VST 2.0 & AU Synth Plug-in [Mac/PC] Algorithms & Synthesis Published in SOS January 2006 Types Oscillators & Filters Print article : Close window Envelopes & Multi Envelopes Reviews : Software LFOs PD/WS Section Saving Presets & Banks Step Sequencers & Mod Matrix Sound designer Rob Papen may be known to you as Sequencer & Effects the name behind amazing sounds for Access's Virus, Easy Edit & Global Windows or Alesis's Andromeda. Now he's moved into In Use & Conclusions

producing software-based synths of his own...

RPCX Blue £133 pros

Nick Magnus

Combines four kinds of synthesis in one package. Clear, uncluttered graphical interface. Amazing amount of modulation possibilities. Excellent presets that you'll actually want to use!

The labelling on the box states 'Virtual Rob Papen Synthesizer', suggesting maybe a virtual Rob Papen that would make the tea, or perhaps do a spot of ironing. Many SOS readers, however, will recognise this name as belonging not to the latest thing in home help, but cons to the well-known Dutch sound Very CPU-intensive. designer. Blue is the latest soft-synth Algorithms are not userproduct from Rob's Netherlands-based configurable. company RPCX, and is designed in The process of saving partnership with programmer Jon Presets could be made simpler. Ayres. They describe Blue as a 'crossfusion virtual synth' delivering FM, Phase Distortion, Waveshaping and summary Blue is an excellent-sounding, subtractive synthesis types. highly individual software synth that is full of surprises, and a joy to use. Despite its heavy CPU overhead, it should prove to be an inspirational tool for intrepid sound designers as well as casual users who simply want a large library of inspirational sounds to fuel their musical endeavours.

Blue requires a VST 2.0 or AU host, and there's no stand-alone version. Installation is straightforward: you just type in your copy's serial number when you first run the plug-in. The review copy supplied was originally at Version 1.0, but the designers supplied an update to v1.1 before I completed my review, which fixed a few bugs and gave me access to some new Preset Bank categories. Apparently, further updates and user banks are planned, so it's worth keeping an eye on www.robpapen.com after you've registered.

information £132.95 including VAT.

file:///F|/SoS/SoS%2001-2006/rpcxblue.htm (1 of 10)12/19/2005 10:20:09 AM

RPCX Blue

Time + Space +44 (0) 1837 55200. +44 (0)1837 840080. Click here to email www.timespace.com www.robpapen.com

Test Spec 2.4GHz Pentium 4 PC with 1GB of RAM running Windows XP. RPCX Blue version reviewed: v1.0 & 1.1.

Overview Blue's promised versatility becomes immediately apparent when auditioning the supplied Presets, which are arranged in Banks of 32 Presets, and include categories with intuitive names like 'Pads', 'Analogue and Digital basses', and 'Sequence Sounds', as well as some less readily comprehensible ones, like 'Diverse'. These feature not only plenty of luscious virtual-analogue and classic FM-synthesis stylings, but also a generous helping of sounds and textures that bring to mind Korg Wavestations, Prophet VS Vector synthesis, PPG wavetables, Casio PD synths and many more. At the heart of Blue are six oscillators which can be routed to interact with one another using 32 possible configurations, or Algorithms — already we can see similarities to six-operator FM synthesis (see the box above). The oscillators can be selectively processed via two analogue-style stereo filters, 13 envelopes, 10 LFOs, a comprehensive mod matrix, two stereo effects processors, three assignable step sequencers and a 32-step 'note' sequencer. This is potentially a complex synth whose features need to be clearly presented to ensure ease of use, so Blue's designers have opted for a simple, clean user interface. As you can see from the previous page, Blue occupies a single window, divided into two halves. The upper half, which is always visible, houses the controls for the six oscillators and the two filters. The lower half offers one of 12 different editing windows in a very much simpler, line-drawn graphic style.

Algorithms & Synthesis Types The different synthesis types in Blue are governed by the way in which the six oscillators are connected to one another. When designing a sound from scratch, the first port of call is likely to be the Algorithm Edit window. Algorithms can be best understood by direct comparison with the well-established Yamaha DX7-style six-operator model, in Algorithm routing is displayed in this easy-towhich operators can either add their understand screen. output directly to the signal, or be routed to modulate other operators, creating harmonically complex waveforms. Blue's oscillators can be structured in exactly the same way to create classic FM sounds — although unlike on the DX7, where operators generated only sine waves, a Blue oscillator can generate all manner of waveforms. To use something analogous to 'traditional' analogue subtractive synthesis, we should therefore choose an algorithm that has one or more oscillators whose signals are added directly to the output. The screengrab below shows two algorithms, and the upper one, algorithm 1, is ideal for this — all six oscillators pass directly to the output, enabling Blue to behave like a monster six-oscillator virtual-analogue synth. The lower algorithm, number 8, shows how two synthesis


RPCX Blue

methods can be combined. The direct signal from oscillators A and B form the virtual-analogue sound element, whilst the output of oscillator F is modulated by C, D and E in series — the FM sound element. Choosing a specific Algorithm is just the starting point, however — there's a huge range of subsequent waveshaping, filtering and modulation possibilities, as we'll see in the rest of this review.

Oscillators & Filters There are six oscillators, labelled A to F. Oscillators A and B have the greater number of controls, and are identical apart from A's lack of a Mode button. Oscillators C, D, E and F are all identical in layout — the difference between these and oscillators A and B is their lack of pulse-width modulation (PWM) and Symmetry controls. This suggests predetermined uses for certain oscillators, and indeed A and B are well suited to typical analogue/subtractive synthesis. Each oscillator has an On/Off button and a waveform selector. Clicking on the waveform button presents drop-down menus revealing 15 analogue-type, 39 additive-type and 32 spectral waveform options. Each oscillators' phase can be inverted, and keyboard tracking can be enabled or disabled — the latter option is often employed to create certain FM and ring-modulated sounds. The Ratio button selects an oscillator's base frequency — when you click here, a drop-down menu offers 64 choices. The default is 1.00, the equivalent of the 8' setting on an analogue oscillator. This can be further adjusted using the Semitone and Fine-tuning knobs, so you can obtain any precise pitch. The Shape and Feed knobs are the key to obtaining a virtually infinite number of variations on the preset waveforms via Waveshaping and Phase Distortion synthesis, the Shape button selecting which of these synthesis types an oscillator uses (more on this later). The Vel knob allows each oscillator an independent velocity response which supplements a sound's overall response to velocity. This means that even if a sound's overall volume is not velocity responsive, the individual oscillators can be made to be, which is an important consideration when modulating one oscillator with another in an FM context. Each oscillator's volume level is set, unsurprisingly, with the Volume knob. The Dest (destination) button selects where the output of an oscillator is routed, but its behaviour (or non-behaviour) will depend on the chosen algorithm. If the oscillator is output directly (as explained in the box on algorithms on the previous page) you have seven routing options — dry, filter A or B, both filters, built-in effects units A or B, or both effects. However, if an oscillator is being used as a modulator, the Dest button displays that oscillator's routing, and you cannot change it, as its output has already been determined by the algorithm. The final controls for Oscillators A and B are the PWM and Sym (symmetry) knobs. Symmetry is directly comparable with the pulse-width control of an analogue oscillator, except that it affects all selectable waveforms except Pink and White noise. The PWM (amount) knob becomes active once a mod source is defined, such as an LFO or envelope.


RPCX Blue

The oscillator Mode button's function is also dependent on the chosen algorithm. If the oscillator is output directly, it simply toggles between Normal and Sync To A settings — The PD/WS section, where you can draw hence the absence of a Mode button your own waveforms and then sculpt them on oscillator A, as there's no point in further. the oscillator sync'ing to itself! However, when an oscillator is acting as a modulator, the Mode options become FM, Ring (modulation) and Sync To A. As if all this choice wasn't enough, right-clicking on any knob allows you to associate it with a MIDI controller; you just select 'latch to MIDI' and move the appropriate wheel or fader on your controller. Two independent filters are available, each offering 13 types: 6dB-per-octave lowpass and high-pass affairs, 12dB- and 24dB-per-octave low-pass, band-pass, high-pass, and notch types, and Ring, Comb and Vox filters. These can be set to operate in series or parallel, and as described above, each oscillator can be routed to either or both of the filters. The controls usually associated with analogue synths are all here: Q, envelope amount, key velocity and key tracking, with additional niceties such as mod-wheel sensitivity, distortion (pre-filter saturation), filter output volume and pan position. Each filter can be routed to the built-in effects units A, B, A and B (more on these later), or simply output as dry. Two of the filter controls depart from the expected norm when using the Ring, Comb or Vox types. The Q control appears to function as a gain control for the Vox filter, whilst Distortion (according to the manual) adjusts the vowel sound. The latter effect can be quite subtle, though, and I found that the interplay between filter cutoff position and Distortion was crucial towards fine-tuning vowels, as indeed was the choice of waveform being treated. The Ring and Comb filters both require the Q control to be set somewhere other than zero, and whilst a static filter cutoff frequency produces audible results with the Ring type, the Comb type is at its most demonstrative when the cutoff is being modulated by an envelope, LFO or other source. The Distortion control, incidentally, becomes inactive when using the Comb filter. In short, there's a vast wealth of tone-generating possibilities available from just the oscillators and filters — but these can also be manipulated using the various parameter-editing windows...


RPCX Blue

Envelopes & Multi Envelopes The Envelope window accesses nine different envelopes — a volume envelope for each of the six oscillators, two filter envelopes and a global volume envelope. All are of the AHDSR type, which should be familiar to many. Blue's AHDSRs, however, add several natty enhancements. A Pre-delay time adds a temposync'ed delay before the attack phase, ranging from 'off' to 1/32nd triplets (fast) to 16 measures of 4/4 (er... slow). Curiously, this appears to run just ahead of the beat. With a one bar pre-delay, the attack phase begins just before the following measure; the designers may want to investigate this. The envelopes' An AHDSR envelope in Blue (top), and speed can respond positively or (above) one of the more flexible userdefinable multi-stage envelopes. negatively to key position and velocity, the individual modulation amount for each envelope can be set, and when sync'ed to tempo, the envelopes can be retriggered to useful musical time values. Best of all, the curve of the attack, decay and release slopes is fully variable from extremely exponential to reverse exponential. In addition, the sustain level can either 'plateau' at the set level, or it can subsequently fade to zero or back to full level using the 'Fade Level' amount. This parameter's description (and its display) are slightly misleading — the value's percentage amount actually refers to the time it takes to fade up or down, not the level to which it fades! Four Multi Envelopes, named Free A, B, C and D, are assignable to synth parameters using the mod matrix. They earn the 'Multi' title because of their customisable, multiple envelope points. Up to 16 points can be set, including the start and end. You just click on the envelope to add points, move them to where you like, and adjust the curves. Multi Envelopes can be sync'ed to tempo, and the time scale over which they run can be set to musical time values. Envelopes can either be one-shot or looped — you simply designate which are the start and end points. For example, in an eight-point envelope, if the start is at the third point and the end is at the seventh point, then you will have an 'attack' phase of two points. The envelope will then loop around points three to seven while a note is held, and if 'Release Stage On' is selected (assuming your sound has a volume release envelope) the envelope will play the final stage when the key is released. I have one request here — when Tempo Sync is on, it would be helpful if the envelope points could optionally snap to musical time divisions.

LFOs Of the 10 LFOs, six have dedicated functions. Oscillators A and B each have their own LFO for PWM duties, there is one each for filters A and B, one for global vibrato and one for global tremolo. The remaining four (Free A, B, C and D) are for use in the mod matrix to assign as you wish. Each has six waveforms,


RPCX Blue

with various controls to adjust Attack and Decay time, Key position, Speed, Symmetry and a parameter called Humanise (random Speed and Depth variations). There is also a smoothing parameter, intended to 'soften' the edges of the square and S&H waveforms, but this seems merely to reduce the mod depth. The LFOs can be sync'ed to the host tempo, and have three modes: Poly (separate LFOs for every note) Mono (one LFO for all notes whose phase resets with each new note) and Free (one LFO for all notes with free-running phase). The LFO Attack/Decay time parameters only work when the mode is set to Poly, which I suspect is a bug.

PD/WS Section This gives you access to a whole galaxy of waveforms, and is one of Blue's most interesting aspects. You can hand-draw waveforms in this set of six windows (one for each oscillator), and when the relevant oscillator's Shape parameter is changed, the harmonic content of the original waveform is dramatically altered, as you can see in the screenshot above. In its default mode (a diagonal line) no change will occur when the Shape knob is moved, but when you draw a wiggly line in this box, and move the Shape knob, new waveforms pour forth. Whenever you redraw the line (or change the Shape amount) the two smaller boxes on the right are updated to show what the resulting wave will be, according to whether the oscillator is in PD (Phase Distortion) or WS (Waveshaping) mode. The manual doesn't attempt to explain the differences between these, and it's not clear how closely related Blue's version of PD is to the method used by Casio in their 1980s CZ-series synths. Nevertheless, the two modes can produce very different results, although just how different depends on the original waveform and what you draw in the window. When you vary the Shape amount in WS mode, it sounds much like adding or subtracting harmonics on an additive synth. Doing the same in PD mode is closer to the effect of oscillator sync: a more convoluted, dramatic sound. If you modulate the Shape amount using an envelope, LFO, MIDI controller or any other of Blue's mod sources, things get pretty lively. And all of this comes from just one oscillator — don't forget you've got five more to play with!


RPCX Blue

Saving Presets & Banks Saving an edited Preset in Blue can be confusing. A Preset is a fundamental part of the Bank it comes from, so if you have edited a Preset and wish to overwrite the original version, you must either re-save (overwrite) the entire Bank, or save it as a new Bank. If you select a new Bank before saving the current one, your edited Preset will be lost! If you edit a Preset, and then decide that you want to return to the original, this is possible using the Compare function. But if you edit a Preset and then select a new Preset, your previously edited Preset cannot be returned to its original state, nor can it be reloaded on an individual basis — you must reload the entire Bank. The simplest way to avoid this confusion is to save your edited patch as a separate FXP file (Banks are saved as FXB files) and then reload it into any other convenient Preset location. This way, you get to keep the original, but of course if you want your new, reloaded Preset to become a permanent part of the Bank you've loaded it into, you still have to save the Bank again!

Step Sequencers & Mod Matrix The three Step Sequencers are, like the Multi Envelopes, intended as modulation sources to be accessed via the mod matrix. Up to 16 steps can be specified, the levels of which are set by drawing in the window with the mouse (see the screenshot below). The transition between steps can either be abrupt, or be made to ramp smoothly from step to step with the Smooth parameter. The main differences between the Step Sequencers and the Multi Envelopes are that the time divisions between steps are fixed (Multi Envelope points are freely moveable), the Sequencers are always sync'ed to host tempo (if Tempo Sync is selected in Blue's Global page), and looping is permanently active. If you want to use a Step Sequencer to modulate oscillator pitch, the value readout just below the edit window is very helpful, as it displays the bars' levels as percentages and semitones, thus enabling simple repeating melodies to be set up. The step sequencer is an excellent tool for creating sounds reminiscent of the Korg Wavestation's wavesequences or PPG wavetables — by drawing waveforms in the PD/WS window, and modulating the oscillators' Shape amount using the Step Sequencer, very similar effects can be achieved. If you want to go that extra mile, why not set up all three Step Sequencers, running on different timescales? One could modulate an oscillator's Shape amount, the second could modulate filter cutoff frequency, whilst the third could modulate the oscillator frequency ratio of an FM pair, or even the pulse width of a 'virtual analogue' oscillator. Devilishly complex rhythmic textures can be made this way! Although Step sequences cannot be saved individually to the hard drive, you can copy and paste between Presets. Up to 20 routings can be made in the Mod Matrix window, presented as two pages of 10 routings per page. Routings must be entered from the top downwards — if you make an assignment in slot 2 with no assignment in slot 1,


RPCX Blue

nothing will happen! Similarly, if you leave an unassigned slot between assigned slots, the ones below the unassigned slot will not function. 35 modulation sources are available: four Multi Envelopes, four Free LFOs, three Step Sequencers, the Sequencer 'Free' line (see above right), as well as 23 pre-determined MIDI control sources. These can be routed to 91 destinations, so things can get very busy indeed! Once you've set up the source and destinations, you can choose a negative or positive modulation amount. This value is displayed as a percentage or in semitones, depending on the destination.

Sequencer & Effects This sequencer is modelled after the monophonic sequencers often associated with modular analogue synths. Sequences of up to 32 steps can be generated, either sync'ed to the host tempo or freely running. The parameters are presented on two pages, each containing three 'control lines' (note pitch, volume, slide on/ off, filter A amount, filter B amount) and a 'Free' line which can be used as a control source in the modulation matrix. Creating a sequence is quite straightforward — you simply click in the Step box on the top line to activate a step (whereupon it turns white) or click again to deactivate it (it turns blue). These blue spaces in a sequence are treated as rests — unlike on some analogue sequencers, you cannot 'tie' notes across the gap. If you're thinking you can compensate for this by setting a long release time to cover the gap, think again — I tried, and the CPU meter shot through the roof! Once you have a sequence that you like, you can save it by right-clicking in the sequencer window and archiving it to your hard drive. Previously stored sequences can, of course, be reloaded into any Preset.

One of the Step Sequencer's displays.

Every good synth has onboard effects, and Blue has two such modules, A and B (see above right), which may be routed in series or in parallel. When set to series, effects unit A feeds into B. Since each oscillator is independently routable to either or both effects processors (depending on the algorithm used) this allows for a reasonable amount of flexibility, despite the simplicity of the effects layout. Each built-in processor offers the same effects: mono delay, stereo delay, chorus, flanger, phaser, distortion, lo-fi, a stereo widener, reverb and a comb filter. They all sound great (well, apart from the lo-fi effect!) and have an unusually wide range of fine-tuning parameters. The distortion notably delivers an extremely wide range of colours, spitting venom by the bucket-load. Fans of Roland's Dimension D will love the stereo widener, and the Comb effect is very interesting, offering two modulatable frequency bands, each with individual feedback amount. I was about to lament that effects parameters cannot be set as destinations within the modulation matrix when I discovered that they can all be controlled via MIDI. Try this with the Comb effect by setting up controller curves in your host sequencer to sweep the frequency bands — you get truly bizarre


RPCX Blue

results!

Easy Edit & Global Windows As its name suggests, the Easy Edit window allows for quick, one-fader editing of 18 useful parameters. Each fader applies a global offset to all occurrences of a particular parameter — so Global Filter Freq, for example, adjusts the cutoff frequency of both filters up or down in parallel. Similarly, Global FM offsets the total FM amount across all six oscillators, which is ideal for when you have to fix something quickly. The Global page sets up the general behaviour of a Preset as a whole. Here we can set tuning, voice assignment, portamento and bend range, for example. Amongst the parameters There are a few more effects on offer than in provided here, Oscillator Precision is a standard soft synth, including reverbs and worthy of special mention. This allows comb filters. you to set an amount of 'tuning drift' for each oscillator, to emulate the behaviour of an analogue oscillator drifting over time. At 100 percent, the tuning is 'digitally perfect', whilst 0 percent gives the greatest amount of drift. Maximum and minimum drift speed controls govern the range of random drift speed variation — you can set both to low for the subtlest effect, or set both to high to simulate a very unwell synth! A Filter Smoothing fader reduces the zipper noise that would otherwise occur when moving the Cutoff frequency, and if you're plagued by aliasing noises at high frequencies, an optional anti-aliasing filter can be engaged. Further tonal control is supplied by a two-band, sweepable EQ which, like all of Blue's variable parameters, can also be controlled via MIDI. And speaking of MIDI-controllable parameters, there is a very neat extra feature hidden in the corner of this page, enigmatically named 'Esc'. This allows you to save and load any Preset's MIDI control assignments, which can be invaluable if you need to make the same assignments time after time when creating Presets.

In Use & Conclusions As you'd expect from a synth bearing the Rob Papen name, the Preset sounds are of the highest order. There are seemingly endless ways in which to animate sounds using the modulation matrix to control the oscillators, envelopes, LFOs, and the step sequencer, as well as some very tasteful effects, all of which makes Blue a great tool for creating interesting textures. If you were to use it only for pad sounds, there would be plenty here to inspire you for a long time to come.


RPCX Blue

All of this good stuff makes Blue's high CPU usage all the more frustrating. Playing four-note chords with particularly complex sounds sometimes took up more than half the available CPU headroom on my 2.4GHz Pentium 4 PC, which is not what I'd think of as underpowered. Running multiple instances of Blue on this system, especially with other plug-in instruments present, is not really an option unless you stick to simple sounds and restrict yourself to monophonic parts. However, that's not to say that you couldn't run one instance at a time, and 'freeze' the results to audio as you go. This isn't exactly convenient, but it is at least a viable proposition. To be fair to the designers, they point out that all unused elements (such as redundant oscillators and filters) should be switched off to conserve CPU power, and they even suggest using separate, low-CPU plug-in effects instead of Blue's own. That aside, for people who really enjoy sound designing, Blue should provide an excellent voyage of discovery. Editing is surprisingly uncomplicated thanks to the clutter-free interface, and FM sounds in particular are very easy to program in Blue — even easier than in NI's FM7, in my opinion. And for those people who simply want to load up some inspiring, ready-to-go sounds with an alternative edge, it also delivers the goods, standing apart from the usual rash of 'me-too' virtual synths. Published in SOS January 2006 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



Sample Libraries: On Test

In this article:


Funk University **** Hot New Releases Koncept & Funktion **** Published in SOS January 2006 Chopped Guitars **** Downtempo Guitars Volume 2 Print article : Close window **** Reviews : Sound/Song Library Talkbox Guitar *****

Star Double First Names ***** Debbie Harry **** James Joyce *** Vera Lynn ** Hugh Grant * Julie Andrews

Funk University **** MULTI-FORMAT This library from Big Fish Audio is all about drum loops. There are no other instruments, or even single drum hits. The 800 loops are served up in three formats and are organised into tempo-based folders across a range of 55150bpm. Within a particular folder, the loops are subdivided further based upon the drum kit used, each kit having a distinctive sound. Aside from the tempo and kit number, there is no further guidance regarding each loop's content, for example to indicate which are fills. The various kits provide plenty of sonic variety. While the snare sounds used are all typical of funk — mostly quite tight, dry, and restrained — there are also kits where the snare has more of a 'crack' or 'ping' to it — suitable for a tight rock sound as well as funk. There is a mixture of both 'dry' loops and those where the kit has been recorded within a more lively space. Amongst the lower-tempo loops the playing is not too busy, but definitely has a funk feel. As the tempos get up to the 70bpm range, things start to liven up a little, and there are some quite aggressive-sounding loops with a bigger snare sound. However, there is generally a nice mix of fairly straight loops, busier loops (with plenty of examples of syncopated snare or hi-hat work), and loops containing fills. At tempos above 120bpm, the action gets a little busier, while right at the top end of the tempo range there are one or two loops that would obviously make your limbs ache. On the whole, though, the playing is very tasteful and restrained. Most of the loops are in four/four time, but a smattering of six/eight loops appear, as do a small number that feature brushes rather than sticks. Obviously these loops would suit funk-based music, but I could also imagine some of them working in soul, R&B, or even more commercial hip-hop —

file:///F|/SoS/SoS%2001-2006/sampleshop.htm (1 of 6)12/19/2005 10:20:15 AM


perhaps anything from Jamiroquai through to the Black Eyed Peas, with a few stops in between. The more aggressive and ambient loops would also work in modern rock (the tight snares, for example, would not be out of place in some nu metal) or pop styles. The library has plenty of content and, given the multi-format package, it certainly offers value for money. However, it might have been nice to see some single hits included, particularly some cymbal crashes to add as accents. I'm not sure there is anything radically new on offer here in terms of musical content, but if you are looking for a 'funk drums 101' loop library, Funk University is a pretty good starting point. John Walden Apple Loops, REX, and WAV DVD-ROM, £59 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.bigfishaudio.com

Koncept & Funktion **** MULTI-FORMAT Reason users requiring a large, accessible drum & bass loop collection should start here. Over 1GB of samples is squished into this Refill, and the eponymous duo (aka David and Nic Higham) know their stuff. This material is fast, so breathe deeply and get ready for a world where the resting pulse is 172-175bpm. Overwhelmingly loop-heavy, this collection (also available in Intakt Instrument format) welds woofer-flapping bass, on-the-edge beats, and 'distorted in a good way' leads to soundscapes and textures both mellow and industrial: a perfect showcase of the two sides of drum & bass. Central to the collection are 40 construction kits, each featuring related beats, pads, textures, and grooves. For a start, there are REX-format loops which can be loaded into Dr:Rex for playback at any tempo — multiple Dr:Rex instances are needed to assemble a track. In addition, a kit's raw samples are collected into both NN19 and NNXT sampler patches. This is bizarre, as NN19 patches can easily load into NNXT, and NNXT's facilities are hardly being exploited. You won't be able to alter the tempo of these patches, but you do have the handy advantage of being able to mix and match loops from your controller keyboard — a nice option in a live environment. Curiously, many sampler patches contain more samples than there are REX loops in a given kit. The Refill is more than rounded out by plentiful non-construction-kit drum and groove loops, pad loops, and effects. The base tempos remain dizzying, though file:///F|/SoS/SoS%2001-2006/sampleshop.htm (2 of 6)12/19/2005 10:20:15 AM


a handful of 100bpm and half-tempo-feel examples throw you a curve ball. This material is again split between REX loops and NN19/NNXT patches. There is no duplicate material: even the NNXT groove loops group appears to have nothing in common with the Dr:Rex folder of the same name. That's great for variety, but less so for tempo-matching, and although many samples aren't tempo specific, plenty are. I also find it odd that entire NN19 and NNXT patches contain just one loop or sample each. Given that no NNXT facilities have been exploited, there doesn't seem to have been much point in using this CPU-heavy device in the first place. There's some very nice work in this Refill, but sadly there's no indication of where the loops come from or how they were created. This is a sample-focused collection — and genre — but it would have been interesting, and quite enlightening, to hear some beats as Redrum patterns and some sounds as Reason device patches. That said, the Refill includes ten hard and heavy Redrum kits. One last thing: amidst all the industrial textures and odd effects are some folders of vocals — Female, MC, Spoken, and FX. Perhaps I'm missing something, but this portion of the set borders of the satirical, despite being nicely recorded. That and some organisational oddities aside, I found much to like, and there's a lot of material here that will be grist for any adventurous musical mill. Derek Johnson

Reason Refill or Intakt Instrument (including VST, DXi, Audio Units, RTAS, and standalone versions), £59.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.zero-g.co.uk

Chopped Guitars **** Downtempo Guitars Volume 2 **** MULTI-FORMAT These two sample collections are in many ways like two different moods of the same entity. Chopped Guitars uses rhythmic gating combined with radical processing to produce effects that wouldn't be out of place in an adrenalinefuelled video game based on antisocial driving habits. Downtempo Guitars, on the other hand, while still dripping with processing, has a somewhat more laidback feel. Both libraries include more than 400 loops, all pre-sliced so that they can be played back over a wide tempo range in REX-compatible applications and Spectrasonics Stylus RMX without sounding any more unnatural than they're intended to! Chopped Guitars includes some atmospheric and melodic examples, but a lot of it is taken up with rhythmic material processed in a way reminiscent of libraries file:///F|/SoS/SoS%2001-2006/sampleshop.htm (3 of 6)12/19/2005 10:20:15 AM


such as Spectrasonics Distorted Reality. Electric guitars are subjected to distortion and heavy filtering of all kinds, combined with reverb, rhythmic echoes, rotary speakers, and so on. If you think of the sequenced synth line at the start of The Who's 'Won't Get fooled Again', that will give you some idea of the general character and vibe of these grooves, and indeed many of the sounds are so heavily processed that they're barely recognisable as guitars at all. In most instances, the sequences are based on a musical chord, scale, or short phrase, and there's normally only a choice of one or two chords, so your composition may need to be built to follow what's available rather than vice versa. Nonguitarists who would like to inject a more organic feel into their music should find this collection inspirational, and even guitar players will find sounds here that are difficult if not impossible to replicate themselves. Even though the sounds are extensively processed, they still retain a 'real' character that synthesized sounds often lack. Downtempo Guitars Volume 2 is more relaxed, but seldom gets as far as being dreamy or ambient. The processing is just as creative as with the other library, but is less obviously rhythmically based and tends to be optimised for lower tempos. Again the limited choice of chords and scales could pose a creative problem, but then I'd image many of these loops will find use as intros or breaks rather than main musical beds. The general sound quality of these loops is extremely good and many of them are great for kick-starting ideas. Both libraries are well worth considering, especially if you have Stylus RMX or a REX-compatible application. Paul White

Acidised WAV, Reason Refill, REX, and Stylus RMX DVD-ROMs, £39.95 each including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.ninevoltaudio.com

Talkbox Guitar ***** GIGASTUDIO Whether it be the hallucinatory tones of 'Sparky's Magic Piano', the dreamy vocoded incantations of 'Mister Blue Skies', Jimi Hendrix's conversational wah-



wah on 'Still Raining, Still Dreaming', an android-esque filter sweep from a Minimoog, or the Absynth's vowel-like mutations, people have always been fascinated by hearing an instrument 'talk'. One '70s device, the Talkbox, allowed players to vocalise their performances in a unique way: by running a guitar signal into a small speaker sealed in a metal box and thence through a length of plastic tubing into the player's mouth, allowing the lips and vocal cavities to shape and modulate the tone. This simple but effective gizmo has been used by Peter Frampton, Jeff Beck, Dave Gilmour, Joe Walsh, Joe Perry (Aerosmith), Richie Sambora (Bon Jovi) and Dave Grohl (Foo Fighters). There are no records of a keyboardist using one, presumably because as a breed we're too fastidious to handle any item that might once have been in a guitar player's mouth. Like a wah-wah, the Talkbox effect works best when in motion, with one vowel eliding into another as in speech. To replicate this with samples requires real-life performances, and Sonic Implants have done us proud by supplying a large number of phrases in 'E', 'G', 'A', and 'D', in a choice of three tempos (92, 104, and 120bpm). Comprising low-pitched, gutsy rhythmic riffs, wailing lead fills, widdly bits, minor third trills, and a nice smattering of soulful pitch bends, these funky, bluesy licks sound fine, though I noticed the tuning of one or two low notes was a little sharp. There's also a handy section of sustained and choked power chords. If you feel inspired to create your own licks (as I did), try the Talkbox multisamples. These are great fun and cover a wide sonic range: closed, open, 'wah', and 'boh' sustains, and short 'boo', 'bop', and 'wow' articulations. The 'wow' deliveries are tight and superbly coordinated, evoking the comedy mute sound of '30s brass. With a different vowel sound on every note, the vibrato samples are equally engaging. Generally speaking, the Talkbox's tone is thin and cutting, lying somewhere between a muted trumpet and a wah-wah clavinet, but the metallic edge is softened by its trademark elastic, throaty quality and tonal shapeshifting. This library is highly enjoyable to play, good quality, ferociously funky, and cheap as hell. The guitarist on the sleeve may look like a young Rolf Harris sans beard, but he has an excellent technique and a great rhythmic feel on a par with Rolf's wobble-board virtuosity. So I'm giving Talkbox Guitar five stars for its sheer entertainment value. It will funk up your track without screwing up your bank balance, and, though it contains only 266MB of samples, it covers a lot of musical ground and provides some unusual and distinctive guitar timbres. Documentation is provided on a PDF file — for more extensive talkbox-related info, visit the excellent site www.blamepro.com/talkbox.htm. Dave Stewart



Gigastudio 2 CD-ROM £38.45 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.sonicimplants.com Published in SOS January 2006 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



Scan 3XS

In this article:

Looking Good Specifications Of Review System Drive Faster Service & Support Audio Performance

Scan 3XS Dual-core Athlon 64 PC For Music Published in SOS January 2006 Print article : Close window

Reviews : Computer

Scan 3XS £1762 pros Incredible amount of processing power! Huge 500GB RAID 3 array that provides both high performance and data security. Excellent value for money considering the specification.

Dual-core CPUs promise a huge jump in performance at a modest price, while RAID disk arrays can provide both faster and more secure storage. Scan Computers' Athlon-based system features both technologies. Martin Walker

cons Slightly noisier than some specialist music PCs.

There's been a lot of talk about dual-core PCs over the last few months (including my own PC summary Musician article in SOS July 2005). By placing two This Scan 3XS dual-core processor cores into a single piece of silicon, Athlon 64 PC is significantly manufacturers can provide significantly faster more powerful in every performance than a single processor, even when respect than any other PC I've reviewed to date, yet its price under-clocking them and running them at lower voltages so they don't run hotter than the singleremains extremely competitive. Only the acoustic core variety. You can now buy dual-core models noise level lets it down slightly from both AMD (the Athlon 64 X2 range) and Intel for the musician who wants to (in the Pentium D range), although AMD-based record in the same room, and systems tend to be somewhat cheaper, especially this could be lowered fairly as many existing Athlon 64 motherboards can have easily with a few tweaks. their BIOS updated to run the dual-core versions. information Basic system as reviewed without monitor, keyboard/mouse, music hardware or software £1762 including VAT. Full system as reviewed including monitor, keyboard/ mouse £1980. Scan Computers +44 (0) 870 755 4747. +44 (0)870 755 4747. http://web6.scan.co.uk/ Support/Query.ASP?

Photos: Mike Cameron

First off the blocks with a dual-core AMD Athlon music PC are a company new to SOS, although the Scan name will be familiar to anyone who knows the mainstream PC world. Scan Computers International, to give them their full name, were founded way back in 1989, and now employ 150 staff at their Bolton main office. Their 3XS brand name stands for Specification, Service and Satisfaction, and they provide PCs optimised for specific applications such as home, office, CAD, audio, video, gaming and servers. When I visited their web site, there were 10 audio systems on offer covering starter studios, tracking and post-production, and catering for songwriters, guitarists and DJs. As with most specialist music retailers, each system can be

file:///F|/SoS/SoS%2001-2006/scanpc.htm (1 of 8)12/19/2005 10:20:21 AM

Scan 3XS

QueryType=S http://3xs.scan.co.uk www.scan.co.uk

further configured from its default specification by the customer if you want a different case, CPU, amount of RAM, hard drive, and so on. However, Scan take the options further than most, since for an additional £300 you can also specify one of eight custom paint schemes, although their list of Pro Audio Recording interfaces is currently restricted to models from M-Audio.

Looking Good Housed in an attractive black Chenbro SR10569 Hi-End Workstation case, the review 3XS System sported a smart red latching front panel, although unlike some, it was not thick enough to reduce noise from the front-mounted drives when closed. A cutout about halfway up exposed the two front-mounted USB 2.0 ports, which are handy for dongles or flash drives, and the door concealed a Sony DW-Q28A DVD writer and 3.5-inch floppy drive in the upper two 5.25-inch drive bays, and a large air intake grille in front of the four hard drives (more on these shortly). Inside, this Scan PC was one of the tidiest I've seen. Round IDE and floppy cables had been fitted, and all the cables had been firmly attached to the chassis using ties, with any excess length carefully looped or folded to keep it well out of the way of the cooling fans for maximum airflow. The vast majority of the motherboards launched specifically for dual-core processors use either the nVidia nForce4 or Intel 945/955 chip sets, which cater for expansion via the new PCI Express format, but as previously mentioned, dual-core Athlons can be slotted into most existing Socket 939 motherboards after a BIOS update. Scan have, perhaps wisely, decided to stick with just such an all-PCI-based motherboard, the Abit AV8 VIA K8T800 Pro, which as its name suggests uses Via's K8T800 Pro chip set and supports a single AMD Athlon 64 Socket 939 processor. This avoids possible PCI Express audio latency issues, as well as keeping the price down. This motherboard has five PCI slots and one AGP 4x/8x slot, and supports up to 4GB of dual-channel DDR400 RAM, Gigabit Ethernet, plus two ATA133 IDE and two Serial ATA150 connectors. There are also six USB 2.0 ports and one IEEE 1394 Firewire port, and the passive (rather than fan-cooled) chip set heatsink is a plus point.

One of the PCI slots in the review machine is taken up by a dummy backplate which has a speed control for the CPU fan.

The most important component is the CPU, in this case an AMD Athlon 64 X2 dual-core 4800+ model with both cores running at 2.4GHz. A fast dual-core CPU like this takes some cooling, so I was pleased to see that an exotic Akasa Evo 120 cooler had been fitted, complete with copper heat pipes to draw the heat away, plus a huge aluminium heatsink and associated 120mm twin ball-bearing fan noise-rated at just 15dBA for its slowest speed. A fan-speed control on a file:///F|/SoS/SoS%2001-2006/scanpc.htm (2 of 8)12/19/2005 10:20:21 AM

Scan 3XS

dummy PCI backplate is also provided to adjust the optimum noise/heat setting, although if you slow it down too much you'll hear frantic beeping from the BIOS fan-speed alarm. From the CPU fan, the cooled air is encouraged to leave the case via another Akasa 120mm rear-mounted case fan, while a third 120mm temperaturecontrolled fan is mounted in the 430 Watt Tagan PSU. It's a shame that the two Akasa fans aren't temperature-controlled, since this could reduce the overall noise level when idling, but larger fans are quieter anyway. Scan had fitted two 1GB sticks of the well-respected Corsair low-latency PC3200 DDR2 RAM, and an NVidia GeForce 6200 graphics card, again with passive cooling. This particular review system was completed with one of M-Audio's Delta 66 soundcards, a 17-inch Neovo TFT monitor with a hardened glass panel mounted on top of the LCD panel to enhance image quality, plus a Logitech silver/black cordless keyboard with rechargeable cordless mouse.

Specifications Of Review System Case: black Chenbro SR10569 HI-End Workstation case. PSU: Tagan TG430-U15 430W PSU fitted with 'Silence Control Technology' and 120mm low-noise fan. Motherboard: Abit AV8 VIA K8T800 Pro with one Socket 939 for AMD Athlon 64 processor, Via K8T800 Pro chip set running 2x1020MHz front side buss and 2x204 MHz memory buss, four 184-pin DDR DIMM sockets supporting up to 4GB of PC3200 SDRAM memory, five PCI slots, and one AGP 8x/4x slot. Processor: AMD Athlon 64 X2 dual-core 4800+ with dual 2.4GHz clock speed, 1MB cache. CPU heatsink and fan: Akasa Evo 120 incorporating heatpipe, heatsink, and 120mm fan with 15dBA noise rating. System RAM: two 1GB sticks of Corsair low-latency DDR2 PC3200. System drive: Samsung HD160JJ, 160GB, 7200rpm, SATA II, 8MB buffer. Audio drives: three Samsung SP2504C, 250GB, 7200rpm, SATA II, connected to a threeport XFX Revo 64 RAID card operating in RAID 3 mode. Graphics card: NVidia GeForce 6200, passive cooling and 256MB RAM. Optical drive: Sony DVD RW DW-Q28A burner, 16x DVD and 48x CD read speeds, 48x CD, 16x DVD+/-R, 4x (DVD+R DL) write speeds, 24x CD-RW, 8x DVD+RW, 6x DVD-RW rewrite speeds, ATAPI Ultra DMA 4 interface, 2MB buffer. Active system ports: PS/2 mouse and keyboard, six USB 2.0 ports, one Firewire port, serial port, parallel port, LAN port. Keyboard and mouse: Logitech silver/black cordless rechargeable desktop system. Installed operating system: Windows XP Professional Edition with Service Pack 2. Also supplied with review system: AG Neovo E-17 monitor, black, with 17-inch diagonal, 1280 x 1024 native resolution; M-Audio Delta 66 interface; Steinberg Cubase SX 3.0.


