Sound on Sound (December 2005)

In This Issue December 2005 In This Issue Click article title to open Reviews People Access Virus TI Composing For F...

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In This Issue

December 2005 In This Issue Click article title to open Reviews

People

Access Virus TI

Composing For Films

Modelled Analogue Synth

Harry Gregson-Williams

The Virus TI promises to bridge the divide between hardware and software instruments, and create a world of Total Integration, while still offering the classic Virus sound. Is it a hard reality, or have Access gone totally soft?

Harry Gregson-Williams's drive to explore original ideas and sounds has made him one of Hollywood's leading composers, scoring everything from romantic comedies to spy thrillers and historical dramas.

AEA R88 Stereo Ribbon Microphone This new monster mic incorporates two separate ribbon diaphragms, allowing you to use either Blumlein or M&S stereo recording configurations.

Buchla 200e Patchable Analogue & Digital Synthesizer

PART 1: Alongside Bob Moog, Don Buchla is one of the founding fathers of synthesis, and yet much less is known of him and his instruments. With this two-part review of Buchla's latest synth, and a history of some of his pioneering work, we hope to redress the balance...

Cakewalk Sonar 5 MIDI + Audio Sequencer [Windows] The new version 5 sees Cakewalk's Sonar becoming a more complete production package than ever, with the addition of new synths, a convolution reverb, 64-bit support and Roland's celebrated Variphrase vocal processing technology.

Digitech Artist Series Digital Guitar Effects Pedals These new pedals model celebrated guitar tones, including those of Eric Clapton and Jimi Hendrix, in unprecedented detail.

ESI Pro Maxio XD 192kHz Audio & MIDI Interface For PC file:///F|/SoS/SoS%2012-2005/Contents.htm (1 of 4)11/23/2005 3:00:54 PM

Leader Paul White's Leader Is the modern sequencer/DAW user interface now crying out to be streamlined?

Recording Hard Rock Toby Wright He took an unusual and unhurried career path, but Toby Wright has helped to create some of the most influential hard rock records of the last 20 years, including Metallica's definitive ...And Justice For All, and is now one of America's most sought-after engineers and producers.

Sounding Off Roger Thomas Won't somebody please think of the audio equipment?

Studio SOS Dave Rogers This SOS reader was having trouble with his monitoring, so the SOS team sped over to his home studio in Bristol, England to sort things out. Technique

Advanced Timing Correction In Pro Tools Pro Tools Notes & Technique We all know that Beat Detective can be used to fix up dodgy drumming. But how about creating a tempo map from a freely played keyboard part? Or replacing a piano track with note-fornote accuracy? You can achieve amazing results when you know how...

Audio Interface Manufacturers' Round Table PC Musician

In This Issue

With interface standards and user requirements changing all the time, the audio interface marketplace is a volatile one. We catch up with representatives of eight leading manufacturers for the inside track on the future of audio I/O hardware. Firewire and USB 2 interfaces have their advantages, but if you need serious channel counts at high sample rates, the PCI card still rules, and ESI's heavyweight recording system caters for a huge range of input and output formats at up to 192kHz.

Genelec 8020A Active Monitors Despite their diminutive size, these new nearfield monitors still share the Genelec family sound.

Korg OASYS: Part 2 Workstation Synth

PART 2: We finish our examination of Korg's new megaworkstation, taking in the remaining synth engines, the sampler, the KARMA algorithms and the onboard sequencer, and draw our conclusions about it...

Little Labs Multi Z PIP Instrument Preamp Aimed at professionals, the Multi Z PIP combines premium DI box, mixer, and re-amping device in a single small unit.

M Audio iControl Control Surface For Apple GarageBand 2

There are dedicated (and expensive) control surfaces for Logic, Pro Tools and Cubase... so why not an affordable one for Apple's semi-free entry-level application GarageBand? M Audio must have thought exactly the same thing...

Sample Libraries: On Test Hot Releases Assessed We check out the latest sample libraries on the block: True Strike 1 ***** Flatpack 2 **** Drumdrops In Dub Volume 1 *****

Sony Oxford Limiter Mastering Limiter Plug-in For Pro Tools Like many of Sony Oxford's plug-ins, their new limiter takes a familiar concept and applies a novel twist.

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CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven' Producer/Engineer: Gil Norton With their oblique, short and often brutally noisy songs, The Pixies reinvented rock music at the turn of the '90s, and influenced almost everyone who picked up a guitar in the following decade. Producer and engineer Gil Norton helped them to shape their breakthrough single.

Dual-core/Dual-processor G5s Apple Notes Christmas came slightly early this year for Mac enthusiasts, with significant product announcements, including new dualcore, dual-processor Power Mac G5s. But just what do the new high-spec computers mean for musicians?

Granular Synthesis How It Works & Ways To Use It Granular synthesis is the core technology behind the latest time-stretching and pitch-shifting algorithms, but it can also be used to generate extraordinary evolving soundscapes. We explain how the process works and show you how to get the best from the software that uses it.

Making A Living From Music For Picture Part 1 Writing music for picture seems like the ideal career. You get to work in your studio for a living, you can earn good money, and there's so much potential work: action films, travel and nature documentaries, romantic comedies, cartoons, lowbudget sci-fi, even breakfast cereal ads. But how do you break into this lucrative world? As we find out in the first part of this new series, the first thing you need is determination...

Mixing Live Recordings In Logic Logic Notes & Techniques Mixing live band recordings within Logic presents a unique set of challenges, so we show you how to get great results with the minimum hassle.

PC Notes XP x64 News, PC Tips & Updates The 64-bit Windows XP x64 edition is on the shelves, but musicians should stick with their trusty 32-bit OS for the moment. PC Notes explains why, as well as offering some constructive soundcard feature suggestions to manufacturers.

Streamlining Your Workflow In Ableton Live Ableton Live Notes & Techniques We begin this new series on using Ableton Live by examining how you can increase your productivity whilst using the

In This Issue

TL Audio M4

program as a writing tool.

Valve Mixer

The Lost Art Of Sampling

Part 5 Nearly all modern samplers have powerful synth engines concealed inside them — and sometimes they're so well hidden that their users are unaware of their existence. But This new mixer caters for those people who found TL Audio's M3 too small, and the flagship VTC too big. But is then why would you want a synth in your sampler? Let's find out... the M4 just right?

Tube-tech MEC1A & MMC1A Valve Recording Channel & Multi-band Compressor These two new valve units offer unusually powerful processing for professional studio use. Competition

WIN Berklee College of Music Production Course

Tuning Drum Loops In Reason Reason Notes & Techniques Can't quite get your Reason rhythm section kicking with the rest of the track? If you've never considered tuning your drum samples and loops to help create a tight and harmonious mix, now may be the time to try it...

Using Digital Performer With External Hardware

Digital Performer Notes & Techniques Few of us use our software sequencers in isolation — we all Q. Is it possible to record in surround on need associated hardware, such as monitors, external effects, only two tracks? and favourite MIDI synths. This month we take a look at using DP with such hardware. Sound Advice

Q. Is there something wrong with my vintage spring reverb?

Q. What factors affect the quality of a microphone capsule? Q. What makes some interfaces more expensive than others? Q. Why do my mixes clip when I apply a high-pass filter? Live Sound

AER Acousticube 3 Acoustic Instrument Amplifier AER's Acousticube has been at the pinnacle of acoustic instrument amps since 1992. Does the latest revision live up to the legacy?

AKG D22 & D11 Instrument Microphones The new Crystal Clear Sound range of mics includes two models intended for instrument miking. They've got the prestigious AKG name, but they won't break the bank.

M-Audio Aries Hand-held Capacitor Microphone M-Audio are perhaps better known for their studio and computer-based peripherals, but they now have a microphone range, to which has just been added a stagespecific model. We put the Aries through its paces.

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Using MIDI Functions In Sonar 5 Sonar Notes & Techniques Cakewalk have strengthened the MIDI side of Sonar 5 considerably, in recognition of the rise of software synths that benefit from enhanced MIDI controllability. We run through some of the new features and suggest how you might want to use them, as well as rounding up the usual Sonar news and background info.

Working With Video In Cubase SX & SL Cubase Notes & Techniques This month we take a look at building tempo maps for writing to picture in Cubase, using Markers, Time Warping and the Process Tempo command. Music Business

Music Publishing Everything You Wanted To Know (But Were Afraid To Ask) If you want to make money as a songwriter, composer or lyricist, the obvious answer is to find yourself a publisher. But what do music publishers actually do for their clients? Why do you need one, and how can you find the right one?

In This Issue

Meet the Sound Guy Jonathan Lucas : Freelance Engineer As well as doing regular stints at Camden's Barfly and other London venues, Jonathan Lucas engineers for a successful gigging band, whom he also records in the studio. We find out how it all came together.

PA Basics Power & Electrical Safety on stage Staying safe on stage is more than a matter of simply making sure that willing hands are available before taking a dive. Knowing how to properly handle the mains power we all need is also crucial to performance health...

RCF ART 322A Active PA Speakers Italian company RCF have a good pedigree. They helped design and manufacture for Mackie when the latter first entered the live sound market. We check out RCF's own entry in the portable powered PA stakes.

Stage Monitoring & Monitor Mixing Workshop Although monitor engineering is often thought of as subordinate to handling the FOH sound, in reality it's at least as important. We take a tour around this most crucial of live-sound subjects.

Voice Systems Eclipse Active PA Speakers The PA market isn't short of powered 'plastic' cabinets at the moment, but not many use the coaxial speaker design approach — which is what sets apart this Italian-made model.

Yamaha EMX512SC 12-channel powered PA mixer Clever lightweight amplifier technology and exceptional effects, plus several other useful features, distinguish this practical and powerful live mixer.

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Access Virus TI

In this article:

Access Virus TI

Total Interface Modelled Analogue Synth Chips With Everything Published in SOS December 2005 Total Initialisation Virus TI — A Second Opinion Print article : Close window Total Installation Reviews : Keyboard The Rear View Total Integration The Hub Of The Matter Total Impression Total Instantiation The new Virus TI promises to bridge the divide Stop Press — OS v1.0.3 between hardware and software instruments, and

Access Virus TI pros

create a world of Total Integration, while still offering the classic Virus sound. Is it a hard reality, or have Access gone totally soft?

Two words: Total Integration. The new TI engine sounds great, and the Hypersaw and Mark Wherry Wavetable oscillators add a great deal of new colour to the 'Virus sound'. I The new operating system, better LCD screen and the t's fair to say that there are very few electronic improved control layout makes the Virus easier to use musicians who haven't heard of Access Music's and clearer to understand. Virus synth, and still fewer who wouldn't like to own Did I mention it sounds one. When you think about it, that's a pretty great?

cons

amazing achievement for a hardware synth that grew up during the software revolution.

Latency isn't great when using the instrument in Sequencer Mode and routing the audio via USB, and this could be a problem for realtime performance unless you set a low buffer size. Not every feature is ready in the first version of the Virus TI OS and Virus Control: nonreal-time off-line bouncing and the Remote mode are noticeable omissions.

The Virus reminds me of The Doctor from Doctor Who; firstly, it seems to be regenerated every couple of years, and secondly, if you got into synths at any point over the last eight years, you might fondly remember 'your Virus', which would be whichever one was available at the time of your interest. The latest incarnation was announced last year, previewed in SOS December 2004 (see www. Photos: Mark Ewing soundonsound.com/sos/dec04/articles/virus.htm), The Virus TI Polar, with the Virus Control plug-in running and first shown at Winter NAMM at the beginning under Cubase SX. summary of this year. Its release was promised less than 50 Access' implementation of the days later, but it finally showed up on October 3rd. Virus TI as the centrepiece of The phrase 'hotly anticipated' is over-used by almost everybody, but in the case your computer music system of the Virus TI, as the new model is called, it's appropriate. The reason can be is commendable, and the summed up by two words: Total Integration (hence Virus TI). company seems to have covered all the possibilities for file:///F|/SoS/SoS%2012-2005/accessvirusti.htm (1 of 13)11/23/2005 3:01:24 PM

Access Virus TI

total integration. The TI is also the best-sounding Virus to date.

information Virus TI Desktop model, £1199; Virus TI Polar model, £1499; Virus TI Keyboard model, £1499. Turnkey +44 (0)20 7419 9999. +44 (0)20 7379 0093. Click here to email www.turnkey.co.uk www.access-music.de

Test Spec Virus TI OS reviewed: v1.0.2 (Build 26). MAC REVIEW SYSTEM Dual 2.7GHz Apple Mac G5 with 2.5GB of RAM running Mac OS 10.4.2, plus Emagic Unitor8 MkII MIDI interface. Apple Logic Pro v7.1.1. PC REVIEW SYSTEM Dual 3.6GHz Xeon-based PC with 4GB of RAM running Windows XP Pro (SP2), plus an Emagic Unitor8 MkII MIDI interface and an RME HDSP 9652 audio interface. Steinberg Cubase SX v3.1. ALSO USED Virus C, Virus B, Virus Indigo, Virus Powercore, and Virus Indigo TDM.

The TI brings together Access' experience with both hardware and software versions of their synth for the first time in one package. The concept is brilliant; the Virus TI combines the best of both worlds from previous Virus products and is able to run as both a stand-alone synth, and as a sample-accurate instrument plug-in within your computer-based sequencer. If Access had stopped there, I'm sure we we would have been happy, but when they say total integration, they really mean it. The Virus TI can also act as an audio recording device for your computer (making use of the onboard audio input and output), or as a MIDI interface (using the built-in MIDI ports and the keyboard, if your TI model has one). And lastly, the Virus's hardware control surface can control the Virus software running in plug-in mode (and the plan is to make it usable as a generic control surface for other plug-ins and applications). Best of all, though, the Virus TI can do almost all of these operations at the same time. Access have always offered Viruses in multiple versions, and the TI is no different. Three TI models are available: Desktop, Keyboard, and Polar. Desktop is the Virus's original tabletop-style design, but the TI incarnation has a slightly darker, sleeker appearance than that of the previous Virus C. The dark, wooden panels on either side can be replaced with the supplied kit, so you can install the synth in a standard 19-inch rack. One annoying thing about racking previous Virus models is that once in the rack, the ports that are normally on the back of the unit are now on top, making them hard to access and meaning that you have to leave two or three units of space above the Virus. To solve this problem on the TI, Access have made it possible for you to rotate the physical position of the ports so that they will be on the back when rackmounted. Very neat! The Virus TI Keyboard, which is the only model of the Virus TI I've yet to see in person, has a 61-note keyboard with mod and pitch wheels, and replaces the previous Virus kc model. However, arguably the most desirable member of the family is the mostly white Polar, which replaces the previous 'lust-have' Virus Indigo and Indigo 2 models, and was used for most of the photos in this review. As you can see, the Polar looks totally, well, cool! Like the Indigo, the Polar's compact design offers a three-octave, 37-note keyboard (which felt of better quality to me than that of the Indigo, with a lighter action), and there's now a elegant wooden edge to the casing underneath the keyboard that adds a touch of class to the instrument. This is basically the Virus you'll want to sell your granny for.

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Access Virus TI

Total Interface All Virus TI models feature identical controls and features, and the actual control surface has been redesigned slightly (see close-ups overleaf); eight buttons have been added, including a Shift key. The most obvious change is the new 128x32-pixel LCD display, which is a huge improvement over previous Virus screens: it's easier to read, and offers more information. On the other hand, when you switch on the Polar, it's apparent that its white LEDs are rather bright. While this would be great on a darkened stage, it can be a little distracting in the studio. Fortunately, you can adjust the LED brightness in one of the menus! The relative brightness of the BPM LED can be adjusted separately. Underneath the new display are three soft knobs (up from two on the Virus C), which are now much more useful, as they have become an integral part of configuring parameters in the new operating system. On the main page, the sound-altering function of each soft knob is labelled above on the lower part of the screen, and by pressing Shift the three soft knobs enable you to quickly dial up different patches, adjusting Category, Bank and Program settings respectively. This speeds up patch navigation and is better than the Value knob on previous Virus models. Each editing and configuration page now has up to three parameters that can be adjusted by each of the soft knobs, which makes them much easier (and quicker) to work with. If you have a synth with many rotary controls and memories, there's always the issue of how to reconcile the physical positions of the knobs with the stored parameters. Access's solutions have always been pretty good, and the three knob responses implemented in the very first Virus are still present in the TI: Jump, Snap and Relative. Jump means that the value is set to the exact position of the knob when you make an alteration, Snap means that the value won't change until the knob passes the point of the original value, and Relative means that the knob adjusts a value relative to the original value. On the TI, this is displayed more clearly than before, thanks to the new screen; when you adjust a knob, a window pops up to tell you what the current value is, what the former value was, and, if you're in Snap mode, which way to turn the knob to set it to the original value. The window disappears if there's no further adjustment after a few seconds.

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Chips With Everything Under the bonnet, Virus synths are powered by Motorola's 56k family of DSP chips, which is commonly used in pro audio products, including, for example, Digidesign's Pro Tools TDM. This architectural similarity is what allowed Access and Digidesign to announce a software Virus plug-in for Pro Tools 24 Mix systems as long ago as 2000. After this, Access released a version of the Virus for TC's Powercore DSP engine platform in 2002. This made sense, because Powercore is also based around Motorola 56k DSP chips. Unlike the Pro Tools version, Virus Powercore was (and still is) sold in two versions: a 'base' licence, allowing you to run just one instance of the Virus plug-in on one DSP chip, offering 16 voices, and an 'unlimited' licence that allows you to run multiple instances of the plug-in on as many chips as you have available in your Powercore system.

Total Initialisation The architecture of the Virus TI has been completely overhauled from previous Virus engines, and now features a dual-DSP configuration of faster processors to offer more power than ever before. At its most basic level, this means more voices, and Access quote a polyphony of 80 voices 'under average conditions', which you may remember was the number of voices quoted for running the original Virus plug-in on Pro Tools Mix systems five years ago. I didn't get near this number of voices when using Multi mode (up to 16 parts simultaneously), and despite the increased DSP power, the polyphony is dependant on how many oscillators you set and what effects you're using. It's therefore hard to be precise about the polyphony you can expect from the TI. If you use seriously DSPintensive patches, you might run out after three or four parts with only slightly more polyphony than the Virus C (maybe about 40 voices). Leaner patches (fewer Unison voices, for example, and fewer effects) will give you greater polyphony. But in the studio context, it was enough that I didn't run into problems. The Virus C's physical power button has been replaced with a soft power button on the TI, so the synth is in Standby mode when you plug it in, and one of the Transpose LED blinks to indicate this. To power it up, you press the two Transpose buttons together, and you do almost the same to put the synth back into Standby mode, except you have to hold the buttons down for two seconds while a countdown appears on the Virus' display. Once the Virus is powered up, it behaves just like a stand-alone synth, and if you don't want to use your Virus TI as a plug-in, you don't have to. The new TI sound engine is worth the price of admission, even if you forget the computer integration aspect. Comparing the TI to several previous Viruses, I felt the TI sounded better, partly due to improvements in its effects algorithms and also to having access to greater polyphony for more Unison voices. This impression may

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Access Virus TI

also be due to the TI's new 192kHz D-A converters (with optional soft limiting), and also to the improvements in the DSP engine. Although the Virus' D-A converters are now specified as being 192kHz, the Virus' internal clock offers either a 44.1 or 48kHz sampling rate. So my assumption is that the engine itself still functions at either of these sampling rates, rather than at 192kHz, and that this high value is more indicative of the quality of the converters used (it's not clearly stated anywhere, but my guess is that incoming 88.2 or 96kHz audio is currently sample-rate converted at the input, so that the voice count isn't halved). The A-D converters have also been improved, and now operate at 24-bit resolution, as opposed to 18-bit. One really nice improvement on the TI is that you now have independent delay and reverb effects for each of the 16 parts, and this is great when working in Multi mode, which embeds all the data for every part in one patch, rather than simply referencing Single mode patches as in previous Viruses. But perhaps the highlight of the new Virus TI engine is the addition of two new oscillators: Hypersaw and a true Wavetable oscillator. There are 72 Wavetables to choose from, plus a sine wave, and these cover the foundations for creating interesting pad sounds, gritty, noisy stabs, or those FM/bell-like cascading sounds that you remember from Waldorf synths. One nice touch is the ability to adjust which wave in the table you're using via the Wavetable Index parameter, and the waves are crossfaded to allow for smooth transitions, which works really well. If you assign Osc 2 as a Wavetable oscillator, you can also use the FM features of the second oscillator to sonically destroy anything! Hypersaw is basically a sawtooth oscillator, except that it's able to generate up to nine sawtooth waves in parallel, and you can add more and take away these additional oscillators in real time with no glitching, which is really neat. Needless to say, this is great for huge-sounding patches, and there's a detune option on the Hypersaw oscillator to really thicken the sound, along with a sync toggle. However, because Hypersaw is effectively 'nine oscillators in one', you can also use the Virus' Unison mode to have up to eight Hypersaw oscillators per note, which means you could have 72 oscillators per note! Or, do the same with the second oscillator and get 144 oscillators per note, and add a bit of the suboscillator for bass! This is just plain wrong, but so addictive! Other features of the new TI engine include six modulation matrix slots with one source and three destinations, giving six sources and 18 destinations in total. There are, however, a few problems that were present in the engine that Access are gradually fixing through updates. There are some issues with the arpeggiator when trying to slave the Virus to an incoming MIDI Clock signal (which older Virus users may remember from previous 'first' versions), although I didn't find this a problem when using the Virus in TI mode as a plug-in, presumably since the sequencer coordinates the sample-accurate sync. And there seems to be a problem with the tuning in the arpeggiator occasionally, which completely foxed me for a while, but I later found other users having the same problem, which has been described on forums as the 'drunken' arpeggiator.

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In short, aside from a few teething troubles, the new sound engine is amazing. However, it's now time to look at the seriously impressive aspect of the new Virus — the computer integration.

Virus TI — A Second Opinion The cold realities of having to move house several times in quick succession, plus the difficulties and costs of keeping vintage synths well serviced, have conspired to whittle my studio mercilessly. Recording and sound design is now a 99-percent computer-based experience for me. I have no regrets; I like contact with the hardware that's producing the sound I want, but I'm able to run more synths, samplers and effects in software than I've ever owned at one time. And I like a life which isn't drowning in leads and means that I have desk space to spare. But working solely in software presents its own problems. New software demands more RAM and the latest CPU speeds, while your own computer stands still. Thus, the idea behind the latest breed of hardware synth — the type that adds DSP resources to your system rather than demanding more of their host — is one I like. Access's Virus TI is a particularly attractive implementation of this idea. The Virus family has become a classic in a relatively short time, and software implementations have been quite demanding, running on higher-end DSP-based audio systems. Host-based options would suffer and, being frank, no doubt Access would suffer from piracy. The hardware-plus-software approach provides manufacturers with security, and users with power. I haven't had the TI long, but even in that short time, digging deeper has been rewarding. The only thing I didn't like on the desktop version I examined was the external PSU. Everything else — its solidity, weight, layout, and sound — I loved. And then there was Total Integration. It's rather uncanny being able to tweak and organise a synth of this power from within your favourite audio environment (Ableton Live, Steinberg Cubase SX and Cakewalk Sonar for me), just like a normal plug-in, and without maxing out the CPU. That the synth can reliably offer basic audio I/O is the icing on the cake. The MIDI interfacing makes up the little piped bits around the edges, and when the control surface software comes on stream, the Virus TI will be covered in hundreds and thousands! Like Mark Wherry in the main part of this review, I found it best to not demand too much of this side of things: audio in and out running simultaneously with busy multi-part synth playback is best avoided. What's more, my ageing 450MHz Mac has only USB 1.1 ports, so although the plug-in wasn't too demanding, there was a little too much data moving up and down the pipe for it to be totally happy. Despite this, the sonics are great. The Virus TI's modelling gives you access to a wide palette of sound; whether you like your analogue acid-fat or modular-clinical, and want to mix in modern digital textures that jump to the top of the mix or blend wistfully, it's here. The presets even include drum sounds! With no less than 17 128-strong ROM patch banks, the Virus TI is impossible to summarise. In the 'leads' category, A095: 'Syncer' is classic with a modern edge, and C075: Click here to email just sings! Pads also abound, from textured backgrounds to upfront movers. The outright simulations of instruments are variable, but this was always the way with real analogue or FM. Here, the fake

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Access Virus TI

pianos, erzatz contrabasses and so on have the playability and 'feel' of the original without being fully imitative. The Virus's forté has always been serious bass. Practically anything in this department that comes up on the category search will impress, but try shaking your woofer with A008: 'Bombasdr'. B126: 'YamahaFB1' also does what it says — it's classic four-op FM 'thud bass'. But by way of contrast, there are also incredibly delicate sounds, such as D009: 'AprilPad'. The Virus TI arpeggiator is particularly good, many factory patches showing this off. Add a drum loop and it's instant dancefloor-filling material. Try the solid, traditional K031: 'Donkey', the jangly A002: 'AoilioA' or the Kraftwerk tributes of patches G042 to G047. And the Virus TI's three LFOs are also capable of some pretty funky arpeggiator-like tricks, as evidenced by A124: 'Zimoux' and A101: 'Thr3sum', for example. I could use up another few boxes to summarise favourites, but I have some programming to do! This is one inspirational synth. Derek Johnson

Total Installation Installing the Virus TI for use with your computer sequencer is easy, and although a CD-ROM is supplied containing the installation software, Access recommend checking their web site to make sure you have the current version. Once you've launched the installer, the appropriate software will be installed, and during this process Windows users will be asked to connect the Virus to a USB port on their computer (Mac users can simply connect the Virus after the installation). The Windows installer advises you to 'choose the USB port wisely' since, as with all USB devices, Windows will try and install the drivers again if you plug the Virus TI into a different USB port later. The Windows installer also informs you (and this advice concerns both Mac and Windows users, as described in the manual) that you cannot connect the TI to your computer via a USB hub — see the box over the page for more info. Once the drivers are installed, the Windows installer will ask you to disconnect and reconnect your Virus TI, and after this the installation will be complete. If the installer is supplied with a firmware update for the Virus TI OS, the cross-platform Virus TI Firmware Update application will run for about 10 minutes, during which you can't do anything to the synth. Following this, the TI will reboot and you'll need to restart your PC or Mac. It's now time for the real fun to begin. I mentioned earlier in this article that the Virus TI can operate as an audio and MIDI interface, and during installation Direct X, MME and ASIO drivers will be installed for Windows users, and Core Audio and MIDI drivers for Mac users. This means that you can use the TI as a front end for your computer music system with no extra MIDI or audio hardware required, which is useful. And the stand-alone Virus synth remains operational, even when the TI is being used as an audio and MIDI interface. In your sequencer, two additional MIDI ports show up as TI MIDI and TI Synth.

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Access Virus TI

Sending MIDI to the first port will cause the data to be output from the Virus' MIDI Out port, while choosing TI Synth will trigger the actual Virus synth. And what's really nice is that the audio driver output will be mixed with the output of the Virus synth engine, which already gives you a much more integrated approach than ever before with a hardware synth. The only things I found annoying were the rough nature of the user interface for the ASIO Control Panel for Windows users (see right), and the lack of documentation describing how to use it. It's not obvious how the options should be configured, and the layout looks like a throwback from Windows 3.1! Performance-wise, though, I didn't have any issues with the USB audio, and using the smallest buffer size possible, Cubase SX 3.1 reported an input and output latency of 3.968 and 4.898ms respectively. On the Mac side, I set the buffer size in Logic to 256 samples (approximately 3ms at 44.1kHz) and this seemed workable.

The Rear View All three TI models feature the Virus's standard complement of three stereo output pairs on jacks, a stereo input pair, a stereo headphone output, and MIDI In, Out, and Thru connections. All Virus models also now include RCA connections for S/PDIF input and output, which means that for the first time, you can get audio into and out of the Virus digitally. There's also the all-important USB 2.0 port for computer connectivity. In addition to these facilities, the Polar and Keyboard models also feature a built-in PSU (the Desktop has an external brick), two jack connections for Control and Hold pedals... and a blinking light in the shape of the Klingon-like Access logo. It beats in time with the current tempo of the Virus, and you can set how dramatically the blinking appears above the normal lighting of the logo by setting the Logo Groove parameter in the System 5/5 Config page, where '0' makes the Logo always on, and '127' causes the logo to blink and fade completely to darkness after each beat. It'll look good on stage!

Total Integration Of course, the feature we've all been waiting for is to run the actual Virus synth engine as a plug-in. And if you're wondering whether the Virus TI can still operate as an audio and MIDI interface when running the Virus engine as a plug-in, the answer is yes, which is really, really useful! The Virus TI's plug-in application is referred to as Virus Control, and when you start the plug-in on your host, the Virus TI will switch into Sequencer mode and no longer work as a stand-alone synth. The Virus control surface itself effectively becomes a control surface for the Virus Control plug-in, and this control is achieved by internal communication between the TI and Virus Control, so there's no additional work required by the user. And despite the internal nature of this communication, you can still automate Virus Control with your host's own automation system. Again, this requires no setting up: just enable automation in your host, adjust parameters on the control surface, and the host will record the movements as if you were file:///F|/SoS/SoS%2012-2005/accessvirusti.htm (8 of 13)11/23/2005 3:01:24 PM

Access Virus TI

adjusting on-screen controls directly. The beauty of this system is that you can control the plug-in at any time from the TI, regardless of which Track is selected in your host. A further mode Access have created for the Virus TI is Remote mode, and this enables the Virus' front panel to control other software instruments and effects in addition to the TI — or rather it will. Unfortunately, at the time of writing, Remote mode isn't implemented in the current version of either the TI OS or Virus Control. Virus Control (shown overleaf) appears to the user as if it were any other The somewhat unfinished-looking Control instrument plug-in. Behind the scenes, Panel for the Virus TI ASIO driver. MIDI data from the instrument plug-in is sent to the synth engine via USB, and audio from the synth engine is sent back to Virus Control so that the plug-in outputs audio to the host application, allowing you to use other plug-in effects to further process the Virus' audio output. Like many products running over USB, the TI seems to work variably from computer to computer, depending on configuration and host. Check out the unofficial Virus user forum at www.sunesha.nu/virusforum/ and you'll read all manner of horror stories, while other users claim their TI is working OK. I fell into the latter category, aside from the non-USB related sound engine problems I've already mentioned, and a slight problem with latency. If you're using the TI as your main audio interface as well, the audio has to travel back down the USB cable again, and although the audio would be have to be sent out by the host at this point no matter what interface you were using, I noticed the latency when using the Virus as both a plug-in and an audio interface was greater than when using another main audio device. Even with an additional audio interface, though, you have to be careful to keep your buffer sizes small (no greater than 256 samples) to keep the TI plug-in playable. However, the upside to Virus Control is that latency is only an issue when performing in real time: on playback, the TI, like any other instrument plug-in, is capable of sample-accurate operation. And another bonus is that the audio doesn't have to be routed back to your host sequencer when using the TI in Sequencer mode. Like the Virus hardware, the Virus Control plug-in also has multiple outputs: there are two stereo outputs available to your host, and in Virus Control's Common page, you can set whether the main and secondary audio outputs are routed to an output on the plug-in, or directly to an output on the Virus TI itself. This latter option has a few pros and cons; firstly, it's more useful if you're using the Virus TI as your main audio device, so you don't need an additional mixer, file:///F|/SoS/SoS%2012-2005/accessvirusti.htm (9 of 13)11/23/2005 3:01:24 PM

Access Virus TI

and also, it prevents you from further processing the audio output of the Virus in your host. On the plus side, though, you could use the direct routing for real-time performance, and then switch to the plug-in's output for playback and mixing. It's a nice touch that addresses a potential problem. Here you can see Virus Control running as an Audio Units plug-in within Logic Pro and showing the Easy page, which offers quick access to some of the most common parameters for real-time performance. Notice also that Access Virus TI is selected as the Core Audio driver in Logic's Audio Preferences.

The Virus Control interface is pretty well laid out, and, as the manual points out, if you're familiar with using a hardware Virus (or a plug-in version), the TI plug-in is pretty self-explanatory — which is just as well, since there is little documentation available for Virus Control at the time of writing, although more is planned. I particularly liked the Arp page, which makes programming the arpeggiator so much easier than on the Virus' control surface, and the new Easy page looks pretty cool, with the ability to adjust both the cutoff frequency and resonance of the filter with the mouse at the same time. All 16 parts of the Virus are displayed in a column to the left of Virus Control, and here you can load and save individual patches, adjust the volume and pan of a part, and select a part for editing in the main display. Sequencer mode, which is the mode the TI uses when you're working with Virus Control, is similar to Multi mode, in that you have 16 simultaneous parts, but it actually works like having 16 simultaneous Single modes rather than one Multi mode. Virus Control also allows you to access any of the ROM and RAM banks on your Virus TI. In addition to storing patches directly on the Virus, Virus Control can also save patches to your local disk if you run out of space on your Virus, and as on the Virus Powercore plug-in, these are stored as banks of patches in MIDI file format. Actually, the Browser page of Virus Control is the only aspect that slightly lets the side down. Access chose to display the patch list as if the patches were on an LCD screen (a bit like NI's FM7), and style aside, this just makes it hard to read the patch list. Some better (faster) search facilities would also have been good, although I hear this is another area Access are working on.

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Access Virus TI

The Hub Of The Matter While it's true that using audio devices with USB hubs can often be problematic, I decided to ignore this warning initially and see if the Virus TI would still be functional connected in this way. However, I wasn't being awkward: the Windows computer on which I was installing the Virus TI was in a machine room some 2030 feet away from the main workstation where the monitors, keyboards and Virus TI were located. Since the maximum length for a USB cable (to allow for reliable transmission) is about 16.5 feet (five metres), not being able to use a hub was potentially a problem, and using a powered USB extender wouldn't help, unfortunately, as these are effectively implemented to behave as if they are USB hubs. So I ignored the initial warning during installation, and at first, all was well. The installer detected that I had connected a TI (even though it was via a hub), and the drivers were still installed correctly. At this point I was thinking 'hub, shmub!' — until I tried to use the Virus Control plug-in in Cubase SX3, that is. Virus Control opened displaying the Total Integration Status Page, and while the Audio and MIDI Communication sections contained green ticks, the USB Communication test had failed, and been awarded a large red cross, with the words 'You are using a hub...'! While this is annoying, it's understandable. Given the number of incompatibilities that users could encounter using the many possible types of USB hub, it does make a degree of sense for Access to ensure that the TI is used in a way that always gives the best possible results. On the other hand, if you're using a computer with a limited number of USB ports (such as an Apple Powerbook), it would be handy if an Expert mode could be added for those users who want to take their chances!

Total Impression I really love the Virus TI. It's not perfect; there are some issues to be resolved, and features to be implemented. But, for me at least, there's nothing about the TI in its current state that would prevent me from having a good time or making a purchase, despite the fact it's not uncommon to see the odd bug. For example, I noticed that when I was selecting patches from the TI in Sequencer mode, Virus Control didn't always display the correct patch name. Still, it's reassuring to see Access releasing regular updates on the Internet for users who have already bought the new Virus, continuing their good practice of offering regular updates for previous Virus models. The Virus has always been a fine-sounding synth, and it's always been fun and intuitive to program. The TI takes this to the next level; the new engine, with its additional DSP resources, is fantastic, and the improvements to both the Virus' hardware user interface and the new software interface, in the form of Virus Control, make the TI a dream to program. The 'total integration' features have been well thought out, and, for the most part, well implemented — everything you could conceivably want to do with a Virus attached to your computer seems to be possible. The only missing piece is the 'still to come' Remote mode, but I don't think this is a serious omission. file:///F|/SoS/SoS%2012-2005/accessvirusti.htm (11 of 13)11/23/2005 3:01:24 PM

Access Virus TI

I think I'm going to buy a TI Polar because, for me, it's the perfect desktop instrument. I can have it right next to my computer keyboard and mouse, it's the perfect master keyboard for generating MIDI data when an 88-note keyboard isn't required, and, of course, it has the best-sounding Virus synth engine to date. It's not particularly cheap, and the Polar costs the same as the 61-note Keyboard version, but it's such a great instrument that I can't really criticise the price, because you really do get what you pay for. Access' latest Virus incarnation was worth the wait: users for whom the TI will be 'their Virus' and seasoned Access veterans alike are in for a treat. A bit like fans of Doctor Who and the BBC's new series, really

Total Instantiation While it might be obvious, you can't run more than one instance of the Virus plugin simultaneously, since you only have one Virus engine attached. If you try to add another instance, you'll see the Total Integration Status Page in Virus Control informing you that Audio, MIDI and USB Communication have failed. Fortunately, this does no harm, and you can simply remove the second plug-in instance and carry on using the first. One potential way around only having one Virus TI (and let's not forget the TI is 16-part multitimbral!) would be to use the Freeze feature found in modern sequencers, where a plug-in's output is rendered as an audio file, freeing up the resources occupied by that plug-in, and allowing you to open additional instances. This usually liberates the computer's CPU, but in this case, it's the TI that would be freed up. Using Freeze in conjunction with the TI could be rather neat; imagine being able to use one instance, freeze it, move onto another instance, freeze that, move back to the first instance and make changes, and so on. So long as all TI instances were 'frozen' before opening or unfreezing another, it could all work rather well. Unfortunately, Freeze-style features aren't compatible with the Virus Control plug-in at present — not even if you're using the Polar! The release notes promise it for 'a future upgrade'. This didn't stop me trying it nonetheless, but I got very corrupted-sounding audio, almost as if the Virus was trying to play at a faster tempo. This makes sense, because most Freeze functions work by performing what is basically a faster-than-real-time bounce. Even if Freeze had worked, though, I noticed that the plug-in wasn't always released under Logic Pro; the Virus TI stayed in Sequencer mode even after the original plug-in instance was frozen. Cubase SX has an 'Unload Instrument when Frozen' option for use when freezing, and this will come in handy when TI Freeze support arrives. Even if you'll never need to use the Freeze function in your sequencer, the behaviour just described could still affect you, as it applies to all non-real-time, offline bouncing, not just freezing. This means that if you do a bounce down or file:///F|/SoS/SoS%2012-2005/accessvirusti.htm (12 of 13)11/23/2005 3:01:24 PM

Access Virus TI

export of your finished mix inside your sequencer, you'll need to make sure that the real-time mode is selected for the time being.

Stop Press — OS v1.0.3 Just as I was about to submit this article, Access posted a version 1.0.3 update on their web site. According to the company's release notes, v1.0.3 fixes a problem where the tempo was displayed instead of Patch Panorama on the Common page, has improvements when the host sampling frequency is 96kHz, and fixes an issue where some pop-up menus displayed the wrong items. The Remote functions and Freeze support are still to be added in a future release, though. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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AEA R88

In this article:

Funky Ribbon Technical Specifications Using The R88

AEA R88 £1526

AEA R88 Stereo Ribbon Microphone Published in SOS December 2005 Print article : Close window

Reviews : Microphone

pros Unique stereo ribbon configuration. Easy to aim accurately. Sounds warm, yet detailed, and very natural. Consistent polar responses. Competent built-in shockmount and supplied angle adaptor. Superb stereo imaging. Supplied with protective mic case and dust sock.

This new monster mic incorporates two separate ribbon diaphragms, allowing you to use either Blumlein or M&S stereo recording configurations. Hugh Robjohns

While touring the recent AES convention in New York, I was struck with how many companies were showing or launching Large and heavy. ribbon microphones. Although good ribbon microphones deliver a beautiful and unique sound in the right circumstances, they were summary A unique stereo ribbon mic in largely usurped by capacitor mics as the engineer's weapon of choice a long time ago. Advances in magnet technology and lowthe classic Blumlein mould, combining the characteristic noise preamplification have made ribbons far more usable than attributes of ribbon technology they used to be, allowing the benefits of an inherently low noise — a warm, pleasing sound, floor, high SPL handling, and excellent transient response to be crisp transients, and natural enjoyed to the full. dynamics — with superbly cons

well-focused stereo imaging. This mic makes an excellent package for anyone who appreciates the virtues of the ribbon and the convenience of a fixed-stereo mic.

information £1526 including VAT. Affinity Audio +44 (0) 1923 265400. +44 (0)1923 266103. Click here to email

One of the more prominent names associated with ribbon microphones is Audio Engineering Associates (AEA), an American company owned and run by Wes Dooley, a highly regarded recording engineer with an exhaustive knowledge of Photo: Mark Ewing ribbon microphones old and new. Sound On Sound has carried reviews of AEA ribbon mics before (the RCA44 and the R84), but this review is a little different, because the new R88 is the only stereo ribbon mic I have ever come across.

www.affinityaudio.com

Funky Ribbon

www.wesdooley.com

The AEA R88 is, in rather simplistic terms, two R84 figure-of-eight mics mounted one above the other in a common housing. As a result, this is a seriously large (and heavy) microphone. It measures 2.6 inches in diameter — the same as the

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AEA R88

R84 — but a whopping 15.9 inches long (including the fixed mounting frame), and it weighs a stand-bending 5lbs, including the weight of the captive cable. However, although this is one of the largest microphones around, it is still considerably shorter than a pair of R84s mounted one above the other! The mic is supplied in a black nylon case, lined with a velvet material, and is protected from dust by a separate nylon bag. The latter helps to prevent stray ferrous particles — 'tramp iron' to use the American vernacular — from being attracted by the magnets and eventually fouling the ribbon gap when the mic is not in use. The User Manual reminds owners to store the mic vertically in its case to prevent the ribbon sagging under its own weight. The output cable is fixed to the bottom of the mic body, and extends for four metres before breaking out to a pair of Switchcraft male three-pin XLRs (pin two is hot). The cable is secured to the supporting frame as well, to provide some additional vibration isolation. The XLR plugs are undifferentiated, but the wires themselves are clearly numbered to identify the two capsule outputs. The body's metal end caps are supported top and bottom by a cunning metal frame, and are isolated with rubber pads. The frame features a 90-degree offset across the middle to ensure that the vertical portions lie in the nulls of each capsule. Covering the two ribbons across the central portion of the microphone is a tight shiny black fabric supported by an internal wire frame. The base of the support bracket is fitted with a 5/8-inch threaded socket for direct mounting on a stand or suspension cable, but the mic is also supplied with a very robust knuckle adaptor. This can be screwed into the support to allow the mic to be tilted at any desired angle. Serrated teeth between the two halves of the adaptor ensure that the mic won't droop.

Technical Specifications Internally, the dual-aluminium-diaphragm design is unique to the R88, but is similar to that of the R84 — being tweaked slightly to improve the off-axis frequency response. Each ribbon measures 2.35 x 0.185 inches (60 x 4.7mm), and is 1.8 microns thick. The two ribbons are mounted at 90 degrees to each other in the classic Blumlein manner, and cannot be adjusted at all. Markings on the body identify the two capsules and helpfully show the relative polarities of the front and rear lobes. In normal left-right stereo use, the microphone's AEA logo would be pointed at the centre of the musical ensemble or instrument, with output one connected to the left channel and output two to the right. For the more adventurous, the mic could be rotated 45 degrees to the left and used as an M&S pair, with output two providing the Middle channel and output one the Sides channel. This may seem counterintuitive to those unfamiliar with M&S practice: most would probably assume output one to be the Middle channel and output two the Sides channel. However, for correct M&S decoding, the lefthand lobe of the side mic must be in the same polarity as the front of the Middle file:///F|/SoS/SoS%2012-2005/aear88.htm (2 of 5)11/23/2005 3:01:31 PM

AEA R88

mic. The only way to achieve that with fixed capsules is to use the left channel as the Sides mic, and the right channel as the Middle mic. The frequency-response chart for the R88 shows a significant tilt from left to right. Relative to 200Hz, the mic is about 2dB down at 1kHz, 5dB down by 8kHz, and 10dB down by 20kHz — a rather steeper response tilt than that of An included mic-stand adaptor allows the microphone to be angled as required. the R84, although it doesn't sound 'dull' in use. The response below 200Hz wasn't shown, but judging by ear I'd say the level falls smoothly and gently, extending a very long way indeed — the specs claim a usable output at 20Hz and I can well believe it! The diaphragm resonance is 16Hz, and I did find the mic quite prone to low-frequency rumbles, despite the built-in shockmounting. The figure-of-eight polar response of each capsule is very consistent, with horizontal coverage over the usual ±45 degree range (front and back) and side nulls which are smooth and deep. The relatively long ribbon means that the vertical acceptance angle is slightly narrower than usual — especially at high frequencies — spanning maybe ±30 degrees. However, the high-frequency response falls smoothly, with no nasty off-axis colorations to worry about. The user manual quotes the maximum SPL as over 165dB above 1kHz (for one percent third-harmonic distortion), although care must be taken to protect the mic from direct wind blasts. This is not a mic to place close in front of brass instruments or bass drums! The output sensitivity is given as a respectable 50dBu/Pa, which shouldn't tax decent preamplifiers too much. Having said that, the crossed figure-of-eight design of this stereo mic means that relatively distant placements would have to be used in many situations, and that could easily call for mic gain settings in excess of 60dB. The R88's output impedance is surprisingly high at 270(omega), and it requires a minimum load impedance of 1.5k(omega), which isn't likely to prove a problem.

Using The R88 Obviously, the R88 is designed for use as a stereo mic, and its performance has been optimised accordingly. This is most notable in its proximity effect, which is very strong for close sources in order to maintain a healthy bass response for the typically distant sources encountered by a Blumlein stereo mic. Given the size and weight of this mic, a robust stand is mandatory, and I preferred to use a stand without a boom arm for most applications, employing the supplied angle bracket when I needed to tilt the microphone. I auditioned the R88 with various instruments and ensembles: an amateur string orchestra, a small file:///F|/SoS/SoS%2012-2005/aear88.htm (3 of 5)11/23/2005 3:01:31 PM

AEA R88

church choir, a pipe organ, a grand piano, and a small drum set. Sadly, the opportunity did not present itself to try it with a brass section, where I would have expected it to fare extremely well. The piano recording was particularly pleasing with the mic placed classically — about six feet away in the curve of the piano, and slightly below a line extending from the axis of the fully open lid. The sound was very natural, with a coherent tone and impressive transient dynamics, and the bass extension was not far short of my usual omni capacitor mics. A brief experiment with the mic inside the piano (about 12 inches away from the hammers and looking down onto the strings from about eight inches) provided a typically 'pop' sound, but without the fierceness than can mar this technique when using some capacitor mics. The AEA logo on the mic made it easy to aim accurately, and stereo imaging was pin sharp, as demonstrated clearly with the choir, orchestra, and organ tests, the last allowing the location of individual pipes to be identified with ease. Amateur string orchestras would normally fill me with dread, but the R88 managed to capture a well-rounded and natural string tone with plenty of body and detail, minimising the 'screechy' artefacts that many condenser mics would have emphasised. This is a common characteristic of most ribbons, and is partly to do with the inherently low resonant frequency — most condenser capsules tend to resonate at around 8-10kHz, which is often a factor in the characteristic capacitormic 'edge'. Ribbons are quite popular for drum overheads in the studio, and the R88 performed superbly when placed behind the drummer (looking down from about two feet above his head), delivering crisp, clean cymbals without splashiness. The drum dynamics were captured very well too, and the rear lobes of the mic seemed to do a very nice job of picking up the room acoustic in perfect proportion, negating the need for separate 'room mics' on this occasion. I also tried mounting the mic directly in front of the kit at head height, and obtained very natural 'jazz style' results. The overall character of the R88 is decidedly 'warm' rather than 'dull', yet it still manages to capture dynamics and transients with a very natural clarity. The stereo imaging is faultless, and when partnered with good mic preamps, the noise floor is extremely low. I used GML and Focusrite ISA428 mic preamps, both offering plenty of gain. I did have to crank the gain to nearly 70dB when recording the small string orchestra, but for all the other sources I rarely exceeded 60dB, so most preamps would probably be able to cope with the R88 in most situations. The Focusrite also allowed some experimentation with input impedances, but I found the best results were obtained with the highest setting. The classic Blumlein stereo technique can deliver superb results in the right settings, although placing the mic in the ideal position can sometimes prove difficult for a variety of tedious practical reasons. In such cases, being able to operate the mic as an MS array — allowing the stereo width to be manipulated — can be very useful. To prove the point, I rigged the mic up to a Yamaha DM1000 digital console and switched a pair of input channels to handle the M&S outputs file:///F|/SoS/SoS%2012-2005/aear88.htm (4 of 5)11/23/2005 3:01:31 PM

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directly. As you would expect, there was a small difference in tonality from the conventional left-right configuration due to the frequency-response variation with incident angle — in the left-right mode, central sources are picked up by the edges of the diaphragms' polar patterns, whereas in the M&S mode, they are picked up directly on-axis to the Middle mic. This small tonal variation might influence me to mic up solo instruments within a nice acoustic space using the M&S configuration rather than the left-right mode — but that would be being very picky! This is a very impressive microphone and one which is without equal as far as I know. It will appeal to the purist classical-music engineer without doubt, but it also has a much wider role in the studio. Long live the ribbon revolution! Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Buchla 200e

In this article:

Buchla 200e

Overview The History Of Buchla & Patchable Analogue & Digital Synthesizer Published in SOS December 2005 Associates Analogue or Digital? Print article : Close window Original Series 100 Reviews : Modular Synth Modules Original Series 200 Modules Patching & Routing Making Connections Alongside Bob Moog, Don Buchla is one of the founding That Reminds Me... fathers of synthesis, and yet much less is known of him Buchla & Evergreen and his instruments. With this two-part review of State College Buchla's latest synth, and a history of some of his Pricing

pioneering work, we hope to redress the balance...

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

Gordon Reid

Despite the success of Hollywood, some icons never cross the Atlantic successfully from West to East. Say 'Babe Ruth' to the average Brit, and you'll conjure an image of a small girl too young to play ball games. Likewise, say 'Don Buchla' in the UK, and you'll probably be asked whether he was a character in The Godfather. Strangely, that description is not as far from the truth as you might imagine. In the USA, there are three 'godfathers' of synthesis: Alan Pearlman, Bob Moog, and — largely unknown in the Auld Country — Donald Buchla. Buchla was a contemporary of Moog, and like Dr Robert, he produced his first synthesizers in the 1960s. He continued to do so throughout the '70s and '80s Photos: Mark Ewing (unless otherwise stated) (see the box on the history of Buchla & Associates overleaf), but the commercial acclaim and recognition afforded to Moog eluded Buchla, and he concentrated on controllers in the '90s. By the early years of this century, he had slipped into the backwaters of the music industry, but in 2002, he decided to reinvent his most successful synthesizer, the Series 200 from the early '70s, bringing it up to date while retaining as much backwards compatibility as possible. Three years later, the result has arrived. It's a feature-packed synthesizer with a staggeringly huge price tag (see the final page of this article). It's the Buchla 200e.

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Buchla 200e

Overview Before looking at its modules, which I'll do in detail next month, there are many global aspects of the 200e that need discussing. That's because there's little about the instrument that's obvious. For example, it's not a modular synthesizer as you would normally use that expression, nor is it an integrated synth, nor is it seminormalled in any conventional sense. Furthermore, despite misleading marketing that describes it as employing 'straight analogue synthesis', the 200e is not a pure analogue synth. But it's not purely digital either, nor is it what we would normally describe as 'digitally controlled-analogue'. It's a hybrid, but not in the same way as other hybrid analogue/digital synths. Confused? I don't blame you. The 200e is remarkably small, taking up about the same amount of room as a Minimoog with its control panel flipped up, but it feels significantly lighter. It comprises three rows of modules mounted in three cases (known as 'boats') arranged in a neat wooden design that flips open for use. When the 200e was first announced, owners of existing Series 200 systems speculated that the two systems would be completely inter-compatible, allowing you to run 200 and 200e modules in the same cases, and off the same power supplies. However, that has proved not to be the case. While the depth of the boats is just sufficient to house the deepest of the 200e's modules, some of the earlier 200-series modules are too deep to fit. There are 16 modules installed in the review instrument (see the detailed picture over the page) but their functions are not always obvious, because Buchla (as on many of his products) describes modules and functions with rather obscure, nonstandard names. But the overriding impression is one of density. There are more knobs, buttons, sockets, LEDs and screens per square inch than on anything else I can recall. This makes the 200e The insubstantial plastic clip holding the incredibly rich in features, and has an 200e together sheared while the review attractive side-effect: it lights up like the system was at the SOS offices. It's certainly Oxford Street decorations at Christmas. not the strongest way to keep the synth Mind you, I would have thought that, on an closed! instrument costing this much, it would have been reasonable to ensure that all the modules lined up perfectly with nicely finished edges, and that it would be finished with solid wood end cheeks (and nicely polished ones, at that) rather than cheaply stained nine-ply. But nine-ply it is. And don't get me started about the cheap plastic clips that hold the 200e closed for transportation (see picture, below right). My concerns about the build quality do not end with the cosmetics. The pots wobble

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Buchla 200e

to an alarming degree, and while I accept that there are 30-year-old Buchlas still working out there, I admit to concerns about the reliability and longevity of the 200e's controls. In the same vein, Buchla has maintained his time-honoured habit of differentiating between audio signals and control signals (as explained in the 'Making Connections' box overleaf) by using 3.5mm sockets for the former and banana sockets for the latter. The banana sockets require a significant amount of force to insert and remove the plugs, which ensures a good connection, but I just feel that, if I push or pull at an angle, something is going to snap. This makes me nervous. Round the back, each boat has two cutouts for I/O sockets. On the lowest boat, both cut-outs are covered with blanking plates. The middle boat has one cut-out blanked off, but the second offers MIDI In and a second — presumably MIDI Thru — socket. I say 'presumably', because neither is marked. There's also a small, blanked-off and unmarked space for a third socket. I suspect that this is for USB, because it lies behind the Model 225e MIDI/USB Decoder module, but once again, there's no legending. The uppermost boat also has one cut-out blanked off, but the second offers four quarter-inch audio signal outputs and a female XLR microphone input (shown opposite). You only get these if you have the Model 227e System Interface Module installed. If it is not, you have to use 3.5mm front-panel sockets as outputs.

The review 200e in full, with the following modules (top 'boat', left to right): Model 260e Duophonic Pitch Class Generator, Model 291e Triple Morphing Filter, Model 266e Source Of Uncertainty, Model 227e System Model Interface, and another Model 291e. The middle boat features a Model 281e Quad Function Generator, a Model 292e Quad Dynamics Manager, a Model 225e MIDI/USB Decoder, a Model 210e Control and Signal Router, and another 281e and 292e. The lower boat contains four Model 259e Complex Waveform Generators, arranged two either side of the impressive Model 249 Dual Arbitrary Function Generator.

There has been some concern expressed on the Internet about the electrical specifications of the 200e, and its ability to interface with other modular synths. This is well founded; the 200e's pitch CV scaling conforms to neither the common 1V-per-octave or Volt-per-Hz standards. To maintain compatibility with the original System 200s, the scaling is a little less than 1.2V-per-octave, so you're not going to be able to take a pitch CV from the 200e and use it to drive other manufacturers' synths, nor vice versa. Indeed, if you don't have the 200e's own MIDI/CV converter, you're going to find it very difficult to play conventional melodies on this synthesizer. Happily, the other voltages lie in standard regions. Control voltages and timing pulses are +5V (signals with sustain) and +10V (transients only), and summed audio signals peak at around 10V peak-to-peak, so these should be compatible with most other manufacturers' devices. However, the 200e's 'wall-wart' power supply is rated at just 12V, which means that — notwithstanding the depth of the boats — the 200e file:///F|/SoS/SoS%2012-2005/buchla200e.htm (3 of 13)11/23/2005 3:01:37 PM

Buchla 200e

is not as compatible with Series 200 modules as many people first thought. That's because some of the original 200 modules ran off 15V rails. Indeed, Buchla's web site admits that the 200 and 200e are only compatible with 'some physical constraints, and occasional power supply restrictions'.

The History Of Buchla & Associates Donald Buchla was born in California in 1937, and proved to be an eclectic talent, with interests in music, physics and physiology, working at various times in the fields of biophysics research, music composition languages, biofeedback and physiological telemetry systems. He even invented aids for visually impaired people. But it is for his developments in the field of music synthesis that he will, perhaps, be best remembered. Aided by a $500 grant from the Rockefeller Foundation, Buchla built his first synthesizer in 1963. Called the 'Model 100 Series' Electronic Music System, this was very different from the instruments being developed contemporaneously by Bob Moog. Buchla's approach concentrated on the development of innovative sounds rather than the performance of traditional melodies. His ideas were (and remain) esoteric, appealing to educational establishments and to avant-garde musicians, rather than to the mainstream. But it would be a mistake to dismiss the Model 100. The '50s had been a decade of fascinating experimentation into electronic music composition and replay, but progress had been incredibly slow, limited by the cumbersome tape-based methods available for sculpting sounds. Buchla's 'voltage-controlled' synthesizer, while abstruse and impenetrable by today's standards, was a huge step forward that allowed composers to control sounds in real time, to connect to external devices, and to reproduce results with a modicum of consistency. In addition to the Model 100 and 101 cases, which respectively hosted up to 15 and 25 of the 7.5 x 4.25-inch modules, Buchla's new company, Buchla & Associates, set about manufacturing an increasing number of building blocks for the 100. I have identified 39 modules in the series (see overleaf for a list). Of special note are

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Photo courtesy of Don Buchla Don Buchla with a Series 100 system in the 1960s.

Photo: Peter Randlette The Buchla Series 100 system at Washington State's Evergreen College, USA.

Buchla 200e

the Model 117 dual proximity detector (with its Theremin-style antennae), the Model 123 and Model 146 sequencers (which could simultaneously control the pitches, amplitudes, and durations of sequences of notes), the Model 148 harmonic generator (which synthesized sounds from their first 10 harmonics), the Model 185 frequency-shifter, the Model 195 Octave format filter (which divided a signal into 10 frequency bands), and the Model 196 phaseshifter. When you consider that the first of these appeared in the year that JFK was assassinated and the Beatles first topped the charts, you get some idea of the pioneering nature of Buchla's early work. Despite this, it was not Buchla's synthesis that differentiated him most from other synth pioneers, nor which limited his penetration into the mainstream. This was determined by his 'Touch-controlled voltage sources'... or, as you and I would call them, keyboards. Buchla was a true disciple of the avant-garde, and did not believe that the potential of his new musical instruments should be limited by the constraints of the 12-note, eventempered octave, or by a black and white piano keyboard. He therefore eschewed conventional mechanical keyboards, and provided pressuresensitive touch-pads such as the Model 112, which had contacts arranged in a straight line, and the Model 113, whose concentric circles generated voltage changes at its outputs rather than absolute voltages. Although you could tune the outputs of the 112 to a chromatic scale, its geometry made conventional keyboard playing impossible, and forced players to think differently about the composition and performance of electronic music. In the May 1994 edition of SOS (see www. soundonsound.com/sos/1994_articles/ may94/suzanneciani.html), Suzanne Ciani (who had at one time worked on Buchla's production line) explained the appeal of this, saying, "I saw the black and white keyboard as an inappropriate interface. With the Buchla synthesizer, I

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Photo: Peter Randlette Evergreen's Series 200 system.

Photo: Peter Randlette 1972's Music Easel.

Photo courtesy of Don Buchla 1971's digitally controlled Series 500 system.

Photo courtesy of Don Buchla The Thunder MIDI Controller.

Buchla 200e

used a touch keyboard, and I could have 20 different things happen and not just one note." Having said that, she then admitted that her Buchla had ended up under her bed after she had sampled its sounds into a Synclavier! Buchla's controllers were well suited to the experimental 1960s, but musicians still wanted to play tunes, and the Model 100 was unforgiving in this regard. Even if you tried, the oscillators were unstable and prone to drift, so Buchlas remained sidelined in colleges, or used by a tiny handful of electromusic pioneers such as Morton Subotnik and Walter (pre-Wendy) Carlos. Even a brief manufacturing arrangement with CBS failed to bring commercial success, and by the time our story enters the 1970s, Buchla's instruments were being heavily outsold by new synthesizers from Moog Music, as well as by newbies EMS and ARP.

Photo courtesy of Don Buchla The Lightning II motion-to-MIDI converter, with two of its wand controllers.

The Model 100 remained in production until 1970 or thereabouts, when Buchla replaced it with the Series 200 Modular Synthesizer. Buchla was now looking to the mainstream, and he bowed to market Photo courtesy of Don Buchla pressure by adding the Model 218 and 219 touch-sensitive keyboards (later Buchla performing sonic alchemy in the early 1970s. superseded by the Model 221) which had their pads arranged in conventional keyboard fashion. Later, he even introduced two polyphonic, velocity- and pressuresensitive mechanical keyboards for the series; the three-voice Model 237 (three octaves) and the four-voice Model 238 (five octaves), the latter of which even incorporated a digital output for connection to mini-computers. But, long before the Model 238, Buchla had in 1971 embraced digital technology and developed the world's first 'digitally controlled analogue' synth, the Series 500. We know that this was based on an Interdata 7/16 mini-computer and that it used a pianostyle keyboard as an event-input device but, other than that, little information exists in the public domain. It's likely that only three were built, and only two are known to have survived. In many ways, the Series 500 was a diversion. It embodied many innovative ideas that would later be refined by other companies, but throughout the 1970s, the Series 200 remained Buchla's only commercially viable product range. To make this more accessible, the company supplied a range of pre-configured systems. The smallest and most affordable of these, introduced in 1972, was the System 200-081 Music Easel. Costing a tad under $3000, and housed in a briefcase much like an EMS Synthi AKS, the Easel's sound generation was provided by a single Model 208 module. This combined an oscillator with voltage-controlled waveshaping, an external signal input, a modulator that provided amplitude modulation and filtering, a noise

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Buchla 200e

source, a contour generator, a clock/envelope generator, an output mixer, and a reverb. The Music Easel could even store patches on 'program cards' — a startling idea in a decade when most musicians' 'patch memories' were pieces of paper with scribbled pictures of knobs and sliders. However, you could only store your patch by soldering the appropriate values of resistors onto the card! Buchla supplied six blank cards and a pack of resistors with each Model 208, leading some to claim that it "offered six memories"! Buchla returned to digital technology for the Series 300, which was a marriage of Series 200 modules and a computer system comprising an 8080 eight-bit CPU, a floppy disk drive, video monitor, interfaces to the synth modules, and a music language — Patch IV — developed for the system. Then, in 1978, he developed these ideas further, with Touché, a duophonic, three-oscillator-per-voice, eight-voice polyphonic, quadraphonic digital synthesizer based on a 16-bit processor and another dedicated music language. This embodied many radical ideas, such as the crossfading of one sound to another, and the ability to record phrases in real time and then loop and transpose them during live performance. At $8500, Touché was more expensive and far less intuitive than a Prophet 5 or Oberheim OBX, so it was never going to be a huge commercial success, despite its groundbreaking technology — and it wasn't. Only a handful of units were made. By the end of the 1970s, Buchla was very much an outsider in the market that he had helped to create. Nevertheless, he continued to push at the edges of synth design. His next instrument, the Buchla 400, incorporated three CPUs, each dedicated to a specific area of sound generation. The first performed housekeeping and userinterface duties. The second controlled the sound generator, and the third — which may have been based on the Touché — generated the sound itself. All of this was controlled by yet another language, 'MIDAS', written in a version of the FORTH programming language. The 400 was capable of interfacing with the Series 200, and many of its functions, such as the display and editing of voices, analogue-style editing, programmable FM, multiple scalings, notation, and its SMPTE timecode facilities, were radical at the time. Its six-voice sequencer was also worthy of note. Designed primarily to play the voices within the 400 itself, this offered note editing, insert, copy, move, loop and delete commands. You could even plug a CRT directly into the 400, and view the sequence in grid form. This was in 1982, a year before MIDI, and long before home computers (let alone Macs and PCs capable of running sequencers) became commonplace. In 1987, the 400 was replaced by the Buchla 700, which for a long time appeared to be the last of its dynasty. This incorporated four CPUs, the extra one of which handled incoming/outgoing analogue and digital data, including that from dual RS232 serial ports and multiple MIDI inputs and outputs. More powerful than the processors on the 400, these allowed Buchla to increase the number of sound-generating variables and their resolution. Nevertheless, the 700 was still recognisably a Buchla, with touch-sensitive pads rather than a keyboard, and (I quote from the brochure) 'position-sensitive transducers used to implement conceptual potentiometers, flywheels, switches, ribbon controllers, and other gesture-sensitive paraphernalia'. By the end of the 1980s, Buchla had started to turn away from synthesis itself, and all his products in the 1990s were controllers of one sort or another. The DSP-based Thunder MIDI controller provided a hexagonal playing surface with 26 touch-sensitive pads, and this was complemented by the following year's Lightning and its successor, Lightning II. These units translated the motion of two hand-held wands into MIDI information that could be transmitted and/or used to control an internal 32-voice synth. The final Buchla of the 20th century was the Marimba Lumina, a self-contained file:///F|/SoS/SoS%2012-2005/buchla200e.htm (7 of 13)11/23/2005 3:01:37 PM

Buchla 200e

instrument comprising sets of illuminated bars, pads and strips played using mallets. Inevitably, it had a few facilities not available on traditional marimbas, including an internal synth and the ability to recognise which of four types of mallet had struck any given bar, each having a different, programmable action upon the sound. A smaller version, the Marimba Lumina 2.5, was released in 2002. Shortly after this, Buchla elected to return to his designs for the Series 200, which eventually resulted in the release of the 200e under review here.

Analogue or Digital? The ability to patch the 200e as a conventional, analogue, modular synthesizer does not mean that it is a conventional, analogue, modular synth. The documentation describes the 200e as an analogue synthesizer because, as it states, "we are describing the aspect that the user contacts as analogue". In other words, because the 200e presents you with knobs and analogue patch points, it's an analogue synthesizer. Many people have great difficulty accepting this, and I count myself as one of them. As Buchla admits, the sound generation itself is an ad hoc mix of analogue and digital techniques, and the use of it was to some extent determined by economic factors, as well as by the obsolescence of some of the components used in the original 200-series modules. Take the 259e Complex Waveform Generators as an example. Buchla states that 'there are no compelling reasons to employ analogue circuitry in the oscillators', so both oscillators within a 259e are digital. In essence, only their controls and outputs are analogue. The same is true for the 260e Shepard tone generator and the various S&H sources in the 266e Source Of Uncertainty. In contrast, many of the remaining modules use digital control signals coupled to analogue signal paths. These facts alone are enough to cause paroxysms in analogue purists, but what limited information that exists about the digital side of the implementation also concerns me. That's because, while Buchla claims that there is no zipper noise in the 200e, he stated some time ago that control parameters are quantised at between eight and 12 bits, depending upon their purposes. The FAQ page on The audio outputs and XLR audio input on Buchla's web site states that the range of the rear of the review 200e's top 'boat'. resolutions in the ADCs and DACs in the 200e is six to 16 bits, but I understand that the 16-bit converters are used only for audio signal conversion, not CVs. Anyway, although 12 bits are adequate for many functions, they may be insufficient for others. To illustrate this, consider the example of two oscillators tuned to almost, but not exactly the same pitch. The differences in beat speeds between subtle degrees of detune is extremely important when creating 'chorused' timbres and, depending upon how demanding you are, even 12-bit resolution could be insufficient for adequate control over such sounds. As regular readers of SOS will know, I'm not an analogue purist. Provided that it's

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Buchla 200e

advanced enough, I don't think that the use of digital technology in the audio path is anything to be ashamed about, especially when it offers sound-generation opportunities that would not otherwise be possible. Consequently, I'm mystified as to why Buchla — who has been designing digital oscillators since the Model 500 in 1971, and who seems willing to use the most appropriate technology for the task — would seek to obfuscate the issue by describing the 200e as 'straight analogue synthesis'.

Original Series 100 Modules

Original Series 200 Modules

100 Cabinet.101 Cabinet.102 Dual stereo locator.106 Six-channel mixer.107 Voltage-controlled mixer.110 Dual voltage-controlled gate.111 Dual ring modulator.112 Touch-controlled voltage source.113 Touch-controlled voltage source.114 Touch-controlled voltage source.115 Power supply.117 Dual proximity detector.120 Distributor.123 Sequential voltage source.124 Patch board.130 Dual envelope detector.132 Waveform synthesizer.140 Timing pulse generator.144 Dual square wave oscillator.146 Sequential voltage source.148 Harmonic generator.150 Frequency counter.155 Dual integrator.156 Dual control voltage processor.157 Control voltage inverter.158 Dual sine/sawtooth oscillator.160 White noise generator.165 Dual random voltage source.170 Dual microphone preamp.171 Dual instrument preamp.172 Dual signal leveller.175 Dual equaliser line driver.176 Dual hiss cutter.180 Dual attack generator.185 Frequency-shifter.190 Dual reverberation unit.191 Sharp cutoff filter.192 Dual low-pass filter.194 Band-pass filter.195 Octave format filter.196 Phase-shifter.Note: the modules adorned with a red star are not shown on Buchla's own web site, but other sources suggest that they existed nonetheless.

204 Quad spatial director.205 Matrix mixer.206 Dual mixer.207 Mixer/ preamp.208 Stored program source.212 Dodecamodule.217 Touch keyboard.218 Touch keyboard.219 Touch keyboard.221 Kinesthetic input port.226 Quadraphonic monitor/interface.227 System interface.230 Triple envelope follower.232 Frequency detector.237 Polyphonic keyboard.238 Polyphonic keyboard.245 Sequential voltage source.246 Sequential voltage source.248 Multiple Arbitrary Function Generator (MARF).256 Dual control voltage adder.257 Dual control voltage processor.258 Dual oscillator.259 Programmable complex waveform generator.264 Sample & hold/polyphonic adaptor.265 Source Of Uncertainty.266 Source Of Uncertainty.270 Quad preamplifier.275 Dual reverb/ equaliser.280 Quad envelope generator.281 Quad function generator.284 Quad voltagecontrolled envelope generator.285 Frequency-shifter/balanced modulator.291 Dual voltagecontrolled filter.292 Quad low-pass gate.294 Four-channel filter.295 10channel filter.296 Programmable spectral processor.

Patching & Routing

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Buchla 200e

What's not apparent until you start to use the 200e is that there are 14 busses within the synth, some provided as patch points, and others running between the modules and boats along cables tucked away within the chassis. You manage and control these from the upper panels on the Model 225e MIDI/USB Decoder. This is, for most purposes, the heart of the synth. Ten of the busses provide voltages derived from MIDI/CV conversion, and all of these appear on banana-socket outputs on the face of the 225e. Those named E, F, G and H are transposable note busses that respond to individually user-defined MIDI channels and velocity curves, and each offers pitch, velocity and gate outputs. The other six (J, K, L, M, N and P) each provide the analogue equivalent (with a zero to +10V range) of two user-selected controllers derived from the channel chosen for each. Unfortunately, although aftertouch is one of the menu options, it does not seem to be functional on this system. Indeed, there seem to be a number of unfinished functions and bugs in the review unit — the manual even admits that 'MIDI is only partially implemented. We'll finish soon.' The other four busses (A, B, C and D) are not accessible via patch points but, with all the appropriate Remote Enable switches on, are hard-wired to their destinations, conveying (on the SOS review configuration) pitch information to the four 259e modules, velocity information to the A, B, C and D sockets on both 292e modules respectively, and Gate signals to the A, B, C and D sockets on both 281e modules respectively. In many ways, these busses are the keys to the 200e, because they cause the oscillators to track incoming MIDI notes, they trigger the contour generators, and they provide velocity information to the combined filter/amplifier modules. They should also respond to pitch-bend messages, but, again, this would appear not to be implemented yet.

The review 200e when closed up. This position allows you to see the MIDI sockets on the underside of the middle 'boat'. The handle is also on the back of the middle boat, and the synth is held closed by the plastic clips, one of which can be seen at the right edge of the synth. When opening it out, you would undo the clips and pull the top boat shown here towards you, and then flip the hinged wooden supports on the back outwards so that they hold up the middle and top boats. The 200e would then be open facing away from you.

Of course, you won't get a peep out of the 200e unless you patch its modules together (the bussing provides only the converted MIDI control signals, not the sounds and CVs themselves) but with the busses set up correctly and the right connections between the modules, you're in business. For example... Given that you can set each of A, B, C and D to an independent MIDI channel, and that this 200e system incorporates four primary oscillators, eight contour generators and eight filter/ amplifier pairs, it's simple to patch it as four independent monophonic synthesizers. Alternatively, set each of the A, B, C and D busses to the same MIDI channel and to 'Poly', and incoming notes will be distributed correctly to the oscillator/contour/ file:///F|/SoS/SoS%2012-2005/buchla200e.htm (10 of 13)11/23/2005 3:01:37 PM

Buchla 200e

amplifier sets of modules, which you can then mix into a conventional four-voice polysynth. Once you've done so, you'll be able to develop sounds that are unlike anything you'll obtain from any simple, integrated MIDI synthesizer.

Making Connections One significant difference between Buchla's approach and that of Bob Moog was his separation of the signals used in synthesis into three distinct classes. First, there were the audio signals, which could be generated by oscillators, or injected into the system from devices such as microphones or tape machines. Secondly, there were the control voltages. Finally, there were timing pulses, which we nowadays call clocks, gates, and triggers. By today's standards, the audio levels were quite low (about 1V peak-to-peak) and the CVs and pulses were rather hot, with a maximum voltage of around 15V, but the strangest thing about them was that Buchla used different types of sockets for each class, so that you couldn't interconnect them. In contrast, Moog saw every signal as simply a signal, without differentiation, and it was this approach that would later become the overriding model of analogue, subtractive synthesis. While Buchla accepted that there were advantages to the non-differentiation of sounds and the signals controlling them, he justified his approach on engineering grounds: specifically, that if a signal has to work in both ways, the circuitry has to be a compromise. For example, he suggested that DC offset is irrelevant in the audio domain — a view that I don't necessarily accept — but is important in the control domain. Conversely, he stated, a certain amount of harmonic distortion is largely irrelevant in the control domain, but has obvious consequences in the audio domain. Those arguments have some merit, but his assertion that using two types of sockets and cords made it easier to see what was going on is, to me at least, more questionable. On the 200e, the CV and timing sockets are colour-coded as follows: the CV inputs are black and grey, and the CV outputs are blue, violet, and green. The pulse inputs are orange, and the pulse outputs are red. There seem to be no differences between the colours used for CV inputs, and between those used for CV outputs, so I suspect that the multiple colours were chosen for nothing more than aesthetic reasons.

That Reminds Me... It's clear that this 200e system is not only a fantastically powerful modular monosynth, it can be a four-part multitimbral synth, or a four-voice polysynth. But hang on... haven't I seen this before somewhere? Yes, I have. The philosophy of the 200e is remarkably similar to that of the Oberheim 4-Voice, the world's first 'integrated' polysynth, and an instrument that was revered as a technological marvel when launched in 1974. Some of these Oberheims were later modified with individual patch-points in each of their SEM monosynth modules, so the parallels are even closer. With this in mind, I thought that it would be interesting to compare the prices of the two. The earliest retail price I have for the 4-Voice is around $5500, which equates to around $40,000 at today's prices, and is therefore double the asking price of the file:///F|/SoS/SoS%2012-2005/buchla200e.htm (11 of 13)11/23/2005 3:01:37 PM

Buchla 200e

200e. Of course, the Buchla offers facilities undreamed of in 1974, and many of its modules could justify reviews in their own right. This isn't possible, but next month I'll look at each of them in turn, patch a few sounds, and try to draw some sort of conclusions about this remarkable synthesizer.

Buchla & Evergreen State College Founded in 1971, Evergreen State College is a liberal arts and sciences college in Washington state, in the USA. Since the earliest days of the college, music technology has been part of its curriculum, and it has always offered its students access to recording facilities. There are three acoustic recording studios with classic API desks, plus the four so-called Electronic Music Labs. The latter rooms are packed with equipment from all ages of music technology, including Oberheim, Peter Randlette with Evergreen College's ARP, Moog and Emu analogue synths, Series 200 Buchla synth. analogue multitracks, and computers running MOTU's Digital Performer, as well as many other synthesis and recording packages. Three of the labs have had rare Buchla synths in them for years, which are still in everyday use: a Music Easel, a Series 100, and a large Series 200 system. Evergreen Media Services staff member Peter Randlette, 50, arrived at Evergreen as a student in 1975 and now runs the Music Technology Labs at the college. He's been responsible for looking after the Buchlas and keeping them in working order for many years. He even acquired the Series 100 from the University of Washington for the college and oversaw its refurbishment and reassembly. It's necessitated a trip or two to Buchla's laboratory and much scouring the world for spare parts, but he says it's all been worth it. "The students connect with these instruments immediately," he says. You can spend a morning with them explaining how Performer works, whereas when you put them in front of the Series 200, they get it straight away, which is a testament to Don's user interface design." The college now also owns a 200e. "It sounds amazing, and the DARF [Dual Arbitrary Function Generator] is a real hit with the students," says Peter. For more on the vintage goodies at Evergreen, see www.evergreen.edu/media/musictech/home.htm.

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Buchla 200e

Pricing As the 200e is a modular system, there is no set price for it, but it's fair to say that the constituent modules and their case do not come cheap. It's sold in the UK by distributors RL Music, and a detailed, module-specific price list is available from their excellent web site, www.rlmusic.co.uk. However, there are no UK sterling prices for the 200e — the prices are shown in dollars, so the cost to UK customers fluctuates with the exchange rate. What's more, the prices shown on the web site do not include UK customs duty, which is payable, nor UK VAT at 17.5 percent, nor the cost of transporting your purchase to the UK from California — and safely shipping a large, delicate system to the UK like the configuration SOS used for review could cost quite a bit. Suffice it to say that the total cost of the modules and case comprising the SOS review system comes to a not-insignificant $19,850. At the time of going to press (late October 2005), this equates to approximately £11,115. And don't forget, that's not including UK customs duty, VAT, or shipping costs. It's certainly not what you'd call an impulse purchase! While it's safe to say that there are few people the world over who will pay for this kind of system, many more modular aficionados might be keen to add a little of the 200e's unique character to existing setups. Understanding this, Buchla & Associates also supply four smaller cabinets. As the numbers suggest, the 201e12, 201e6, 201e4 and 201e2 will house 12, six, four or just two standard modules. Perhaps the most interesting of these is the 201e6, which is the middle 'boat' from the full 200e18 pictured here. This costs $700 (currently around £400 excluding shipping, VAT, and UK duty), and I understand that, if you later upgrade to a full system, the company will allow you to trade this in for credit against the purchase of the three-boat cabinet, currently quoted at $1400 (about £800 without the shipping, VAT, and duty). Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Cakewalk Sonar 5

In this article:

Cakewalk Sonar 5

Overview MIDI + Audio Sequencer Manual Choices Published in SOS December 2005 Getting Started 32 Bits Good, 64 Bits Better? Print article : Close window New Features Reviews : Software Not-so-new Features Spaced Fever Pitch The Fifth Element

Cakewalk Sonar 5 £369/ £229 pros Powerful, comprehensive and flexible DAW package. V-Vocal pitch correction is well implemented and effective. Perfect Space convolution reverb sounds very good. Additional virtual instruments are a bonus. 64-bit compatibility there for those who want it.

cons User interface can seem cluttered. The five new instruments aren't entirely new.

summary Sonar 5 adds some nice extras to the already powerful Sonar package, along with some useful user-interface enhancements. It's an undoubtedly impressive package, although it may not seem like an essential upgrade to every Sonar user.

information Producer Edition £369; Studio Edition £229. Prices include VAT. Edirol Europe +44 (0)20 8747 5949. +44 (0)20 8747 5948.

[Windows]

The new version 5 sees Cakewalk's Sonar becoming a more complete production package than ever, with the addition of new synths, a convolution reverb, 64bit support and Roland's celebrated Variphrase vocal processing technology. Paul Sellars

Less than a year ago, in these very pages, Derek Johnson began his review of Cakewalk's Sonar 4 with the observation that less than a year had elapsed since the release of Sonar 3. What, he wondered, might have happened to justify such a bold and rapid whole-number increment? Now, less than a year later, I find myself faced with Cakewalk Sonar 5. Where did the time go? What further developments will the intervening months have brought? Where will it all end? Restricting myself to the answerable questions, I can tell you that among the key new features of Sonar 5 are a collection of software instruments (synths, a Soundfont sampler, a REX file player), a new convolution reverb, updated MIDI effects plug-ins, a new 64-bit 'double precision' floating-point audio engine, support for 64-bit processor architecture and operating systems, the integration of Roland's Variphrase vocal processing technology, enhanced MIDI step recording, and a range of minor user-interface refinements. It's still recognisably the same application as Sonar 4, though: this is a case of enlargement rather than reinvention, if you see what I mean. It's worth mentioning that many of the new gadgets and features described here

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Cakewalk Sonar 5

www.edirol.co.uk www.cakewalk.com

Test Spec PC with 1.8GHz Athlon CPU and 512MB RAM, Emagic Audiowerk 2 soundcard, VIA onboard audio hardware and MOTU Fast Lane MIDI interface, running Windows XP SP2.

are exclusive to the Producer Edition of Sonar 5. The more affordable Studio Edition offers fewer of the high-end whistles and bells (see www.cakewalk.com/ products/sonar/studio.asp for more details), although it's still a very creditable package in its own right.

Overview For the benefit of any newcomers, let's run through the basics. Sonar is a powerful Digital Audio Workstation (DAW) application for Windows XP, designed to handle just about every audio and MIDI task you might think of, from MIDI sequencing to audio recording, editing, mixing and beyond. Unlimited MIDI and audio tracks are available, at sample rates as high as your soundcard can cope with. VST and Direct X plug-ins are supported, with full delay compensation, and Rewire instruments can also be used. There are sync-to-video capabilities, powerful loop-based composition tools, and plenty more besides. As such, Sonar is competing for the same ground as applications like Cubase and Samplitude, or on the Mac side, Logic and Performer. In fact, while finding my way around Sonar, I experienced a couple of disorientating Cakewalk's MIDI FX have been overhauled, flashbacks to the Samplitude review I with a smart new look. wrote for SOS a while ago. This has less to do with any particular similarities between Sonar and Samplitude, I think, than it does with the way in which all the major DAW applications seem to be converging on a common feature set, and on implementations that are at least superficially similar. Sonar is supplied on a single DVD-ROM, which auto-runs to show a very nicely presented installer menu where you can choose to install the application proper, a few additional utilities, some sample/loop content and various other bits and pieces. After choosing to install the main Sonar 5 application, you're presented with a dialogue box showing the customary End User Licence Agreement, beneath which you're required to tick a couple of boxes, the first acknowledging that you are only permitted to install and use the software on one machine at a time, the second acknowledging that you are not permitted to sell or transfer the software. You may or may not be happy with these licence terms. If you're not happy with them, you unfortunately have no other option besides not installing the software. Having agreed to the licence, you then have to enter your serial number, which is supplied in the DVD case. After that, the final hurdle to clear is registration, which must be completed within a 30-day 'grace' period. Registration is easy if your audio computer has an Internet connection, and requires a phone call if it doesn't.

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Cakewalk Sonar 5

At least there isn't a dongle.

Manual Choices As with previous versions of Sonar, customers in the UK can choose whether to buy the US or European version of the package. The difference is that the US version ships with a full printed manual in English only, while the European version includes only the more basic 'getting started' guide, but in French and German as well as English.

Getting Started The first time Sonar is started, a dialogue box appears offering to run some diagnostics on your audio hardware and make some default settings. Sonar 5 uses Cakewalk's VST Adapter (version 4 of which is included in the bundle) to enable VST plug-in support, although it's more closely integrated than in previous versions, and will automatically scan, load and configure any new plug-ins when Sonar is started. The basics of Sonar are fairly straightforward, and should be more or less familiar to anyone who's worked with a MIDI and audio sequencing package before. Sonar's Track View shows a list of the MIDI and audio tracks in the project, while an 'inspector' pane displays more details about the currently selected track. Tracks run horizontally from left to right, and are populated with 'clips', which may be short, single-hit sounds, or extended takes. Groove clips are audio clips which have pitch and tempo data stored in them, as in Sony's Acid, and Sonar can import and export Acidised WAV files. MIDI events and data are also stored in 'clips', although these are handled slightly differently, for obvious reasons. The Console View is a window containing the customary graphical representation of a mixing desk, with all the virtual faders, knobs and so on laid out much as you'd expect them to be. It's actually quite possible to mix tracks without ever opening the Console View, as the inspector pane in the Track View provides a pretty complete 'channel strip' for the currently selected track. Of course, it's sometimes useful to be able to see all your faders at once, and the Console View allows this.

RGC Audio's Pentagon I is a supremely flexible and good-sounding virtual analogue soft synth.

Sonar's a complex application, and at first glance can seem slightly cluttered. file:///F|/SoS/SoS%2012-2005/cakewalksonar5.htm (3 of 12)11/23/2005 3:01:47 PM

Cakewalk Sonar 5

Fortunately the detailed and rather weighty printed manual contains several clear, step-by-step tutorials, which help clarify things for the beginner. Sonar's user interface perhaps still has a slightly steeper learning curve than other similar applications, but once you've become familiar with where everything is, it begins to seem quite logical. While by no means restricted to loop-based composition, Sonar is very well equipped to deal with looped clips, of both the audio and MIDI variety. If you're familiar with Sony's Acid, you'll feel right at home with Sonar's handling of loops. Matching the tempo and pitch of Groove clips is made very easy, and Sonar's inbuilt tools handle the business of creating or importing Acidised clips admirably. Recording MIDI and audio tracks is quick and easy, and the Folder track facilities in the Track View allow you to assemble even quite large and complicated arrangements without creating too confusing a mess. The Console View is reasonably clear and intuitive, although I personally found the controls for the built-in EQ a little fiddlier than they need have been. Overall though, Sonar provides a comfortable environment in which to work. Since I have only limited space, I'll be concentrating on the new features added in Sonar 5, although there's still plenty that could be said about some of the older features. Sonar's surround mixing facilities, for instance, are very well implemented. The Sonitus FX bundle is also impressive, as is the Lexicon Pantheon reverb plug-in. For a more thorough look at Sonar up to and including version 4, I'd recommended Derek Johnson's January 2005 review at www. soundonsound.com/sos/jan05/articles/sonar4.htm and, of course, the regular Sonar workshops in this and every issue of SOS.

32 Bits Good, 64 Bits Better? One potential source of confusion around Sonar 5 has to do with its 64-bit features. Sonar 5 ships in two different versions, both included in the same package. There's a 32-bit Windows application of the kind we're all used to, and a 64-bit version aimed at users running the 64-bit version of Windows XP on a computer with 64-bit processor architecture. Personally I'm still languishing in the 32-bit Dark Ages, and in all probability so are you. Nevertheless, 64-bit systems are apparently on their way, and the computer industry being what it is, we'll probably all find ourselves having to upgrade eventually. By building a viable 64bit version now, Cakewalk have ensured both that the application is as futureproof as possible, and that eager 'early adopters' have a strong incentive to either stick with, or defect to, Sonar when making the 64-bit switch. One of the new features that Cakewalk are keen to advertise in Sonar 5 is their new '64-bit double-precision floating-point audio engine' which, confusingly, is available to both the 32-bit and 64-bit applications. In the 32-bit version, 64-bit mixing can be activated by selecting Audio from the Options menu and activating the 64-bit Double Precision Engine tick box in the dialogue box that appears. The 64-bit audio engine should, in theory, offer a superior signal-to-noise ratio and improved dynamic range. This applies not to recording, where the same 16, 24 file:///F|/SoS/SoS%2012-2005/cakewalksonar5.htm (4 of 12)11/23/2005 3:01:47 PM

Cakewalk Sonar 5

and 32-bit options are available, or playback, but exclusively to mixing. The extra bits of resolution are intended to provide better performance when combining multiple signals and when scaling (ie. adjusting the volume or panning of) signals; the 64-bit mixing engine provides extra headroom, so that rounding errors should occur less often when mixing projects containing large numbers of tracks. Theoretically, at least, the improved resolution should provide improved sound quality, but in practice the perceived difference can be very slight. To be honest, I struggled to find much to distinguish mixes rendered with the 64-bit option enabled from those rendered without. By and large, both sounded equally good to me. People with more sensitive ears and more demanding requirements may be able to discern a more marked difference, however. One final point to bear in mind here is that many of the third-party effects plug-ins inserted at various points in your signal paths will be incapable of handling 64-bit audio. Fortunately Sonar can tell which plug-ins need to be fed a 32-bit diet, and will make the necessary arrangements for you.

New Features One of the user-interface enhancements introduced in Sonar 5 is the new Inline Piano Roll View, which allows MIDI notes and controller data in a track to be viewed and edited directly from within the Track View. Simply choose a MIDI track and click the the 'PRV mode' button, and the clip redraws itself as a miniature piano-roll display containing the data in that track. When the PRV mode button is activated, a small piano-roll toolbar is displayed, which allows you to choose which kinds of MIDI data are hidden or displayed, and how editing is handled. It's all straightforward, and quite easy to use. A little judicious zooming of the Track View is required to make the data comfortably visible, though, and some Sonar users may decide it's actually quicker and more convenient to double-click a MIDI clip and have it open in the Piano Roll View 'proper', as in Sonar 4. Still, it's nice to have the choice. Another nifty 'workflow' enhancement comes in the form of Sonar 5's Track Templates feature, which allows you to create templates for recalling groups of track settings. This may not sound very exciting but it can be quite handy. For example, you might have a favourite combination of distortion and With the RXP soft synth, you can take a compression plug-ins you like to use sliced loop, apply different processing to each slice and trigger the slices for recording lead guitar parts. In this independently over MIDI. case you could create a 'lead guitar' Track Template, with the desired plugins already inserted and the EQ tweaked just as you like it. Then, whenever you want to record a lead guitar take, you can simply go to the Insert menu, select Insert From Track Template and recall your saved template.

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Cakewalk Sonar 5

Track templates can store information about the track type, the mute, solo or record state of the track, its hardware input and output destination, any buss send settings, effects settings, instrument bank and patch settings and track name. If you habitually assemble arrangements in a certain way, Track Templates can serve as genuinely useful shortcuts, saving you plenty of mouseclicks. A simple idea, but nonetheless useful. It's also now possible for real-time effects to be non-destructively applied on a per-clip basis. In other words, you can right-click a clip (or clips), choose Insert Effect from the context menu that appears, and create an 'FX bin' for that clip. Depending on whether you're working with MIDI or audio clips, you choose from either MIDI or audio effect plug-ins. Clip effects can toggled on and off, and if more than one effect is loaded, you can change their order. It's also possible to destructively apply clip effects, in which case the real-time clip effects are automatically removed afterwards. Speaking of MIDI effects, the Cakewalk MIDI FX plug-ins have all been overhauled for Sonar 5, and now sport improved user interfaces designed to make them quicker and easier to set up. Window clutter can be reduced by enabling a new 'tabbed' option which allows any of Sonar's windows (except the Console View) to be neatly 'docked' in the bottom right-hand corner of the screen. When multiple windows are docked you can switch between them either by clicking their tabs (shown along the bottom of the screen) or via a keyboard shortcut (Ctrl + Shift + the left or right arrow). Another visual enhancement is the new waveform preview mode for busses and virtual instrument tracks. When this option is enabled, a waveform of the buss or track's audio output is drawn in real time while the song plays. The waveform stays drawn after playback, and appears in red wherever clipping occurs. This is either frivolous eye candy, or an elegant refinement of the user interface, depending on your point of view. Personally, I quite like it. Sonar's automation features have also been improved. The Envelope Draw tool now allows proper freehand editing, and can also be used to create preset shapes (sine, triangle, saw, square and random), which can produce temposync'ed LFO-type effects such as tremolo or auto-panning.

Not-so-new Features Although the Cakewalk web site is keen to advertise five new virtual instruments included with Sonar 5, little or no mention is made of them in the manual. There are some general observations about working with soft synths, and a quick nod in the direction of the Sound Canvas-esque TTS1 from Sonar 4, but beyond that you're basically left to discover things for yourself.

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Cakewalk Sonar 5

One of the first things I discovered was that two of the five new instruments aren't new at all, and are available independently of Sonar. Both the Pentagon I virtual analogue synth and the SFZ Soundfont sampler were developed by independent software developers RGC Audio; Cakewalk have bought up RGC, but they continue to distribute Pentagon and SFZ from their web site at www.rgcaudio. com. There's nothing wrong with this, of course, and although the SFZ Soundfont player is distributed as freeware, Pentagon I usually sells for 99 Euros. If you factor this into the cost of, say, upgrading to Sonar 5 Producer Edition from Sonar 4 Producer Edition (£119), the upgrade looks all the more attractive. Even so, it's as well to be aware of your options. Pentagon I is certainly an impressive beast, however you arrive at it. Its user interface is crammed with a startling quantity of knobs (one hundred and three, if I counted correctly), which will terrify the faint-hearted and delight the most dysfunctionally obsessive twiddlers. It features four independent oscillators, each offering an impressive 13 'alias-free' waveforms including saw, sine, square and pulse waves, as well as noise. There are independent LFOs for pitch, pulse-width modulation, filter and amp, allowing for very complex parameter modulation within a patch. There are some good on-board effects, including a very nice-sounding chorus, a tempo-sync'ed delay, simple EQ, a 'Drive' effect with Gain and Tone parameters, and a clever formant filter effect, which can be used to create surprisingly voice-like vowel sounds. There's also an amp and cabinet simulator, and an unusual Voice Modulator option, which allows you to modulate a patch with your voice or any other audio signal, using Pentagon I much as you would a vocoder.

Cakewalk's partnership with Roland means that many classic Roland sounds are now available to Sonar and Project 5 users courtesy of the software Groovesynths.

Pentagon I definitely has more to offer than just a load of knobs. It also sounds very good: fat and warm, and often quite convincingly 'vintage' and 'analogue'. The built-in presets are all very good, and ably demonstrate Pentagon I's flexibility. There's a lot this synth can do, and while I wasn't able to spend as much time with it as it really deserved, I nevertheless came away from Pentagon I impressed by its character. Sticking with RGC Audio, the SFZ Soundfont sampler is a good deal simpler, but nonetheless effective. It can load and play standard WAV files, OGG compressed files, SF2-format Soundfonts, and instruments saved in its own SFZ format. It supports direct-from-disk streaming and offers a multi-mode filter, a couple of LFOs, on-board chorus and reverb effects, and more. It's an impressive instrument, which doesn't demand much in the way of CPU resources, and is very easy to use. You needn't take my word for it though: SFZ can be downloaded for free from www.rgcaudio.com/sfz.htm. file:///F|/SoS/SoS%2012-2005/cakewalksonar5.htm (7 of 12)11/23/2005 3:01:47 PM

Cakewalk Sonar 5

The Psyn II subtractive synth is unique to Cakewalk, but not to Sonar 5: it is also included in their Project 5 all-in-one sequencing package. Psyn II is a more straightforward synth than Pentagon I. It features two oscillators, each with a choice of six waveforms, which can be configured for ring modulation and frequency modulation. There are five envelope generators and three assignable LFOs, allowing plenty of flexibility in terms of modulation. Cakewalk's web site suggests that Psyn II may be suitable for 'rap, hip-hop and dance musicians who need warm and edgy bass and lead sounds', and indeed it may be. It's certainly a functional little synth, capable of producing some good, usable sounds. Compared to some of the sounds produced by Pentagon I, though, Psyn II's output occasionally struck me as a little too 'polite' or 'measured'. I can't offer any kind of rational justification for this: it's purely a subjective impression. I'm not saying that Psyn II doesn't sound good. I'm perhaps just saying that Pentagon I tends to sound better, from where I'm sitting. The Roland Groovesynth also ships with Project 5, but is still a welcome addition to Sonar. It's a simple, sample-based sound module that features '100 percent genuine Roland sounds from their genre-defining grooveboxes and synthesizers'. In practice this means a good selection of clean and usable sounds, including pianos, basses, organs, strings and plenty more. Best of all, for my money, are the drum machine sounds, which include all the obligatory 808 and 909 hits, along with various others. These are well known — even arguably over-used — sounds, but they still have plenty of character, and it's nice to have them so conveniently available 'on tap'. The RXP REX Player 'groovebox' is described as a 'tempo-sync'ing drum machine and groove box that plays REX and SFZ loops and single hits', which pretty well sums it up. It allows you to trigger individual hits from within a Recyclesliced loop with MIDI notes. You can rearrange the order of slices within a loop, pitch the whole loop up or down, and even set a pitch randomisation factor for slices. There are envelope generators and resonant filters which allow you to shape the sound of the slices. MIDI sequences can be extracted from REX files and imported into Sonar by simply dragging and dropping from RXP, allowing for more flexible editing of patterns and phrases. This is a nice touch, and it works well. If you're not interested in working with sliced loops, RXP probably won't be of much interest to you. If you are, on the other hand, it could prove to be a useful addition.

Spaced Moving from instruments to effects, for me one of the highlights of Sonar 5 is the new Perfect Space convolution reverb. Convolution reverbs work by capturing

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Cakewalk Sonar 5

the reverberant properties of a space in an 'impulse response' file. An impulse response is typically just a standard WAV-format audio file, often created by firing a starting pistol in the desired location and recording the resulting sound. A convolution reverb processor uses the data in an impulse response file to calculate what kind of reverberation happens in a particular space, and employs some frightening mathematics to apply the reverberant characteristics stored in the impulse response to incoming audio signals. Native convolution reverb plug-ins have been growing in popularity ever since Altiverb first appeared for the Mac a few years ago, and many Sonar fans will no doubt be delighted by the arrival of Perfect Space.

One of the major selling points of the Producer Edition of Sonar 5 is the Perfect Space convolution reverb.

In practice, it's not a difficult effect to use. You simply insert the plug-in in a track or buss, load an impulse response file (a good selection is included, and more are available from www.noisevault.com), and away you go. The quality of the reverb is immediately impressive, and the supplied impulse responses cover all the required bases from the conventional ('Blues club', 'chapel') to the bizarre (springs, bathrooms, maracas). At their best, the 'real' room sounds really do sound, well, real. The more unusual impulses can also yield some interesting effects — although for my money it's the realism of the rooms and halls that's most impressive. You can tweak the wet/dry balance of the effect, adjust the length and offset of the impulse response, or even reverse it. More complex edits are possible by enabling and tweaking one or more envelopes on the impulse. There are envelopes for volume, width, pan, low- and high-pass filtering, and EQ. These are non-destructive, in that they only affect the loaded impulse data, not the underlying file. A fairly mind-boggling variety of different effects can be produced once you begin experimenting like this — and that's before you start downloading new impulses, or creating your own. One other parameter worth mentioning here is the global latency control which sets the internal processing delay, in samples. Convolution reverbs, as a breed, tend to make a fairly substantial dent in your CPU power, and Perfect Space is no exception. Longer latencies reduce the CPU hit at the price of making the plug-in less usable during tracking. The lowest available latency is 64 samples (1.5ms at 44.1kHz), which is certainly snappy enough for real-time use. You wouldn't want too many instances open at this setting, though, as the processor load is considerable. Sonar's Freeze Track function (which automatically creates a bounced track with effects applied) could come in handy here!