Scan 3XS

Drive Faster As mentioned previously, a total of four hard drives were fitted in the review system, and their arrangement deserves special mention. All four were Samsung SATA II models from their Spinpoint P series which are claimed to be quieter than the competition, as well as running significantly cooler, which helps keeps overall noise levels down. The first drive was a HD160JJ model of 160GB capacity connected directly to a SATA port on the motherboard for system drive duties, while the other three drives were all larger SP2504C models with 250GB capacity, again with 7200rpm spin speed and 8MB buffers. However, this time they were connected to a three-port XFX Revo 64 RAID card operating in RAID 3 mode, creating a 480GB array. Musicians investigating RAID invariably desire RAID 0, which splits reads and writes across two drives, thus doubling the sustained transfer rate and therefore the potential number of audio tracks or soft-sampler voices. The problem is that if one of the drives ever goes faulty you may not be able to access the data stored on either of them. Conversely, RAID 1 uses two drives in 'mirror' mode to provide greater security, but with no increase in drive speed. RAID 0+1 does provide the benefits of both, but requires four drives.

With plug-in and soft-synth performance that eclipses every previous PC I've reviewed, I predict that this Scan dual-core Athlon 64 is destined to become highly desirable to musicians, especially as it doesn't come with a huge price tag.

RAID 3, as used here, provides both additional speed and security. It employs one drive for parity, storing extra data that can replace any that is lost on the other two, so even if one drive fails altogether the other two should be able to carry on regardless until it's replaced. Most RAID systems require careful setting up in the BIOS as well as driver support from Windows, but this Revo 64 card is clever in that as far as Windows is concerned it looks like a standard IDE controller, with all the RAID aspects implemented in hardware. Apparently setup is also incredibly easy, requiring a single key-press to configure the three drives ready for formatting by Windows. This card makes using RAID far easier for the musician who requires a vast number of audio tracks or streamed instruments, the only down side being that you have the noise of four hard drives to contend with. However, with such exotic cooling fans I wasn't expecting this Scan 3XS machine to be particularly noisy, and it isn't. It's not the quietest PC I've reviewed to date (that honour still goes to the Prescott 3.2GHz system from Phil Rees reviewed in SOS January 2005, closely followed by Inta Audio's Opteron 146 system reviewed in SOS November 2004), but it's as quiet as various PCs I've received from other specialist music retailers, despite including four hard drives file:///F|/SoS/SoS%2001-2006/scanpc.htm (4 of 8)12/19/2005 10:20:21 AM

Scan 3XS

and not having any acoustic material lining the case. A quick look in the BIOS confirmed that the Sony DVD burner had been connected as IDE Primary Master, the Onchip Audio Controller had sensibly been disabled, and as far as I could see all other ports had been left at their default settings, with the serial, parallel, USB and Firewire ports all active. I downloaded and installed Abit's uGuru utility software for Windows XP, and was then able to monitor temperatures, fan speeds and voltages while performing various stress tests. The BIOS had been carefully set up to sound continuous warning beeps on various fault conditions, including the temperature of the CPU reaching 75 degrees Centigrade or the CPU fan speed The DVD-RW and floppy dropping below 1200rpm, so I set the rear-panel drives are located behind the fan speed control to the minimum value that orange door on the front panel. avoided the BIOS warning, when it was scarcely audible next to the case and PSU fans. After an hour or two the system CPU temperature was idling at just 31 degrees Centigrade, and when I ran my usual 'torture tests' to establish the highest temperature of the CPU under extreme load I couldn't get this to rise above a very safe 55 degrees Centigrade. I'm impressed with these results considering the huge amount of processing power 'under the bonnet', but there are several other things that could further reduce noise levels. Providing a speed control for the case fan as well as the CPU one would help, while having both fans continuously temperature-controlled would optimise noise levels without the user needing to make manual adjustments (this was the main strength of the Phil Rees system). Those particularly interested in low noise might also like to investigate the various system options on the Scan web site — the Silverstone case with the more solid aluminium front panel might well reduce hard drive noise, as could some acoustic lining panels (Scan offer the Akasa Sound Proofing Kit as an optional extra).


Scan 3XS

Service & Support All 3XS systems have a one-year on-site parts and labour warranty for UK mainland customers, which for many musicians will be more useful than the more typical RTB (return to base) warranty that forces you to repackage the system and be without it for at least several days. However, Scan do provide an additional two years' RTB service, with the first year covering the cost of return, parts and labour, and the second covering return and labour. There's also an Extended Systems Warranty available for a further £115 that provides an additional two years of insurance-backed cover that you can take out at any time, offering on-site repairs from Monday to Friday, 8.30am to 5.30pm, and which covers any internal, electrical, electronic or mechanical failure of any any desktop PC system valued up to £2000. Tech support also seems very good, with a dedicated telephone support line where a support engineer will guide you through various technical solutions, and if a hardware fault is suspected, an on-site engineer can be sent out to examine your system and hopefully resolve the problem. If this doesn't prove possible, the system will then be returned to base for further examination. Once the fault has been rectified and the system fully checked it will then be returned to the customer.

Audio Performance The Dskbench utility measured 51.7MB/second and 51.1MB/second for the sustained write and read transfer rates on the 'C' system drive, along with a projected 144 tracks of 16-bit/44.1kHz audio with a low 2 percent CPU overhead, and I confirmed these figures with HD Tach. However, I was far more interested in the results for the RAID 3 audio drive 'D', which produced excellent figures of 91MB/second and 84MB/second for sustained write and read transfer rates, and a projected 320 tracks of 44.1kHz/16-bit audio (equivalent to 98 tracks at 24bit/96kHz). This doesn't quite reach the staggering 124 and 95 MB/second that I measured on Carillon's twin Seagate Barracuda RAID 0 array in SOS September 2004, but this time your data is a lot safer, as even if one of the three drives on this system fails altogether you can still carry on regardless. I did notice that the CPU overhead when Dskbench read its round-robin blocks from multiple tracks was around 10 percent, but while HD Tach produced a similar result for average read speed of 92.8MB/second, this time the CPU overhead was only around 4 percent, so I don't think this is anything to worry about. Taking a closer look at the Windows settings relevant to audio performance, I was reassured that the most important, processor scheduling, had been correctly set to 'Background services', and although System Restore was still active, along with all the visual effects, I very much doubt that you'd notice any performance difference on such a powerful machine.


Scan 3XS

The various Sisoftware Sandra CPU benchmark tests turned in results that were almost identical to the reference Athlon 64 X2 4800+ model, proving that the review system was performing exactly as expected, and the memory bandwidth was an excellent 5800MB/second. However, for most musicians, the other important test is just how powerful this dual-core processor is when running plug-ins and soft synths. I decided to stick with Timo's Cubase SX Performance Test version 2, since although a version 3 is now available, version 2 is what I've run on all the other SOS review systems, so you get a direct comparison of how each one compares. The Akasa Evo 120 CPU

Initially I had a strange problem where the ASIO cooling system involves Multimedia drivers provided great performance at copper pipes, a heatsink and a large fan. very high latency of several hundred milliseconds, but as soon as I chose the ASIO drivers the CPU overhead rose drastically, even at the highest 2048-sample buffer setting, and kept on increasing as I lowered the buffer size so that I couldn't run the test song at all down at 3ms. Re-installing the latest M-Audio drivers resolved this anomaly, and I began to get the results I was expecting. When running at 44.1kHz and 23ms latency and above, the test song required just 11 percent CPU when stopped (thus measuring only the processors and effects), rising to 20 percent in play mode (with the soft synths also active). Dropping the latency to 3ms increased these two values only slightly to 13 and 24 percent respectively. This is significantly faster than any other PC I've reviewed, even beating by a considerable margin the dual Xeon 3.06GHz PC from Red Submarine that I reviewed back in SOS 2004. We've all been waiting to see how much improvement dual-core systems would offer when running music applications, but even I found the truth startling. Back in July 2005 I said that 'Performance benefits for the musician are uncertain at the moment, but it seems that a dual-core Athlon X2 could provide more than 60 percent more welly than a similarly clocked Athlon 64.' In fact, this Scan system provides almost exactly double the raw CPU performance of the Dawsons Athlon 64 3700+ system reviewed in SOS March 2005, and by extrapolation you would need a single-core 7700+ processor to match it — if one existed! Overall, this Scan system turns in a stunning performance, but not at the expense of a luxury price: £1762 is extremely competitive considering that the 4800+ CPU retails at over £600, especially as this includes a three-drive 500GB RAID array that should achieve 100 audio tracks at 24-bit/96kHz, 2GB of RAM, a DVD burner, and some exotic cooling options. Those who don't intend to run loads of audio tracks at 192kHz sample rates or streamed video alongside their music could abandon the XFX Revo 64 RAID card and two of the hard drives, knocking about £250 off the total price and bringing it closer to the £1500 mark


Scan 3XS

— I think the other specialist music retailers have some serious competition on their hands. Overall, this Scan system turns in a stunning performance for a very reasonable price. I think we can safely assume that dual-core systems are going to be extremely popular with the PC musician, and that Scan themselves should expect plenty of customers for their 3XS audio PC systems. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Tannoy Reveal 8D

In this article:

Reflex Loading Digital Interfacing Round The Back Critical Auditioning Reveal Revelations

Tannoy Reveal 8D Active Midfield Monitors Published in SOS January 2006 Print article : Close window

Reviews : Monitors

Tannoy Reveal 8D £699 pros Extremely good sound quality for the price. Tight bass and extended highs. Versatile input connections. Unusually flexible roommatching equalisation. Free spectral analysis and alignment software.

Updating their successful Reveal range, Tannoy have included new high-resolution tweeter technology, sophisticated room-correction EQ, and digital interfacing. Hugh Robjohns

cons Disappointing stereo depth information.

The original Tannoy Reveal Actives were reviewed back in SOS July 1999, and have had a large and summary loyal following ever since amongst the home-studio brigade. The latest models in the Reveal series are This latest incarnation of the Reveal includes useful available in three forms: the 6D, which has a siximprovements all round, while inch woofer; the 8D (under review here), which has retaining the cost an eight-inch woofer; and the 66D, a horizontal effectiveness. There are both centre-speaker design with a pair of six-inch analogue and digital inputs, woofers. and very impressive roommatching EQ facilities. Distortion and resolution have both been improved, as has power handling and overall bandwidth. The new Reveal is destined to remain a favourite of home studio owners.

information Reveal 8D, £699 per pair; Reveal 6D, £499 per pair; Reveal 66D, £299 each; Activ-Assist microphone and cables kit, £49. Prices include VAT. Tannoy +44 (0)1236 420199. +44 (0)1236 428230. Click here to email www.tannoy.com

Reflex Loading Photos: Mike Cameron The Reveal 8D is quite a substantial midfield monitor measuring 425 x 262 x 364mm (hwd) and weighing 16kg. The reflex-ported cabinet is constructed mainly from 18mm MDF, although the front baffle is 40mm thick, partly to enable the characteristic corner chamfering which combats cabinet edge-diffraction effects. Internal bracing helps control panel resonances, and a single flared port vents to the rear — so you shouldn't place this speaker too close to a rear wall.

The two drivers are both new to the updated Reveal series, and both are recessed into the front baffle to minimise diffraction from their fittings. The bass driver, despite initially appearing to use a spun-metal cone, is actually a fairly traditional multi-fibre doped-paper design. However, it does feature a very

file:///F|/SoS/SoS%2001-2006/tannoyreveal8d.htm (1 of 5)12/19/2005 10:20:25 AM

Tannoy Reveal 8D

Test Spec 2GHz Pentium 4 PC with 1GB RAM, running Windows XP (SP2).

complex four-layer voice-coil assembly to improve the driver's sensitivity. Harmonic distortion has been improved too, through the use of eddy-current damping at the extremes of cone excursion. This is achieved by aluminium fluxcontrol rings around the voice coil and copper caps on the pole pieces. The titanium-dome tweeter is protected behind a strong grille, and is one of Tannoy's favoured 'wideband' units which boasts a response up to 50kHz. The quoted system frequency response is an impressive 47Hz-51kHz at the -3dB points (measured anechoically), and shows gentle peaks of around 2dB centred on 8kHz and 15kHz, a 1dB scoop-out between about 500Hz and 2000Hz, and a rising bass level below 300Hz peaking at +3dB at 100Hz. Below this, the response falls steeply (typical of a reflex design), although there is also an odd secondary resonance in the response at 50Hz. The smaller versions of this Reveal speaker show similar characteristics. The crossover is set at 2.5kHz, and the built-in power-amp pack provides 120W for the bass driver, with 60W for the tweeter, providing an impressive maximum SPL figure of 117dB (for a driven pair of monitors at one metre).

Digital Interfacing The 'D' in the Reveal 8D model name indicates the speaker's acceptance of a digital input, in this case a 24-bit coaxial S/PDIF signal via an RCA phono socket. A parallel feed-through socket is provided to route the stereo signal on to a second speaker. Sample rates between 44.1kHz and 96kHz are supported, and a yellow LED illuminates when a suitable digital signal has been recognised and the D-A synchronised. Another miniature slide switch selects the required audio channel from the stereo D-A converter: options are left, right, and mono sum. I have some reservations about monitors equipped with digital inputs. While many people will doubtless find the feature attractive, the technical disadvantage is reduced D-A resolution. Because in practice you rarely, if ever, listen to the monitor running flat out, some digital attenuation will have to be applied to the signal feeding the digital input, so it won't be using anything like the full 24-bit resolution theoretically available. Furthermore, if that digital attenuation isn't processed correctly with dither, you could end up with nasty truncation distortions. Not nice! It's easier to control monitoring levels in the analogue domain, but the 8D is equipped only with a rear-panel trim control and it only has a range of +6dB to -12dB, so some sort of outboard monitor level control will be necessary.

Round The Back Although the large rear-panel heat sink only became warm during my tests, Tannoy supply a plastic heat shield to protect the user from coming into contact with uncomfortably hot surfaces when adjusting any of the controls. Mains voltage is fixed according to the sales region and, unusually, the IEC socket is an earth-free two-pin type, because the circuitry is double-insulated. The benefit of this is that there is no chance of developing ground loops through the mains safety ground when using this speaker, even if using unbalanced connections. A


Tannoy Reveal 8D

proper mains power switch is provided adjacent to the mains inlet, but the speakers can be switched to Standby mode from the front baffle by pressing the Tannoy badge. An LED on the rear panel shows when the unit is powered, and one on the front baffle illuminates green when the speaker is fully on, and red when it's in Standby mode. The amplifier chassis accepts both analogue and digital inputs, although only one source can be active at a time, selected by a miniature slide switch. The balanced analogue input is catered for with a combi jack/XLR socket, and full output is achievable with a 0dBu input signal. The input impedance is surprisingly low at a nominal 600(omega), although this is only likely to be problematic with passive monitor controllers. For use in audio-visual or satellite-plus-subwoofer installations, the Reveal 8D has a switch to introduce a 12dB/octave high-pass filter at 80Hz. Like most active monitors, the 8D features frequencycontouring facilities to match the speaker's response Alongside the analogue and to the room, but these are unusually extensive on the digital connections on each new Reveal — two 10-way DIP switches on the rear speaker's rear panel are 20 DIP switches which set up panel offer no less than 2250 different EQ settings! the unusually flexible roomThankfully, Tannoy offer Activ-Assist software as a correction equaliser. free download from their web site, and this can be used on a PC or Mac in conjunction with an omnidirectional mic and soundcard. (You can use your own, or purchase a kit of suitable mic and interface leads from Tannoy.) The software generates a test signal for the speaker, the mic captures the in-room response, and the software recommends the most appropriate DIP-switch settings — all very straightforward and user-friendly, and it does help. However, it is no replacement for decent acoustic treatment to control standing waves and reflections.

Critical Auditioning With the onboard equalisation set 'flat', I powered the speakers up and played some familiar tracks. Like most speakers in my room, the Reveal 8Ds sounded fractionally bright to my ears, but that was quickly tamed using the -1dB highfrequency contour switch. My initial impression was that the bottom end was very well controlled, if a little dry and with limited extension, and the mid-range was very clean and neutral sounding, with a lovely open top. Stereo imaging was excellent and stable, but didn't seem to convey depth information as well as some other monitors in this price range — mixes I know to have excellent depth came across rather flat and one-dimensional. Balancing familiar multitrack material was surprisingly easy, there being plenty of file:///F|/SoS/SoS%2001-2006/tannoyreveal8d.htm (3 of 5)12/19/2005 10:20:25 AM

Tannoy Reveal 8D

resolution throughout the mid-range and top end. The lack of low bass distortion made judging the relative levels of bass and kick drum fairly straightforward too, and I didn't develop any sense of fatigue after extended listening — always a very encouraging sign! Indeed, it became clear quite early on that the 8Ds were surprisingly accurate and revealing monitors, far more so than their UK price would suggest. An optional measurement mic and interface cable bundle allows you to use Tannoy's Activ-Assist software to match the response of the Reveal 8Ds to your monitoring room.

Heavy-handed mastering was revealed very clearly as such, while wellrecorded and dynamic material was portrayed gloriously, with detail, precision, and involvement. I'd have liked a little more low-end extension, but you can't have everything from a relatively small ported box, and I was A/B-ing them with PMC TB2 monitors which have a prolific but natural-sounding bottom end. The Reveal 8Ds are blessed with both power and efficiency, allowing them to fill any reasonably sized room with ease and with only mild power compression at full bore. However, I found that they also needed a certain minimum volume to really gel together. At low levels the balance changed significantly — as is the case with most reflex speakers — with low bass all but disappearing. However, this was countered to a degree by the very good mid-range and low mid-range resolution, which allowed the upper harmonics of bass instruments to be heard clearly, giving useful clues as to the true balance, even if the low-bass energy wasn't really there.

Reveal Revelations Overall, I was very impressed with the Tannoy Reveal 8Ds. These are monitors which have been finely honed over the years, retaining the best elements of the earlier incarnations while benefiting from new technological advances which improve upon their weaknesses. I was a little disappointed with the limited ability to convey stereo depth cues, but overjoyed with the level of resolution and neutrality given the price. The Reveal 8D offers a good blend of properties: high sensitivity and power handling, low distortion and fine detail resolution through the mid-range, an open and airy top end, and a nicely controlled bass. Add to that an adequate D-A converter to allow flexible interfacing, and unusually comprehensive onboard EQ to aid room-matching, and the Reveal 8D is a very attractive and cost-effective monitor for the home studio.


Tannoy Reveal 8D

Published in SOS January 2006 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



Tascam FW1804

In this article:

Rack Attack Bundle Of Fun In Use Recommended System Requirements Conclusions

Tascam FW1804 Firewire Audio & MIDI Interface [PC/Mac] Published in SOS January 2006 Print article : Close window


Tascam FW1804 £479 pros Very good audio quality. Good driver performance. Easy to use.

cons

If you fancy Tascam's FW series of Firewire interfaces, but don't need their control-surface features, the new FW1804 might be just what you're after.

Only basic input level monitoring on front panel.

summary Tascam's latest offering is very easy to use, the software drivers seem solid and it provides high-quality audio performance. It might not be the coolest-looking piece of hardware in your rack, but if the feature set fits your particular combination of I/O needs, the FW1804 is worthy of serious consideration.

information £479 including VAT. Tascam +44 (0)1923 438880. +44 (0)1923 236290. Click here to email www.tascam.co.uk www.teac.co.jp

Test Spec PC with 3.2GHz Pentium 4 CPU, 2GB RAM, Echo Mia 24, Egosys Wami Rack 24 and Yamaha SW1000XG soundcards, Adaptec FireConnect 8300 PCI Firewire card, running Windows XP Pro SP2.

John Walden

Firewire seems to be the interface protocol of choice for large numbers of musicians on Mac OS and Windows, and the pages of SOS have featured a growing number of such devices over the last two of years. Tascam have featured a couple of times in this regard. The FW1884, a well specified unit combining a control surface, Photos: Mike Cameron comprehensive analogue and digital audio interfacing and four sets of MIDI I/O, was favourably reviewed by Paul White back in the November 2003 issue. More recently, the FW1082 was reviewed in May 2005. This unit, which also combines a control surface with audio and MIDI interfacing, could be seen as the FW1884's more affordable baby brother, with four rather than eight mic preamps, two analogue outputs to the 1884's eight, and just two MIDI In and Out ports. Tascam's latest offering is the FW1804, and if it is possible for a baby brother to have a baby brother, then this is it. With a list price under £500 (and being advertised at considerably less than that by some of the major retailers), it features a very similar input and output specification to the FW1082 but without the control surface; instead, it's housed in a 2U rack format. The relationship between the FW1082 and FW1804 is, therefore, not dissimilar to that between Digidesign's 002 and 002 Rack.

Tested with Steinberg file:///F|/SoS/SoS%2001-2006/tascamfw1804.htm (1 of 5)12/19/2005 10:20:30 AM

Tascam FW1804

Rack Attack

Cubase SX 3.1.0 and Wavelab 5.00a, Sony Acid Pro 5.0c and Sound Forge 7.0b.

The front panel of the FW1804 is almost minimalist in comparison with some other audio and MIDI interfaces. The left-front is dominated by eight Trim controls for setting the input levels for the eight analogue inputs. Underneath each Trim control are two LEDs labelled Signal and OL, indicating the presence of a signal at each input and whether the input is being overloaded — useful, but in terms of front-panel level monitoring, that's your lot. A global switch for the phantom power for inputs 1 to 4 is located between and beneath the Trim controls for inputs 2 and 3. Beneath Trim 8 is a quarter-inch unbalanced jack which can be toggled between Line and Guitar levels: given that all the other inputs are on the back panel, this is useful if you just need to hook up a guitar, synth or microphone quickly. Aside from the power switch, the righthand side of the front panel contains various LED indicators to show Firewire, Clock, Digital In, ADAT and MIDI activity, all of which are useful for seeing quickly that signals are going in and out of the FW1804 correctly. A headphone output with its own level control is also present. The Monitor control sets the level of the signal going to the stereo out, while the three large buttons labelled Computer, Inputs and Both dictate which signal is routed to the stereo outputs — just the audio from the host computer, just the audio received at the inputs or both.

The FW1804 Settings page of the Control Panel provides settings for latency, how a footswitch should respond and the level at which the front-panel OL indicators will light, amongst others.

The rear panel is somewhat busier. Inputs 1 to 4 have balanced XLR-TRS combo jacks. Very usefully, these inputs also feature separate Insert jacks so that external processors such as compressors can be patched into the signal chain if required. Inputs 5 to 8 are provided on balanced quarter-inch TRS jacks, as are the Stereo Out jacks. Two physical MIDI In and four MIDI Out connectors are provided and a quarter-inch jack can be used to attach a footswitch, the function of which can be assigned through software. The rest of the rear panel is all about digital, with word clock, ADAT optical and coaxial S/PDIF connections. There are two Firewire ports, although the manual advises against daisy-chaining the FW1804 with other Firewire devices. Aside from a printed Setup Guide and Owner's Manual — both very concise — and the necessary power adaptor and two-metre Firewire cable, the only other contents of the box are three CDs. The first of these is the ubiquitous driver installation disc, while the other two contain bundled versions of Cubase LE (for both Mac and PC) and Gigastudio LE (PC only). The latter two are described more fully in the 'Bundle Of Fun' box, while the former provided a very painless installation of the unit when hooked up to my test PC. file:///F|/SoS/SoS%2001-2006/tascamfw1804.htm (2 of 5)12/19/2005 10:20:30 AM

Tascam FW1804

Bundle Of Fun Tascam supply a very useful bundle of software with the FW1804, particularly for PC users. For both Mac and PC, Steinberg's Cubase LE (v.1.07 in the review package) provides up to 48 audio tracks with 96kHz support for those who want it, 64 MIDI tracks, VST, VSTi and Rewire support and a good subset of the editing facilities found in SL or SX. There is also an upgrade path to the full version of Cubase SX. Tascam's own Gigastudio 3 LE is also included for PC users. Some regard the full version of Gigastudio as the best software sampling instrument currently available and it certainly has some of the best sample libraries. The LE version offers 64voice polyphony, 16-channel multitimbral MIDI, Rewire and VST plug-in support. It is compatible with all Gigastudio sample libraries.

In Use Given that much of the I/O hardware on offer is identical to that found in the FW1082, I expected both the audio and driver performance of the FW1804 to match the positive findings of Paul Sellars in his May 2005 review of that unit. Some basic listening tests using a range of commercial recordings in various styles — including orchestral, solo classical guitar, pop, hip-hop and rock — produced consistently good results. Playback was always clear, with no obvious noise, and the sound both detailed and focused. In testing with Acid Pro 5, Sound Forge 7, Wavelab 5 and Cubase SX v3.1, the audio drivers performed flawlessly. The front-panel Computer, Input and Both switches enable input monitoring to be done directly or via your DAW. Easy access to direct monitoring will be appreciated by many, although latency is not really an issue given the performance of the drivers. Even with a fairly busy mix, I was able to get down to 4ms latency value in SX without any noticeable glitches in performance.

The 1804's first four input channels feature phantom-powered mic preamps and analogue insert points.

Direct monitoring of the inputs is easy to set up, although input metering is restricted to Signal and OL LEDs.

Recording tests in SX also suggested that the quality of the mic preamps is very good, while the line inputs and the switchable line/guitar input located on the front panel all provided very clean recordings. As long as due care is paid to other elements of the signal path, the FW1804 is capable of very good results and, in the context of a home or project studio environment, I'd have no qualms in using it for recording material aimed at commercial or broadcast applications.

file:///F|/SoS/SoS%2001-2006/tascamfw1804.htm (3 of 5)12/19/2005 10:20:30 AM

Tascam FW1804

Although I didn't get to test the ADAT connectivity, I had no problems with the MIDI I/O or the S/PDIF I/O — both worked exactly as would be expected. For potential purchasers weighing up the FW1804 alongside the various other Firewire audio and MIDI interfaces available, two features are worth emphasising. First, the front-panel monitoring of the audio input levels provided by the Signal and OL LEDs for each input can, at best, be described as 'basic' — it does the job without fuss, but if you need more precise information on input levels, this will have to be achieved through your software. Second, unless you can make use of the digital connectivity, the FW1804 is only stereo-out. Obviously, if you envisage needing multiple analogue outs to feed a surround monitoring system or integrate external effects units, then this particular Tascam unit is not going to be for you.

Recommended System Requirements PC: Windows XP or 2000; six-pin Firewire port. Mac: G4 or G3 with OS 9.2 or above or OS 10.2.8 or above, Firewire port.

Conclusions Tascam's FW1804 is solidly built, very easy to use, offers reliable drivers and provides audio performance that will be more than a match for most of the other equipment in the average home and project studio, all in a neat 2U rack unit. With the addition of the software supplied, it also provides a complete hardware/ software recording package — although PC users get a slightly better deal here given the inclusion of Gigastudio LE to go with Cubase LE. The market for Firewire audio and MIDI interfaces is getting crowded, with products at a range of different prices. Digidesign, RME, Echo and Yamaha all have excellent units at or above the price of the FW1804, while for those on a tighter budget, Edirol, M Audio and Focusrite (with their new Saffire) all provide stiff competition. While these units may vary in their absolute audio quality, they also offer different audio and MIDI input/output options, so it's a question of whether the combination adopted by Tascam in the FW1804 fits your needs. The unit itself performs to a high standard and is very straightforward to use. Published in SOS January 2006


Tascam FW1804




Yamaha MG8/2FX

In this article:

Channel Facilities Internal Digital Effects Processor In Use

Yamaha MG8/2FX £149

Yamaha MG8/2FX Analogue Mixer Published in SOS January 2006 Print article : Close window

Reviews : Mixer

pros Inexpensive. Provides all the basics for live mixing on a small scale. Decent effects with some adjustment.

cons Uses knobs instead of faders. No PFL buttons or pre-fade sends.

summary If you simply want to mix some line and mic sources, add a bit of decent reverb, and have basic EQ on the various channels, the MG8/2FX is ideal. It's only real shortcoming is that the lack of pre-fade sends means you can't set up a separate monitor mix.

information £149 including VAT. Yamaha-Kemble Brochure Line +44 (0)1908 369269. +44 (0)1908 368872. www.yamahamusic.co.uk www.yamaha.co.jp/ english

Straightforward facilities, built-in effects processing, and an affordable price make this new mixer a good choice for entry-level studio and live setups. Paul White

Mixers come in all kinds of sizes and with all levels of complexity, but sometimes you need something really simple that just gets the job done. Yamaha's MG8/2FX seems to have been designed with simplicity in mind and has absolutely no unnecessary frills, though it does have a very capable effects processor on board that can rustle up some suitable live reverbs and other basic effects. Powered from an external PSU with a screw-locking connector, the MG8/2FX Photos: Mike Cameron has eight main inputs plus a stereo return and a two-track input, so it is conceivable that you could mix up to 12 sources.

Channel Facilities The first two channels are mono mic/line, while the second two (3+4 and 5+6) can double as mono mic or stereo line inputs. Two further line-only inputs are available for channels 7+8, which offer a choice of both jack and phono inputs, and all five channel strips include a three-band equaliser with a mid-band control set at 2.5kHz. There's no EQ bypass, but the controls have a centre detent. The high and low controls operate at 10kHz and 100Hz respectively, and all three

file:///F|/SoS/SoS%2001-2006/yamahamg8.htm (1 of 4)12/19/2005 10:20:35 AM

Yamaha MG8/2FX

bands have a gain range of ±15dB. Input gain trims and switchable 80Hz low-cut filters are fitted to the first four strips (the ones with mic input XLRs), and all five channel strips have main rotary faders, effects sends (feeding the internal effects processor), and pan/balance controls. No input gain controls are available for the channel 7+8 inputs, so their levels must be controlled at source if they are not in the right range for the mixer. Note that there is no pre-fade foldback send, and the post-fade effects send knob feeds the internal effects processor, though it is also sent to the outside world via a jack socket if you feel the need to connect an external processor instead. Although this mixer seems best suited to live applications, it has both a main output and a control-room output, as well as RCA phono tape outs that mirror the main output. The tape input and stereo return both have level controls, and both these sources feed directly into the main mix. A headphone output, which is controlled by the C-R/Phones Level knob, provides plenty of level for monitoring, but there are no mute or solo buttons, so all you really hear is the same thing as appears at the main output, but with independent level control. The master section is gratifyingly sparse, with level controls for the main output, the phones, the effects return from the internal processor, the stereo return input, and the two-track input. A pair of 12-section LED meters monitors the mixer's output level and a prominent button activates global 48V phantom power for use with capacitor mics or active DI boxes.

Internal Digital Effects Processor Also within the master section is the effects processor, where a 16-way rotary switch selects the effect type and a further Parameter control adjusts the most important parameter of the selected effect. Typically this will be reverb time or delay repeat time. All the effects types are listed on the front panel. A switch with a large yellow status LED is used to switch the effects on and off, though another practical concession to the live performer is the provision of a footswitch socket for bypassing the effect using an optional Yamaha FC5 footswitch. Note that whenever a new effect is selected the Parameter value is set to minimum, whatever the physical position of the control, so you need to move the control if you prefer a different value. When I first tried the reverbs, I thought they sounded more like early reflections programs, because the Parameter value default meant they were set to minimum decay time. Once the decay is increased they sound reasonably convincing, and they're certainly good enough for most live applications, though they don't rival Yamaha's studio reverbs.


Yamaha MG8/2FX

The first eight programs explore various types of reverb (Hall, Room, Stage, Plate, and Drum Ambience types) so you get a rather better choice of practical treatments than you get with some mixer effects sections, and in most live situations reverb or delay is likely to be the effect of choice. Amongst the other effects are two types of delay, the usual modulation variants (phase, flange, and chorus) plus distortion — which sounds pretty nasty to me, though somebody might find it useful.

In Use There's nothing at all complicated about using this mixer, so even a newcomer can approach it with confidence. The audio path is clean, and the main line inputs and outputs are balanced, so long cables can be used with no problems. I found the EQ to sound musical enough, but the choice of a 2.5kHz mid-band frequency isn't always practical for live use, as you're often likely to want to pull out some of the low mid-250Hz range to cure room coloration or close-miked boxiness, but on a mixer of this price it isn't a big deal. Not having PFL buttons makes the process of optimising gain structure a little less straightforward than it might be, but my usual ploy is to set all the channel controls somewhere near unity gain, then adjust the input levels using the Gain trims so that the mix can be handled with all the faders in something like the same position. Using rotary knobs for the fader controls rather than sliders saves on cost and space, but isn't as ergonomic, though the white caps and thick pointer lines make it very easy to see what gains are being used, even in poor lighting conditions. All small mixers like this involve compromises to keep the costs down, but what you're left with here is still a very practical package for the solo act or duo needing a mixer to run as part of a small PA system. Having a good choice of reverbs plus the ability to adjust the reverb time also counts for a lot. As I said earlier, these aren't the best studio reverbs, but they are fine for most live applications where the artificial reverb is further diffused by the room's own reflective properties. There's nothing really unusual about this little mixer, and in the UK it has inexpensive competition from the likes of Behringer, Samson, and Peavey, but I think its better-than-average effects section makes it a strong contender. It doesn't have enough facilities to be used as a serious recording tool, but for simple live performance where you don't need to drive a foldback system from a separate pre-fade mix, it's really very good and covers all the basics. I'm not a big fan of external power supplies, but, to be fair, most small mixers use them and this one at least has sensibly thick cables and a locking connector. Small analogue mixers will always tend to be a compromise between size, features, and cost, but in this case I feel that Yamaha have probably got the balance just about right. file:///F|/SoS/SoS%2001-2006/yamahamg8.htm (3 of 4)12/19/2005 10:20:35 AM

Yamaha MG8/2FX

Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q. How can I achieve phase inversion?

Q. How can I achieve phase inversion? Published in SOS January 2006 Print article : Close window

Sound Advice

In the October 2000 issue of Sound On Sound there was an article on improving a stereo mix. In example three ('Ye olde phase trick'), you explained a technique that increases the perceived width of the stereo mix using phase inversion. I want to use this technique, but as my mixer doesn't have phase-invert buttons I'm not sure I can. I have a Signex CP44 unbalanced patchbay in which I've wired some sockets in parallel, so splitting the signal is no problem. On the returns, could I use balanced cable and swap the hot and cold pins to create the phase inversion, or would this not work, as the original signal and patchbay are unbalanced? Dez Ford Technical Editor Hugh Robjohns replies: The technique you're referring to works by adding some of the right-hand signal to the left-hand channel out of phase and some of the left-hand signal to the right-hand channel out of phase. You can read the original article on-line at www.soundonsound.com/sos/oct00/ articles/stereomix.htm, but I've included the original diagram here (see left) for reference. Many phase-inversion tricks aren't mono-compatible, but one advantage of this widening technique is that when the left and right channels of the final mix are summed to mono, the effect disappears without causing any problems. Swapping the hot and cold pins at one end of the cable feeding a balanced input will indeed invert the phase of the signal. Another way you can achieve the same thing is to use a pair of aux outputs from the main channels (one for the left signal, the other for the right) instead of splitting the signal at the patchbay. The advantages are, firstly, that the aux outputs are properly buffered and, secondly, if you make these post-fade sends, the inverted return channel levels will follow the input-signal level, making mixing easier. You may also benefit from the fact that in many budget consoles the aux outputs are phase-inverted relative to the main inputs, which would avoid the need to make up inverting cables! You'd have to experiment to find out if your mixer provided inverted aux outs. Published in SOS January 2006

file:///F|/SoS/SoS%2001-2006/qa0106_5.htm (1 of 2)12/19/2005 10:21:25 AM

Q. How can I achieve phase inversion?




Q. Should I be using my mixer's group outputs or its direct outs for recording?

Q. Should I be using my mixer's group outputs or its direct outs for recording? Published in SOS January 2006 Print article : Close window

Sound Advice

I recently started teaching music technology in a college and was asked to rebuild one of the studios. It uses a 32-channel mixing desk, patchbay and Alesis HD24 to record to, as well as outboard gear. The desk has eight group busses arranged in four stereo pairs. There are 24 mono group output sockets, three per group buss, so that group 1 goes to outputs 1, 9 and 17, group 2 goes to 2, 10 and 18, and so on. The way it was set up previously was that these 24 group outputs were normalled through the patchbay to the 24 inputs on the HD24. The students were being taught that the signal should come into the desk and then be routed through the relevant group to get to the HD24. For instance, if your mic is plugged into channel 3 and you want to go to track 5, you have to route it to group 5-6, pan it hard left and bring up the channel fader and group fader. However, I changed it so that the direct outs of the first 24 channels are normalled through to the 24 inputs of the HD24, which seems to make more sense. One of the lecturers is kicking up a fuss, so my question is: which practice is most common in professional studios? Thom Corah Reviews Editor Mike Senior replies: You're both right after a fashion, but I'm afraid that I think the lecturer is probably more right in this case, as you appear to be using a group desk, rather than an in-line one. Your approach has two main limitations. Firstly, you can only route channel 1 on the desk to channel 1 on the recorder. This is admittedly less of a limitation with a digital recorder, where you can swap tracks digitally, but it's still quicker to do this from the desk than from the recorder. The second (and more serious) limitation is that you can't record a mix of several channels to the same track on the recorder. Although 24 tracks is quite a lot to work with, you might need to submix a number of microphones to, say, a stereo pair — when layering up a string quartet a few times to make a composite string sound for a pop production, for example. Another problem is that you can't use the mixer's EQ on the way to the recorder, as direct outputs are often taken from before the EQ circuitry. Also, you couldn't bounce down a group of tracks through the desk in this way without sending them all to a group first, and then patching from the group output to a further channel. So you'll have more flexibility if you do things the lecturer's way.


Q. Should I be using my mixer's group outputs or its direct outs for recording?

One reason that you're not completely wrong is that you're implementing a kind of in-line methodology, treating the input stage up to the direct output as the input path and the rest of the channel as the monitor path. However, a group desk isn't really sufficiently well equipped to do this properly, most notably because there is no routing matrix between the input channels and the recorder inputs, as there would be on an SSL or similar. There's only one routing matrix per channel on a group desk, and that is situated after the channel fader. There's no real alternative, given the facilities, but to have separate channels for the input and monitor paths. In your case, as you have only 32 mixer channels, this means repatching for mixdown and monitoring purposes, I imagine, but I don't know all the details of your setup. One situation where you can get away with using an in-line configuration on a group desk, exactly as you have, is where the recorder is actually a computer system. In this case, given the powerful processing facilities a computer offers, there's little advantage these days in pre-processing audio before it reaches the computer, so the lack of input EQ would not really be a problem. Also, there are comprehensive input routing and mixing facilities built into most modern audio-recording packages, so a hardware routing matrix would also be unnecessary. Perhaps you could justify your routing scheme as just being a little ahead of its time? You are simply anticipating the happy day when the college moves to a more flexible computerised system! At the end of the day, which is the more appropriate arrangement depends on how many tracks you plan to record at one time. The group routing approach is more flexible when it comes to being able to do track bounces and partial submixes, and it is an important way of working to teach students. However, the down side is that you can record no more than eight (different) tracks at a time because there are only eight groups. Taking the direct outs approach allows up to 24 different tracks to be recorded at the same time and is ideal in areas designed purely for tracking, but you are then in for lots of replugging when it's time to mix. In any case, students should definitely be made aware of both techniques and configurations. One possible solution that you could consider is using the patchbay to normal the group outputs to the recorder inputs, as before, but also send all of the desk's direct outs to patchbays on the row above, so that when you need to patch direct outs straight into recorder tracks it's just a case of plugging in some patch cords. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q. Should I opt for active or passive monitors?