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Cakewalk Sonar 5

Fever Pitch Another impressive new addition is V-Vocal, which is based on Roland's Variphrase technology (it's the first time this technology has been made available in a software product) and has been designed to perform pitch, time and formant manipulation on monophonic sounds, particularly vocals. It's quite similar, in both concept and presentation, to Celemony's Melodyne. To use V-Vocal, you simply select some audio data in the Track View, right-click and choose Create V-Vocal Clip in the pop-up menu that appears. Sonar automatically creates a new clip containing a copy of the selected audio, and it's this clip rather than the original that V-Vocal is applied to. The V-Vocal editor window is large and well laid out, and quite easy to work with. The are four distinct modes, to allow control over four different aspects of the sound: pitch, time, formant frequencies and dynamics. Pitch corrections can either be performed 'manually', by setting specific target notes in a pitch curve, or 'automatically', by nominating a scale which the clip will be corrected to fit. Variable-depth vibrato effects can be created, and a sensitivity control allows you to fine-tune the handling of material containing ambiguously pitched notes or phrases. You can audition your tweaked clip from within the V-Vocal interface, with a useful looped playback option available. As you'd expect with an effect of this type, a certain amount of restraint is required in order to achieve really natural-sounding results. That said, VVocal works very well, and is capable of sounding very 'clean' over quite a wide range. With a bit of care, it's possible to perform significant pitch corrections that are all but undetectable. Conversely, if you throw caution to the winds and turn all the metaphorical dials up to 11, a range of 'synthetic voice' effects can be produced, some of which are quite striking.

For the first time, Roland's highly regarded Variphrase vocal processing tools are available in software, in the shape of V-Vocal.

When V-Vocal is switched to 'time' mode, the pitch graph is replaced by a waveform display. Double-clicking creates a vertical green line across the display. Dragging this line left or right compresses or expands the audio on either side of it. Creating several of these markers within a clip allows you to precisely select and expand particular sections of sound, for instance to stretch a vowel sound and straighten out the timing of a phrase. As with pitch correction, this stretching must be done subtly in order to produce natural-sounding effects — although, as with pitch correction, some quite interesting unnatural-sounding effects are possible, if you care to experiment.

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In 'formant' mode a red line is imposed over the waveform, to which nodes can be added by double-clicking. This red line acts as a kind of envelope controlling formant shifts over the duration of the clip. A variety of effects can created by tweaking formant frequencies in this way, ranging from the subtle to the downright peculiar. In some cases it's possible to change (or at least to obscure) the gender of a voice. Highly confusing. Finally, 'dynamics' mode works in much the same way as formant mode, except that the envelope line is yellow rather than red, and it has the effect of scaling the amplitude of the clip up or down to create changes in the volume of a phrase. The waveform display redraws itself to reflect the changes imposed by the envelope, which helps to make editing quick and intuitive. Taken together, V-Vocal provides a set of tools that enable you not only to finetune a vocal performance, but to rapidly warp and distort it beyond all recognition! Of course V-Vocal isn't limited to processing vocals: any monophonic instrument or sound can be handled in the same way. Whether you approach is a tool for a clean, careful error correction, or outlandish sounddesign experiments, V-Vocal has a lot going for it.

The Fifth Element Sonar is certainly an impressive package, offering an intimidatingly comprehensive feature set, and then some. While its features may not be spectacularly original or innovative, they are well implemented and work reliably. The package as a whole offers a formidable array of tools for anyone working with audio and MIDI, all the more so with the new features added in version 5. Anybody in the market for a powerful and flexible DAW package for Windows should give serious consideration to Sonar — and those with a penchant for Acidesque looping ought to pay particular attention. Even so, I might stop short of calling version 5 an essential upgrade for existing Sonar users. If I were already a Sonar 4 user, I imagine I might be a bit hesitant about upgrading immediately. It's not that I think Sonar 5 offers too little; it's more that Sonar 4 already provides so much. The extra features in Sonar 5 seem more like pleasant luxuries than must-have necessities. While the new synths and other instruments are a nice addition, these in themselves arguably don't provide a compelling reason to upgrade — at least not for the likely majority of Sonar users who will have already invested in one or more third-party software synths. The two features most likely to clinch the deal will be the V-Vocal pitch processor and the Perfect Space convolution reverb. The former will be attractive to anyone with a hankering for a convenient tool for fixing wayward vocals; the latter is simply very nice, and sounds good. Even so, Sonar 4 users who already own a third-party pitch-correction plug-in or convolution reverb may think twice.

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Cakewalk Sonar 5

Support for 64-bit operating systems and CPUs will be appealing to anyone convinced that the advantages of 64-bit computing are significant enough to make upgrading a short-term priority. What percentage of the SOS readership fits that description is difficult for me to judge! Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Digitech Artist Series

In this article:

Digitech Artist Series

The Weapon & Eric Clapton Digital Guitar Effects Pedals Crossroads Published in SOS December 2005 Jimi Hendrix Experience Studio Auditioning Print article : Close window

Digitech Artist Series

Reviews : Effects

pros The first pedals to try to model the entire production chain, not just the amps and pedal effects. Straightforward user controls. Mainly good emulations of the original sounds.

cons

These new pedals model celebrated guitar tones, including those of Eric Clapton and Jimi Hendrix, in unprecedented detail. Paul White

Some players are bound to be disappointed when they discover that the pedals only model the sound, and don't necessarily make your playing sound the way you'd like it!

summary This is a potentially exciting evolution of amp modelling, and I expect we'll be seeing a lot more of this technology over the next year or two.

information Dan Donegan The Weapon, £149; Eric Clapton Crossroads, £149; Jimi Hendrix Experience, £199. Prices include VAT. Sound Technology +44 (0)1462 480000. +44 (0)1462 480800. Click here to email www.soundtech.co.uk

We're all pretty much used to the concept of physical modelling, where digital algorithms are used to replicate the behaviour of analogue synths, vintage effects, or specific guitar amplifiers. Digitech have taken the process one step further by attempting to model specific guitar sounds from classic tracks, including emulations of original effects, amp, speaker, and mix processing. The result of their work is packaged as three stomp box-style pedals. All can be used with a regular guitar amp or can be DI'd into a mixer, and each comes with a mains adaptor to save on batteries. Clearly you need to use the same guitar as the original artist (and the same pickup settings) for the best results, but the pedals have a number of controls which can help compensate for your own guitar or amplifier if necessary. With the controls in their central positions you get the most authentic recreation, provided that you're using the right guitar and pickup.

www.digitech.com

Rather than aiming to review these three pedals in great subjective depth, my aim is to explore the potential of this approach to modelling and to comment on what I feel to be its strengths and failings. Clearly the biggest limitation, though no fault of the designers, is that part of the authentic sound of any player is the way they play — and that goes some way beyond just picking the correct notes. This is particularly pertinent given that two of the pedals have been dedicated to

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Digitech Artist Series

legends Eric Clapton and Jimi Hendrix! The third artist is Dan Donegan, and while he may be less well known than Eric or Jimi, he was the first to be modelled — at his request, because he wanted a practical way to take his studio sound on the road. The strengths of this approach are that some of the modelled effects are not available elsewhere, and the development of the sounds was done in collaboration either with the artist or with an engineer who worked on their records. The biggest coup here was the collaboration of engineer Eddie Kramer in recreating some of the classic Hendrix sounds.

The Weapon & Eric Clapton Crossroads Dan Donegan is a member of the band Disturbed, a name which pretty well sums up some of the wonderfully aggressive tones his signature pedal The Weapon creates — basically death metal with an extra side-portion of death! Supplied with a soft zip gig bag and a heavy-gauge Dan Donegan pick, The Weapon also comes with a power adaptor, though battery operation is available. This dual powering arrangement is the same for the Eric Clapton pedal, while the Jimi Hendrix pedal is adaptor powered only, due to it's greater current consumption. The Weapon offers mono processing, with one output jack for DI'ing and another voiced to feed a guitar amplifier. Two recessed pins form the pedal-switch pivot, and these can be pressed in to release the pedal and expose the battery compartment, which takes a regular 9V PP3 cell. By way of controls, there are knobs for Level, Control 1, and Control 2, followed by a rotary selector switch that picks one of seven modelled signature tones. The songs used to model these sounds are 'Stupify', 'Mistress', 'Voices', 'Bound', 'Rise', and 'Intoxication'. Most of the presets feature distortion, but many are composite effects where we're also treated to reverb, phaser, autowah, and Whammy octave pitching. Control 1 and Control 2 perform different functions depending on the preset, generally by adjusting whatever the two most important parameters are for the preset in question. The Eric Clapton Crossroads pedal follows a similar physical format, but the pedal is painted a friendly shade of yellow in contrast to The Weapon's metallic finish. The modelling chain encompasses the speaker cabinets, microphones, effects, and the processing applied in the recording, mixing, and mastering paths. Again there are seven signature tones, including the rotary speaker effect used in the middle section of 'Badge' as well as an electric-to-acoustic simulation for those who care to have a crack at 'Layla' from the Unplugged album. Then there's that 'you'll get fined for playing that in this store!' classic 'Sunshine Of Your Love' with its silky 'woman tone', and of course the sprawling blues epic 'Crossroads'. We also get the original electric 'Layla' sound, the almost rockabilly 'Lay Down Sally', and 'Reptile', so there's a good selection of classic Clapton sounds here.

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Digitech Artist Series

Jimi Hendrix Experience The jewel of the collection for most players has to be the Jimi Hendrix Experience pedal, which again has four knobs, but comes built into a wah-wahstyle pedal. An additional switch pedal input allows you to change to the different sections of the song during performance without having to use the heel and toe switches beneath the pedal. The first three rotary controls are dual concentric in order to offer additional parameters. Thanks to the Hendrix family, the sounds in this pedal were referenced against the original master tapes, as recorded at Electric Lady Studios by Eddie Kramer. There's so much going on inside this pedal that it uses two of Digitech's Audio DNA chips rather than the single chip used in the other two pedals. In the process of voicing this pedal, the designers had to model the plate reverb originally used by Jimi Hendrix, as well as an array of vintage effects pedals such as the Fuzz Face and the Roger Mayer Octavia. They also modelled Jimi's 100W Marshall Super Lead amp, a vintage EMT plate reverb, and a home-built rotary speaker from Olympic studio. Again the pedal has amp and DI outputs, but there are additional switch modes detailed on the bottom of the unit that enable the pedal to be set up for stereo use, and that's how it sounds best. However, if you set it for stereo mode and then use just one output, the channels seem to get summed and the result is rather phasey and unpleasant, so you need to ensure the pedal is set up correctly for your intended use. The songs Digitech have chosen to model are all classics and serve to display the range of tones Jimi used across a wide range of material. These range from the gentle 'Little Wing' and 'Wind Cries Mary' to the aggressive 'Purple Haze', taking in 'Foxy Lady', 'Star Spangled Banner', 'Machine Gun', 'Voodoo Child (Slight Return)', and the seminal 'All Along The Watchtower', which is in my opinion one of the greatest singles of all time — despite Jimi cocking up the lyrics in the second verse and not bothering to fix them... A couple of the effects that have been modelled here initially sound rather rough and unfriendly, especially the Fuzz Face and Octavia, but that's pretty much how the originals sounded, and used in context they work very well for conjuring up those signature tones. Some things are more accurate than others though. As you might expect, the acoustic emulation for the rhythm part of 'All Along The Watchtower' doesn't get that close to the 12-string used on the original recording! On the other hand, the lead line for 'All Along The Watchtower' has the right overdrive and delay to get very close to the original, and it's no surprise that the wah-wah effects are also very authentic. Personally I found the rather stiff heel and toe switches a bit stubborn, so I'd probably cop out and buy the optional footswitch if I decided to use one of these.

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Digitech Artist Series

Studio Auditioning I'm not overly familiar with Dan Donegan's output, but I enjoyed the throaty, aggressive sound The Weapon produced, and it seems well suited to deathrelated music in general, not just covers of Disturbed songs. If death isn't your thing, you can always try the intros to 'Norwegian Wood' and 'Hole In My Shoe' using the sitar preset instead! I found the Hendrix pedal to be a lot of fun once I'd worked out the switching between the various rhythm and lead parts. However, as I said at the outset, if you can't play in Jimi's style, no amount of processing is going to be able to help you sound like him. I felt the 'Purple Haze' fuzz sound could have had a bit more angst, but then part of the problem is that these things sound different depending on who's playing. I loved the 'All Along The Watchtower' lead sound, and the tonality of the 'Little Wing' preset was also very reminiscent of the original. Certainly these pedals have won a lot of praise from guitar players who are far better able to emulate these classic players than I am, and I've always been impressed with how authentic these pedals sound when demonstrated by good players at the various trade shows where they've been shown. Most of my tests were conducted simply by DI'ing the pedals into a mixer, but they also sound good through a guitar combo set to clean if you take the time to set up the amp tone controls appropriately. In my mind, there's no doubt that Jimi Hendrix playing through a £50 practise amp would have sounded far more like Jimi Hendrix than anyone else on the planet playing through this pedal. On the other hand, if you do have a few Jimi licks up your sleeve, this pedal does give them the ring of authenticity — and wah-wah is a lot of fun whichever way you look at it! The only pedal I felt personally disappointed by was the Eric Clapton Crossroads model, but again that could be down to the sound in my head not being the sound on the record. I've always felt that Eric Clapton used fairly conventional

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Digitech Artist Series

sounds and that the magic came from what he played, and I feel much more comfortable playing his licks using my Line 6 PodXT. Come to think of it, comfort and feel might be part of it, because I thought the sounds created by this particular pedal felt a little stiff — guitar players will know what I mean by that. My impression was that I could hear a lot of aggressive filtering going on to turn my sow's-ear playing into a sonic silk purse — for magic to be effective, it shouldn't give away how it works. Then again, when I heard the pedal being demoed at the music shows, it sounded surprisingly authentic. Debating the authenticity of these pedals is only part of the story, as there's a lot more to this concept than being able to play seven clone songs per rock hero! The real point is that these tones also work for your own music, and, because everyone plays in a distinctive way, you can make the sounds your own by using them in your own songs. Different players are likely to feel very different about how effective these pedals are and the way they translate into playing feel is a very important part of that. It's probably fair to say that those who already like the better modelling guitar preamps are likely to be equally impressed by Digitech's new pedals, as they're really extensions of the same idea. Now that we've got this far, I'd like to see this technology in a more traditional multi-effects/preamp format able to recreate all the elements that a specific artist uses as the basis of his or her sound — the amp model would have all the usual controls, as would the vintage effects boxes. Just imagine a complete George Harrison, Jimi Hendrix, Pete Townshend, or Dave Gilmour stage setup in a fully programmable box, possibly with optional 'player personalities' that could be added later. You could still have song-based presets, but the user would also be able to set up custom sounds to recreate the guitar sound from virtually any song the artist had ever recorded, not to mention combining elements from different players. So it's hats off to Digitech for taking the first steps — now we have to see where they, and others, will follow. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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ESI Pro Maxio XD

In this article:

ESI Pro Maxio XD

The Perfect Host? 192kHz Audio & MIDI Interface For The Art Of Installation Published in SOS December 2005 Why PCI? EX8000 Control Panel Print article : Close window Maxio XD Brief Specifications Reviews : Computer Recording System To The Max Driver Performance Conclusions

ESI Pro Maxio XD £1299 pros Excellent system expandability up to 32-in/32out at 24-bit/192kHz. Eight low-noise mic preamps with comprehensive front-panel metering. Incredibly versatile digital options. EX8000 can also be used as a stand-alone eightchannel A-D and D-A converter or digital format converter. Inserts on all input channels.

PC

Firewire and USB 2 interfaces have their advantages, but if you need serious channel counts at high sample rates, the PCI card still rules, and ESI's heavyweight recording system caters for a huge range of input and output formats at up to 192kHz. Martin Walker

Korean company ESI have earned an enviable reputation for rock-solid audio and MIDI interfaces over the years, including their budget Waveterminal and more up-market Wami (Wave cons MIDI) ranges, and more recently, the Julia and ESP 1010 models. They Although the EX8000 box Photos: Mark Ewing provides eight separate have also clocked up some impressive analogue, ADAT, S/PDIF and firsts along the way: their EWDM AES-EBU inputs and outputs, (Enhanced Audio MIDI) drivers are particularly noteworthy for the ability to the PCI host card only supports a maximum of eight- support ASIO 2.0, GSIF, MME, WDM and Direct Sound formats with multi-client in/eight-out operation for each capability, so that multiple applications can access the interface simultaneously, EX8000. and you can even internally patch audio from one application to another using No high-impedance the Direct Wire digital patchbay. instrument option. No rear-panel analogue input sockets. Currently PC-only.

summary With excellent audio quality and a huge number of versatile features, ESI have managed to cram a huge amount into their Maxio XD system. This should make it attractive to lots of different types of musician, especially

Now they are poised to make an impact in more professional circles with the introduction of the Maxio series, which supports up to 32 input channels and 32 output channels at up to 24-bit resolution, at a 192kHz sample rate. Initially there are two models in this range, both based around a 32-bit buss-mastering PCI card to which you attach expansion boxes or breakout cables. The Maxio 032 expansion box provides 32 channels of ADAT-format digital I/O and two analogue mic/instrument/line input channels, plus further MIDI, S/PDIF and word clock I/O, but the Maxio XD (eXtended Definition) under review here is rather more ambitious. Its EX8000 2U rackmount interface offers only eight

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if ESI provide some different expansion options in the future, although some studio owners may find the lack of rear-panel inputs on the current EX8000 box a little frustrating.

information £1299; additional EX8000 units £1099. Prices include VAT. Electrovision Ltd +44 (0) 8700 053053. Click here to email Click here to email www.esi-pro.com www.maxioxd.com

Test Spec ESI Maxio XD Windows XP driver version 1.81. Intel Pentium 4C 2.8GHz processor with Hyperthreading, Asus P4P800 Deluxe motherboard with Intel 865PE chip set running 800MHz front side buss, 1GB DDR400 RAM, and Windows XP with Service Pack 2. Tested with Cakewalk Sonar 4.0, NI Pro 53, Rightmark Audio Analyser 5.5, Steinberg Cubase SX 3.1 and Wavelab 5.01, Tascam Gigastudio 160 v3.10.

simultaneous inputs and outputs, but each can be chosen from balanced analogue, ADAT, S/PDIF and AES-EBU digital formats, and the analogue inputs have phantom-powered mic preamps with metering and insert points. Up to four EX8000 interfaces are supported by a single Maxio PCI card, so the fully expanded 32-channel system would be impressive, especially as the converters offer up to 123dBA dynamic range and the mic preamps are claimed to be very special.

The Perfect Host? The Maxio PCI host card is actually quite diminutive considering its capabilities, and is compatible with both +3.3 and +5 Volt PCI slots. It has Sync In and Sync Out connectors on the card itself, suggesting that further expansion might be possible at a later date, and the backplate features four EDI ports, each supporting an eight-channel in/out audio stream at up to 24-bit/192kHz, plus one multiway MDI connector, which in conjunction with the supplied one-foot-long flying breakout cable provides a single MIDI In/Out, coaxial S/PDIF in/out, and BNC word clock in/out. The EDI ports on both the PCI host card and EX8000 expansion box are standard six-pin IEEE 1394 sockets, which means you can use ordinary Firewire cables — a high-quality five-metre one is supplied. However, you'll have to be careful not to plug any actual Firewire devices into these sockets to avoid possible damage. The EX8000 2U rackmount expansion box is an impressively rugged affair in black with silver legends, with a 3mm-thick front panel. The inside of the case is equally impressive, being almost entirely full of circuitry (in some areas three boards deep), with everything carefully bolted or tied down for reliability. I couldn't see the converter chips, but ESI told me that the ADC is AKM's AK5394A (the same as that used in Digidesign's HD192 interface, as well as Emu's 1212M, 1616M and 1820M) and the DAC is AKM's AK4395 (used in Emu's 0404 among various others). All of the mic/line input sockets and associated controls are on the front panel, which will certainly appeal to the live recordist who wants instant access, but perhaps not so much to The EX8000 rackmount unit offers an studio owners who normally prefer to impressive selection of analogue and digital I/ O, though only eight channels per unit can leave items of line-level gear be used simultaneously. permanently plugged in 'round the back'. For each of the eight inputs there's a Neutrik Combi socket, an associated mic/line button, rotary gain control and very welcome twin 10-segment input and output level meters with a 60dB range. With the button in the line position either the outer XLR or inner TRS-wired jack can be used in balanced or unbalanced modes, with a gain range from 0dB

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ESI Pro Maxio XD

(+4dBu nominal sensitivity) to +29dB. However, the TRS jack has an input impedance of 10k(omega), so it's not ideal for guitars and the like. In the mic position a preamp is switched in, offering overall gain from +25dB to +73dB, and global +48 Volt phantom power is available from a switch on the right-hand side of the front panel, where there's also a dedicated stereo headphone output with its own rotary level control hard-wired to the output 1/2 signal, and LEDs indicating the current sample rate and clock source. The 'Internal' clock option on the EX8000 comes into play when the expansion box is used in stand-alone mode as a preamp and A-D converter. The rear panel is crammed with socketry, with eight analogue XLR balanced/ unbalanced outputs at +4dBu level, eight duplicate analogue outputs using balanced/unbalanced TRS sockets at -10dBV level, plus eight inserts for the input channels to add hardware compressors, EQs, reverbs, and the like (you could at a push use these as unbalanced line-level inputs if you really wanted rear-panel inputs, but would have to pull out the cable if you ever wanted to use the associated front-panel preamp). On the digital side, there's another BNC word clock input, a pair of optical sockets for eight-channel ADAT in and out, four pairs of coaxial S/PDIF ins and outs on phono sockets, eight XLR sockets for the four stereo AES-EBU balanced digital ins and outs, two IEEE 1394 connectors for connecting the PCI host card and a Thru connection for 'future expansion', and a fused IEC socket for AC power. This is indeed comprehensive, but do beware when plugging in XLR cables to the analogue outputs, since the identical-looking AES-EBU outputs are immediately above — you won't do your amp or ears any good if you feed them full-strength digital signals by mistake!

The Art Of Installation The EWDM drivers support Windows 2000 and XP, and I downloaded the latest version 1.81 from the ESI web site. As with many modern interfaces, the drivers installed as four separate devices: the Maxio EWDM Controller, Wave-1, Wave-2 and MIDI. After a reboot the ESI icon appeared in my PC's System Tray and I could use it to launch the Maxio Control Panel utility. The Control Panel is quite complex, but you can save and recall up to five presets storing every setting. On the left-hand side of the Control Panel window is the Input section, with the Output section on the right, each divided into four banks of eight channels with level meters, along with various input and output options. By default, the banks are connected to the four EDI sockets on the PCI host card, but each bank can be switched to 'see' the MDI port instead. In an EX8000-based system, this port is occupied by a flying lead; this provides word clock and MIDI I/O regardless of the bank settings, and S/PDIF I/O that shows up if you switch one of the banks to MDI mode. However, the alternative 032 digital interface connects only to the MDI port, delivering its 32 channels of digital I/O this way.

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ESI Pro Maxio XD

Immediately beneath the meters are three rows of tiny icons. The top row displays the current MME Channel Mapping, and you can decide here which of the 16 stereo pairs of inputs and outputs are mapped to the Maxio Stereo Wave driver for basic stereo recording and playback. Using the middle and lower rows you can optionally assign the MDI analogue and S/PDIF ins and outs to specific EDI channels instead of the multichannel ones from the expansion boxes.

The main Maxio Control Panel has metering for a fully expanded 32-in/32-out system running four expansion boxes, plus zerolatency monitoring and a host of other system options.

Across the bottom of the Control Panel are the Input and Output Monitor buttons (32 in all), which work in conjunction with the right-hand Master Section controls. Here there are two channel faders with a 60dB range, along with routing that lets you allocate them to any one or none of the 16 stereo output pairs. Each of the 32 monitoring buttons has three states — disabled, enabled, and enabled but mixed to mono — and you can set up any combination for your monitor mix. The Channel Limit setting in the I/O pane can be fixed at 8, 16 or 32, and should be left at its default eight-channel setting if you only have a single EX8000 connected. The lower the setting, the lower will be the transfer rate of the Maxio host card across the PCI buss; a fully expanded system will require a fast PC with a good motherboard and associated chip set, as well as care when choosing other PCI devices for your system. I hope that detailed sample system specs will appear on the ESI web site before long — the only guidelines at present are 'Intel Pentium 4 or equivalent or compatible CPU, motherboard with chip set supporting the Intel Pentium 4, and at least 512MB of RAM'.

Why PCI? The Maxio system has been a long time in gestation (ESI first showed it at the Winter NAMM show in February 2004), and some musicians may now be wondering whether the PCI format is the best choice — after all, motherboard manufacturers are now telling us that PCI slots may disappear without trace within a couple of years. Well, as always, the truth is rather more complex than motherboard manufacturers want us to think. First of all, I think it highly unlikely that the PCI slot will completely disappear for at least four years — people worldwide simply aren't going to throw away all their existing PCI expansion cards unless they really can't use them any more, so many motherboards will continue to feature at least some PCI slots. There are some issues with PCI cards installed in motherboards with mixed PCI and PCI Express slots, particularly using the nForce 4 chip set, as I explained in PC Notes September 2005, but a resolution for these now looks more hopeful. Moreover, the Maxio XD can potentially support 32 input and 32 output channels

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ESI Pro Maxio XD

at 24-bit/192kHz with low CPU load and latency, a specification which simply can't be achieved by either Firewire 400 or USB 2.0 (PCI bandwidth is 133MB/second, compared with the 50MB/second of Firewire 400, 60MB/second of USB 2.0, and 100MB/second of Firewire 800). Some musicians are managing to run 80-in/ 80-out setups at 24-bit/44.1kHz with Firewire 400 audio interfaces, but at 96kHz this would reduce to about 40-in/40out, and 20-in/20-out is a more likely limit at 192kHz. Here, Firewire 800 and PCI are the only real options, with PCI still significantly ahead on bandwidth. Motu have apparently achieved 32-in/out operation with their 896HD system at 192kHz with a Mac system by connecting two units to the motherboard's FW800 ports, and a further two to FW800 ports on a PCI expansion card, but on the PC I suspect this to be rather unlikely. In a year's time, perhaps a PCI Express host card might be more sensible, but at the moment so few musicians have PCI Express slots available that it's not a commercial proposition, so PCI becomes the only real option for a PC-based system with this capability. ESI are confident that their Maxio XD system can realistically be used for perhaps four to five years if bought now.

EX8000 Control Panel If you leave a channel group as EDI and have an EX8000 connected you can launch a separate control panel that provides comprehensive choices over its clock source, and a routing table that lets you specify which signals are sent to each of the four stereo pairs. Possible input and output signals are EDI, Digital (AES-EBU and S/PDIF), Analogue and ADAT Optical, and here I found the various options tricky to get my head round at first. If you want to send an analogue input pair to your PC, you need to route them to EX8000's EDI Out, whereas if you want to send a stereo signal from your PC to (for instance) an S/ PDIF output, you route the appropriate Digital Out to the EX8000's EDI In. The Digital Audio input ports can also be either 'RCA (S/PDIF)' or 'XLR (AES/ EBU)'; their current PLL (Phase Locked Loop) status is displayed as Lock or Unlock, and you can individually choose from professional or consumer format. Further controls allow the ADAT I/O to provide eight channels at 44.1/48kHz (Normal), four at 96kHz (SMUX), or two at 192kHz (SMUX2) and the analogue outputs to have their levels individually set anywhere between 0dB and -60dB, while even the software meters offer four different modes, with various peak and display options.

Each EX8000 expansion box has its own Control Panel containing numerous options, although there are only eight channels of I/O between each EX8000 and the PCI host card.

Together, these two Control Panels provide incredibly versatile control, yet the

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ESI Pro Maxio XD

EX8000 has yet more versatility up its sleeve: it can also double as a stand-alone device with six selectable modes. It can be used as an eight-channel A-D and DA converter, either using the four AES-EBU ins and outs, the four coaxial S/PDIF ins and outs, or the ADAT optical I/O with or without S/MUX enabled, and can also convert ADAT to AES-EBU formats and back again, or convert analogue to ADAT and AES-EBU formats simultaneously. Overall, I was well impressed with the versatility of the Maxio XD, although I suspect that a few people will mistakenly think that they can use all the I/O simultaneously — after all, the system is capable of 32-in/out operation, while the EX8000 box has far more than eight physical inputs and outputs. For those wishing to expand beyond a single EX8000, I hope that ESI release a simpler and cheaper EX box that simply has the eight analogue in/outs, as I can see this being a popular expansion option.

Maxio XD Brief Specifications Sample rates: 44.1, 48, 88.2, 96, 176.4 and 192 kHz available on both PCI host card and EX8000 rack unit. Analogue inputs: eight Neutrik Combi sockets with mic/line switch, variable gain controls, channel inserts, plus low-noise mic preamps with up to +73dB gain and optional global +48V phantom power. Input impedance: 1.5k(omega) mic, 10k(omega) line. Analogue outputs: eight XLR balanced at +4dBu level, eight balanced/unbalanced TRS quarter-inch jacks at -10dBV level, stereo headphone jack with rotary level control. PCI host card digital I/O: word clock in and out, MIDI In and Out, coaxial S/PDIF in and out. EX8000 digital I/O: word clock In, ADAT in and out, four coaxial S/PDIF in and out, four AES-EBU in and out, EDI in and Thru. Dynamic range: input 116dBA, output 118dBA. Frequency response: 20Hz to 20kHz ±0.1dB. THD + noise: <0.0003%.

To The Max I experienced no problems while using the Maxio XD system, and my Rightmark Audio Analyser measurements supported the manufacturer's spec quite closely. The D-A converters have a dynamic range of 120dBA, and even once the output circuitry has been added this remains close at 118dBA. The A-D converter chips are even better at 123dBA, but the analogue input circuitry reduces this to 116dBA, and my loopback tests that measure both simultaneously were extremely close to this at 115.7dBA with both 96kHz and 192kHz sample rates. Frequency response was a good 0.3dB down at 9Hz and 21kHz at a 44.1kHz sample rate, extending to 43kHz at both 96 and 192 kHz sample rates, while THD was a very low 0.0008 percent, and stereo crosstalk an excellent -115dB. file:///F|/SoS/SoS%2012-2005/esipromaxioxd.htm (6 of 9)11/23/2005 3:01:54 PM

ESI Pro Maxio XD

My double-blind auditions against my own Emu 1820M and Echo Mia interfaces using a variety of music once again proved revealing, although as often happens, the differences between them and the review interface were subtle. The Mia revealed itself as having a slight harshness and less focused sound, while the 1820M and Maxio XD were both rather more revealing of low-level detail, and proved more difficult to tell apart. I re-shuffled the card outputs and repeated my listening tests at least half a dozen times, and then looked back at my written comments to find that I chose the Emu and ESI fairly equally as providing the best sound, so the fairest conclusion is to declare it a draw on audio quality. The 1820M is considerably cheaper, but the Maxio XD offers far more I/O and other options for its higher price. I also liked the sound of its mic preamps, which were certainly quiet, yet had an extended and natural high end.

Driver Performance

ESI's Direct Wire utility window is as impressive as ever with the Maxio XD — here I've patched the first two analogue inputs for MME recording, the same MME outputs to a WDM application, and the first two outputs of Gigastudio direct to ASIO inputs 3/4.

ESI provide multi-channel as well as basic stereo Wave drivers, and in Sonar's WDM/KS mode I managed the lowest 1.5ms setting at 44.1kHz with no glitching. In ASIO mode I managed the same setting in both Sonar and Cubase SX3. The Direct Sound drivers managed a slightly better than average 30ms Play Ahead setting with NI's Pro 53, with the MME drivers achieving 45ms, while I had no problems using the GSIF drivers with Gigastudio 3. They are only claimed to be GSIF1, but GS3 declared them to be GSIF 2.0, and I could use them to record audio, so GSIF2 they proved to be. Evert van der Poll's MIDI Test utility proved that like most PCI interfaces, the Maxio XD put in a good performance on the MIDI side, with an average MIDI latency of just 0.81ms (the lowest I've measured to date), average jitter of 0.22ms, and maximum jitter of just 0.48ms. These are all excellent results, two to eight times lower than any USB or Firewire MIDI interface I've measured to date, and even the most discerning drummer should have no issues with the Maxio MIDI timing.

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ESI Pro Maxio XD

need to interface with whatever other gear comes along, and of course an audio interface with eight mic preamps built in is the perfect spec for many musicians making live band or other ensemble recordings. Until the company's demise, Aardvark's Q10 was very popular for this reason, but there isn't now all that much direct competition for the Maxio XD. The Presonus Firepod retails at about £600, has eight mic preamps, eight-in/eight-out analogue plus S/PDIF digital and MIDI In/Out, and up to three units can be used simultaneously for 24-in/24-out operation, but it doesn't support 192kHz, or feature ADAT, AES-EBU and multiple S/PDIF I/O. A much closer competitor is MOTU's Firewire-based 896HD, which does support 192kHz, has eight mic/line inputs and 10 analogue outputs plus separate stereo headphone output, and features ADAT optical, stereo AES-EBU digital and word clock I/O, as well as the option of stand-alone use, providing up to 18 ins and 20 outs at 44.1 and 48 kHz. At £995 it is £300 less than the Maxio XD, but doesn't have its inserts, simultaneous +4/-10 analogue outs, multiple AES-EBU or S/PDIF I/O, and using more than two units may not be possible on PCs when running at 192kHz, although four units have been used at this sample rate with some Macs for 32-in/out operation (see 'Why PCI?' box)

The heart of the Maxio XD system is the host PCI card, which supports up to 32 audio input and output channels at 24-bit, 192kHz.

Overall, ESI's Maxio XD is certainly an ambitious and impressive system, and with its high-quality audio, well written drivers and incredibly versatile options it ought to find a home in many rigs, especially for those who want plenty of future expansion potential. It's slightly disappointing that with so much analogue and digital I/O on board the EX8000 you have to choose just eight ins and outs at a time to connect to your PC via the host card, but I feel this is largely redeemed by the versatility of the EX8000 and the ability to turn it into a stand-alone converter box. I'm still not personally convinced of the benefits of 192kHz recording, but it's there if you want it, with up to 32 inputs and outputs on a fully expanded system with four EX8000s. Penty of manufacturers hint that drivers supporting multiple interfaces may be available at some time the future, but ESI are to be commended for having them from day one. Published in SOS December 2005

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ESI Pro Maxio XD

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

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

file:///F|/SoS/SoS%2012-2005/esipromaxioxd.htm (9 of 9)11/23/2005 3:01:54 PM

Genelec 8020A

In this article:

Listening Tests

Genelec 8020A £470 pros

Genelec 8020A Active Monitors Published in SOS December 2005 Print article : Close window

Genelec family sound. Can be augmented by Genelec subwoofer. Surprisingly plausible bass end.

cons No jack input option.

summary Genelec fans will love the 8020As, as they deliver the same family sound as the rest of the 8000 series, with decent bass extension, but at a lower level.

information £470 per pair including VAT. SCV London +44 (0)20 8418 0778. +44 (0)20 8418 0624. Click here to email www.scvlondon.co.uk www.genelec.com

Reviews : Monitors

Despite their diminutive size, these new nearfield monitors still share the Genelec family sound. Paul White

In an ideal world, we'd all have pristine monitoring environments equipped with accurate monitors flat down to 40Hz and capable of 115dBSPL or more, but the reality is that some rooms are too small to accommodate such monitors. Indeed, some jobs simply don't need that kind of level or bass extension. For example, if you have a programming suite that you use for writing sequenced music, then you can probably make do with smaller speakers until you come to do the final Photos: Mark Ewing mix, whereupon you can move to the main studio. Similarly, in a small home studio, smaller monitors can be used both for writing and for producing reasonably accurate mixes, as working at high SPLs may simply not be practical. Genelec's new 8020A monitor is the smallest in the 8000 series, but still features an active two-way design. Controls on the rear panel optimise the audio performance to best suit the environment in which the speakers find themselves. With a decidedly compact 151 x 142mm footprint and a height of just 242mm, the rear ported 8020A features a die-cast aluminium enclosure with integral highfrequency waveguide and rounded corners to minimise diffraction. The shape also makes the case extremely rigid, which in turn cuts down on unwanted cabinet resonances. The drivers comprise a 105mm bass/mid-range driver augmented by a 19mm tweeter firing into the waveguide depression moulded into the front of the case. Both drivers are magnetically shielded. Each driver has its own 20W amplifier on board, fed from an active crossover operating at 3kHz,

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Genelec 8020A

and driver protection is built into the system. Despite being tiny, even compared to the earlier Genelec 1029A, the free-field frequency response is quoted as being 66Hz-20kHz ±2.5dB with a maximum SPL per pair at one metre of 105dB. In situations where the maximum SPL is adequate, but where the bass extension needs to be greater, the 8020As have been designed to work with Genelec's new LSE subwoofer, which provides bass management for surround or stereo and extends the low-end response down to 25Hz. Mechanically, the 8020As are nothing if not solid, weighing 3.7kg each. The drivers are both protected by acoustically transparent metal grilles, and there's a level control (with integral power switch) accessible from the front panel as well as a power LED. The integrated rubbery Iso Pad provides secure antiresonant mounting with a useful degree of 'tiltability', and the rear port is flared to reduce the turbulence Below the reflex port on that can cause noise and distortion. Power comes in the rear panel are via a downward-facing IEC socket, and the input threaded inserts which allow for mounting the female XLR is accompanied by an opposite-sex 8020As on wall brackets version allowing up to six 8020As to be daisy-chained — the screws at the from one line-level source. My own preference would bottom of the rear panel be for the input to be a combi jack/XLR for the sake of will release the Iso Pad flexibility, but a quick rummage in the cable box found desk stands in this case. me the necessary balanced jack to XLR leads I needed to hook up to my monitor controller. DIP switches are available for setting the EQ of the system, with a choice of flat or treble tilt at the high end (-2dB below 5kHz), bass tilt (choice of -2dB, -4dB or -6dB) for half-space mounting against a wall, and an 85Hz high-pass filter — this should always be active when using the speakers with the optional subwoofer. If the 8020As are on a console meterbridge, the recommended setting is to have everything flat except the bass tilt, which should be set at -4dB.

Listening Tests Despite their almost laughably small size, the 8020As maintain the Genelec family sound and also deliver a very credible impression of low end within their power range. When used in the nearfield position, there's adequate level for most types of work, other than perhaps for evaluating dance tracks at 'battle level', and unless you push the speakers too hard the sound retains its integrity and clarity surprisingly well. I can't think of any speaker this size that delivers such clean, wide-spectrum audio as does the 8020A, and although the Genelec sound is considered a little forward by some engineers, the degree doesn't seem excessive and there's a useful amount of adjustment anyway. The bottom line is that if you need small monitors that behave like big monitors, only not so loud,

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Genelec 8020A

then the 8020As come pretty close to being ideal. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Korg OASYS: Part 2

In this article:

AL1 CX3 Value For Money? Sampling Help! Combis & Effects Polyphony KARMA Ins & Outs Adding Fairy Dust Composing The EXs1 Piano Sound Other Matters Conclusions

Korg OASYS £5399/ £5149 pros It looks wonderful. It sounds wonderful. There's so much here, you'll be learning about it for years.

cons There's so much here, you'll be learning about it for years. It's a work in progress. It's not cheap (but see the 'Value For Money?' box overleaf).

summary The OASYS sets a new standard for keyboard workstations. If you're a keyboard player, you'll want an OASYS. If you're another keyboard, you'll aspire to be an OASYS. If you can afford one, I bet you buy one.

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

Korg OASYS: Part 2 Workstation Synth Published in SOS December 2005 Print article : Close window

Reviews : Keyboard workstation

We finish our examination of Korg's new megaworkstation, taking in the remaining synth engines, the sampler, the KARMA algorithms and the onboard sequencer, and draw some conclusions about it... Gordon Reid

Last month, we charted the evolution of the OASYS, and looked at its sample-based HD1 'High Definition' synth engine. But, impressive though this is, it's far from the be-all and endall of the instrument. The raison d'être of the OASYS is its ability to host additional synth engines, known collectively as EXpansion instruments, or 'EXis'. So we'll start this month by looking at the first two of these, the AL1 virtual analogue synth and the CX3 organ. Photos: Richard Ecclestone

Before considering the synth engines themselves, I should explain that an EXi Program is not either a virtual analogue synth or an organ patch. Each Program contains two slots that can host either an AL1 patch or a CX3 patch, so the Program itself can contain two AL1s, two CX3s, one of each, or a single patch of either. This makes an EXi Program incredibly flexible, especially when you use two AL1s in tandem. Hopefully, the same will be true when Korg releases additional EXis. But for now, let's stick to what exists today...

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AL1

Korg OASYS: Part 2

www.korg.co.uk

Test Spec OASYS88 (serial number 000001). OS versions reviewed: v1.0.0 (first part of this review) and v1.0.2 (second part of this review).

I've read much chat-room banter which suggests that the AL1 is no more than a repackaged version of the analogue models in Korg's Z1 or MS2000. This is misinformed. It's a completely new synthesizer with a huge voice structure, and contains so many good features that — even if this whole review concentrated on nothing else — there would be insufficient space to discuss it fully. Let me offer you some examples... The AL1 allows you to 'morph' oscillator waveforms to produce harmonically rich timbres for solos, polyphonic pads and effects. It also offers detuned waves for added depth, and a four-mode ring modulator that allows you to modify the modulator before applying it to the carrier. Moving along the signal path, the filter section offers a cornucopia of goodies, with two resonant, multi-mode filters per voice. The special touch here is an option called 'Multi Filter' that allows you to morph and fade between 22 types of filter profile. It will take me years to plumb all the possibilities here. Not to be out-shone, the amplifier offers Driver and Low Boost controls that make even the simplest patches jump into life. Note, however, that the Driver is very different from placing an overdrive or distortion effect across the output of the AL1, because it affects each voice individually, much like the Hexa-fuzz on Roland's early GR-series guitar synths. In 2001, I was much impressed by the three seven-stage envelopes on the Alesis's A6 Andromeda, because you could see them displayed graphically on the synth's screen, assign them almost anywhere in the synth, and because they offered nine choices of slope for each stage. The OASYS offers five envelopes to the A6's three, five stages to the A6's seven, and 10 selectable slopes instead of nine. The OASYS can thus impersonate the Editing a Program containing two AL1 linear and exponential contour patches. characteristics of different vintage synths, and much else besides. Finally, the AL1 is peerless if you want modulation flexibility. Not only does it incorporate four LFOs offering 18 distinct wave shapes, but you can further shape these using nearly 200 variations, and then direct the results to innumerable destinations. Given that every modulator is an AMS Source and an AMS Destination, and that you can apply further shaping and control in the AMS Mixers, the possibilities are almost boundless. Given this praise, it may surprise you to discover that I was not impressed with the AL1's factory patches when I first heard them. With the exception of some superb string ensembles, everything seemed too smooth, too 'programmed', and too drenched in effects. It seemed to me that the AL1 factory set exemplified much that die-hard fans of analogue synthesis dislike about virtual-analogue instruments. But it would seem that there's a reason for this. Apparently, when designing the sounds for the HD1 engine, Korg's programmers had created enough sounds to occupy six banks of Programs — but the design team elected

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to use only five. So the programmers tried to recreate some of their favourite HD1 Programs using the AL1 instead. While this may have been a triumph for them, I don't regard it as a sound commercial decision. As a result of this, it wasn't until I started to program AL1 patches for myself that its true character became apparent. I removed the step sequencing and other tricks, made less use of the 'supersaw'-type waveforms, removed the effects, and went back to square one with the oscillators, filters and amplifiers. The AL1 was transformed. Polyphonic, without drive, it has the same smooth character as some of Roland's vintage polysynths. Wind up the drive, however, and it's like morphing a Jupiter 8 into an Oberheim OBX. Alternatively, punch the Mono, Unison and Detune buttons, and the AL1 becomes a hugely flexible monosynth capable of everything from unassuming single-oscillator sounds to massive concoctions with the depth usually reserved for large modular beasties. Unfortunately, the AL1 won't appeal to knob-twiddlers. Although you can assign all manner of functions to the OASYS's knobs and sliders, the quantisation is too coarse for manual sweeps and bleeps. This implies that, despite the power available, Korg's engineers have neglected to interpolate between controller values. I hope that they rectify this.

CX3 The second EXi is an enhanced version of the generator used in Korg's CX3 and BX3 Hammond impersonators, controlled using the nine top-panel faders as fake drawbars. The enhancements take many forms. For example, the OASYS CX3 overcomes my only significant complaint when I reviewed the 'physical' CX3: that the keyboard gates the leakage noise, leaving an unHammond-like silence between notes. On the OASYS, there's a Noise Level parameter that creates a welcome background swirl. I also like the fact that all the Hammond-esque effects (amp modelling, overdrive, chorus/ The OASYS's CX3 sound engine. vibrato, and rotary speaker) are contained in the CX3 itself, so you don't need to use up precious Insert Effects to generate them. Furthermore, the OASYS even allows you to create custom chorus/vibrato settings! Mind you, given that the CX3 is designed to emulate the scanner vibrato of the Hammond A100, B3, and C3, this is a little strange, and after a few hours' experimentation, I decided that Hammond had it right in the first place. Nonetheless, it's a boon if file:///F|/SoS/SoS%2012-2005/korgoasys.htm (3 of 14)11/23/2005 3:02:01 PM

Korg OASYS: Part 2

you want to fine-tune the effects. In addition to all of this, Korg have extended their unique EX mode, which now allows you to choose the pitches of the four additional drawbars from a list of 73 footages. The range of new sounds obtainable is immense, if not always musical. Oh yes... and you can now use EX mode when playing two 'manuals' either side of a split point. This may sound obvious, but when you invoked EX mode on the CX3, you could play only on a single manual. As you would expect, the sound of the OASYS CX3 is extremely good but, as I write this, I wish that I had a CX3 here to perform a direct comparison. If my memory is not playing tricks, something appears to have changed at the top end... the characteristic 'scream' seems more muted than I remember. Of course, the difference could be in the OASYS's output stages rather than the CX3 algorithm, but either way, I often found myself using EQ in an attempt to add back a certain quality that seemed to be missing.

Value For Money? In 1995, the retail price of my Trinity Pro was £2795, and, once upgraded with the later MOSS, HD, and ADAT expansion boards, the price headed scarily in the direction of £4000. The value of the pound has decreased by approximately 30 percent since then, so £4000 in 1995 is equivalent to about £5700 today. In this light, the price of an OASYS doesn't look outrageous, despite what some people have been saying on web forums since the cost was announced. Of course, the public perception of what a pound should buy has changed considerably over the past decade, and many people will view £5000 for a keyboard workstation as unpalatable. If you are one of these, don't worry about it. Korg didn't build the OASYS for you.

Sampling In principle, the sampler — which is derived from that in the Triton Studio — occupies the free RAM in the OASYS. In other words, if you load the HD1's EXs2 (grand piano) sample set, you should have approximately 200MB for your own samples. If you load EXs1, you should have approximately 400MB, and if you load neither you should have approximately 700MB. But this is not the case; the memory available is a paltry 3MB when EXs2 is loaded. This holds just 34 seconds of samples, which is almost pointless. With EXs1 loaded, the available memory increases to a little under 195MB, and with neither EXs loaded it reaches 500MB. In every case, this is 200MB short of what it should be, which is strange. I have a standard test for reviewing samplers. It's an audio CD with multisamples of various Mellotron voices, with different sounds on each channel, all badly recorded and without topping or tailing. Turning this into a usable Program

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requires loading, trimming, and keymapping... all of which can take well over an hour on a clunky system. The quickest way to turn these samples into a usable patch is with a software sampler such as NI's Kontakt. Ripping the CD, separating the samples and allocating them to notes is then a doddle, as is the editing. Without a mouse/pointer GUI, operations on the OASYS are inevitably slower than they would be on a PC, although somewhat faster, perhaps, than on many traditional hardware samplers. In particular, once Looping a sample in one of the OASYS's you have created a multisample on the sample-editing screens. A full range of audioediting options is provided, and the large OASYS, you need only click on display makes the process easy. 'Convert MS to Program' for the whole thing to appear as a conventional Program no different from a factory HD1 Program. Well... no different except for one thing. You may think you have saved everything associated with the Program once you have written it to memory, but you haven't. If you then switch off the OASYS, the samples — if loaded to RAM rather than to disk — disappear in a puff of dispersing electrons. If you don't perform the appropriate Save in the Disk menus, you'll have to start all over again. While editing the samples, I discovered a bug in the system. Select Reverse in the Loop Edit page, and the selected sample does not play backwards; instead, you get a continuous burst of loud, digital hash. Returning to good things, the OASYS answers one of my most fervent desires in any sampler; it loads other manufacturers' multisamples correctly. This may sound obvious, but a quick whizz through back issues of SOS will demonstrate that many samplers are unable to do this. Sticking with the Mellotron theme, I loaded the Akai S1000-format Mellotron Archives CD-ROM and... blimey! The OASYS loaded multiple Programs of multisamples, stuck them into separate Program locations, and everything worked. Of course, this CD provides an easy test because it has no loops, and minimal filtering and enveloping, but the mere fact that the OASYS handled it so easily is encouraging. In addition to Akai S1000 and S3000 data, the OASYS will load AIFF and WAV files, plus, of course, Korg's own libraries, either from CD or via USB. To those of us who have piles of Zip drives and clunky old 500MB HDDs lying around, the OASYS is less kind; it does not have a SCSI interface. It also won't load Roland sample libraries, although I hear that Korg have it 'on the list' for inclusion later. In addition to sampling, the OASYS permits resampling of its output. This is a boon when you want to lay stems back as a single audio file, and also allows you to resample complex audio, chop it up, create loops, apply wave sequencing and file:///F|/SoS/SoS%2012-2005/korgoasys.htm (5 of 14)11/23/2005 3:02:01 PM

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KARMA (both of which I'll discuss later) and generally mould sounds into new shapes. If you're into experimental audio techniques or the modern equivalent of musique concrète, this is just the job.

Help! The Help button takes you into a context-sensitive Help system. This is hugely cut down from the paper manual and parameter guide, but is comprehensive enough to guide you through much of the OASYS, and is invaluable when you don't have the manuals to hand. I hope that Korg continue to work on this, improving the indexing and expanding the content, because it's an excellent facility that deserves further development.

Combis & Effects I always found it strange that Korg workstations offered Combis that incorporated up to eight Programs, but also provided a single Sequencer setup that included up to 16. Happily, the OASYS dispenses with the eight-part limitation, and its Combis now boast the full complement of 16 Programs. They also offer a great deal of real-time control over the sounds generated by the Programs within them, without affecting the Programs themselves. And, unlike the Trinity and Triton, which could only have a single MOSS Program in any given Combi, the OASYS lets you draw upon its HD1, AL1 and CX3 engines as you choose. This is all excellent stuff, but I'm going to have a moan, nonetheless. In 1998, Novation launched the Supernova, followed in 2000 by the Supernova II, both of which offered up to seven effects per Part. When you inserted a Part into a Performance, its effects were unaffected, so a sound was identical whether used in isolation or as part of a Combi. In contrast, the OASYS sticks to the model that Korg introduced on the M1. This allows you Editing a KARMA GE assigned to an OASYS to define a new effects structure within Program. a Combi, or to import effects from a Program into a Combi, but only up to the Combi's limit. An OASYS Combi can support many more simultaneous effects than an M1 Combi (and with genuine multitimbrality) but it's still not always possible to preserve the sounds of the Programs within a complex Combi. What's more, you can't import Programs with their effects attached. This methodology is clunky, and it's time that companies stopped using it. As for the effects themselves, the algorithms in modern workstations are so

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extensive as to be unreviewable. On the OASYS, the numbers are impressive. In addition to the effects provided within the engines themselves, and in addition to 32 dedicated EQs (16 each for the synth parts and audio tracks that we'll discuss later) its dedicated effects sections offer 185 algorithms that you can insert into 12 insert effects, two master effects, and two so-called 'Total' effects, per Program/Combi. If I had to pick a handful of highlights (and I do), my list would include the Polysix Ensemble and the 'REMS' modelled microphone, amp and speaker effects. For many years, I toyed with the idea of ripping the effects board from a dead Polysix to build a dedicated ensemble unit, but the OASYS makes this unnecessary because — with four analogue inputs plus S/PDIF, all of which can be routed through the effects structure — you can use it as a sophisticated effects unit. Likewise, you can use the OASYS as a REMS processor that — channels permitting — could replace many of your stomp boxes and rackmount effects. You might think this daft, and in a studio you would probably be right, but in a live context this would reduce the number of bits and bobs on stage, and the number of cables needed to set up your rig. There are only a couple of minor niggles. The vocoder seems a bit of an afterthought. Before laying hands on the OASYS, I wondered whether it would offer something like the voice processing options on the Roland VC2 board for the VariOS and V-Synth, but it doesn't. There's also a tiny omission from the otherwise all-encompassing effects. When experimenting with the Mellotron sounds mentioned above, I tried to insert a spring reverb effect for an authentic 'Mk2' flavour, but I couldn't, because there isn't one.

Polyphony What is the OASYS's polyphony? It's impossible to give a one-word answer. The specification quotes maximum numbers, but the polyphony is affected by the complexity of patches, the effects used, layering, and other factors, and I doubt that you'll attain the maximum values in practice. A meter gives you an indication of how many voices are available, but my experience suggests that you shouldn't rely too heavily on this.

KARMA If I cornered you in a dark alleyway and demanded that you describe Stephen Kay's Algorithmic Real-time Music Architecture, I bet that you couldn't. I don't blame you. Korg describe KARMA as a 'revolutionary performance technology that generates amazing phrases, grooves, and other musical effects that can be altered and randomised in real time', but the same description was true of the analogue sequencers of more than 30 years ago, and tells us little about why Korg make such a big deal of it.

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Korg OASYS: Part 2

The key to understanding KARMA is this. Even if you have (for example) an amazing acoustic guitar patch or a superb harp patch, your compositions will not sound realistic if you do not play those sounds as you would the 'real' instruments. And, as we all know, it's difficult to play keyboards in the same way as guitarists play guitars, or harpists play harps. Confronted with this, Kay tried to quantify how players of non-keyboard instruments do what they do, and then describe this as a set of algorithms ('Generated Effects', or GEs) that could mimic them. I first judged his success when I tested Korg's KARMA workstation in 2001. At first, few of the KARMA algorithms excited me, but then I stumbled upon a Combi called 'Magic Flute'. If I played The OASYS rear panel certainly doesn't an interesting chord in the bottom two skimp on features, or outputs. octaves of the keyboard, this produced a beautiful, picked acoustic guitar part, with KARMA choosing notes and inversions that sounded both believable and musically interesting. The results were not arpeggios; they were performances based upon the chords I was playing. Delving into the menus, I learned how to extend the guitars over the 61-note range of the keyboard, and to use the two 'scenes' in a GE to create a patch that picked on Scene 1 and strummed realistically on Scene 2. The Real Time Controller knobs then allowed me to manipulate parameters including the Rhythm Complexity, Strum/Pick ratio, Note Voicing, and Velocity Accents — things that human guitarists do unconsciously to suit the needs of the music. I was impressed; so much so that I bought a KARMA. But after a while I began to realise that I was always using it for the same purposes. Given that Kay had by this time extended KARMA way beyond the boundaries of imitations of acoustic instruments, creating numerous GEs for dance grooves and other musical effects, this worried me, so I tried to analyse the problem. I came up with three reasons that, in retrospect, I also believe to be the reasons that the KARMA was a less successful product than Korg might have expected. Firstly, the two-line LCD on the KARMA workstation does not lend itself to programming the KARMA GEs. Secondly, because Korg's programmers had used KARMA to create miniature backing tracks, it was hard to view it as anything other than a sophisticated arpeggiator. This, of course, is exactly what it wasn't intended to be, because, as when playing with a human guitarist, a drummer, or whatever, you don't have precise control over the output, merely over the nature of the output. But perhaps the biggest obstacle to using KARMA was the complexity of the system. The manuals were encyclopaediaic in both length and depth, and I wonder whether anybody fully plumbed them. Now, fast-forward half a decade. KARMA in the OASYS is hugely enhanced over previous incarnations, with up to four simultaneous GEs in Combi mode. Additionally, instead of offering two scenes per Program or Combi, there are now file:///F|/SoS/SoS%2012-2005/korgoasys.htm (8 of 14)11/23/2005 3:02:01 PM

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eight per Program and 40 (!) per Combi. But despite the increase in complexity, the KARMA features on the OASYS are no harder to use than they were on the KARMA workstation. Much of this is a consequence of the OASYS's better user interface, although some credit lies in the introduction of a certain amount of standardisation that makes the GEs easier to use, in particular the separation of GEs into categories such as strumming, picking, bass patterns, and so on.

The built-in sequencer's Mixer page.

To become acquainted with the new version, I took Combi A000 (which is the sound that appears when you switch on the OASYS) and experimented with the effects generated by tweaking the knobs and sliders on the control panel. I then selected the KARMA tab to reveal nine further pages of controls. Changing the GEs attached to sounds and changing the Programs in the Combi soon created something completely new, transforming the Combi from a soundtrack-y thing into something closer to 'Fanfare For The Common Man'. Further tweaking generated numerous new Combis, quite different from one another, but all sharing the same fundamental structure. Given that I was relying upon serendipity to create something interesting, the results were more than acceptable. If you're a composer working in strict tempo, KARMA can be a fascinating tool for obtaining inspiration and for capturing ideas quickly and simply. But even if you don't want to delve into the programming system, you might find inspiration from the existing GEs and KARMA-processed Combis, using OASYS as a toolkit of musical expressions, or even as a big auto-accompaniment keyboard.

Ins & Outs The OASYS has a wealth of I/O, including no fewer than eight individual analogue audio outs in addition to a L/R stereo pair. Alongside these, you'll find a headphone socket and four analogue inputs, two of which offer balanced and unbalanced options with mic/line options, level controls, and 48V phantom powering. Digital I/O is also provided, with 24-bit optical S/PDIF input and output. Given the flexibility of audio assignment within the OASYS, it's fair to say that it's well provided with ins and outs. Control options are par for the course, with MIDI In, Out, and Thru sockets, and three control pedal inputs (for a damper pedal, a switch, and a single continuous pedal). Finally, there are four USB2 sockets that you should be able to use for external storage. I was unsuccessful using a 128MB memory stick, but I'm not going to hold that against OASYS version 1. There is a single I/O expansion option. The EXB-DI board will provide an eightfile:///F|/SoS/SoS%2012-2005/korgoasys.htm (9 of 14)11/23/2005 3:02:01 PM

Korg OASYS: Part 2

channel ADAT output that mirrors the individual analogue outputs or the stereo pair (as you choose), and also offers a word clock input. This will allow you to synchronise the OASYS's sample rate to external devices, but is not a substitute for a timecode input.

Adding Fairy Dust I've left two aspects of voice creation to last: wave sequencing and vector synthesis. Like a drum setup, you create wave sequences in the Global area of the OASYS, and apply them by selecting 'Wave Sequence' rather than 'Multi Sample' in the HD1 oscillators. Given that you can velocity-switch up to four such sequences and layer any two when building HD1 sounds, this provides great scope for creating evolving sounds or injecting rhythmic interest. The wavesequencing specification is more advanced than that of the Wavestation family, but anybody familiar with these synths will be comfortable with the OASYS's implementation. Hmm... I should have said that they'll be comfortable with a subset of the OASYS's implementation, because KARMA now includes wave sequencing, too. Just as you can ask GEs to play notes, affect velocities, and so on, you can now attach waves to KARMA steps, thus making it an amorphous wave sequencer as well as an amorphous pattern sequencer. This is appealing because, unlike a pre-defined sequence, there is a degree of uncertainty about the result, which adds further interest to the sound. You shouldn't confuse this with conventional wave sequencing, because you can't hold a note and create evolving sounds that crossfade between waves. However, there are many other possibilities, so the creative potential is immense, particularly when creating rhythms. And you can access and program all of this far more simply than on the original Wavestations. Similarly, anyone who programmed vector synthesis patches on a Prophet VS, Yamaha SY35 or Korg Wavestation will grasp the Advanced Vector Synthesis on the OASYS. The difference here is that AVS is not limited to the PCM-synthesis engine; you can use it to control the balance between EXi1 and EXi2 in a single Program, as well as to control the balance of the oscillators in an AL1 patch. This opens up no end of morphing possibilities. Likewise, you can morph the contributions of Programs within Combis, and modulate effects settings, both of which offer much creative potential if you're working in a surround environment. As always, you can apply vector synthesis using the joystick, or in a repeatable fashion using the five-point vector envelope. The big improvement here is that you can now modify the envelope using modulation sources, and synchronise it to MIDI tempo. And the vector envelope can itself be used as a MIDI CC. It's all good stuff.

Composing I have never much liked the sequencers built into keyboard workstations. There's only so much one can do with limited processing power, limited RAM, and an

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Korg OASYS: Part 2

operating environment reminiscent of attempting to paint the roof of the Sistine Chapel through a letterbox. Consequently, I had high hopes for the OASYS's sequencer. With a PC under the bonnet and a 640x480 pixel screen, it should have been possible for Korg to endow the OASYS with the most remarkable sequencer yet seen on a keyboard. Surprisingly, it's little different from the sequencer in the Triton Studio (see above). It's solid and reliable, and everything falls to hand more easily than before, so you could view this as a case of not fixing what isn't, well, broke. There has been progress, with a handful of extras such as in-track sampling and Song Templates but, ultimately, I feel that this is a missed opportunity to lift the workstation aspect of the OASYS to a completely new level. It's important that I put this concern in context. Some people have suggested that the OASYS should have a sequencer with the power of Digital Performer, Cubase or Logic, but this is irrational. Companies dedicated to these packages have refined them over two decades, and it's unreasonable to expect Korg to have emulated them in the first version of the OASYS. In addition, the user interfaces of Mac- and PC-based packages offer drag and drop functionality and the ability to draw things on screen, two operations that you can't perform on the existing OASYS. So... it's currently impossible for Korg to have stepped far beyond the style of sequencer that they've been offering since 1988. But understanding this and feeling comfortable with the limitations are different things. Text-based event editing is archaic, and the OASYS begs for a better sequencer that offers pianoroll representations and graphical editing of notes and CCs. Happily, I understand that these are also on one of Korg's upgrade lists. If the OASYS's sequencer is currently a slight disappointment, the same cannot immediately be said of this workstation's ability to handle audio; its ability to record up to four tracks simultaneously and play back up to 16 makes it stand apart from the crowd. The OASYS is not Pro Tools HD, but, with its ability to tap its effects, its (albeit limited amount of) automation and audio-event editing, and its use of the top panel as a tactile mixer, it is perhaps the first keyboard to embrace fully the concepts of a digital audio workstation, or DAW. You might ask why manufacturers bother integrating recorders into their workstations. After all, anyone able to afford an OASYS will almost certainly own a computer running a MIDI + Audio package of one flavour or another. I suppose you could argue that, if you're going to stick a 40GB hard drive inside a keyboard, you may as well use it for recording, especially if the instrument already has audio inputs. But to be so dismissive is to miss the point. When a file:///F|/SoS/SoS%2012-2005/korgoasys.htm (11 of 14)11/23/2005 3:02:01 PM

Korg OASYS: Part 2

recorder is incorporated into a workstation, the boundaries between sampling and recording can become blurred in a way that is not necessarily true of dedicated DAWs. The OASYS allows you to manipulate audio fluidly, cutting and pasting it freely in audio or MIDI-based sequences, using it as the basis of conventional sampling, or even resampling it to free up audio tracks, polyphony and effects for further sounds and sequencing. This means that if Korg's developers are indeed working on updates and revisions in this area, as I have been told they are, the OASYS could have had the potential to approach the ideal of an all-in-one production tool for video post, foley and TV. Unfortunately, it has a single deficiency that makes this ideal unobtainable. It has no timecode capabilities. Damn! Mind you, other limitations stop the OASYS performing exactly like a computerbased DAW. For example, you can only record to the internal drive. Furthermore, with the OASYS, Korg have built another workstation with a 48kHz internal synth engine, which incorporates S/PDIF inputs and outputs that support only 48kHz and 96kHz sample rates, and they have, naturally, provided it with a CD burner that writes audio CDs at 44.1kHz. This means that everything you play, record or sample has to pass through an internal sample-rate converter as it's burned to disk. Thus any tiny improvements in audio quality gained by using 48kHz instead of 44.1kHz are lost when converting downwards. So maybe the greatest beneficiaries of the OASYS's recorder will be 'live' performers. The 16 tracks and 80 minutes of track recording time will allow many artists to leave their laptops at home, which has got to be a good thing for simplicity and reliability on stage.

The EXs1 Piano Sound Last month, I suggested that it was possible to take the single-oscillator threelayer EXs1 piano, and add a fourth layer and the piano damper sound to create a dual-oscillator Program similar to the far greedier EXs2 piano. It seems that Korg's programmers had the same thought, because when you upgrade to v1.0.2, this is exactly what you get!

Other Matters For reasons of space, I've had to omit much about the OASYS that I would have liked to have discussed: things such as the control assignments, the use of the touch-pads, the drum kits, and some of the factory sounds. Nevertheless, I hope that I've been able to convey that — while it will benefit from further development in some areas — the OASYS is already a superb synthesizer that sounds fantastic. In many ways, I applaud Korg's decision to make the OASYS sound good first. However, it's important that the developers don't design new synthesis engines file:///F|/SoS/SoS%2012-2005/korgoasys.htm (12 of 14)11/23/2005 3:02:01 PM

Korg OASYS: Part 2

to the exclusion of bringing the workstation aspects up to speed. While they're usable as they are, the sequencer and the hard disk recorder require further development if the OASYS is to fulfil its potential. It's also important for Korg to start bringing the first upgrades to the market quickly, although I understand that the developers are working flat-out in all areas of the OASYS's operating system, so hopefully the first large upgrades won't be long now. If I were asked what synth engines Korg should add first, I would offer two suggestions. Given that the OASYS already offers two modelled instruments, it would seem reasonable to suppose that Korg could develop SAS-style pianos and electric pianos for it. Certainly, the motherboard could handle it; Roland's RD1000 had the CPU power of a dishwasher, yet is still used by aficionados for whom nothing has ever superseded it. Likewise, there should be nothing stopping Korg from rewriting the physical models from the Z1 and incorporating them in the OASYS. The Z1 was a fantastic synth, misunderstood and undervalued by keyboard players the world over, and it deserves another chance, particularly on an instrument with a user interface like that of the OASYS.

Finished mixes can be burnt to CD using the built-in CD writer.

In addition to these, I would like it to be possible to play a CD and the keyboard simultaneously. I spend a fair amount of time learning other people's material from CD, and it would be helpful to be able to do so without requiring an external CD player or having to rip the CD into the OASYS first. Moving on to the physical lump that is the OASYS hardware, you should consider which size of OASYS is most suitable for the type of music you play. Fortunately, both keyboards are pleasant and responsive, so the compromises entailed in choosing one over the other are not as great as they could be. In fact, apart from the hinge on the display, which I mentioned last month, there's only one other part of the OASYS's physical being that concerns me, and that's the CD drive (shown left); the tray feels flimsy, and I would have preferred something more robust. Better still, why not adopt the tray-less design used in my G4 Powerbook? This would seem much more suitable for a flagship instrument, both on the road and in the studio.

Conclusions Despite the technological progress of the past 10 years, the OASYS (Synthesis Studio) is very similar in concept to the original OASYS (Synthesis System). If this sounds like a criticism, it's not. In fact, it demonstrates that Korg were way ahead of the game in 1995. Sure, the hardware of the mid-'90s was unable to file:///F|/SoS/SoS%2012-2005/korgoasys.htm (13 of 14)11/23/2005 3:02:01 PM

Korg OASYS: Part 2

support the company's vision, but the technology of 2005 demonstrates clearly that the vision was a good one. So, who's going to buy an OASYS? Given the substantial price tag, it's clear that the OASYS was not aimed at the hobbyist or the local band. This is a flagship instrument aimed at composers, serious players, and major studios. Nevertheless, it's already finding favour with keyboard enthusiasts. Like a 50inch plasma television, it's something that you can do without, but which — if you have the cash — will do what it does better than anything else. The OASYS's existing voicing is superb (with more to come) and, while the sequencing and hard disk recording facilities are not yet in the same class, they are due to improve considerably. More importantly, the integration works faultlessly, which means that there's no way you should compare the OASYS to a PC plus a controller keyboard and a bunch of software packages. Most important of all, I can overlook my criticisms of the OASYS because I love working with it. Everything falls to hand effortlessly, and if you view a synthesizer as a musical instrument rather than a mere bundle of technology, the OASYS is currently in a class of its own. Other manufacturers are going to have to respond to compete, and that — whether you buy an OASYS or not — is going to benefit all of us. Given that I also expect to see spin-offs over the next few years, I can only conclude with one sentiment: Bravo Korg! Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Little Labs Multi Z PIP

In this article:

Verdict

Little Labs Multi Z PIP £458 pros Exemplary audio quality. Flexible connection options. Effective re-amping option.

cons No speaker thru connection.

summary Just as good mic preamps cost a lot of money, so do good instrument preamps, and there is no denying that this is one of the best. Whether its high cost can be justified in the smaller studio is another question, however.

information £458.25 including VAT. Unity Audio +44 (0)1440 785843. +44 (0)1440 785845. Click here to email www.unityaudio.co.uk www.littlelabs.com

Little Labs Multi Z PIP Instrument Preamp Published in SOS December 2005 Print article : Close window

Reviews : Preamp

Aimed at professionals, the Multi Z PIP combines premium DI box, mixer, and re-amping device in a single small unit. Paul White

This new DI box is built to audiophile standards using extremely high-quality components and metal case work. It has an external power supply with locking connector, and all the knobs and switches are on the front panel, along with the instrument input jack. The outputs and additional inputs are on the rear panel, where the main output is on a balanced XLR and the rest are on quarter-inch jacks. The Instrument Thru socket can send the instrument input on to another device, such as a guitar amplifier, and can double as a main output, a rear-panel main instrument input (overridden by the front-panel jack), or a summing amp input, combining another signal with that of the main instrument input. There is also a dedicated summing-amplifier input jack. The Balanced Only Expansion Output socket can act as a balanced insert send fed from before the instrument transformer. A re-amp input is included to allow recorded parts to be fed via the unit to a guitar amplifier or pedal at a suitable source impedance. A multi-turn trim pot accessible from the rear panel adjusts the gain of the re-amping signal path. The main and re-amped outputs are transformer coupled. Three ground-lift switches affect the instrument input, output XLR, and re-amp output. One of this unit's main claims to fame is its switchable input impedance — the user can select from high-, medium-, low-, or speaker-level input via a rotary switch. A larger rotary control adjusts the output level. The high-impedance setting is optimised for guitars and basses with passive pickups, while the

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Little Labs Multi Z PIP

medium and low impedances work with active guitars/basses and electronic instruments. There's no power soak or speaker simulator here, though, so if you're feeding it with a tube amplifier you need first to split the signal and hang a dummy load on the amplifier output to prevent the damage that might be caused by running it into an open circuit. It is also important not to try to use the unit as a speaker thru connection when working with speaker-level sources, as the high currents involved could toast the circuit board. The balanced line-level output is transformerless, but there is a transformer-isolated output as well, and you can even buy an optional mic-level output transformer. Internal jumpers allow a degree of user customisation, such as using the insert point to connect a volume pedal. As supplied, the buffered output jack comes before the level control, and can provide 6dB of gain for the medium- and high-impedance settings. Changing internal jumpers can also move the buffered output to a point after the level control, or can turn the buffered output jack into an input, allowing a further signal to be summed with the main input prior to the level control.

Verdict There's no doubt that this is a beautifully engineered unit capable of getting the best possible DI'd tone out of an electric guitar or bass. It is also very good at converting a recorded guitar track into a signal optimised to feed a guitar amplifier. I found that some switching noise was occasionally evident even after the unit had warmed up properly, but other than that it behaved perfectly. While professional studios will appreciate the benefits of such a good-sounding piece of kit, I feel that many UK project studio owners will deem the Multi Z PIP out of their price range. Nevertheless, if you want the best possible DI box for instruments, I can't think of one that sounds better. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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M Audio iControl

In this article:

The Look Using iControl With Other Software Double Up Getting Into The Garage Park It Here

M Audio iControl Control Surface For Apple GarageBand 2 Published in SOS December 2005 Print article : Close window

Reviews : Hardware Controller

M Audio iControl £130 pros Integrates snugly, as you would expect, with Garage Band. Clear layout, reflecting Garage Band's streamlined appearance. Good build quality. Inexpensive.

cons

There are dedicated (and expensive) control surfaces for Logic, Pro Tools and Cubase... so why not an affordable one for Apple's semi-free entry-level application Garage Band? M Audio must have thought exactly the same thing... Derek Johnson

Only works with Garage Band — and therefore only works on the Mac. Doesn't control everything in Garage Band.

It's always been a tricky business, designing control surfaces. On the one hand, you can make a controller that Not future-proof — if Garage can theoretically talk to any piece of software, but then what do you make it Band is drastically overhauled, iControl won't be look like? The control layout that works able to keep up. brilliantly for one sequencer or plug-in summary is inevitably no good for another, and the potential for confusion is then huge If you take your life with Garage Band seriously, then as the user hops between different it doesn't get much better than software while using the same this. To try one will be to buy Photos: Mike Cameron controller. On the other hand, you can one. create a control surface that is Spot the difference... M Audio's iControl is information dedicated to working with one piece of clearly modelled to look like a hardware £129 including VAT. version of Garage Band, seen running in the software, and is ergonomically background. But there are some welcome M Audio +44 (0)1923 completely in tune with it... but the differences, too — Garage Band's tiny Track 204010. problem then is that you can't easily buttons are much larger on iControl. +44 (0)1923 204039. use the controller to run anything else. Click here to email Generally speaking, these kinds of www.maudio.co.uk dedicated control surfaces also tend to be much more expensive than the generic type. Test Spec MAC REVIEW SYSTEM 450MHz Apple Mac G4 with 896MB of RAM running Mac

But here's a product that bucks that trend. M Audio's iControl is one such dedicated control surface, and it's a mere £129. As you can tell from its looks, it's designed to control Apple's 'semi-free' Garage Band 2 application (the iLife suite

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M Audio iControl

OS 10.3.9. Apple Garage Band v2.02.

of which Garage Band is a part costs £49 if you don't already have it, but it's supplied free with all new Macs). It's worth noting, though, that the version (2.0) of Garage Band included with iLife 05 needs upgrading to at least v2.0.1 to function with iControl; v2.0.2 is the latest download from Apple's web site. Before we go any further, I need to mention a couple of things. This controller doesn't really work very well with any other software on the Mac (see the box above for more on why), and none at all on PCs. So if you run a PC or don't use Garage Band, turn to the next review and put away your credit card!

The Look The rest of us can now mentally unpack a well-built, cleanly designed add-on to a widely available piece of Apple software. The iControl's size is just right, taking up less area than the magazine you're reading now. It's not so small that it's fiddly to use, and not so large that it dominates a desktop. The hardware is even equipped with faux wooden end cheeks, reflecting Garage Band's own virtual end-pieces! As delivered, the controller comes with a six-foot USB cable and a brief multilingual user guide (the English bit runs to seven pages). Since iControl is a buss-powered plug-and-play device, no driver CD is required. Everything worked first time for me, with Garage Band recognising the controller immediately. In addition to the USB socket round the back, there's a five-pin MIDI In socket, though it's undocumented in the manual. You'll notice it turning up in your MIDI applications as 'iControl Port 2', and it can be used by them. It seems to be there to provide a quick conduit to Garage Band for your MIDI controller keyboard if it's not itself equipped with USB, or if you've only one USB connection free on your Mac. In general, the iControl knob and button collection reflects the on-screen facilities of Garage Band, and allows you to undertake the majority of the important jobs, such as recording and mixing, Instrument tweaking, effects and EQ editing. The layout is really quite clean and straightforward, with loads of space around the controls. You might at first think that the legending is a bit plain, but as soon as the USB connection is made and Garage Band is fired up, subtle backlighting is enabled behind most of the white button graphics. The colour scheme of the backlighting even reflects that of Garage Band. In broad file:///F|/SoS/SoS%2012-2005/icontrol.htm (2 of 7)11/23/2005 3:02:11 PM

The user interface on the iControl is as coolly streamlined as... well, that on a piece of Apple software, say!

M Audio iControl

daylight, this backlighting can prove a little too subtle, but it works well in most other conditions. Straight out of the box, each of the eight 'tracks' is equipped with Mute, Solo, and Record-enable buttons, an endless rotary encoder, and a Sel(ect) button for choosing the track for editing. If your Garage Band song has more than eight tracks, you'll find that the two big arrow buttons in the centre of the iControl (labelled 'Track/Parameter') scroll through the track list in banks of eight. Over on the left, the two Volume and Pan buttons marked 'All Tracks' allow you to determine whether the eight small rotary encoders act as track-specific level controls or pan pots.While a fader for level control would have been ideal, the rotary encoders are a vast improvement over Garage Band's on-screen equivalents. Similarly, the track-specific buttons are refreshingly large, and much better than the miniscule, hard-to-mouse, on-screen equivalents. The jog wheel and transport bar work as you would expect: record, return to zero, rewind, play, fast-forward and loop buttons all reflect Garage Band's facilities, and the jog wheel simply moves Garage Band's playback position back and forth. Incidentally, it's possible to set playback or record loop points from the iControl: engaging the loop button whilst using the jog wheel or pressing the rewind or fast-forward buttons enables a loop, and sets start and end points for the looped region. The one fader on board the iControl drives the master volume control in Garage Band. It has a nice feel, though it's only small, and makes one rather wish that M Audio had been able to budget for faders all round. It'll take you a few goes to get used to the way that the hardware fader doesn't move its software twin until the software position has been matched, though.

Using iControl With Other Software While the main body of this review asserts that the iControl is exclusively dedicated to Garage Band, this isn't wholly true. In theory, the controller does its thing in exactly the same way as any other USB MIDI control surface. This means (again in theory) that if the target software is capable of manually accepting assignments of incoming control data to on-screen parameters, then iControl can do the job. In practice, however, it doesn't work very elegantly. I tried it with Ableton Live and Propellerhead Reason, and discovered that due to the way iControl's hardware is configured, assignments didn't always work properly. Chiefly, some buttons are momentary when they would ideally latch, and the rotary encoders tend to move between the extremes of an on-screen slider's (or knob's) travel with just two clicks. In any case, there's no way to switch 'banks', so even without such problems, you can't assign more parameters than there are physical controls on the surface of the iControl. Users of Apple's Logic may be better off; an Environment could be set up to respond to the incoming data. However, because iControl's knobs send incremental rather than absolute data (source of the 'jumping' problem I just mentioned), the objects in the Environment would have to be carefully tailored to file:///F|/SoS/SoS%2012-2005/icontrol.htm (3 of 7)11/23/2005 3:02:11 PM

M Audio iControl

work correctly. In general, though, I think it's fair to say that you probably wouldn't buy iControl unless you were using Garage Band a lot. Incidentally, Windows XP recognises the iControl when it's plugged in to a PC, but it's not recognised by any software I tried, though it appears as a USB audio (!) device in some Preference windows.

Double Up With this number of hardware controls, some doubling of button and knob assignments is inevitable. With the iControl, though, this doesn't lead to operational confusion; even a complete newcomer to music on computers, with the manual in hand and Garage Band on screen, should be able to get going fairly quickly. Five buttons, grouped together on the left under the 'Selected Track' label, cause various windows to appear in Garage Band, at which point the Sel buttons and rotary encoders perform a range of different functions, depending on which of the Selected Track buttons you've pressed. Of these, the first, the Track Info button, invokes Garage Band's window of the same name (shown opposite). Here, you see exactly what sort of Instrument has been assigned to the track (Garage Band, as you may know, distinguishes between 'real' Instruments, which play back sampled loops or your own audio, and 'software' Instruments). You also see the Instrument itself, and the selection of 'effects' (in Garage Band's broad sense of the word) that have been enabled for each track. On offer, as you may know, are a compressor, two variable effects, EQ, echo and reverb. 'Real' Instruments also have a noise gate as part of their arsenal. The Select buttons enable or disable any available effects, and the rotary encoders alter the global value of that effect. One problem here is that if either of the two effects proper have not been assigned to a track, there's no way to make the assignments from the iControl — it's not quite sophisticated enough. Once you've made those choices, you can turn the effects on or off, but that's it. From this window, you can do nothing more than turn EQ on or off. It's also not possible to select Instruments or change those that have been assigned to a track from iControl. And as for selecting MIDI or sampled loops from the Garage Band library or setting up an audio recording... well, put simply, you can't. In short, you'll still need your mouse and ASCII keyboard to hand for several significant operations. Underneath Track Info, the next button,

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Here's the Garage Band Track Info window for a 'real' Instrument (software Instruments lack audio routing and the gate effect). You'll be able to switch effects on and off and tweak any

M Audio iControl

Generator, can be accessed whether the Track Info window is active or not; it invokes Track Info first if it's not visible. When a software Instrument is assigned to a track, this button brings up the parameter-editing window for the Instrument and the rotary encoders take the part of the onscreen parameter sliders.

of the on-screen sliders from the iControl, but not choose effects, effect presets or select Instruments from scratch. In this window, the audio input controls are also inaccessible from the iControl.

The last three buttons — Effect 1, Effect 2 and EQ — call up the relevant windows for editing those particular Garage Band elements. Again, the rotary encoders do all the work here, by being assigned to each on-screen parameter from the top down. If there's only two parameters in an effect, it'll be the top two encoders that come into play, and so on. These three windows have the same shortcoming as the Generator window: there is no way to select or save presets from the iControl. Incidentally, there's an 'Option' button on the iControl that doesn't seem to do much, but it does allow you to quickly enable or disable Effect 1, Effect 2 and/or EQ no matter which windows are, or aren't, open. You just press Option and the required 'Selected Track' button. No matter which window you call up from iControl, it'll take a short while for you to become accustomed to how the Sel buttons and encoders relate to the various on-screen elements. The correspondence is not always immediately logical, though when working in the Track Info window, the Sel buttons light up when the matching switch is engaged on screen. But as I noted earlier, the backlights are not very clear in daylight, especially on these switches. They also dim when disengaged, so you still may not be sure which hardware button is equivalent to which software button if they're all disengaged. The manual explains this all very clearly, so you can keep that page open until you become accustomed to the layout. Incidentally, there's no 'echo' or 'reverb' edit buttons because these effects are uneditable inside Garage Band: they're either on or off with an amount value, both of which can be set from the iControl.

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M Audio iControl

Getting Into The Garage I was never that taken with Garage Band, even as a freebie. But Garage Band 2 is much better. The initial basic options for recording and crucially, editing, your own MIDI or audio data have been very much enhanced. You still need ideas, but Garage Band is now more useful for someone who might have a little creative ambition, offering more room for growth for those who are introduced to the idea of creating tracks by stacking the supplied loops. The interface is simple and the terminology is made more comprehensible to the novice computer musician ('Align to...' is used in place of 'quantise', for example). Certainly, the loops are there if you really want them (and I must admit that I've borrowed the odd drum loop or two as a way of quickly mapping out a song), but they're no longer Garage Band's main reason for existence. The program also feels a lot less bloated than the first release. The supplied Apple Loops still take up a lot of disk space, and the Garage Band application is huge, but the whole thing seems to zip along much more smoothly — I notice things like this on my soon-to-be-retired 450MHz G4!

Park It Here There isn't really a lot more to say. In describing the available facilities, I've introduced practically everything the iControl does. Of course, as I've already explained, there are things you can't do with iControl, although sometimes this is because Garage Band itself doesn't offer the facility (iControl button presses and controller moves, for example, can't be recorded by Garage Band for automated playback, because the software doesn't support this option yet). But there are plenty of Garage Band features you can't access. There's no way to enable or disable the metronome, access track-edit windows, or change tempo, and there are no file-management options. But these features, and any further metering or display options, would have added more components and increased the controller's cost. And if M Audio had gone this far, you'd still have needed to use an ASCII keyboard and mouse for track editing and file naming, say. In general, the balance is largely right. If there's a problem, it's the same one I mentioned at the start of this article — if you design a dedicated controller for a certain piece of software, you immediately put off all the people who don't use that software. And any major changes to the software can potentially leave the hardware behind. There was considerable excitement a couple of years ago at the idea of a dedicated controller for Propellerhead's Reason, proposed by Novation. However, in the end, what Novation brought to market was the Remote 25 — a generic, non-specific controller. Furthermore, although Garage Band may be installed on all current and many other Macs, not every user will be playing with it — do you use everything Apple puts on your hard drive? And PC users will never, as far as I know, get the chance of using Garage Band, which permanently locks out a large number of file:///F|/SoS/SoS%2012-2005/icontrol.htm (6 of 7)11/23/2005 3:02:11 PM

M Audio iControl

potential users. Despite these concerns, if you do meet the criteria for using iControl, it's a great product. I enjoyed working with it tremendously and found that it elegantly opened up those aspects of Garage Band that it can reach. And working with this controller also caused me to reassess Garage Band in a favourable light. Now, I couldn't imagine working full-time with Garage Band without an iControl in tow. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Sample Libraries: On Test

In this article:

Sample Libraries: On Test

True Strike 1 ***** Hot Releases Assessed Flatpack 2 **** Drumdrops In Dub Volume 1 Published in SOS December 2005 ***** Print article : Close window

Star Cartoon Sidekicks

Reviews : Sound/Song Library

***** Mutley **** Penfold *** Barney Rubble

True Strike 1 *****

** Godzuki

MULTI-FORMAT

* Scrappy Doo

Dutch samplists Project SAM won a lot of friends with their quartet of brass titles (SAM Horns, SAM Trombones, SAM Trumpets, and SAM Solo Sessions, reviewed in SOS February and June 2003, and March and October 2004 respectively). These libraries provide what most MIDI orchestral composers need: realistic, accurate, bright-sounding samples recorded from a choice of three listening perspectives, covering a wide range of dynamics and articulations. Project SAM repeat the winning formula in their new library — the same Utrecht concert hall and mic placements are used, but this time the source material is orchestral percussion. True Strike 1 consists of a large menu of traditional and contemporary orchestral percussion instruments, both tuned and unpitched, supplemented by an array of Latin and effects percussion, metal sounds, and a few unremarkable 'body noises' (claps, finger snaps, and so on). In addition, there are effective unison hits from bass drum and timpani, bass drum and crash cymbals, and bass drum and tam tam. The library's tally of around 50 instruments makes it one of the most comprehensive orchestral percussion titles on the market. Project SAM's percussive debut is available as a Gigastudio sound library or as a virtual instrument. The 16-bit Gigastudio 2 version is 12GB in size, while the 17.6GB Gigastudio 3 option offers the same samples in 24-bit form. The 24-bit instrument is powered by the Steinberg Halion Player, available as a full Professional model or a half-price Starter Edition — the latter omits ten instruments (including marimba), some effects performances, and two of the mic positions.

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Sample Libraries: On Test

I used the Gigastudio 2 version to add stock orchestral percussion to a TV-music cue, and was immediately impressed by the low-end wallop of the timpani and bass drum, so much so that I began to fear for my speaker cones. In the 'far' miking, the hall ambience adds explosive force to these booming hits. At first I feared the piatti samples might prove less potent, but after wading through the obligatory batch of choked and semi-choked hits, I found the bright, triumphal cymbal splashes I was looking for. The library's layout is straightforward and logical, so finding the right sounds took only a few minutes. Timpani single notes consist of two alternative sets of samples (a great help in making repeated notes sound lifelike), mapped two octaves apart to facilitate twohanded programming. These clean, resonant hits are supplemented by a sizeable menu of crescendo and diminuendo rolls of various lengths, glissando (ie. changing pitch) rolls, and looped tremolos. Medium beaters are used on the timps throughout, with hard beater hits adding a fortissimo bang to some programs. Project SAM uphold their reputation for loony experimentalism by playing an upside-down cymbal laid on a timpani skin while bending the drum's pitch with a pedal, creating some intriguing spooky effects. Tom toms often end up sounding emasculated and out of place in orchestral libraries, but SAM's set of four are a knockout, performing strong, clean, melodicsounding single hits, flams, rolls, and smartly executed multi-tom phrases. Sampled at six dynamic levels for maximum realism, the snare and side drums also sound good, covering all the common orchestral styles and throwing in some distinctly unclassical Bruford-esque whangy rim shots for good measure. Experimenting with the cracking three-player snare section samples, I was pleased to find that the three listening perspectives can be layered without introducing any obvious phase cancellation or flamming; adding the close-miked samples to the 'stage' and 'far' sets simply increases the attack definition. The tuned percussion ranges from the sparkling high-frequency tones of glockenspiel, xylophone, and crotales (all superb), to the more mellow timbres of marimba and vibraphone. Although the latter pair sound splendid, both suffer from an awkward lurch in timbre between their softer strokes and the far brighter, more percussive loud hits — an intermediary mezzo forte dynamic layer would have helped bridge the gap. I was unhappy with the programming of the vibraphone's long notes; the samples ring on long after the key is released, making pianistic playing impossible. Let's hope Project SAM fix this in an update. The tuned percussion is completed by a pretty celeste (sounding most effective from the close-miked position) and some vintage, churchy tubular bells. The producers have sampled small items like temple blocks, tambourines, and triangles in reasonable detail, resulting in unusual deliveries such as gradually accelerating woodblock rolls — believe it or not, quite a dramatic effect! Claves, castanets, vibraslap, and rattle are grouped together into one handy program with a few other bits and pieces including eighth-note sleigh-bell loops played at different tempos. Simply add a Noddy Holder sample, and there's next year's Curry's Christmas ad! Effects include entertaining scrap-metal and brake-drum hits, a less-than-thunderous thunder plate (imagine a metallic version of Rolf

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Sample Libraries: On Test

Harris' wobble board), some rather tame whip cracks, and a scary siren which had me cowering under the kitchen table waiting for the 'all clear'. Latin percussion is a nice bonus, and although it's unusual to hear familiar pop sounds like shakers, guiro, bongos, and timbales recorded in such an ambient space, the hall reverb really suits the conga drums. The far east is represented by Chinese cymbals and tam tam, but other parts of the globe go largely unexplored — three deep-pitched Thai gongs, tuned to the notes of 'F', 'A', and 'C', are about as esoteric as this library gets. There is very little dead wood in this percussion collection. Most of its sounds are highly usable, and immediately sound convincing in an arrangement without having to be tweaked. The instrumentation is less exotically varied and the performance options more limited than in the 30GB Vienna percussion set, but this title is cheaper! True Strike 1 hits the mark — its musicality, instrumental scope, and excellent concert-hall sound fully deserve a five-star rating. Dave Stewart Professional Edition: Gigastudio 2 DVD-ROMs, £226; Gigastudio 3 DVD-ROMs, £239; Halion Player instrument, £239. Starter Edition: Halion Player instrument: £117. Prices include VAT. Time & Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.projectsam.com

Flatpack 2 **** REFILL Lapjockey have been busy since their excellent Flatpack Refill was reviewed in SOS July 2003. The group have contributed material to Reason 3's factory Refill, and their experience with the new Combinator device has resulted in two things: firstly, Flatpack 2 was restarted from scratch to include the Combinator; and, secondly, Flatpack 2 has become one of the first commercial releases to really focus on the device. The numbers are good: 300 Combinator patches plus 200 NN19, 50 NNXT, 20 Redrum, 50 RV7000, and 20 Scream 4 patches, not to mention the further 200 REX loops and over 400 drum samples. All is easily accessible, and the user is encouraged to create, but most of us will sample the Combis first. In Flatpack 2, Combinator becomes the basis for five 'shells', each of which behaves as a sort of new mega-instrument complete with custom graphics and thoughtful use of assignable controls. Approach the Kilburn, Scope, Boxmoor, and Rexdex variants as if they were new Reason sound-making devices and I think you'll be jiving with the Lapjockey crew. The fifth shell, Outboard, features chains of vintage-feel send and insert effects. file:///F|/SoS/SoS%2012-2005/sampleshop.htm (3 of 6)11/23/2005 3:02:24 PM

Sample Libraries: On Test

Think synths, think Kilburn: raw synthwaveform samples from the team's favourites are loaded into devices that are wired up to emulate those same synths' signal paths — the patch abbreviations offer clues as to which instruments (CS, SH, Odyss, Source, and so on). The approach is actually rather successful, and manages to extend well beyond what I think of as Reason's signature sound. If your synth and sample tastes are more impressionistic, load up the Scope Combis: this family is dedicated to textures, pads and soundscapes. Here you can really hear what Combinator is capable of, and the epic patch Gienah Vangelis sums up the approach admirably. Drums are handled by Boxmoor. There are plenty of strange hits in the kits, plus distortion that's just the right side of sandpaper, for example in the Slipmat patch. If Boxmoor has a weakness, it's that its central device is Redrum — a novel approach to drums might have been attempted. But this is also the shell's strength: Redrum patterns from anywhere will be in shouting distance of compatibility. Now introduce yourself to Rexdex, which applies DJ-style control to loops in Reason. All the loops, produced by the Dr:Rex and Matrix devices, are nicely energetic and processed — just dig in. Raw samples and programming are excellent throughout, and the Jockeys don't shy away from the unexpected: the set is sample heavy, of course, but NN19 is often used rather than NNXT with its heftier CPU load. Beyond the presets, 'initialised' patches for many of the shells let you programme from scratch — a rewarding process. Overall, the set seems like pretty good value, and a nice development of the Flatpack concept whilst being a great collection in its own right. Derek Johnson

Reason Refill CD-ROM, $95 (around £50). Click here to email www.lapjockey.com

Drumdrops In Dub Volume 1 ***** MULTI-FORMAT This is another library from UK-based drum specialists Drumdrops, whose Fistful Of Drummers and Drops In The Bronx have both already received very favourable reviews in SOS. As with other titles in their range, this one is available in two formats; either a single DVD containing 16-bit mixed loops or, at a slightly

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Sample Libraries: On Test

higher price, a DVD set featuring 24-bit multi-track performances. In this case, the drumming is supplied by Style Scott (who has played with some high-profile reggae artists such as Gregory Isaacs and Barrington Levy), while some additional percussion comes courtesy of Norman Grant (Twinkle Brothers) and Tuca Rainbow. The playing is uniformly excellent and, in keeping with the Drumdrops ethos, the recording process is all analogue, including some classic spring reverb and echo units. I explored the WAV files using Sony Acid Pro 5, and the standard stereo loop collection is organised into drum loops (400 — although some are duplicates with different processing), percussion loops (approximately 200) and a large collection of single hits and additional fills. All the loops are organised into tempo-based folders, ranging between 69bpm and 142bpm. In the case of the drum loops, each folder provides a useful collection of related loops, making it very easy to mix and match for song construction. The sound is also consistent enough between folders for easy combining. A mixture of (very) Dry, FX, and Dub versions of some loops are provided — with the latter having plenty of that classic Reggae-style ambience added! The percussion loops add some further variety, with hi-hat, shaker, tambourine, bongos, and scrapers amongst the instruments used. As with Fist Full Of Drummers, the multi-track version provides complete performances, some running to over four minutes. Each performance includes the kick, snare, hi-hat, tom, overhead, and room microphones. Drumdrops also provide Pro Tools, Logic and Cubase session files for each performance, and these include tempo maps to enable accurate quantising of any MIDI tracks added to the project. Throughout, the character of the playing, the classic sound, and the very distinctive Reggae-based rhythms mean Drumdrops In Dub is certainly not just another 'me too!' drum-loop library. As a collection with its roots firmly in Reggae, this library might sound like it would have limited appeal. However dub influences can be heard in all sorts of different musical styles and I could certainly imagine these loops working in some hip-hop, trip-hop, or even rock (think The Police or some No Doubt material). The sound also has a very authentic '70s Jamaican vibe — very cool! While the stereo collection might seem a little pricey, if you like an extra degree of control the multi-track set is excellent value for money. John Walden 16-bit Apple Loops, REX 2, and WAV DVD-ROM, £75; 24-bit multitrack AIFF 3-DVD-ROM set, £99. Prices include VAT. +44 (0)1273 553106. Click here to email www.drumdrops.com Published in SOS December 2005

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Sample Libraries: On Test

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

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

file:///F|/SoS/SoS%2012-2005/sampleshop.htm (6 of 6)11/23/2005 3:02:24 PM

Sony Oxford Limiter

In this article:

Sony Oxford Limiter

Reading Between The Lines Mastering Limiter Plug-in All In A Dither Published in SOS December 2005 Enhanced Prospects In Use Print article : Close window

Sony Oxford Limiter £347/£229

For Pro Tools

Reviews : Software

pros Easy to use. Can detect and manage inter-sample peaks as well as conventional 'overs'. Unique Enhance function is very effective for maximising loudness.

Like many of Sony Oxford's plug-ins, their new limiter takes a familiar concept and applies a novel twist. Sam Inglis

cons Setting up the right level of enhancement requires very careful listening.

summary A well specified mastering limiter with a special Enhance function that sets it apart from much of the competition.

information TDM version £346.63; RTAS version £229.13. Prices include VAT. HHB +44 (0)20 8962 5000. +44 (0)20 8962 5050. Click here to email www.hhb.co.uk www.sonyplugins.com

Test Spec Centrino laptop with 2.0GHz Pentium-M CPU and 2GB RAM, running Windows XP Service Pack 2. Tested with Digidesign Pro Tools M-Powered v6.8.

The idea behind Sony's Oxford Limiter is similar to that of Waves' L2 and many other wide-band mastering limiters: it can make your music louder without introducing clipping. However, Oxford Limiter also has a unique selling point in Sony's proprietary Enhance function. This, it's claimed, can preserve fragile transient information even when the audio is being pushed hard against the end-stops, allowing you to achieve greater subjective loudness with less damage to the signal.

Reading Between The Lines Oxford Limiter is available in TDM and RTAS formats on Mac OS X and Windows, and is authorised to an iLok key. The limiter itself is a pretty comprehensive design, using lookahead detection to anticipate peaks and intelligent adaptive processing to tame them, and featuring a detection algorithm that is capable of spotting not only sample values that would hit 0dBFS, but also inter-sample peaks that might clip the D-A converter when the signal is reconstructed. The output level meters go to +6dB in order to display these peaks when the Recon Meter button is selected, and a second button labelled Auto Comp automatically brings the level down for an instant whenever such a peak is encountered, providing either a safety net or a second layer of limiting, depending on how far you're pushing the output level. The ability to detect and

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Sony Oxford Limiter

neutralise illegal inter-sample peaks can come in very handy even if you don't plan to use the limiter function at all. Sony say that the limiter section was designed with two separate purposes in mind. The first is the kind of fast peak limiting required in order to maximise loudness without clipping, while the second is management of gain changes over a much longer period. To aid in the latter, the limiter's release time is variable up to 10 seconds, and there is also an Auto Gain button, in essence an automated volume control that can smooth out the kind of dynamic variations in the source material that occur over a timescale of seconds or even minutes. Further control comes in the shape of a Soft Knee parameter, which can reduce the threshold at which limiting begins to take place from 0dB anywhere down to 10dB. As you'd expect, 'harder' settings give punchier results, but 'softer' ones often sound smoother and less obvious in action. The most unusual feature of the limiter section, though, is the Attack time control, which is variable from 0.05ms to 1ms. Most limiters are simply designed to have as short an attack time as possible — in fact a limiter is sometimes defined as a compressor with instantaneous attack and an infinite compression ratio — and this is usually desirable, since the most prominent peaks in an input signal are often due to transient spikes that rise almost instantly. Here, however, the ability to slow the attack time has two functions. First, it can be more transparent when you're using Oxford Limiter solely for long-term gain reduction rather than loudness maximising. And second, it allows transient peaks to pass through the limiter algorithm and be dealt with by the Enhance function.