Q. Should I opt for active or passive monitors? Published in SOS January 2006 Print article : Close window

Sound Advice

I'm interested in buying a pair of Alesis Monitor 1 MkIIs. Should I buy the passive versions and a good amp or just go for the active versions, which cost £100 more? I've always thought that active monitors are a bit of a gimmick and don't give a good sound, but I have now been told that they will give the best sound, as there is no crossover. Can you help me? SOS Forum Post Technical Editor Hugh Robjohns replies: In the middle and upper parts of the monitor market there is no doubt that active models offer significant advantages over passive designs, such as optimised power amps for each driver, optimised driver-protection circuitry, short and direct connections between amps and drivers, more complex and precise line-level crossovers, and so on. However, at the budget end of the market these advantages are somewhat clouded by the inherent problems of achieving a low sale price. Most notably, many models are saddled with poor-quality power amps and power supplies that have been built down to a price rather than built up to a standard. Obviously, I'm painting pictures with a very broad brush here — there are some good and some less good designs out there — but the generalisations are true. Active speakers come in two forms: true 'active' monitors, which have a separate amplifier for each driver, and 'powered' monitors, which have a single amplifier built into the cabinet, feeding both drivers via a normal passive crossover. In examples of the latter, you often get a better amplifier because you are only paying for one amp and not two (or three, in the case of a true active three-way monitor), while retaining the advantages of having an integrated package with very short internal speaker cables and so on. In the case of a well designed two-way speaker, a passive crossover can deliver superb results, and there is often little, if any, quality advantage from employing a complex line-level active crossover instead.

One advantage of passive monitors is that the two components of your monitoring system — the speakers and the amp — can be upgraded separately, allowing a more gradual and less expensive progression to better-quality gear.

However, one facility that's easy to implement in active designs with line-level crossovers is user-adjustable EQ tweaks. These can be helpful sometimes in matching the speaker to the room, but in inexperienced hands file:///F|/SoS/SoS%2001-2006/qa0106_7.htm (1 of 2)12/19/2005 10:21:34 AM

Q. Should I opt for active or passive monitors?

such facilities can often be more trouble than they are worth because they can be mis-set... and usually are! Perhaps a more relevant argument against budget active speakers — for me, at least — is the difficulty of upgrading. When the time comes to move up to a higher standard of monitoring, you will have to change both the speaker and its integrated amps. This inherently means that upgrading has to jump in large financial steps. On the other hand, if you go down the passive route you can upgrade the speaker separately from the amp, and vice versa. That approach allows you to improve the quality of the complete system in several easier and more cost-effective stages. For example, you could start off with the best passive monitors you can afford and a reasonable amp (possibly second-hand — there are plenty on the markets as people switch to the more 'fashionable' active monitors), then maybe upgrade the amp to something that will warrant a better speaker after a year or two, then upgrade the speaker, and so on. For what it's worth, all my 'little speakers' are passive designs coupled to good quality amps, in some cases with the amps fixed to the back of the speaker to make a 'powered' unit. I have found this approach to provide the best-quality result whilst still being very cost-effective and flexible. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q. What is 'aliasing' and what's the cause of it?

Q. What is 'aliasing' and what's the cause of it? Published in SOS January 2006 Print article : Close window

Sound Advice

With reference to A-D/D-A converters, what exactly is an 'alias'? How and when do they occur? SOS Forum Post Technical Editor Hugh Robjohns replies: An alias occurs when a signal above half the sample rate is allowed into, or created within, a digital system. It's the anti-aliasing filter's job to limit the frequency range of the analogue signal prior to A-D conversion, so that the maximum frequency does not exceed half the sampling rate — the so-called Nyquist limit. Aliasing can occur either because the anti-alias filter in the A-D converter (or in a sample-rate converter) isn't very good, or because the system has been overloaded. The latter case is the most common source of aliasing, because overloads result in the generation of high-frequency harmonics within the digital system itself (and after the anti-aliasing filter). The sampling process is a form of amplitude modulation in which the input signal frequencies are added to and subtracted from the sample-rate frequency. In radio terms, the sum products are called the upper sideband and the subtracted products are called the lower sideband. In digital circles they are just referred to as the 'images'. These images play no part in the digital audio process — they are essentially just a side-effect of sampling — but they must be kept well above the wanted audio frequencies so that they can be removed easily without affecting the wanted audio signal. This is where all the trouble starts. The upper image isn't really a problem, but if the lower one is allowed too low, it will overlap the wanted audio band and create 'aliases' that cannot be removed.

Figure 1: The D-A converter's low-pass filter, set at half the sample rate, removes the upper and lower images while keeping the wanted audio.

Figure 2: When the 10kHz signal overloads the A-D converter, the resulting third harmonic at 30kHz creates an alias at 18kHz which will be allowed through by the lowpass filter.

Let's consider what occurs if we put a 10kHz sine-wave tone into a 48kHz sampled digital system. The sampling process will generate additional signal frequencies at 58kHz (48 + 10) and 38kHz (48 - 10). Both of these images are clearly far above half the sample rate (24kHz), so can be easily removed with a low-pass filter, which is the reconstruction filter on the output of the D-A converter, leaving the wanted audio (the 10kHz


Q. What is 'aliasing' and what's the cause of it?

tone) perfectly intact. See Figure 1, above. However, consider what happens if our 10kHz tone is cranked up too loud and overloads the A-D converter's quantising stage. If you clip a sine wave, you end up with something approximating a square wave, and the resulting distortion means that a chain of odd harmonics will be generated above the fundamental. So our original 10kHz sine wave has now acquired an unwanted series of strong harmonics at 30kHz, 50kHz and so on. Note that these harmonics were generated in the overloaded quantiser and after the input anti-aliasing filter that was put there to stop anything above half the sample rate getting in to the system. By overloading the converter, we have generated 'illegal' high-frequency signals inside the system itself and, clearly, overloading the quantiser breaks the Nyquist rule of not allowing anything over half the sample rate into the system. Considering just the third harmonic at 30kHz for the moment, the sampling modulation process means that this will 'mirror' around the sample rate just as before, generating additional signal frequencies at 78kHz (48 + 30) and 18kHz (48 - 30). The 18kHz product is clearly below half the sample rate, and so will be allowed through by the reconstruction filter. This is the 'alias'. We started with a 10kHz signal, and have ended up with both 10kHz and 18kHz (see Figure 2, above). Similarly, the 50kHz harmonic will produce a 2kHz frequency, resulting in another alias. Note that, unlike an analogue system, in which the distortion products caused by overloads always follow a normal harmonic series, in a digital system aliasing results in the harmonic series being 'folded back' on itself to produce audible signals that are no longer harmonically related to the source. In the simplistic example I've explained, we have ended up with aliases at 2kHz and 18kHz that have no obvious musical relationship to the 10kHz source. This is why overloading a digital system sounds so nasty in comparison to overloading an analogue system. I hope this brief explanation helps to clear up the topic of aliasing for you. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q. What should be the next step for my studio?

Q. What should be the next step for my studio? Published in SOS January 2006 Print article : Close window

Sound Advice

I'm looking for advice on what's the best next step for improving my signal chain. I've got a budget of about £1000. I'm writing music to picture (so mainly instrumental, not too 'in your face'), mainly using sample libraries and the odd MIDI sound piped in from my Roland XV5080. I'll occasionally record live sources, but have so far never needed more than two inputs to do so. My setup currently includes a Yamaha 03D mixer and a Focusrite Twin Trak Pro preamp, both with word clock, AES-EBU and S/PDIF I/O), and an M-Audio Delta 1010 audio interface with S/PDIF and word clock I/O. I never use more than two inputs on the 1010, but do use all eight outputs into the 03D for mixing down to a stereo pair. My sequencer is Apple Logic Pro 7 on a Mac G5, and I'm using Adam S3A monitors. I spent a large part of the Summer knocking together traps and absorbers for my room, which now sounds pretty good. So where would I get most bang for my buck? Should I get a word clock generator to help tighten things up? Something to replace the M-Audio interface? A friend suggested a summing mixer, but there seems to be so much discussion about whether these are voodoo or not that I wonder if I'd be capable of hearing the differences through my tin ears! I also appreciate that it might be worth going for better preamps, but I do so little live recording that I can't help but wonder if there's something else that would make more of an impact on my day-to-day studio work. Any thoughts? SOS Forum Post Reviews Editor Mike Senior replies: There are a few different possibilities here, I'd say. I can understand that you're a little reluctant to shell out masses of money on a new preamp, but have you thought of looking at a decent A-D converter instead? Something like the RME ADI2 would be a neat product for your purposes and price range, as it offers both A-D and D-A conversion, so it would not only improve the quality of all input signals, but also increase the resolution of the digital output from the Yamaha desk that feeds your main monitors.



Further up the scale are Apogee's Rosetta 200 and RME's ADI96 Pro, and although these are a little over the price range you stated, you should still give them some consideration, as they both have word clock outputs from their high-resolution internal crystals. Clocking your entire system from the converter's word clock output would then upgrade the sound of all the converters in your soundcard and mixer — in the case of the original 0-series mixers in particular, I've heard that this can make a big difference to the sound.

A monitor controller such as the Mackie Big Knob, a quality A-D/D-A converter such as the RME ADI2 or an external plug-in processor such as the TC Electronic Powercore Compact or Waves APA44M would be welcome additions to just about any studio.

Another possibility to consider is a dedicated monitor controller and a second pair of monitors to go with it. This would allow you to stay objective about your mixes by flipping between the two sets of monitors during the mixdown process. The second pair of monitors doesn't need to be as good as your main pair, and could even function as the 'grot boxes' — monitors that will give you some idea of how your material will sound on lowerquality domestic systems. The other advantage of a monitor controller is that it would allow you to quickly check the mix in mono (still important for broadcast work), and would also allow you to quickly and easily audition external sound sources on your main monitors — perhaps the output of a CD player or television, again for referencing purposes during mixing. Towards the lower end of your price range is the Mackie Big Knob, which is very flexible, but you might also want to have a look at the SPL Model 2381 or Presonus Central Station, the former for its more 'audiophile' bias, and the latter for its built-in digital source monitoring facilities and remote controller. A final suggestion would be to look at some of the add-on DSP processors currently available, in order to increase the number and quality of plug-ins and virtual instruments that you can run using your single G5 Mac. There's lots of choice here within your price range, such as the Universal Audio UAD1 processor card bundles, several varieties of TC Electronic Powercore (both PCI and external Firewire), and the new Waves APA series. Which one you go for will depend on what plug-ins you're most likely to use, but you should see a significant increase in audio processing power in all cases. I imagine you already know how useful it is to have everything running live when the director changes the brief at the last moment! Forum member Tomás Mulcahy also suggests exchanging the G5's audio interface for one with ADAT I/O, and then also buying the ADAT-equipped mini-YGDAI expansion board for the 03D. This would remove unnecessary extra stages of A-D and D-A conversion when transferring audio from the computer to the mixer, and would certainly improve the sound quality of the whole system. RME's Multiface and Digiface are both possibilities well within your price range, and as RME also have an excellent reputation for digital clocking you should still see some improvement in the Focusrite channels' A-D conversion and the Yamaha mixer's D-A conversion. Whether you want the analogue I/O provided on the Multiface will depend on your future expansion plans, but by the sound of things there's not that much need for it in your system, so the Digiface might be the better bet, as well as the cheaper one! Published in SOS January 2006






Q. Which microphone should I buy for recording vocals?

Q. Which microphone should I buy for recording vocals? Published in SOS January 2006 Print article : Close window

Sound Advice

I do recording work with a number of acoustic performers but I'm having trouble finding a suitable mic for recording vocals. For recording acoustic guitars I use AKG C1000s and I get a great recorded sound with no problems. For vocals I was using an SE Electronics large-diaphragm condenser mic, which sounded OK, but I thought I'd upgrade and went for an AKG C3000. What a mistake! The recorded sound was very harsh, sibilant and unnatural — all the performers hated it! I've now resorted to using AKG C1000s for vocals as well as guitar, and they do a good job. However, I do want to get a purpose-designed vocal condenser mic, but it must be one that has a warm and smooth sound. As always, I'd be grateful for your thoughts and comments. Having researched the available mics, I think the Rode NT2A seems to fit the bill, and it got a good review from Paul White in SOS, but would it be the correct choice? I have a budget of around £200. John Ablitt Editor-In-Chief Paul White replies: If you need a smooth-sounding vocal mic, the Rode NT2A is a very good choice within its price range, but don't get rid of all your other mics, as different voices benefit from different mic characteristics. I've found that the SE Electronics mics have a pretty 'in the middle' kind of sound, so they should suit a wide range of singers. Also take a close look at your recording environment itself, to see whether that is colouring the sound excessively. Hanging duvets behind the singer, to reduce the amount of reflected room sound reaching the vocal mic, is always a handy quick fix if you don't have a proper vocal room or booth set up. Published in SOS January 2006

Both the Rode NT2A and the SE Electronics Z3300A can help provide good quality vocal recordings on a budget.


Q. Which microphone should I buy for recording vocals?

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


Q. Why does my Mackie Control make strange noises in Cubase?

Q. Why does my Mackie Control make strange noises in Cubase? Published in SOS January 2006 Print article : Close window

Sound Advice

I'm using a Mackie Control control surface with Cubase SX, and it works fine on audio tracks. However, whenever I select a MIDI track within Cubase, pressing buttons on the Mackie Control seems to trigger random MIDI notes, and using the other controls sometimes seems to make my synths go out of tune. What's going on? Jeremy Carter Features Editor Sam Inglis replies: Mackie Control and similar control surfaces communicate with Cubase via MIDI, and they use ordinary Note On and Continuous Controller messages to tell the computer that a button has been pressed or a fader moved — but not ones that will have any musical relevance to your song! Meanwhile, the default preference in Cubase SX is that whichever track is selected is automatically record-enabled, and all MIDI tracks default to accepting MIDI input from all connected sources. This means that if you have, say, a controller keyboard and a Mackie Control connected, Note On and controller messages from both will be recorded on the selected track. Even when you're not recording, all MIDI messages from all sources will be routed to whatever synth is attached to the selected track. The answer to this is to change the input selection for each of your The Mackie Control works via MIDI, so keep MIDI tracks. In the track Inspector, change the MIDI input from 'All' to an eye on the input assignments of your MIDI tracks. a specific device that's not the Mackie Control, or 'None' if you don't want them to accept any MIDI input. If you're not planning on recording any MIDI, you could also achieve the same result by visiting Cubase's Preferences and deselecting the 'Record enable selected track' box. Published in SOS January 2006


Q. Why does my Mackie Control make strange noises in Cubase?





In this article:

Taking Wing Love-hate Relationships Travellin' Man Logical Progressions

From 4AD To Nine Inch Nails John Fryer Published in SOS January 2006 Print article : Close window

People : Artists/Engineers/Producers/Programmers

The likes of Depeche Mode, Cocteau Twins and Nine Inch Nails all owe a sonic debt to engineer/producer John Fryer. Tom Doyle

For over 25 years, John Fryer has managed to sustain a career working as a producer in such diverse fields as proto-electronic pop, ethereal mood music and alternative rock. His discography runs to dozens of albums, singles and remixes, involving everyone from Depeche Mode to the Cocteau Twins to Nine Inch Nails, and most recently, Finnish multi-platinum goth rockers HIM. But now, after a quarter of a century spent in darkened studios, Fryer has largely turned his back on the world of commercial recording facilities, particularly since 2003 when he set up his label Something To Listen To Records. Instead, he prefers to run his operation from his Ladbroke Grove flat, where he works on an Apple Mac G5/Logic Pro 7 setup. "I'm not a gear snob," he states. "If I think something's got a good sound, I'll use it, I don't care. I know from talking to record companies that sometimes when they employ a producer, he wants this studio and this desk and he's got to have these preamps and so on. Whereas I'll work anywhere through anything and I'll still get a good sound. I've made records through Peavey desks, Mackie desks, Harrison desks, SSL desks, Amek desks. I don't care about numbers and what they are. The sound is all I care about." Having first started working as an assistant engineer at Blackwing Studios in London's Southwark at the beginning of the 1980s, John Fryer has witnessed the

file:///F|/SoS/SoS%2001-2006/johnfryer.htm (1 of 8)12/19/2005 10:22:29 AM


rapid technological developments of the last 25 years first-hand. "I've been through a lot of changes, from pre-MIDI, pre-SMPTE to total computers." From his perspective, he argues that the low point of audio advances came in the 1990s when he feels that equipment began to get in the way of the actual process of making music. "It got too technical," Fryer says. "And to be honest with you, I think the less technical stuff you have in the way, the better the songwriting is. The first Depeche Mode album was made on eight-track. Just because you've got 48 tracks doesn't make a better song. And with a computer, just because you've got 164 tracks, it doesn't make a better song."

Taking Wing Back in 1980, John Fryer more or less stumbled into the recording industry when some friends of his in a long-forgotten band booked into a London studio to record some demos. That studio happened to be Blackwing, the facility where Fryer would cut his teeth as an engineer and carve his reputation as a producer over the next nine years. "It was a total accident," he remembers. "The guy who ran the studio, Eric [Radcliffe] was looking for an assistant and a few weeks later I was made redundant. And so he gave me a job. It's where I grew up in the music world. I had a very good education because it was the alternative indie world, working for Mute and 4AD and Beggars Banquet. There wasn't so much major work, it was all independent work and so it was very artistic, working with Ivo [Watts-Russell, 4AD] and Daniel [Miller, Mute]." Fryer admits that when he first entered Blackwing, his knowledge of recording technology was incredibly basic. "I knew what most people knew which was recording onto a cassette or a reel-to-reel and playing it back. That was about it."

The outboard rack in John Fryer's home studio, with (from top) ART Pro MPA preamp and Pro VLA compressor, three guitar processors — Line 6 Pod XT Pro, Sansamp PSA1 and Washburn WSR42 — and MOTU 896 recording interface.

But at Blackwing, the technical rulebook had pretty much been thrown out of the window anyway. "The bands I was working with at the time, it was like the rules were there to be broken. If it made a noise, it was an instrument. It was a sound to be used."

Blackwing, he recalls, was a hive of hyper-productivity, with lack of cash forcing bands to make their records at a breakneck rate. "Back then you'd make an



album within about five days," Fryer says. "The turnover was quick. Exciting but exhausting. Every record was totally different, the styles of music were totally different. The average day then was probably about 14 hours. You had to work very fast if you had a week to make an album, or two if you were lucky." Can Fryer confirm or deny the long-standing rumours that Blackwing, situated in a church, was haunted? "Well, there were stories, yeah. I didn't actually see or hear anything, but people said they did. You'd make noises and within the noise you could hear other noises and odd voices." Home to spooks or not, for a period of time in the early '80s, chiefly thanks to Depeche Mode, Blackwing became the home of the hits. The Basildon quartet recorded their first two albums there, 1981's Speak & Spell and 1982's A Broken Frame, with Fryer engineering using a Tascam eight-track and rudimentary sequencing. Favoured synths of the day included the Kawai 100F, Korg Minikorg 700s, Moog Prodigy, Roland Jupiter and SH1, Yamaha CS5, and perhaps most importantly, the ARP 2600. "Back then the equipment was so limited, you had to work out ways of getting the most out of everything," Fryer recalls. "There were no sync tones, so we were using the ARP and its analogue sequencer, and because it worked on CV and Gate, we devised a way of recording the click from that and feeding it back on itself, so you'd get a couple of chances of running sequences in time with the tape. You'd record a kick and snare on it and everything was played live over the top. The drums were the ARP and then it was Moog bass and so on. We used whatever was available. If it wasn't there, it was hired in." Soon, Fryer had moved up from engineering and was becoming more involved in production. "As was quoted somewhere on the Internet, someone said that as an engineer I couldn't keep my mouth shut. But people liked working with me and then they'd come back and say would I co-produce or produce their records." Among Fryer's first clients were the Cocteau Twins. The fledgling producer helped give birth to a sound that was utterly unique, built around Robin Guthrie's cavernous, echoing guitar work and Liz Fraser's otherworldly, if often indecipherable, vocals. He worked on their first two albums, Garlands in 1982 and Head Over Heels in 1983. Photo: BBC archive / Redferns John Fryer made his name as an engineer "The first one we did at Blackwing and on the first two albums by Essex synth-pop the second we did at Palladium in innovators Depeche Mode. Edinburgh. On the first album they were very shy — I don't know if they'd even been out of Scotland before. But, yeah, it was a unique sound and



obviously Robin developed it the more they went on. A lot of it back then was done with Boss pedals — particularly the delay and the chorus. And we used the old AMS for chorus and delays and, of course, Lexicon reverbs. All the good stuff that's still in studios today, like the MXR Harmoniser and the Roland Space Echo. "The second album we experimented with putting the guitar through anything and everything. At one stage we fed a guitar through a Yamaha electric grand piano and miked it up. That's the way I've always been with the bands I've worked with — you put anything through anything and see what sound you get." Working with 4AD's Ivo Watts-Russell as part of the label boss's This Mortal Coil offshoot, Fryer helped create a classic with Guthrie and Fraser's atmospheric take on Tim Buckley's 'Song To The Siren', originally the 'B' side to 1983 single 'Sixteen Days (Gathering Dust)'. "It still stands up today," the producer rightly points out. "But it wasn't something that we consciously put a lot of effort into. As you can hear it's very simple and it was done, like a lot of classic tracks, as a throwaway 'B' side, so there was no pressure. It wasn't overproduced or over-mixed. We always used Neumanns on Liz's voice and we didn't normally have to do much — add a bit of top, take some bottom out and the vocal usually sounded brilliant. "It's funny when you record things — a lot of the time you think everything's special and when you stand back from it some time later, you realise that sometimes it wasn't as good as you thought it was or sometimes it's better than you thought it was. But that one just always sounded beautiful."

Love-hate Relationships Come 1989, John Fryer decided to leave Blackwing after a prolific nine-year run and went freelance. In the first year of going it alone, his clients included Lush ("It was down to me that they had distortion on their guitars") and Nine Inch Nails. Trent Reznor, confessing to being an unlikely fan of some of Fryer's more ambient works, asked him to produce the ground-breaking industrial rock of Pretty Hate Machine. "A lot of bands like records that I've made and get in contact," Fryer explains, "and a lot of the industrial scene in America liked all the This Mortal Coil ethereal side and the electronic records I made for Mute. Pretty Hate Machine was sequenced using a Mac — the little one with the built-in screen — though I can't remember which software he was using at the time. The guitars were recorded using the Yamaha SPX50, just straight into there and out to the desk. "We were trying to make the hardest record we could make. It was very strange because we made it, we thought it sounded brilliant, we had it on the big



speakers just blowing us away. Then someone from the record company came in — and because the demos were more synthy and not as industrial as the album, he listened to it and his mouth dropped open and he said 'You've ruined this record.' But of course it's gone on to be a classic. It was done in 20 days. I think it was a good thing that we made records so quickly back then because there's a lot of energy in there and mistakes are left in, so it sounds human and it's not blanded out over time."

Travellin' Man For most of the '90s and into the current decade, John Fryer spent much of his time in America, working in various locations with the likes of Stabbing Westward, Raging Speedhorn and Cradle Of Filth. He says that, particularly after years of working in the same studio, this itinerant lifestyle suited him fine. "Normally it was easier for me to fly to a place than it was to fly the band. But I always tried to end up back in Battery Studios in New York to mix 'cause they had my favourite Boxer speakers there. They're just big, true speakers. You could have them loud all day and they didn't tire you out. "The thing I find with working in studios is that every studio you go to sounds completely different, and so Battery was like having a reference point. When I worked in Blackwing all the time, I knew the sound that I could get. We had an Amek 2500 desk which I really enjoyed, the EQ was very good, you could kind of alter everything. But when you go to different studios every week and they all sound completely different, you don't know what you're doing. You think you're doing one thing that sounds good on that speaker and then you take it away and it sounds like shit." In Fryer's experience, US recording studios also vary wildly in terms of quality. "They have a funny attitude to A Rooms and B Rooms in the States," he says. "The A Room is like a million dollars a day and the B Room is like five bucks a day — and it sounds like five bucks a day. It's usually just a desk and speakers and you have to rent in other gear. In England, when you rent a studio, you get everything in the studio." One of the reasons why Fryer says he enjoys working at home, especially when it comes to mixing, is because he's incredibly pernickety — an expensive business when working in top-flight facilities. "I like to mix at home rather than the studio because you're not pressurised for time. Also, when I used to work in Battery on the SSL, you'd do a mix and then you'd listen to it and want to change something, so you'd have to come back a week later and do a total recall. And remember, a total recall on SSL was only the desk — you couldn't recall all the outboard. It was close but never the same. "Now it's easy because if I mix something at home and send it to the artist on CD and then they ask for changes, I'll just open the file, lift the vocal or whatever and send it back to them. I love the fact that I can come back to a mix a day later or a week later, open the folder and it's exactly how I left it."

Logical Progressions



Fryer is a devout Apple Mac and Logic man and says he can trace his loyalty to the latter back to his days using its Atari-based predecessor, Creator. "The thing about Logic is as they've developed it, they've just enhanced the program, they haven't completely changed it. So it still functions as well as the first version and still has a lot of the same commands. Even though we've progressed 15 years or whatever, the MIDI side of it is still the same, just as good as it ever was. I was talking to someone the other day who only gave up using Creator four years ago, 'cause it was rock solid. I just find it easy to use, I find it totally logical. Even when they bring out a new version, it only takes probably an hour or so and then you're up and running again and everything makes sense." Anything he doesn't like about it? "Well, I think some of the old functions were a lot more user-friendly than the things they've got now. In the mixer page, you used to be able to drag across a few channels, select them all and move the faders up and down. Now you can't and you have to go in and select them all individually or go into your groups, which is time-consuming. It was something that I found totally useful when you'd pushed your mix too high." Over recent years, Fryer has purposely got rid of a lot of his software effects. "This is the thing I was talking about with technology being distracting," he says. "When I had the earlier versions, I'd sit there all day and have a sound and try this plug-in and that plug-in. So now I just use all the Logic stuff because they all sound brilliant and you don't have to spend all day going through banks and banks of plug-ins to find something to manipulate the sound. Everything that's in there will get used, whether you multi-layer the plug-ins or record it and then manipulate it again. It depends on what kind of music I'm working on — if it's just straight acoustic stuff, I want to make it sound as pretty as possible and if it's power noise stuff, I distort the hell out of everything." Fryer's setup is built around his dual-2.3GHz G5 and MOTU 896 Firewire interface. Monitor-wise, he uses the Mackie HR824s. "They just have a beautiful sound, it's very clear. You can hear the EQ, you can hear the frequency changes. With some speakers, you can sit there twiddling the knob and not hear a thing. They've got a lot of bottom end and they do seem true to the outside world. Before I had a pair of Fostex speakers, the ones with the orange cone [NF Series]. They were more like the NS10s, but they sounded a bit grey. You couldn't really hear all that well, they were a bit too middly and I don't think they were that true and that flat." Like many producers, Fryer has an ambivalent attitude towards the onceubiquitous NS10s. "They became the norm, the standard reference and you'll find a point somewhere in the late '80s to '90s where every record had the same sound. Then when everyone started using different speakers, records sounded different from each other. You obviously mix to the speaker. If it's not giving you a lot of mid, you overdo the mids. So now I think it's much better." For a second pair of speakers, Fryer relies on a pair of Harman Kardon Soundsticks. "They're a good reference. They used to be USB powered but now they're on a jack and you can plug them into anything. They're a great reference file:///F|/SoS/SoS%2001-2006/johnfryer.htm (6 of 8)12/19/2005 10:22:29 AM


point." A Mackie fan through and through, Fryer finds he can mix everything using his 1202 VLZ Pro. "I just like the sound it makes. I used to have a little Soundcraft Folio and then when I was over in Germany, Tom from [Something To Listen To band] Sundealers owns a music shop. I'd mixed some stuff through the Mackie over there and it broke, so we got in the Soundcraft and it was very smooth-sounding and clean. The Mackie Among the acts on Fryer's STLT seemed to add a bit of edge, a bit of dirt. I think label is his own band Esoterica. this is the one [the Prodigy's] The Fat Of The Land was mixed through... you get those kind of drum sounds with it. I separate the channels out and don't mix totally internally because the single stereo channels compress and change the sound too much. So I split the signal down to four stereo channels and then put it back into itself. If I had a bigger desk, I'd probably split it up a bit more." While the producer admits that "there's not much left in my rack these days", he sings the praises of his ART Pro MPA preamp and Pro VLA compressor. "They're just very clear-sounding and inexpensive. My favourite compressor is the Dbx 160. You can make them subtle or you can have them quite extreme. I think the Logic compressors have a kind of similar sound to them. You can really over-compress stuff and it doesn't really damage the sound." The rest is made up of guitar processors, including the Washburn WSR42 ("I used that back in the day with the Cocteau Twins — graphic EQ plus double distortion"), Sansamp PSA1 ("Great for a certain heavy guitar sound which I do believe Rammstein were using for a while") and Line 6 Pod XT ("A great selection of amp simulators"). But even with his stripped-down setup, Fryer admits that he can still get lost in a mix for days on end. Such, it seems, is the lot of the perfectionist record producer. "I'll be mixing something and I'll come back to it again and again," he grins. "In a studio, a mix might take a couple of days. The assistant will be saying 'OK, let's run the mix down,' and as we're halfway through, I'll be saying 'One more change, one more change.' We can go on all day doing that. Now of course at home I can almost go on forever. That's the problem with producing records. There always seems to be one more change you can make..." Published in SOS January 2006






Futurism...

Futurism... Paul White's Leader Published in SOS January 2006 Print article : Close window

People : Industry/Music Biz

Due to a temporal anomaly in the SOS server caused by a stray tachyon pulse, we managed to get hold of this Leader column from the year 2016, and thought you might like a peek too... It's five years since we completely discontinued the paper version of Sound On Sound, following the introduction of Apple's now-ubiquitous iSlate, which many would argue is close to rendering most printed material obsolete. The latest version of the iSlate, in case you haven't seen it yet, provides two A4 pages, folds up to a size smaller than most paperbacks (it's less than a quarter of an inch thick) and has a battery life of around 28 days with a following wind — longer if you're reading on the beach, as the built-in photocells charge the battery for you. For me, the iSlate's biggest breakthrough is its dual-mode screen — conventionally backlit for indoor use or switchable to reflective mode when reading in bright sunlight. I can hardly believe we used to use the old TFT screens outdoors, where they were virtually useless! Apple's original plan, of course, was that the iSlate would augment traditional reading libraries, by allowing users to download books without having to leave home; even the storage capacity of the very first iSlate allowed anyone to carry hundreds of heavily illustrated books around with them at the same time, and I've lost count of how many times its memory has been upgraded since then. We've recently heard that the next software revision will enable the iSlate to read the books to you, if you're foolhardy enough to try to go anywhere in a car! As far as I'm concerned, one of the best things about Sound On Sound becoming a SlateZine is that we've been able to use its audio capabilities to add fully integrated audio and video content directly to the magazine. The iSlate's ability to run music software directly is fantastic for interactive reviews and demos, and of course it handles audio and video as well as any mainstream computer. The fact that you don't have to go out to the shops to buy it any more is also a bonus — not that it's too easy to find a shop these days! Still, at least we've seen a big increase in dealer advertising since most high streets became heritage centres. I think we'd all agree that Internet shopping, the £20 gallon of petrol and the £100 per-day congestion charge in large towns and cities are responsible for taking most walk-in retail stores (other than food markets) off the map. Not all of us are entirely happy about this, but from the magazine's point of view it's great that this state of affairs has prompted more people to stay at home and make music, especially following the controversial (to say the least) ban on alcohol in all public houses. We're finding that now the hypernet is crammed full of competing eSales outlets, targeted SlateZine marketing is the best way to get end users to visit retail sites. Virtual shopkeepers are waking up to the fact that they need more than their own virtual store to bring in business. file:///F|/SoS/SoS%2001-2006/leader.htm (1 of 2)12/19/2005 10:22:32 AM

Futurism...

'Zines' such as SOS also remain the only source of unbiased opinion, as the net carries more 'advertorial' material than ever. To turn to another subject, I'm pleased to report that the SOS Readers' Records on-line 'label' has just topped the £10,000,000 turnover mark. Of course, we're still very small potatoes compared to Apple's iTunes (confirmed recently as the biggest record company ever to exist, since it now handles over 90 percent of the world's music distribution and download business). Indeed, since the legal ruling that iTunes had to put up for sale any music submitted unless disbarred for profanity reasons, their turnover has doubled, most of the previously big-name record companies have 'left the building' and the word 'unsigned' has completely disappeared from the musician's vocabulary. Oddly, though, surround sound still hasn't caught on and my wife still won't let me have that nice 12-foot flat-screen TV in the lounge! Paul White Editor In Chief Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


file:///F|/SoS/SoS%2001-2006/leader.htm (2 of 2)12/19/2005 10:22:32 AM

Sounding Off: Monitors

In this article:

About The Author

Sounding Off: Monitors Mike Senior Published in SOS January 2006 Print article : Close window

People : Sounding Off

Oh, monitors, so much to answer for... Mike Senior

Let's be honest: there are few things less inspirational in the home studio than monitors. They aren't the best synth for trancebag happycore. They can't create a daft-mask-metal riff from a couple of lame barre chords. And they won't convolve pan-pipe solos into the deepest of underground caverns, however much this might be considered a step in the right direction. In fact, short of heaving your speakers out of a hotel-room window, their rock and roll rating won't win you many games of Studio Top Trumps. The result of this is that a large number of homestudio owners spend more on cheese than they do on monitors and acoustic treatment, and, despite the importance of good dairy products in everyone's diet, I think this is a false economy. But before you turn back to the Readers' Ads with a dismissive comment about bears and Catholicism, there are a lot more reasons for this than most people stop to think about. I'd imagine that the most common reason you think you need good monitors is because you want to mix down or even master your recordings at home. Without decent monitoring, you've no real choice but to mix somewhere that does have it (unless you're pioneering a new 'broken' sub-genre of Industrial). The problem with this is that all studio control rooms sound different, however professional they might be. So how do you know how your mix should sound in the one you've chosen?

file:///F|/SoS/SoS%2001-2006/soundingoff.htm (1 of 3)12/19/2005 10:22:35 AM

About The Author By day Mike Senior is SOS Reviews Editor, but by night he transforms into the amazing Acoustic Foam Kid! Or occasionally back into a pumpkin. If you would like to air your views in this column, please send


By listening to a selection of reference records your submissions to soundingoff@ before and during the mixing session, of course. soundonsound.com or But if you're paying for this time, that's got to hurt! And even if you've pulled a freebie (you sly devil), to the postal address listed in the front of the you're hardly going to spend as long in that control magazine. room as you spend in your own studio. If you can hear what you're doing properly at home, every record you play for fun will help to train you for mixdown. Conversely, if your monitoring sucks, every CD you play in your studio will move you closer to monkey-butt status. And before you say 'I'll get someone else to mix my track!', how will you tell whether they've done a good job? Or, to put it another way — could you even afford Spike Stent's toenail clippings? What I'm saying is that the main reason I think your monitoring system is so important is that it affects the entire development of your listening and engineering skills. On every session you make a hundred little decisions based on what you're hearing, and that is the basis of the learning process. The ropier your monitoring, the dodgier will be the techniques you're acquiring. And when you finally have decent monitoring, you'll need to spend a lot of extra time working out why your normal tricks don't seem to work any more, as well as scrabbling around for new tricks to replace them. Oh, and that's after you've finished listening to all those reference tracks, of course. And don't forget all those gear-buying decisions you make. If the bass in your room is all over the place because your monitors and room acoustics suck, how exactly are you going to tell which bass synth is the phattest, for phuck's sake? Or whether BFD, DFH, or WTF has a better kick drum? Every erroneous purchase you make is likely to cost you money (to replace), time (in trying to compensate for its deficiencies), production quality, or a cosy combination of all three. And I speak from experience. Fresh from working in a professional studio, I decided to set up a home rig and grudgingly opted for the cheapest active monitors I could find. I quickly became unable to mix my way out of a wet paper bag, despite having found mixing a very natural and logical process before. To make things worse, however, I initially made the mistake of believing the new monitors and doubting the rest of the equipment, and spent happy hours unlearning a whole load of useful things I'd picked up in professional studios, while learning a load of rubbish as a replacement. Add a couple of ill-advised equipment purchases, and you can see why I'm still kicking myself for wasting all that time, energy and money. I'd get Paul White to kick me too, but he might enjoy it too much... So let me be crystal clear: monitoring and acoustic treatment may be dull as digital dither through a low-pass filter, but unless you can hear what you're doing you'll be haemorrhaging money, time, and engineering skills. Do yourself and your music a favour: sort it out! Don't make me set Paul White on you... Published in SOS January 2006






Studio SOS

In this article:

Control-room Ergonomics Adjustments Monitoring & Acoustics Overhaul Listening Tests

Studio SOS South Published in SOS January 2006 Print article : Close window

People : Studio SOS

The SOS team get busy at the band South's London studio, transforming the sound of their troublesome monitoring room. Paul White & Hugh Robjohns

This month's Studio SOS visit comes from an industrial building in London, where the band South have set up their own studio. The band comprise Joel Cadbury (vocals, bass, guitar, synths), Jamie McDonald (vocals, guitars), and Brett Shaw (drums, keyboards), and they have enjoyed a certain amount of commercial success since releasing their first album From Here On In on the Mo Wax label back in 2001. Their second record, With The Tides, was released by Sanctuary in 2003, and they are now working on their third album, which should arrive in the shops in March 2006. They're doing all their recording in their own studio, and will have mixed eight of the tracks from their new album there, including two with producer Dave Eringa. (Dave is also mixing two of their tracks at Brit Row studios.) The rooms had already been treated with some thin acoustic foam tiles Joel had bought on Ebay, and these had been positioned mainly on the front and right walls of the new control room, file:///F|/SoS/SoS%2001-2006/studiosos.htm (1 of 5)12/19/2005 10:22:37 AM

Studio SOS

with a few on the ceiling. The left wall was mainly taken up by the controlroom window. The rear of the room featured the semicircular shell of a spiral staircase intruding into the room so, quite sensibly, they'd covered this convex surface with a duvet and some heavy drapes to cut down on reflections. A narrow space to one side of the stairwell structure accommodated a set of free-standing shelves, while the entrance door was at the other side. The front stud wall of the room was also angled in several segments, presumably to help control reflections when it was used as a studio. With their Dynaudio BM6A speakers facing down the longer axis of the 12 x 8.5foot room, the symmetry was also thrown out a little by the ceiling, which was sloping from left to right rather than from front to back as would have been preferable.

Control-room Ergonomics Adjustments It was immediately obvious to us that the monitoring balance was inconsistent at different points around the room, and we also noticed that the band's Mackie eight-buss console was set up on a very high desk, barely a couple of inches below the monitors, which were on their sides on stands. This arrangement produced very strong early reflections from the console, while the sideways monitor mounting narrowed the sweet spot. It also caused ergonomics problems, because one computer screen was in front of the mixing position, but another was mounted on a shelf fixed to the desk quite a long way off to the right.

file:///F|/SoS/SoS%2001-2006/studiosos.htm (2 of 5)12/19/2005 10:22:37 AM

Here you can see how the main workstation desk was improved (left to right). Initially the mixer was on a higher shelf, which made it more difficult to use, interfered with the sound from the monitors, and meant that the two computer screens had to be separated. The first step towards a better layout was to lift the mixer from the desk so that the top shelf could be removed. The redundant shelf supports were then sawn down, allowing the mixer to be reinstated on the main desk surface. Finally, the monitors were replaced, but vertically to provide a wider monitoring sweet spot, and the two computer screens were placed side by side on a stand behind the mixer.

Studio SOS

The existing foam treatment was put up asymmetrically and it wasn't really doing the monitoring environment any favours, The amplifier packs on the so we band's Dynaudio BM6A removed monitors had high-frequency it all trimmer controls, but the except for plastic spindle of one had snapped off, and was even a couple beyond Paul's powers to of panels repair. above the console. We then moved the mixer onto a keyboard stand, so that we could extricate the supporting desk and modify it. With the upper shelf of the desk removed, we could drop the lower shelf to a sensible height for the mixer, the only problem being that the mixer wouldn't fit between the two rather substantial metal upright shelf supports. Not to be thwarted, the band hacksawed the right-hand support to size! With the support gone, the mixer could be set up on the desk at a more practical height. A keyboard stand from the studio was then placed directly behind the desk, supporting the redundant upper desk shelf so that the two Apple monitors could sit side by side on it.

Monitoring & Acoustics Overhaul With the speaker stands now a little further forward and angled inwards, we file:///F|/SoS/SoS%2001-2006/studiosos.htm (3 of 5)12/19/2005 10:22:37 AM

Studio SOS

moved the speakers to a vertical position and used a couple of Auralex Mo Pads to angle the speakers towards a spot just behind the head of anyone working in the mixing chair. One of the monitors had a damaged high-frequency trimmer, so we removed the amplifier assembly to see if this was fixable. Unfortunately, the plastic adjustment spindle had been snapped clean off the potentiometer inside the case, so there was nothing we could do other than show Joel how to fit a replacement when he could get one.

Transforming the control room's acoustics required a number of different steps (top to bottom). The band had already installed some thin acoustic foam tiles, but these weren't placed effectively, so they were removed. Thick Auralex acoustic panels were then glued in a horseshoe shape on the walls behind the monitors, as well as to the ceiling, in order to reduce direct early reflections reaching the listening position. The previous thin tiles were rolled up and packed into a corner of the room as improvised bass trapping. With the acoustic treatment done, listening tests allowed the tonal balance of the speaker system to be optimised with the frequency contour controls.

With all the cheapo foam taken down, the room was of course rather reflective, so we used four-inch Auralex foam panels to form a horseshoe shape extending behind the monitors and out to a little way behind the mixing position. (Paul Eastwood of Audio Agency arranged to get this shipped directly to the studio at very short notice in time for our visit, so thanks for that Paul!) A loose Auralex panel was also placed over the controlroom window, supported with string from small cup hooks screwed into the window frame so that it could be removed when sight lines were required during recording. In all, we used five panels to treat the walls, leaving us one spare to put on the ceiling centred just forward of the mixing seat. These measures audibly improved the sound, but did little to resolve the problems in the bass region, so we decided to improvise some bass trapping by rolling all the old foam we had removed into cylinders, secured with tape, filling the area between the stairwell and the side wall to a depth of two or three feet.

Listening Tests Hugh produced his familiar BBC test CD and we used it to balance the levels of the two speakers. We also checked that the undamaged speaker had the same subjective high-frequency level as the damaged one by switching to mono and listening for a stable central image on cymbals and vocal sibilance. Joel had turned down the bass trimmers in an effort to control the room-mode problems, but we now found that these were best set to their flat positions. As expected, the stereo imaging was greatly improved now that the new foam file:///F|/SoS/SoS%2001-2006/studiosos.htm (4 of 5)12/19/2005 10:22:37 AM

Studio SOS

was in place. However, what we hadn't quite expected was quite how dramatic the improvement in bass-end evenness turned out to be, presumably as a result of our improvised bass trap, assisted by the convex shape of the back wall. With most small rooms we've worked on, there's still been a dip in bass at the absolute centre of the room and there have also been hot spots near the walls and corners. Here, though, the bass seemed more or less even everywhere in the room, except very close to the walls and corners, where the boundary effect inevitably increased bass levels. Complex bass parts on the test CD showed up no unduly loud or quiet notes, and the overall tonality and imaging of the monitoring setup was as good as we could have hoped for, with a very wide sweet spot. But, most importantly, all three members of the band were impressed by the improvement in monitoring accuracy and also approved of the more streamlined appearance of the mixing area. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


file:///F|/SoS/SoS%2001-2006/studiosos.htm (5 of 5)12/19/2005 10:22:37 AM

Anyone got time to make music?

In this article:

Is Your PC Secure? Malicious Software Defined Spyware Doctor PC Audio Driver News Microsoft Vista News

Anyone got time to make music? PC Notes Published in SOS January 2006 Print article : Close window

Technique : PC Notes

We'd probably all prefer to keep our music PCs insulated from viruses, spyware, adware, phishing and the general nastiness of the Internet, but the way music software is developing makes this increasingly difficult — so let's be careful out there. Martin Walker

In a perfect world, we would never lay our music PCs open to attacks from viruses, trojans, spyware, adware, phishing, trackware, browser hijackers, keyloggers, diallers, spam, and all the other variations of nastiness that are dreamt up by hackers. And we'd never have to install additional Internet software, firewalls, virus checkers, spyware detectors and so on, when we would prefer to leave our computers as lean and stripped-down as possible for the highest performance.

If your PC is connected to the Internet, Spyware Doctor is currently the most thorough way to keep it free from a wide variety of on-line threats.

However, music software developers are making it more and more difficult for us to maintain this approach, offering easy one-click Internet access to registrations and updates from within their music packages. If you, like me, have adopted a multi-boot setup with a generalpurpose Windows partition or drive that has Internet access, and a second instance of Windows on a separate partition or drive, to keep your music setup safe from such infections, you'll have noticed that you have to jump through more and more hoops to manage this approach. I'm referring to tedious cutting and pasting of URLs and web pages between partitions or drives, or resorting to longer-winded email registration. I suspect many people now regard having Internet access on their music PC as almost inevitable, whatever their personal

file:///F|/SoS/SoS%2001-2006/pcnotes.htm (1 of 6)12/19/2005 10:23:20 AM


preference.

Is Your PC Secure? My general-purpose (Internet-enabled) partition already has a raft of freeware protection software installed to cope with 'malware' (the collective term for all the malicious software listed above). I use Zone Labs' Zone Alarm personal firewall (www.zonealarm.com), while for virus checking I have both AVG Anti-Virus Free Edition (www.grisoft.com) and the slow but sure Clamwin Free Antivirus (www. clamwin.com). I've also been running Spybot — Search and Destroy (www.safernetworking.org) and Lavasoft Ad-Aware SE Personal (www.lavasoft.com) to root out spyware. Many people haven't yet cottoned on to the fact that no virus checker or spyware detector has a 100 percent detection rate, so while yours may declare your PC free of infection, there's still a chance that some deeprooted items might be lurking somewhere that could be picked up by another utility. While software firewalls dislike rival products being installed alongside them, it's perfectly possible to install and run several virus checkers and spyware detectors side by side. An estimated 20,000 PC spyware threats are now claimed to exist in the outside world, and they can change your browser home page, install unwanted custom toolbars, track your surfing habits and sell them to marketing companies, slow your Internet access and even record your every keystroke. These threats aren't dealt with by most virus checkers, so if you haven't yet installed a dedicated utility to root out spyware on your PC you should seriously consider it.

Malicious Software Defined Most musicians are already well aware of the perils of viruses, which are selfreplicating programs that spread by inserting copies of themselves into other executable code or documents. However, there are now lots of other types of 'malware' that can affect your PC, most of which aren't dealt with by a viruschecker utility. Here are the main offenders that can be eradicated by running a good spyware utility: Spyware: Any software that covertly gathers user information through the user's Internet connection without the user's knowledge. Adware: Similar to but less threatening than spyware, adware displays persistent (and mostly unwanted) adverts, often in pop-up windows. Browser Hijackers: These take control of your browser and give it a different home page, as well as changing various settings. They may not be detected by firewall software as they can appear to be part of the browser program itself. Cookies: Generally beneficial, retaining settings for when you next visit a web site. However, 'tracking cookies' and particularly 'malicious cookies' monitor your behaviour across different web sites and provide third parties such as spammers with this data, so that they can target you with their unwanted advertising. file:///F|/SoS/SoS%2001-2006/pcnotes.htm (2 of 6)12/19/2005 10:23:20 AM


Diallers: Software covertly downloaded. Once installed, the dialler can disconnect your Internet connection and then reconnect via a different long-distance or premium-rate phone number. Keyloggers: These send personal and password details that you type into your browser to third parties, or may take covert screenshots to see what software you're running. Trojans: Small programs covertly downloaded to your PC. Once running, they leave a 'back door' open in your system so that someone can remotely monitor and control your PC, erase files, alter your Registry contents or even read your email. Phishing: The name for Internet scams that use email 'bait' to encourage you to divulge personal passwords or financial data. Can also refer to similarly malicious web sites that pose as legitimate businesses in order to obtain credit-card details.

Spyware Doctor Given this escalation in spyware, this month I thought I'd investigate Spyware Doctor from PC Tools (www.pctools.co.uk), which has been getting some excellent publicity in the mainstream PC press for its detection and removal capabilities. Spyware Doctor detects and removes spyware, adware, trojans, diallers, keyloggers and trackware. It also provides optional but comprehensive background monitoring to prevent any attempts to execute unwanted files, plus immunisation against 1800 Active X objects known to be malicious (although Firefox users won't be quite so worried about these as those of you running Internet Explorer, since Firefox doesn't support Active X controls). SD's 'Live Update' function lets you keep abreast of the latest nasties, with new updates posted every few days, and while a musician wouldn't want the 'On Guard' monitor running alongside music software, it's easy to enable it just before you go on-line, to deal in real time with browser hijackers, pop-up ads, malicious cookies, keylogging and so on, and it even prevents you from accidentally accessing known malicious web sites that may be masquerading as legitimate businesses, to avoid 'phishing' fraud. To see whether or not Spyware Doctor was more effective than my existing utilities, I made sure I had the most up-to-date versions of both Ad-Aware and Spybot S&D, ran them both, and then followed up with Spyware Doctor, to see if it could find any infection they had missed. Sure enough, it discovered and eradicated 42 additional problems — a sobering thought when you already think your PC is 'clean'! I've also had On Guard monitoring active whenever I've been on-line for the past couple of weeks, and my PC hasn't contracted a single spyware infection in all that time, which is most reassuring. Spyware Doctor will run happily on Windows 95, 98, ME, 2000 or XP, and costs just $29.95, complete with one year's worth of unlimited live updates and support. For the added peace of mind it's given me I think it's well worth the money, and I intend to look more closely at the other utilities in the PC Tools file:///F|/SoS/SoS%2001-2006/pcnotes.htm (3 of 6)12/19/2005 10:23:20 AM


range in future.

PC Audio Driver News Universal Firewire Drivers: At the October AES (Audio Engineering Society) show, CEntrance were demonstrating their new Universal Firewire audio driver software for Windows XP. This software provides robust, low-latency ASIO support at 16bit or 24-bit, with 44.1kHz, 48kHz or 96kHz sample rates, and is compatible with all major Firewire chip-sets. However, its major claim to fame is the ability to run several interfaces from different manufacturers side by side, all connected to the same host application. While Michael Tippach's freeware ASIO4ALL driver already provides an ASIO overlay for multiple devices that already have suitable WDM drivers, not everyone has managed to get its multi-device support to work successfully, and because it runs in 'User Mode' it relies on Windows to detect and configure the soundcard. The CEntrance Universal Firewire ASIO

The CEntrance drivers are rather drivers could provide the first professional more ambitious in offering multiway to combine the functions of several application as well as multi-device Firewire interfaces from different support, and they completely manufacturers. replace the Windows drivers with their own low-level 'kernel-mode' drivers, allegedly bringing latency down significantly. Due for release in the first quarter of 2006, the CEntrance drivers should attract a lot of interest from musicians who want to add new interfaces to their current setups, while CEntrance are also hoping that interface manufacturers will contact them for customised versions of the drivers. Find out more at www.CEntrance.com. Emu Audio Interface Update: Emu (www.emu.com) have released yet another significant software update for their audio interface range. For many users, the most important improvements added by these latest Version 1.81 drivers are support for 88.2kHz and 176.4kHz sample rates and support in the WDM drivers for multiple playback channels (up to eight) to play back surround files and so on (WDM recording is still restricted to stereo operation). There are also lots more ASIO buffer settings between 20ms and 2ms, so you can fine-tune performance to your PC more easily and, for those who can use it, 64-bit support is also available. The Patchmix DSP mixer now lets you import/export Core and Multi FX presets and offers the same 'Load FX on Startup option' offered by the new 1616 series, to speed up boot time by only loading all the presets when you actually launch the mixer for the first time. If audio interface manufacturers would like me to publicise significant PC driver updates in this column, just let me know about them via an email to [email protected].



Microsoft Vista News Microsoft's forthcoming Vista operating system has now had a beta release so that developers and beta testers can evaluate it. The final product release is still destined for summer 2006, although there have already been so many slippages that few industry experts are confident that this date will be achieved (Microsoft originally hoped to ship this OS at the end of 2004). Despite its flashy new look and underlying power, many industry analysts also feel that users will be slow to get their credit cards out to buy it. The traditional route to getting a new OS onto our PCs has been to pre-install it on new machines, but all the indications are that PC sales growth is dropping: many users simply don't need computers that are any faster than they already have, and businesses are understandably wary of making fundamental changes to their networks unless there's an obvious improvement to be had. Here lies the biggest obstacle: while Windows Vista does look different, the desktop is functionally identical to Windows XP, with the Start button, System Tray and Recycle Bin in exactly the same positions as before. There's no obvious feature that screams 'buy me!', as there was with the multimedia features of Windows 95 or the fresh start and extra stability of the Windows XP experience. And while Microsoft claim that Vista simply needs a 'modern' CPU and 512MB of RAM, it draws a distinction between a 'Vista-capable' PC (one that can run the new OS but not necessarily all its new features) and the 'Vista-ready' PC (with a modern graphics card, to provide the full-on experience). In other words, for the musician who chooses to install a basic graphics card, the Vista experience may appear little different from the current XP one. I suspect that many people faced with the prospect of having to buy a new PC to provide the full Vista experience will decide that what they already run is perfectly adequate. I would also guess that most businesses will wait for at least six months, and many a year or more, before taking the plunge, until the inevitable bugs have been picked out and service packs have been released. Finally, although Apple have insisted that their forthcoming Intel-based OS X will only run on Apple-based hardware, we may yet see OS X made available to other PC users, and I can see some musicians being more interested in this than buying Windows Vista. We'll have to wait and see what happens. Oh, and by the way, Microsoft have confirmed that they will bundle anti-spyware technology into Vista. Better late than never! Published in SOS January 2006






Avoiding The Blue Screen Of Death

In this article:


The Blue Screen Of Death Hints & Tips For Trouble-free PC Musician Published in SOS January 2006 Working Mains Power Supply Print article : Close window Problems Technique : PC Musician The Show Must Go On: Power Issues On Stage Surviving A Power Cut Random PC Reboots Specifying A UPS If you've ever been

confronted by the dreaded Blue Screen Of Death, suffered random reboots or faced the frustration of inexplicable PC crashes, read on for some preventative measures... Martin Walker

Using a PC to record, mix down and master your own music ought to be a streamlined and pleasurable process, and for many musicians it is exactly that. Indeed, some have run their computers for months or even years without a single problem. However, others unfortunately find that many of their initially creative sessions descend into another round of fault-finding frustration. This month we're going to explore some measures to help the PC musician minimise the chances of crashes, reboots and other interruptions, so that they can simply get on with the most important task — making music!

The Blue Screen Of Death Fortunately, one of the big improvements gained when you run Windows XP (and both 2000 and NT) rather than Windows 98 is that each application runs in its own 'protected area' of RAM. Should an application encounter problems and crash, you (or, rather, Windows) can safely shut down that particular application without having to reboot the entire PC and potentially lose unsaved data from

file:///F|/SoS/SoS%2001-2006/pcmusician.htm (1 of 8)12/19/2005 10:23:23 AM


other running applications. In some cases you can then restart the offending application, although it's generally safer to save any open files and reboot anyway, in case any processes started by the application that crashed are still running. Despite the above, the dreaded BSOD (Blue Screen Of Death) can still put in an occasional appearance with Windows XP, 2000 and NT. It signals a nonrecoverable condition, and that you've lost all data that hasn't yet been saved. This blue screen is still one of the most frustrating aspects of Windows use, since there's absolutely nothing you can do about it except reboot your PC and start again. The most common reason for blue screens under Windows 98/ME was incompatible versions of DLL (Dynamic Link Library) files, but with Windows 2000/XP blue screens tend to happen because of driver problems, and when using older versions of applications with the latest Service Pack. The last, once again, underlines the importance of either keeping everything up to date or leaving everything well alone. There's an area in the Windows Startup and Recovery section (Advanced page of the System applet in Control Panel) devoted to what happens if a System Failure occurs. Here you can specify whether or not the event is added to the system log, send an Administrative Alert (not much help to most musicians, who are already the administrator of their PCs), or Automatically Restart. It's helpful to un-tick the last option, since that gives you a chance to read the BSOD. Further options include various 'Write debugging information' memory-dump sizes, the choices being None, Small (64KB), Kernel and Complete (entire physical RAM). I've chosen None, on the grounds that unless you know how to interpret the memory dump it's not much use to you, and there's no point in cluttering up your hard drive with dump files. However, if you're consistently given a BSOD by (for example) a particular audio interface driver, the manufacturer may ask you to send this memory dump so that their experts can try to analyse what's going on. If your PC ever presents a blue screen, don't force the PC to reboot immediately: first, note down any file name that may be mentioned, and/or any error number, and try to remember whether you've recently installed new software, a driver update or new hardware. Even if you don't manage to track down the culprit first time, if the blue screen happens again you may notice a common factor that will help to solve the problem. The quickest way to obtain more information on a specific error message is to visit Microsoft's Knowledgebase (http://support.microsoft.com) and enter the error number that you saw on your blue screen, in the format 'STOP 0x000000D1'. You should then be provided with a long list of possible culprits that, at the very least, will give you more ideas on what might have caused the problem and may point to a specific application that's known to cause it.



Other related web pages that I've found useful include Windows XP Shutdown & Restart Troubleshooting (http://aumha.org/ win5/a/shtdwnxp.htm) and Troubleshooting Windows Stop Messages (http://aumha.org/ win5/kbestop.htm).

Hints & Tips For Trouble-free Working While the main causes of untimely interruptions are discussed in the main text, there are various others that can scupper your chances of making music while the inspiration is fresh. Here are some to bear in mind: Power Management: The phenomenon of audio interfaces spontaneously 'disappearing' from laptops running Windows has been blamed on IRQ sharing but is more likely to be due to power management issues, as I explained in SOS October 2005. For instance, if your USB interface periodically goes AWOL and crashes your laptop, try un-ticking the box labelled 'Allow the computer to turn off this device to save power' in the Power Management page of the USB Root Hub that it's plugged into. Protect Your System: If you must download and install demo versions of unknown software, create a dual-boot system with one Windows partition that connects to the Internet and has all the demos installed on it, and another that only ever houses your trusted music applications. This is what I do, and it's why my music partition has always been so stable. When your music partition is working reliably, make an image file using a utility such as Norton's Ghost. Then if the partition does become less stable in the future, you can restore that image for an immediate return to normality. Mouse Muscle: If you have a wireless mouse, make sure you keep spare batteries standing by, and if it's a rechargeable mouse try to get into the habit of leaving it on charge at the end of each session, so that it's well topped up for the start of the next one. There's nothing more frustrating than your mouse dying in the middle of a session! Even better, keep a standby wired mouse plugged in as well. It won't interfere with the wireless one, and if the wireless one dies you won't have to reboot to carry on.

Mains Power Supply Problems Spontaneous crashes and reboots can be caused by hardware or software problems (see 'Random PC Reboots' box), but some are obviously due to mains power problems. A brownout, for example, is a short-term drop in voltage (your mains light bulbs will dim simultaneously). Depending on its severity, your PC may survive unscathed (I had three or four such 'blips' while writing this feature and my PC didn't grumble at all). However, it depends on how low the voltage dip is, and how long it lasts. Your PC could suffer a 'frozen keyboard', a complete lockup or even a spontaneous reboot. A blackout, as the name suggests, is a complete loss of power (your lights will go out), and the result is exactly the same as switching your PC off at the wall socket. All data not saved will be lost, and as soon as the power comes back on file:///F|/SoS/SoS%2001-2006/pcmusician.htm (3 of 8)12/19/2005 10:23:23 AM


your PC will either stay in its 'off' state or reboot itself. Spikes and surges are sudden momentary increases in voltage, either caused by nearby equipment being Although you can buy a cheap surgeswitched on or (more often) off, or (in protected mains distribution board, products more serious cases) by a nearby such as Furman's PL8 II, shown here lightning strike. They can cause (available in 120 and 240V versions) provide more sophisticated filtering and surge catastrophic hardware damage, often protection. This one is presented in a burning out motherboards and compact rackmounting unit with pull-out connected devices. A strike to lights (very useful on dark stages!). telephone lines can also easily result in a burnt-out modem if it's connected to them, and can damage the rest of your PC at the same time. I heard of one case where the PC itself was physically unplugged from the mains outlet but its modem was still connected to the telephone line. An incoming transient not only took out the modem, but also the motherboard it was connected to, the processor and most of the other devices, plus all the data on the hard drives. Many musicians switch off their PCs as soon as they hear a rumble of thunder or notice the first flash of lightning, then physically unplug it and their modem from the wall sockets. Better still, if, like many musicians, your mains wiring is all connected via distribution boards to a single mains socket, pull these plugs from the wall so that all your gear is protected. However, surges can happen at any time, so although PC power supplies generally have integral mains filtering to deal with incoming RF interference, it's sensible to fit some sort of external surge protection for all your gear, to cope with more vigorous influxes. There are lots of mains distribution boards available that are fitted with surge-protection devices, and even a cheap one is better than nothing. However, if possible, avoid those with 'sacrificial' MOV (Metal Oxide Varistor) components that are partially or completely destroyed in the event of a strike, or at least pay a little more for one with indicators that show when the MOVs have failed. Lots of suppliers stock the respected Belkin range in six and eight-way versions with added protection for telephone and modem, in various configurations, for under £25. These are far better than the anonymous 'under £10' products you tend to find in DIY centres, but for even more protection look for products such as those from SurgeX (www.surgex.com). The next step up from a surge-protected board, to be considered if you suffer from brownouts or regular surges, is a power conditioner, which has the totally different function of filtering out mains noise, as well as stabilising and regulating the output voltage. Stand-alone units are available from companies such as APC (www.apc.com) from about £75. Power conditioners from companies such as Furman (www.furmansound.com) also offer more advanced MOV surge protection that's claimed not to degrade. Models such as their 1U rackmount PL8II E, with 10 IEC outlets, are available from about £170. The more sophisticated AC Line Regulator models can also deliver a constant output voltage of 120/240 volts AC wherever you are in the world.



The Show Must Go On: Power Issues On Stage Lots of musicians now perform live with various sorts of PC, but it has to be said that your computer is probably at its most vulnerable when on stage. I discussed ways of making rackmount PCs more roadworthy in SOS June 2004, and tried to dissuade people from gigging with desktop PCs, which aren't really designed for much bouncing about on long journeys.

Teaming your laptop with a buss-powered audio interface, or one that offers the option of buss-powering, such as MOTU's Traveler, could save lots of hassle in the event of mains problems, as the interface can carry on regardless, drawing power from the laptop's battery.

The majority of musicians now adopt a laptop as the ideal stage companion — laptops will carry on regardless even if the mains power on stage momentarily sags or conks out during your set. Nevertheless, it's also well worth buffering your stage gear from spikes and other nasties caused by nearby stage lighting and the like. A filtered distribution board is a wise investment, and a power conditioner or UPS is even more sensible. Another consideration is your choice of audio interface format. A PCMCIA card will also benefit from your laptop's battery backup if anything goes wrong, as will a USB or Firewire interface that's buss-powered. A mains-powered USB or Firewire interface will 'disappear' even during a momentary mains problem, and if your laptop sequencer application carries on smoothly with battery power it will suddenly be without an interface. This will cause the sequencer to stop and you may have to close it down, re-launch it so that the interface is recognised once more, then reload your song. Even worse, your sequencer may crash or completely lock up your laptop, requiring a complete reboot once the power has come back on. Neither of these scenarios is the recipe for a relaxed stage performance!

Surviving A Power Cut The ultimate protection for any computer system would allow it to carry on regardless even if the mains disappeared altogether. Given the rarity of power cuts in many countries, most project studio owners just swear if all the lights go out and they lose whatever PC data they were working on, while waiting for the lights to come back on again. However, if you live in a part of the world where power blackouts are more commonplace, regularly suffer from spikes and brownouts that crash your PC, or have a commercial studio where it's simply not an option to risk your clients' data, investing in an Uninterruptable Power Supply (UPS) is a wise move. It's also vital for anyone involved in broadcasting, running Internet servers or any other applications where you just have to keep on trucking! Anyone who owns a laptop PC will have seen the principle of the UPS in action. While the laptop is plugged into the mains, its internal battery is trickle-charged. If the power fails, or if you unplug the laptop's mains power supply, the battery file:///F|/SoS/SoS%2001-2006/pcmusician.htm (5 of 8)12/19/2005 10:23:23 AM


instantly takes over to provide a smooth continuation of power for as many hours as the battery's charge allows. Some Intel Centrino models can manage four to five hours when running non-demanding office applications, while 'desktop replacement' laptops may only last for an hour or two when running a sequencer and lots of plug-ins and soft-synths. A bona fide desktop system with multiple drives and expansion devices consumes even more power, making it largely impractical to fit integral batteries for backup purposes. Instead, you'll need a stand-alone UPS, rated such that it can supply enough power to keep your gear going for the required length of time. Some people may be content to protect their PC for just a few minutes (enough to save whatever they're working on), while others may want to keep their entire studio powered up for the remainder of a session. At the very least, computer hard drives should be allowed to power down gracefully, as modern hard-disk controllers tend to cache data that may be lost in the event of a power loss, causing possible file corruption. Failure to boot after a power cut is usually caused by such corruption or a damaged hard drive. UPS devices usually come with software that communicates with your PC via a serial or USB cable and guides it through an automatic controlled shutdown (generally the cheapest way of avoiding damage or data loss). Several computers can be controlled from one UPS by this means, and for more ambitious setups it's even possible to manage a UPS via an Ethernet network. The UPS may instruct your PC to enter its Shutdown state, or go into Hibernation. In the latter state, the entire contents of RAM are saved as one big file on your hard drive, and when power returns the system reloads this file and carries on from where it left off. A Standby state is also a possibility (the PC drops into a low-power mode with the monitor and some other devices powered down, but able to quickly resume when power returns). The musician may need to consider these options more carefully than most users, particularly if a Firewire or USB audio interface is being used (see 'The Show Must Go On' box for more details).

Random PC Reboots Spontaneous reboots can occur as a result of viruses or using elderly driver versions. However, more common reasons are hardware-related. Faulty RAM is one possibility, especially if it wasn't handled carefully during installation. The easiest way to test your memory, to eliminate this possibility, is to run a freeware utility such as Memtest86 (www.memtest86.com) or the two memory benchmarks from Sisoftware's Sandra Burn-In Wizard for at least several hours. Alternatively, temporarily remove or replace the RAM to see if the problem goes away. RAM timing could also be an issue, so avoid overclocking and try reducing the memory speeds in the BIOS. A poor-quality power supply (or one that's running near to 100 percent capacity) is another possibility, as is overheating, particularly of some CPUs, so check internal temperatures in your BIOS or with a suitable Windows utility to make sure. Even file:///F|/SoS/SoS%2001-2006/pcmusician.htm (6 of 8)12/19/2005 10:23:23 AM


bad contacts can be a cause, so rule this out by re-seating your expansion cards, RAM, CPU and so on, and unplug/re-plug all the internal cables, which should help to clean all the connections.

Random reboots may be due to a RAM problem, so running some memory tests like these in Sandra's Burn-in Wizard is a useful check.

Specifying A UPS Although there's not really such a thing as a 'typical' power cut, 90 percent are said to last under five minutes and 99 percent less than one hour, which may make it easier to choose a suitable UPS. Deciding on the VA (Volt-Ampere) rating you will need your UPS to have is simply a matter of deciding which of your gear needs protecting and then totting up the VA ratings that you should find somewhere on the equipment rear panels. Many PCs will be happy with about 500VA, although it's safer to over-specify if you can afford it, as computer and audio gear can draw significantly greater peak currents. It can be a confusing business trying to find the most suitable UPS, as there are several basic types. The cheapest type is the Standby (off-line) device that makes no attempt to regulate the mains supply while it remains within certain predefined limits, and which only switches its inverter (the circuitry that converts DC from a battery into an AC output waveform) on-line after a short break, of typically several milliseconds, in the mains supply, or if the voltage varies significantly from its nominal value. Its normal output waveform is often a square wave. Such UPS supplies are available for loads between about 350VA and 1kVA, and are ideal for keeping your PC going during a power cut if you don't want to spend too much money. The next step up is a line-interactive UPS. This type of device offers some 'conditioning' of the mains supply, to provide a clean, stable output voltage, free of spikes and electrical noise, that is normally either a sine or stepped wave. Again, this type of supply switches to its inverter circuitry only on mains failure, after a few milliseconds. They're available with power ratings from 500VA to to 5kVA and are suitable if you regularly notice your lights dimming, you suffer from incoming spikes, or you want to prevent your PC from crashing because of poor mains quality or power cuts. However, their output waveform may not suit highquality audio gear such as power amps, if you want to use such a UPS to power your whole studio, and you may hear an audible click during the change from file:///F|/SoS/SoS%2001-2006/pcmusician.htm (7 of 8)12/19/2005 10:23:23 AM


mains to battery power. Prices for off-line and line-interactive supplies range from £50 to £300, depending on power rating and battery duration. The ultimate solution (and, therefore, the most expensive, starting at about £350) is an on-line UPS, where the inverter circuitry is permanently in circuit, powered either by the mains or from the battery when mains power isn't present. No audio changeover clicks will be heard and the output waveform is a true sine wave, mostly cleaner and more stable than the incoming mains supply, and An uninterruptable power supply with regulated to much closer tolerances. This power conditioning needn't be expensive. type of UPS is normally the only one of This line-interactive Riello Galatrek Plug Dialog 350 model could supply the the three that offers an automatic bypass average PC for several minutes in the in the event of a fault condition, so your event of a power cut, for under £50. gear should carry on regardless to the end of the session, when you can arrange to get the UPS repaired. It can also be used for 50/60Hz frequency conversion when powering foreign gear. An on-line UPS is the only type of UPS I would recommend for powering studio audio gear as well as your PC. They're available with power ratings of between 700VA and 800kVA. The length of time for which a particular UPS can run when the full rated power is being drawn from it is, of course, dependent on battery capacity and thus varies greatly from model to model, although some have optional extra battery packs that can extend the nominal time. Recommended UPS manufacturers include APC (www.apc.com), Emerson-Liebert (www.emerson-ups.co.uk), MGE (www. mgeups.co.uk) and Riello Galatrek (www.riello-ups.co.uk), all of whose products can be bought from a variety of computer and electrical suppliers. One of the best suppliers I've come across is UPS Systems (www.upssystems.uk.com/ acatalog/index.html), who specialise in supplying standby power and can provide advice, service and support. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



CLASSIC TRACKS: The Staple Singers I'll Take You There

In this article:


Staple Diet The Learning Experience Muscling Down Producer: Al Bell; Engineers: Before The 1176 Published in SOS January 2006 Invisible Mending Print article : Close window Beyond The Memphis Sound Technique : Recording/Mixing Getting Away From It All

Terry Manning, Jerry Masters

For the Staple Singers' landmark 1972 album, Terry Manning and producer Al Bell employed the talents of Memphis's finest musicians and two of the South's most famous studios. Richard Buskin

"I really think it's an art form to use production in a way that ends up pleasing the listener," says Terry Manning, whose credits as a producer and/or engineer include ZZ Top, Led Zeppelin, Isaac Hayes, Albert King, George Thorogood, Joe Cocker, Joe Walsh, Johnny Winter, the Fabulous Thunderbirds, Al Green, Elton John, Shakira, Lenny Kravitz and, most pertinent to this article, the Staple Singers. "Although I hear everything I have ever worked on critically, and there's always the element of looking back and saying 'Why didn't I do that instead?' I can now enjoy some of those things and actually sit back and just let them go." Manning was still a high-school freshman in El Paso, Texas, when he began playing guitar and befriended a local attraction named Bobby Fuller — in 1966, Fuller would enjoy brief fame with 'I Fought The Law' before dying in mysterious circumstances. Manning played several gigs with Fuller, who had his own nightclub, record label and two-track home studio, but then relocated to Memphis, Tennessee, when his minister father was assigned a church there in 1963. Since many of the 15-year-old's favourite recordings, by the likes of Rufus Thomas and the Mar-Keys, hailed from that city, he was more than happy to go there, and no sooner had he arrived than he went to Stax Records and snagged a job copying tapes and sweeping floors.

file:///F|/SoS/SoS%2001-2006/classictracks.htm (1 of 10)12/19/2005 10:23:26 AM


At the same time, Manning also befriended John Fry, another go-getting highschool sophomore, who ran his own Ardent Records label and whose home studio would soon evolve into one of Memphis's top facilities. Not long after Fry cleaned up some demos by a band called Lawson and Four More, of which Terry Manning was a member, the two youngsters began working together alongside future producer Jim Dickinson, and it was the simultaneous employment by Stax and Ardent that provided Manning with his training as an engineer. "Neither studio had a problem with this arrangement," he says. "In the musical sense, Memphis was fairly isolated — it wasn't associated too much with Philadelphia, Chicago, New York or Los Angeles. Its music style was homegrown, even though technically we were trying to try to emulate the big boys in London, New York and Los Angeles. Musically, Stax was doing what it liked, and together with Ardent we were just one big happy family of people wanting to do music."

Staple Diet It was in 1971, after having engineered on albums by Isaac Hayes, Albert King, Led Zeppelin and Billy Eckstine, that Terry Manning first assumed the same role for the Staple Singers, pushing the faders in addition to playing guitar, harmonica, melodica and echo harp on their Staple Swingers album. This record, the first to be produced by Al Bell and tracked in Muscle Shoals, saw the Staples moving into funk, having started off as a traditional gospel quartet. Roebuck 'Pops' Staples and daughter Mavis shared lead vocals, backed up initially by Mavis's sister Cleotha and brother Pervis. After flirting with white folk music during the mid-'60s, the Staples had signed with Stax in 1968 and brought themselves up to date by way of 'message' songs on their first two albums for the label, both produced by Steve Cropper. Then, after Pervis was replaced by sister Yvonne, Al Bell took over the production reins and funked things up, and the result was the Staples' halcyon period. This peaked Terry Manning (right) at the desk in Ardent with their fourth Stax album, Be Studios in 1969, with James Taylor and Peter Asher. Altitude: Respect Yourself, containing the Luther Ingram/Sir Mack Ricecomposed hit 'Respect Yourself', as well as the soul-and-reggae-flavoured number one smash, 'I'll Take You There', written by Bell on Mavis Staples's livingroom floor with some uncredited input from her. "Al Bell was a terrific arranger," asserts Terry Manning. "He was not a musician — he couldn't sit down at a piano and start playing a song and know which chord file:///F|/SoS/SoS%2001-2006/classictracks.htm (2 of 10)12/19/2005 10:23:26 AM


was which. If he did, it was quite rudimentary, but he really had an ear. He knew what was commercial, he knew what he liked, he knew what sounded good, and he was really interested in mixing musical styles. So, when we hooked up, we did many projects together and he came to rely on me quite heavily from a partial production standpoint, and certainly from a musician's standpoint, and totally from an engineering standpoint. And I would rely on him for his incredible intellect and knowledge of musical styles and feels and how things could meld together. Again, he was someone I learned so much from and to whom I'm so appreciative. "Al really liked the fact that, in addition to me living in Memphis and knowing the so-called 'Memphis Sound', I also came from a rock standpoint. I was completely into the 'British Invasion' led by the Beatles and the other great groups of the era — in fact, I was possibly into that type of music even more than the music I was working on at the time. So, in his eyes, I would bring a different viewpoint, a more rock & roll viewpoint, to R&B. He would branch out and look at places like Jamaica for the beginnings of ska and reggae music — they would be on other beats than the two and the four; they might be on the one and the three — and he would try to bring those things into the music. "Al had been on a vacation to Jamaica, and he'd gone to Montego Bay and seen a couple of bands, and he'd also visited the Bahamas and heard different kinds of music in the Caribbean region. When he returned, he had a few records, mostly from Jamaica, and he would play those for me and say 'Listen to the way these musicians are kind of hopping and skipping along instead of just laying down a solid groove.' I'd say 'OK, let's get the guys to try that.' "We were actually trying to consciously meld things together, to do something different and bring a breath of fresh air to the way we were doing R&B, and I think that really culminated on the Be Altitude album where I overdubbed guitars that were heavily distorted and used Moog synthesizers. Those things just weren't being done at the time in R&B. On 'I'll Take You There', Photo: Chris Walter / Photofeatures especially, we were really trying to get The Staple Singers in action in 1973. that jumpy Jamaican, early reggae feel, and it's kinda funny because among my good friends now are those two great Jamaican players and producers, Sly & Robbie. I told them 'Hey, you know, we were trying to copy you guys with "I'll Take You There",' and they said 'Really? We got "I'll Take You There" and we were trying to copy that on some record we were doing!'" Manning, meanwhile, fulfilled more of a co-production role for many of the Be Altitude sessions, leaving the tracking to engineer Jerry Masters at the Muscle Shoals Sound Studio in Alabama where the rhythm tracks and guide vocals were recorded.



"I gave advice on certain things, both technically and musically, and I also played some guitar, but basically I stayed out of the picture while Jerry did his thing," Manning says. "The Muscle Shoals studio was like a factory at that time, and it had incredible players: Roger Hawkins on drums, David Hood on bass, Barry Beckett on keys, and Jimmy Johnson on guitar. Mavis was the only member of the Staples to attend those initial sessions, laying down guide vocals surrounded by windowed baffles that allowed her to see the other musicians. You see, the Muscle Shoals guys had their own thing going really well, so there was no point for anybody coming in there — even if it was an Aretha Franklin or Rolling Stones session — to impose on them and say 'Hey, here's how I want it recorded!' You went there for what they did, and that's why we went there. As great as they were, Al wanted to get away from the Stax players, and that's because we were looking for a different feel and different types of things going on."