All In A Dither If you're using a mastering limiter, it should always be the last element in any signal-processing chain before any word-length reduction that might be necessary to fit your material to a particular release format such as CD. Like many mastering limiters, Oxford Limiter builds in a dither section at the output stage to convert its output to the appropriate bit depth for whatever medium your music is intended — since the plug-in's internal resolution is higher than 24-bit, dither is applied whether you choose 24- or 16-bit output. Five noise shapes are available. The default high-pass Triangular Probability Density Function shape distributes the dither noise fairly evenly across the frequency spectrum, while noise-shaping Types 1 to 4 offer different distributions which concentrate the noise in particular areas of the spectrum. Applying more shaping to the dither noise can make it less audible, but can also have undesirable consequences, so Sony also provide a Depth control which allows you to achieve the best balance by varying the extent to which the noise is shaped.

Enhanced Prospects The manual states that the Enhancement process 'enhances the perceived loudness and presence of the programme by modifying the dynamic and file:///F|/SoS/SoS%2012-2005/oxfordlimiter.htm (2 of 4)11/23/2005 3:02:27 PM

Sony Oxford Limiter

harmonic content of the signal', somehow preserving the impact of transients in a heavily limited signal whilst ensuring that they never exceed 0dBFS. No explanation is forthcoming as to how the process actually works, but I suspect that it involves momentarily cutting energy from the low-frequency end of the spectrum in order to keep more of the high frequencies that are characteristic of most transient sounds. The manual suggests that Enhance will be more effective on fuller mixes than on solo instrument tracks and other sparse recordings, and my tests bore this out: there was obvious distortion on a solo bass or acoustic guitar at high Enhance values, but with full band recordings, the function allowed me to add several dBs worth of perceived loudness over and above what could be achieved through limiting alone. However, I found it essential to listen to the results very carefully, and preferably on headphones. When you go slightly too far with the enhancement, the distortion that results is unpleasant but not always obvious; what's more, it usually shows up only in a few small sections of the programme material, and not always where you might expect.

In Use As with most mastering limiters, Oxford Limiter's interface is very simple, and I only have a couple of very minor niggles: one is that parameter values are set by the position of the top of each slider, not the centre point, which takes getting used to; the other is that I think the gain-reduction meter would benefit from slightly more resolution at the bottom end of the range. In general, though, I found Oxford Limiter easy to set up and flexible enough to cope with all of the obvious applications. Of course, there will be mastering jobs where a multi-band design is needed in order to tackle a particular problem with the programme material, and the likes of Waves L3 offer more powerful tools for problem-solving, as well as more radical ability to change the sonic character of a mix. However, I didn't find that L3 offered a noticeable advantage over Oxford Limiter when it came to achieving maximum loudness: the latter's Enhance function is a simpler but effective alternative to L3's psychoacoustic processing. The ability to seek and destroy illegal inter-sample peaks is also handled very straightforwardly, and the Soft Knee option complements some material nicely. In most 'normal' mastering situations I found that the Auto Gain function made no difference, but there are situations where it comes in handy: if, for example, you have to apply heavy limiting to a very dynamic classical piece, perhaps for broadcast use, it does help make sudden transitions from loud to quiet sections sound more natural. Oxford Limiter offers all that you'd expect from a conventional wide-band limiter along with the unique Enhance option, and I can see it becoming a musthave for mastering studios. Published in SOS December 2005

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Sony Oxford Limiter

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

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

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TL Audio M4

In this article:

Connections & Digital Interfacing Options Channel Facilities Output Section In The Studio With The M4 A VTC For The Masses?

TL Audio M4 Valve Mixer Published in SOS December 2005 Print article : Close window

Reviews : Mixer

TL Audio M4 pros Characteristic TL Audio sound. Excellent build quality.

This new mixer caters for those people who found TL Audio's M3 too small, and the flagship VTC too big. But is the M4 just right?

Musical and controllable EQ. Optional digital I/O cards. Separate sockets for insert Hugh Robjohns sends and returns. Switchable I/O operating TL Audio are well known for their levels. various ranges of hybrid valve and Nice, spacious control ergonomics. solid-state audio processors. They Competitive UK pricing. also one of a very small set of

are

manufacturers still making valve-based mixers. Although TL Audio's consoles Poor monitoring facilities. are not 'true' valve desks, they manage Split master output faders. to achieve modern performance No groups or routing. specifications with classic 'tube' sonics No slate talkback. at pragmatic prices by using hybrid summary designs combining solid-state The M4 fills the gap between Photos: Mark Ewing electronics with dual-triode valve the M3 tracking console and stages. Purists might hanker for the full VTC flagship, designs that only employ valves, but in reality achieving modern performance providing up to 32 channels with versatile and musical EQ, expectations with all-valve designs costs far more than even the most fanatical excellent I/O facilities, and old- are now willing to pay. cons

world ergonomics. The output section is disappointingly limited, but for many the console's distinctive sound quality will outweigh its monitoring deficiencies.

information 16-channel model, £4694; 24-channel model (as reviewed), £5869; 32channel model, £7044. TL Audio +44 (0)1462 492090.

The company's first offering in the valve console market was the impressive VTC desk (reviewed back in SOS February 2000), which has acquired a strong reputation. Its baby brother, the M3 8:2 tracking console has also become a popular addition to more affluent DAW-based home studios, maintaining much of the design ethos and sonic quality of the VTC, but in a smaller and more appropriate package. The new M4 is a larger and slightly more sophisticated version of the M3, on which it is closely based, neatly bridging the gap between the M3 and VTC. Like the M3, the new M4 is essentially a tracking and stereo mixdown console. It

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TL Audio M4

+44 (0)1462 492097. Click here to email www.tlaudio.co.uk

is available in three frame sizes from 16 to 32 channels, and I reviewed the 24channel version. The input channels are constructed in sub-frames of eight strips which bolt either side of a narrow master, control, and output section. An external (and quiet — hurrah!) 2U rackmounting power unit supplies 300W into the biggest 32 channel frame, and 200W into the smaller 16 channel console. A twometre connecting cable is supplied as standard. The console is raked at a comfortable angle that allows all the control settings to be seen easily, and the controls themselves are well spaced and easy to adjust. In fact, the whole style and ergonomics are reminiscent of a console from the '60s or early '70s, with lots of space between the nicely weighted knobs and the man-sized buttons. Everything feels solid and well built, right down to the oiled oak end cheeks and armrest. Everything, that is, apart from the faders, which feel strangely lightweight and insubstantial compared to the rest of the controls.

Connections & Digital Interfacing Options On the rear panel, the mic inputs and main stereo outputs are on XLRs, but everything else is on balanced TRS jack sockets. Most inputs and all outputs are individually switchable for +4dBu or -10dBV operation, which allows easy connection with a range of pro and semi-pro gear. A nice touch is that each channel features an insert point with separate balanced send and return sockets. Optional DO8 digital boards can be installed in each input module to provide ADAT lightpipe I/O for simple connection to a DAW or other recorder, and an optional DO2 card can be installed in the master section to provide an S/PDIF stereo output. The latter has a fixed 24-bit word length, but the sampling rate can be selected from 44.1, 48, 88.2, or 96kHz. Each DO8 and DO2 card is entirely autonomous, but external word-clock inputs are provided for synchronisation. Rather unusually for a console of this size, the M4 has no groups and no bussrouting facilities, other than to the main stereo outputs. If you want groups and routing you'll need to move up to the VTC. Instead, every channel has a direct output that also feeds the optional DO8 card's A-D converter. This direct output can be switched before or after the EQ, and is independent of the channel fader and Mute button. A dedicated Track send-level control optimises the headroom of your recording device, providing ±15dB trim and a maximum output level of +22dBu — sufficient to fully drive any professional A-D converter. While this direct output arrangement is great for tracking, allowing every channel to be recorded separately to 'tape' with or without EQ, it does complicate things slightly when it comes to overdubbing at a later date, or when building up a mix by recording each track in isolation. You either have to re-connect the mic to the appropriate channel to access a specific track on the recorder, or you have to reallocate inputs on your DAW. Neither is ideal for anyone accustomed to the flexibility of buss- or track-routing switches.

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TL Audio M4

Channel Facilities Each input channel has separate mic and line inputs (the DO8 card returns being normalled to the line inputs) selected with a front-panel switch, so that a multitrack recording can be remixed through the console simply by flipping all the input source switches. The mic input section affords a gain range of 44dB from +16dB to +60dB, and includes a switchable 30dB pad — it sounds a lot, but with a minimum gain setting of +16dB, a hot kick-drum mic could exercise the full 30dB of attenuation on offer. Every channel has individual phantompower switching, but the button is tucked away on the rear panel next to the input XLR, and with no indication on the channel strip to confirm whether phantom is switched on or not. A polarity reversal switch and 90Hz highpass filter (12dB/octave) apply to both mic and line inputs. The line input features a gain swing of ±20dB, with a maximum input level of 26dBu — more than enough to cope with the hottest of D-A converter outputs. The rest of the channel facilities are very similar to those of the M3, except that where the smaller console had two aux busses the M4 boasts four, the first two independently switchable pre/post-fader and the last two permanently post-fader. The EQ section is identical, with fixed-frequency high (12kHz) and low (80Hz) shelf equalisation and a pair of swept mid-bands (50Hz-2kHz and 500Hz-18kHz). The whole EQ section can be bypassed, and the ±15dB gain controls are all centre-detented. The two shelf equalisers have unusually steep slopes at 12dB/ octave, although they work very well in practice, and the mid-band sections have a bandwidth of 1.5 octaves (corresponding to a Q value of 0.7), which is narrow enough to allow precise tonal correction without sounding too peaky. The insert point is normally pre-EQ, but can be switched post-EQ from the front panel, and the return can also be switched out of circuit if required. Channel Mute and PFL buttons (with LEDs) are mounted on the fader panel above each 100mm fader, along with yellow Drive and red Peak LEDs. The Drive light gives some idea of how hard the valve in the input stage is being driven, glowing progressively brighter with signal levels between +6dBu and +16dBu. The harder the input stage is driven, the greater the characteristic harmonic distortion the valve provides. The red Peak LED illuminates at +21dBu, 5dB below the actual clip point, and indicates the peak-level signal monitored at the input amp and post-fader buffer of each channel. The desk has an impressive internal headroom, but I found that when it did finally clip it did so in a way typical of solid-state op amps, rather than giving the benign crunch of an all-valve design.

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TL Audio M4

Output Section The reasonably well-equipped input channels are, to my mind at least, let down slightly by the rather limited facilities of the narrow output section. Starting at the top, a pair of retro-styled round VU meters (with individual calibration trimmers recessed behind the front panel) are assisted by separate Peak LEDs. These meters follow the monitored signal, so indicate the two-track return or PFL channel levels instead of the mains stereo mix when appropriate. The main stereo mix output level is controlled with a pair of 100mm faders, spaced in the same way as the channel faders, and with the same 10dB of gain in hand. The main outputs are equipped with an insert point to allow the easy patching of a buss compressor, for example. The four Aux Master output-level controls each have associated PFL buttons and LEDs, and there are two stereo effect returns with level and balance controls and PFL buttons. If a signal is patched only to a left effect-return socket it is automatically normalled to the right as well, providing a centred mono return which can then be panned as required. The main monitoring level control has a delightfully large knob which is associated with a Mute button and a two-track return switch. There is a separate volume control for the headphone output (located on the fader panel next to the main faders), and the headphone output signal is always the same as that driving the main monitors. A PFL level-trim knob is provided to allow the monitoring level to be matched between the full mix and a soloed channel (over a ±20dB range). The only other facility on the control section is talkback, with an unpowered XLR socket, a volume control, and a pair of routing buttons. These allocate the talkback signal to either or both of the first two auxes. These master-section facilities are the minimum that would be required, and I would expect many potential customers to find them frustratingly basic — even the cheapest stand-alone monitor controller offers a more complete palette. For example, why is there no power to the talkback mic socket, when phantom is provided throughout the desk anyway? Although suitable dynamic mics are available, there are also a lot of electret gooseneck designs that require power. Why no facility to slate talkback to the main outputs? Why no facility to route the main stereo mix, two-track return, or effect returns to the aux outputs? Why no Dim control on the monitoring? Why no mono-check facility? Why only a single two-track return and no secondary monitor output? It's all a bit underwhelming and inflexible for a console in this UK price bracket. Another concern I have is to do with the output faders. In the first instance, their wide spacing makes it extremely difficult to keep the stereo image stable when fading up and down — even if you use a plastic fader clip or the infamous 'pencil and sticky tape' alternative! It is also most unusual to have 10dB of gain available above unity in a master fader, and this makes fading up less straightforward than it would normally be, as well as allowing one fader to be knocked relative to the

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TL Audio M4

other more easily, resulting in off-centre images on the main output too. There's nothing as reassuring (and easy to check) as having the main output faders pushed up hard against their end stops. Overall, a single ganged stereo master fader, with unity gain at the top, would have been far more practical.

In The Studio With The M4 The M4 has many appealing features, and the control-surface ergonomics are excellent. In particular, having plenty of space around the controls means you can adjust things without having to use a pencil sharpener on your fingers first! All the controls feel like high-quality units, and there is only the faintest of clicks when switching the inserts and EQs in and out of circuit. Although, as I mentioned earlier, the faders feel light and flimsy in comparison to the rotary controls, they are silent in operation and deliver the goods well enough. The overall sound quality is typical TL Audio, nicely warmed by the high-anode-voltage valve circuitry, but not overblown at all. Of course, cranking up the input levels drives the input-stage valve harder to deliver a much richer, more obviously 'valve enhanced' sound. However, while this can be useful for the odd effect, the real strength of the console is in the very subtle addition that the valves make when operating at more conventional levels. The input section struggles a little to provide enough gain (or a low enough noise floor) to cope with low-output ribbon mics, but with typical condenser and dynamic mics coupled with normal close-miking techniques the 60dB on offer is perfectly sufficient. The mic input EIN specification figure of -127dBu is not the best around, but is quite respectable for a console stuffed full of valves, and more than adequate for the applications this desk is likely to find itself employed in. While I'm reciting specifications, the overall distortion from line input to main output is quoted as 0.4 percent. This is a pretty high figure compared to most modern solid-state equipment, but it is entirely commensurate with the design intention of the console — it is a deliberately 'coloured' desk, and the distortion is mainly comprised of second harmonics generated by the valve stages. The output noise is quoted as a rather modest -72dBu, but if the output levels are being driven fairly hard (as they would have to be to fully modulate the input to a professional A-D converter) you could add easily another 15dB to that figure. The EQ section is excellent, always sounding musical and complementary, but with sufficient bite and overlap in the swept mid-bands to really get to grips with any tonal source deficiencies when necessary. Switchable bandwidth on the midbands — as available on the VTC — would have been a nice additional feature, but that really would have been only the icing on an already very tasty cake. Its file:///F|/SoS/SoS%2012-2005/tlaudiom4.htm (5 of 7)11/23/2005 3:02:31 PM

TL Audio M4

nice that the EQ bypass button was retained too , because so many desks leave this vital facility out, and the ability to bypass and reposition the insert point is also welcome. The separate balanced connections for the send and return also make life easy — so much better than having to use unbalanced 'Y' cables — and the ability to fine-tune the direct output levels from the console surface is a very worthwhile feature. Four aux sends should be enough for most applications, especially given that a lot of people tend to use internal DAW plug-in effects rather than outboard processing and reverbs these days. Clearly, the desk is designed such that the first two auxes can be used for headphone cue mixes. However, this facility turns out to be frustratingly difficult to use in many situations. For example, there is no easy way of routing the main stereo mix or the two-track return to the appropriate aux outputs. Similarly, if you are using outboard reverbs, it would make sense to bring them back to the desk via the two effect returns, but there is no way you can then apply a little reverb to the headphone cue to help a vocalist.

A VTC For The Masses? It seems a shame that a console with so many good points should have an impractical master-fader configuration and such a surprising lack of flexibility in the monitoring section. To me this is the ha'porth of tar that has spoiled this particular ship. The front end of the console is excellent for its intended task — the channel strips are well equipped and sound superb, and although the routing arrangements (or rather the lack of them) force a specific way of working, I don't see that as a significant practical problem at all. Essentially, the front end is a bigger and more flexible version of the M3, and an ideal large-scale tracking mixer for working with a DAW or hardware digital recorder. However, whereas limited functionality might be acceptable in the monitoring section of the compact M3 — and a separate monitor controller is a reasonably practical and affordable solution — this cannot be true in a console as large and relatively expensive as the M4. Most of the facilities anyone would reasonably expect are all provided in the VTC: switching for alternative speakers, mono check, multiple external monitoring returns, the ability to slate talkback to the main output, source selection, and effects-return routing to cue-mix headphones — so it's not as if this is uncharted territory for TL Audio. So overall, then, the M4 is a qualified success. Fabulous sound, solid construction, nice ergonomics, good input handling facilities, and excellent I/O flexibility. Balanced against that is a disappointing monitoring section and some frustrations in terms of the main outputs and headphone cue-mix provision. If the M3 doesn't offer you enough input channels and the VTC is too expensive, the M4 may be the ideal solution, and it is certainly the antidote for anyone who feels that digital recording still sounds too sterile and pristine. Published in SOS December 2005

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TL Audio M4

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

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

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Tube-tech MEC1A & MMC1A

In this article:

MEC1A Channel MMC1A Compressor On The Session Tube Masters

Tube-tech MEC1A £2890

Tube-tech MEC1A & MMC1A Valve Recording Channel & Multi-band Compressor Published in SOS December 2005 Print article : Close window

Reviews : Recording Channel

pros Fantastically musical sound. Unfussy operation. Beautifully engineered.

cons More metering would be useful in places. Shared mic/line input invites phantom-power mishaps.

These two new valve units offer unusually powerful processing for professional studio use. Paul White

summary

The Danish company Tube-tech recently celebrated their 20th birthday, making them just slightly older than Sound On Sound, and the reason they're still around is that they build exceptionally good kit, albeit at a Tube-tech MMC1A £2890 premium price. If you want tube Photos: Mark Ewing preamps and processors that deliver pros the goods sonically, then Tube-tech's range has always been a good place to Sounds fantastic. start looking. Built like a tank, but a very This a piece of top-tier audio equipment with a price to match, but in an environment where you're doing serious vocal recording on a regular basis, it is definitely worth it.

good-looking tank! Easy to operate.

cons Could use more metering, specifically in the preamp section and on the output. Shared mic/line input invites phantom-power mishaps.

The two units under review here are, like all Tube-tech processors, impeccably engineered, with retro Bakelite-style knobs, metal switches, and red power lamps that could double as aircraft warning lights! Tubes are used throughout the audio path — no hybrid circuitry here — and mains is fed in both cases through conventional IEC inlets with adjacent fuse holders and 115V/230V voltage selectors.

summary I wouldn't normally think of using multi-band compression on vocals, but after trying this box I'm sold on the idea. It is a lot of money, but within its league I think Tube-tech have a winner on their hands.

information MEC1A, £2889.93; MMC1A, £2889.93. Prices

MEC1A Channel The MEC1A recording channel is a combined mic, line, and instrument preamplifier combined with an equaliser and an optical compressor. To the left of the 2U processor's front panel is the preamp section. The mic signal enters via a screened step-up transformer, giving it a 10dB boost before it hits the two dualtube preamps that provide the rest of the gain. A pair of rotary gain switches with 10dB and 1dB steps respectively provide a total gain of 20-70dB. Switchable

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Tube-tech MEC1A & MMC1A

include VAT. Systems Workshop +44 (0)1691 658550. +44 (0)1691 658549. Click here to email www.systems workshop.com www.tube-tech.com

phantom power is included, along with a 20dB pad and a phase-reverse switch. The low-cut filter can be set at 20Hz (12dB/octave) or 40Hz (6dB/octave). The microphone input can handle levels of up to +6dBu without the pad, so with the pad switched in line-level signals up to +26dBu can be accommodated. The high-impedance DI input jack is unbalanced and is located on the front panel for easy access. It feeds into the preamp circuitry directly after the input transformer, and plugging into this socket disables the mic input. The gain range for this input is 10-60dB. Both the mic and line inputs share the same XLR input on the rear panel, where you'll also find the balanced XLR output. There are no insert points or direct outputs from the preamp. The equaliser comprises low and high shelving-filter sections and a switchedfrequency mid-band, which is controlled via a rotary 20dB gain control and a cut/boost switch. There are 12 mid-band frequencies between 40Hz and 10kHz, and an uncalibrated bandwidth control. Both the shelving and peaking filters employ a dual-tube operational amplifier topography, and both the low and high shelving sections The audio circuitry of the MEC1A uses three ECC81 and three ECC83 valves, as well as have six switchable frequencies (20160Hz and 4-26kHz) with a ±15dB gain a single ECC82. range, again achieved using a cut/ boost switch. The equaliser can be bypassed when not in use. Following the equaliser is an optical compressor, though a switch in the master section can place the compressor before or after the equaliser in the signal path. The compressor may also be bypassed when not in use. A toggle switch selects between manual or fixed attack and release times (in fixed mode the attack time is 1ms and the release time is 50ms). An Output Gain control offers up to 10dB of gain make-up, while a moving-coil VU meter can be switched to read the output level or the compressor gain reduction. Two TRS jack sockets on the rear panel allow you to link the compressor sidechains of multiple units for ganged operation. By using the front-panel Link 1/Link 2 switches, the user can decide which of the daisy-chained compressors should be linked, in two possible groups, without having to do any re-patching of the TRS sockets. The attack, release, threshold, and ratio are set by whichever unit is working hardest, so the threshold should normally be set fully anticlockwise on the slave compressors. The ratio and gain settings on the slaves should be made the same as those on the master unit for proper operation, but the attack and release controls on the slaves have no effect in this mode. However, as I only had one unit for review, I couldn't test all this myself.

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Tube-tech MEC1A & MMC1A

MMC1A Compressor The 2U MMC1A is a bit of an odd product, in that it combines a mic preamp with a multi-band compressor. Although multi-band compression is usually associated with mastering, apparently engineers have found that Tube-tech's previous multiband compressors sounded great on vocals, so they talked the manufacturer into building the MMC1A. The mic preamp again utilises a screened input transformer to provide the first 10dB of gain, and this is followed by two dual-tube preamplifiers. The gain is set using the same pair of controls, for an maximum gain boost of 70dB, and the mic and line inputs are both on balanced XLRs on the rear panel. Output is via another balanced XLR. The microphone stage has a switchable 20dB pad, 48V phantom power, phase reversal, and 20Hz or 40Hz high-pass filtering. Unlike on the MEC1A, the mic input impedance of this unit can also be switched between 600(omega), 1200 (omega), and 2400(omega) to suit the mic being connected. A high-impedance, unbalanced DI input (0-60dB gain range) is again available on the front panel. After the preamp, two crossover networks split the signal into three frequency bands for feeding to the compressor. The filters are each based on simple circuitry to preserve accurate summation of the three bands at the output after compression. There's no insert point between the preamp and compressor, which some people might have found useful. The lower crossover frequency is continuously variable from 60Hz to 300Hz, but also has a x4 multiplier switch taking it to a range of 240-1200Hz. The upper crossover frequency is variable from 1.2kHz to 6kHz. In all three bands, the compression ratio can be switched between 2:1, 5:1, and 10:1. Each of the bands is controlled from its own side-chain, and after compression the three signals are fed to separate gain controls before being recombined and fed to the master Output Gain control. The compressors utilise an optical gainmanagement system and have rotary controls for ratio, threshold, attack, and release. Bar-graph gain-reduction meters are fitted to each band showing the gain reduction from zero to -20dB in 11 steps, but there's no overall output-level meter. There's also no level meter for the mic preamp, so I guess the thinking behind this is that you watch the meters on whatever the MMC1A is feeding and adjust the levels accordingly. Even so, I would expect a processor of this quality and sophistication to have all the necessary metering, as that's really the only way to optimise the gain structure properly. A toggle switch allows the compressor to be bypassed.

On The Session Although the circuitry uses tubes, which are traditionally noisier than solid-state

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Tube-tech MEC1A & MMC1A

components, the use of an input transformer has kept the noise down to a level where it is insignificant for most practical purposes, especially in the studio, where few things are miked from any great distance. The mic preamp sound is solid, confident, and open without seeming coloured, but one practical point that worries me slightly is that both the mic and line inputs share the same socket, so it would be all too easy to switch on the phantom power with a line-level source connected. I can imagine some pieces of gear not taking kindly to phantom power being stuck up their output! I loved the sound of the MEC1A's equaliser, especially when using the mid-band for cutting in the 1kHz1.5kHz region to smooth out vocal harshness. However, having to switch between cut and boost feels unnatural until you get used to it. I can't argue All rear-panel audio connections on both with the sound though — it does units are via balanced XLRs, although the exactly what a good equaliser should, MEC1A (above) also includes TRS jack and adds polish, punch, and clarity to a sockets which allow the linking of multiple sound without you having to add very units for stereo and surround work. much at all. The same is true of the compressor, because although it does flatter the sound in some ways, it also manages to sound very unobtrusive, keeping tight control over the vocal level in a very effective way that leaves you wondering whether it was actually compressed or not. You can coax more attitude out of it by hitting it hard, of course, but it is surprising how much gain reduction you can apply before you give the game away. When it came to the MMC1A, I wasn't entirely sure what to expect from a multiband compressor designed for vocals, but it turned out to be a useful problemsolver as well as a creative tool for setting up larger-than-life vocal sounds. For example, if you have a singer whose vocals tend to get very honky and nasal when they sing loud, you can use the crossover filters to bracket the area between 1kHz and 2kHz, then apply more compression to the mid-band by increasing the threshold. The louder the singing, the more the mid-band gets squashed, so those honky frequencies are suppressed. Need a more intimate vocal sound? No problem — set the upper crossover to 4kHz or above, then apply more compression to the top band and balance the levels accordingly. During quieter passages, the top end will open up and become more breathy, but when the level increases the compressor steps in to keep it under control. Like most multi-band compressors, the MMC1A works like a combined compressor and equaliser, because balancing the contributions of the three bands is effectively EQ'ing the signal. With just a little practice, you can enhance just about any vocal in a musical and creative way without the processing reducing the clarity or focus of the sound at all. Light voices can be given more low end density by adding extra compression and gain to the low end, while the magical airy quality associated with top-division studio recordings is easy to achieve. In all this is a lovely box, and the preamp is just as solid and transparent

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Tube-tech MEC1A & MMC1A

as that of the MEC1A. My only real concern is over the miserly metering.

Tube Masters I love the effortless way the MEC1A enables you to polish a sound without spending ages readjusting the controls. If I dealt with top-division vocalists on a regular commercial basis, then the MEC1A is one piece of kit that I'd very much like to have on my side. The MMC1A impressed me even more with its extraordinary ability to coax different sounds out of the same mic. It was like having a cabinet full of different mics to choose from! This unit can turn a good vocal recording into a glorious one, and it's interesting that, although you do need to record your vocals in a suitably treated room (the compression will just exaggerate any boxiness), you don't have to be using a world-class microphone to hear the difference. But let's not beat around the bush — both these units cost a packet. Here in the UK you could buy a fast computer, an audio interface, a capacitor mic, a passable preamp, and some audio software for the same money, and still have enough change for fish and chips on the way home from the shop! Nevertheless, if you're in a professional situation where recording impeccable-sounding vocal or instrument parts justifies the outlay, then there's little not to like about these units. In particular, the MMC1A stands out from all the esoteric gear I've tried over the past few months as being something a little bit special, producing more creative results than the usual raft of signal processors we use on a daily basis. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Q. Is it possible to record in surround on only two tracks?

Q. Is it possible to record in surround on only two tracks? Published in SOS December 2005 Print article : Close window

Sound Advice

Would it be possible to use two figure-of-eight mics to create a surround sound recording on a two-track recorder, which could be decoded and mixed later on? The two mics would be placed at 90 degrees capturing left and right and front and back respectively. I'm after a quick and portable way to make surround recordings in the field, and portable recorders with more than two inputs are expensive! As far as I can see this could work, unless there's something I'm missing about figure-ofeight decoding? SOS Forum Post Technical Editor Hugh Robjohns replies: There is something you are missing — a third mic to resolve the ambiguity inherent in a figure-of-eight mic over which side of the diaphgram the sound strikes. You cannot derive front-to-back directional information with just two crossed figure-of-eight mics. Imagine I'm in a studio and you are in a control room, without sight lines between the two. There is a single figure-of-eight mic in the studio and I'm talking into it. How could you tell whether I was talking into the front or back of the mic just by listening? The answer is that you couldn't — a figure-of-eight mic provides signals of identical level from front and rear sources. Yes, the polarity of the signal is inverted between front and rear sources, but without a reference to know which polarity was which, you are no better off. If you think about it, when you combine the two coincident figure-ofeight mics mounted at 90 degrees, you effectively end up with another 'virtual' figure-of-eight pointing midway between the two original mics. You can use that to provide left-right discrimination — which allows the arrangement to be used for stereo — but you will get the same level of output from a source behind and to the right of the array as you would from one to the left front. That doesn't matter in stereo: in fact it is often very useful that rearward sounds are folded back onto the front. However, it is obviously a problem in surround because we need to be able to have rearward sounds coming out of the rear speakers! file:///F|/SoS/SoS%2012-2005/qa1205_4.htm (1 of 3)11/23/2005 3:03:02 PM

This simplified polar pattern diagram shows how a figure-of-eight mic (red) and a coincident omnidirectional mic (blue) can be combined to produce a directional cardioid pickup pattern (green). Flipping the polarity of the omni reverses the direction of the cardioid pickup.

Q. Is it possible to record in surround on only two tracks?

So, what we need to be able to do is create virtual polar patterns that have front-back discrimination, and that basically means creating 'virtual' cardioid patterns. These can be derived by combining an omnidirectional mic's polar response with a figure-of-eight (see my reply to the question on capsule design on page 20 for a little more on this subject). So, if you have two coincident figure-of-eights, facing left/right and front/back, and you add to that a coincident omnidirectional mic, you can combine them in various ways to create virtual cardioid patterns pointing in almost any direction you like. Let me explain how. Consider what happens if you mix together the output of an omnidirectional mic and figureof-eight, the two capsules being coincident. The omni mic picks up sound from all directions equally. The figureof-eight picks up sound only from the front and back, rejecting sound sources to the sides, and its rearward pickup is in the opposite polarity to the front. If we arrange the polarity of the omni to be the same as the front of the figure-of-eight mic, then when you mix the two mic outputs together their contributions will add together for frontal sound sources. So the resulting 'virtual mic' will be very sensitive to frontal sound sources. Sources to the side are not picked up at all by the figure-of-eight mic, but the omni still hears them. So the 'virtual mic' created by the combination of the two is not as sensitive to sounds from the sides as it was from the front, but it does still hear them. Sources to the rear are heard by both the figure-of-eight and omni, but the figure-ofeight's output is of the opposite polarity to the omni. Hence, when the two outputs are combined they will cancel each other out. Thus the virtual mic hears nothing at all from the rear, and if you draw this polar pattern out accurately you'll discover that we have just created a cardioid microphone (facing forwards). You can also create other first-order polar patterns (sub-cardioid, hypercardioid and so on) by varying the ratio of the omni and figure-ofeight contibutions (in other words changing the gain of each). The four coincident capsules of a B-format Soundfield mic, and a diagram showing the intersecting pickup patterns of the three figure-of-eight capsules (X, Y and Z) and one omni-directional capsule (W), courtesy of Soundfield.

So you see that by introducing an omni mic into the array, you can resolve the inherent front/rear ambiguity of the figure-of-eight pickup patterns, by converting them into cardioid patterns, and by adjusting the ratios and polarities of the signals from the two figure-of-eights, you can make those cardioids face in pretty much any direction you like. You can now generate any number of virtual microphone outputs which 'hear' only the sources in front of them — front left, centre front, front right, rear left and rear right, for example. This is the basis of horizontal Ambisonic encoding — the 'B-format' — as used by the Soundfield mic. The file:///F|/SoS/SoS%2012-2005/qa1205_4.htm (2 of 3)11/23/2005 3:03:02 PM

Q. Is it possible to record in surround on only two tracks?

omni component is called W, the front/back figure-eight is called X and the left-right figure-eight is called Y. The Soundfield mic also adds a third figure-eight element for the up/down axis (called Z), although this isn't really needed in most surround applications. You can read more about the Ambisonic approach in the October 2001 edition of SOS: www.soundonsound.com/sos/oct01/articles/surroundsound3.asp. Another (arguably more practical) way of recording horizontal surround sound is the MSM format. This uses the same basic concepts, but is constructed from a pair of matched cardioids facing to the front and rear, plus a sideways-facing figure-of-eight. Again, all three should be coincident. The front cardioid and the sideways figure-of-eight are decoded as a conventional M&S pair for the frontal sound stage, while the rear cardioid and the same figure-of-eight are decoded as another M&S pair for the rear channels. However, whether you adopt the crossed figure-of-eights plus omni approach (WXY format), or the front/rear cardioids plus figure-of-eight approach (MSM format) you do need to be able to record at least three channels, and there is no getting away from that! If you really want to encode surround onto a two-track machine, you have to use some form of phase/ amplitude matrix system like Dolby Pro Logic or RSP Circle Surround. However, while these formats are acceptable for final mixes delivered to the end user, they are far too restrictive for source recordings because you can't easily manipulate the signals to alter the spatial surround characteristics later on, as you can with either of the three-channel surround recording techniques described above. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Q. Is there something wrong with my vintage spring reverb?

Q. Is there something wrong with my vintage spring reverb? Published in SOS December 2005 Print article : Close window

Sound Advice

I have just bought an old spring reverb unit called the Great British Spring off eBay. It sounds great but if I try and put any drums through it, or a percussive synth sound, it makes a weird 'ping' sound. I've had a look around the Internet and can't find much if any info on the thing. Can you help? Rob Pope SOS contributor Steve Howell replies: The Great British Spring was very popular in the '80s — I had one myself. One of the first affordable, decent-quality spring reverbs, it arrived at a time when Fostex were bringing fairly serious eight-track reel-to-reels to the market — it was a marriage made in heaven for the emerging home studio market. That said, the GBS was of serious enough quality to have been adopted in 'proper' studios as a cost-effective way to add extra reverb channels to supplement the main plate reverb. Spring reverbs work by feeding the input signal, typically from an effects or aux send, to a transducer that 'excites' one or more of the springs. The signal travels down the spring and is picked up by another transducer at the other end, then sent to the output and on to the effects return. But it's not quite as simple as that, as the signal also 'bounces' back along the spring, colliding with other signals on their way down and causing complex pseudo-reflections. We perceive this as a reverb effect, and the more springs a unit has, the more diffuse the reverb effect is. The length of the spring dictates the reverb length and density — the GBS's springs are quite long and give a nice hall reverb effect. However, as with all spring reverbs, percussive attack transients can cause the springs to become temporarily unstable, generating all sorts of unpleasant audio artifacts, as you've found out. The simplest solution is just to reduce the level of the signal going to the GBS. This will prevent the springs from getting over-stimulated and thus will eliminate (or at least reduce) the 'ping' effect. The down side to this is that to have the same level of reverb on the sound, you will have to increase the reverb return level which will, of course, increase the amount of noise — these electro-mechanical devices are not known for their noisefree operation! However, even that can be overcome. You see, the frequency range of the springs is limited so, by bringing the reverb returns back through channels that have EQ, you can roll off the top end to reduce the hiss coming from the unit without adversely affecting the reverb sound too drastically, if at all. In fact, given the simplicity of the GBS (and spring returns in general), using EQ can add a lot or creative as well as correctional possibilities.

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Q. Is there something wrong with my vintage spring reverb?

A more elaborate solution is to run the effects/aux send that is feeding the GBS via a limiter set pretty hard, so that the signal never reaches the level that will cause the springs to become unstable. Many more expensive spring reverbs had just such a facility built in. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Q. What factors affect the quality of a microphone capsule?

Q. What factors affect the quality of a microphone capsule? Published in SOS December 2005 Print article : Close window

Sound Advice

I am curious to know more about the design and construction of capacitor mic capsules. For example, what is it about the capsule or the way it is mounted that dictates the polar pattern of the mic? If the capsule is sturdy and made from good-quality parts, what other factors come into play which affect its sound quality? Do capsule designs really differ that much from mic to mic? Paul Curtis Technical Editor Hugh Robjohns replies: A capacitor mic capsule is an extremely complex thing, and the very best are expensive and time-consuming to make. There are two basic types of capsule, working according to two different principles — pressureoperated and velocity-operated (also known as pressure-gradient). The former is constructed a bit like a snare drum — the capsule is, in essence, a sealed box with a diaphragm stretched across one side. The diaphragm acts like a pressure sensor, comparing the pressure changes caused by passing sound waves with the static internal pressure inside the box. The result is an omni-directional polar response —the direction of the sound waves don't matter, the diaphragm is only sensitive to the fact that they pass by. The other way of doing things is to suspend the diaphragm in free space so that sound waves can get to both sides. In this case, the diaphragm moves (hence 'velocity') as a result of the pressure difference (pressure gradient) between the two sides. This arrangement gives a figure-of-eight response — the capsule is sensitive to sounds from front and back, but insensitive to sounds from the sides.

Above: As Paul White discovered when he visited the Rode Microphones factory (see SOS August 2005), the utmost precision is required for drilling holes in a cardioid mic's backplate. Left: Besides the capsule itself, the design and construction of the mic body and internal electronics also shape the sound of the mic.

Often, it is more useful to have a mic that is sensitive to frontal sounds but rejects rearward ones — the familiar

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Q. What factors affect the quality of a microphone capsule?

cardioid polar pattern. A cardioid pickup pattern is produced by combining equal proportions of pressure operation and pressure-gradient operation, and the earliest cardioid mics actually did have both an omni and figure-of-eight capsule side by side in the same box, with their outputs summed together before reaching the output terminals. These days, most cardioids are 'phase shift' or 'labyrinth' designs which are constructed with a single diaphragm, like a pressure-operated mic (the snare drum), but with special convoluted passageways in the rear plate which allow sound to find its way through to the inside of the diaphragm after a time delay. The way this works is rather less obvious than the two prime capsule designs, and would take more space to explain than I have available here, but you can learn more about the subject by reading this article from September 2000 on the Sound On Sound web site at www.soundonsound.com/sos/sep00/articles/direction.htm. In terms of construction, there are literally dozens of different parameters to consider. There's the material the diaphragm is made from and its shape, thickness and tension, there's the spacing between the diaphragm and the back plate, the damping arrangement, the isolation dielectrics, the polarising voltage and so on and so on. In the case of a cardioid capsule, there is also the complex arrangement of the rear chamber labyrinth to consider, and how that affects the polar pattern and the linearity of the capsule's off-axis frequency response. Entire books have been written on this subject alone! Then, once the capsule has been designed and built, it has to be mounted in a mic body, the size and shape of which (along with the grille) affects the response of the capsule. And then there is the impedance converter circuitry, the powering circuitry and the output circuitry to consider, all of which affect the sound of the mic further. This is why it is relatively easy for manufacturers in the Far East to reverse-engineer established mics and build copies very cheaply. But it is extremely hard for them to design new models from the ground up because the real science involved is known by a relatively small group of people. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Q. What makes some interfaces more expensive than others?

Q. What makes some interfaces more expensive than others? Published in SOS December 2005 Print article : Close window

Sound Advice

When it comes to computer audio interfaces, what is it that we are really paying for and how does the price relate to the quality of the A-D/D-A converters? Devices like the MOTU Traveler and the RME Fireface 800 cost more than, for example, the Focusrite Saffire or Digidesign M Box 2, so what does the extra money get you? When I look at the A-D/D-A specifications (sample rate, dynamic range and so on) of interfaces which differ quite a lot in price, they often seem very similar. So do more expensive units sound better? SOS Forum Post PC music specialist Martin Walker replies: When it comes to audio quality, there's a lot more to computer audio interfaces than the choice of A-D/D-A converters — having a low-jitter clock is vital if the sound is to remain 'focused', and the design of the analogue support circuitry (the input preamps and output stages) also modifies the final sound to a lesser extent, including the choice of op-amps, some of the capacitors, the power-regulator design... the list goes on! Many manufacturers start the design of a new audio interface by establishing a rough feature list along with a likely price point, and then the engineers have a complex juggling act to perform to meet this brief. Entering the equation are the quality and price of the converters, the quality of the analogue circuitry (particularly the mic preamps, if there are any), the quality of digital circuitry, plus the controls, connectors, casework and so on. However, when it comes to the converters, many companies tend to choose exactly the same components from one of a handful of manufacturers like AKM Semiconductor, Cirrus Logic and Burr Brown. The converters may only end up contributing a tiny part of the overall build cost, but their specifications often become an important part of the marketing process, particularly when new features like 192kHz support are available (though in the real world I still regard this as a red herring for most recording musicians). Some audio interface manufacturers also quote specifications for the converter chip alone, which can be misleading, since once all the support circuitry is added this inevitably compromises overall performance to some extent. Others quote real-world performance for the entire interface, which is far more helpful. With many audio interfaces you are predominantly paying for the array of features on offer, so an eight-in/eightfile:///F|/SoS/SoS%2012-2005/qa1205_2.htm (1 of 2)11/23/2005 3:03:20 PM

Q. What makes some interfaces more expensive than others?

out interface will cost a lot more than a stereo one simply because there's nearly four times as much circuitry, socketry and controls. You will also pay more for additional features such as mic preamps, built-in limiting, word clock I/O and so on, which is why I always stress the importance of choosing the interface that best suits your The Focusrite Saffire and the MOTU Traveler are both 24-bit/192kHz Firewire needs. A £1000 interface with loads of features may not benefit you interfaces, so why does one cost twice as if you really only need one with basic stereo in/out capability that much as the other? could give you similar audio quality for half the price or less. On the other hand, if two interfaces with similar features and I/O are at wildly different prices, the more expensive one is almost bound to offer better audio quality, although whether or not you'll really benefit from it depends to some extent on the rest of your gear. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Q. Why do my mixes clip when I apply a high-pass filter?

Q. Why do my mixes clip when I apply a high-pass filter? Published in SOS December 2005 Print article : Close window

Sound Advice

There's something happening when I master my mixes that I can't find an explanation for. This has been bothering me for nearly two years, so I thought it was time to call the experts! After maximising my mix, so that the level of the audio is just below the point of clipping, if I insert a highpass filter at, say, 40Hz, suddenly the audio starts to clip. I thought that after inserting a high-pass filter the level should drop, but this is not the case. I've tried it with several EQs but always with the same result. What is happening? I've been using digital EQ, but will this still happen with analogue EQ? Johan De Visser Features Editor Sam Inglis replies: It's surprising, but true, that using EQ can cause clipping whether you are cutting or boosting. There are two reasons for this. One is that EQ changes the phase relationships between the different frequencies that make up a complex signal such as a full Here we see the characteristic 'hump' just above the turnover frequency of a shelving mix. The result of this is that even though you're cutting the low frequencies, you could be shifting other frequencies around in such a filter. way that they reinforce one another at a point where they had not previously done so. It's also possible that whatever signal element you removed was actually serving to 'cancel out' another element at some points, so removing it has created larger peaks at these points. The other reason is that some EQ designs actually have a resonant peak at the corner frequency. If this is the case then applying a low-pass filter at, say, 40Hz might actually introduce more energy than it removes, assuming there was nothing below 40Hz in the signal to start with. It's possible to experience gain increases when cutting with both analogue and digital EQ — it's in the nature of equalisation that this will happen. The answer is to always do loudness maximising as the last process in the mastering chain (apart from dithering) — always use EQ before the maximiser, not after. Published in SOS December 2005

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Q. Why do my mixes clip when I apply a high-pass filter?

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

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

file:///F|/SoS/SoS%2012-2005/qa1205_3.htm (2 of 2)11/23/2005 3:03:24 PM

AER Acousticube 3

In this article:

Small is beautiful Editing effects via USB Preamp facilities Jargon explained Even more connectivity Remote control Acousticube in action

AER Acousticube 3 Acoustic Instrument Amplifier Published in SOS December 2005 Print article : Close window

Live Sound

AER Acousticube 3 Pros Excellent sound quality. Extremely compact.

AER's Acousticube has been at the pinnacle of acoustic instrument amps since 1992. Does the latest revision live up to the legacy?

Rugged and well built. Comprehensive connectivity. Now with effects editing. Reviewed by Dave Lockwood

Cons

Photographs by Mike Cameron

Poor documentation for a premium product. No Mac version of the editor yet.

Specialised amplifiers for acoustic instruments seem to fall broadly into two factions: those that are designed Summary primarily to mitigate the known defects Quite rightly, the Acousticube of the most common pickup types, by is the benchmark product in imposing a character of their own, and its field. Quite rightly, the those that assume you're using a high– Acousticube is the benchmark quality pickup system that you would product in its field. Quite rightly, the Acousticube is the like to hear amplified as faithfully as benchmark product in its field. possible. The line between products in the latter group and PA systems really Information is a fine one, with perhaps only the £1495 including VAT. need for a wider range of input Westside Distribution matching in the instrument amp +44 (0)141 248 4812. differentiating the two. German company AER (Audio Electric Research) Click here to email definitely make amps that fall into the 'virtual PA' category, with their acclaimed www.aer-amps.de flagship Acousticube series having carved out an enviable reputation for itself as the benchmark product in its field for several years now. As an owner and very happy long–term user of an Acousticube IIa, I was pleased to be able to check out the latest Acousticube 3 model, complete with a new speaker design, new preamp configuration and a USB connection allowing reprogramming and organisation of the on–board 32–bit digital effects processor.

Small is beautiful

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AER Acousticube 3

Measuring just 13 inches square by 10.4 inches deep, the Acousticube is amongst the smallest acoustic instrument amps on the market, but this certainly doesn't seem to compromise its performance in its intended applications. The cabinet houses a single Hexacone–based eight–inch driver with Neodymium magnet and a concentric one–inch dome tweeter. Hexacone is a composite material based on Kevlar, which allows the cone to be both stiffer and lighter than conventional paper or polymer cones, giving less cone break–up and an overall faster response, for a tight low–end and smoother mid–range. Only a small port on the rear panel identifies this as a bass–reflex design. The 120W RMS power amp resides at the rear of the box, with the preamp sub– chassis slotted in the top. A ribbon cable joins the two, lying exposed on the surface of the back panel for a couple of inches, creating what seems like an unnecessary vulnerability in an otherwise pretty bullet–proof design. At a little over 28 pounds, the Acousticube is an easy one–handed carry via the side– mounted, recessed handle, or the twin webbing handles of the impressively padded gig bag provided with the amp. The open–cell acoustic foam that covers the speaker actually conceals a steel mesh grille underneath, so there should be no danger of accidental damage in that area. Where previous Acousticube models have offered, effectively, a dedicated instrument channel and a microphone channel, the preamp is now configured with two apparently identical channels, each with a three–pole quarter–inch Both input channels have the same range of jack input on the front and all input– sensitivities and Colour switch, although only selection options available to both. channel two can access the rear panel XLR. A balanced XLR mic input is still The newly-added Pre-Master control governs available, but it is now on the back the line-level outputs without affecting the panel, and although both channels feed to the power amp. Confusingly, the tiny legending that appears beneath the LED have a Mic input setting, only Channel chains in fact pertains to the switches below. two can in fact access the XLR. Channel one's Mic setting seems to switch its quarter–inch jack to mic sensitivity. The additional flexibility that the two almost identical channels confers will, I'm sure, be welcomed by most users, particularly for working with dual–source pickup systems. The jack inputs can each be cycled around four modes, optimising the connection for passive piezo pickups, which require a very high input impedance; line input; mic input; and 'E/P' — an electret mic and piezo combination, with pickup on the tip and mic on the ring circuit. Phantom power at 9V can be activated on the line input and mic/pickup combo, and at 48V on the mic input, by holding in the Mode selector switch for three seconds. The huge disparity in sensitivity between some of the modes makes it all too possible to give yourself a nasty surprise when you inadvertently cycle back from the last one to the first — the most sensitive of all — unless, that is, you actually follow the advice in the manual about making all connections and settings with the master volume control well down!

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AER Acousticube 3

Preamp mode and phantom status can be seen at all times, thanks to a row of multi–coloured LEDs, and a 15dB pad can be inserted to optimise gain range with exceptionally hot signals, although I didn't find anything at all that required it amongst my extensive arsenal of pickup systems, and I rarely provoked the Clip LED into action once the gain setting was optimised. There's a lot of headroom on these inputs.