The Learning Experience Once Ardent Studios had moved out of John Fry's home into a rental space on National Street, Memphis, it virtually became the 'B' studio for Stax Records and artists such as Isaac Hayes, Booker T & the MGs and the Staple Singers, in addition to others like James Taylor and Ike & Tina Turner. "One reason they recorded at Ardent was because it was by far the most technically advanced studio in the area," Terry Manning explains. "We had the first four-track recorders, the first 16-track recorders, the first 24-track recorders, the first of everything. We would jump ahead of the technology and stay at the level of what the biggest studios were doing in other Photo: Terry Manning cities. What's more, there was an incredible musician base in Memphis: Isaac Hayes, Booker T, Steve Cropper, Teenie Hodges, the Memphis Horns. People wanted that sound, and when they also wanted the best technology we got those gigs. Well, that gave me quite a grounding early on when it came to working with all styles of music, while also dealing with plenty of local jingle work taught me to do it fast. "At the age of 16 and 17 years old, I was thrust into sessions with 60 to 80 musicians: strings, a horn section, the great drummer Ronnie Tutt, percussionists, two guitars, bass, three keyboards, marimbas, you name it. It was a full studio, and they were doing 10- to 30-second songs, so there wasn't much time for me to mess around getting a sound. Fortunately, having done quite a bit of studying and learning, John Fry was quite the technician and the teacher, teaching me and others what propagated an audio wave, what it looked like, what it sounded like, where it went, what type of microphone captured that wave, where that microphone should be placed for optimum use, what components made up the console... this was all thrust on me very early on. 'You've got to learn this now. You've got 80 musicians, they're ready to play, they're being paid by the hour and



we're gonna do 20 songs in this three-hour period, so get it down.' "It was a training that has served me well through all the years that I've done this, and I always like to give the real credit to the musicians and the singers, because if you don't have something good to record, there's no point in recording. We were so lucky at that time to have absolutely incredible talent around us, whether it was Otis Redding, Isaac Hayes, Booker T & the MGs or the Staple Singers. I mean, three of the first drummers I ever worked with were Al Jackson, Jr, the Stax session player; John Bonham; and Ronnie Tutt, who played for Elvis Presley. To me, getting drum sounds was easy... Little did I know that years and years and years would follow with rarely having that calibre of drummer available!"

Muscling Down Having founded Muscle Shoals Sound Studios in 1969, and with the help of legendary producer/A&R man Jerry Wexler, the aforementioned white quartet of Hawkins, Hood, Becket and Johnson moved the studio into a former casket warehouse they called the Burlap Palace. And it was there, inside a basic control room with a 16-input Flickinger console and MCI JH16 tape machine, that Manning looked through the glass towards a wood-floored live area above a basement where the caskets used to be stored. "The fact that it wasn't solid provided the room with a little bit of a thump and a ring," he says. "Roger Hawkins's drums were inside a booth that they had just built at that time, and it was very tight in there, so we were really into closemiking drums and there were no room mics at all. The overheads back then would have been Neumann U87s, placed on either side of the cymbals; on the snare and toms there would have been dynamic mics, either Electrovoice RE15s or Shure 545Ss — the 545S looked like an early 57. An RE20 was on the bass drum, and there was a Neumann KM84 on the hi-hat. "Barry Beckett's electric piano was a Wurlitzer, and there was an RE15 actually miking the speaker, while David Hood's bass went through a passive direct box. Nothing fancy at all. Very rarely did we use an amp to record the bass. David's bass solo on 'I'll Take You There' was recorded live at Muscle Shoals; a great, great part. Eddie Hinton played guitar on the song, and he had a small Fender amp, miked with an 87 positioned directly on Photo: Terry Manning the 12-inch speaker, and there was The control room at Ardent Studios, based also the guitarist Raymond Banks who around a Spectrasonics console and 3M tape machine. came down from Stax — we'd joke because Raymond would play almost nothing. He'd be on the session playing just four or five notes, and we'd look over and say 'The guy's not playing.' Then we'd listen back to the song and go 'Wow,



those were the notes to play.' He was the minimalist. It was all about feel. "Mavis, meanwhile, was also miked with an 87 — I tell you, we used those for everything — and she was just the incredible diplomat and party person. What a personality. She'd pal around with everybody and have so much fun, putting everyone at ease, that everyone wanted to play for her. In fact, on 'I'll Take You There' you can hear her talking to the musicians. In the middle part, she's actually calling their names out when they play different things — that's live from her guide vocal on the tracking sessions. She was so into it and had such a rapport with the musicians, and that's one of the things that made those sessions so great. There was a tremendous feel because everyone was really having fun." All of the rhythm tracks for the album were recorded at Muscle Shoals within four to five days, and consisted of complete takes without edits. Indeed, no more than five takes were required for each song, with the results successfully realising Al Bell and Terry Manning's aim of creating a sound that was, to their minds, distinct from that of Stax. "The Stax sound, as we perceived it back then, was pretty much straight ahead R&B," Manning explains. "It probably featured the greatest players to ever perform that style of music — Booker T & the MGs playing the rhythm, the Memphis Horns playing the horn parts — and it was wonderful and instantly recognisable. Well, Al and I maybe fooled ourselves a bit when we thought that the sound we created at Muscle Shoals was different — listening now, it's actually far more similar than it is different — but we wanted to go for more of an open sound, more of a rock sound, more of other ethnic musical sounds mixed in with the traditional 'Memphis Sound'. That's why we used the Muscle Shoals musicians, who were playing on records by Bob Seger and the Rolling Stones."

Before The 1176 Once work had been completed at Muscle Shoals, the project reverted to Ardent, where Al Bell turned Terry Manning loose to add his contributions on guitar and keyboards. "Obviously, although none of my parts matched the virtuosity of people I respected and admired like Jimmy Page and Eric Clapton and Jeff Beck, I tried to capture their essence by using overdriven and distorted guitar effects, blending them with horns and generally trying to use a lot of production techniques in a more sophisticated way than just straight-ahead R&B." At Ardent, the control room housed a 16-input Spectrasonics console, a 3M M56 16-track tape machine and JBL monitors. "The Spectrasonics board had a really good sound," Manning says. "It was very direct and the sound of the mic pres and the signal path was really, really clean and excellent. The tape machine was the second 3M 16 ever made — it was brought in for the mix of the Led Zeppelin III album — and we used it for many years. It was awesome.



"The effects we had back then were quite simple, but the thing that I really liked and used on almost everything was a Universal Audio 176 limiter. This was the valve precursor to the 1176, and I still have one today and have created my own brand of tube limiters somewhat based on that. To me, Bill Putnam's 176 was one of the defining sounds of music — a great, great piece of equipment — so we used that and we also had outboard equalisers. We had some Pultecs, we had some Langevins, and I'd usually patch in one of those when we went to EQ as the Spectrasonics in-board EQ was good and usable but not terribly powerful. "In terms of delays, I would use the tape machine Photo: William Eggleston and either delay something once with that, use Terry Manning, in a photo varispeed if I wanted to change the delay time, or taken on a miniature Minox 'spy camera' around 1972. bring it back into the mix for feedback, while for reverb I used an EMT 140 plate for virtually everything. Stax had echo chambers that were quite interesting-sounding — good for some things, not for others — but the EMT 140 was pristine, and I used that a lot."

Invisible Mending Once the sessions for the album had moved to Ardent, Mavis Staples laid down her lead vocal parts, Manning compiling a performance from three or four takes of each song. "People look back on those times and say 'Oh, it's classic, things weren't manipulated,' but I will admit I was certainly comping the vocals," Manning states. "The reason I didn't comp the rhythm track was because, not only were the musicians so good, but cutting two-inch tape also wasn't the most fun thing to do and you could easily hear a splice if you weren't very careful. Additionally, after you'd edited the multitrack, everything else you recorded was going over that splice. On the other hand, once we had a good, solid master take of the rhythm track, comping the vocals didn't require splicing with a razor blade. It was just a case of choosing the part and punching it in, dubbing it down. I didn't get microscopic and work on every syllable like people do today in the digital world, but if there was a better line or half-line from a different take I would certainly use that. "I recorded Mavis with an 87 going through the UA 176, standing out in the middle of the floor with some tall baffles behind her to minimise the big room a little bit. Since it wasn't a terribly live room, this worked quite well, and I actually interspersed the overdubbed vocals with bits of the guide track. Mavis was so spontaneous and she did so many things right off the cuff, it was hard to discard file:///F|/SoS/SoS%2001-2006/classictracks.htm (7 of 10)12/19/2005 10:23:26 AM


them, so some of that vocal on 'I'll Take You There' — as well as on the other songs — was taken from the guide track. "Once I had a comp of Mavis, she and the rest of the Staples recorded the response vocals, and we also doubled the harmonies to thicken the backing track underneath everything. For those harmonies, they would be grouped around one 87 — if it was a small enough group I would have it in cardioid and have them out in front, whereas on other occasions I would do a figure-eight and have them facing each other. They were so talented, I couldn't mess up. It was just a matter of getting a good level and not over-limiting."

Beyond The Memphis Sound Terry Manning did not only engineer albums such as Be Altitude: he would also play numerous instrumental parts, most of them overdubbed onto the basic tracking sessions after the fact. "Al [Bell] would give me the general concept," Manning recalls. "He would say 'I really want something layered. I want something more. I don't want just three pieces. I want little things that come in and out. I want texture.' That's what he wanted on all of these songs; texture. He thought he was creating an R&B classic, and he was, so he was looking for textural things, and it was with these guidelines in mind that I'd experiment with different sounds. I was quite lucky to have a studio full of great equipment and great instruments, and to be left alone to use them. "I had all sorts of instruments at my disposal — I had Mellotrons and vibraphones and marimbas and every kind of guitar and every kind of amp. Al would say 'Look, take some of these things and see what you get,' and he would just leave. I'd stay there all night, all by myself, locked in the studio, and I'd just experiment, trying things and arranging things, adding strings to a song with the Mellotron or creating synth sounds with a Moog IIIC. I'd come up with all sorts of ideas, and most of the time Al would come in the next day, listen to them and say 'Yeah, I like that. Use it.' Occasionally, he'd say 'Nah, that doesn't fit,' but he'd approve of about 80 percent of things and say 'Yeah, let's go with that.' "So, I was just working, working, working on overdubs for any one song, and on 'I'll Take You There' we added four, five, maybe six instruments, often just mixed in for tiny little pieces that would pop in and pop back out. For example, I would sometimes use a very, very high, tinkly little Moog synthesizer sound, injecting it into certain places to try to mentally capture people. If young kids heard that, they might be attracted to it and get drawn into the rest of the song. And that's what we were trying to do: psychologically manipulate the listener with production techniques. That wasn't the overall thrust and concept at other R&B places like Stax back then, who were trying to capture the listener by way of incredible virtuosity and a strict groove and very simple production. "A song like 'I'll Take You There' has become so ingrained in the consciousness of those who have heard it, it's hard to now say which, if any, sounds shouldn't be there. Of course, some of the synthesizer sounds got so over-used later on that they started to sound somewhat trite and were passed over for other things, but at the time of the Be Altitude album they were brand-new and no one had done things this way before."



"On 'I'll Take You There', after the vocals had been recorded, I remember playing quite a bit of guitar, some of it distorted, and adding my simulated version of lead towards the end of the song with a Fender Telecaster on which I had put a humbucking pickup. I miked that with a Shure SM57 and used an early Gibson Fuzz-Tone box, playing in the control room with a long speaker lead running out to a Fender Bassman amp... That's now a well known guitar because I later sold it to a guy named Michael Toles of the Bar-Kays and he used it on 'Shaft'. "I did a few little instrumental things, experimental things, and saved some of those for the very end, and then we brought in the Memphis Horns: specifically, Wayne Jackson and Andrew Love, the two guys who have always been in the group and copyrighted the name, as well as several other players. On that session we had two trumpets, two saxes, a baritone sax and a trombone as well. They were recorded in the main room at Ardent, again without much baffling, the trumpets miked together with a KM84, the saxes with an 87, the baritone with an 87 and the trombone with a KM86. "The Memphis Horns make up most of their own parts on the spur of the moment. That's their MO. They come in, they listen to a song, they quickly get a part in their heads — something that jumps at them, because they're so used to doing it — and they then build around that. They were doing that in the early days, too, unless they were working with Otis Redding, who would hum them the horn parts. However, on 'Respect Yourself' I did have a horn line in my mind, so I demoed the part with the Moog IIIC near the end, and I also played it on guitar in a spot or two, and then when the Horns came in I had them copy that line at the end. Later on, however, Al Bell said 'Look, I'm so used to hearing the synth, can't we keep it, too?' So, I had the synth starting the line and then the horns pick it up before I also pick it up on guitar later." As for 'I'll Take You There', a number largely built around just two chords, there were a few other additions on the part of Terry Manning: a harmonica and several guitar parts that he recorded late at night and then submitted to Al Bell the following day. "Doing things like that took time and took a lot of thinking, but I just loved the whole mix, with the great playing by the rhythm guys, the horns that were so cool, and all of the different types of sound blending together."

Getting Away From It All Terry Manning mixed at Ardent using a Scully 280 quarter-inch machine, at a rate of about two songs a day. "A lot was left to the mix," he says. "I rarely mix as I go along. Every time I go to a mix, I turn my brain around in a completely different way and build the mix piece by piece, usually starting with the bass and drums, and then bringing things in gradually before I start soloing things." In total, the Be Altitude album took about six weeks to record, and it was a joyous experience for Terry Manning, who remained at Stax until its closure in 1975 and then spent the next 17 years dividing his time between London and Memphis, since when he has run Compass Point Studios in the Bahamas with his wife Sherrie. "I cannot stress enough how much fun it was to work with the Staples," he declares. "They just loved life, they had a great time, there was a lot of laughter file:///F|/SoS/SoS%2001-2006/classictracks.htm (9 of 10)12/19/2005 10:23:26 AM


and excitement during so many sessions, and while we knew it was a business and we were in there for a purpose, at that time we didn't think so much about budgets or about schedules. We just tried to do things as best as we could, the Staples came into the studio to enjoy themselves, and that made things so easy." Photo: Patrick Cromwell Bell and Manning's attempts to get away from the 'Memphis Sound' initially Terry Manning now runs the renowned Compass Point Studios in the Bahamas. met with mixed responses. When cowriter Sir Mack Rice (who also wrote 'Mustang Sally') first heard 'Respect Yourself', he said "You've ruined my song!" — but he revised this attitude after the recording hit the Top 20.

"'Respect Yourself' prompted some funny reactions," recalls Terry Manning. "When 'Pops' Staples first heard the mix, he said, 'Oh, it's horrible. My voice is so low and the band is so high. All I'm hearing is music.' It was different to how people expected it to be. We've now grown used to it, and it does sound right... at least, I hope it does." Published in SOS January 2006 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



Making A Living From Music For Picture

In this article:

Reeling Around Inspire Me! Pretty In Pink What's Going On? Structure, Mood & Variety How Much Theory Do You Need? Knowing Me, Knowing You Next Month Learn From The Masters

Making A Living From Music For Picture Part 2 Published in SOS January 2006 Print article : Close window

Technique : Composing/Arranging

If you're ever going to make it in this game, you need a calling card, a way of impressing potential clients with your musical ability. You need a showreel! We explain what to do to create one... Hilgrove Kenrick

If I failed to scare you off last month — and if you're reading this, that must be true — you've obviously accepted the idea that a career in music for picture is no picnic, but want to know more about how to get one going nevertheless. The first thing you'll need is a showreel, and it's these that we'll be considering this month, along with how best to use one to capture the attention of the production executives that commission music-for-picture work. Next month, we'll be looking at the kind of equipment you need to create your music. The month after that, we'll look more closely at the process of pitching for work, the people you need to target and what they all do, and finally, we'll demystify some of the jargon associated with this business — what exactly are 'underscore', 'beds' and 'stings'? After that, we'll start to consider how you go about conducting yourself once you've actually got a commission.

Reeling Around First things first — what is a showreel, and why is it important? Simply put it's a display of what you are musically capable of, a noisy CV/business card, if you like. Much like a CV, it will change over time, sometimes several times a week, and at other times you may not need to fiddle with it for a few months (see below for a few of my attempts to get it right over the years).

file:///F|/SoS/SoS%2001-2006/musicforpicture.htm (1 of 9)12/19/2005 10:23:29 AM


The ever-changing face of the author's showreel (from left): in 1998, as a fresh-faced hopeful with, clearly, no idea of what would make a good cover; in 2000, with at least some idea of what would fit on a CD label, but still no idea about font selection; in 2003, having discovered the Font menu and gone rather overboard with the knowledge; and the current, older, wiser, stripped-back model, which is designed for printing on a business-card-shaped CD blank.

The first question is how to deliver this information about yourself, and the answer is: on a CD. Not so long ago, the wisdom was that your showreel had to be on cassette, so that interested executives could listen to it in the car. The same principle still applies — but you try finding an executive who still has a cassette player in their car! So it has to be a CD, and it has to contain a little bit of everything you're capable of. 'Little' is the operative word here. You need to keep this disc really short; five or six minutes are enough. The people you're hoping will listen to this have neither the time nor the inclination to plough through lengthy symphonies. What's more, you'll need to include several different tracks on there, so that the disc showcases a variety of styles. Whilst this requirement may at first appear restrictive, it's actually a godsend. No longer are you plagued by worries of how to start or finish a track; you simply pick your best ones, extract the very best bits from them, and crossfade them together (not fading to silence between tracks will help to keep the disc short, too). You should still leave track IDs in there, though — that way, if you're demonstrating something using your own reel, you can easily flick back and forth. And, if you do get lucky and the showreel elicits a return call, the interested parties can say they liked Track 8, for example.

Inspire Me! One of the best sources of musical inspiration (or desperation) is the Internet. Let's face it, you're not the only composer out there, so go hunting and find some others. Like you, they're trying to attract producers, so you can use them and their web sites for research. Does their work sound better than yours, and if so, why? Is it their production values, or the musical quality of what they have composed? Go for it — tear their work apart, and remind yourself why you're better than them. Then try doing the same with your own work — and be just as harsh. Perhaps someone has a extra style or two under their belt. Have you put some Tongan Neck Blues on your showreel? If not (and if you can find out what it is), get to it! If you think you can do better, then do it, and make sure it's on the reel.



If you're brutally honest with yourself, you'll find composers who utterly out-class you. When that happens, don't be despondent — work out why, and what you can do to catch up. You should never stop learning new musical tricks and skills, and you won't suddenly be able to out-do Poulenc or The Prodigy — these things take time. Learn from them, and then one day it might be you. Don't forget that you can also obtain feedback by posting compositions to the multitudes of forums on the Internet. Of course, everyone has a different opinion, and you should brace yourself for an assault of contradictory comments, but the feedback can be useful nonetheless, if only to gauge general opinion on your direction. Choose your forums carefully depending on the input you want; some will only cover the technical side, while others will have little to say about the audio quality and more about the composition. A blend of both types might be to your advantage.

Pretty In Pink Spare a thought for how the CD will be presented. Is it something you're only ever going to send, or might you hand it around in person? Either way, you don't want a genericlooking branded CD-R with hastily scribbled details across the top — your disc should stand out! If you can get access to a CD printer and printable blanks, that's great. Failing that, there are hundreds of packages available for creating CD labels which can be stuck over the top of a generic CD-R. Either way, you can put your crucial contact details on the CD itself, so that even if your business card or covering letter are mislaid, execs will know how to get hold of you if they still have the CD. Your mobile number and email will do; you can squeeze in more if you want, but try to keep it clean. In the same vein, don't fuss over fancy graphics and pretty colours — no one is taking you on for your art skills! On the other hand, anything you can do to make your disc stand out (a nice case, or a decent, simply designed inlay with perhaps a few credits) is a good idea. And as showreels are just a few minutes long, you're not restricted to full-sized CD blanks — you can now get half-size, multi-coloured, unusually shaped, and even (my personal favourite) business-card-sized blank CDs. However, there's a drawback to these — they're unusable in slot-loading CD players, like, say, the ones in most cars! Will this make your disc the one that's left behind in the office, while your rivals' efforts get the full in-car treatment?

What's Going On? So, you know what to make it look like, and how long to make the contents — but what do you put on it? Firstly, if it isn't up to scratch, forget it. Initial impressions count, and you need to be ruthlessly self-critical. Remember, most of the executives you're targeting will be getting piles of these discs across their desk every week. What they're listening out for is the sound of someone who can deliver, and that goes for technical and technological ability as well as the musical goods. Unlike a band demo, it's important that your showreel sounds at least like a competent recording. It's no good hoping that your prospective file:///F|/SoS/SoS%2001-2006/musicforpicture.htm (3 of 9)12/19/2005 10:23:29 AM


clients will be prepared to ignore the quality of a poorly recorded Portastudio hiss-fest and appreciate the musical genius beneath — it's far more likely that they simply won't bother to listen to it. But how good does your kit have to be to produce something convincing? This is one of those debates that keeps the SOS forum going late into the night: does better kit make for better music? We'll look at this issue again next month, but as regards showreels, well, it can, but it's not essential. Yes, recording at 24/192 on some monster console with a shed-load of vintage outboard can make for a greatsounding recording. But conversely, it's perfectly possible to create a fabulous showreel with a few free software synths and plug-ins, plus a two-quid MIDI keyboard from a car-boot sale. A compositional genius with one black box will get better results than an SSL-owning creative dunce. The crucial difference, as in any artistic field, is made by talent and skill. In this area of endeavour, talent is being able to compose. You either can or you can't. Skill, however, is being able to use whatever kit you have at your disposal, and use it well. Having said that, when you're starting out, if you can improve your chances by borrowing a better synth to use for your showreel, or make it on a friend's PC or digital multitracker in favour of your ailing cassette-based four-track, so much the better. Don't get carried away, but do your best to get the most out of what you have access to.

Photo: Mr Bonzai

Photo: Richard Ecclestone You don't have to have a recording studio with a grand piano, five Yamaha 02Rs, and clocks on the wall showing different time zones like film composer Harry GregsonWilliams (whose amazing setup is shown above) to make music — as chart topper Mylo's battered East London studio (left) shows beautifully. Highly successful music comes from both of these rooms — it's a question of finding out what works for you.

Unfortunately, when you're starting out, you have no idea what the people you're pitching to are looking for — musical ability, technological brilliance, both, or even neither (they might not care what it sounds like, so long as you're cheap to hire). Many of the people you'll end up working for will know very little about music — but not all of them! On the other hand, most of them will be able to tell when something sounds badly recorded. The only answer is to polish everything — practice until your fingers and ears bleed, and make sure you know your recording kit inside out. That way, you can create the best recordings of the best work you've done — and that's what goes on your showreel.



Structure, Mood & Variety As hinted earlier, your showreel should contain a variety of styles. You should also be acquainted with the various types of score that will be used with moving pictures, such as stings, phrases, underscore, themes, and beds, and know which of these to include on your reel, and where. We'll say more about these later in this series, but if this is gibberish to you, just try watching TV or films for a few days. Whether it's a drama, documentary or a game show, listen out for the music. Look for repeated themes, altering styles, when emotions are evoked and where they've used underscore. The latter is musical wallpaper, designed for sections where music is needed, but where it literally has to sit under what's going on and not get in the way. No one is going to listen to a minute of underscore, let alone five, so by all means practice it in your own time, but for sake don't put it on your showreel — it has no place there. What you're aiming for is a collection of emotive moments and styles (fast and slow pieces containing crescendos and diminuendos), and transitions (major to minor, fast to slow, and so on). I'm not going to tell you to have three orchestral, three electronic and three ethnic tracks, or whatever, as that's up to you and what you consider to be your style. But in the early days, you'll need to prove clients that you can do whatever they ask, so try to cover as much ground as possible. By all means have more wind-quintet pieces than happy hardcore if that's your home turf, but remember that if the recipient is looking for drum and bass and you give them brass band, the showreel will go in the bin, and that will be the end of you. You could hedge your bets, and blend styles. This is not as daft as it might sound; in the current climate, musical crossover is becoming increasingly popular. We've moved away from the stark electronic scores of the '70s and '80s, and now the orchestra is back in a big way — but with electronic rhythms and pulsing synth atmospheres. It's never been so exciting, but to do it well, you need to be on intimate terms with two or more genres. When you're happy with your selected styles, and everything's crossfaded and ready to burn to disc, my advice is that you compress the living daylights out of it. I know this isn't the done thing for an SOS writer to suggest, and rightly so! However, you need your showreel to jump out of the CD player and grab the listener by the throat, and nothing aids that in quite the same way as sheer volume.



How Much Theory Do You Need? As Tom Stoppard would have it, "Imagination without skill gives us modern art." I'm not suggesting you create a performance art piece comprising a synthesizer engraved with the names of all the people you snogged behind the bike sheds — only that some musical grounding is better than none. The old guard will tell you that unless you stick to The Rules of composition, and studiously avoid consecutive fifths, for example, you'll never be a 'proper' composer. What utter rubbish. If you want to silence such doubters, you can try pointing out that Mozart or Beethoven were geniuses because they broke The Rules — or at least, The Rules as they were back then. Who says there are any rules, anyway? To impose rules is to stifle creativity or worse, emotion. One failed composer I know, very highly trained, can write you a perfect score in almost any style. The trouble is, all the 'T's are crossed, the 'I's are dotted, and all the emotion has been stripped clean out, leaving a soulless noise. Pull apart an Oscar winner or two. Hans Zimmer uses consecutive fifths and octaves all over the shop, John Williams continually reprises Johann Strauss when he gets excited, and on the stage, Andrew Lloyd-Webber has a nasty habit of sounding disturbingly like Puccini reincarnated. The thing is, we can sneer at them all we want, but they've each been hugely successful in their chosen fields of work. There are plenty of cyclical arguments about musical training, but clearly some is better than none. You're going to be called upon to write in so many disparate styles it'll make your head spin. Sure, once you're further up the ladder, you'll be hired for your own unique sound, but until that day, you need to be capable of coming up with whatever is musically required of you, be it Westlife or a waltz. A basic understanding of harmony, rhythm, meter and scales will get you a long way in deciphering an unfamiliar style. Practice and determination will handle the rest. And what if you don't have that basic understanding? Well, the sole purpose of music in TV and film is to amplify emotion, so if it lacks soul, it'll detract from what's on screen. Thus even though you may not have your scales and modes the right way up, if you can move people, you are 90 percent of the way there. Don't forget, the director or producer is likely to be much less of a musical pedant than you, so they won't complain about your use of harmony, but they will if you are wrecking their scene. On one hand, it's easier to break rules if you don't know they exist, but on the other, knowing the ground rules gives you a place to push off from. As with most things in life, it's a question of balance.

Knowing Me, Knowing You So, there's a stack of perfected showreels on your lap — now who should you send them to? It's here that anyone who knows anyone is going to be of use to you — call in every favour. Convincing executives to hire an unknown is no mean feat, and nothing beats a personal introduction. Some of my best leads have come from an ex-girlfriend's new beau; difficult, yes, but you get used to putting other concerns aside and single-mindedly pursuing your goal. You're looking to stay ahead of everyone else — remember, if you've thought of targeting a particular person, you can bet others have too. Anything you can do to press your showreel straight into the hand of the intended recipient is worthwhile — it matters not whether it's you somehow getting into their office, or asking someone else to do it on your behalf. Whichever way you do it, you're ahead if you can ensure you're at the top of the pile to listen to, and not lost in the heap. file:///F|/SoS/SoS%2001-2006/musicforpicture.htm (6 of 9)12/19/2005 10:23:29 AM


Obviously, it doesn't stop there. You don't just send out a thousand CDs and then sit back waiting for the phone to ring. You need to chase, cajole and charm your way past countless receptionists to try to get hold of the executives you sent them to. For me, this is the hardest bit of all. It's the composer's equivalent of being a cold-calling insurance salesman: no-one wants to talk to you, and they will produce every excuse in the book to avoid doing so. Also, these executives are genuinely busy people, and will be fixated on whatever project they currently have on the go. As so often, it's a fine balance between annoying them enough to persuade them to speak to you, and irritating them so much that they lob your showreel into the deepest recesses of the incinerator... If you can't get through, and they don't return your calls, the doubts I mentioned last month will start to nag. Did they get the CD, are they on holiday, are they after disco and not Dvorak this week, or are you just utterly useless and the best man got the job? All you can do to combat this is make that showreel the best you possibly can. Inevitably, your opinion of what's best will vary over time, so don't be afraid to revise the contents frequently, pulling sections out and putting new ones in. Keep it fresh, keep it exciting, and above all, keep going!

Next Month As promised, next month we'll look at equipment — what you need, what you don't, and streamlining your setup for maximum effect and minimum working effort.

Learn From The Masters For some affordable inspiration, simply raid your DVD collection, or start one! Along with endless 'featurettes' and celebrity back-slapping, an increasing number of DVDs have interviews and even commentaries with the score composers. These can range from twominute superficial affairs to revealing exposés of how they put these scores together and why — and the latter type provide an excellent means of picking the brains of those at the top of the tree. Here are a few of them, and what the discs included that I found useful. GLADIATOR: HANS ZIMMER How Hans Zimmer and the director approached scoring, chose instruments and found the right themes. LORD OF THE RINGS, PART I: HOWARD SHORE



You can't study with established composers, but you can watch them talk about their experiences on DVD for the price of buying these films. Howard Shore spills the beans on what it was like to Photo: Mr Bonzai compose for Lord Of The Rings in the Special Edition box set of The Fellowship Of The Ring (left), Alan Silvestri does so in Van Helsing (below left) and Hans Zimmer (below) talks in detail about creating the score for Black Hawk Down in the box set edition of that film.

The Extended Edition DVD boxed set of this film has a long documentary about the whole scoring process, from the first notes through to recording and mastering. THE MATRIX: DON DAVIS A feature-length commentary on the score. A VIEW TO A KILL: JOHN BARRY A long documentary about music in the 007 franchise. VAN HELSING: ALAN SILVESTRI A short look at scoring for action/horror films. RED DWARF VI: HOWARD GOODALL A great, honest interview about scoring for TV comedy. BLACK HAWK DOWN: HANS ZIMMER An excellent long documentary about this cross-genre score. A recent release of note is John Williams' score to Revenge Of The Sith — there's a special edition with a DVD which features video montages from all six films, set to excerpts from the scores. It's a great opportunity to listen to the score along with relevant scenes, and hear the London Symphony Orchestra in full flow under the baton of a (Jedi) master! Published in SOS January 2006






Making The Most Of Digital Performer Plug-ins

In this article:

Side-chains Other Side-chain Plug-ins Third-party Plug-ins Ring Modulator MIDI Control Digital Performer News 'Playing' Plug-ins Trim

Making The Most Of Digital Performer Plug-ins Digital Performer Notes & Techniques Published in SOS January 2006 Print article : Close window

Technique : Digital Performer Notes

Can you honestly say you use all the features of DP's plug-ins? If not, prepare to be intrigued, as we dig deeper into Dynamics, Multimode Filter, Sonic Modulator and more, in search of the facilities you didn't know were there. Robin Bigwood

In Digital Performer, there's more to some plug-ins than meets the eye. Just when you thought you knew what you could expect from a given plug-in, you discover that it has side-chain audio connections, can have its parameters controlled by MIDI, or has other unexpected features that open up all sorts of sonic possibilities. This month, we blow the lid offf the hidden lives of plug-ins both mainstream and a little more specialist.

A variety of plug-ins in Digital Performer take on new or enhanced roles when you investigate their audio and MIDI side-chain capabilities, while others are simply more interesting and useful than you might first suspect.

Side-chains The vast majority of audio plug-ins take an audio signal, process it in some way and spew out the resulting modified signal. But a few — notably dynamics processors such as compressors and gates — can accept an additional audio signal via a 'side-chain' connection, to influence their operation in some way. DP has its share of side-chain-equipped plug-ins, all of which can do some interesting things when their side-chains are pressed into service. Perhaps the most conventional manifestation of side-chain routing is in the file:///F|/SoS/SoS%2001-2006/performernotes.htm (1 of 8)12/19/2005 10:23:33 AM


Dynamics plug-in. Using this, it's possible to set up compressor pumping and ducking effects and the sort of rhythmic keyed-gate effects that you often hear applied to vocals or synth pad sounds. Dynamics' side-chain is implemented in a very straightforward way and is easy to use. To demonstrate this, let's consider how you'd set up a so-called 'keyed' gate effect. The idea is to place the Dynamics plug-in on a track that is playing back a sustained sound (such as a pad or string section), and then control the opening and closing of its gate with a signal from elsewhere in DP — perhaps a hi-hat track or even a live input — routed to the Dynamics side-chain input. The result is an unnatural stuttering of the sustained sound as the gate opens and closes, which can really spice up an otherwise boring pad part. The hi-hat (or other signal) that is 'keying' the gate is not routed into the audible signal chain; it's just a control signal, which is exactly how MOTU refer to it. 1. The first step is to place the Dynamics plug-in on a track that contains some sustained (or at least fairly continuous) audio and switch Dynamics into Gate mode, using the buttons at the top of the plug-in window. 2. The default behaviour of Dynamics is not to use a side-chain at all, and this is the case when its Control Signal pop-up menu (to the right of the Control level meter) is set to 'Input'. This is the setting you'd use when using the Gate in the normal way, such as when cleaning up a noisy vocal track. However, we want to disassociate the action of the gate from the level of the input signal, so instead choose a buss from the pop-up menu. 3. Now you need to route your new control signal to the Dynamics sidechain input. Taking a suitably rhythmic or percussive audio track (ideally one with some silence between hits, or at least a good, wide dynamic range), choose for its output the same buss as you chose in the Control Signal pop-up This is the basic Dynamics side-chain set up menu a moment ago. However, if you to achieve a rhythmic gate effect. The Dynamics plug-in itself is placed on a track want to also keep your control-signal track audible and routed into your main containing a sustaining pad sound, while audio from a more percussive track is routed mix, you can just as easily route it to to its side-chain via one of DP's busses. the Dynamics side-chain via one of the track's sends. This is best configured as pre-fade (so that the send level won't change if you subsequently move the track's fader) and with a good healthy send level. 4. Now you need to play your sequence and view the Dynamics plug-in window once more. The Control Level meter will now show the level of your side-chain 'key' signal and you can drag the transparent Threshold 'handle' to fine-tune the action of the gate to it. You're aiming for transient peaks to go well over the Threshold, but for the gaps in between to fall well below it. You should end up with the transients opening the gate, and therefore allowing your sustained sound file:///F|/SoS/SoS%2001-2006/performernotes.htm (2 of 8)12/19/2005 10:23:33 AM


to be heard, while the gaps between the transients cause the gate to close again. 5. Now it only remains to adjust the Attack and Release parameters to fine-tune the stuttering effect. Of particular importance is the Release setting: with too low a value you may sometimes notice a strange kind of distortion caused by the gate opening and closing very frequently as the control-signal level passes the Threshold. This could be a good or a bad thing, depending on what kind of music you're working on!

Other Side-chain Plug-ins If you've mastered setting up a side-chain in Dynamics, you can try the same thing with the Masterworks Gate. It works in exactly the same way, except that MOTU use the (rather more correct) terminology 'Key Source', rather than 'Control Signal', to refer to the signal that triggers the gate. The Gate has generally more sophisticated controls (discussed in detail way back in the June 2002 Performer Notes), the side-chain input is equipped with a 'key listen' facility that can make working with and identifying your side-chain key signal rather easier, and there's a pair of low- and high-pass filters. These filters are useful when, for example, you're using a whole drum-kit signal to key a gate, as they allow you to remove unwanted frequency content from the key signal. By setting the LF parameter to, say, 1000Hz, you're dialling in a high-pass filter that would remove most of the kick drum signal, leaving the gate to be triggered only by snare, hi-hat and so on. As well as dynamics processors, other Digital Performer plug-ins offer side-chain audio inputs. Two, Multimode Filter and Sonic Modulator, have a built-in envelope generator that's used as a modulation source for other sections of the plug-in. In both cases it's possible to set this up so that it's triggered by an external audio signal arriving at the plug-in via a side-chain. In the case of Multimode Filter, that means you can set up funky rhythmic wah-wah effects, and for Sonic Modulator the possibilities are even wider. As with the Dynamics plug-in example given above, the steps to setting up your routing go something like this: 1. Place the plug-in on the track you want to treat and switch the envelope section's pop-up menu from 'input' to a buss. 2. Now configure the track you're going to use to key the envelope generator, so that its output (or one of its sends) is routed to the same buss as you chose in step one. Make sure it's putting out a decent level. 3. Play your sequence — and maybe consider setting up a Memory Cycle loop around the section you're working on — and adjust the plug-in's envelope parameters to suit. For a wah-wah effect in Multimode Filter, try envelope generator settings of file:///F|/SoS/SoS%2001-2006/performernotes.htm (3 of 8)12/19/2005 10:23:33 AM


100ms Attack and 250ms Release and set a centre frequency of about 600Hz, with Resonance to taste. Depending on your sidechain signal, you may also have to alter the scale and trigger values, which determine modulation amount and tendency for the envelope to retrigger, respectively. Finally, the Range parameter controls the depth of the wah, and you may want to try flipping the polarity of the envelope by clicking the little arrow button; sometimes negative works better. Some tricks to try in Sonic Modulator (which, incidentally, can only be instantiated on mono tracks) include modulating the Pitch or Delay sections of the plug-in with a side-chain-keyed envelope. The results can be spectacularly unusual (!), but as long as your side-chain audio signal is nice and rhythmic it's nearly always possible to keep things musical and usable.

DP's Multimode Filter plug-in also accepts a side-chain audio signal, but uses it to trigger its built-in envelope generator, which in turn modulates the filter's cutoff or centre frequency.

Third-party Plug-ins MOTU's MIDI-controlled plug-ins are an interesting and useful part of DP, but if you want to see what can really be achieved with MIDI control you have to turn to other plug-in developers. Sadly, not that many plug-ins implement it, but the ones that do, do it beautifully. Without a doubt, the best of the bunch are Audioease's Nautilus and Rocket Science plug-in bundles, with all of their plug-ins implementing MIDI control extensively. Perhaps most impressive is the way MIDI control is implemented extensively in Audioease plug-ins such as Roger. the granular synthesis plug-in Riverrun behaves, with virtually all of its parameters controllable via MIDI note number, patch-change and controller messages. Using a MIDI keyboard and control surface of some kind, you could really play Riverrun like a 'proper' software instrument. Intriguingly, too, the phaseaccurate equaliser Periscope, which you'd be forgiven for thinking was just a pretty boring, static mastering tool, has extensive MIDI control options. One of the craziest is having a MIDI note number select a frequency band and key velocity set the level of boost or attenuation. If you thought you knew what dynamic EQ was, think again...

file:///F|/SoS/SoS%2001-2006/performernotes.htm (4 of 8)12/19/2005 10:23:33 AM


For me, though, still the most downright beautiful and charming MIDI controlled plug-in is Roger, from the Rocket Science bundle. Here, MIDI notes switch the plug-in between the 30 vowel forms shared between the virtual vowel processors (Roger, Patty and Cindy). Velocity can also control the filter bandwidth or the 'morph' time between vowels. My advice is not to try this out if you have any deadlines looming. Whole hours seem to vanish as you play Roger on as many sound sources as you can lay your hands on!

Ring Modulator Amongst DP's bundled plug-ins, there's one more — Ring Modulator — that can accept a side-chain audio input, but the side-chain signal doesn't 'key' anything, instead becoming an integral part of the effect. Ring modulators are 'sum and difference networks', meaning that they require two inputs to work. These inputs are then processed so that the resulting sound consists of literally the sum and difference of their frequencies. DP's Ring Modulator uses the track it's placed on to supply one of the inputs, and normally the other comes from a built-in tone generator, called 'Internal Oscillator' in the Modulation Source section. You can select a different modulation source, though, and set it to one of DP's busses. This opens up the possibility of using audio from other DP tracks as the additional input. Although it's hard to make any firm recommendations about settings, since the ring-modulation effect is so materialdependent, you might try sending sporadic very low synth notes (50Hz or lower) to the side-chain input as a possible treatment for vocals, other melodic signals, or even drums. The presence of a note would trigger Dalek-like gurglings and shimmerings, and as long as the Mix parameter is set to 50 percent you won't lose all your input signal when your side-chain is silent. If anyone comes up with any other good effects using this method, I'd love to hear about them!