Editing effects via USB The one area of the earlier Acousticubes that invariably came in for criticism was the lack of any facility to edit the effects programs. Even if you were quite happy with the programs themselves, you would usually want different mix settings for, say, a reverb, at maybe 15 percent effect, and perhaps a chorus effect where you would probably need a 50/50 balance. Even this most basic requirement was beyond the scope of the Acousticube's single Return level control, often rather undermining the benefit of having two memory locations at all. At the back of the new Acousticube 3, however, lurks a USB connector that finally lets us get in there and tweak. A software editor program is supplied on CD ROM (PC-only at present, although a Mac version will allegedly be coming soon), complete with a conveniently lengthy USB cable. The editor's graphical front-end lets you design your own effects program by selecting and arranging modules within a block diagram and then send the preset to the Acousticube. Doubleclicking a module opens a detailed view, where you can tweak the available parameters. It is advisable to save the preset locally before sending, so you can perform further edits, as you can't get a program from the Acousticube into the editor. The software editor includes the complete factory set, though, so you've got a starting point for every existing preset. Once you've sent a preset to the editor, the parameters for that preset are all live and can be tweaked in real time, with immediate audible result. It's not the most elegant software editor I've ever used; parts of it are still in German, even when you select English as the language, and its use of separate Mixer objects, rather than having a wet/dry mix parameter within the effects blocks themselves, does little for the visual clarity of the interface. But this is only version 1.0 and I'm just pleased to have it at all, especially once I discovered that it would run perfectly happily on a Mac under Virtual PC! It seems to be possible to randomly lose the connection to the amp, whilst the software still shows it as connected — you only find out when it tells you it can't send a preset — but restarting the application and activating the 'Connect to amp' routine got things back on track every time, so it's not too much of a headache in practice. I suspect that many users will remain content to never delve into the inner workings of their effects processor, but, like so many of the Acousticube's extensive facilities, it's nice to know that it's there for when you need it.

Preamp facilities A Mute switch is a welcome new addition to the preamp section, allowing silent file:///F|/SoS/SoS%2012-2005/live_aeracousticube.htm (3 of 9)11/23/2005 3:04:04 PM

AER Acousticube 3

instrument changes on stage, and the useful 'Colour' voicing switch of the earlier models now features on both inputs. This offers a preset EQ curve, simultaneously dipping the mid–range and lifting the top end, giving a sparkle and transparency that particularly suits finger–picking and lighter playing styles. I find the Colour switch well–voiced for taking excessive mid–range out of magnetic pickups, or sweetening an aggressive–sounding under–saddle pickup. The preamp stages all seem subjectively a little quieter than the IIa — the IIa is itself a very quiet amplifier— and activating the Colour switch now raises the hiss level barely at all, unless you have a noisy source connected. A three–band EQ completes the channel facilities. Channel one's frequencies are Bass, 60Hz (+/–10dB); Mid, 600Hz (+/–6dB); The effects editing software allows you to and Treble, 13kHz (+/–13dB). This is design your own programs as a block a rather different configuration to most diagram. instrument amplifiers, working mainly at the two extremes of the spectrum, with a low–Q soft slope and limited gain range in the mid band. Channel two's EQ is subtly different, with more conventional turnover frequencies at top and bottom (100Hz and 10kHz) and a much wider mid–range gain (+/–12dB at 1kHz). If your pickup needs radical surgery, neither of these EQs alone will do it. It would be better to get an outboard EQ/preamp with a sweep mid designed for the purpose and use the AER's EQ for what it does best: adjusting the amp's overall tonal response to different rooms, without producing anything too peaky or coloured that might provoke earlier onset of feedback. To further help combat the latter, there is a notch filter — still located at the back, as on the IIa, and still not sweepable. Being an analogue filter, it offers nothing like the pinpoint notching we have become used to with digital anti–feedback processors. A notch that is 24dB down at 120Hz, recovering to –12dB by 36Hz, makes a sizeable hole in the bottom end of most guitars — so much so that I have only ever found myself using it with naturally boomy big–bodied instruments or in particularly adverse monitoring conditions. Even then, I always put back a little bottom end with the Bass control, to avoid over–thinning the sound. One further tonal adjustment is available in the form of the Presence control, again located on the rear panel. The documentation implies that this is for fine– tuning the response of the tweeter, affecting frequencies above 4kHz, but I have never felt the need to set it anywhere other than maximum. Still, it's good to know it's there. The Acousticube 3 has sprouted a new Pre-Master control, alongside the main master volume, governing the Left and Right line–level outputs and dedicated recording outputs independently of the main master volume settings, adding still more flexibility to this range's legendary interconnectivity.

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AER Acousticube 3

The on–board 32–bit digital effects processor has a very simple three–control interface: a continuous rotary encoder for program selection; return level; and pan to assign the effect between the two channels. A comprehensive selection of treatments is offered, ranging from different sizes of dark, bright and soft rooms, halls, churches and cathedrals to ambiences, unusual spaces such as corridors, a swimming pool and a railway station, and assorted choruses, flanges and delays. A lot of subtle variations are offered on the same basic themes within the 100 programs.

Jargon explained Bass reflex: A cabinet design with a small opening, or 'port' that allows some of the sound from the rear of the loudspeaker to escape from the enclosure in phase with the signal from the front of the speaker, thereby enhancing bass response. Cone break-up: An ideal loudspeaker cone acts as a piston, increasing and decreasing air pressure over its entire surface area, but the designer always has to strike a balance between stiffness and a light enough weight to respond at high frequencies. When a cone receives too much level it first bends and then 'breaks up', in that part of it can be moving in one direction, while the rest is moving the other way, producing modes of distortion not harmonically related to the input signal, and an audibly unpleasant result. Neodymium magnet: Very high strength 'rare–earth' magnet type made out of neodymium, iron and boron. Its use in a loudspeaker allows the coil attached to the diaphragm to be made smaller and therefore lighter, thereby increasing efficiency. Phantom power: Standardised scheme of providing an invisible (hence 'phantom') power supply voltage using the same cable as the balanced audio output. RMS (Root Mean Square): A calculation (using the square root of the average of the squares of a group of numbers) for obtaining the effective average voltage or current of an AC signal. Assuming the source is a sine wave, the rms value will be 0.707 times the peak value, or 0.354 times the peak-to-peak value (from full negative to full positive).

Even more connectivity Acousticubes have always been renowned for their comprehensive connectivity, yet the latest version has still managed to add some more, with the addition of a pair of phonos for connecting a CD player or other –10dBV line–level source. A level control allows you to balance this source independently of the main signal paths. Although the Acousticube itself is mono, the aux signal, like the on–board effects, is sent to the left and right line outputs and the phones output in full stereo. A master insert point has also been added (while retaining the Effect 2 send and return loop). This appears on a stereo jack (tip send, ring return), allowing for the file:///F|/SoS/SoS%2012-2005/live_aeracousticube.htm (5 of 9)11/23/2005 3:04:04 PM

AER Acousticube 3

insertion of a permanent master effect such as a compressor, limiter or EQ, without having to tie up the footswitchable Effect 2 circuit. I must admit that this was one of my few frustrations with the IIa configuration and it's great to see it solved in the latest model. Two sets of line–level outputs are provided: Left and Right line outputs for connecting to active extension cabs or a PA system; and two Recording outputs, which allow you to access the separate signals of channels one and two, with their EQ, but without effects and independent of the master level The diminutive size of the Acousticube, setting. The Left/Right line outputs coupled with its comprehensive connectivity include anything patched into the inevitably leads to a fairly busy rear panel. Effect 2 return circuit, but not the The logical layout helps a bit, but once it's internal effects unit, with the front– populated with a few fat Neutrik jacks, some of the legending is obscured. Take a torch panel Pre–Master control determining and a copy of the relevant sheet from the the level independently of the master manual on your first few gigs! monitoring level. There's also a separate mono Line output which both tracks the Master volume and includes the on–board effect signal, for expanding your rig with another amp (AER would probably like you to use their CX8 powered extension). An electronically balanced XLR forms the main DI output for connection to a PA. DI level is independently adjustable (via a rather too small pot shaft on the back), and the signal can also be switched between dry with no EQ, or post–EQ and with the internal effect signal. The Effect 2 circuit allows the connection of an external effect in addition to the on–board processor, with both being controllable from the supplied footswitch. The return is in stereo, which is retained in the line outputs (and headphones), though not, of course, in the main amp, which is mono. The send is switchable between series and parallel operation, allowing you to set the mix locally, using the Return level control, and also keep at least half your signal entirely in the analogue domain if you are not entirely convinced about the integrity of your effects unit's A–D converters. A Pan control determines whether the send to Effect 2 is sourced from channel one, two or both. Staying with the theme of signal integrity for a moment, the presence of a dedicated Tuner output allows you to keep a tuner permanently in circuit without having to have it in the primary signal path. Helpfully, the line–level output is independent of all level controls except the input gain setting. A dedicated line–level sub–woofer output is slightly unusual in this type of amplifier. This one is optimised for connection to AER's active auxiliary bass– box, the Sub12/400A, and automatically engages a high–pass filter to relieve the Acousticube's own speaker of everything below 200Hz. The signal includes tone file:///F|/SoS/SoS%2012-2005/live_aeracousticube.htm (6 of 9)11/23/2005 3:04:04 PM

AER Acousticube 3

controls and effects and follows the master volume. The comprehensive connection facilities are completed by the stereo phones output, which cuts out the internal speaker, the eight–pin DIN master control socket and the newly added USB connection (see 'Editing effects via USB' box).

Remote control An eight-pin DIN cable connects to the supplied Acousticube two-button footswitch, allowing remote selection between the two user-stored effects in the memory one and two slots. Effects are stored by dialling up the one you want on the amp's rotary encoder and simply pressing the pot in towards the chassis. While one footswitch toggles between programs, the other activates or deactivates either the internal effects processor, via a short tap, or the Effect 2 loop, via a longer press. Very clever, and far more intuitive to use than to describe! The footswitch also features a couple of controller inputs giving access to a VCA in each channel. A simple 'shorting' footswitch will act as a channel mute here (although, cleverly, the tuner output stays active), or you can use a volume pedal. There's nothing in the documentation to suggest an optimum pot value, but my empirical messing about settled on 22k Log as the best combination of offering both a progressive taper and ensuring no loss of volume when the pot is fully turned up. The nearest off-the-shelf component is an Ernie Ball 25k 'line' pedal which worked well enough, although I felt that the usable range was too concentrated at the toe-end compared to using the same pedal inline with the signal. Both channels can be controlled together if only channel one's jack is connected, or separately, using two pedals. There's also a neat trick you can try when using an external processor in the Effect 2 circuit: if you return the effect only to channel two (using channel one as your primary signal), the VCA foot volume on channel two then acts as an independent effects level control.

Acousticube in action The Acousticube 3 delivers everything that its predecessors have always offered, and then some, offering smooth, neutral sound that is every bit as at home with acoustic guitar, vocals, or even amplified upright bass. Power delivery seems more effortless than the IIa, for while there's still a speaker protection limiter in the system it now seems virtually impossible to trigger it with normal music signals. The overall response of the new speaker system seems better, too, the subjectively smoother sound being confirmed by a higher initial feedback threshold.

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AER Acousticube 3

In my primary application, acoustic guitar amplification, the Acousticube 3 surpasses my IIa version in areas that I really hadn't expected, such as lower front–end noise and higher headroom, whilst retaining the same intrinsic quality of sound. Specific settings that worked with certain pickups on the IIa still work on the '3'; they just work Double-clicking any of the on-screen effects a bit better. This is a remarkably blocks opens a comprehensive parameters versatile, high–quality amplifier, equally view, allowing the block to be tweaked in detail, but still reasonably intuitively. adept at filling a room on its own in a 'reinforced acoustic' context, or slotting into a larger rig as a performer's monitor feeding an FOH system. Despite the diminutive driver, there's a reasonable amount of volume on tap. I use mine with a small band line–up including drums and electric bass and never need to ask for any guitar in the monitor mixes. The single (actually dual–concentric) driver and narrow cabinet frontal dimension serve to create a very predictable and even spread, both vertically and horizontally, with none of the beaming effects of larger boxes or multi–driver systems. There are practical limits to how much level a single eight–inch driver can generate, however, and I wouldn't expect to be able to keep up with a less subtle drummer, or an electric guitar– based line–up, unless it was just with background strummy filler. The on–board effects, with a few exceptions, are mostly the kind you can just dial–in as a sweetener and leave alone. I know some people criticise them as not radical — not 'effecty' — enough, but I think they have been well chosen as subtle enhancements rather than show–stealers. Of course, now you can 'roll your own' anyway, via the USB link and supplied software. Granted, the Acousticube 3 is by no means cheap — there are perfectly respectable acoustic guitar amps on the market for a third of the cost of one of these — but high–quality components and engineering cost. With the release of the new version, at least there might now be a few IIa models on the second– hand market, as owners look to upgrade! Will I be upgrading? Now that I've heard the new model, yes. Will everyone want to upgrade? Probably not. The IIa is still a great acoustic amp, and unless you're perceiving limitations with it, why spend more money? AER make a whole family of PA/acoustic instrument products, including a number of rather more affordable models such as the Compact 60 amplifier, but manufacturers tend to be judged on the performance of their flagship products, so, as a big fan of AER's 'quality is everything' company ethos, it's nice to be able to report that 'the best' just got even better. Published in SOS December 2005

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AER Acousticube 3

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

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

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AKG D22 & D11

In this article:

D22 spec check Testing, testing Jargon explained The D11 Sound words

AKG D22 & D11 Instrument Microphones Published in SOS December 2005 Print article : Close window

Live Sound

AKG D22 & D11 Pros Inexpensive without being too cheap! Very solidly engineered. Good sound quality within their price range. Included cables, integral swivel mounts and case. The D22 also has a drum rim clip included.

Cons

The new Crystal Clear Sound range of mics includes two models intended for instrument miking. They've got the prestigious AKG name, but they won't break the bank. Reviewed by Paul White Photos by Mike Cameron

No obvious cons at the price, although I would have appreciated more technical information.

Have you ever seen some famous actor in the flesh and been surprised at how much smaller they look than they do on Summary television or in the cinema? It's a bit like While not top-drawer mics, that with the AKG D22 instrument mic. these models give near-pro performance at a near-budget Intended for applications such as tom price. The D22 is a great little miking, percussion, guitar amps and all-rounder, while the D11 is some wind instruments, the D22 is probably best suited to kick a surprisingly compact dynamic drums and bass amps. cardioid microphone clearly designed to Information get in 'under the radar' of flying drum D22 £59.99; D11 sticks! Without the XLR plugged in, the £79.99. Prices include VAT. mic's body is barely longer than its Harman Pro UK +44 (0) basket, and a built-in swivel stand-adaptor further helps maintain a low profile. 1707 668222. Even better, a plastic clip is included that should slot onto the rim of most types Click here to email of tom, enabling the mic to be mounted above a drum without the need for www.harmanprouk.com a separate stand. A five-metre AKG-branded cable also comes as part of the www.akg-acoustics.com package, and everything comes stored in a rigid, translucent plastic case with positive-acting snap fasteners.

D22 spec check The D22's technical specification is as simple as the mic itself, which has no

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switches or pads to complicate things. Its frequency range extends from 60Hz to 18kHz, while sensitivity is 2.5mV/Pa, but no frequency curve is provided in the brief manual and there's no clue as to how many dBs 'down' the mic is at the specified limits. Clearly, this is one microphone that will have to be evaluated almost entirely subjectively. While the D22 is laudably compact, plugging in a standard XLR cable effectively doubles its length, as I mentioned above. While I wouldn't want the designers to use a non–standard connector or a fixed cable, I still can't see why mics of this type can't be designed with their XLR socket at an angle, to keep the cable connector below the line of fire when the mic is fixed to a drum rim and angled down towards the head. Having said that, even though the D22 is an inexpensive mic, it is very nicely engineered, with a heavy cast body and a very tough single– layer basket lined with acoustic foam — so it should safely withstand a few whacks! As you'd expect, the capsule is internally shock–mounted, although when I removed the basket to investigate I found the resilient mounting stiffer than I'd expected. The swivel mechanism has a fairly small thumbscrew, for tightening purposes, but the mic showed no tendency to droop in normal operation.

Testing, testing The D22 comes over as a fairly honest, flat dynamic mic and doesn't even sound out of place handling vocals. It does a workmanlike job of miking electric guitars, drums and hand percussion, with a smooth, even tonal balance that responds well to gentle EQ. In tight situations, its small size is also beneficial, especially around a drum kit. Given that it retails at well below the cost of more commonly used microphones in this market sector, it performs extremely solidly on drums, percussion and guitar amps.

Jargon explained Cardioid: Literally, 'heart-shaped'. When applied to microphone pickup patterns (ie. the patterns in which particular mics pick up sound), the bottom bit of the 'heart' points forward, while the cleft in the middle at the top represents the direction of lowest sensitivity and points to the rear of the mic. Dynamic microphone: Microphone design based on the principle that a wire coil suspended in a magnetic field will generate a voltage when the coil is moved. If that coil (called the voice coil) is attached to the diaphragm of a microphone, a proportional voltage will be created when sound causes the diphragm to move in and out. Dynamic mics are generally rugged, but with limited sensitivity and frequency response, due to the mass of the coil/diaphragm assembly restricting its speed of movement.

The D11 file:///F|/SoS/SoS%2012-2005/live_akgd22d11.htm (2 of 4)11/23/2005 3:04:09 PM

AKG D22 & D11

The D11 is styled in a similar way to the D11 but is significantly larger, with an extended bass end that allows it to be used with kick drums and bass guitars, as well as all the sources that the D22 is designed to handle. Again there's a swivel mount, but this time the XLR connector is built into the plastic moulded swivelling section, so the effective mic length isn't altered by plugging in a cable. A cardioid design, the D11 has a frequency response that extends from 20Hz to 20kHz — although, once more, no frequency plot was included in the documentation. I found this especially frustrating, as kick mics tend to have 'interesting' response plots that are more to do with musicality than with accuracy, and I would have liked to see what the designers have chosen to do. A similar type of construction to the D22 has been employed, although the mic capsule itself is clearly very different and the basket is almost twice the diameter. There's no drum-rim clip with this mic, as it's probably a bit on the heavy side, but it comes with the same cable and plastic storage case as the D22. One way the cost has been kept down in both cases is to manufacture these microphones in Taiwan, although the design is all original AKG and done in Vienna.

Sound words As expected, the slightly more expensive D11 has a very different sound, which might best be described as 'scooped', no doubt because it is tuned to augment the upper–mid thwack and the low thump of a kick drum, while suppressing the somewhat boxy frequencies that lie between. Because of this coloration it works well on kick drums and larger toms, and can also suit certain bass guitar amps, but by the same token it is likely to be less well suited to those tasks that require the mic to deliver a more accurate representation of the instrument. However, at around half the cost of the better established AKG D112, the D11 is very attractive for smaller bands who want to get close to the classic 'weighty but defined' AKG kick sound. When you take into account the price, build quality and performance of these two microphones, there's no getting away from the fact that they do a very solid job. You even get a mic cable, which is a worthwhile bonus. For serious studio work, I'd probably still pay the extra for a D112 for kick miking, but for live work or applications where the budget is tight, these mics deliver better than adequate results and seem rugged enough to keep doing so for many years to come. Published in SOS December 2005

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AKG D22 & D11

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

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

file:///F|/SoS/SoS%2012-2005/live_akgd22d11.htm (4 of 4)11/23/2005 3:04:09 PM

M-Audio Aries

In this article:

First impressions Let's get physical Gig test

M-Audio Aries Pros Comfortable to hold. Robustly built. Easy to clean. Inexpensive.

Cons Poor feedback rejection. Rather unpredictable polar pattern. Proximity effect quite pronounced.

Summary Intended for hand-held use, this is an inexpensive and robust entry-level vocal condenser but is marred by poor feedback rejection.

M-Audio Aries Hand-held Capacitor Microphone Published in SOS December 2005 Print article : Close window

Live Sound

M-Audio are perhaps better known for their studio and computer-based peripherals, but they now also have a microphone range, to which has just been added a stage-specific model. We put the Aries through its paces. Reviewed by John Gale Photographs by Mike Cameron

When the M–Audio name is mentioned, a selection of well– built mid–priced products springs to mind: soundcards, MIDI Information peripherals, studio monitors and the like. I wouldn't normally have considered M–Audio as a first choice for a stage £99 including VAT. M-Audio UK +44 (0)1923 microphone, as there are so many already on the market by 204010. manufacturers that I know and trust, but as the Aries under Click here to email review here is their first mic to focus on live sound, as it's a capacitor and as it has an RRP of just under £100, I was www.maudio.co.uk particularly interested to see how it would fare in a gig environment against some of my old favourites.

First impressions I had mixed feelings when I first received the Aries. It comes in a cardboard box, which is nothing new, but both the box and the manual provide very little information about the characteristics of the microphone. The accompanying blurb states that the Aries is a "cardoid design for stage and studio vocals," but claims such as "20Hz-20kHz frequency response" seem rather generic and make me wish for more data. A closer inspection of the manual yields no further information, and there's no polar plot illustrating the cardoid response, either. For the more anally retentive amongst us, I suppose this could be considered a bad sign.

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M-Audio Aries

However, a few quick tests later, things become a little clearer. Lending itself to close–miking applications, the Aries has a flat frequency response between 40Hz and 2kHz when placed around 5cms from the sound source, and an overall lift of around 3dB across the 2kHz–15kHz range. As is common to many vocal capacitors on the market, I found a noticeable (6dB) boost in the higher frequencies at both 6kHz and 12kHz. At greater miking distances, the frequency response of the Aries naturally rolls off quite steeply below 200Hz, with the proximity effect audibly more prominent than one would expect, perhaps as a result of the high–frequency boost I've just mentioned. (My test measurements were made against a B&K 4007 reference microphone.)

Let's get physical The Aries is supplied in a standard zip–up soft case with a robust microphone clip (although mine doesn't include a thread–adaptor insert for the clip). The microphone looks very similar in design to a Shure Beta 57, although it is slightly larger, at 40 x 168mm. Weighing in at 260g, the Aries feels light, but it's well balanced for hand–held use and appears durable enough to take a fair amount of road punishment. Something I particularly like about this mic is its sturdy metal body, which is solid, does not scratch easily and is moulded to sit comfortably in your hand. Slightly less appealing is the grille that hides the pop shield and which is designed to protect the three–quarter–inch gold–evaporated diaphragm. The grille feels and looks a little inexpensive, although it is clearly strong enough to serve its purpose, and removable for cleaning purposes. Most microphones of this nature boast at least a two–layer 'pop shield' but the Aries only appears to incorporate a single layer of foam, which does leave it susceptible to extreme plosives (such as 'b's and 'p's). While handling noise is reduced with an internal shockmount, the Aries shockmount seems less effective than the ones fitted to some other microphones on the market. The shockmount places the capsule within a rubber ring on the end of three rubber shock– absorbers, rather than suspending the capsule with rubber strips. The Aries' handling noise didn't concern me unduly, but a perceivable metallic 'clunking' occurred when I tapped the end of the XLR, and this was not present with my other microphones. As far as general specifications go, the Aries claims a low self–noise of 17dBA, making it comparable with some of the more expensive contenders on the market. It can accommodate sound pressure levels (SPLs) of 134dB (for 0.5 percent distortion) which is not bad at all, and has plenty of output, requiring noticeably less gain than a Shure SM86, for example.

Gig test

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M-Audio Aries

I put the Aries through several tests and concluded that it sounded pretty useable when I listened to it through the front–of–house system. I found it slightly lacking in warmth, perhaps due to the bright high–end, but a little desk EQ soon brought it into line with my preferred vocal sound, and although I did feel it lacked a little definition, for the price I was impressed. However, when I put some stage monitoring into the equation, a problem became evident. Using monitoring I trust (EM Acoustic M12 wedges and a BSS Graphic EQ), I found that the Aries required more than normal EQ'ing across the monitors in order to prevent feedback at a medium stage–noise level. Problem areas were the 400–500Hz range and the 6–8kHz range, with the microphone quickly on the edge of feedback before I would normally expect this. Additional spill from drums led me to conclude that the cardoid response of the Aries is not as tight or predictable as it should be for a stage microphone, which was a disappointment. When speaking further than an inch away from the microphone, I noticed that the sound quickly became quite thin, and although with a full band playing I could make the microphone cut through in the mix, it was a less than full–bodied sound. I concluded that an inexperienced engineer or a vocalist with poor microphone technique would suffer in a loud environment. Substituting the microphone with a range of others for comparison purposes, I could easily get 6dB or more headroom from the monitors with no additional EQ. I think it would be wrong to write off the Aries completely, because for the money you get a durable capacitor microphone that can give a good sound at front of house. However, for those seeking a vocal microphone for loud stage use, in this price bracket a good old-fashioned dynamic such as a Shure SM or Beta 58 might yield a more appropriate result. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Meet the Sound Guy

In this article:

Fly on the wall Jonny's tips for getting into live sound Band of brothers Sound Czech Using compressors and gates live The great outdoors Stage to studio

Meet the Sound Guy Jonathan Lucas : Freelance Engineer Published in SOS December 2005 Print article : Close window

Live Sound

As well as doing regular stints at Camden's Barfly and other London venues, Jonathan Lucas engineers for a successful gigging band, whom he also records in the studio. We find out how it all came together. Feature by David Greeves Photographs by David Greeves and Karl Nathan

Leafing through the pages of Sound On Sound and Sound On Sound Live you'd be forgiven for thinking that live engineers and studio engineers are two very different species. But outside the very top echelons of the profession, few engineers are so specialised. For one thing, work is work and we all need to make a living, but there's also the fact that some engineers simply don't want to be restricted to either discipline. One such engineer is Jonathan Lucas — Jonny Turbo to his friends — a man who's equally at home behind a live console or behind the glass. Jonny's first definite step towards professional sound engineering took the form of a two-year National Diploma course in Music Technology at Leeds College of Music (LCM). This came after four years working for Tele-Products Ltd in York, an electronics company who manufacture test equipment for the telecoms industry. The job turned out to be advantageous, with hindsight. "Working there prepared me really well for studying music technology, in terms of understanding circuit diagrams and generally having a technological approach. So when I started studying audio equipment in depth, looking at circuit flow and so on, it didn't phase me, as I already had that grounding in electronics". Seeking broader horizons, he relocated to London and began another two-year course, this time at the SAE Institute. "I was originally going to do a full

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Meet the Sound Guy

Recording Arts degree but I cut it short after a year and came out with an Audio Engineering diploma. The bulk of the practical work was in the first year, and a lot of the second year was industry study, which I'd already covered at LCM. It was a very expensive place to study and paying five grand a year to sit in a classroom and have someone tell you about copyright law didn't seem worthwhile. But the first year was an invaluable experience, having the opportunity to use big professional consoles like Neves and SSLs." Other opportunities were already presenting themselves too. "I was getting offers of work from the London Barfly [the well-established Camden venue is a landmark for up-and-coming bands] and I started doing shifts there at night and then going in to study in the morning, which wasn't great. Nobody in the industry cares whether you've got a degree or not, so it seemed topsy-turvy to be turning down work because I was studying, when the aim of the game was to get work at the end of it!"

Fly on the wall Jonny's opening at the venue came, as is often the way, via working initially as a volunteer. "I got involved in a series of charity gigs they were doing, called Passport Back To The Bars. It was a gig a night for a week with the likes of the Cure, the Darkness, Craig David, David Gray — big shows. I was just doing bits of security and stage management, and generally helping out the production managers because they didn't have enough staff, but it was a good way into the setup there. I shadowed one of the engineers for a few shifts to get the feel of what was going on, and as my confidence grew and I learned more and more, I eventually hit a point where they were asking me to do FOH sound, which is absolutely terrifying the first time you do it." So did he learn how to engineer a gig just by watching other people? "As far as the practical side of it goes, I suppose so, but then I had the theory in place to back it up. A lot of what you learn in a studio can cross over into live sound, but the pressure is completely different. In a studio it's all quite laid back, whereas with live sound, soundchecking and so on, everything's got to be done now — Sevenball, the band for whom Jonathan is you're not spending hours getting resident engineer, on stage at the United exactly the right snare sound, you're Islands Festival in Prague this summer. just flying through it. I just bit the bullet and jumped in. Slowly you get stronger and stronger and eventually you feel confident about what you're doing rather than just feeling panic!"

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Meet the Sound Guy

As well as working in-house at London venues such as the Barfly, the Underworld and the Hard Rock Café, Johnny is practically the fifth member of London-based four-piece band Sevenball (www.sevenball.co.uk), engineering for them wherever they play. "I got to know the band through mutual friends and started plugging away at Luke [Ritchie, lead singer] about all the benefits of having your own engineer. In fact, they had previously had a regular engineer before moving to London and were really waiting to find the right person, and I was waiting for them to realise it was me! Eventually the opportunity to do FOH for them came up. That felt kind of like a live audition, but they were happy with how it went and so was I, and feedback from the crowd was really positive, so that was that."

Jonny's tips for getting into live sound Obtain your own gear

"Getting some equipment of your own is important. I think you learn the most just from experimenting, and the more time you can spend using the stuff the better. If that involves hiring equipment or buying studio time it's massively expensive. If you can find a room or garage where you can set up some monitors, a mixer and a graphic EQ, you can start finding out what frequencies do what and try ringing out monitors. You don't need top of the range gear and you can get a lot for your money these days." Volunteer at venues

"If you know any approachable sound engineers, or if you just go and get chatting with someone at a local venue, ask if you can go and shadow them one evening just to get a feel for what's involved. There's really no standard career path — maybe even less so than in the case of studio engineering. A lot of venues use stage managers. It doesn't take a lot of technical know-how to do that, just good organisation and communication, but it's an 'in'. You can lend a hand and get to know the people that run the venue." Find yourself a band

"If you know a band who aren't very technically minded, they'd probably welcome someone to look after that side of things. And having some willing guinea pigs for both live engineering and recording is great experience."

Band of brothers Jonny has a lot to say on the benefits of being a band's permanent live engineer, both for the band and the engineer. "It's a huge boost in terms of gaining experience. If you do FOH in-house in one venue for lots of different bands you'll get to know that venue really well. But if you go to lots of different venues with the one band as a constant, you learn how to get them sounding good in lots of different rooms on lots of different setups. Also, if someone offers you an inhouse gig somewhere, you might have already been there once or twice, so you'll have a better idea of what you're dealing with.

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Meet the Sound Guy

"It really helps to know the songs, too. If you're mixing a band and you don't know the material, and, say, a guitar solo suddenly jumps in, you'll have to look up and think, 'Now, which one of the guitarists is doing that?' That bit of reaction time isn't a factor when you know the songs — you can have your finger on the fader before they hit the first note of the solo. And if one of the band needs to swap guitars between songs, it looks so much more professional if you know it's coming and just take out that channel at the end of the song, rather than them having to wave to the back of the room and make 'is it OK to unplug this?' signals. "Once the gig is under way, knowing the band makes communicating with them much easier, and that's a side of live engineering which is massively underestimated. It's one more thing that helps things run as smoothly as possible and look professional. If you don't know the band and they have a problem on stage or need you to change the monitor mix, there's that a bit of a barrier that's not there if you know the band well. I actually get quite a kick out of it if, in the middle of song, someone in the band can give the subtlest of signals so that no one in the room has probably noticed it but I know they want a little more vocal in their monitor. More than anything, it puts the band at ease. They know that the person they've taken along to do the FOH is going to do the best they possibly can with the equipment and they just have to worry about playing. I think they get a better performance out of it. And knowing roughly what sort of monitor mix the band are going to need also means that the time you spend soundchecking is cut down a lot."

Sound Czech One of the most challenging jobs Jonny has encountered with the band so far was an outdoor gig at the United Islands festival in Prague in June 2005 (www. unitedislands.cz), where the band were booked to play two different stages over two days. Arriving off a delayed flight half an hour before they were due to perform, with no prior knowledge of what to expect, can hardly be described as an ideal situation, but Jonny seems to have taken it in his stride. "It was quite a scary experience for all involved, but most enjoyable! I didn't have a clue what kind of equipment I'd be using — we'd tried to find out, but hit a brick wall with the organisers — and the band didn't have their own amps and kit with them either. As it turned out, there was a fantastic Midas desk at the site and their outboard gear was fantastic — better than most of the venues I've worked in. They had TC Electronics effects, digital graphic EQs and some multi-band compressors, which you rarely find in the smaller file:///F|/SoS/SoS%2012-2005/live_jonathanlucas.htm (4 of 8)11/23/2005 3:04:18 PM

Having arrived at the Prague gig with little time to spare, and with no knowledge of what equipment was on offer, Jonathan was

Meet the Sound Guy

venues in London and are a real luxury.

pleased to find top-class gear, including a Midas console and TC Electronic effects.

"As well as not knowing what to expect in terms of equipment, I was a bit worried that there might be problems with the language barrier. Luckily, the in-house engineer was Irish, although the monitor engineer who was up on stage didn't really speak any English. But the band managed to get by through pointing and using the international sign language for guitar, drums and so on! "Not having to deal with the monitor mix has good and bad sides to it. In this case, it meant that I could just concentrate on the FOH, which was great, but it's also an element of control that's taken away from you, and given the language problems I'd probably rather have done it myself. Having said that, there wasn't that much time to set up. We had no soundcheck at all — just 10 minutes to linecheck everything, then we went for it." Jonny even manages to find an up-side to having little time for pre-gig preparation. "In a normal situation, it would be a case of different bands and engineers soundchecking in turn, each noting down all their settings carefully and then recalling them later on. When it comes to your turn, another engineer will probably have used some, if not all, of your channels. Sometimes you'll have your vocals and guitars left for you, but you'll usually be sharing the bass and drum channels with everyone else, even if the bass amp and drum kit are being changed during the gig. In this kind of situation, I think there's a danger that you'll end up worrying too much about recalling your settings absolutely exactly and get lulled into a false sense of security. Lots of things can change between the soundcheck and the performance so you need to be on your toes. As well as changes to band's equipment, people make a huge difference. People just soak up sound, so if you soundcheck when the venue is empty, you'll find a full venue sounds very different — usually much better."

Using compressors and gates live "Depending on the size of the venue, you'll usually have between four and eight compressors to play with. Obviously, the type and amount of dynamics processing required varies a lot from band to band, but in general I don't use much compression. "I used to always put a compressor over the drum group as a matter of course, but I very rarely compress drums at all any more — I find that it's usually unnecessary in smaller venues. I do tend to gate kick-drums and toms, though. Most drummers don't seem to tune their toms very well, so they tend to ring on for too long. By using a gate, I can control how much or little tom ring comes through. Gating the drums also means that you're only picking up the hits and not everything else in between. "I compress DI'd bass because the output is usually so spiky. If I'm using a compressor on a lead vocal, I'll put it on a group rather than in-line, so that it doesn't affect the on-stage monitors — if the singer starts singing really loud, the compressor will kick in and they won't be able to hear themselves through the file:///F|/SoS/SoS%2012-2005/live_jonathanlucas.htm (5 of 8)11/23/2005 3:04:18 PM

Meet the Sound Guy

monitors. I'll also sometimes use a compressor as a dedicated limiter if, say, someone is using a stomp-box effect that causes a sudden big rise in level. "As far as gear goes, I really like BSS's stereo and quad compressors. They're really transparent and they've got built-in de-essers. I'm not so keen on the Behringer and Samson ones, which you can hear working. Drawmer gates are great. On the MX40 Punch Gate there's a Peak Punch mode that adds a very brief increase in level when the gate is triggered. So it boosts the start of the hit too and really makes the drums kick through the mix."

The great outdoors On top of everything else, the United Islands festival was the first large-scale outdoor gig that Jonny had engineered. "It was the first time I'd used a proper rig outdoors — I'd only done little acoustic acts before. It's very strange. It always feels quite quiet when you're outside, because you don't have the reflections off walls that you get indoors, but that's also a good thing, because you can drive the hell out of a vocal and you're never going to get feedback through the FOH. No reflections means there are no standing waves to worry about either. "Wind caused a little bit of a problem, because the top cabs weren't fixed very securely. They were hanging on chains but weren't tethered at the bottom, and when they moved in the wind you'd get an odd phasing effect. I heard it before I realised what was happening, which caused a bit of a panic until I looked up! Obviously, there was nothing I could do about that."

The Barfly's small upstairs studio, which is put to good use recording bands performing on the venue stage. Some shows are then broadcast on XFM.

The FOH mix position — a raised platform constructed from scaffolding facing the stage — had some peculiarities of its own. "Compared to most venues, where you're tucked away in a corner and have to deal with some weird acoustics, I was expecting that being outside would be much easier. You'd have thought that, being out in the open, you'd hear the same thing as everyone else from the mix position, but the board I was standing on behind the desk was developing a huge amount of bass. I could feel it through my legs and as soon as I stepped off, the bass disappeared. But whether you're inside or outside, you should be stepping out from behind the desk and moving around to see what other people are hearing, rather than just mixing for yourself in the mix position. You can never have a perfect mix everywhere in the venue, but you can aim for a compromise. "In those kind of conditions — when you're standing in a bass-heavy area — mixing feels very unnatural. The kick drum sounds really heavy and the guitars and vocals don't feel like they're cutting through, then as soon as you step out, it file:///F|/SoS/SoS%2012-2005/live_jonathanlucas.htm (6 of 8)11/23/2005 3:04:18 PM

Meet the Sound Guy

brings all your confidence back and you can hear it's all OK. There are some venues where the crowd are down low and the engineer is very high up, and you might be tempted to boost the top end, as the vocals don't seem to be cutting through where you are. But if you were to mix to make things sound good behind the desk, the vocals would be really shrill and piercing for the crowd. You just need to keep making small adjustments and popping out to check."

Stage to studio Despite his obvious enthusiasm for live engineering, Jonny hasn't left the studio behind. Far from it — he's set up a studio with Sevenball in the band's erstwhile rehearsal space, an industrial unit which is split into a performance space and small control room. "The band had already set up an HD recorder and I had a lot of equipment at home that I'd been collecting for a while. Sevenball had been sharing the space with another band, and when the other band decided to move out it was just at a time when I was moving house and needed to get all of the gear out of my flat! You can't really mix loud at 3am in a flat anyway, and in a business unit you can do what you like. It works out for all of us, because I share the rent with the band, they have a place to rehearse and record and I use it to record other bands when they're not there." Working at the Barfly also provides a more direct opportunity for Jonny to put his studio skills to use, in a small studio above the venue used to record the live shows. "The multicore going into the FOH desk splits so that everything plugged in on stage also gets sent upstairs to a seperate desk in an isolated room. Obviously, the FOH engineer is just mixing for the audience in the venue, whereas the engineer upstairs can do an entirely different mix for recording purposes. XFM regularly broadcast shows from the Barfly — I've had the chance to record all sorts of people, from Mel C to Carbon Silicon (Mick Jones from The Clash and Tony James from Generation X), Willis and The Others. There's a digital phone line going to XFM, so 10 minutes after the show you can listen to your mix on the radio, which is pretty cool. "It's a funny mixture of studio and live work, in that the performance is all happening live, so the pressure is still there, but you're in a slightly more controlled environment. Of course, since what you're doing is being recorded, you can scrutinise it afterwards, whereas with a normal live show, once it's gone it's gone. You can't go over it again and beat yourself up about it!" So if Jonny was forced to choose between live and studio work, which would it be? "I really couldn't just do one or the other. I love the studio side of things but you don't get the adrenalin rush like you do when you've got a crowd in front of you and you're responsible for what's coming out of the PA, which is absolutely awesome." You can view video of Sevenball in action in Prague at www.sevenball.co.uk.

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Meet the Sound Guy

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

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

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PA Basics

In this article:

How much mains power? Working it out User beware Going through a phase Distribution deal Safety checks Care and maintenance Summing up

PA Basics Power & Electrical Safety on stage Published in SOS December 2005 Print article : Close window

Live Sound

Staying safe on stage is more than a matter of simply making sure that willing hands are available before taking a dive. Knowing how to properly handle the mains power we all need is also crucial to performance health... Feature by Mike Crofts Photographs by Mike Crofts

Whatever the size, complexity or cost of your live sound rig, one of the first — if not the first — question on your mind when you get to a venue will usually be "where do I plug it in?" Depending on the venue, the answer can vary from a wall-socket behind a plant pot to a dedicated and professionally-installed supply that is reserved for your exclusive use, fully tested and certificated, and for which (with any luck) you'll have brought an appropriate connector. Whatever you encounter, you'll need to know some basic rules. When it comes to portable live-sound systems, this means firstly, using a suitable electrical supply; secondly, using suitable equipment; and, thirdly, connecting and using that equipment safely.

How much mains power? What constitutes a suitable supply will depend, of course, on what you need to plug into it: if it's your own equipment you'll presumably know what supply

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capacity is required, but there may be other factors to consider if additional gear needs to be connected to the same supply. Such gear might include a visiting disco, a lighting rig, or other event equipment — for example, fridges at summer events. A good first step, then, is working out what current your equipment will draw from the mains. The power rating of each piece of gear should be stated on a panel fixed close to the mains connector, or where a fixed mains lead enters the equipment. The power rating may be expressed as a current (in Amps) or as a power figure in Watts. It's generally best to work out the total current your gear will draw, How much power will the average band's gear actually need? The only way to know adding up all the individual figures to for sure is to add up the power requirements find the total load you'll be connecting of each individual item. to the mains. To convert Watts to Amps, divide the Wattage figure by 230 (mains voltage). As an example, a piece of equipment with a mains power rating of 100 Watts (not 100W of audio power) will draw a little under half an amp. In a small venue that is only offering 13-Amp sockets of the normal domestic type, you can then work out how you need to wire up. If the total connected load of your system — including the backline equipment — is comfortably within the rating of a single or double 13-Amp socket, it's perfectly all right to connect it all from a single point. After all, that's what they're designed for! Try to avoid too many connections between this point and your equipment. It's much better to have a single power lead of the required length than two shorter ones joined together: less to go wrong!

Working it out One common mistake is assuming that audio output power is the same as the mains power required to operate the gear. If an amplifier were 100 percent efficient, you could, in theory, use all the mains power as audio output power, but this is not the case in practice, as some of the power used by the amplifier is dissipated as heat. A typical full-range 'active' speaker with built-in amp modules, rated at 240 Watts audio output, would have a mains power rating somewhere around 350 Watts. A useful rule of thumb (if you don't have the manufacturer's file:///F|/SoS/SoS%2012-2005/live_pabasics.htm (2 of 7)11/23/2005 3:04:26 PM

A professionally made distribution box with meters to indicate AC mains voltage and

PA Basics

current. stated figures) is to multiply the audio output power by 1.4 to get an idea of how much mains power would be needed, then divide by 230 to find out the current consumption.

The table opposite gives a rough guide to the supply current likely to be required by a band with three backline amps and a vocal PA. Bear in mind that equipment may demand a much bigger supply current when it is first switched on, so don't be tempted to turn everything on from a single switched socket — you wouldn't want to do this anyway, for many other reasons, such as risking a huge pop through your speakers! Also consider that the power that you can safely run your system on may not be enough to realise its full performance capability. Any system capable of delivering good bass power will need to draw a hefty current from the mains, and if, in the above example, we were to replace our typical small speakers with, say, a pair of Mackie SA1521s, the makers recommend that each speaker's mains supply is capable of providing seven Amps at 230 Volts! This is, of course, not a constant current requirement, but it does illustrate how important a good power source is for getting the best from your gear.

Equipment

Mains power needed

Mains current needed Amps

2 x 240W active speakers 480W x 1.4 = 672W

672 / 230 = 2.92 Amps 2.92

Mixer

100W (stated)

100 / 230 = 0.44 Amps 0.44

2 x rack processors

20W each (stated) = 40W

40 / 230 = 0.17 Amps

3 x 100W backline amps 300W audio x 1.4 = 420W

0.17

420 / 230 = 1.83 Amps 1.83

TOTAL

5.36

ROUNDED UP

5.5

User beware While we're talking in Amperes, it's worth remembering that electrical current is a dangerous animal; a current of only 50 Milliamps (0.005 Amps) can be fatal, and our typical small rig above is using over a thousand times more current than this. Safety is thus a huge consideration, and the use of a suitably rated supply is only the beginning. The best way to stay safe is to use only well-maintained equipment (including cables and connectors) that are properly designed for the task in hand, and to make sure that they are used as the manufacturers intended. If the venue in question is unfamiliar to you and you are responsible for providing and operating the PA, always check that the supply you're asked to use is suitable. Just because it's a 13-Amp socket doesn't mean that it's capable of supplying 13 Amps: it may have been DIY-installed as a spur from a domestic ring main, originally to light a garden shed or run a fountain or something! If you're operating in any kind of business or commercial premises, they should have an up-to-date electrical safety certificate. A quick look at the distribution

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board or consumer unit should show the overall current rating of the circuit you'll be using, and you can also see if it uses old-style wired fuses or the more modern MCBs (Miniature Circuit Breakers), which react more quickly if the rated current is exceeded. Fuses and MCBs do not protect you from electric shock, so always make sure that your system is fed via a residual current device (RCD). This could be at the main board/box, on the A professionally made distribution box with socket itself, or at the point where a four 16-Amp outputs, all RCD protected, for separate spur is fed. If you're not sure limited outdoor use. that this is the case, use your own RCD, either as a plug type or one of the RCD plug-in adaptors readily available for a few quid from any electrical retailer. The RCD should be as far 'upstream' as possible so that it protects as much as possible, and wherever it is, make sure you test it before use, by using the built-in test button. If it doesn't seem to work, find another! A final word on RCDs: they are there as a backup in case anything goes wrong, not as a substitute for poorly-maintained, faulty or unsuitable equipment.

Going through a phase Most small venues are likely to have a single-phase supply, as will normal domestic premises, and for the purposes of this basic article we won't be looking at the whys and wherefores of 'three-phase' systems, other than to point out that all the sound equipment should be connected to the same phase. Any other electrical equipment, such as lighting, should also share the sound-system phase if it is possible for a person to come into physical contact with both systems — for example, to touch the lights and a guitar at the same time. It's a given that the venue's technical staff should supervise any connection to a three-phase supply.

Distribution deal Having found a suitable supply point, you now have to feed it to all your equipment. For all gigs where a 'proper' supply is available, I use a professionally made portable distribution box, which has a single 32-Amp inlet and 32-Amp breaker, feeding four 16-Amp outlets, all via separate combined RCD/MCBs. Although, on the face of it, I've got four 16-Amp outlets, giving a total of 64 amps, I can only use 32 Amps overall, with each outlet limited to 16 Amps. I run my front-of-house speakers from two of these feeds, the monitors and desk from the third, and the stage backline from the fourth. This splits up the load and ensures that each feed is fully RCD protected. As mentioned earlier, it is always best to file:///F|/SoS/SoS%2012-2005/live_pabasics.htm (4 of 7)11/23/2005 3:04:26 PM

PA Basics

have an RCD as far upstream as possible, and I would ensure that my original 32-Amp source incorporated suitable protection if available. For smaller indoor events, a single 13Amp fused RCD plug feeding into a multi-way distribution board (four or six sockets) is fine, as the total current can't exceed the 13-Amp fuse rating in your RCD plug. From this distribution board you should try, where possible, to connect direct to equipment, or feed A damaged mains plug recently discovered at the bottom of the cable trunk — to be the equipment in logical groups. disposed of straight away! Normally, you can take the initial feed from a socket at the back of the stage and run all your backline straight from this, with one feed going off to the PA system. If you need to use more than one socket in a small venue, ensure that all your signal connections are balanced, and never, ever remove an earth connection to get rid of hum or noise. Also take care when using those 'flying saucer' extension reels. They are very useful and neat, but remember that their maximum current-carrying rating only applies when the cable is fully unwound.

Safety checks Your visual examination, before connecting any equipment, every time you're about to use it, should include checks for: Damage to cables or plugs, including cuts, cracks, abrasions, bent or missing pins. Previous repairs or modifications, including exposed or taped-up cable joins and unsuitable connectors. Exposed inner conductors where the cable enters the mains plug. Signs of damage to casing and covers. Obvious signs of previous problems; for example, signs of water, moisture or heat damage.

A visual check on a regular basis (by a competent person, such as a qualified electrician or someone with appropriate training) should include taking the cover off each mains plug and checking that: All wires are firmly attached (screws nice and tight) to the correct terminals, with no bare wires showing. The cable outer sheath is firmly gripped by the cord grip. There is no debris or signs of damage internally.

Electrical testing on a regular basis (by a professionally-qualified and suitably trained person) normally includes all of the above, plus: Additional testing of earth integrity and insulation. Test results recorded and appropriate labels attached to the equipment.

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Failed equipment identified for disposal or repair.

Care and maintenance All leads, connectors and equipment should always be checked before use, even if this is a quick visual check for any obvious signs of damage. If it's your own gear, you'll know it's all correctly fused, but it's best to check if you're not sure. Cables should be undamaged along their entire length and plugs should be securely clamped on, with no inner conductors visible. Cables with moulded plugs are a common sight nowadays, but these plugs cannot ever be re-used, and if Never be tempted to 'lift' the earth wire for any reason. If you find a plug like this one, damaged or removed for any reason with the earth disconnected, don't use it. they must be thrown away — preferably after destroying them so that an unaware person can't find one and plug it in. If anything looks faulty, then it probably is. Remove it from service and make sure it can't be used again until it has been repaired and tested. All electrical equipment, including cables and connectors should be stored and used in dry conditions unless it is designed for outdoor wet weather use and carries an appropriate IP rating (for mains connectors this will usually mean industrial 'Ceeform' types — coloured blue — either rated IP44 (which is spashproof) or IP67 (which is waterproof).

Summing up We've covered the basics of finding a suitable supply and connecting the gear to it, but there are other things to consider when rigging. Cable runs need to be thought out to avoid or minimise trip hazards, and a generally neat cabling job will be much easier to troubleshoot than a spaghetti surprise. Don't forget the rule 'signal before mains'. Connect the power leads last and switch on after everything has been connected in the signal path (with the master levels down, of course). Turn on your power amps last of all, file:///F|/SoS/SoS%2012-2005/live_pabasics.htm (6 of 7)11/23/2005 3:04:26 PM

All portable electrical equipment should be periodically tested for electrical safety, and 'PAT' (Portable Appliance Testing) records kept. Some venues will not allow you to use

PA Basics

and switch them off first when powering down the system.

anything which hasn't been properly tested. Get a quote from your local electrician for testing; it's not expensive and is well worth it for the peace of mind.