MIDI Control If you're thinking that audio side-chains open up some great sonic possibilities, even more are on offer in the form of the slightly mind-bending 'MIDI side-chain'. Since an application such as DP unites audio and MIDI environments, it's easy to get MIDI signals to control the parameters of an audio plug-in, and that's just what the MIDI side-chain concept is all about. Three bundled plug-ins offer MIDI control, all of which have already been mentioned. First of all, Masterworks Gate can have its Key Source pop-up menu set to 'MIDI Notes', so that MIDI note-on and note-off messages, rather than an audio signal, key the opening and closing of the gate. For many DP users, this makes setting up rhythmic gate effects far easier to achieve, and great deal easier to modify and edit once they're up and running. 1. Place Masterworks Gate on the audio track you want to treat with the gate file:///F|/SoS/SoS%2001-2006/performernotes.htm (5 of 8)12/19/2005 10:23:33 AM


effect and switch the Key Source to 'MIDI Notes' 2. In the Tracks Overview or Sequence Editor, create a MIDI track, click on its output pop-up menu and choose 'MW Gate : [Track Name] : [Insert Slot]' to correspond with the track and insert slot in which you just instantiated it. If you've only one Masterworks Gate in your mix there will only be one choice here. Virtual Instrument plug-ins are driven with 3. Now record-enable the track and MIDI messages, so why shouldn't audio record in some MIDI notes. processing plug-ins be too? Here DP's Ring Masterworks Gate isn't at all fussy Modulator is set up so that its Internal about which MIDI notes it receives — Oscillator frequency is being controlled in real time by a MIDI track. it's really just interested in note-on and note-off events — so pitch and velocity are unimportant, and you only need single notes. It may be best, though, to ensure that there are clear gaps between your notes, to trigger the gating cleanly, so a staccato part can work well.

4. For ultra-rhythmic effect, quantise the MIDI notes you just recorded, then hit play and get ready to tweak the Gate parameters to achieve the desired effect.

Digital Performer News Wave Arts Update Plug-ins: Exciting developments are afoot at Wave Arts, the US-based software developer very much a favourite in DP circles (as their plug-ins are offered in true MAS format and are always reassuringly reliable, useful and light on the CPU. Wave Arts' entire range is being updated to version 5, with superb new user interfaces, and even more features. Already powerful weapons in many a DP A new version of Trackplug, a oneusers' plug-in armoury, the range of Wave stop solution for channel EQ, gating Arts plug-ins has been updated to version 5, and limiting, is already available as I with tidy new GUIs, more features and write, along with the Finalplug and improved sound quality. Multidynamics mastering processors. What looks to be a greatly enhanced Masterverb should also be available by the time you read this, sporting comprehensive early reflection control and improved quality throughout. More info is available from www.wavearts.com.



DP 4.6 fixes some bugs: If you haven't downloaded it already, head over to www. motu.com to download the version 4.61 update to Digital Performer. This offers various bug-fixes and a few new features, amongst which is a genuinely useful keyboard shortcut for the 'Expand' button in edit windows, which opens and closes the track selector list. By default this is Apple-alt-E, and it should come in very handy.

'Playing' Plug-ins The other two bundled plug-ins that offer MIDI control, Multimode Filter and Ring Modulator, work in a slightly different way, whereby the pitch of the MIDI notes you play is important. That's because, in both cases, the MIDI 'side-chain' controls a frequency-based parameter, and this is set to a frequency corresponding to the pitch of the MIDI note received. In the Ring Modulator, you get to play the Internal Oscillator via MIDI. No setup is required in the plug-in window, other than to make sure the Modulation Source is set to 'Internal Oscillator'. After you set a MIDI track's output to drive Ring Modulator, MIDI notes control the Frequency parameter — it's as simple as that. Perhaps for the first time in history, this opens the possibility of ring modulation becoming a truly melodious effect! Bear in mind, though, that the oscillator you're playing is only monophonic, so if you play a chord it'll still only produce one note. If anything, the MIDI control of Multimode Filter is more useful. Again, no setup is required in the plug-in window, so you just need to set a MIDI track's output to Multimode Filter and play (or play back) some MIDI notes. These now control the Center frequency parameter, and that allows a number of interesting effects to be set up. First, you might try playing alternately very high and very low notes, to achieve a kind of rhythmic 'filter muting' effect. Because you can alternate the notes in any rhythm you like, this can be much more interesting that just setting up an LFO to do a similar thing. What's more intriguing, though, is turning up Resonance and then playing the resonant peak that results. Depending on the Center frequency setting and the source material, this can be a very musical effect, with proper pitched notes as an outcome, but all the while very dynamic and pleasantly uncontrollable. Watch your speakers, though...

Trim Before I conclude this month's exploration of DP plug-ins, I can't leave out one which, while it might not do anything really radical to your sound, is wonderfully useful in all sorts of situations and is a real problem solver. You might be forgiven for thinking that Trim is a sort of digital gain stage, just for boosting or attenuating signal level in DP. It can indeed do this, but it can do so much more besides. To start with, it has great metering, with momentary,



average and peak displays, all across a user-definable range from -144dB to +20dB. I use this all the time to find out precisely how much headroom I've got in a track, or how much I'm overdoing it with EQs and other level-boosting processors. Then there's phase inversion, which can be so useful in dealing with snares miked from underneath, wiring discrepancies and other studio disasters. The humble Trim plug-in, so much more than a gain stage, is your friend when it comes to metering, balancing stereo, controlling stereo width and fixing any number of audio problems.

But it's the stereo version of Trim that's really worth its salt. The independent gain controls, which can be unlinked by clicking the button that sits between them, are often better at balancing a stereo channel's levels than its pan knob in the Mixing Board, since boosting one channel doesn't cause the other to be attenuated if you use Trim. The pan controls are also tremendously useful. When linked, they can progressively narrow a stereo recording until it's mono, and turning them still further causes a stereo track's channels to be reversed. If you ever needed to subjectively 'narrow' a stereo recording of a drum kit or piano to fit your mix better, this is the plug-in for you. Published in SOS January 2006 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



Mastering Your Album In Logic

In this article:

Incorporating External Hardware The First Stages Logic News Processing Tips Mastering The Art

Mastering Your Album In Logic Logic Notes & Techniques Published in SOS January 2006 Print article : Close window

Technique : Logic Notes

Current Versions Mac OS X: Apple Logic Pro v7.1.1 Mac OS 9: Emagic Logic Pro v6.4.2 PC: Emagic Logic Audio Platinum v5.5.1

We look at how to combine hardware and software processing within Logic for mastering purposes. Paul White

Since the demise of Digidesign's Sound Designer II as a mainstream editing platform, I've ended up doing virtually all my mastering within Apple Logic Pro, compiling and processing individual tracks before switching to Roxio's Jam to compile my album master. Although I've tried a whole range of mastering plug-ins, including the System 6000-inspired TC Electronic Powercore ones, the Waves suite, and independent products such as the PSP Vintage Warmer, I'm constantly amazed that my Drawmer DC2476 Masterflow hardware box still beats the pants off them all. I don't know why, but it sounds more 'analogue' than any other digital processor I've used.

Incorporating External Hardware Accessing the DC2476, or any other external hardware processor for that matter, within Logic can be neatly achieved using the I/O plug-in (available from the Helper plug-in submenu), provided that you have at least two spare inputs and outputs on your interface. The plug-in lets you select a physical output to use as an insert send, and a physical input to use as your insert return. For mastering, you'll obviously need the stereo version, and if you're digitally connecting external hardware, you'll either need to set the unit to external sync or else run everything in your system from a master clock generator. It's probably worth mentioning here, though, that the current version of Logic

file:///F|/SoS/SoS%2001-2006/logicnotes.htm (1 of 5)12/19/2005 10:23:36 AM


seems to have a bug that prevents the I/O plug-in working properly when inserted into the main stereo output Audio object. One workaround for this is to route all the tracks and busses to another Output object (3+4 for example), and then send to your external hardware from that, returning the externally processed audio via an Aux Audio object set to unity gain. The Aux object can then feed the main stereo outputs.

The First Stages However, even if this bug did not exist, there is still an advantage in inserting your mastering processing into a normal stereo Track Audio object. That's because, when you're mastering the different tracks that will comprise an album, it helps to pick one track, optimise that, and then use it as a yardstick for the other tracks you need to work on. As a rule, I work on the loudest track first, optimise it using whatever processing is required, then bounce it to a new 24-bit interleaved AIFF file. Note that it can help to open your bounced files in the Sample Editor, so that you can trim any excess 'dead air' from around the song or fade out the thumps and bangs that are occasionally introduced at the start of the file by certain plug-ins. However, do leave at margin of at least a half a second at the start of each track for the benefit of those CD players that are a bit slow at cuing tracks — you can always make this truly silent by using the Silence command in the Sample Editor window's Functions menu.

Once I'm happy with the first track, I drop it onto a spare stereo track in the Arrange window, where it can be played for direct comparison with the next track while I am fine-tuning plug-in and processor settings, the aim being to get the second track to sound as if it The I/O plug-in is the key to accessing belongs to the same album as the first. external processing, such as the multi-band Obviously it's important to make sure tube emulation of the Drawmer DC2476, that the faders of both Track Audio while mastering in Logic. objects are initially at unity gain so that there's no level mismatch between them, but you may find that you need to increase the second track's limiter gain to achieve similar subjective levels. I usually start work with tracks that have similar instrumentation and style. I also favour working on all the heavier songs first, before working down to the ballads if the album is that varied. Working from loudest to quietest helps you match the subjective levels of the different tracks making up the album with less risk of running out of headroom. You can make final level adjustments in your CD



burning program, but if you turn down a track that has already been limited, you're wasting headroom that might have been better utilised by limiting less in the first place. After all, limiting very rarely makes anything sound better, it just makes it sound louder, so there's no point in limiting the quieter tracks more than you have to.

Logic News There have been a couple of Logic-related upgrades recently. The Mac OS X Waveburner utility has reached version 1.1.2 and is now almost up to the feature and performance levels of its OS 9 counterpart. It's definitely more stable now, the undo feature finally works as expected, and plug-ins no longer crash or lose their parameter settings. Speaking of plug-ins, Waveburner can use all of your thirdparty AU effects plug-ins as well as Logic's own, but with the strange exception of the Space Designer reverb. Apple's latest Pro Application Support 3.1 update is also recommended if you are running Logic, although it often doesn't show up in Software Update unless you've installed Final Cut Pro as well. Another well-hidden Apple item is the availability of a free trial of Logic Express downloadable from www.apple.com/logicexpress/trial. It's a 30-day time-limited version, and a perfect way to dip your toes into the choppy waters of Logic, even if you're thinking of going for Logic Pro in the end. More problems with Logic's half-hearted plug-in delay compensation (PDC) are appearing as users start to put it through its paces. The one I find most annoying revolves around the use of Universal Audio's UAD1 plug-ins and, as far as I can tell, has only appeared since v7.1.1. If you insert a UAD1 plug-in into a track or buss and have PDC set to All, only the UAD1 plug-in inserted into the first slot will have its latency compensated for, which renders the feature almost useless in a lot of cases. One workaround seems to be to reorganise Logic's memory by double-clicking the memory value below the tempo display in the Transport — although this doesn't always work for me. The only other option is to Freeze the UAD1 tracks with PDC set to Audio Tracks And Instruments to get everything back in time. Apple really need to get PDC sorted out properly in the next release of Logic. On a side note, the memory reorganisation tip I mentioned above is often useful when you're having spurious memory-related or CPU problems. It's often helped me to get a Song that's started to complain running again. Another tip is to start and stop Logic's transport, and then start it again. This has the added advantage of making sure Logic realises that both CPUs are actually being used on a dualprocessor Mac, so you don't get CPU overload errors. Stephen Bennett

Processing Tips To me, the process of mastering is about making the songs on an album sound as good as they possibly can, but at the same time ensuring that they sit comfortably next to each other in terms of tonal balance and subjective loudness. This can be tricky when working on songs recorded in different studios, or when assembling compilation albums. Often the bass end will need a little equalising to compensate for deficiencies in the original monitoring environment. Because I file:///F|/SoS/SoS%2001-2006/logicnotes.htm (3 of 5)12/19/2005 10:23:36 AM


don't have a 'top of the tree' acoustically designed mastering room I'll also compare my results with similarly styled commercial releases that I deem to be well recorded. After that the album also gets checked on several other sound systems to ensure that it 'travels' well. Some digital equalisers seem to need lots of cut or boost to get the necessary tonal change, but the DC2476 sounds a lot more analogue, and often a decibel here and there is all that's needed. Most times I'll add only a little 80-100Hz boost to beef up a weak bass, but this may be teamed with an additional filter to roll off very low frequencies that might otherwise compromise the bass end and make it unmanageable when played over The TC Electronic Powercore's Dynamic EQ plug-in is great for adding weight to bass smaller speakers. A useful tip, if you instruments without otherwise changing the have access to it, is to try dynamic EQ tonality of the overall mix to any great extent. to add punch to bass notes in situations where you don't want to affect the tonality in the spaces between the bass notes. Dynamic EQ can apply cut or boost whenever the signal exceeds a certain threshold in a particular frequency range, so you can, in theory, add more EQ to problem sounds without disturbing the rest of the mix too much. There's a dynamic EQ in the DC2476, but I tend to use the Powercore Dynamic EQ plug-in instead, because of its intuitive graphical interface. Adding a gentle 'air' EQ boost at 12-14kHz is also very commonplace, as it helps to separate the instruments in the mix, but it should be subtle. If you do this with a really good equaliser, the sound will become better focused, but without any excessive tonal changes being introduced. However, every mastering job is of course different, so you have to train your ears to pick out the areas of the spectrum that really need attention. There is no fairy dust setting that works on everything.

Mastering The Art That's all for now, but stay tuned for more mastering tips in next month's workshop, where I'll be looking at how to get the best from Logic's built-in plugins in this context. Plus, I'll be revealing how to get track levels and inter-song gaps just right. Published in SOS January 2006






Mix Rescue

In this article:

Rescued This Month... Introducing Ouja Hear The Differences For Yourself! Preliminary Edits & Bass Fixes Mandola & Djembe Removing Unwanted Background Noise Remix Reactions The Final Result

Mix Rescue + Audio Files Published in SOS January 2006 Print article : Close window

Technique : Recording/Mixing

Can't get your mix sounding right? Let SOS sort it out! Paul White

Everyone who's tried their hand at mixing will know the feeling. You work on a mixdown for hours, carefully balancing faders, tweaking compressors and EQs, and automating every last aux send, but you still can't get the sound you can hear in your head. You end up doubting your monitoring, your ears, and even the quality of the song itself. You're not alone. In fact, it's because this kind of creative burn-out is so common that so many commercial productions end up requiring the services of a specialist mix engineer. The idea behind this is that someone with a fresh pair of ears and a wide-ranging experience of different productions can bring valuable objectivity to the final product, focusing the mix around those elements which really make the production stand out. But such luxuries are not really an option for most home studio owners, so in reality most just go it alone and hope for the best. We hear the results of this every month, either via CDs sent direct to the SOS office or via MP3s posted in the dedicated My Sound Files area of the SOS Web Forum, and in a significant number of cases the mix is the main weakness of the final result. So we've decided to tackle this malaise head-on by taking problem mixes from real readers and remixing them from scratch ourselves to try to get closer to the sound they had in mind. This, of course, means that we're not going to entirely reconstruct tracks and add significant new parts in the way that a dance remixer would — instead we'll try to work wherever possible with the

file:///F|/SoS/SoS%2001-2006/mixrescue.htm (1 of 8)12/19/2005 10:23:39 AM

Mix Rescue

existing recorded material.

Rescued This Month... Ouja are Georgina Clarke (violin, vocals) and Paul Adams-Groom (mandola, vocals), and they have played to many audiences, including beer festivals, folk festivals, folk clubs, and even art groups. They have been playing festivals and venues together for over four years now, re-arranging traditional tunes from numerous cultural backgrounds and developing their own unique sound. Click here to email www.ouja.co.uk

Introducing Ouja The first candidate for our remixing challenge was an instrumental track recorded by violinist Georgina Clarke as part of her music technology college course. Outside college, Georgina is one half of a duo that goes under the name of Ouja, playing material, much of it original, based on the style of eastern European folk music. Her partner Paul plays Mandola and, for this particular recording, Georgina had brought in a couple of friends to add a bass guitar line and a djembe drum part. A solitary MIDI track was used for a sampled accordion counterpoint line in the middle section, and this was originally added at the college using IK Multimedia Sample Tank. Before starting out, I asked Georgina how the track had been recorded. She explained that both the mandola and the violin had been recorded using two mics, one close to the sound hole and one further away, but the choice of mics was probably not optimal for these instruments, as she'd used a combination of AKG C1000 backelectret mics and Shure SM58 dynamics. These are fine mics for their intended live applications, but a more sensitive capacitor mic with a better high-end extension would have been preferable. Using two mics on a single instrument also brings up the potential problem of phase cancellation affecting the tone in a negative way, while placing one mic very close to the sound hole runs the risk The Bass Amp plug-in settings file:///F|/SoS/SoS%2001-2006/mixrescue.htm (2 of 8)12/19/2005 10:23:39 AM

Mix Rescue

of over-emphasising the body resonance of the instrument. As the tracks I was provided with comprised the mix from both mics, I wasn't able to separate these.

Paul used for processing his replacement bass-guitar line.

The bass guitar was simply plugged directly into the line input of the 02R used for recording, while the djembe was recorded by pointing a mic into the open bottom end of the drum. The final recording was to an Alesis HD24 hard disk recorder, which made it easy for Georgina to bring me the separate tracks as WAV files, which I could then import into my own Apple Logic Pro system for mixing and processing. The MIDI part for the accordion line wasn't on the CD-ROM she brought along, so she played that part in again. She also brought along her original mix, which to my ears was over-EQ'd, making it sound rather dull, and she'd also used a lot of reverb with the aim of creating a big and spacious sound. The balance Georgina had set up was pretty good, but because of the way the track was treated, it sounded almost like listening to the band from a distance, or from outside the room in which they were playing.

Hear The Differences For Yourself! Judge the changes made to Ouja's mix for yourself by checking out the following audio examples I made during the session — they can be found at www. soundonsound.com/sos/jan06 plus following filepaths: /audio/OriginalMix.mp3

This is the original mix sent in by Georgina when she asked SOS for help. /audio/OriginalBass.mp3 /audio/NewBass.mp3

The unprocessed original bass part and its replacement. /audio/OriginalDjembeTrack.mp3 /audio/ProcessedDjembeTrack.mp3

A section of the original unprocessed djembe track, and the same part processed using Apple Logic's Enhance Timing plug-in, Noveltech Character, Universal Audio LA2A, and Waves L1 limiter. /audio/OriginalDjembeLoop.mp3 /audio/ProcessedDjembeLoop.mp3

A loop edited from the original djembe track, and the same loop processed via Apple's Logic SubBass, Waves L1 limiter, and Apple Logic Tremolo used as a tempo-sync'ed audio chopper. /audio/Version1Remix.mp3

The final remix as agreed by Paul and Georgina. /audio/RMXDjembeGroove.mp3 /audio/Version2Remix.mp3


Mix Rescue

After finishing off, I couldn't resist tinkering a little more, and added in another djembe groove from Spectrasonics Stylus RMX in order to add more dynamics to the track as a whole. I think I prefer it, but you decide what you think...

Preliminary Edits & Bass Fixes Fortunately for me, Georgina had decided to record by overdubbing one instrument at a time to a click track rather than playing everything live, so separation wasn't a problem, and, because the tempo was fixed to the click, I was able to do some basic editing. After loading the WAV files into a Logic Song, the first thing I did was to try to establish the tempo at which they had been recorded, because Georgina didn't have a record of this. Logic's tempo counter actually made a pretty good estimate working from the djembe track, but I found I could also get the same result by adjusting the tempo manually while listening to Logic's own click. Fine tuning could then be done by looking at the waveform display for the last bar of drums (which of course would drift the furthest out of time in the case of any error) and then making very fine tempo adjustments until these sat exactly on the bar and beat lines. I did this check anyway, just to ensure that there was no drift, and ended up with a very odd tempo value to three decimal places, so apparently their click didn't run at exactly the same rate as the one on my system. The next check was to listen to the individual tracks to see if there were any problems, and it turned out that the bass-guitar part was going to be the hardest to deal with. This had been DI'd and the tone was rather dull, but with a lot of high-frequency fret noise and rattle that stifled any attempt to use EQ to better focus the sound. The timing was also a bit on the rough side, and to my mind the player had picked the notes too gently, so there was no real definition to the sound. After a few minutes playing with EQ, compression, and Logic's Enhance Timing plug-in, I decided that it was never going to sound entirely satisfactory, so I replaced it with a new bass-guitar part played with a pick and following the notes of the original as closely as I could make them out. This was DI'd via the high-impedance instrument input of my MOTU 828MkII interface and then shaped using the Garage Band Bass Amp plug-in, which now comes as part of both Logic Express and Logic Pro.


Mix Rescue

Users of other platforms could achieve much the same result using any ampmodelling software or by DI'ing via a modelling guitar or bass preamp. The end result was a more middly bass sound with more definition, and I felt it would sit better in the mix as a result. Compression was added after the amp plug-in to even out the sound, but otherwise it seemed fine for this track, where it played a basic supporting role.

Mandola & Djembe

Although the mandola part had been wellplayed, there was quite a bit of room tone on the recording. However, this was soon sorted out with some low-frequency EQ cut and a dash of ambience reverb from the TC Powercore Classicverb plug-in (bottom). A second reverb, created using Apple Logic Space Designer (top) was applied to flatter just the violin part.

Checking Paul's mandola track showed it to be well played, with no timing or tuning problems, though the low end seemed excessively boomy, presumably because of the location of the close microphone. There was also a certain amount of room tone in the sound, because of the use of a second more distant mic, but it wasn't too serious. Where artificial reverb is going to be used at the mixing stage, it is invariably best to eliminate as much room tone as possible at source by using acoustic absorbers (foam or duvets, for example), as room reverb tends to dictate the tonal character of the sound no matter what artificial reverb you apply later. My strategy in this case was simply to use some lowfrequency EQ cut to clean up the low end, adding a hint of high-end boost to put some gloss back into the sound. I also applied compression to even up the level using the Waves Renaissance Compressor plug-in. A short ambience reverb was then set up on one of the sends to add life to the mandola and djembe drum without clouding the mix. We made the artistic decision to drop out the mandola for one short section in the middle of the song, in order to add variety and to improve the general dynamic of the piece. This was achieved simply by cutting the file at the appropriate places, then fine-tuning the cuts so that they fell exactly before new 'strums'. As the rhythm part was fairly staccato anyway, the lack of sustain at the drop-out point wasn't really noticeable, and the ambience reverb helped conceal it further. All that remained to be done then was to clean up the starts and finishes of the track to ensure there was no unnecessary noise. Turning to the djembe track, the timing turned out to be slightly loose in places — it's not easy playing to a click track when you're not used to it, as the conscripted bass player no doubt also discovered! The tone was also lacking in attack and brightness, but with a little cheating I figured that it was probably going to be usable. To establish a more solid, confident-sounding rhythm, I located a bar of straightforward four-to-the-bar playing, copied it to an adjacent track, and then looped it throughout the song so that it underpinned the generally more busy sound of the djembe part. I also used Logic's Enhance Timing plug-in (a type of


Mix Rescue

audio quantisation) to tighten up the main djembe part. This worked pretty well, except for some sections where very fast fills were being played. I could have used EQ or a harmonic enhancer to add more attack to the djembe sound, but settled on a rather clever plug-in called Character, which is part of the TC Electronic Powercore suite of plug-ins. This emphasises certain parts of the spectrum after the manner of an enhancer, but it does so dynamically and in a way that isn't simply confined to bringing out the high end. As the name implies, it seems to lock in on whatever gives a sound its character, then emphasises it. I tuned this by ear to bring out more of the sound of the hand hitting the drum, then compressed the result to even up the levels — I used the Universal Audio UAD1's LA2A compressor plug-in for this, but there are lots of other compressors that would have done a perfectly adequate job here. The same treatment was applied to the drum loop, and both parts were then treated with a short ambience reverb to give them a sense of space. In the mix I panned the two drum parts slightly to each side to give an impression of width, but without separating them too much.

Removing Unwanted Background Noise

Here you can see the complete processing chain for the Djembe drum part. After some quantisation from Apple Logic's Enhance Timing plug-in, the Noveltech Character plug-in was used to bring out the attack of the sound dynamically. The Universal Audio UAD1's LA2A compressor then reigned in the overall levels a little, while the Waves L1 Ultramaximizer kept tight control over signal peaks.

Hearing Georgina's violin part in isolation exposed some bow noise, but, given her fairly assertive 'gypsy' style of playing, this wasn't surprising, and for me it was part of the character of the sound. Again, I would have liked to rerecord her using a single good mic in an optimal position, but the sound was at least natural enough to work with. Although the sound was a little bright, I felt that it would be better to work with it rather than agressively shaping it with EQ.


Mix Rescue

The only technical problem with the violin track was a low-level ticking noise that could be heard during the decay of the final note of the piece. I don't know where this came from, but it was a similar sound to what you hear when interference from a computer disk drive is picked up on the output of a soundcard. However, whatever the reason for this, I was more concerned with hiding the problem than discovering the true cause. To achieve this, I used a high-cut filter set to around 10kHz for the main body of the song, taking just a little edge off the violin sound, and then drew in a control envelope on the final note which rapidly closed the filter to 1kHz as the sound decayed. This turned out to be enough to hide the ticking sound without changing the subjective tone of the violin, even when I added compression to smooth out the level peaks in Georgina's playing. I added a little of the short ambience reverb to the violin to help it integrate with the other parts of the mix, but then added a second concert-hall reverb, as this really flattered the sound and came closer to what Georgina had envisaged. Logic's Space Designer was used for both reverbs, but any number of hardwareor software-based reverbs would have done the job. With the tweaks we'd made, it was now possible to have the djembe drums higher in the mix to drive the track along, as the timing was now more accurate and the articulation of the hand playing was more clearly audible. Having the straight rhythm loop underneath also helped keep the beat going, whereas originally it faltered a little when the player was adding complicated flourishes. The final touch was to set up a sampled accordion to play the counterpoint Georgina had recorded onto the MIDI track, and I also added a low-level sampled tambourine part played with no quantisation to give more of a human feel. This helped reinforce the rhythm part and also added a bit of interest higher up the spectrum, which is something Georgina would have liked to have done originally given the studio time. We ended up copying the accordion part over three octaves to thicken the sound, making the lower line legato to produce a more believable drone. Once this was mixed to just support the violin melody, it sounded perfectly natural, even though it sounded less so when soloed!

Remix Reactions Georgina: "Recording this track was a bit of a last-minute job, because we didn't have much time in the studio and we'd been having problems with the equipment. I also originally used a lot of EQ on the violin because I wanted a warmer sound, but I wasn't filtering out the really low frequencies, so the sound just became boomy. In only used a little compression on the violin and mandola parts, just to keep the levels consistent throughout, but I also managed to drown everything in reverb! As for the djembe line, Paul and I were never happy with the way it came out at all. "Now that Paul's worked on the mix, everything's sounding much brighter, and you can hear from comparing the two mixes how much Paul has bought our tune to life. There's also better separation in the mix, because there's less reverb now — the individual instruments are much more distinct. I particularly like the way


Mix Rescue

Paul has made the most of the djembe line by selecting the best bits and then processing them to dramatically improve the sound of the whole tune. I originally wanted to use more percussion to give the feeling of lots of musicians playing, but we ran out of time before we could do this. Now that the djembe part has been changed it has really beefed up the track without us having to add much else. "It was very helpful to go through Paul's mix in the studio, because he demonstrated to us how he had used simple techniques to get big results. One particular point that stuck with me was how to EQ the bass to allow the other instruments space in the mix — I had simply turned it up and swamped the overall sound." Paul: "This remix session was an intriguing insight into how much you need to know about sound and technology to do a decent recording, but also into how much is on offer in terms of sound quality and capabilities once the technology has been mastered."

The Final Result While not perfect, the final mix was more open and 'present', and also closer to what Georgina wanted. But it also showed the limitations of the 'fix it in the mix' approach, so I can only reiterate that the way to get a great mix is to get sounds right at source. Published in SOS January 2006 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



PCI Express: What Does It Mean For Mac Musicians?

In this article:


Digi Catch The Express What About Everyone Else? Apple News In Brief Apple Notes Audio Units Meet VST Published in SOS January 2006 Instruments Print article : Close window Jack Plugs In

Technique : Apple Notes

With the new single-processor, dual-core Power Macs shipping, we continue to investigate the impact of PCI Express on the Mac audio and music world. Mark Wherry

Forget about crocodile wrestling — writing about technology is perhaps one of the most hazardous tasks you can undertake. If being technically accurate isn't hard enough, the fact that technology changes constantly makes the job nigh-on impossible. The ink barely had time to dry on last month's Apple Notes (discussing the introduction of new Power Macs featuring PCI Express expansion slots that would make it impossible to run Pro Tools HD) when Digidesign announced PCI Express versions of their Core and Accel cards required to run the ubiquitous music production system.

Digi Catch The Express According to Digidesign, the company "is now concluding its qualification process of the PCI Express systems with the range of Apple Power Mac G5 computers." The new PCI Express Core and Accel cards are expected to ship before the end of the year (they may be available by the time you're reading this column) and will cost the same as the current PCI models, which will continue to be sold. Digidesign also confirmed that there will be "a crossgrade program for PCI users who wish to switch to a PCI Express solution." One point to bear in mind, though, is that 7.1 will be the first supported version of Pro Tools with the new PCI Express hardware, and while 7.x Sessions aren't compatible with 6.9 directly, it is possible to export them in a compatible format (see our Pro Tools 7 review, page 90, for more). Since Apple's new Power Macs all have three PCI Express slots, those running an HD3 system will be fine. But what if you use an expansion chassis with a

file:///F|/SoS/SoS%2001-2006/applenotes.htm (1 of 5)12/19/2005 10:23:50 AM


larger HD system? Hold on to your PCI cards, because Digi will also be introducing a new product, the Expansion HD six-slot PCI chassis, at the end of the year. That will be available with either a PCI Express or PCI card to attach to your host computer and can be effectively viewed as a PCI Express to PCI bridge. It will be interesting to see if an upgrade is offered for existing chassis users if the chassis itself isn't compatible with the new PCI Express bridge. Whether this new Digidesign chassis will be a re-badged Magma (www. magma.com) device, as with previous Digidesign chassis products, is unclear, but Magma themselves also have a PCI Express to PCI six-slot chassis "coming soon." Interestingly, Magma are going to offer a 'conversion path' between the PCI-to-PCI chassis and the PCI Express model, so this might be of use to existing Pro Tools users who have a chassis already.

What About Everyone Else?

Digidesign will support Apple's move to PCI Express slots by offering PCI Express versions of their Core and Accel cards (the original PCI Accel card is pictured here), plus a new expansion chassis.

While the number of Mac owners using a Firewire or USB-based audio interface is probably quite large compared to those using PCI offerings, there are some products with certain specifications that are only available as PCI cards, RME's HDSP MADI card being one example. I know many Logic users who use this card with RME's ADI648, to integrate their sequencing and mixing environments, for example, and these are precisely the people who would benefit the most from the increased power of the Power Mac Quad. I spoke with RME's Matthias Carstens [see also the audio interface manufacturers' Round Table feature in our last issue), who confirmed that "Naturally we will add PCI Express versions of existing products to our line", and expected the first announcements to be made at next year's Frankfurt Musikmesse. "Using the latest FPGAs [Field-Programmable Gate Arrays], we will be able to fully implement all currently known RME features. An example is the HDSP 9652, where the FPGA is completely filled in the current model. This card doesn't have Steady Clock, and also misses phase inversion and the optional +6dB gain in [the] Total Mix [mixer], but the PCI Express version would have these features. We will also make a PCI Express version of the current HDSP PCI card for Digiface and Multiface users, but this card will be function-identical to the existing PCI model, for compatibility." There are no plans to offer an upgrade program at this time for those wishing to file:///F|/SoS/SoS%2001-2006/applenotes.htm (2 of 5)12/19/2005 10:23:50 AM


replace their PCI cards with the PCI Express alternative, and Matthias was keen to point out that Apple's announcement had no impact on the company's plans to bring PCI Express products to market. Although RME use FPGAs to implement their products, the reason why other manufacturers haven't specifically announced PCI Express cards could be the fact that there haven't been suitable off-the-shelf PCI Express solutions available, such as a PCI Express to PCI bridge chip, or more complete solutions like Via's Envy24 PCI audio controller chip, which M Audio use in many of their products.

Apple News In Brief Apple released Mac OS 10.4.3 at the end of October, offering numerous fixes for most areas of the system, including unspecified fixes for Core Audio. Quartz 2D Extreme, by the way (which will bring hardware acceleration to the 2D graphics used by applications such as Cubase SX), is still disabled and unsupported in Tiger. You can download the new version via Software Update or on the web at www.apple.com/support/downloads/ macosxupdate1043.html, where you'll also find more detailed information. Digidesign hadn't qualified 10.4.3 at the time of writing and still recommend 10.4.2, although I have used 10.4.3 with Pro Tools 7 and didn't experience any obvious problems. RME have released an updated driver for users of the HDSP MADI (www.soundonsound. com/sos/mar05/articles/rme.htm) and AES32 audio cards. The new release brings the Mac software into line with the Windows version and includes new Total Mix and Settings applications that enable Mac users to configure features such as the DDS (Direct Digital Synthesizer) for pull-up and pull-down options (related to film frame rates), with full varispeed support in 56-channel mode on the MADI card. The Total Mix mixer now supports a monitor panel, submix view, updated preferences and remote control via MIDI.

Audio Units Meet VST Instruments In October 2005's Apple Notes (www.soundonsound.com/sos/ oct05/articles/ applenotes.htm) we looked at the free AU Lab Audio Units host application that Apple include as part of the company's developer tools package, supplied free of charge with every copy of OS X. After reading the article, someone at the studio where I work asked if AU Lab could be used with Cubase, to run Audio Units alongside VST Instruments, and the answer is yes. One nice thing about AU Lab is that for every Audio Unit Instrument you add, the application will put a virtual MIDI port on your system (for as long as AU Lab is running) that lets you trigger each Instrument from another application. So, for example: Open AU Lab and create a new Project with the default options. Select Edit / Add Audio Unit Instrument. Choose a plug-in and make sure the MIDI Input Source is set to 'None',



because we want to trigger the Audio Unit via another application, not directly from a hardware MIDI input. Load Cubase and create a new MIDI track. You should be able to set the MIDI output of that track to the AU Instrument in AU Lab (a port will be available with the same name as the plug-in you added). Play some MIDI data to this virtual port and you should hear the AU Instrument being triggered. I've noticed on occasion that if there's no sound at first, you need to double-click in an empty area of the Instrument Channel in AU Lab, set MIDI Source to a physical port, trigger some data from that port, then set it back to None. After this, you should be able to trigger the AU Instrument from Cubase.

Jack Plugs In So now you can trigger an Audio Unit plug-in from Cubase via MIDI. But wouldn't it be great if there was some way of routing the audio output of the AU Instrument back into the Cubase Mixer instead of going directly to a hardware audio output? You might remember a utility called Jack, which I've written about in previous columns, that can do just what we need. Jack is similar to Rewire in that it's basically a virtual cable for routing audio between applications, but it also supports physical audio hardware and is compatible with any application that makes use of Core Audio. To start with, download and install Jack (www.jackosx.com), run the included Jack Pilot application, then click Start in the Jack Pilot window. Next, run AU Lab, start a new Project, choosing Jack Router as the Audio Device, and add an Audio Instrument as before. In Cubase, set up a MIDI track, route its output to the port labelled with the name of the AU Instrument you added in AU Lab, and create an audio track. One really nice thing about the Jack OS X installation is that it installs a special plug-in (in VST and AU formats) that allows you to route the audio output of an application into a plug-in (or the audio output of the plugin to another application). So by adding file:///F|/SoS/SoS%2001-2006/applenotes.htm (4 of 5)12/19/2005 10:23:50 AM

Here you can see an AU Instrument plug-in, running in AU Lab, being triggered from a MIDI track in Cubase. Its audio output is being routed into a Cubase audio track, via Jack and a clever VST plug-in included with the Jack OS X distribution.


Jack-insert as an insert plug-in on our newly created audio track in Cubase, we'll be able to stream AU Lab's output into that track. In Jack Pilot, click the Routing button on the main window and you should see the Connections Manager window, which consists of three columns: Send Ports, Receive Ports and Connections. Send Ports shows all audio ports (whether applications or physical hardware) that can send audio to a Receive port, and Connections shows whether there is a connection between the currently selected Send Port and a Receive Port. Jack uses your Mac's built-in audio hardware by default, but if you want to use additional audio hardware you can set this via Jack Pilot's Preferences window. If you click on AU Lab in the Send Ports column, you'll see that its output has automatically been routed to the physical audio hardware output. To remove this assignment, double-click each entry in the Connections column To route the output of AU Lab into Cubase, expand the Send and Receive Port entries to show all available audio streams, by clicking on the little triangle next to a Port's name. Next, select AU Lab 'out1' and double-click Cubase 'VSTreturn1' to make the assignment for one stream in AU Lab to the Jack-insert plug-in within Cubase. Repeat the process for AU Lab 'out2' and 'VSTreturn2'. Cubase itself talks to Core Audio directly, so Jack is only needed to route the output of AU Lab into the Jack-insert plug-in. That's it — and it should be less complicated to do than to explain. Play the AU Instrument from your MIDI track in Cubase and you should notice hear the audio come back in via the Jack-insert plug-in to the audio track. Pretty neat. And while this example is for AU Lab and Cubase, the technique will work with any application that supports Core Audio in OS X, so there's plenty of room for experimentation. Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Recording & Remixing With Ableton Live

In this article:


Set Up, Look Sharp Recording Processed Signals Ableton Live Notes & Techniques Published in SOS January 2006 Let's Boogie Name & Number Print article : Close window

Technique : Ableton Live Notes

In the first instalment of this two-part article, we examine how you can use Live as a conventional digital audio workstation. Ingo Vauk

Although Live is mainly known as a loop-based sequencer, it has a surprisingly powerful set of features that make it a useful tool for recording and arranging audio more conventionally. This month, we'll look at how to approach a multitrack recording session using Live as the front end — and next month we'll take it apart and explore how you can use Live's unique features to remix and A multitrack take in Live's Arrangement view rework the arrangement you've created. looks very similar to any other DAW.