In this article I've taken a very basic and superficial look at the power side of live sound. There's a lot of additional good advice to be found, and it's well worth taking a professional approach and discovering as much as you can. After all, if you were going to jump out of an aeroplane you would, presumably, want to know that your parachute was (a) of the correct type; (b) correctly installed on your person; and (c) recently tested! Electrical power is a serious business, so if in doubt, ask a qualified electrician. If you don't know one personally, someone you know will, or you can look one up in the phone book. There are also plenty of useful pages on the Internet. The Health and Safety Executive web site, for example, has a lot of relevant information and links to some very good guidance publications. Check out www.hse.gov.uk. Published in SOS December 2005

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

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

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RCF ART 322A

In this article:

Speaker anatomy Jargon explained Power points Testing times Bringing up the rear Summary

RCF ART 322A Active PA Speakers Published in SOS December 2005 Print article : Close window

Live Sound

RCF ART 322A Pros Sensibly priced. Compact and manageable. Generally good sound quality. Remote control option.

Cons As with most plastic speaker systems, the lower mid–range isn't as tight as from a good wooden cabinet design.

Italian company RCF have a good pedigree — for one thing, they helped design and manufacture for Mackie when the latter first entered the live sound market. We check out RCF's own entry in the portable powered PA stakes. Reviewed by Paul White Photography by Mark Ewing

Summary RCF have built a mid–priced plastic speaker system that performs as well as more expensive systems and also features a remote–control option. It is ideal for band vocals in small and medium venues, and in conjunction with a suitable sub it makes a convincing full–range system.

Information £750 per speaker including VAT. RCF MI Pro Sales +44 (0)1753 655566. Click here to email www.rcfaudio.com

The Italian–based RCF loudspeaker manufacturers were bought by Mackie just before the latter moved into live sound equipment manufacturing. RCF then became involved in the design and manufacture of the popular Mackie SRM450 PA speaker (see review in Sound On Sound Live August 2004). However, when Mackie restructured following its acquisition by Loud Technologies, RCF was cut loose, allowing it to market new products of its own design, and the model reviewed here seems to be a head–on competitor for the SRM450. RCF already have a reputation for building good–quality speakers in moulded cabinets, and their existing ART 300As (the ART line was started in 1996) are used extensively by hire companies around the world. The new ART series reviewed here comprises four models, two with 12–inch woofers and two with 15–inch woofers. There's little difference in maximum SPL between the four models: the lowest powered one has an SPL just 2dB below that of the highest rated. The reason why there are four models in the range but only two woofer sizes is that two variations per woofer size are available, offering one– or two–inch compression drivers at the high–frequency end. The models with the smaller tweeters also cut off at 50Hz at the low end, as against 45Hz for the two–inch models.

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RCF ART 322A

Speaker anatomy The ART 322A reviewed here is an active system based around a moulded cabinet and featuring a 12–inch woofer teamed with a two–inch compression driver. The 312A and 315A have one–inch compression drivers, while the 'big gun' of the range is the 325A, delivering 129dB SPL from a 15–inch woofer and a two–inch HF compression driver. All fit into the same size of cabinet (680 x 405 x 345mm). Physically, the cabinet is of a similar size as the Mackie SRM450s, delivering 400W of power with a maximum SPL of 128dB. However, RCF have developed new amplifiers and transducers, with the aim of delivering both quality and power without excessive weight (23kg), so there seems to be little in common with the Mackie product other than general shape and styling. Both rear corners are bevelled, to allow the cabinet to be used as a floor monitor at 45 degrees, and the bass ports are divided into six parts, four to either side of the horn and two in the bottom corners below the woofer. All vulnerable parts are recessed, including the rear panel and its controls. The drivers both use Neodymium magnets, and the compression driver has a titanium diaphragm feeding a constant–directivity horn (fitted with a phase plug) producing 90–degree horizontal coverage and 60–degree vertical coverage. This is typical for a speaker of this size and is pretty much optimal for the type of venue in which these speakers are likely to be used, although it may be too wide for installations where multiple speakers need to be mounted in arrays. An edge– wound, copper–clad aluminium voice–coil drives the woofer cone, and large magnets are used on both drivers to ensure that they operate as linearly as possible at high power levels. A frequency response of 45Hz–20kHz is claimed, but with no limits specified (that is, how many dBs down the output is at these frequencies), the figure is of limited use.

Jargon explained Cabinet loading: The resistance to movement of the air within the cabinet. When a speaker cone moves inwards it must either compress (sealed cabinet) or displace (ported designs) the air inside the box. The cabinet loading therefore directly affects the performance of the speaker, most notably in terms of bass output. Class-A/B amplifier: An optimum combination of the power efficiency of Class B and the linearity and low distortion of Class A. Class A/B is by far the most commonly used audio power amplifier. Class–H amplifier: High-efficiency power amp design in which the input signal amplitude determines the power supply voltage, delivering the optimum supply to the output devices at all times.

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RCF ART 322A

Compression driver: Specialised high–frequency loudspeaker with a small, domed diaphragm and usually attached to a directional horn. The narrow horn 'throat' constricts the volume of air into which the driver radiates, hence the 'compression' part of the name. Constant directivity horn: A specially shaped loudspeaker horn design that achieves a relatively uniform dispersion of sound over its intended operating frequency range. Crossover: A crossover is an electronic circuit designed to separate high– and low–frequency signals from each other so that each can be fed to speakers optimised for the role — ie. large and robust speakers for bass, small and light (and therefore fast) ones for high frequencies. If a crossover is placed between power amps and speakers (usually built into the speakers), it is said to be 'passive'. An 'active' crossover is used to divide signals before the power amps, so separate amps are then used for each band, further adding to efficiency. Crossovers can be two–way, just splitting highs and lows; three–way, adding a mid band; and occasionally four–way. DC offset: Any AC signal (such as a music signal) has positive-going half and a negative-going half to its waveform. To maximise the headroom of the circuit stage, this 'voltage swing' should occur around the centre line between maximum positive and negative voltage, ie. zero volts. If the signal is not centred, but 'offset' towards either side, clipping of that half of the waveform will occur earlier than necessary. 'Flying': Suspending from a suitable load–bearing point in the ceiling. Common in installations and large touring rigs. Neodymium magnet: Very high strength 'rare–earth' magnet type made out of neodymium, iron and boron. Its use in a loudspeaker allows the coil attached to the diaphragm to be made smaller and therefore lighter, thereby increasing the speaker's efficiency. Phantom power: Standardised scheme of providing an invisible (hence 'phantom') power supply voltage to capacitor (condenser) microphones using the same cable as the balanced audio output. Phase plug: In a horn-loaded compression driver the phase plug is a small physical device designed to ensure that all components of the sound from the diaphragm surface arrive at the horn's throat at the same time, or 'in-phase'. SPL: Sound level calculated in decibels compared to a reference sound pressure, commonly 20 micropascal (20 uPa), defined as the threshold of human hearing (0dB SPL).

Power points As usual, separate amplifiers feed the two drivers, and these are themselves fed from an active crossover circuit, operating at 1.2kHz, that also incorporates low– cut filters to attenuate any frequencies lower than the speakers can comfortably handle. This is important with any ported design, as the cabinet loading on the rear of the driver ceases to be effective below the port's cutoff frequency, so there's nothing to prevent the speaker cone from moving too far on loud bass notes. In this model, a 350W Class–H amplifier powers the woofer, while a 50W Class–A/B amplifier powers the horn driver. The amplifiers themselves are file:///F|/SoS/SoS%2012-2005/live_rcfart322a.htm (3 of 6)11/23/2005 3:04:31 PM

RCF ART 322A

documented as being monolithic, which suggests that they are designed around purpose–built all–in–one chips rather than being made from discrete components, and they employ a high–efficiency design that can handle short peaks around six times higher than their nominal rating. Cooling is via a rear–panel convection heatsink, so no fans are needed, and amplifier protection is built in to guard against short–circuited speakers and DC offsets. DC–offset protection is important because if an amplifier fails and the output subsequently rises to one of the DC supply rails, as often happens in such cases, the speaker would fry very quickly. The quality of a moulded cabinet can make or break the sound of the complete system, and it is clear that RCF have put in a lot of work to keep resonances in this cabinet to a bare minimum. A polypropylene material is used for the main moulding, with recessed handles top and side, as well as four captive nuts (hidden beneath blanking plugs) for the fitting of flying hardware. A pole recess with pole clamp is fitted into the bottom of the cabinet, and the woofer is protected by a powder–coated, perforated steel grille.

Testing times Side handles are ideally

I checked out these speakers using full–range pre– placed for easy lifting. recorded material, live vocals, DI'd acoustic guitar and live electric guitar. They work well either alone or in conjunction with a sub–woofer. I tried them with a Mackie 15–inch SWA1501 sub and experienced no problems. Unlike many systems that use titanium horns, the ART 322As have a relatively smooth high end, rather than being incisive and scratchy, but the sound is still detailed and clear. They handle vocals in a very neutral way with their EQ set flat (see 'Bringing up the rear' box for details of the built–in EQ), with better mid–range clarity than many plastic cabinet designs, and even the low–end response of the cabinets when used without a sub is acceptable at lower power levels. For vocals and guitars the range is fine, but if you need to mic kick drums or amplify bass instruments an active sub that filters out the low end feeding the main speakers is probably your best option. When the system is being used primarily for vocals, the Vocal EQ setting works well, giving more presence. The Music setting is fine where you want that 'loudness button' effect, but sounds too 'scooped' in the mid–range for my liking when vocals are a key part of the mix. At one venue, the speakers were working at close to their maximum level and the vocals were very loud, with surprisingly little in the way of feedback problems. Immediately after the set I checked the temperature of the rear panels and found them to be comfortably warm to the touch — not piping hot, as I'd feared. Ergonomically, I like the packaging of these speakers, especially the positioning file:///F|/SoS/SoS%2012-2005/live_rcfart322a.htm (4 of 6)11/23/2005 3:04:31 PM

RCF ART 322A

of the handles, which makes it easy to lift or carry them. They're also not too heavy. The locking screw that bears down on the mounting pole prevents the cabinet from rotating on its stand, and if you need to stack two cabinets on the floor, the feet of one fit snugly into moulded recesses on the top of the cabinet below.

Bringing up the rear Doubling as a heatsink, the rear panel of the 322A plays host to the input connectors, the power socket and the volume control, as well as additional switches that require a little explanation. The input is on a balanced XLR (with hardwired male and female options to allow the daisy–chaining of cabinets) and there's also a standard jack that can accept balanced or unbalanced sources. This degree of choice means that you can connect either sex of XLR or any line–level jack and still get the system working, which can be useful when 'improvising' at non–standard gigs. A rotary control adjusts the input gain, but there's also a sensitivity switch that allows the user to select mic– or line– level inputs. Normally you'd feed in a line–level signal from a mixer, but if pushed you could simply plug in a dynamic mic to get up and running. As well as the usual input connectors, mains power There's no phantom power, so you can only really inlet and volume control, the use dynamic or battery–powered electret mics rear panel hosts a switch for direct into the speaker. A built–in limiter prevents the amplifiers from being driven into clipping (which, the speaker's three basic EQ modes. in turn, protects the speakers) and, as is invariably the case, the red clip–warning LED is on the rear panel where the band can see it but the sound engineer can't! A green LED on the rear panel shows that the speaker is powered up. Many active PA speakers have switchable EQ options, and this one is no exception, but what is a little out of the ordinary is that you can buy an optional ART RC01 remote control to switch the level, mute and EQ modes via infra–red, if you feel that would be useful. While the gigging musican hauling a pair of these in and out of pubs may not need remote control, I can see many permanent installation situations where it would be desirable, especially when speakers are being 'flown' in inaccessible positions. For manual operation, the single rear– panel button can be used to step through the three EQ options, with EQ status denoted by the colour of the front baffle LED. When this LED is green the response is flat, which is how it leaves the factory. Red denotes a Music EQ 'smile curve' with top and bottom boost, while a yellow LED denotes an EQ with more of a presence character that is designed for optimum vocal intelligibility. This is all simple enough, but it means that you have to look at the front of the cabinet while your hand is around at the back pressing the button. Pressing and holding the button enters and exits remote–control mode. Mains power comes in on a conventional IEC connector with adjacent switch and fuse, and there's also a recessed voltage–selector switch to allow 230V or 115V operation. If the unit is switched from 230V to 115V, the fuse value needs to be

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RCF ART 322A

increased (presumably doubled) accordingly, though for some reason the manual states that this should be done by an authorised RCF service centre rather than by the user.

Summary While they're not the cheapest plastic speakers on the market, the ART 322As are by no means the most expensive either, yet their performance is up there with the best. A good wooden system would probably beat them, but then wooden cabinets also tend to be heavier. Though there is a matching RCF sub (the ART 705AS), I didn't get to try this, but — as explained earlier — the 322As worked fine with my Mackie SWA1501, which is of a similar size and power to the RCF model. The remote control option (see box above) is more relevant to installations than to gigging bands, but you don't have to buy it if you don't need it. The fixed EQ settings are useful in specific applications, but I prefer to stick to 'flat' and then use the EQ on my mixer to get the right sound. Having said that, if you want a 'tizz and boom' disco sound, you can select the Music setting, and for announcement work or band vocals the Voice setting may offer some advantages. To sum up, then, these RCF models combine portability and convenience with good sound quality and sensible pricing. Used in pairs, they're ideal in pub and club venues for handling vocals and instruments, and teamed with a suitable sub they're capable of impressive full-range results. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Stage Monitoring & Monitor Mixing

In this article:

Stage Monitoring & Monitor Mixing

Requirements of musicians 'Self-adjust' monitor mixing Workshop Published in SOS December 2005 Sound quality requirements Print article : Close window Wedges and side fills Live Sound Amplification for monitors Soundcheck protocol Mixing monitors from the FOH console Suitable consoles for Although monitor engineering is often thought of as monitoring subordinate to handling the FOH sound, in reality it's Sending signal to the at least as important. We take a tour around this most monitor console crucial of live-sound subjects. Positioning of monitor console Just for drummers Feature by David Mellor Creating the monitor mix Original header photo by Jyoti Mishra In-ear monitoring Feedback To create a musical performance, two Contacts things have to happen: your audience Career opportunity? needs to hear you, and you need to

hear yourself. If you can't hear yourself clearly, how will you know that you're playing or singing well? In purely acoustic music, being able to hear oneself is often taken for granted. But there are situations where this doesn't happen as it should. For example, in an orchestra performing on stage, each musician needs to hear his or her own instrument clearly and distinctly from the other instruments around them. But sometimes the acoustics on stage make this difficult. Suddenly, one's ability to perform well has been diminished severely by the inability to hear one's own playing. The same applies to amplified music. In the early days of pop and rock bands it was common to provide only front-of-house (FOH) amplification, commonly mixed from the stage by a member of the band (who, of course, couldn't hear the FOH PA properly). Although many exciting performances (and undoubtedly many distinctly unexciting ones) have been given in this way, the fact is that no-one is properly in control of what the audience hears. The one advantage is that the band can angle the speakers and set their levels so that they can hear themselves and each other reasonably well, most of the time.

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Stage Monitoring & Monitor Mixing

Fortunately, progress has been made and we now recognise that it is essential to have the mix position at front-of-house, placed centrally amidst the audience area. The FOH engineer is now ideally placed to control the sound the audience hears. The problem now is that the band are no longer in any kind of control whatsoever of what they hear. Clearly, in an ideal scenario, there should be a completely independent system to provide the band with crystal clear sound so they can hear their own individual performances and the overall sound of the ensemble. This is what stage monitoring should provide. Stage monitoring is taken very seriously by top professionals, and should be by anyone working in live performance, right down to pub gig or theatre foyer level. Good monitoring consists of having the right equipment, suitable for the nature of the venue and performance, setting it up well and, of course, operating it effectively.

Requirements of musicians Above all, musicians need to feel that they are making great sounds. If they feel that the performance is good, the performance will be good and the audience will go away whistling the tunes. Also, performers need to feel secure. Security comes from knowing what the other band members are doing, knowing where they are in the song, and being certain that the notes and rhythms they are playing fit in with the rest of the band. So let's imagine you're the lead singer of a band. The lead singer needs to feel that his or her voice is strong, in tune, and communicating emotion to the audience. Clarity and good tone of voice are paramount. Also, the lead singer needs to hear the band, so that they know they are in tune and are fully comfortable that the band are following them precisely. If the band are playing to a click track or a recorded backing, strict tempo will be an issue and the lead singer may need the band to be more emphasised in the foldback, since now everyone has to follow the click (even though only the drummer would normally hear it) or recording; the band cannot follow the singer. The other band members have their own individual requirements, but in general they also need to feel that they sound great. They need to hear the vocal, too, otherwise they might have a blank moment and forget whether they're in verse two or verse three (that's scary when it happens). They will also have a preference about which other instruments they need to hear most clearly, to feel as though they're 'gelling' with the rest of the band.

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Stage Monitoring & Monitor Mixing

'Self-adjust' monitor mixing Over the years, the topic of 'selfadjust' monitoring has waxed and waned several times. The idea is that if musicians can never be happy with the mix the monitor engineer gives them, why not let them adjust their own individual monitor mix? Systems such as the Intelix Psychologist allow up to 16 individualised personal mixes [and The Hear Technologies Hear Back system is SOS Live reviewed the Hear one of several on the market that offer Technologies Hear Back personal performers the option of self–adjusted monitor mixing system in our monitor mixing. September 2005 issue]. Sounds like heaven, and surely nothing can go wrong? Well, yes, something can go wrong. As tempting as it might seem to eliminate that pesky monitor engineer from the signal chain, there are several problems with self-adjust monitoring. First and foremost is that the musicians now have something else to do as well as perform. Second, it takes years of experience to become a good monitor mixer. When are the musicians going to get that experience for themselves? Third, the natural instinct of any newcomer to a mixing console is to increase the level of anything that seems too quiet. So eventually everything gets boosted and all the faders are at the top end of their travel. Fourth (should I stop counting now?), monitor mixing is largely about managing the clutter of sound that can so frequently occur on stage. With individuals controlling their mixes, this is just so not going to happen. Having said that, the idea still seems to be a good one, so perhaps it just requires further work. Or maybe musicians will become more savvy in this area and be able to use their new power wisely and responsibly. One area where self-adjust seems most likely to succeed is in-ear monitoring. Here, the performer can't spoil the sound for anyone else through spill, and the clarity of in-ear means that it should not be necessary to apply too much level through ignorance or inexperience. Time will tell on this point.

Sound quality requirements There are also certain technical requirements for stage monitor sound. The first thing to remember, though, is that this is a means to an end. The audience doesn't get to hear the monitors (or at least they shouldn't) and no-one is going to be releasing a live album of the monitor mix. So whatever gets the job done best is the best monitor sound. It should go without saying that the 'no faults' criterion always applies — no noise, hum, buzz, clicks or distortion. Beyond that, there are two features that deserve consideration. The first is that although monitors don't need a fantastically extended low-frequency response, they do need a good firm, punchy bass, subjectively. Bass tends to wash around the room and there will be a lot of file:///F|/SoS/SoS%2012-2005/live_stagemonitoring.htm (3 of 15)11/23/2005 3:04:42 PM

Stage Monitoring & Monitor Mixing

'mush' floating about, which the monitors need to be able to overcome. The other is that the sound from the monitor speakers should not rip your ears off with its harshness. In the quest for clarity it's very tempting to wind up the high-mid and high frequencies, or choose monitor loudspeakers that emphasise the top end (piezo tweeters are offenders in this respect). This is a sign that something is wrong. Either the equipment is sub-standard, it is inappropriately installed or the engineer doesn't really know what he is doing when it comes to monitors. It's also worth considering treating the monitor sound. EQ, of course, is a must, just as you would EQ anything else. Compression of certain individual channels can also help improve the fullness of the sound, although this has to be done with care — as with compression of anything going to the FOH PA — as it reduces the margin before feedback. Also, singers often perform better with a little reverb in their foldback, whether or not it is used for FOH. Any such treatment would be on an instrument-by-instrument basis, rather than used on the whole of the monitor output, and you should carefully assess whether it really is required. Over-complicating a monitor mix unnecessarily is unlikely to bring good results.

Wedges and side fills The classic stage-monitor design is, of course, the wedge. The wedge shape angles the sound upwards from floor level to the musicians' ears. Typically, a wedge would contain a 12-inch (30cm) or 15-inch (38cm) low-frequency drive unit and a one-inch (25mm) compression driver with horn, a classic example being the Martin Audio LE400C. Some interesting features here include the asymmetry of the cabinet's cross section. Place it one way up and you will have an angle of fire of around 40 degrees, but turn it over and the angle will be around 50 degrees. The choice depends on how far away the performer will be from the wedge. The horn is mounted with its long axis vertical, which means that it has a wide dispersion vertically and a narrow dispersion horizontally (70 degrees x 40 degrees at the -6dB points, says the specification). This gives the performer freedom of movement in the forwards/backwards direction. The tighter horizontal distribution allows monitors carrying different mixes for different performers to be placed closer together. Some wedges (but not this one) allow the horn to be rotated according to whether you want a broad spread of sound vertically or horizontally. Just remember that, counter-intuitively, it is the narrow axis of the horn that delivers the broader spread of sound. Many wedge monitors allow the option of passive or active crossovers. In passive mode, they are fed by a single amplifier and an internal crossover separates the high and low frequencies to their respective drive units. In active mode, the internal crossover is not used and each drive unit is fed by its own external amplifier. Naturally, an active crossover has to be used before amplification, to separate the low and high bands of frequencies. One might expect that wedges would be available with internal amplification, which would seem to simplify matters. However such models are rare, one notable exception

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being the Meyer Sound USM1P, which incorporates amplification capable of a maximum of 550W of output in short bursts. Another feature of the Martin Audio LE400C is that it comes in left-hand or right-hand models. Does this mean that they come in stereo pairs? Well, yes — and if a performer of sufficient This Meyer Sound wedge monitor is selfstature in the musical hall of fame powered and is capable of power output up to 550W. However, most dedicated monitors asked for stereo monitoring, doubtless are passive. they would get it. For a lead vocalist, it would be beneficial. But even if monitoring is all mono and two wedges are used to widen the area of coverage, if the wedges were identical the field of coverage would be asymmetrical. If the wedges are mirror images of each other, the field will be symmetrical. Other features of wedges include the option to choose between models that have the HF drive unit positioned above the LF unit, and those that are side-by-side. The thinking here is that a vertically orientated wedge will give a wider and more consistent sound from side to side. However, this is at the expense of a higher physical profile. Obscuring the performer's knees somewhat may be no big deal on stage, but for broadcast the trend is towards lower-profile monitoring. Where there's a wedge, there's often a 'side fill'. The best way to use wedges is to give each performer their own, and place them as close to each individual performer as possible. However, some performers are not content to stay rooted to the spot and want to take advantage of other areas A possible stage layout showing positions of of the stage to strut whatever stuff they monitor wedges, side fills and the monitor happen to be in possession of. So engineer with console. these areas are covered by larger 'sidefill' monitors. Unfortunately, side fills are where everything starts to fall apart. A five-piece band with five separate wedge monitors puts a lot of sound on stage, and since it is impossible to focus sound precisely, there's a lot of spill flying around which does nothing but confuse the sound for everyone. Add side-fill monitors whose whole purpose is to fill the stage with sound and you have a recipe for a sonic disaster. One common complaint among musicians on stage is that the sound is loud but they can't hear anything. This may seem like an oxymoron, but it can easily happen, so side fills are not a category of equipment that should be used automatically. They should be used when they are needed, and precisely where they are needed. The rule is only to direct sound at parts of the stage that will actually be used, so if a performer wants to be spontaneous and use an area of

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Stage Monitoring & Monitor Mixing

the stage that wasn't planned to be used and isn't covered by side fills, he or she will have to adapt to that localised situation. The alternative is to compromise the sound all round.

Amplification for monitors The first law of amplification applies to monitors too. 'What is the first law of amplification?', I hear you ask. It's the one that states that the amplifier should have more power than you would ever possibly want to use, for the particular application you have in mind. There's still a general feeling around that if you have a speaker that claims to be able to handle 100W, the amplifier should be no more powerful than that, 'to be on the safe side'. If that were the case, it would make sense to manufacture cars that could go no faster than the maximum speed limit. But with a car we all know that it's better to have power in hand, and it's exactly the same with amplifiers. Small amplifiers struggle, big amplifiers breeze through — but the engineer has to be in control. The power-handling capability of a small wedge monitor might be a mere 100W, which honestly ought to be easily enough, given that they are always used close to the performer. Some, however, claim to be able to handle up to 500W, and the odd one even more. The McCauley SM9502, for instance, claims RMS power handling of 1100W and a peak SPL of 136dB. Wow! Some monitors feature 'loop-through' input connectors so that the amplifier can be connected to one wedge, then on via the loop-through to another wedge. This is useful for small-scale systems, as you don't need so many amplifiers, and in general any decent amp should be able to drive two wedges without trouble. However, this reduces flexibility and should only be considered as a 'rung on the ladder'.

Soundcheck protocol Anyone who has played live will understand the rules of the game when it comes to soundchecks. The headline act gets as much time as they like, the support act gets barely enough. If there are additional support acts, some might even get no soundcheck at all. I've seen that happen. But a good rule of thumb is that things should come together within three songs. For that to be possible, both the band and the engineers need to have good soundcheck technique. For the band, this will mean choosing songs file:///F|/SoS/SoS%2012-2005/live_stagemonitoring.htm (6 of 15)11/23/2005 3:04:42 PM

This Allen & Heath ML5000, with an impressive total of 16 auxiliary sends, is well suited to the task of monitor mixing.

Stage Monitoring & Monitor Mixing

that reflect the typical sonic content of the show as a whole, and having an individual awareness of what they want to achieve from the monitors, which they should already have communicated to the monitor engineer. For the engineers, really it's down to experience and getting more and more practice, leading simply to knowing what to expect in the majority of circumstances. Granted, there may be the occasional Guatemalan marimba orchestra to take care of, but vastly more often it will be a common-or-garden band line-up on stage. One inconvenience of the soundcheck is that both the FOH engineer and the monitor engineer have to do their work simultaneously. Whereas the FOH engineer is left alone to achieve the sound he or she thinks is appropriate, possibly guided by the band's management, the monitor engineer will be in tight communication with the band. One typical complaint made by bands, however, is that although the monitors sounded good during the soundcheck, everything fell apart during the show. Obviously, the monitor engineer must have done something different. But the monitor engineer swears blind that everything is exactly the same. The cause of this conflict is spill from the front-of-house system on to the stage. When the auditorium is empty, during the soundcheck, there's a considerable amount of reflected sound coming back to the stage. The musicians will base their requests for adjustments to the monitoring on the sound from the monitor system, plus the spill from FOH. But during the performance, when the audience is present, that spill is largely absorbed. Human beings are excellent absorbers of sound and suitable volunteers would make highly effective acoustic treatment. Now that the FOH spill is absent, naturally the monitors will sound different. The solution to this is to soundcheck at least one song with the FOH system either completely down or attenuated by 20dB or so. This will give a much more accurate representation of what the monitors will sound like during the performance. The smaller and more reflective the auditorium is, the more relevant this tactic will be.

Mixing monitors from the FOH console In a small-scale system it might be necessary to mix the stage monitors from the front-of-house console. This certainly can be done, but there are certain drawbacks. Firstly, let's consider where the monitor signal comes from... The monitor signal could be exactly the same as the FOH mix, and in fact for side fills it often is, but the requirements of monitoring are different from the audience's requirements. So the monitor mix really needs to be a different mix from the FOH mix. Fortunately, every PA mixing console has the ability to let you set this up, in the form of pre-fade auxiliary sends. Each channel has an auxiliary send control, which is like another fader that mixes into an auxiliary buss and separate output from the console. 'Pre-fade' means that the signal comes from a point in the channel prior to the fader, so the fader has no effect on the level of that channel in the monitors. Using the pre-fade aux sends on all of the channels, a completely independent monitor mix can be created. But it would be an impoverished console that only had one set of pre-fade auxiliary sends; most have at least two pre-fade aux busses. This means that, providing there are at least two wedges and two channels of amplification, there can be two completely independent monitor mixes — one for the vocalist and one for the rest of the file:///F|/SoS/SoS%2012-2005/live_stagemonitoring.htm (7 of 15)11/23/2005 3:04:42 PM

Stage Monitoring & Monitor Mixing

band, most likely. Some consoles have even more pre-fade auxiliary sends. What are not useful, however, are post-fade auxiliary sends. Post-fade auxes take their signal from after the fader. Plainly, this is totally unsuitable for monitoring. The mix the performers hear would be affected by levels that were changing according to the requirements of the FOH mix. When this happens, either by accident or incompetence, it's very uncomfortable for the performers. So we've established that a monitor mix can usually be made from a properly specified FOH console. However, there are some good reasons why monitors should not be mixed from front-of-house but by a dedicated monitor console and engineer: The front-of-house console may only have a small number of pre-fade auxiliary sends. The multicore cable might already be close to being fully occupied with incoming signals and outgoing FOH PA signal to the amplifiers. It's another job for the FOH engineer to look after. There's a likelihood of poor communication between the performers and the FOH engineer.

Suitable consoles for monitoring Deciding on a mixing console that is suitable for monitoring is easy: it has to have plenty of auxiliary sends. There isn't necessarily a difference between an FOH console that is suitable for monitoring and a dedicated monitoring console in this respect. As long as there are plenty of aux sends, it's good for monitoring. Of course, where an FOH console is to be used with a separate monitor console, the FOH console doesn't need all those auxes, or they can be used for effects sends if desired. Typical of a console that is suitable for monitoring is the Allen & Heath ML5000, with no fewer than 16 auxiliary sends. Will a band really require that many different monitor mixes (remember that it's common for two or more performers to share a mix, through separate wedges)? Probably not, but auxes 9-16 on this console are configurable as four stereo sends, stereo sends being eminently suitable for in-ear monitoring.

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Stage Monitoring & Monitor Mixing

Sending signal to the monitor console If you have a very long memory stretching all the way back to the 1970s, or if you watch archive performances on TV, you will probably have wondered why there were often two microphones taped together for each performer. An ungainly solution, but what was the problem? In the case of an archive performance, the reason is possibly to take the signal from one mic to the FOH PA console and the other to a separate console for the film sound. But the other possible reason is to have one mic for front-of-house and the other for monitors. Clearly, this is an inelegant solution and it's much better to use one mic for both purposes, or all three purposes if there is monitoring and recording as well as FOH sound. One possible solution is to solder up a 'Y-cord'. A microphone can supply two inputs reasonably well, if not at tip-top quality, due to the extra loading. The drawback, however, is firstly that a fault in the cable run to one console could short out the feed to both Schematic of a transformer mic splitter. consoles. Also, the earth of one console is now connected to the earth of the other, resulting in a potential earth loop, creating hum. This can be cured by removing the earth connection on one feed, but then that could sever the phantom power from a capacitor mic. Y-cords have their place — mainly in the bin — and there is a better solution, which is to use a transformer mic splitter. The diagram on the right shows a schematic of a transformer mic splitter. You could make one if you're technically minded — check with Jensen or Sowter for suitable transformers — or you can buy one from BSS Audio, EMO or a number of other manufacturers. They don't seem to be widely promoted, which is odd given that they're so useful. You can see if you trace the signal that the mic is connected directly to one mixing console, but via the transformer to the other. Using two transformers, a third split can be created. The ground-lift switch isolates the earths of the outputs if necessary. You might wonder which output goes to the FOH console and which to the monitor console, and it would seem that the FOH console is most important and requires the direct-wired output. However, this means a longer run for the phantom power for any capacitor mics on stage, hence it is better, by a small margin, to take this output to the monitor console. The sound quality will be indistinguishable in a live environment.

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Stage Monitoring & Monitor Mixing

up to you. The monitor engineer might have a personal preference, it might be better to be closer to certain musicians, or there might be technical facilities or people on one side of the stage that you need to be close to. The reason for having the console right there at the side of the stage is, of course, to facilitate good communication with the band. There's nothing worse than band members having to make obvious hand gestures that are visible to the audience. When the monitor engineer is closer, a simple look in their direction with a suitable facial expression can convey the message that something needs attention (of course, a really good monitor engineer would have pre-empted any problem in the first place!). Naturally, the monitor engineer needs to monitor the monitor mix. In fact, the monitor engineer needs to be able to monitor any of the several mixes he or she is creating. This can be done through soloing of the auxiliary outputs, on headphones so that the monitor signal is isolated from the clutter of sound on stage. Suitable headphones are the Extreme Isolation headphones by Direct Sound, which offer up to 29 decibels of attenuation of sound from the outside world. They are as good as Monitor engineers have to be able to hear the ear defenders used by operators of the monitor mix (or mixes) clearly even in the high-noise surroundings of a typical gig. pneumatic drills and by airport workers Direct Sound's Extreme Isolation on the tarmac. One of the great things headphones, which offer up to 29dB of about headphones such as these is attenuation of outside sounds, are one that you don't have to turn the volume solution to the problem. up too much and possibly risk damage to your hearing. Without such good isolation, the volume has to be turned up to compete. The other alternative is to monitor on a wedge, just like the musicians, and there's a good case for hearing the sound exactly the way they do. If I can be allowed a personal anecdote at this point, there once was an occasion when I was playing keyboard on stage and I was having considerable difficulty because the monitor mix I was hearing kept changing suddenly. "Incompetent monitor engineer" was my thought, (which, to be honest, isn't all that unusual at small gig level). However, further consideration revealed that he was monitoring the various mixes on a speaker placed on a table upstage of the console. This speaker was firing directly at the back of the PA stack on that side and reflecting back towards me, so every time the engineer solo'd a different mix, I got it too! The moral of that particular story is that the engineer should monitor on a wedge, or position his speaker downstage of the console.

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Just for drummers Monitoring on wedges is generally workable. But there is one musician in the band for whom wedges might not be enough. Yes, it's the drummer. The problem with drums is that they are so damned loud, so the wedges have to be even louder to compete. In fairness, the rest of the band should really be following the rhythm of the drummer, so perhaps the drummer doesn't need to hear them so clearly. But he or she does need to hear the lyrics, so that he knows where he is in the song. Who hasn't, at one time or another, 'drifted off' and forgotten how many verses have been played so far? (Trouble is, it's sometimes the singer!) The drummer also often has an additional need: to play to a backing track or a click track. If it's a backing track, it's OK to monitor this on wedges. But if it's a click, no-one else wants to hear that click, particularly not the audience. Playing to a click has surprisingly wide applications. One, for instance, is musical theatre shows that incorporates dancers who sing. They actually don't sing, they mime to a pre-recorded vocal track, to which the drummer must keep time using a click. The answer to all of the above is for the drummer to monitor on headphones. This looks ungainly, but it works, particularly with the Extreme Isolation variety of headphones (see previous page) that considerably attenuate sound from outside. In-ear monitoring can work well too, but the earpieces don't provide so much isolation. One further refinement for drummers is the 'ButtKicker'. Yes, really. The ButtKicker contains a linear motor very much like a conventional loudspeaker drive unit, but instead of having a conical diaphragm it attaches to the underside of a seat, upon which the drummer places his or her, er... butt. Low frequencies are therefore channelled to the drummer's brain in the most direct way possible! At present, there is no word on possible side effects...

Creating the monitor mix In creating the monitor mix, the monitor engineer clearly has to take into account any stated requirements of the band. But he or she also needs to work on his or her own initiative — the band may express certain requirements, but they are not engineers themselves and cannot be expected to understand the whole of the process. The difference between the basic monitor mix and front-of-house mix is that the monitor mix has to be effective. It doesn't need to be a wonderfully musical mix, but it must: Allow the musicians to play well together rhythmically.

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Tell the musicians where they are in the song. Allow singers to sing in tune, for which they need to hear themselves and harmony instruments. Allow string players (in particular) to play well and in tune, for which ideally they need to hear themselves, not just the whole string section. Keep the drummer in time with the backing track or click track, if necessary. There are also artistic requirements: Band members should feel that they are performing well. The overall sound of the band has to be good for those performing on stage. If there is any spill into the front rows of the audience (which there will be), it should not spoil the experience for them. Much of the above is handled by the selection and placement of monitors, bearing in mind that closer is nearly always better. However, the mix needs to be handled sensitively too. Just as in a FOH or recorded mix, the priority is to evaluate which are the main instruments and sounds, and get a rock-solid mix of those. In a typical rock band, then, clearly these will be the kick drum and bass (you can't get far without getting those right), then the snare and hi-hat, the guitar and the keyboard (and if there is more than one keyboard, the one that provides the harmonies most of the time will be the most important). All the other instruments are precisely that: 'other', and not nearly as relevant to the monitor mix. As an example, how often do cymbals, other than hi-hat, make an effective contribution to a monitor mix? Hardly ever, and you may not even need to include them. It is very well worthwhile considering also how much sound is pumped on to the stage by the backline. Perhaps the guitarist doesn't need any additional level from his own instrument, or maybe just a little extra clarity rather than a fullblown contribution. One way to approach a monitor mix is to bear all of the above in mind, then construct one generic mix that should suit everyone, copy it to the other banks of auxiliaries and then add and subtract level from individual instruments in certain sends, according to the performers' various requirements. Otherwise you'll have to construct each mix entirely from scratch and you probably don't have nearly enough time for that during the soundcheck. Above all, the vocal should be available with ultimate clarity to whichever performers have need of it. Some performers might be happy for it to stay in the background so that they can concentrate on their rhythm. But if someone really needs that vocal, it had better be clearly audible.

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In-ear monitoring In-ear monitoring is certainly all the rage at the moment, but time will tell whether it is eventually replaced by something different or better. Basically, it does what it says on the tin: instead of wedge monitors and side fills, musicians have earpieces through which they hear their monitor mixes. In-ear systems can be 'off the shelf' or can have custom moulded earpieces individualised to each musician. The custom solution is, of course, more expensive. The benefit of in-ear monitoring is obvious and considerable: there's no need to 'spray' sound around the stage, which so often results in extreme lack of clarity, particularly where side fills are involved, and there's no undesirable spill from the monitors into the audience area. With in-ear monitors, the sound goes directly to where it is needed. So that's all good then? Well, actually, not necessarily. The In-ear monitoring offers various advantages problem with in-ear monitors is that over monitoring on wedges, but it may also they isolate the performer from the make the user feel isolated from the audience. A performer with a pair of audience. earpieces is in his or her own little universe of sound, aurally disconnected from what's going on around them. In-ears don't look as visually obtrusive as headphones, but they are there for all to see. And it's not just the front rows any more, with the common use of large screens in big concerts. It is, of course, possible to use just one earpiece, and many performers do. But it is difficult to get this to work without at least some conventional monitor sound from a wedge. Perhaps eventually the solution would be for musicians to have aural implants, like George Bush (allegedly), so they can get their sound directly, with no visible side effect. In the meantime, the balance of virtues between wedges and in-ear monitoring will have to be carefully assessed for each case. As I said, it's possible to use one earpiece or two. If two are used, there are a number of possibilities for delivering the signal. The obvious thing to do would be to deliver stereo sound which, for preference, requires the monitor console to have stereo auxiliary sends with level and pan controls, rather than having to balance two mono sends, which is tricky. But the performer might do better with a mono mix of the band plus a feed of their own vocal, panned apart to separate them aurally. This is an area that is ripe for experimentation. Whatever gets the best performance from the musicians will be the correct solution.

Feedback What would live sound be without feedback? An undiluted pleasure, many sound engineers would say. But feedback is a fact of life and engineers have to deal

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with it, monitor engineers included. As is commonly known, feedback is caused by sound from a loudspeaker entering the microphone and being amplified back to the speaker again. If there is any gain in this loop, that familiar howl will be heard by all. The problem for monitoring is that the speakers are very much closer to the microphones than are the FOH speakers. So the FOH engineer doesn't have too hard a time combating feedback, but it is the bane of the monitor engineer's life. And if the monitor engineer allows feedback to occur, the FOH engineer will want to have words, because the audience and the promoter will think he messed up. The keys to avoiding feedback are these: Place the microphone as close to the sound source as possible. If the sound source is loud, that helps. Position the speaker as far from the microphone as possible. Angle the mic away from the speaker, and the speaker away from the mic. Keep the level from the speaker as low as practical in the circumstances. Equalise out the worst-offending frequencies. Now, one of these points is clearly at odds with the other requirements of monitoring. For the purpose of avoiding feedback, it is better for Contacts the speaker to be further away. But placing www.martin-audio.com monitors further away creates a confused www.meyersound.com sound for the performers. Sorry, Mr or Ms Monitor Engineer, you're just going to have to www.mccauleysound.com deal with this. Apart from the above, if there's a www.allen-heath.com 'top tip' for avoiding feedback, it is to have a graphic equaliser for every wedge (and each www.jensen-transformers.com wedge, of course, powered by its own amplifier www.sowter.co.uk channel). Each wedge will display different feedback characteristics, even if they are www.bss.co.uk identical models, due to placement, www.canford.co.uk (EMO) microphone selection and nearby reflecting surfaces or objects. So if each wedge can be www.extremeheadphones.com optimised in this way, the risk of feedback will www.intelix.com be significantly reduced. The other thing that's worth trying is inverting the phase of the signal www.thebuttkicker.com from the microphone. Whether this works or not depends on the distance of the wedge from the mic and the worst-offending feedback frequencies. If, at the feedback frequency, the direct sound into the mic destructively interferes with the sound from the wedge, you're in luck — less risk of feedback. But if the performer wants to hand-hold the mic and move around, there's no chance of this working.

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One further problem regarding feedback in monitoring is that the monitor system and FOH system can form a link. So sound from the wedge enters the mic, comes out of the FOH speakers, bounces back into the mic, then back into the wedge. Ahh... the joys of feedback. The remedy is to apply all of the techniques mentioned above. Oh, and you can cross your fingers too. The topic of feedback naturally brings feedback suppression devices into consideration. They are not magic bullets, and sometimes even if they give a useful extra margin, the onset of feedback is even more sudden and horrifying than without the device in circuit. Yes, such devices can be useful, but not until the other techniques of feedback control are mastered and applied. At all times, the monitor engineer needs to be on guard for feedback and be ready to pull down the correct fader. He or she needs to know where the limit of feedback is for each channel, and know which ones are most at risk. Needless to say, the monitor engineer must never leave the console during the performance.

Career opportunity? The odd thing about monitor engineering is that it is more difficult than front-ofhouse engineering, yet it is the junior position. To become a front-of-house engineer, either you work with small bands and get bigger, or you work monitors for a big band first. Monitor engineering is one of the most responsible jobs in the whole of sound engineering (second only to flying a centre cluster above the lead singer's head!). It's certainly not a second-best activity, and if you can please the performers with the quality of your monitor mixes, you can be proud indeed of a job well done. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Voice Systems Eclipse

In this article:

Physical features Jargon explained Performance testing Conclusion

Voice Systems Eclipse

Voice Systems Eclipse Active PA Speakers Published in SOS December 2005 Print article : Close window

Live Sound

Pros Excellent vocal clarity. Practical size and weight. Reasonably priced.

Cons Low end becomes very coloured when pushed. Side handles uncomfortable.

The PA market isn't short of powered 'plastic' cabinets at the moment, but not many use the coaxial speaker design approach — which is what sets apart this Italian-made model.

Summary The Eclipses are a departure in design from the usual powered PA speaker and have a more natural sounding mid-range than many of their competitors. They do, however, need to be used with a sub to get the best out of them in full-range applications.

Information £499.99 per speaker including VAT. Turnkey +44 (0)20 7419 9999. www.turnkey.co.uk

Reviewed by Paul White Photographs by Mike Cameron

If it seems that a disproportionate number of 'plastic' PA speakers come from Italy, the Voice Systems Eclipse Active speakers do nothing to disabuse one of that notion, as they're built in Mondaino in that very country. Many of the PA systems we've looked at so far in SOS Live follow a fairly standard layout, but the Eclipse departs from tradition by using a coaxial speaker — one where the tweeter is located in the centre of the bass/mid cone driver. Coaxial speakers have the benefit of being more like a true point-source of sound than the more conventional split systems, where the tweeter is invariably above the cone driver, and this may make arraying multiple speakers easier, as the polar pattern of each speaker is very predictable. In this particular system, a one-inch tweeter feeds a spherical waveguide horn at the centre of a 12-inch bass/mid driver, the shape of the horn producing a 60degree x 60-degree coverage pattern. This can be a good thing or a bad thing, depending on the venue. Most horns have a rectangular flare, producing a pattern that is wider than it is high, and that can make it easier to direct the sound onto the audience, while minimising the amount that hits the ceiling. On the other hand, if the audience are very close to the speakers the high end from a typical system may go right over the heads of the first few rows, so there are benefits

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Voice Systems Eclipse

and disadvantages to both approaches.

Physical features Powering each Eclipse is a 300W amp feeding the bass/mid driver and a 100W amp feeding the tweeter. An electronic crossover operating at 2kHz splits the signal between the two power amplifiers. Little information is given regarding the type of amplifiers used, although we are told that each is fed via its own compressor, which can control levels in a transparent way, to avoid clipping. This 'Multicomp' compressor can be bypassed, but I can't see any advantage in doing so, as it offers a useful degree of protection against speaker over-excursions and audible distortion. Because the coaxial design means that there's no need for a horn flare above the main driver, the Eclipse cabinets (made from a material called HDPE) are shorter and stockier than most of the systems we've looked at. There are moulded handles on each side, as well as on the tops, and a standard polemount socket is fitted at the base of each cabinet. Recesses on the top of the cabinets allow stacking, and as the optional 12-inch active subwoofer is The coaxial speaker design means that the available in the same size of cabinet, the main boxes can be stacked directly Eclipse speakers disperse sound in the same way whether used upright or on their on top of the subs if desired, rather sides as wedge monitors. than being placed on stands. On a practical note, the side handles are less than ideally positioned for lifting the speakers onto stands single-handed, although the weight of these speakers is sensible enough, at just 21kg each. The speakers are well protected by a perforated steel grille, and there's a blue LED to show when the power is on. As is now almost obligatory, the rear of each Eclipse cabinet is angled to allow sideways mounting as a stage monitor, and this is one area in which the symmetrical dispersion pattern can be a real advantage, as the polar response is exactly the same as when the cabinet is standing 'right way up'. The rear panel of the Eclipse, as usual for the active 'plastic' systems, hosts its few controls and its heatsink. Here you'll find buttons for putting the Multicomp compressor in or out of circuit, a ground-lift switch and a 'Loud' button for scooping mid frequencies, to produce a bright, punchy sound. A rotary control sets the input gain and there's a combi XLR/jack connector that can take a balanced or unbalanced input, as well as a hard-wired link XLR of the opposite gender for feeding additional speakers. The input XLR is sensitive enough to file:///F|/SoS/SoS%2012-2005/live_eclipsepa.htm (2 of 5)11/23/2005 3:04:47 PM

Voice Systems Eclipse

directly accept a dynamic mic (30dB maximum gain), should this be necessary — a useful feature if your mixer fails or if you need a simple voice-amplification system. Status LEDs show signal present and the operation of the high and low compressors, as well as the presence of power. The last comes in via the usual IEC socket, with switch, and there's a delay circuit to prevent thumping when power is switched on or off.

Jargon explained Crossover: An electronic circuit designed to separate high– and low–frequency signals from each other so that each can be fed to speakers optimised for the role — ie. large and robust for bass, small and light (and therefore fast) for high frequencies. If a crossover is placed between the power amps and the speakers (usually built into the speakers), it is said to be 'passive'. An 'active' crossover is used to divide signals before the power amps, so separate amps are then used for each band, further adding to efficiency. Crossovers can be two–way, simply splitting highs and lows; three–way, adding a mid band; and occasionally four– way. Power compression: An unfortunate side-effect of the laws of physics. When the voice coil of a loudspeaker or compression driver is subjected to high signal levels, its temperature will rise, causing the electrical resistance of the coil to rise. If the coil gets hot enough, this can significantly affect the power transfer from the amplifier, reducing the efficiency of the speaker: 'compressing' the output. In a multi-driver system, power compression won't affect the drivers in every band equally, thereby skewing the tonal response of the system. Speaker designers have devoted years to trying to find ways to keep voice coils cool in normal operation. SPL (Sound Pressure Level): Sound level calculated in decibels compared to a reference sound pressure, commonly 20 micropascal (20 uPa), defined as the threshold of human hearing (0dB SPL). The human ear is capable of hearing an enormous range of sound pressures — the ratio of the sound pressure from the minimum up to damaging levels from even short–term exposure is more than a million — necessitating a logarithmic scale, which also corresponds roughly to our psychological perception of loudness.

Performance testing By way of performance, the manufacturers of the Eclipse (who, incidentally, also make the Viscount range of products that includes master keyboards and digital organs) claim a 50Hz to 20kHz frequency response, but this is measured at 10dB, not the usual -3dB, so don't expect earth-moving bass unless you use the optional subwoofer (around £320). There's no maximum SPL figure, which is somewhat remiss, although a sensitivity of 98dB/Watt/metre is quoted. As this takes no account of power compression at higher levels, it actually means very little, but in our field test (well, pub test, really) the speakers proved capable of plenty of level.

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Voice Systems Eclipse

Our tests involved the usual variety of acts, both electric and acoustic, and the first thing I noticed about the Eclipses was that they sounded nice and neutral on vocals and acoustic instruments, with a well-defined mid-range complimented by a smooth high end. The high-end clarity fell off a bit at a distance, but in the main they handled mid-range sound sources extremely well. At higher levels, or when faced with much in the way of bass content, the low end (by which I really mean the upper end of the bass register, at around 120 to 150Hz) became somewhat confused, with the cabinet resonance kicking in more than I'd have liked, but then if you're expecting these speakers to handle bass they really should be used with a sub. The spread of sound was also more even than I expected, with that 60-degree The simply-furnished rear panel. field I mentioned earlier, so there were no obvious dead areas in our pub-sized venue. I think we'd have managed an even better sound if we'd used a graphic equaliser to roll off the lower-mid a bit, but my own system doesn't have one so I didn't get a chance to try this. Incidentally, the sound from behind these speakers is extremely dull and boxy, so you need to be using stage monitors to have any idea of what the sound out front is like. This is to be expected, as PA speakers are designed to send the sound out to the front, not back onto the stage, but standing behind these speakers produced an even duller, more coloured sound than I remember from other systems I've tested in the same room.