Set Up, Look Sharp Before we begin, there are a few settings in the default preferences that need to be adjusted in order to turn Live into a multitrack DAW with multiple active inputs. To activate inputs from your interface, you can go to the Preferences panel (found in the Live menu on Macs, or the Options menu on PCs), or use the 'Configure' option from the pull-down menu in the Input section of the I/O module, which opens the Audio Device preferences (see the screenshot overleaf). It's worth planning a 'track sheet' in advance to optimise the use of your computer's resources — think about the maximum number of inputs you'll be recording at one time and limit the settings accordingly.

file:///F|/SoS/SoS%2001-2006/livenotes.htm (1 of 6)12/19/2005 10:23:54 AM


A quick aside here — once you've finished your main multitrack takes, it's worth closing down any inputs that are no longer required so as to cut down on CPU overhead (this also holds true for output assignments). Another aspect of resource management is the file format You can you opt for. Obviously the higher the bit depth and change your sampling frequency are, the harder both the drive and the audio interface CPU will have to work. Generally speaking, your drive will settings by handle anything in playback mode that it has recorded clicking the 'Configure' simultaneously without fuss, but remember that your option in the track count will go up once you start overdubbing. Audio From Judging the CPU load is a little more tricky, because of drop-down the plug-ins and additional processing that will be menu. introduced later on. Real-time time-warping is by its nature processor-intensive, and you are going to be doing a lot of that in Live. As with the track count, it is worth having a plan as to what you are trying to achieve with the overall production. If it's sonic purity and high dynamic range across a limited amount of tracks, then by all means increase the sampling frequency and use the 24-bit or 32-bit coding available to you. If, however, you are going to process and tweak the sound extensively, and are planning on track counts to rival those of the finest '80s producers, stick to 16-bit at 44.1kHz. As you might expect, these parameters are to be found in the Preferences as well. Confusingly, though, they're located in two different places — Audio / Settings for the Sample Rate, and Default / Audio Format for bit depth and file type. Generally speaking, a normal 7200rpm drive can reliably record at least 24 tracks of (16-bit/44.1kHz) audio at a time, but it is worth testing your particular system before you invite a bunch of musicians into your studio. I also find that disk maintenance such as defragmentation (or even wiping the drive before a big session) is worth doing, regardless of any claims your operating system might make in that department! While you are setting up the Preferences, it is also worth checking that AutoAssign Colours is set to 'On' in the Default section (see the 'Name & Number' box over the page), and that you've disabled both the Arm and Solo Exclusive buttons in the Misc / Behaviour section. This ensures that Live will allow you to arm and/or solo more than one track at a time. Alternatively, you can Apple/Ctrlclick the Arm or Solo buttons on screen to achieve the same result. It's also a good idea to disable 'Auto-Warp Long Samples' in the Clip Default preference menu, otherwise Live will attempt to quantise your recording after it's been made. With live recordings, it's best to hear what you've actually got without any manipulation — at least at first. This is even more relevant when you're not working to a fixed tempo or click track; in cases like these, quantised playback won't bear any resemblance to what was played in the first place! But of course you can always disable Warping afterwards to restore the unadulterated audio.



The next step is to create the required number of Audio channels. This can be done in either the Arrangement or Session Views using the 'Insert Audio Track' command (Apple/Ctrl+T) from the Insert menu. You do not have to specify mono or stereo tracks, since Live will automatically record in the format determined by the input settings. You can choose the input and output for each channel in the I/O section of the mixer using the key command Alt+Apple+I (Mac) or Alt+Ctrl+I (PC), which toggles this display on or off. It's also sensible, although not strictly necessary, to name each channel/track, since Live will then neatly name all your subsequent takes (again, see the 'Name & Number' box over the page). Now is also a good time to save the Live Set. If you put it into a folder on your Audio drive called 'RED', Live will automatically place any audio you subsequently record into a folder named 'RED Sounds' on the same level as the Live Set itself.

Recording Processed Signals What, no subgroups? No dedicated input channels? Just Audio and Return Tracks? The fact is that because of the clever routing architecture, these two audio objects are all you'll ever need when working in Live. Audio Tracks can basically be any type of console component you want them to be; output busses, input channels, subgroups, or anything else you might think of can be configured by choosing the right source and destination. In the screenshot, you can see the incoming audio being routed through some dynamics processors in the first channel, before being sent on to the second channel for recording to disk. Channel 1 ('In+Plug') takes its input ('Audio From') from External Inputs 1+2. The channel is set to monitor its input at all times, which means that there is always level present on the meter. The channel is running two of Live's dynamic processors — Compressor II, which is set to be a fast-acting limiter to take care of occasional peaks, and Compressor I, which compresses the signal more gently for overall level consistency. In Auto Monitor mode, the channel listens to the input signal when the sequencer is stopped or recording, and during playback, the signal comes off disk. This is the most common monitor setting. The output ('Audio To') of channel 1 is routed to


If you want to record a processed signal you need to set up two separate channels. The external signal arrives at channel 1, where it is processed by the two compression units, while channel 2 records and monitors the signal. This kind of arrangement can produce considerable latency, so it is best to monitor the signal directly from the input of your mixer.


Sends Only. This means that the signal is not present in the mix buss or any of the outputs of the audio interface. Channel 2 ('Post Rec') is the actual recording channel. As you can see the 'Audio From' input is set to receive the signal from '1 In+Plug', which is the processed signal. The track is armed and ready to record; all that remains to be done is to trigger one of the empty Clip Slot buttons in Session View, or switch to Arrangement View in order to record using the Global Record function. Channel 2's 'Audio To' output is routed to the mix buss in the usual way. As you can imagine, the processing on the input side will produce considerable latency, so you cannot use this signal during recording for the musicians' monitor mix. The advantages of recording processed signals are that you free up CPU power for later on, and you capture the polished sound — so there's less painful decision-making required later, at the mix. This is just one example of how routing can be used to create virtually any signal path in Live.

Let's Boogie Although it sounds complicated written down, all of the above should only take about 10 minutes, after which you're ready to record. Naturally you will have to go through the usual soundchecking routines, monitoring the signal both at the input and output stages, and you might want to record a 'dry run' of each instrument to check your mic placements and instrument sounds. From here on, you should stay in the Arrangement View (the Tab key toggles between the Arrangement and Session views). Arm all the required tracks, either via the View / Mixer menu, or by hitting Alt+Apple+M (on Macs) or Alt+Ctrl+M (on PCs). If you are aiming for a tight, steady performance, it makes sense to use a click. Live's own metronome (situated in between the Swing and Ext sync controls, in the top left corner) does the job, but given the amount of loops and grooves Live ships with, you shouldn't have any problem coming up with something a little more inspiring. To set the level of the metronome, use the Cue Level control (headphone symbol) situated next to the master fader. If you don't want to use a click, you don't have to, but if you're planning to tighten up the performance later using Live's time-warping facilities, it makes life a lot easier. Having said that, it is usually easy to force most material into some form of synchronisation using Live, but keep in mind that if you start off with a relatively tight performance, you will have fewer audible artifacts when using the Warp functions. When you are ready to record a take, you press the Global Record button in the centre of the transport panel, or F9 on Macs or PCs, and hit Play (the Space bar). This will drop all armed tracks into Record mode. If you're recording multiple takes, it is probably best to record them sequentially; this will make it much easier to splice multiple takes together. The alternative is to save different takes in different Live Sets, but this is much less simple when it file:///F|/SoS/SoS%2001-2006/livenotes.htm (4 of 6)12/19/2005 10:23:54 AM


comes to editing. Unlike a lot of DAW systems, there is no way of grouping tracks together for editing purposes. This means that you have to be careful to select all tracks when you move them around, so that they stay locked together and you don't run into phase problems. Apple/Ctrl+4 toggles the Snap To Grid function on and off — this is very useful when moving unquantised audio around.

Using the Punch In and Punch Out buttons ensures that you create smooth drop-ins. Spontaneous, manual drop-ins can sometimes cause audio playback glitches, as your CPU may have difficulty handling the 'unexpected' data.

If you find it is necessary to drop into takes, it is best to use the Punch In and Punch Out buttons (either side of the Loop Switch at the top of the screen, as you can see in the picture to the left). Length and position of the drop-in section can be set numerically or by dragging the Loop Start/End bars underneath the time ruler on the arrangement display. Spontaneous drop-ins work best with individual or small numbers of tracks — with multiple tracks, they can be slow and lead to audio glitches while the software is 'getting ready' to record (although this problem varies depending on your hardware and the performance of your system). Once you are happy with any edits and drop-ins you may have made, it is very useful to consolidate the sections of audio into continuous files, so that you don't lose any of the edits by mistake. If you have a multitrack recording that you plan to time-correct, you should create audio files of equal length, since this will enable you to superimpose the Once a recording has been completed with a timing of one track onto all the others number of overdubs and drop-ins, it is useful (we'll look at this in greater depth in the to compile each track into one continuous second part of this article). If, for audio file. example, you have cut the beginnings and ends of tracks to different lengths to get rid of unwanted noise, insert short slices of silent audio at the beginning and end of each track until they're all the same length again. If you now Select All (Apple/Ctrl+A) and then consolidate your selection into Clips by hitting Apple/Control+J, Live will create new audio files of the same length containing all your edits. And there you have it — one finished recording, ready to be mixed or remixed. As I said at the beginning, Live enables you to work very much as you would with a traditional DAW, and up to this point, what I've described doesn't differ much from sessions you might run in Logic or Pro Tools. From here on, however, it's a very different story... In part two of this article, next month, we will look at how to correct timing, and file:///F|/SoS/SoS%2001-2006/livenotes.htm (5 of 6)12/19/2005 10:23:54 AM


deconstruct a recording like the one made this month in order to mix and remix your material in ways that only Live can...

Name & Number It's worth mentioning Live's way of naming and archiving newly recorded audio. As with most DAW systems I have come across, Live names newly recorded audio files after the channel/track they were You can see here how each take is recorded on. It also adds the take automatically named by combining the track number to this name for easy name with the number of the take and identification. Live can identify audio channel. The automatic colour assignment used in a Set even after it has been also helps to identify the take the material renamed in the browser; however, it has come from. is best to label material clearly for your own future reference. The easiest way to do this is to name the recording channels suitably ('Piano', 'Lead Guitar' and so on) prior to the first take by selecting the top of the track and hitting Apple/Ctrl+R, and then let Live do the rest. Enabling the 'Auto-Assign Colours' preference (found under the Defaults menu in the Options menu on PCs, and in the Live menu on Macs) adds a visual guide to the chronology of your takes, assigning a new colour to each one as you add takes and record drop-ins and overdubs. It is often useful to save Sets under different, descriptive names as the project develops, since this gives you the opportunity to return to a certain point in time (before a mix was bounced to stereo, for example). Another advantage of doing this is that Live will create an audio folder in the browser for all new data created with that name. This is useful when you consolidate audio into fresh files after tweaking the raw material, since the consolidated tracks often end up with the same name as the source material. If you have saved the Set under a unique name, Live will have written the consolidated audio into a dedicated folder that has the name of the Set with the suffix 'Sounds'. By clearly labelling tracks right from the start, you will build up a clean browsing architecture that you'll be thankful for as your audio collection grows. Published in SOS January 2006




The Lost Art Of Sampling

In this article:

From Dull To Dynamic A Little History — Multitimbrality & Samplers Killer Composites Positional Crossfading Multi Mode And Finally...

The Lost Art Of Sampling Part 6 Published in SOS January 2006 Print article : Close window

Technique : Sampling

We continue looking at what your sampler's synth engine can do to liven up bland self-sampled sounds, and explain the concepts of layering and multitimbrality. Steve Howell

Last month, I introduced the powerful synth engine features present in most samplers. Ironically, these days, when samplers have more sound-shaping capabilities than ever before, these features are remaining unused by many modern sample-library developers, who are increasingly using multi-layered velocity-switched samples instead of the built-in synthesis capabilities. The increasing realism of these sounds comes at a price, of course, in terms of memory load and processor strain, and it has also contributed to many people feeling there's no point sampling their own sounds any more, as their own efforts inevitably sound 'flat' and 'lifeless' compared to those in commercial libraries. However, by using the oft-ignored synth facilities, even short and often static-sounding self-made samples can be brought to life quite dramatically, as we'll see this month.

From Dull To Dynamic Most samplers present users with a fairly traditional synth-type signal path, except that samples are used in place of the usual oscillators — see the block diagram on the right. They're not shown, but typically the final output amplifier will also feed a set of multi-effects for adding the final polish to any sound.

file:///F|/SoS/SoS%2001-2006/lostscience.htm (1 of 7)12/19/2005 10:23:57 AM


If I have a dull, bland-sounding self-made sample, the first thing I do to make it more interesting is give it some 'shape', a sense of development over time, by reaching for the amplitude envelope controls. Surprisingly enough, simply adjusting these, combined with mapping playing velocity to amplitude, can often transform an otherwise 'lifeless' sound into something quite playable. If the sample you've taken is very short, and you've artificially extended it by using some of the looping techniques described in the fourth instalment of this series, you can even make a difference to how good the loop sounds with the envelope. If the loop is imparting a 'static' quality to the sustain portion of the sound, nudging the level of the sustain portion of the envelope down will help to make it less so. Once I have something that's playable, the filter is usually my next port of call. Whether the sounds are sets of acoustic samples or samples of synths, real drums or drum machines, I've found that they nearly always benefit from having playback velocity applied to control the filter cutoff. This is simple to achieve by reducing the filter's cutoff frequency a little, and slightly A highly simplified block diagram of a increasing the extent to which playback sampler (or indeed any modern samplevelocity affects the cutoff. Working with based synth or workstation). Many also feature multi-effects before the final audio (say) a control parameter range of 0 to output. 100, setting the cutoff to 80 and the velocity-to-cutoff amount to +20 will give a modest degree of tonal control, and you could be more extreme if the situation demands it. These programming routings give a dull, flat-sounding sample a pronounced envelope, with velocity control of amplitude and timbre — much like most acoustic instruments! Additionally tweaking the filter's envelope can also further help to minimise any dodgy-sounding loop points. Sweeping the filter cutoff very slightly with an envelope varies the tone of the sound over the course of a note, and this timbral movement can help to obscure the loop point, or even render it inaudible. If this helps, but the 'static' nature of your loop is once more revealed when the filter goes into the sustain portion of its envelope, you can reach for LFOs to add some amplitude and timbral movement to the sound during the sustain period. For example, by applying some slow sine-wave modulation to the amplifier, the sustained level will rise and fall slightly, adding some subtle interest to the sound. To prevent the modulation being present all the time, you can usually set an LFO delay, so that it only kicks in during the sustain cycle of the sound. Only small amounts of LFO amplitude modulation should be applied, just to make the sound subtly fluctuate in level and add some movement and interest without it sounding too artificial. The same subtle LFO modulation trick can be applied to the filter in moderation if you wish to create a similarly subtle variation in tone during the



course of a sustained note. However, it's important not to modulate amplitude and filter cutoff with the same LFO — the ear can quickly pick up on this repetition and, in some cases, it will actually highlight a loop rather than suppressing or masking it. Fortunately, most modern samplers offer you at least two LFOs to play with — Native Instruments' Kontakt allows you to add as many as you wish! If you use separate LFOs for filter and amplitude modulation, and set them to slightly different rates, you can achieve asynchronous tonal and amplitude modulation that rarely (if ever) repeats in an audible fashion. Sometimes, depending on the nature of your sample, it can even be appropriate to add further movement to a sound by using an LFO to slightly modulate pitch. However, our ears are particularly sensitive to such changes, even very small ones, and it can be over-obvious that the sound is being artificially 'wobbled', so this technique has to be used with care. Similarly, sounds can be enlivened and loop points further disguised or 'smudged' by adding multi-effects such as reverb, chorus or delay, but care must be taken not to overdo it and make the sound too 'artificial' or 'processed' (unless that's what you're after). When coupled appropriately with the kind of synth-engine tricks I described in the third instalment of this series, such as mapping playback velocity to the sample's start point, or velocity to attack time (and modern samplers allow you to control all kinds of other sound-shaping parameters, such as decay, sustain level, release times, and resonance), even the dullest of source samples can be made much more expressive and interesting, and without having to use umpteen memory-hungry velocity layers in the sound.

A Little History — Multitimbrality & Samplers Samplers were fairly slow to adapt to what we would now regard as multitimbrality, although of course you could map different samples across the span of a keyboard from the Fairlight onwards. When Akai first dabbled with the concept on the S900, they implemented it in a very strange (and not very intuitive) way. You had to set up a Program that contained all the samples you wanted in your 'Multi' and then set their Keygroups to the MIDI channel you wanted them to play on! It was very tedious and difficult to use. Things improved with the S1000 — you could renumber Programs so that they had the same Program number, and then set each Program to have its own MIDI channel. Thus, if 'Drums', 'Bass' and 'Piano' were all renumbered to be Program number 1, when this Program was selected, all three Programs would become active at once. It was fairly cumbersome to set up, but it worked well enough, especially live, where a single remote MIDI Program Change command could recall a complete multitimbral/layered collection of Programs. However, synth manufacturers had already established the concept of the Multi mode, where Programs or Patches are assigned to 'slots' on different MIDI channels, and Roland introduced the same simple concept on their S-series samplers in 1986. Other sampler manufacturers followed suit, but it took Akai longer to come around. A 'proper', synth-like Multi mode wasn't introduced until the S3000XL arrived in 1995, and even then, an old-style 'renumbering' multitimbral mode was retained until the release of the S5000 and S6000 in 1998, file:///F|/SoS/SoS%2001-2006/lostscience.htm (3 of 7)12/19/2005 10:23:57 AM


which offered only a 'proper' Multi mode capable of handling up to 128 parts. Even then, many die-hard Akai users complained about the 'new' implementation, and a conversion function was provided to convert the old Multis into the new format.

Killer Composites Creators of synth and sample libraries often layer several samples together to create great composite sounds — a killer orchestral string sound may well be made up of several different string samples, detuned, panned and mixed to create a lush ensemble sound — and there's no reason why you shouldn't do the same with your own self-made samples. Such layering is easy in a modern sample-based synth, where you just select different (say) sampled string sounds in several different oscillators. With a sampler, it's not quite so straightforward, but it's far from impossible. We've already looked at Keygroups as a means of creating multisampled sounds, where you might have one instrument sample handling a range of three or four semitones across the keyboard, and then a different source sample handling the next few semitones in the next Keygroup up. But by deliberately overlapping the Keygroups, you can layer several samples so that they are triggered together by the same keys. A good way of building a bigger, lusher sound is to place samples on top of each other, detune each one (one slightly up, one slightly down, for example) and maybe pan each one to the left and right respectively to create a 'pseudo-stereo' effect, especially if the samples are mono. This technique works well to fatten electronic-sounding synth tones, but it tends to sound artificial if you try it on 'acoustic' sounds such as string or choral ensembles, especially if your self-made samples have very similar attack portions and loop points. In such cases, it's better to layer two different but similar sounds, such as an orchestral string sound with a smaller ensemble string sound, 'real' strings with a warm synth string sound, or a male choir with a female or synth choir patch. When you try this, you often find that detuning (and the phasing/chorus effect it inevitably creates) is no longer necessary to fatten the sound, and you can create composite samples that sound much greater than the sum of their parts. It's by using such techniques that the creators of factory libraries for synths can make such great sounds using such relatively minimal sample-sets as their source material.



Positional Crossfading One of the biggest problems with multisampled instruments and mapping them out across the range of a keyboard is that there can often be an abrubt tonal change as you pass from one Keygroup to another. If your sampler permits this, one way around this is to use a Positional or Keygroup crossfade (as usual, the precise terminology varies from sampler to sampler). When you do this, the ranges of your Keygroups overlap, and as you play up or down the keyboard in the area of the transition from one Keygroup to another, one Keygroup sample fades out as the other fades in, resulting in a smoother transition between Keygroups. This feature is implemented differently on different samplers. On some, such as Akais, you simply overlap the Keygroup ranges and switch on the Keygroup Crossfade A diagrammatic representation of a simple function. Then, if your Keygroups Program with samples on the 'G' of every overlap, they are automatically octave, but using Keygroup overlap/ crossfaded, and if they don't, they behave as normal. Modern software crossfade to smooth the transition between adjacent Keygroups. samplers allow you to overlap the Keygroups and then drag handles with the mouse so that crossfades can be set independently for each Keygroup as required. This is more flexible, but equally, it's more time consuming to set up. As always, however, there can be pitfalls for the unwary. Firstly, if there is a slight pitch difference between the samples in the overlapping, crossfading Keygroups, you might hear anything from a mild flanging effect to a stronger chorus effect in the overlapping region. You can overcome this by tuning one of the Keygroups slightly, but then you run the risk of trouble with the tuning of all the other Keygroups! Secondly, if there is any 'movement' in the sound (such as natural vibrato, for example), then when the sample is transposed as you play higher up the Keygroup, that movement speeds up. The problem is that in a Keygroup crossfade, the adjacent sample is likely to be playing more slowly, so you can get a jumbled mess as the two interact. In difficult cases like this, you simply have to decide whether this is preferable to the abrupt transition that might occur without any Keygroup crossfade in place.

Multi Mode I mentioned last month that I felt samplers to be the ultimate synthesizers, because they can employ any sound as their sonic starting point. They can replay acoustic sounds very accurately, or process and transform them, they can operate like complete synthesizers in their own right, and they can be used to chop up, stretch and mangle drum (and other) loops, and trigger long recordings or spot sound effects. And all of this power can be brought together in the Multi Mode found in all modern samplers.



In this mode, the various single sample-based Programs we have created can be combined, and each one can be played on a different MIDI channel. For example, you could have a sampled piano sound on part 1, on MIDI channel 1; sampled drums in Multi part 2, on MIDI channel 2; bass (synthesized in the sampler from raw synth waveforms) in Multi part 3, on MIDI channel 3... and so on. But unlike on a typical multitimbral synth, these individual multitimbral Parts are not limited to purely instrumental sounds, and could contain anything: Recycle loops, vocal 'spin-ins'... whatever you like. Unlike other software instruments, where you insert different instances on different sequencer tracks each time you want to use them, most modern software samplers operate in a form of Multi Mode by default, and Programs are played simply by assigning them to a Part (usually a 'drag and drop' operation) and addressing them on the appropriate MIDI channel. However, unlike hardware samplers, which typically have to share their internal effects complement across all their multitimbral parts, multitimbral software samplers typically allow you to assign effects at the Program level, so that each part can have its own effects tailored for the sound (provided the power of the host CPU can handle the processing load of all the effects, of course!). However, whilst most people associate Multis with multitimbral sequencing (where different sounds are played on different MIDI channels), they can also be used more simply to layer sounds. If you assign two Programs to two Parts, and set those parts to the same MIDI channel, the two Programs will play together. This is a great way to create composite sounds, as described earlier, and it saves the tedium of having to layer individual samples at the Program and/or Keygroup level. Most samplers have the ability to individually tune and pan the multitimbral Parts (the latter function allowing you to create 'pseudo-stereo' effects on mono samples). The individual Parts in the Multi can, of course, be mixed and balanced, and most samplers also allow you to set key ranges for each Part, too, so you can set up keysplits, where different ranges of the keyboard trigger different sounds. Whilst this may all seem a little complicated at first, it allows a great deal of flexibility when combining sounds in different ways.

And Finally... Next month, we'll be bringing this series to a close by looking at the thorn in almost every samplist's side, namely format compatibility, and considering the future of sampling. Until then, I urge you to start experimenting with the powerful synth engine at the heart of your sampler. Published in SOS January 2006






Tomorrow's Musicians & What They'll Be Playing

In this article:

Keynote Speakers The Tools That Make This Possible Instrumental Innovation Digicon 83 Music By Any Other Name Glimpsing The Future Controllers & Games

Tomorrow's Musicians & What They'll Be Playing Controllers Of The Future Published in SOS January 2006 Print article : Close window

Technique : Miscellaneous

The New Interfaces for Musical Expression conference has been running for five years, and is a great place to see and discuss new ideas that may provide the musical controllers of the future. SOS was in Vancouver to learn more... Paul D Lehrman

Where are the musical instruments of tomorrow going to come from? Surely, now that every computer and game console has more musical capabilities than an entire synth studio of 10 years ago, there should be a flood of new electronic instruments coming from factories worldwide. But making new instruments is risky: you have to design them, build them, and perhaps hardest of all, teach people how to play them — and not lose your shirt in the process.

Photos (from top left): Dan Overholt, Bernardo Escalona Espinosa, Gil Weinberg, Bernardo Escalona Espinosa, Juan Pablo Cáceres Just some of the fascinating new instruments and controllers demonstrated at 2005's NIME conference (from top left): Dan Overholt's Overtone Violin, some of the Speak And Spell-based 'circuit-bent' instruments used by composer Giorgio Manganensi in a NIME concert performance, a trio of MIT's Beatbug controllers, the Bangarama headbanging controller from the University of Aachen, and Stanford University's tuba-based SCUBA system.

Fortunately, there are plenty of people in universities and research facilities who are employing new, cheaper technologies, and are hard at work thinking up and building the next several generations of electronic music controllers. How do you find these people? One way is to go to their conference: New Interfaces for Musical Expression, or NIME. This year's NIME, held at the University of British Columbia in Vancouver, was the fifth, and it offered plenty to see for the musicians and scientists who showed up from all over the Americas, Europe, and Asia. There are some very creative people out there — and some of them are really out there.

file:///F|/SoS/SoS%2001-2006/nime.htm (1 of 7)12/19/2005 10:24:00 AM


Keynote Speakers The tone of the conference was set by Bill Buxton, a Canadian who has been one of the pioneering forces in both computer music and video for the last 25 years, and who was one of the first to propose the concept of 'gesture controllers' as musical instruments at another conference in Vancouver over 20 years ago (see the 'Digicon 83' box over the page). At the start of his keynote speech he threw up a slide of an old Revox reel-to-reel tape deck and proclaimed, "This is the enemy." The problem with electronic music concerts historically, he said, is that someone would walk onto a stage, push a button on a tape deck, and walk off. Today, people do the same thing with laptops. "If you're sitting at a computer and typing, why should I care?" he lamented. If we're going to a live performance, he opined, we want to see the performer doing something interesting to create the music. "The goal of a performance system," he stated, "should be to make your audience understand the correlation of gesture and sound."

Photo: Bernardo Escalona Espinosa Music and video artist Bill Buxton on stage at NIME, 'then' (above) and 'now' (below).

Another keynote speaker was Don Buchla, who has been actively developing new electronic instruments for four decades. He presented a comprehensive history of electronic performance instruments, including many less well-known devices like the Sal-Mar Construction. Built in the late 1960s by composer Salvatore Martirano, it resembled a giant switchboard surrounded by 24 ear-level speakers, and had almost 300 switches so sensitive they could be operated by brushes. Buchla also discussed a number of his own designs, some of which made it to market and some didn't. One of these was an EEG-to-control-voltage converter that responded to alpha waves. "I did a performance in Copenhagen," he recalled, "but while I was up on stage, I found I couldn't generate any alpha waves, so I didn't make a sound for 15 minutes. It was the longest silence I've ever heard."



The Tools That Make This Possible As you may gather, there's a lot going on at the college level in the world of new electronic musical instruments, partly because the tools for building custom performance systems are continuing to become cheaper, more plentiful, and easier to use. On the hardware side, there's now a host of different sensors that respond to various physical actions, developed for hi-tech industrial, security, and medical applications, but easily adaptable to musical ones. These include force-sensing resistors (FSRs) such as you would find in MIDI drum pads, 'flex sensors' that change resistance as you bend them, like the fingers in a Nintendo Power Glove, and piezoelectric elements, which send a voltage when they're struck, flicked, or vibrated. There are also tilt switches, airflow sensors, accelerometers, and infrared and ultrasonic distance sensors. Slightly more expensive, but within the budgets of many education and research labs, are electromyography (EMG) sensors, which detect muscular tension. What's more, diagrams and tutorials for hooking these devices together are available in many locations on the Internet. On the software side, most of the presenters used one of two musical 'toolkits'. Max, for Macintosh, is the older and more sophisticated of the two, and was written originally in the 1980s by American mathematician Miller Puckette. Max, now called Max/MSP, is a commercial program, frequently updated by and available from the San Francisco company Cycling 74 (one of the sponsors of this year's NIME). A few years ago, Puckette released a freeware program for Mac, Windows, Linux, and Irix called Pure Data, which is effectively a simplified version of Max, although without any fancy user-interface tools or formal tech support — but at a price even university students can afford!

Instrumental Innovation The three-day conference was packed with stuff to do and see: there were three dozen papers and reports, five large interactive sound installations, and four demo rooms showing a wealth of gadgetry. There were also jam sessions, which bore more than a passing resemblance to the cantina scene in Star Wars: Episode IV, and each night there was a concert. The presentations were primarily about new one-of-a-kind instruments and performance systems, as well as new ways of thinking about performance. Some of the instruments were variations on conventional instruments. For example, Dan Overholt of the University of California Santa Barbara showed his Overtone Violin, a six-string solid-body violin with optical pickups, several assignable knobs on the body, and a keypad, two sliders, and a joystick where the tuning pegs usually go. In addition, an ultrasonic sensor keeps track of the instrument's position in space. It's played with a normal violin bow, but players wear a glove containing ultrasonic and acceleration sensors which allow them to make sounds without ever touching the strings. The whole thing is connected to a wireless USB system so performers don't have to worry about tripping over cables.



Then there was the Self-Contained Unified Bass Augmenter, or SCUBA, built by researchers at Stanford University. It starts with a tuba, adding buttons and pressure sensors to the valves. The sound of the instrument is picked up by a mic inside the bell, and sent through various signal processors like filters, distortion, and vibrato, which are user-configurable and controllable. The processed sound is then piped through four bell-mounted speakers and a subwoofer under the player's chair.

Photo: Bernardo Escalona Espinosa Golan Levin of Carnegie Mellon University demonstrating his projector-based Manual Input Sessions project on stage.

A hotbed of development for new musical toys has been the Media Lab at Massachusetts Institute of Technology. One of the more well-known of these is the Beatbug, a hand-held gadget with piezo sensors that respond to tapping. A player can record a rhythmic pattern and then change it while it plays using two bendable 'ears'. An internal speaker makes players feel they are actually playing an instrument. But Beatbugs work best in packs, as Israeli-born Gil Weinberg, who is an MIT graduate and now director of a new music-technology program at Georgia Tech, demonstrated in a new system called 'iltur'. In this system, multiple Beatbugs are linked to a computer through Max/MSP software, and produce various sounds from modules in Propellerhead Reason. The Max program allows interaction and transformation of the input patterns, encouraging the players to bounce musical ideas off each other. Some of the presentations used interfaces borrowed from completely different fields to produce musical sounds. Golan Levin, also recently of MIT and now at Carnegie Mellon University in Pittsburgh, Pennsylvania, described in his keynote speech his famous 'Dialtones: A Telesymphony'. In this project, some 200 members of the audience have their cell phones programmed with specific ringtones and are told where in the hall to sit. A computer performs the piece by dialling the phones' numbers in a pre-programmed order. Describing himself as 'an artist whose medium is musical instruments' while admitting that he can't read or write a note of music, Levin told the audience, "Music engages you in a creative activity that tells you something about yourself, and is 'sticky' — you want to stay with it. So the best musical instrument is one that is easy to learn and takes a lifetime to master." He also made the point that, "The computer mouse is about the narrowest straw you can suck all human expression through." As an example of a means of performing music that many non-musicians can master, Levin showed his 'Manual Input Sessions' project. Using an oldfashioned overhead projector and a video projector, the player of this system makes hand shadows on a screen, while a video camera analyses the image.



When the fingers define a closed area, a bright rock-like object filling the area is projected, and when the fingers open, the rock 'drops' to the bottom of the screen. As it hits, it creates a musical sound whose pitch and timbre are proportional to the size and the speed of the falling rock. It's fascinating to watch.

Digicon 83 For me personally, Vancouver has long been a symbol of dramatic change in the music industry. I was there once before, in 1983, at a meeting billed as the First International Conference on the Digital Arts, or Digicon, which was one of the most important events of its time. Before computers had taken over NAMM and Musikmesse, Digicon demonstrated many of the developments that would revolutionise every part of our industry, including the Sound Droid, an early (and never released) digital Photo: Paul Lehrman multi-channel editing and mixing system, and the video-processing Bob Moog prepares to demonstrate the revolutionary new technology 'MIDI' in technology behind the first longVancouver, 1983. form movie CGI sequence, the Genesis Project sequence from the then-recent Star Trek II — The Wrath Of Khan. Who was there? Bob Moog, Herbie Hancock, Bill Buxton (who first articulated the concept of 'gesture controllers' at Digicon), Todd Rundgren, and pioneering computer composers Barry Truax and Herbert Brün. And what was there? The first Fairlight CMI with hard disk recording, PPG's Wave, and pre-release models of the Yamaha DX7 and DX9. Oh yes: and a demonstration by Bob Moog, which caused jaws to drop in astonishment and delight all through the lecture hall, of a brand-new technology called 'MIDI'. You can read a full report on Digicon 83 at http://paul-lehrman.com/digicon.

Music By Any Other Name Some of the systems shown at NIME didn't require the 'player' to use any hardware at all. A group from ATR Intelligent Robotics in Kyoto, Japan, showed a system for creating music by changing one's facial expressions. A camera image of the face is divided into seven zones, and a computer continuously tracks changes in the image, triggering different MIDI notes in response. And then there was 'Bangarama: Creating Music With Headbanging', from Germany. This extremely low-budget project uses a guitar-shaped plywood controller. Along the neck are 26 small rectangles of aluminum arranged in pairs, file:///F|/SoS/SoS%2001-2006/nime.htm (5 of 7)12/19/2005 10:24:00 AM


which are used to select from a group of pre-recorded samples, most of them guitar power chords. Players wrap aluminum foil around their fingers so that moving them along the neck closes one of the circuits. The headbanging part consists of a coin mounted on a metal seesaw-like contraption, which is attached to a Velcro strip on top of a baseball cap. When players swing their heads forward, the metal piece under the coin makes contact with another metal piece in front of it, closing a circuit, which triggers the sample. Moving your head back up breaks the circuit, and ends the note. There was much, much more at NIME, but the last item I have room to mention here is McBlare, a MIDI-controlled bagpipe designed by Roger Dannenberg of Carnegie Mellon University as a humorous side-project. Based around a genuine set of bagpipes, McBlare incorporates a computer-controllable air compressor which will precisely match the breathing and arm pressure of a human piper, and a set of electrically controlled mechanical keys on the 'chanter' pipe. Under the control of an old Yamaha QY10 sequencer, McBlare could not only do a convincing imitation of a real piper, but could create frenetic, complex riffs no human could (or would ever want) to play. At the end of Dannenberg's perfomance, a wag in the audience (OK, it was me) asked him, "Why are bagpipers always walking?" and straight away, he responded correctly: "To get away from the sound!"

Glimpsing The Future The NIME conference wasn't the biggest, or the longest, but it was certainly among the most informative conferences I've ever attended. As happens at any good meeting of like-minded creative types, I made new friends, heard new theories and concepts, saw some amazing performances, and most importantly, left with my head buzzing with things to try out in my own work. The NIME conference is not for everybody — but all of us who work with music and electronics will be hearing from the people who were there in years to come.

Controllers & Games The market for new musical instruments is not an easy one to break into. Very few of the instruments shown at NIME will ever make it into commerical production. But there is a strong market for the ideas being presented, as one NIME speaker, Tina Blaine, a percussionist, vocalist, and inventor from Carnegie Mellon University, explained. Blaine's presentation was called 'The Convergence of Alternate Controllers and Musical Interfaces in Interactive Entertainment', which translates to 'Everything you're doing, the game industry can use'. There are a number of categories of games that have interactive musical components, she explained, and they are growing all the time. The most common are beat-matching games, but there are also games that require one of a variety of specific actions by the player in response to musical cues. And there are other games that make music when



players move their bodies or hands in free space, like Sony's Groove and Discovery's Motion Music Maker. But Blaine's most salient point for NIME attendees was that none of the game manufacturers develop their own controllers: all of their technology is licensed from someone else. And who better to develop that technology than the musicians and scientists who are tinkering away in their labs trying to build the next generation of electronic musical instruments? Published in SOS January 2006 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Using Sonar 5's Cyclone Loop Tool

In this article:


The Right Controller For Sonar Notes & Techniques Cyclone? Published in SOS January 2006 Using Cyclone As A Loop Machine Print article : Close window Sonar News Technique : Sonar Notes Cyclone As A Loop Warper Custom Waveform Previews Cyclone As A Drum Machine

The Cyclone 'groove sampler' DXi, bundled free with Sonar, is a powerful tool that allows loops of all kinds to be dissected, manipulated and generally bent to your will... Craig Anderton

Spectrasonics' Stylus RMX is one slick piece of software. The way in which it lets you experiment with loops and build rhythmic tracks, coupled with its massive library of material, is outstanding. Even better is how you can add filtering, envelopes and processing, the last being particularly interesting because you can actually get 'inside' the loop, and extract particular hits for treatment. But Sonar's own Cyclone provides many of the same functions — and it's free. Let's insert Cyclone and check out what it can do. When inserting, I strongly suggest you tick 'All Synth Outputs' in the Insert Soft Synth Options dialogue box (for reasons you should see shortly). You'll end up with 18 tracks in total, but if that's too much clutter, you can always stuff them into a Track Folder. Cyclone has several main areas of importance. The heart of the instrument is 16 pads (labelled '1' in the screen shot overleaf) that can access Acidised WAV files, one-shot samples or non-Acidised WAV file (if you use the last, files of different tempos won't sync together). Each pad can feed its own audio output and appear on a separate Sonar track, so you can process the output of each individually. The Pad Inspector (2) shows MIDI-triggering characteristics for the selected pad. Triggering can be restricted to single notes, note ranges (which also transposes file:///F|/SoS/SoS%2001-2006/sonarnotes.htm (1 of 8)12/19/2005 10:24:04 AM


what's on the pad) and velocity ranges. The Pad Inspector's Output field is where you assign each pad to a particular output. The Loop Bin (3) is where you can drag loops on to pads, delete loops, save them, audition them, and so on. If you click on a loop in the bin, the window in the middle of the screen (4) shows the loop's waveform, and also where it has been sliced into individual elements for time-stretching. In the Pad Editor (5), each slice appears as what Cakewalk call a Grain. This is a rectangle that can be dragged anywhere along the Pad Inspector's time line, or even to a different pad. When dragging, the Grains can be quantised to a specific rhythmic value in the Slice Inspector (6). You can also alter the pitch, level and pan position of each Grain. Cyclone really shines for three main applications: As a loop machine: When the pads are loaded with loops, you can do MPCstyle triggering and improvise bringing loops in and out of the mix. The fact that loops will sync up is vital for this kind of application. As a loop 'warper': Load a fairly simple loop into a pad and use the Pad Editor to drag specific hits to separate outputs. Thus the snare could go into a track processed by heavy reverb and the kick into a pad processed by distortion and delay, while the rest of the loop remains unprocessed. This is somewhat tricky to do, but fun. As a drum machine: Although the pads aren't velocity-sensitive, you can load one-shot samples on to each pad and trigger them. You can fake velocity sensitivity by placing samples recorded at different velocities on different pads and setting the pads to respond to the same MIDI note but different velocity levels. For example, with 16 pads you could have four velocity-switched snares, three levels of open hi-hat/closed hi-hat/kick, and three additional pads for toms and so on.

The Right Controller For Cyclone? You can trigger Cyclone's pads from a standard MIDI keyboard, but my favourite triggering option by far is M Audio's Trigger Finger (reviewed in SOS September 2005). In fact, when you use one, it almost seems as if it was developed in conjunction with Cakewalk specifically for this application (I checked and no, it wasn't). The Trigger Finger has 16 pads that are ideal for triggering the Cyclone pads, but there are also eight programmable knobs and four programmable sliders. As I rarely file:///F|/SoS/SoS%2001-2006/sonarnotes.htm (2 of 8)12/19/2005 10:24:04 AM


load more than eight loops or use more than eight tracks of Cyclone, I generally assign the knobs to track volume (more on this later) and use the sliders to control effects inserted in the various tracks. Another advantage of using the Trigger Finger is the Enigma librarian, which makes it easy to assign pads, knobs and sliders to whatever is needed by Cyclone. For not much more than £150, the Trigger Finger is a very worthwhile investment if you're a Cyclone fan.

Using Cyclone As A Loop Machine For MPC-style pad-triggering fun, first specify a separate audio output for each pad, so that each pad can be assigned to a single output for additional processing. 1. Drag the 'groove clip' files you want to use to the desired pads. This automatically places them in the Loop Bin. Or drag the loops into the loop bin, and from there on to the desired pads. 2. Click on a pad and use the Pad Inspector to assign its audio output and the MIDI note and velocity range to which it will respond. Do not re-assign the Root (root note) parameter, as it's assigned when the loop is imported. To keep things simple, set Key Unity, Key Low, and Key High to the same note so that only one MIDI note will trigger the loop.

Here's Cyclone's 'front panel', labelled to show the main areas of interest: Pads (1); Pad Inspector (2); Loop Bin (3); waveform window (4); Pad Editor (5); Slice Inspector (6). See the main text for more detail on these features.

3. While you're in the Pad Inspector, turn the Latch function on if you want the pad to toggle (trigger once to turn on, once to turn off), or off if you want the pad to sound only for as long as it is being triggered. A handy shortcut for turning on the Latch function for all pads is to hold down the Shift key as you enable Latch for any pad (this also works for 'unlatch'). Leave Tails and Pitch Markers off for now. 4. Repeat steps two and three for each pad. 5. On each pad you're using, click on the sync and loop buttons (which look like an analogue clock face and a bent arrow). This ensures that the loops will all be in sync and that they will for loop as long as they're being triggered. 6. If you want to control track volumes with hardware knobs, such as the knobs on the Trigger Finger (see box on previous page) or other control surface, rightclick on a track volume control and select Remote Control. Move the knob, click file:///F|/SoS/SoS%2001-2006/sonarnotes.htm (3 of 8)12/19/2005 10:24:04 AM


on Learn, then click on OK. Note which controller numbers you used (as indicated in Cyclone's Controller field when you clicked on Learn) to control various parameters, as you'll need this information in step two of the next section.

Each pad has several controls. The top knob controls volume and the lower knob panning. Buttons along the bottom strip are (from left to right) 'open sound in pad', mute, solo, loop sync and loop on.

7. Save your settings as a Cyclone file by clicking on the Save button in the Cyclone Toolbar. (Note: I assume that this is a bug, but when you call up a Cyclone file, the first file in the loop bin is placed on all the pads. I've found no workaround other than dragging files back on to pads when opening a project.)

To record the results of playing with Cyclone, enable recording on its source MIDI track. Controller motions will be saved as controller data within the MIDI track, but on playback they will not control the automation because they are not automation signals. Fortunately, there's a way to convert MIDI signals into automation signals. 1. Select the MIDI track driving Cyclone by clicking on its track number so that the entire track is selected. 2. Go Edit / Convert MIDI to Shapes. For the Type field, select Control; for Value, enter one of the controller values you used in the last step-bystep procedure; ignore the Channel parameter, then click on 'OK'. 3. The MIDI Controller curve has now been converted into a shape with nodes. Click on the MIDI track number to deselect the track, then click on it again to select it. This 'resets' the track so it recognises that the controller is now track automation instead of a MIDI controller.

A simple drum loop has been loaded into pad one. Note the waveform display in the middle with the 'slice' markers, and the representation of the loop in the Pad Editor toward the bottom. Each slice is shown as a 'Grain' in the Pad Editor.

4. Go Edit / Copy. In the Copy dialogue box, tick Track/Bus Automation only, then click on OK. This copies only the 'shape' envelope. 5. Set the Now time at the beginning, then click on the track in which you want the envelope to end up. 6. Go Edit / Paste, and under 'What to Paste' (you may need to click on the Advanced button to see this), tick only Track/Bus Automation. Click on 'OK'. The file:///F|/SoS/SoS%2001-2006/sonarnotes.htm (4 of 8)12/19/2005 10:24:04 AM


envelope is copied into the track. 7. Right-click on the envelope, and go Assign Envelope / Volume. The envelope will now control the track volume. 8. Return to the MIDI track, right-click on the shaped envelope, then select Delete Envelope. 9. Repeat steps 1-8 for each of the other controllers contained in the MIDI track.

Sonar News OK, Sonar 5 is the big news. But maintenance version 4.0.4 has been released for Sonar 4, so if you're still waiting to upgrade this should help to tide you over. Note that version 4.0.4 will update only version 4.0.3, so if you're running an older version of 4, you'll need to proceed through the upgrade path. The biggest improvement fixes the clip-envelope editing anomalies that could occur under a specific set of circumstances involving clips inserted after a tempo change, where nodes were added before the Now time (assuming that the Now time was over the clip). Another fix for a problem often pointed out in internet forums is that 'Split Repeatedly', under the Edit-Split command, now works. Previously, muting a cropped clip with the Mute tool meant that the clip would not be included in an export or bounce operation, and this problem has now been solved. There's also a fix for the rare but annoying problem of tracks in playback mode dropping out while new tracks were being recorded. There are several other fixes, many relatively minor, as detailed on both the Cakewalk web site and in the Read Me file that can be viewed after installation. And while you're in an update frame of mind, if you're using Windows XP SP2 with Sonar 4, download the Windows XP SP2 Hotfix from the Downloads / Patches / Updates section. This prevents a memory leak in your PC that could cause your system to run out of resources over an extended period of time. For more information on the fix, go to http://support.microsoft.com/default.aspx? scid=kb;enus;319740

Cyclone As A Loop Warper Suppose you have a great drum loop. Or, it would be a great drum loop, if only you could add a cool reverb splash on the snare, distort the kick a bit for a tougher sound, and while you're at it, add a bit of delay on the kick too. But you can't, because it's an audio file that's already mixed, looped and Acidised...so there's nothing you can do about it. Right? Wrong — at least if the loop isn't too complicated. Let's take a simple trance-type drum loop as an example (see the screen on the left). In the Pad Editor, we can drag the 'Grains' of the loop to other pads. For

file:///F|/SoS/SoS%2001-2006/sonarnotes.htm (5 of 8)12/19/2005 10:24:04 AM


example, there's a kick sound on beats one, two, three and four, along with a snare hit on beats two and four. Unfortunately we can't isolate the kick and snare, but we can take kick beats one and three and drag them to pad two, and beats two and four and drag them to pad three. Next, assign pad one to channel one, pad two to channel two, and pad three to (no surprise!) channel three. Each now feeds a separate track in Sonar and we can add individual processing to each of the tracks. In this example, the kick on track two has been processed with the Cakewalk FX2 Tape Sim to give it a harder sound, plus delay to add some bounce. The snare, track three, is processed by the Pantheon reverb set for a nice long splash (see screen below). Even though this may seem pretty cool — and it is — there's even more you can do. Click on a Grain and use the Slice Inspector to alter its pitch, level and/or pan. For example, you could split the two snare hits so that one is panned left and the other panned right, or drop the first kick hit down a couple of semitones and raise its gain for a massive downbeat. And did you notice the little white arrows on measure five in the screen shot below? They indicate the loop end and can be moved. For example, move the snare loop end from measure five to halfway through measure four and the snare will do an interesing off-rhythm effect. MIDI-wise, you can either leave the pads unlatched and have long MIDI notes that trigger the pads and leave them on, or latch the pads and just add MIDI notes where you want the pads to turn on or off.

Custom Waveform Previews The waveform preview function, wherein Sonar 5 draws a waveform for busses in real time or while you're recording in tracks, is tremendously helpful. I've now become thoroughly dependent on drawing the master buss preview, as it lets me see where any overloads might be occurring that would affect the two-track mix. I also use it with aux busses, to obtain a 'reality check' on the signal waveform going into any signal processors or external audio outputs. However, it's possible to have too much of a good thing. If you have a slower computer with less-than-wonderful graphics abilities, drawing all those waveforms in real time can affect performance. If you do experience problems, you'll be glad to know it's possible to change the rate at which the preview graphic is updated. Here's how: 1. Locate the CAKEWALK.INI file in the Sonar 5 folder (typically under C: \Program Files\Cakewalk). 2. Open the CAKEWALK.INI file using Notepad. 3. Under [WinCake], enter the following: WavePreviewSampleFrequency=10 4. Save CAKEWALK.INI Entering 10 sets the longest possible update time. One, the default, is the shortest update time. You can set any number between one and 10; experiment to see



which value gives the best compromise between smoothness of preview drawing and overall computer performance.

Cyclone As A Drum Machine Although this is the least sexy of the Cyclone applications — a plug-in like NI's Battery 2 runs rings around it, because Cyclone's pads aren't velocity-sensitive — Cyclone is still a good instrument for triggering one-shot samples. As mentioned earlier, you can spread sounds with different velocities over multiple pads, trigger them from the same note and restrict each pad to a specific velocity range. It's a bit clunky, but it works. Along the same lines, Cyclone is good for firing off sound effects, especially since the Slice Inspector lets you mess around with each effect. And for bass lines, you can take a single bass sample, apply it to each pad and use the Inspector to create different pitches. That's not all. One of the most useful The loop warping is now complete. Two things when you're using Cyclone to kicks are isolated on pad two, two snare hits play back one-shots (and here you're on pad three and the remaining beats on pad much better off with a conventional one. Note that in Sonar's mixer tracks three keyboard rather than a controller such and four (channels two and three) have their own processors. as the Trigger Finger) is that each pad can be triggered by a range of notes (Key Low to Key High in the Pad Inspector), and the sound is transposed according to those notes. For example, a single tom sound could be triggered over an octave of notes, giving you 12 tuned toms. When they're assigned to a conventional keyboard, you could (for example) pitch the kick over half an octave, the snare over half an octave, and so on. This helps to make up for the lack of velocity by at least offering pitch versatility. Cyclone really is a terrific instrument, whether you use it with Sonar or Project 5. I've never understood why Cakewalk didn't make the pads velocity-sensitive, allow you to control pad level and pan with MIDI controllers, and sell it as a VST instrument to serve as a 'virtual MPC'. Now that I've mentioned it, maybe they will! Published in SOS January 2006






Using The Logical Editor In Cubase SX & SL

In this article:

Not Quite Aristotle Cubase News 15-To-1 Countdown

Using The Logical Editor In Cubase SX & SL Cubase Notes & Techniques Published in SOS January 2006 Print article : Close window

Technique : Cubase Notes

Cubase contains many powerful features for processing MIDI data, such as the Logical Editor and Macros. This month we look at how using them together can create some powerful solutions to potentially tedious problems. Mark Wherry

The Logical Editor has been a part of Steinberg's sequencing software products since the days of Pro 24 on the Atari, and its latest incarnation in Cubase SX/SL/SE remains a powerful way to process MIDI data. We've covered an introduction to the Logical Editor in previous editions of SOS, so for a complete refresher, you might want to check out May 2003's Cubase Notes at www.soundonsound.com/sos/ may03/articles/cubasenotes0503.asp. The Logical Editor is made up of three basic However, I'm going to start this month's parts: the Function Menu, the Filter Condition Cubase workshop with a brief overview List, and the Action List. The black status lines underneath the two List boxes show an of the Logical Editor and cover a few overview of the expression you're describing points not described in the previous with the commands in the respective List, indicating an error if you make a mistake. article, in addition to covering some of the same ground for new users. The first example will also be the same, so beginners can read a more in-depth explanation, making this article backwardly compatible if you will(!); after that we'll look at some more complex examples.

Not Quite Aristotle file:///F|/SoS/SoS%2001-2006/cubasenotes.htm (1 of 7)12/19/2005 10:24:08 AM


The basic principle of using the Logical Editor window is fairly straightforward. As with most off-line MIDI processing operations in Cubase, you can use the Logical Editor to process MIDI Parts in the Project window, and in this case you need to specify the MIDI Parts you want to process by selecting them and choosing MIDI / Logical Editor to open the Logical Editor window. Bear in mind that the Logical Editor option will be disabled in the menu if the MIDI Parts you've selected don't contain any MIDI Events to be processed. You can also use the Logical Editor to process MIDI data in a MIDI editor window. If you have an open MIDI editor displaying MIDI Events with nothing selected, the Logical Editor will process all MIDI Events in the MIDI Part (or Parts if you're displaying multiple Parts in the editor window and have Edit Active Part Only disabled on the editor's toolbar). If you have Events selected in the MIDI editor, only these Events will be considered by the Logical Editor. The Logical Editor window contains three main elements: the Function Menu, the Filter Condition List, and the Action List. The Filter Condition List specifies which events should be processed, the Action List specifies how the events should be processed, and the command selected in the Function Menu defines the mode of operation for the Logical Editor and precisely how the Filter Condition and Action Lists should be used. There are seven different commands available from the Function list: Delete erases all events specified by the Filter Condition List. Transform is the default (and most commonly used function) and modifies the events in the Filter Condition List by the actions defined in the Action List. Insert is similar to Transform, but instead of the filtered notes being modified to produce a different Event, the original Events are kept and the transformed Events are created as new, separate Events. Insert Exclusive is, again, based on Transform, except that notes that do not meet the conditions specified in the Filter Condition List are deleted. Unlike the standard Insert function, this one creates no new Events. Copy is like Insert, but instead of the newly transformed Events being created in same Part on the same track, the new Events are added to a new MIDI Part on a new MIDI track. Extract is similar to Copy, except that any Events matching the Condition List are deleted and the transformed versions are created in a new MIDI Part on a new track. Select selects any Events that match the Filter Condition List (and if any Events were already selected, deselects any that don't match the condition); the Action List is ignored. file:///F|/SoS/SoS%2001-2006/cubasenotes.htm (2 of 7)12/19/2005 10:24:08 AM


The Filter Condition List is made up of a series of Lines, where each Line represents one expression that is part of the overall condition (if you have multiple Lines). A Line basically consists of a Filter Target, a Condition, and some data that is in one or more of three columns: Parameter 1, Parameter 2 and Bar Range, although Here is the Logical Editor window showing these data columns are never used the preset for transforming the second simultaneously. The Filter Target is a velocity layer, as described in the main text. general category, such as a type of Event, the first data property, the length, and so on; the Condition is what evaluates the Filter Target against the data, such as Equal, Unequal, All Types, and Greater Than; and, finally, the data column specifies what the Filter Target should identify. For example, if you were looking for all notes, set Filter Target to Type, Condition to Equal, and Parameter 1 to Note. Each Line in the Filter Condition List can only evaluate one Filter Target against one element of data, so if you wanted to find all notes equal to C3, you couldn't simply add a note pitch in the Parameter 2 column of the Line I just described. To do this would require two expressions — type is equal to note, and pitch is equal to C3 — so you'll need two Lines in the Filter Condition List. Add another line by clicking the Add Line button next to the upper List and then, on the second Line, set the Filter Target to Value 1 (which should automatically change to Pitch in the Line), make sure Condition is set to Equal, and set Parameter 1 to C3. The Action List is very similar to the Filter Condition List in that it comprises a series of Lines that tell the Logical Editor what to do with the Events identified by the Filter Condition List. Each Line is again split up into several columns: an Action Target, an Operation, and some data (Parameter 1 and Parameter 2). Action Target specifies which part of an Event should be processed, whether it's the type, a property or value (such as pitch); Operation sets what you want to do to the Action Target, such as add, set it equal to something, and so on; and the data specifies what the Operation applies to the Action Target. For example, to transpose found notes up an octave, the Action Target would be Value 1, Operation would be Add, and Parameter 1 would be 12 (semitones). Additional Lines can be added to (or deleted from) the Action List, but bear in mind that while each Line in the Filter Condition List is an expression of one overall condition, each Line in the Action List operates independently, one after the other, like a recipe. So if you had two Lines that were both 'Value 1 Add 12', this would be the equivalent of one Line that states 'Value 1 Add 24'. Once you've set the Function of the Logical Editor, specified the Events to operate on with the Filter Condition List, and set up what happens to the identified Events with the Action List (unless you've chosen Select as your file:///F|/SoS/SoS%2001-2006/cubasenotes.htm (3 of 7)12/19/2005 10:24:08 AM


Function), you press the Do It button for the Logical Editor to work its magic.

Cubase News Of interest to Cubase users this month will be Steinberg's announcement of Hypersonic 2, an update to the popular virtual workstation released by Steinberg and Wizoo at the end of 2003 (www. soundonsound.com/sos/feb04/ articles/steinberghypersonic.htm). Although the original point of Hypersonic was to be an economical all-purpose sound source that you could get good Hypersonic 2 features an updated user results with even if you were using interface that allows full access to all the an older computer system, many parameters of the underlying sound engine. people wanted the gigabyte sound library to compete with products like IK Multimedia's Sampletank. So Hypersonic 2 comes with a 1.7GB sample library (up from the original 250MB content) with 800 new patches and revised versions of the original 1000 presets supplied with Hypersonic 1. Hypersonic 2 also includes Hyperphase, a rather good polyphonic arpeggiator that includes 200 phrases, and the ability to import your own as a Standard MIDI File. What's more, the updated user interface makes it possible to edit every parameter in Hypersonic's different sound engines, which you couldn't do in the previous version. One user interface change I didn't like so much was that the semi-circle value displays in the Mix page of version 1 have been replaced with more traditional horizontal rectangular bars, which personally I didn't find so effective — but if that's the only gripe I have, it's probably not a serious criticism of the product!

15-To-1 I recently encountered a curious problem when a composer I was working with needed to use newer versions of some multisampled orchestral instruments, featuring many velocity layers, which were being played back via MIDI from various Cubase Projects. The composer in question is a meticulous programmer and had balanced the velocity layers very precisely so the appropriate samples within the instrument were triggered; but in the newer versions of the instruments the velocity ranges had changed, even though the samples were the same. So the problem was how to rebalance the velocities used for notes that had been programmed with one version of the instrument so they would play back using the same samples in the new versions. And since the Cubase Projects that needed altering were large and numerous, this was definitely a job for the Logical Editor.

file:///F|/SoS/SoS%2001-2006/cubasenotes.htm (4 of 7)12/19/2005 10:24:08 AM


One of the instruments in question had six velocity layers and the old version had these velocity layers programmed with the following splits: 0-30, 31-52, 53-74, 7598, 99-113 and 114-127. And as I explained, the new version had the velocity layers programmed slightly differently, so the new splits were as follows: 0-80, 8190, 91-100, 101-110, 111-120, and 121-127. Cubase's Logical Editor is a great way of processing MIDI data, but its Transform feature is only capable of transforming one condition into one result. An example of one condition would be 'find all the velocities between 0 and 30' and an example of one result would be 'scale these velocities to 0 and 80'. If I had multiple conditions for different velocity layers, they'd all have to be processed by the same Action List, so all velocities would end up between 0 and 80, which would be useless. Therefore, each velocity layer transformation has to be dealt with using a separate Logical Editor operation; so what we'll end up learning here is how to combine multiple Logical Editor operations into one command to create more powerful (and convenient) Logical Editor commands. The first step is to open the Logical Editor window by selecting MIDI / Logical Editor. Start by selecting the Init Preset (click on the Presets pop-up menu and choose 'init') to make sure any previous operations in the Logical Editor are reset, and make sure that the Transform Function is selected. The first Line in the Filter Condition List should read '(Type Is Equal Note)', and next we need to set the range for the incoming velocities by adding two further Lines to the List, clicking the Add Line button twice. Once you've created Logical Presets, you

Set the Filter Target for the lower two can combine them together into one Lines to 'Value 2' by clicking command by creating a Macro in the Key Commands window. underneath the column in the appropriate row and choosing 'Value 2' from the pop-up menu. Once you make the selection, you should notice how Cubase will display this entry as Velocity in the Line. Next, set the Condition on the second row to 'Bigger or Equal' and the third row to 'Less or Equal', and enter the velocity boundaries for the first velocity layer (0 and 30) as Parameter 1 on the second and third rows. Now the Filter Condition List has been set, we need to set up the operation to scale velocities between 0 and 30 to 0 and 80 instead. To do this, on the first Line of the Action List, Action Target should be set to Value 2, Operation to Multiply by and Parameter 1 to 2.67. To get this scaler value (2.67) you divide the range of velocities in the destination (80) by the number of velocities in the source (30), and I rounded the result to two decimal places, which is easily accurate enough for the seven-bit integer values used to describe velocity in the file:///F|/SoS/SoS%2001-2006/cubasenotes.htm (5 of 7)12/19/2005 10:24:08 AM


MIDI protocol. And that's it. Store the Preset by clicking Store, entering a name, and clicking OK, and we have a Logical Editor operation that can scale velocities between 0 and 30 to 0 and 80. Creating an operation to scale the next velocity layer is almost the same: the process of setting up the condition is identical, except Parameter 1 in the lower two rows should now be 31 and 52 to represent the range of the second velocity layer. The most significant change this time is that the Action List is a little more complicated, because we need to first subtract the lowest velocity (31) from the velocity of the note we're processing, so we can scale the velocity correctly, and then add the value of the lowest velocity for the new instrument to this number so that it's put back into the correct range again. Therefore, the Filter Condition List should now consist of three Lines: 'Value 2 Subtract 31', 'Value 2 Multiply by 0.43', and 'Value 2 Add 81'. 0.43 is the scaler because it's the division between the range of velocities in the old second later and the range in the new second layer ((90-81=9)/(52-31=21)=0.43). So now we have two Logical Editor Presets to deal with the first two velocity layers, and by substituting the numbers from the remaining velocity layers you should be able to build presets for the other transformations, as they're all based on the second layer Filter Condition and Action lists described in the previous paragraph.

Countdown Once you've finished, you'll have ended up with six different velocity layers with which to process each MIDI note where you want to rescale the velocities. To execute all these operations simultaneously you can build a Macro (which we covered in January 2003's Cubase Notes: see www.soundonsound.com/sos/ jan03/articles/cubasenotes0103.asp), as Logical Editor Presets appear in the Process Logical Preset Key Commands category. The important thing is that the Presets are run in reverse order, so you process the highest velocity layer first — otherwise, each Preset would operate on the results of the previous operation, so a small velocity could end up scaling to 127 if you weren't careful. By running the Presets in reverse order, there's less chance of this happening. So to build the Macro, open the Key Commands window (File / Key Commands), click Show Macros, click New Macro, type in a name for the Macro and press Enter. Find the Logical Presets you created in the Process Logical Preset category in the Commands list and select the first Command to add to the Macro (the last Preset you created, the one to process the highest velocity layer). Now, click the Add Command button to add that Logical Preset to the Macro, and then repeat this process for the rest of the Logical Presets. When you've finished, you can assign a Key Command to the Macro itself if you want, or just click OK to close the Key Commands window.



If you haven't assigned a Key Command, you can run the Macro by selecting it from the Edit / Macros submenu — just remember to have some MIDI Event selected before you run the Macro, and also to make sure the Logical Editor window itself is closed. Triggering Logical Editor Presets via Key Commands or Macros with the Logical Editor window open will select those Presets for editing in the Logical Editor rather than actually running the process on available MIDI data. Hopefully this will have helped you to get a better handle on the Logical Editor, and will have made you think about the possibilities of combining Logical Editor Presets with Macros. Join me next month for some action in the exciting world of Boolean operators... Published in SOS January 2006 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



Working With Video In Pro Tools

In this article:

Quicktime Movie Tips Opening Windows Decisions, Decisions Useful Links Video Latency Importing Video The Upgrade Dilemma

Working With Video In Pro Tools Pro Tools Notes & Technique Published in SOS January 2006 Print article : Close window

Technique : Pro Tools Notes

This month's Pro Tools workshop is the first in a series where we will explain how to use Digidesign's DAW to work to picture. First of all, we look at the decisions you need to make in setting up your system. Mike Thornton

There are three basic ways to work to picture in Pro Tools. First, you can lock Pro Tools via time code to an external video playback machine and then Pro Tools can 'chase' the VT machine. This process doesn't put any extra load on the computer but it is slow, because you have to wait for the VT machine to spool backwards and forwards and cue to the correct position on the tape before you can work on a section. You will also need some sort of timecode synchroniser like Digidesign's Sync I/O.

Pro Tools can play back Quicktime movies in its Movie window, but only at the original resolution specified in the Quicktime file.

Second, you can import video files directly into a video track in Pro Tools. This is fast, because there's no waiting for the VT machine to catch up: Pro Tools can continue to work in a 'non-linear' way and will jump to the correct part of the video file as you move the cursor around the Session, just as it does with the audio files. However, handling the video file puts a load on the computer, and you will find that with high track counts and plug-in counts, Pro Tools is more sluggish when using a video file as your picture source. Third, you can use a separate non-linear video player. This can be either a second computer running an application like Virtual Video Tape Recorder from Gallery Software, or a dedicated non-linear player like Rosendahl's Bonsai Drive. file:///F|/SoS/SoS%2001-2006/ptworkshop.htm (1 of 8)12/19/2005 10:24:14 AM


This has the benefits of both of the first two options, with the only down side being cost. In this series, however, I'm going to concentrate on the second option, as it is both the easiest one to implement and the most affordable. Pro Tools, on both Mac and PC , can use any type of movie file you like as long as it is a Quicktime movie! However, you have to be careful which codec is used to create the Quicktime movie — see box above for details.

Quicktime Movie Tips Pro Tools can handle most QuickTime movies, but not all. If the video card used to create the QT movie has hardware data compression, Pro Tools will be unable to play the movie unless the correct card is available in your computer. This tends to make taking the Session from system to system difficult unless they all have the same video card. Second, when exporting from Avid Xpress DV, always use the Apple DV codec in the Export dialogue box. Do not use the Avid codec — if you do then it won't play in Pro Tools. Third, Pro Tools currently doesn't support DV stream files, so make sure you have a DV movie and not a DV stream.

Opening Windows The great news is that without any additional hardware Pro Tools will play and display a Quicktime movie in a special movie window on your computer screen. The size of this window is set by the pixel size of the Quicktime movie, and cannot be changed in Pro Tools, which is where Quicktime Pro comes in very handy. It costs £20 to unlock Quicktime Player, which enables you to carry out numerous tasks such as video resizing, file conversion and so on. You can buy the 'unlock' code for Quicktime Pro for Windows or Mac OS from the Apple web site. If you prefer to view your movie on a separate screen or video monitor, there are several options available to you (but see the 'Video Latency' box). First, you can use a DV Firewire bridge transcoder to play out your movie via a Firewire port, through the transcoder, and into a video monitor. Digidesign approve the Canopus ADVC 100 and the newer ADVC 110 model, but for this to work the movie files must be DV movies. No other type of codec can be used, and if you have a movie in any other format you will find that the Play DV Movie Out Firewire Port menu option in the Video menu in Pro Tools is greyed out. Second, if you have a Mac and a spare PCI slot, Digidesign have approved the use of the Aurora Igniter LT/IgniterX Lite video capture card. You can use this to route your movie file out to a video monitor, and it gives you a broader choice of movie codecs that you can use. The third option is the use of a Miro Motion DC30 or DC30+ video capture card. Although these are not approved by Digidesign, a number of users have found this option to work very successfully

file:///F|/SoS/SoS%2001-2006/ptworkshop.htm (2 of 8)12/19/2005 10:24:14 AM


using freeware OS X drivers from Squared 5, and Miro cards can be picked up very cheaply second-hand.

Decisions, Decisions In terms of the materials needed to work to picture in Pro Tools, you will encounter three basic elements, some or all of which will need to be delivered to you. First, you'll obviously need the movie file. Second, you'll want an EDL (Edit Decision List) from the video edit, if one exists. This is a list that records how and where the video was edited. Third, there is what's called 'sync audio', if any exists for the project you're working on. Adverts, for instance, are often shot 'mute', with all sound added in post-production. This workflow has been made a lot simpler with the introduction of electronic delivery of these elements. Before, you would get the video and audio on tapes which would then need digitising and loading into your audio editing system, and then the EDL (which often used to come on a floppy disk) would be used to 'conform' the audio material to match the video edit against the original timecode data from each edit. This process is rarely plain sailing, and is thankfully less common these days.

Useful Links www.bonsaidrive.com www.gallery.co.uk www.apple.com/uk www.canopus-uk.com/ www.auroravideosys.com/ www.alfanet.it/squared5/dc30xact.html www.pharoahaudio.com/ www.kenstone.net www.avid.com/content/7697/OMF v 1.0.doc www.northbeachpost.com/ audio_primer/omf_export_for_avid.html

The EDL now more often comes in the form of an OMF (Open Media Framework) file: there are other formats but this is still the most common one. These are interchange standards that enable the user to import, edit and export information to and from different brands of editing station, whether video-toaudio, audio-to-audio or audio-to-video. OMF files can come in two types, 'embedded' or 'referenced'. An embedded OMF is one large file that includes all the audio used in the project and the EDL consolidated into one file. A referenced file is where the OMF file is simply the EDL data, which then points to all the individual files, in the same way as a Pro Tools Session does. The down side of the referenced format is that it is very easy to lose a few files in the transfer process, and I always ask for an embedded OMF file. These can easily be created from video workstations like file:///F|/SoS/SoS%2001-2006/ptworkshop.htm (3 of 8)12/19/2005 10:24:14 AM


Avid or Final Cut Pro. If the video editor is unsure on how best to handle the OMF export, there are loads of help guides available on the Internet. For example, for a guide for exporting from FCP see Ken Stone's site, or for exporting from an Avid have a look at Avid's own site (see the 'Useful Links' box). These files tend to be delivered on DVD ±R or on a portable hard drive as the DV video files can be quite large — around 200MB per minute of video. My last project came in at over 50GB of data! There may be situations where you will still receive the video on a 'tape', whether it is VHS or DV or Beta. You will then need to capture this video from a player into the computer and so create your own Quicktime movie. Use If you want to work with non-linear digital your preferred video editing application video but need to take the strain off your Pro Tools machine, options include Rosendahl's to capture the video. Mac users will probably have iMovie somewhere, as it Bonsai Drive and Gallery Software's Virtual VTR (running on a second computer). comes free with most Macs, or you can use Final Cut Pro — you don't need the latest version and so you can pick up a second-hand copy for a song. Make sure you export it as a DV Quicktime movie, not a DV stream, and all should be fine. On Windows I would recommend getting Quicktime Pro and using that to convert the video files from, say, Adobe Premiere to Quicktime format. If you get your video on a DVD Video disc, you will have to rip the disc and then convert the results into a Quicktime movie, which is a slow process. Wherever possible, then, get the client to supply a Quicktime movie as a video file. Pro Tools will be happy and it won't take you forever to get the video file into the correct format. Note that if the file is delivered to you on a DVD±R, you'll need to copy it off onto a hard drive before trying to import it into Pro Tools. Put the DV movie onto a different drive to your audio files — your boot drive will be fine if you don't have another drive other than your audio drive. While you're on the 'phone to the client, try to make sure they send video with 'burnt-in' timecode. This is where the timecode is overlaid, in numbers, on the video so at any point you can see the exact timecode point. This is very useful to check that the video file matches the Pro Tools timeline by going to the end of the video track and comparing the burnt-in timecode with the Pro Tools counter.



Video Latency If you use any external display to view video files in Pro Tools, it's important that you take into consideration video latency — the delay between video being output by Pro Tools and appearing on a screen. This will be different depending on which method you use to view your video file and what device you use to view it with. For example, some people use video projectors, which tend to have a higher latency than other display devices. Why does this matter? Well, if you are trying to get things in sync with the video and the video you are viewing is delayed, then all your carefully positioned 'hit points' will be out of sync on the finished product and you will have a very unhappy client on your hands! Note that video latency is an issue with both TDM and LE systems. Pro Tools has a feature in the Movie menu where you can compensate for this latency to make sure everything is bang in sync, but first you need to know how much compensation to apply! There are some extremely useful documents on the amounts of latency on the different systems at the Pharoah Editorial site (see 'Useful Links' box), under the Products/Papers section. Richard Fairbanks compares all the different options outlined above with their relative performance and has been very supportive on the Digidesign User Conference on these matters.

Importing Video Importing a video file into Pro Tools is simple. Select Import Movie... from the Movie menu in Pro Tools, navigate your way to the location of the movie file. Click on the Open button and Pro Tools will create a video track and open the Movie window. The size of the Movie window will depend on the pixel size of the movie and the resolution of your computer screen. It is not possible to have 'full screen' video from within Pro Tools unless you dedicate one computer monitor to it and set that monitor's resolution to a size in pixels similar to that of the movie. For instance, for a standard 4:3 TV movie, the movie file is 720x576 pixels, so by setting the screen resolution of your second monitor to 800x600, the movie will almost fill the screen. For most purposes it's best to route the video out of the computer to a video monitor (see above) and so view the movie on a screen designed for the job — this could be a domestic TV using the AV input, so it doesn't need to break the bank. You can set the video track to show either blocks or frames, which displays thumbnails along the track in the Edit window. Although this can help navigate around the movie it does put extra demand on the computer, so I tend to leave it in blocks mode. You can also hide the video track using the Show/Hide menu, to free up both computer resources and screen real estate. When you import a video file into Pro Tools it will place the start of the video file at the start of the Session, but the timecode in the video file might not start at zero. You can compensate for this by changing the Session Start time in the Session Setup Window to match the start time of the video file. Some people like file:///F|/SoS/SoS%2001-2006/ptworkshop.htm (5 of 8)12/19/2005 10:24:14 AM


to have some space before the video file starts for line-up tone and so on. No problem: just set the Session Start time back, say, 1 minute. You can then use the Spot tool to position the start of the video file at the correct timecode point. You can choose whether Pro Tools' video

Now you may well be asking: so how track displays your video file as Blocks (top) or Frames. The latter is helpful, but demands does all this stuff actually help me do more of your computer. something useful with my Pro Tools system and working to picture? Well, this month we've covered what you need to get to a position to start work. In the rest of this series we will look at how you can use Pro Tools to handle the nuts and bolts of audio post-production for TV documentaries, drama and films as well as composing music to picture. We will also be exploring how it is possible to work to picture with nothing more than a laptop, an M Box and Pro Tools LE even though there is no Timecode ruler in PTLE, and also how Digidesign's DV Toolkit package can help. If you want to go into the topic in even more detail, you might want to take a look at Ashley Shepherd's book Pro Tools For Video, Film And Multimedia (Muska Lipman: ISBN 159200069X).

The Upgrade Dilemma A few days after having ordered the Pro Tools 7 upgrade for my HD system I was checking the Digidesign web site for compatibility information, to find to my horror that my 933MHz G4 Mac was no longer on the list! What's more, Digi have issued a very clear statement recommending against any of the Quicksilver models from 2001/2002. I had expected I would need to upgrade some of my plug-ins and upgrade from Mac OS 10.3.8 to Tiger, but I certainly hadn't bargained on having to replace my computer at this point! So what computer to upgrade to? There are a couple of things that complicate matters. All Macs are now being produced with PCIe slots instead of old-style PCI slots except for one model, and from June 2007 Apple are going to move to Intel rather than Power PC processors. Digidesign have announced they will be producing PCIe-compatible HD Core and Accel cards, with a crossgrade route for existing users. However, these will be identical in power to the existing cards, so I was reluctant to spend the money for no gain in performance. What's more, even this system would only be an stopgap: my guess is that within 12 months Digidesign will have new cards that will only be available in the PCIe format and Apple will have Intel-based Macs. Apple have stated that Power PC models won't be dropped from the range until late 2007, but again Digidesign are surely going to put their efforts into developing new software and hardware for the Intel-based Macs with PCIe slots in them. So bearing in mind what the future holds, what's the best option now for a system that can run Pro Tools 7? It seemed to me that there were five options to choose



from: 1. Stop at Pro Tools v6.9 running on my G4/933 until the world settles. 2. Fit a processor upgrade to my existing G4/933 so, unofficially at least, it would be fast enough to run Pro Tools 7. 3. Buy a second-hand 'Mirror Door' G4 Mac that is on the Pro Tools 7 approval list and sell my G4/933. 4. Buy a new G5 with PCI slots before stocks run and sell my G4. 5. Buy a new G5, crossgrade my Digidesign cards to PCIe ones when they come out, and sell my G4. Option 1 would cost me nothing, but would build a brick wall around my facility that will get higher and higher as the world moves onwards outside. Option 2 would cost between £300 and £500 depending on what sort of card I got, but it would still be a G4 processor and of course totally unapproved by Digidesign. I did some research and found mixed reports as to the success of processor upgrade cards and Pro Tools — and of course no one was able to report success with Pro Tools 7 so I would be completely on my own there. Option 3 would seem to cost around £500 to £700 for a well specified G4 Mirror Door machine, to which I would probably need to add some more RAM and buy a copy of Tiger. I would be able to sell my G4/933 for around £400 to £450, ending up with a net cost of between £200 to £400. This would give me an approved machine with some reasonable life in it but it would still be a G4 rather than a G5. Option 4 would have a net cost of around £800 and I would end up with a dual2.3GHz G5 computer with PCI slots, but it would be a new computer with built-in obsolescence. Option 5 would give me a G5 with the most life but at an as-yet-unknown cost. Digidesign have as of the time of writing not published the crossgrade price for going from PCI to PCIe cards. My guess is that it will be around £500 per card, so for my HD2 system this would give me a net cost of around £1800 for this option, and it still doesn't get round the problems of the Intel Macs coming over the horizon. My gut feeling, having researched this as thoroughly as I could, is the world will change completely within the next 12 to 18 months. In the present, however, I need to have a system that allows me to make a living and keeps me close to the front of the technology race. In the end, I decided to go with option 3 as being the least expensive solution that keeps me well inside the supported hardware window until the revolution comes. When that happens I plan to go for a well specified new Intel Mac with PCIe slots, as well as whatever new Pro Tools hardware is current at that time, having had my money's worth out of this system and built up a nice reserve out of current earning to pay for its successor. That's my choice — what's yours? Published in SOS January 2006






Sound on Sound (January 2006)

Sound on Sound (January 2004)

Sound on Sound (January 2005)

Sound on Sound (September 2005)

Sound on Sound (December 2005)

Sound on Sound (November 2004)

Sound on Sound (March 2004)

Sound on Sound (June 2005)

Sound on Sound (October 2004)

Sound on Sound (May 2005)

Sound on Sound (July 2004)

Sound on Sound (November 2005)

Sound on Sound (June 2004)

Sound on Sound (February 2004)

Sound on Sound (September 2004)

Sound on Sound (February 2005)

Sound on Sound (July 2005)

Sound on Sound (March 2005)

Sound on Sound (May 2004)

Sound

Sound

Sound

Sound Advice on Mixing

Sound Symbolism

Sound Insulation

Sound Doctrine

meyer sound design reference (for sound reinforcement)

Sound Advice on Microphone Techniques

Carly's Sound

Secondary Sound

Sound Reading

Sound on Sound (January 2006)

Sound on Sound (January 2004)

Sound on Sound (January 2005)

Sound on Sound (September 2005)

Sound on Sound (December 2005)

Sound on Sound (November 2004)

Sound on Sound (March 2004)

Sound on Sound (June 2005)

Sound on Sound (October 2004)

Sound on Sound (May 2005)

Sound on Sound (July 2004)

Sound on Sound (November 2005)

Sound on Sound (June 2004)

Sound on Sound (February 2004)

Sound on Sound (September 2004)

Sound on Sound (February 2005)

Sound on Sound (July 2005)

Sound on Sound (March 2005)

Sound on Sound (May 2004)

Sound

Sound

Sound

Sound Advice on Mixing

Sound Symbolism

Sound Insulation

Sound Doctrine

meyer sound design reference (for sound reinforcement)

Sound Advice on Microphone Techniques

Carly's Sound

Secondary Sound

Sound Reading

Recommend Documents