Conclusion There's a lot of choice in the integrated PA-speaker market, but nevertheless the Eclipses impressed me with their good mid-range clarity, especially on vocals. The use of a coaxial tweeter may well have helped in this respect.They're also a practical size (430 x 560 x 410mm) and weight for throwing into the back of a car, although the design of the side handles leaves a little to be desired. Like any small speaker, these are a compromise, but used on their own they're very good for small-to-medium venues where the prime requirement is amplification of vocals or mid-range instruments such as guitars. Attempts to force them to handle any significant amount of bass result in a more coloured sound, so use with a subwoofer or two is highly recommended if you need fullrange coverage. A graphic equaliser could also be an advantage to help tame venue problems. However, given their mid-range price, the Eclipses turn in a very creditable performance for speakers of this type. Published in SOS December 2005

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Voice Systems Eclipse

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

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

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Yamaha EMX512SC

In this article:

Yamaha EMX512SC

Overview 12-channel powered PA Music, non-stop Published in SOS December 2005 Effects and processing Jargon explained Print article : Close window In use Live Sound Conclusion Feedback Channel Locator

mixer

Yamaha EMX512SC Pros Very light but still tough. Musical sound and exceptional effects. Flexible without being complicated. Good value for money.

Cons

Clever lightweight amplifier technology and exceptional effects, plus several other useful features, distinguish this practical and powerful live mixer. Reviewed by Paul White Photographs by Mark Ewing

Phantom power only 15 Volts. No headphone output.

In the small PA market, convenience and size are two of the most important factors influencing purchasing Summary decisions, and a lot of people are now Providing eight mic inputs is using either powered speakers with a enough for your needs, the separate mixer or passive speakers EMX512SC is a great solution driven from a powered mixer. In both to high-power amplification, cases, there's been a movement mixing and seriously good effects, in a box you can carry towards mixers with built-in effects, in one hand. which keeps setup time to a minimum. Information Yamaha's EMX512SC is just such a mixer, the largest in a new series of EMX512SC £419; three powered consoles that offers unusually good on-board effects. EMX312SC, £339; EMX212S, £269. Prices include VAT. Yamaha-Kemble +44 (0) 1908 366700. www.yamahamusic.co.uk.

Overview The 512SC has a pair of integral 500W amplifiers on board, although anyone needing less power can check out the 2x200W EMX212S or the 2x300W EMX312C. I couldn't get anyone at Yamaha to shed any light on the amplifier technology used, but given the light weight of the 512SC and the huge amount of power it produces, I'm assuming that some kind of class-D digital design is hiding under the hood. The amplifiers are rated at 500W each into 4(omega) or 350W into 8(omega), although these are maximum power ratings. No continuous power ratings are given but these will inevitably be somewhat less.

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Yamaha EMX512SC

The three new EMX models replace the EMX62M, EMX66M, EMX68S and EMX88S, and whichever one you choose you get a total of 12 input channels. Four are mono mic/line inputs and four can function either as mono mic inputs or stereo line inputs, with RCA phonos handling the stereo inputs on the last two stereo channels, as well as the main tape outputs. All channels have balanced XLR inputs, pan/balance controls and a main level control knob (rather than a slider), with a further control for the pre-fade monitor send. Channels 1-4 have mic/line switching, while the two largest models also have compressors on these channels. Weighing around 22lbs and measuring just 442 x 274 x 286mm (around 18 x 11 x 11 inches), the wedge-shaped package of the 512SC incorporates a moulded carrying handle at the rear (a rackmounting kit is available for those who prefer that format). Both Speakon and jack speaker outlets are available on the rear panel, while the mains comes in via a standard IEC connector, also on the rear panel. A sensibly quiet cooling fan is built into the amplifier section, venting through a grille in the rear panel, which also forms part of the heatsink. The physical design of the mixer isn't unique: this monitor wedge shape is now quite common, because it's practical and ergonomically suited to both front-ofhouse engineers and bands who need to adjust their own sound from the stage. A durable plastic is used for the outer shell, leaving a large panel area at the front for all the necessary controls and inputs. The mixer offers a fixed three-band EQ on each input channel, teamed with a seven-band graphic EQ (125Hz, 250Hz, 500Hz, 1kHz, 2kHz, 4kHz, 8kHz) in the master section. I couldn't find any mention in the documentation of the channel EQ frequencies or characteristics but the channel EQ sounds sweet enough, with the proviso that the mid is used mainly for cutting rather than boosting. A feature common to many powered mixers is the ability to switch the function of the two amplifiers between stereo front-of-house use and a configuration where one channel drives the front-of-house speakers in mono while the second channel drives a separate monitor system. In the case of the EMX512SC, you can also split the graphic equaliser in monitor mode, giving the opportunity to separately equalise the main (FOH) and monitor outputs. For FOH use, the lower graphic equaliser controls function in stereo, whereas in monitor mode the lower controls handle the mono FOH signal and the upper controls take care of the monitor EQ.

Music, non-stop A neat feature of the EMX512SC is a standby mode that leaves a music input feed live so that you can play background music when not performing, without having to mute all your mics individually. Note, however, that this only mutes the mono channels, so if you also have mics plugged into the stereo channels you may still need to mute those separately.

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Yamaha EMX512SC

Effects and processing Lots of mixers include built-in effects, but few are as sophisticated or good sounding as the ones found here. The reason for their high quality is that they're based on the established Yamaha SPX range of outboard signal processors. In addition to extremely good reverberation you also get echo, chorus, flanger, phaser and distortion. In all, 16 effect types are available, all with 24-bit resolution and 32-bit internal processing, plus the facility to connect an optional footswitch for bypassing the effects. To keep things simple, you call up one of 16 effects types via the rotary switch, then adjust a single main editing parameter using the knob below. In the case of the reverbs, the editable parameter is decay time, while for delays it's delay time. Personally, I'd have liked a tap tempo button for setting delay time, but you can't have everything. The amount of effect per channel is set using a post-fade send control, in the usual way, and there's also a master effects return level control, as well as a manual effects-bypass button. A separate effects output jack allows the output from the effects section to be routed separately, something you might wish to do if you need to use the effects section while making recordings. One problem I've found with other effect-equipped mixers is that very few of the reverbs have the right character for live performance — cathedrals and caverns are provided but there's often little that's usable. Yamaha haven't fallen into this trap, and although you can create those huge spaces there's also a first-rate choice of plates, rooms and chambers with variable decay time. With all controls and audio connectivity on the front panel, the 512SC is very convenient to use. The channels each have access to simple three-band EQ, but there's also a seven-band graphic equaliser (centre right) for the master. Compressors on channels 14 are controlled by a single knob, for speed of adjustment, and perform well despite their simplicity.

The compressors on each of the mono input channels are controlled by a single knob at the bottom of the channel strip. Compression can be a tremendous ally in levelling out a vocal part, but users must appreciate that for every 1dB of compression used you get 1dB closer to the threshold at which the system will feed back, so using more than minimal compression on the mic channels in smaller or acoustically difficult venues may not be a good idea. However, you can still safely use it on guitars, basses and backing tracks, if necessary. The technical spec of the 512SC is reassuringly good, with a frequency response extending from 20Hz-20 kHz (-3dB, +2dB) with THD + noise at less than 0.5 percent when the desk is working hard. Equivalent Input Noise is quoted as -115 dBu, which, although not as good as the better studio mixers, is absolutely fine for live use, where close miking is the norm. Phantom power is available, but this file:///F|/SoS/SoS%2012-2005/live_yamahaemx512sc.htm (3 of 6)11/23/2005 3:04:54 PM

Yamaha EMX512SC

is only at 15V rather than the standard 48V, and may not be enough for some capacitor mics and DI boxes. A limiter is available to protect the speakers from being driven with clipped waveforms, and Yamaha's proprietary speaker processing is built in to smooth out the high end and enhance the low end. Details on exactly how it does this are rather sketchy!

Jargon explained Capacitor microphone: Used to be called condenser microphones. In microphone applications, a thin conductive diaphragm is suspended in front of a metal disc (called the backplate) to form a capacitor which is given an electrical charge from the mic's power supply (on-board battery or phantom power). When sound causes the diaphragm to move, this changes the distance between the two surfaces and therefore the capacitance, varying the backplate voltage in proportion with the sound pressure. A built-in preamp amplifies the small signal and provides the necessary impedance matching. Because the diaphragm has no coil attached to it, it can be very light, resulting in high sensitivity and extended frequency response. Class-D amplifier: A class–D amplifier is one in which the output transistors are operated, effectively, as switches, being either fully off or fully on, with the audio signal modulating (controlling) the switching action. Class D is very efficient, allowing smaller power supplies to be used. The 'D' is sometimes mistakenly said to stand for 'digital' when in fact it was simply the next letter in the alphabetical series after Class A, Class B (and A/B) and Class C. Equivalent Input Noise (EIN): The amount of noise added to the input signal. Because the actual output noise of an amplifier stage will always vary with the amount of gain applied, 'EIN' is measured at maximum gain with an agreed industry–standard source impedance of a 150(omega) resistor standing in for the microphone. Phantom power: Standardised scheme of providing an invisible (hence 'phantom') power supply voltage to capacitor (condenser) microphones using the same cable as the balanced audio output.

In use The EMX512SC dishes out an incredible amount of power for such a ludicrously compact and lightweight unit, and though fixed EQ frequencies are quite limiting the sound quality is generally excellent, with particularly good effects. The reverbs really are great, with plenty of decay range available via the parameter knob, but I'm not convinced that distortion has any place as a PA send effect! Aside from the mid EQ, which inevitably works better for cutting than for boosting, the EQ is well-focused and musical. Even the graphic EQ sounds very smooth and natural, if used sparingly, and that's not something that can be said for every graphic equaliser! Having a built-in graphic EQ that can be split between the main and monitor feeds is definitely useful, providing you appreciate its limitations, and the same can be said of the feedback detector (see box above), which at least warns of problems and gives you a chance to turn down file:///F|/SoS/SoS%2012-2005/live_yamahaemx512sc.htm (4 of 6)11/23/2005 3:04:54 PM

Yamaha EMX512SC

the offending channel. I tried the speaker-processing button but found the result extremely subtle, lending only a slight increase in warmth and smoothing of the high end. Finally, I tested the compressors and was surprised how good they sound, especially given that they have only one control. Providing you don't push your system into feedback, they really add that studio density and polish. The settings the user doesn't have access to, such as ratio, attack and release, seem to be optimised for vocal use but also work well enough on instruments such as DI'd acoustic guitar and bass guitar.

Conclusion Yamaha's EMX512SC is a great example of just how far PA technology has come, thanks in part to innovations in efficient amplifier design. When highpowered audio amplifiers first appeared, they were incredibly heavy, yet this whole package can be carried effortlessly in one hand and it doesn't even run that hot. I can find little to dislike about the EMX512SC, although I found the lack of a headphone output a little puzzling, as it can be useful to check the monitor mix via headphones, not to mention setting up a balance when an audience is in the room without feeding the sound through the speakers. I've also made my feeling known about the low-voltage phantom powering. However, on the whole this is a superbly flexible and surprisingly affordable powered mixer for club and pub acts who have to carry their own gear. The quality of the reverbs is exceptional and comes far closer to what you'd expect from a studio processor than just about any of the competition, and virtually every aspect of the signal path is quiet and musical, even the graphic EQ. In fact, the most frustrating thing for me was not the unit itself but the paucity of technical information, as I really would like to know more about how the power amps are rated and how they work, and what frequency ranges are affected by the channel EQ. As I said at the outset, there are two ways you can go for an easy life: you can buy a powered mixer and add passive speakers or you can buy powered speakers and use them with a regular mixer, ideally with integral effects. You might feel that the powered speaker approach is more flexible, as you can always add on more speakers when you need more power, but as this mixer has line-level outputs as well as speaker outputs, you could feed additional powered speakers from there if you need more coverage. Having said that, one kilowatt goes a long way when teamed with efficient speakers! Add a couple of decent two-way passive speakers and you have a great little PA rig that should fit into the back of just about any car and delivers a very polished sound.

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Yamaha EMX512SC

Feedback Channel Locator Yamaha's new Feedback Channel Locator (FCL) indicator is featured in the EMX512SC. This feature shows which of the input channels is responsible for any feedback problem, although there's no automatic 'feedback killer' on board, which would have made the package even more attractive. (A seven-band graphic EQ such as the one on board is really far too coarse for notching out feedback-prone frequency bands, though it is very useful for overall tonal shaping to compensate for room acoustics.) Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

file:///F|/SoS/SoS%2012-2005/live_yamahaemx512sc.htm (6 of 6)11/23/2005 3:04:54 PM

Composing For Films

In this article:

Composing For Films

Early Promise Harry Gregson-Williams Back In Time Wavecrest Music Equipment Published in SOS December 2005 List Print article : Close window Doing Violence People : Artists/Engineers/Producers/Programmers Clearer Thoughts

Harry Gregson-Williams's drive to explore original ideas and sounds has made him one of Hollywood's leading composers, scoring everything from romantic comedies to spy thrillers and historical dramas. Richard Buskin

"In an odd sort of way, I think I'm aiming to perfect one composition through this entire journey," says Harry Gregson-Williams. "There are certain chord changes that move me — I don't know about anybody else, but there are ways that a melody can move from left to right that appeal to me, that attract me and let me sound like me. Why this is, who can say? You can trace things back to God knows what, but I come back to them time and time again, and the joy of working on such a variety of films is that the music's always going to come out differently." With more than 40 movie scores currently to his credit, Gregson-Williams has learned not to overPhotos: Mr Bonzai analyse the creative process. His track record includes animated blockbusters Shrek, Shrek 2, Antz and Chicken Run, a trio of Tony Scott actioners — Man On Fire, Spy Game and Enemy Of The State — and Joel Schumacher's Veronica Guerin and Phone Booth, while his most recent projects include Tony Scott's Domino, Andrew Adamson's The Chronicles Of Narnia: The Lion, The Witch & The Wardrobe for Disney, and the Ridley Scott epic Kingdom Of Heaven. The man is on a roll, and for now he's happy to proceed with the journey, wherever that should take him.

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Composing For Films

Early Promise A native of Chichester in the south of England, Harry Gregson-Williams learned to read music by the age of four and toured Europe as a choirboy with an ensemble from the music school of St. John's College in Cambridge. By the age of 13 he had been on over a dozen recordings, and he went on to study music at the Guildhall School of Music and Drama in London before spending a few years teaching music to children in England, Egypt and Kenya. Thereafter, GregsonWilliams's professional career was launched as an orchestrator and arranger for well known composer Richard Harvey, before he went on to score Nicolas Roeg's Full Body Massage and Hotel Paradise. Following his relocation to Los Angeles in 1995, he became a protégé of Oscar winner Hans Zimmer, providing additional music for films such as The Rock, Broken Arrow and Armageddon before spending the past decade involved with the aforementioned diverse list of major projects. "So," I ask, "while the time and the place in which a movie is set obviously help define the instrumentation and the arrangements, how is the music itself conceived inside Harry GregsonWilliams's head?" "Oh hell," comes the reply, "I'm often concerned that I've forgotten how I do that, and it terrifies the crap out of me! The process is always pretty identical, Harry Gregson-Williams's work area at and Veronica Guerin was a perfect Wavecrest — featuring no fewer than five example in this regard. That Joel Yamaha 02R mixers (right). Schumacher movie was alarming for me, because here I was, a skinny white English boy, being asked to score this film that was entirely set in Dublin with an Irish cast. However, rather than try to put a different hat on compositionally and think of myself as Irish, I used my own wherewithal and compositional technique, sprinkled liberally with what I perceived to be an Irish influence. The same thing happened on Kingdom Of Heaven — my process was just the same. By utilising different forces and melodies, I could duck and dive slightly differently to something that was entirely Western. "When I worked on Shrek, the director really liked the tune but he chucked it out because it didn't suit the main character. Shrek was kind of lumpy and clumsy and a bit shy and unsure of himself, and suddenly I was stuck, wondering where the hell I was going wrong when the tune was perfectly good. Well, it turned out that the orchestration that I'd chosen was the problem, so overnight I had a friend of mine — a wonderful cellist — rent a double bass and come in and play the theme on that. He was not a double-bass player, he was a cellist, so he found it quite difficult, but what came out was lumpy and a little bit halting, and there you go, Bob's your uncle. I played it for the director the next day and he thought it was great." file:///F|/SoS/SoS%2012-2005/harrygregsonwilliams.htm (2 of 9)11/23/2005 3:05:14 PM

Composing For Films

OK, but what is the compositional process that runs through all of HG-W's projects? "An intense period of calm before the storm," he responds, somewhat enigmatically. "I frequently run the film in my studio and try to pick out the nuances that might give me a clue as to where I can go with certain characters. I write at a grand piano, a Disklavier that I often turn on to record everything I do, because frequently during inspirational moments when I hit on something I don't want to stop playing, and 10 minutes later I'll be wondering what the hell I played. It's good to have method to one's madness, and I spend quite a lot of time thinking about what I'm trying to do before I actually do it." So, he doesn't fall out of bed with a tune in his head... "No, not at all. I bloody force it, banging my head with a hammer until something comes out. It's music to order. I've got a deadline and I've got to come up with it. To come up with something original, all one can do is draw on one's own experiences, so thank God I've done such a variety of work — if all I had done during the past 10 years was romantic comedies instead of all these different films, I'd probably just be repeating myself time and time and time again. I know I've got one or two decent ideas, but I'm not a bottomless pit. "I remember last year, doing Bridget Jones's Diary 2 and being very worried to begin with that all I could do was write a conventional romantic comedy score with orchestra, strings and piano. I beat myself up for months, thinking 'How can I avoid doing that? What is it that I can do that's going to give the audience the same feeling, give the director what she wants and still come away thinking that was an original thought?' In the end the director came to my rescue, because I'd written some fairly decent cues already for her, but I wouldn't orchestrate them in the way that they wanted to be orchestrated. I felt that was just too conventional. However, there was nothing wrong with leaning on convention in that score at all, and although what I did in the end didn't break down any new barriers, and no one wrote to me saying 'You've reinvented film scores,' it worked perfectly for the film, and I realised that as long as it feels right for the film and feels original to me, that's fine. "I mean, you can sit there and say 'Well, look Harry, you've never written anything original in your life,' and I'd have to disagree with you — I did recently when I finished Domino. I've never done anything like that before. There was little orchestral writing, I wrote and recorded a song with a friend, Lisbeth Scott, which was sung by Macy Gray, and it was just a totally different beast: much more eclectic and much, much more contemporary. I love that variety. "One minute I'm doing a Fatboy Slim hat and the next thing I'm doing Kingdom file:///F|/SoS/SoS%2012-2005/harrygregsonwilliams.htm (3 of 9)11/23/2005 3:05:14 PM

Composing For Films

Of Heaven or Shrek 2 or Man On Fire. They're all so very different, and I think that's the only thing that keeps me from a collision course with myself, having a problem with trying to knock out something original. If I can feel that something's original for me, that's very different to what you might perceive as original, because for me the experience is original — I've never done something based entirely in the 12th century before, or that heavily concerns religion, as was the case with Kingdom Of Heaven... I never got to work with so many choirs or with Turkish musicians before."

Back In Time Kingdom Of Heaven, co-starring Liam Neeson, Jeremy Irons, David Thewlis and Orlando Bloom, released in May 2005, focuses on a 12th-century blacksmith named Balian (Bloom) who, having lost his family in medieval France, travels to Jerusalem, gets drawn into the Crusades, tries to broker peace between the Muslims and the Christians, and strives to establish... yes, a kingdom of heaven on earth. Replete with the ingredients that appear to be de rigeur for modern-day historical epics — a sweeping narrative, spectacular battles, exotic landscapes, a cast of thousands (many of them computer-generated), plenty of heroics and just as much blood and guts — the movie also benefits greatly from GregsonWilliams's soaring musical score, a masterful blend of the new with the not-asold-as-you-think that does the job in terms of capturing the essence and conveying the feel of Dark Age intrigue. "When I joined the project, which was six months before I finished it, Ridley Scott already had a pretty good cut of the film," Gregson-Williams recalls. "It was about three and a half hours long, and that's the version I began scoring, but the film ended up being under two and a half hours, so quite a lot of music ended up on the cutting-room floor, along with quite a lot of footage that'll probably pop up on the DVD. In that kind of situation I'm often playing catch- Gregson-Williams's main sequencing tool is up, but it's all part of the process. No Steinberg's Cubase SX. one is trying to trip you up, even if it might sometimes feel like that. The film-maker's got to be allowed to make his movie. "Ridley had a very open mind about it all. He was full of encouragement and he left it open for me to discern the scenes I wanted to start with and then directed me from there. He was very clear about what he wanted to achieve, which basically related to the main character, the Orlando Bloom character, who starts off as very naive and pretty anonymous, while filmically the screen is filled with blues and whites while it is snowing constantly in northern France — very cold

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Composing For Films

and a lot of blue light. Ridley therefore asked me to make the music as chilly as the footage, and to that end I used a consort of viols, which were basically forerunners of the violin, viola and cello." Initially unfamiliar with the viol, a six-stringed instrument with a fretted fingerboard, flat-backed body and curved bow that is played underhand, GregsonWilliams came upon it during the course of his research and, despite the fact that it was popular during the 16th and 17th centuries, used it anachronistically within the 12th-century context of Kingdom Of Heaven. After all, aside from attentive music historians, who would notice this musical sleight of hand? "OK, so the viols are not absolutely authentic in terms of the date," the composer reasons, "but then neither are orchestras, and if I was only going to limit myself to 12th-century instruments, we would have ended up with pretty much a racket on screen. Viols have this rather coarse, edgy sound, played without any vibrato, and I decided they would be perfect for the beginning of the film. Look, this is a movie, and there are no prizes in Hollywood for being authentic. As long as something provides the listener and the viewer with the right feeling, that's all that matters."

Wavecrest Music Equipment List Harry Gregson-Williams's LA studio complex, Wavecrest Music, commenced operations in Venice Beach in August 2003, and comprises three floors of editing suites, mix rooms and a live room, as well as Harry's own studio on the top level. This houses the following gear: Sequencing and digital audio Steinberg Cubase SX and Ableton Live running on dual 3.6GHz Xeon server with 4GB RAM, two RME HDSP 9652 soundcards, 75GB mirrored SATA system drives and 400GB mirrored SATA audio/sample storage drives. MIDI is streamed over LAN to Gigastudio PCs, and an Emagic Unitor 8 and AMT8 are used for hardware synths. Pro Tools HD Accel system running on dual 2GHz Apple G5, with one 192 Analog and seven 192 Digital interfaces.

Samplers 14x Emu E4X. 24x Roland S760. 7x 3.0GHz Pentium 4 PCs with 80GB SATA system drives, RME HDSP 9652 soundcards and 250GB SATA sound drives with custom and commercial sound libraries, running Tascam Gigastudio 3.

Software synths and plug-ins Arturia ARP 2600, Minimoog V and CS80. Antares Filter and Auto-Tune. Applied Acoustics Lounge Lizard. Gmedia Imposcar and Oddity. GRM Tools. file:///F|/SoS/SoS%2012-2005/harrygregsonwilliams.htm (5 of 9)11/23/2005 3:05:14 PM

Composing For Films

Korg Legacy Collection. Native Instruments Absynth 3, Battery 2, FM7, Kontakt 2, Pro 53, Reaktor 4. Ohm Force Ohm Boyz. Spectrasonics Stylus RMX, Atmosphere and Trilogy. Steinberg D'Cota, X-Phraze, Virtual Guitarist, VG Electric Edition, Groove Agent and Halion.

Hardware synths Access Virus, Indigo 2 and Redback. Clavia Nord Rack, Nord 3 KB and Nord Modular. Korg Trinity Pro X, Wavestation A/D and M1R. Novation Supernova II Pro. Roland JP8080, JD990, JV1080 and MKS80. Studio Electronics SE1, ATC1 and Omega 8. Waldorf Q, Microwave II, Microwave XT and Pulse.

Recording, mixing & outboard Avalon 727 preamp. Eventide DSP4000 effects. Lexicon PCM80 and 90 reverbs. Quested HQ108 speakers in 5.1 setup. 5x Yamaha 02R digital mixers.

Doing Violence Accordingly, Gregson-Williams turned to Fretwork, a six-piece ensemble of highly acclaimed London-based musicians who, over the course of nearly 20 years, have performed English music for viol consort on radio and record, as well as in concert halls around the globe. "It's kind of odd to see and hear these stunning, world-renowned players performing on really outdated instruments," he remarks. "However, I plonked them in the middle of Abbey Road Studio One and got a crash course in how to write for viols from their leader. I mean, if I was trying to write for a flute and I was writing really low notes, a flautist might turn around and say 'I don't have that note on my instrument.' So, in this case, I had to be careful and take note of the viols' range and ensure that I wasn't writing anything that would be impossible to play. Obviously, if we were using traditional orchestrations I'd know where I was, but I'd never studied viols and so this whole exercise was pretty fascinating to me. Instead of using them by themselves I kind of surrounded them with a regular orchestra, and that seemed to work for the first act of the film. They made a glorious sound. "Ridley wanted me to musically track the movements of Orlando Bloom's file:///F|/SoS/SoS%2012-2005/harrygregsonwilliams.htm (6 of 9)11/23/2005 3:05:14 PM

Composing For Films

character. He starts off in this chilly, grubby place and arrives in Jerusalem, where the sights, sounds and smells are all new to him. You know, he touches silk for the first time, he sees beautiful women who aren't dirty, he hears instruments playing in the marketplace that he's never heard before, and so musically I was able to transition from northern Europe to more Middle Eastern sounds. And that was a wonderful thing to hang onto, because as long as I charted Balian's progress through the film, I felt that I had a secure anchor on where I should be musically. "As the action progressed I introduced these more ethnic and Middle Eastern sounds, and I spent two or three days with a small group of Turkish percussionists and singer at Abbey Road. That was interesting. Starting to write music around what I heard them do, I integrated them into the score, and then as we got deeper into the film an emotional tug took place between several of the central characters, particularly Balian and his father, who dies in the film's second act — everything that the young crusader does is for his father, and this tight Other rooms at Wavecrest include the 'blue bond between them carries us through room' (top), editing suites (middle) and a as we go towards the battle for large live area (bottom). Jerusalem, when it was necessary for me to lean more heavily on traditional orchestrations to give me the power that I needed to illustrate what was going on. I went for a lot of authentic percussion instruments and also started doing quite a lot of choral writing. After all, it was Kingdom Of Heaven, and there's a certain thing that choral music can bring that instrumental music perhaps doesn't sometimes. "I utilised two very different choirs that had two very different effects. The first one was very small, with 16 singers, called the King's Consort, performing in eight-part harmony that I wrote for the more tender and reflective moments, two on each part. That provided a very intimate, extremely church-like sound. It had to be very accurate, and they were very, very, very good singers. Then, I also hired London's Bach Choir simply because there were 120 of them, enabling 30 or 40 to sing each part, and I utilised their power and their gravitas to give a greater focus to the battle scenes later on in the film. file:///F|/SoS/SoS%2012-2005/harrygregsonwilliams.htm (7 of 9)11/23/2005 3:05:14 PM

Composing For Films

"In amongst all that I also worked with a number of different soloists and various extraneous singers whom I found — I wrote a hell of a lot of music for the movie, and that's not the sort of thing that one would want to be doing too often. It's quite a lot to take on. And technically it was very difficult for me, too. You know, I've spent 10 years over here in Hollywood, doing my thing, and when I've worked in London it's been to record the orchestra after I've written the music. In this case, however, I was required to be in London during the writing process, and that meant I had to freight my studio over there, find someone to set it up, hire a whole load of technical assistants, find somewhere to live and so on. It was quite an undertaking. "During the course of all my previous scores I had felt my way and got into a pattern, setting conditions to bring out the best in myself, and I can assure you it isn't easy to freight an intricately balanced studio over to a new location, set it up, hire a whole new support staff, and walk in and expect it all to work and sound as great as it usually would. That's what I had to do, because since it was an entirely English production I wasn't allowed to have any of my own trusty guys who have worked with me for years, as they aren't English for the most part. It was very challenging. What's more, I'd never written with Ridley before — I'd only met him a couple of times. Still, the whole exercise was fantastic, trying to integrate sounds and instrumentation that I'd previously never had the chance to do. I mean, I have scored some movies that, for one reason or another, have required me to take in the Middle East, but not ones that were set entirely in the 12th century."

Clearer Thoughts All of which brings us back to the ongoing process that Harry Gregson-Williams mentioned at the top of this interview — the unwitting journey towards the one perfect composition. "There's a score that I wrote for a tiny little independent movie called The Magic Of Marciano," he says, referring to a 2000 film written and directed by Tony Barbieri, starring Natassja Kinski. "I just hit upon a theme on that movie that I sometimes come back to and feel drawn towards — it sticks in my mind as probably the most lucid piece of composition that I've done. It's so clear and so clean, when I listen to it I can hear what I was trying to do. And I've drawn on it from time to time, as I still think that I can improve on that first idea that appeals to me. "You see, I nearly 'did it', and it's probably a good job that I didn't completely do it, otherwise I don't know what I'd be doing now — when I say 'do it', I don't necessarily mean the melody or the harmony, but a combination of the two and the way it speaks to me. There's something about it that reassures me I'm not completely in the wrong job here, and it eggs me on to do better or to complete the germ of an idea that was enough for that film but can still be expanded file:///F|/SoS/SoS%2012-2005/harrygregsonwilliams.htm (8 of 9)11/23/2005 3:05:14 PM

Composing For Films

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

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

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Leader

Leader Paul White's Leader Published in SOS December 2005 Print article : Close window

People : Industry/Music Biz

I have just returned from yet another audio trade show, brimming with new and exciting recording equipment and software, and although there were several things that had me reaching for my credit card, I've come to realise that what I really value is simplicity and audio quality rather than yet more features. For example, the automatic OS software updaters that come as part of operating systems like Mac OS X and Windows XP make it easy to update software relating to the OS itself, but what I'd really welcome is a similar system that would check on all my music software, look on the corresponding web sites and then call up the appropriate updates automatically. Naturally this would require a certain degree of co-operation between companies to make it work, but perhaps the major 'dongle' companies such as iLok could act as nodes for sending out the update requests on behalf of their customers? If this is in any way feasible, then surely it would persuade more people to use iLoks, as the simplicity of updating would be a major attraction. After all, it can take up a significant part of your life checking around various vendors' web sites to see if there are new and relevant updates, so to replace this tedium with a single button called 'check my music stuff for updates now' or similar would be a real bonus. Another area that I feel is crying out for streamlining is the sequencer, and though I wouldn't promote the removal of any features, I would venture to suggest that for anyone recording multiple audio tracks at one time, there is a very obvious page missing from all the major programs. All the mainstream sequencers have an arrange page, a mixer page, an event list and a piano-roll editor, and most offer score editing, but none that I know of offer a basic 'record page' of the type I feel is needed. By this I mean a page that offers the simplicity of a tape recorder for tracking and overdubbing, complete with meters, track arm buttons and auto-locate points. Each track's Level control should be visible on this page for setting up a basic monitor mix and a nice graphic of tape running around to show when the 'machine' is recording wouldn't go amiss either. At the tracking stage, simpler is better, so designing this page to look and behave like the software equivalent of an ADAT with gapless drop-ins (both manual and auto), would be an ideal starting point. Not only would this make the program more accessible to those musicians who inherently mistrust computers, it would also benefit the rest of us. This simple paradigm could usefully be carried over to other entry-level recording packages, because although we already have simple programs such as Garage Band and Traktion, they still look like sequencers and not like the 'Portastudios' or tape machines that most musicians started out on. As it stands, if you're recording a band, you're probably still better off tracking on hardware and then porting the session over file:///F|/SoS/SoS%2012-2005/leader.htm (1 of 2)11/23/2005 3:05:17 PM

Leader

to the computer for mixing, but I really hope that will change. Turning to the issue of recording quality, that is something that money can buy, and as with most other areas of recording, the cost is falling. If you're serious about making good-sounding music, then choosing really good audio converters is one of the best investments you can make alongside decent microphones, preamps and monitors. Basic acoustic treatment wouldn't go amiss in most home studios, either. You really have to hear the difference in quality between systems to appreciate it, but once you've heard the best, it's very difficult to make do with 'OK'. Most audio systems are sufficiently open-ended to be expanded in some way, especially those built around software, but at some point you need to take a long hard look at what you have, identify the weak links in the chain, then replace them with something better. Simply adding more bits to a mediocre system isn't going to do you any favours. Paul White Editor In Chief Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Recording Hard Rock

In this article:

Once A Maintenance Engineer... Moving To Maintenance Enter Alice New Sounds From Old Techniques Stomp Guy

Recording Hard Rock Toby Wright Published in SOS December 2005 Print article : Close window

People : Artists/Engineers/Producers/Programmers

He took an unusual and unhurried career path, but Toby Wright has helped to create some of the most influential hard rock records of the last 20 years, including Metallica's definitive ...And Justice For All, and is now one of America's most sought-after engineers and producers. Dan Daley

Patience is a virtue not in overabundant supply in the music businesses. But when it's allied to a large amount of native talent, as it is in Toby Wright, it can make the difference in forging a lasting career. It was patience that led Wright to start his career as a maintenance engineer, rather than seeking engineering gigs straight away, when he landed his first job at New York's Electric Ladyland Studios in 1979: "I figured I was going to be around for while, so I bided my time, learning and listening."

Above: Toby Wright puts the finishing touches to a mic array for vocal recording. Left to right: an original custom-made Soundelux U95, a Telefunken (not Sennheiser, though they are identical) 421, and a Shure crystal mic dating from around 1942.

It's an unusual statement, and an unusual way of looking at the world when you're 18 years old, as Wright was, when he entered the professional audio programme at the Institute for Audio Research at New York University that year. But patience would pay dividends; within a few years he was assisting on sessions for artists including Kiss, Damn Yankees, Sammy Hagar, Heart, Michael MacDonald and Cheap Trick, with producers such as Ron Nevison and Mike Klink, and two decades later would be the producer and/or engineer for records for Alice In Chains, file:///F|/SoS/SoS%2012-2005/tobywright.htm (1 of 8)11/23/2005 3:05:21 PM

Recording Hard Rock

Machine Head, Metallica, Tantric, Third Eye Blind, the Wallflowers, Fishbone, Korn, Primus, Sevendust, Slayer and 3rd Strike. It would become a career worth waiting for. While attending school, Wright and his roommate were sitting around one afternoon, plinking out songs on the guitar and drinking beer. "I remember saying to him 'This is retarded, we need jobs,'" Wright recalls. So they picked up the Manhattan phone directory, with Wright starting at 'A' in the listings for recording studios and his mate starting with 'Z', and each working towards the middle. They cold-called studios for two days. Wright got as far as the 'E's, landing a job as a runner at the famous Electric Ladyland for the grand sum of $5 a day. His friend took a bit longer to hit paydirt, working back all the way to the 'P's before getting a similar gig at Power Station. The low pay was understandable: pro audio schools were still in their infancy then, and studio owners almost universally derided them as not preparing graduates for the real world of music recording. "They tended to focus a lot on electrical theory," Wright remembers. "But we did get to work on big Neve consoles and there were a lot more Class-A electronics floating around then. There were no computers, but the Neve was fitted with one of the earliest versions of the NECAM automation, which was before Flying Faders. That prepared me pretty well to work on the Neve 8078 in Studio A at Electric Lady."

Once A Maintenance Engineer... Although he had been employed in major studios for the best part of a decade, it was not until he engineered Alice In Chains' Jar Of Flies that Toby Wright's career behind the desk really got going, and he secured management with the Lippman agency. "It was around this time that I really began to think of myself as a producer," he says. "I started letting go of maintenance." Or at least he tried to. The music business has always been big on pigeonholing and categorising people and music, and engineers are not immune to that phenomenon. When most people have known you as a maintenance engineer, you are a maintenance engineer and always shall be in many minds. "It was a hindrance at first," Wright agrees. "People can be very closed-minded about things like that. But I went back to the philosophy that got me into engineering in the first place: if I can fix it, I can work it better; if I can work it better, maybe I can be more creative with it. "Roy Thomas Baker once asked me, when I was an assistant, if I knew what an oboe sounds like. Then he asked if I knew what a tuba sounds like, and so on. Then he asked me to record an acoustic guitar, and he really liked what I came up with. What he was telling me was, if you can envision an instrument and its sound in your mind, then you can record it. Visualisation really does help in music recording. Think about it: how many people really know what an oboe sounds like? Music is a language you learn to speak. Once you do, then it doesn't make a difference if you're recording Alice In Chains or a pop band or whatever. A lot of careers have been stunted by people categorising others for only being able to do certain types of music. But when you do that, you're not only limiting them but limiting yourself. Who knows what someone is capable of? That's why I always file:///F|/SoS/SoS%2012-2005/tobywright.htm (2 of 8)11/23/2005 3:05:21 PM

Recording Hard Rock

was eager to do music that I wasn't always associated with. Right now, I'm developing a hard rock act, a country artist and a techno band."

Moving To Maintenance Early on Wright was little more than a runner — a gofer, in the studio parlance of the time. But he did get to see sessions chaired by engineers such as Neil Kernon, Hugh Padgham and Michael Frondelli. "I was realising I had a hunger for this stuff," he says. A critical point for him was a friendship with Sal Grecco, then Electric Lady's maintenance engineer (now head of maintenance at Ocean Way Studios in Nashville). "He used to come into the room screaming about incompetent assistants not knowing what the hell they were doing, and I didn't want to get yelled about like that," Wright says with a laugh. "So I figured out that if I knew how to fix the equipment, I'd be better at operating it when the time came. So I joined Sal's maintenance team and wound up doing maintenance for the next 10 years." In 1982 Wright moved to Los Angeles, driven out of New York by the city's notorious decline in the 1980s. He specifically recalls the moment he made his decision: "I was coming home at six am from a long, long session and as I'm walking along, a homeless person threw up on my shoes. And then she started laughing at me. That was the final straw." In LA, Wright had maintenance gigs at Village Recorders and One On One Studios, but was still recording music only rarely. He worked on the construction of One On One, and when the other engineer on the project left suddenly, Wright became the sole engineer at the facility, doing maintenance as well as assisting and occasionally engineering sessions. Most of those were demos, to be sure. But it was a demo session for a nowA typical Toby Wright guitar amp setup... forgotten band on Warner Bros Records Canada, Brighton Rock, that got him his first album engineering and production credit. "The manager heard the demos I'd done with them and asked me to produce the band on the spot," he says. "Even though I'd been at studios for hundreds of sessions, I had been doing maintenance and I didn't have a lot of experience actually doing music. But I figured that at least if anything broke I'd be able to fix it." Brighton Rock's record was a non-starter, and Wright went back to assisting in addition to maintenance work, working closely with producers Ron Nevison and

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Mike Klink for a while. Klink was about to start production on Metallica's ...And Justice For All LP in 1987 and chose One On One to record it. However, Klink and drummer Lars Ulrich had serious disagreements about the album's recording and Klink left the project, with Metallica's original producer, Flemming Rasmussen, coming back to direct it. Wright was in the right place at the right time. ...And Justice For All would go on to become one of the sonic signature records of the 1980s, entering the Billboard Top Ten with virtually no radio or video support, despite almost every song being an extended opus of mixed time signatures and odd sounds, including the unlikely Top 40 single 'One'. Considering the recording approach, one can see why it's considered such a unique album. "Lars wanted to do a record that required very, very heavy editing," Wright explains. "Flemming had always been a proponent of hitting the drums very, very, very hard. He said that that's where you get the sound out of them: sheer brute force and volume. The harder you hit it, the better it sounded. Now Lars is very well built and muscular, but at this level of velocity he couldn't play for more than a minute or so at a time. So we recorded in bits and pieces. None of the songs on that record were recorded all the way through in one pass. We ran a click track and [guitarist] James Hetfield played along to it, laying down a scratch track of rhythm guitar. Then Lars went out and played drums to that, stopping each time he got tired and then picking it up again after he rested up. All of those snippets of songs were edited together by me later. It was a pretty remarkable way to make a record, and I can see why perhaps Mike Klink wasn't nuts about it. But I was up for anything. To me, it was like fixing things. You have a bunch of parts and you put them all together and you make the thing work." Wright was also up for flouting convention. Working with Ulrich, he spent days experimenting with drum-miking techniques and combinations to discover the elusive power Ulrich and Rasmussen were seeking. "In the end, it was all microphone placement and EQ," Wright comments. "I put a 412 and a D112 inside the 24-inch kick drum and ran them through a Technics DN360 31-band graphic equaliser. That was the whole signal chain. The EQ was pretty twisted — lots of bottom and top and no middle. Lars was pounding the drums; after one tom roll I'd have to go out and change the heads. I still have those drum heads, signed by everyone in the band."

Enter Alice ...And Justice For All was a big hit, but Wright went back to assisting, working with Nevison again on projects for Heart and Ted Nugent's Damn Yankees. But more than a decade in studios had also introduced him to some of LA's A&R contingent, and the rep for Alice In Chains decided to pair the band with Wright for two songs on the 1993 Arnold Schwartzenegger film Last Action Hero. Wright went on to engineer the next record release, Jar Of Flies, the first EP to ever debut at number one on the charts, and whose seven songs were cut in 10 days.

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Recording Hard Rock

"They called from the road during Lollapalooza that they had lots of new songs and were ready to go back into the studio," Wright remembers. "We had 10 days booked at London Bridge Studios in Seattle. We got there and I asked [guitarist] Jerry [Cantrell] to play them for me. He looks at me and smiles and says 'That's funny — we don't have any songs. Do you mind if we just jam for the next 10 days?' I said sure — I didn't have anything else to do. I figured, I get to listen to a bunch of great musicians play around for a week. Incredibly, they began writing as they were playing. On the eleventh day, I began mastering what would become Jar Of Flies."

New Sounds From Old Techniques The 1990s were explosive for Wright. He produced Korn's Follow The Leader with the band, having been promoted from engineer after initial producer Steve Thompson departed the project. "This was a very experimental project, sonically," Wright recalls. "They wanted very much to get sounds that hadn't already been used on records before. They thought I had the technology background, from fixing such a range of stuff, and was open-minded enough to let them get pretty far out there. We wanted to make sounds without using plugins on Pro Tools. "One sound in particular, on 'Freak On A Leash', was interesting and kind of illustrated what we were doing on that record. There's this whirring guitar sound, which is a guitar played through a Pignose amplifier placed in front of your garden-variety house fan. I put a Shure 57 and a Sennheiser 421 on the other side of the fan from the Pignose, some dynamic mics with pop filters on them to take the edge off the moving air from the fan. It took a while to get Another way to get a novel guitar sound: the the right position for the mics, but it mic is taped to a barbeque grille and a miniature amp is rested on it. was similar to what you go through when miking a Leslie cabinet — you're looking for the spot where you get the most sound but the least amount of useless moving air. Oh, and the fan was set on low. "That kind of set the tone for the rest of the record. We were constantly looking for ways to get new sounds without using processing, just using microphones and placement techniques. When I was at NYU, I took a course in acoustics. That has really helped me understand the nature of how sound moves around in a space, and has been a big part of how I approach microphone choice and placement." Sevendust's second album, Home, was recorded by Wright at the residential file:///F|/SoS/SoS%2012-2005/tobywright.htm (5 of 8)11/23/2005 3:05:21 PM

Recording Hard Rock

studio Longview Farm, in Massachusetts. There, he had a lot of space to play around with. "There's about 50,000 square feet of space in the barn, the main recording area, which was built in the 1840s," Wright explains. "The building is old and wonderful and it acts like a huge violin in terms of how it reacts to sound and volume. When you play loud, the entire structure vibrates and resonates, and it becomes part of the music, and on Home we used a lot of it. If you listen closely to the guitar tracks, you can hear it. I had the band play for a while I went around the room looking for the frequencies at which the walls resonated most and where the vibrations were the most intense. The I took some PZM microphones and put them on the walls, covering them so they didn't pick up any direct reflections. Soloed, [those tracks] sound kind of gargly, but when you mix them into the sound, it adds this dimension that's incredible."

Stomp Guy This is good point at which to ask Wright about how he approaches effects. A dedicated analogue tape fanatic — he always records basic tracks to a Studer A800 MkIII with 16-track heads running Quantegy tape — he likes to think ahead. "If I'm going to be mixing the record, I'm less inclined to print effects," he says. "Because I know I'll be able to restore the sound we had originally envisioned in the mix stage. If I'm not mixing, I'll print effects to another track and make notes to whoever is mixing that they're there and what they're for. But I love effects. I'm known as the guy who will stomp on the flanger pedal for you during a take." Recording Jakob Dylan and the Wallflowers with producer T-Bone Burnett was an exercise in minimalist effects. "Everything was organic," Wright recalls. One Headlight was done at Jackson Browne's studio and Sunset Sound. "T-Bone said he wanted very classic sounds on this album," Wright says. "Classic drums and guitar sounds, using the kinds of microphones and placements I learned years ago at Electric Lady. When you hear a B3 sound on this record, it's a B3, not a B3 patch on a DX7. The sounds are all genuine, and that really helps push the music along. This is the kind of stuff I learned from guys like Ron Nevison and Eddie Kramer. The band was recorded ensemble, no overdubbing of basic track parts. And the musicians you use make a huge difference. We had Matt Chamberlain on drums for that record and he is a one-take drummer. If he plays it again, it might sound a bit different, but it'll be just as good as the previous take. A guy like that in the band puts a very good edge on the entire band and the entire record." Wright has a basic setup he likes for drums: a Shure 57 and AKG 451 taped together on a single stand on the top of the snare, with a Sennheiser 441 on the bottom snare head. There'll be a small condenser microphone, like a Neumann KM84, on the hi-hat, AKG C414s as stereo overheads, a combination of a Sennheiser 421 and an AKG D112 on the kick drum, and 421s on the toms. But this is not written in stone, and he's willing to adapt at a moment's notice. "On Sevendust we used about six different drum placement setups," he says. "We'd file:///F|/SoS/SoS%2012-2005/tobywright.htm (6 of 8)11/23/2005 3:05:21 PM

Recording Hard Rock

move the kit into different rooms, too. The kitchen there gives this great smallcombo jazz sound to the kit." His basic recording methodology is also adaptable. While he's a dyed-inthe-wool analogue fanatic, Wright generally takes the basic tracks and moves them to one of the three Pro Tools systems he has for editing. He'll re-record many of the parts on Pro More creative engineering from Wright: the Tools, but will fly them back to tape guitar amp is directed at the piano's before mixing. "If you do it all in Pro soundboard in order to record the resonance Tools, you'll get this unhumanly tight of the strings. performance from the band," he cautions. "If you use Pro Tools at the front end of a session, you'll lose the warmth. But it's amazing for editing, and tape simply adds too much to the cost of a record these days. Sometimes I might go straight to [Pro Tools] HD, depending on the type of session, but I'll always go to tape before mixing in any case with the basic tracks, and then mix from both tape and Pro Tools locked together with a Lynx synchroniser." Wright owns a Neve 8081 console and is partial to studios that have Class-A equipment like the Trident A-range modules, Pultec EQs and other vintage goodies he keeps in the six 24-space racks that travel with him. He likes to keep the old gear close at hand and is comfortable with it, just as a master mechanic is unperturbed driving a 1958 C-type Jaguar simply because he's also capable of fixing it. He mixes to Yamaha NS10M monitors — no tissue paper over the tweeters. Wright has one other new project, and it's not a band. Mixlab is a software package he developed with a codewriter friend who spent months analysing where Wright puts elements in the surround soundfield, such as the 18-track video DVD he completely remixed from the original tracks for Alice In Chains. The resulting algorithms will, Wright asserts, do the surround panning placement automatically for a record. The user enters various bits of information into fields on the screen — this is a rhythm guitar here, drums on these tracks, here's a vocal, and so on — and after processing the entered information plus the audio data, the software will create a surround mix field using the console's panning structure. "It'll go through about 2.5 million possibilities from 24 tracks and figure out where everything makes the most sense," he explains. "You can also adjust the preset algorithms to have it create panning templates suited to your particular way of doing surround mixes. It speeds up the surround mix process, which I hope will encourage more people to do surround mixes. I used it to do the surround mixes on [Phish guitarist] Trey Anastasio's album. It took me 16 days to do the stereo mix at Electric Lady; it took me four days to do the surround mixes. That's fast." Speed doesn't seem to be a primary characteristic of Toby Wright, though. After

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Recording Hard Rock

putting in nearly a decade in the solder-filled back rooms of studios, he knew his day would come and he didn't rush it. The accumulated wisdom of all that time waiting was worth the time it took to acquire. As he says, "When you think about it in the long run, what's the rush?" Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

file:///F|/SoS/SoS%2012-2005/tobywright.htm (8 of 8)11/23/2005 3:05:21 PM

Sounding Off

In this article:

About The Author

Sounding Off Roger Thomas Published in SOS December 2005 Print article : Close window

People : Sounding Off

Won't somebody please think of the audio equipment? Roger Thomas

I've just acquired a pair of Gemini DJX headphones, and I'm very pleased with them for all sorts of reasons. Value, comfort, durability and the kind of sound I like all figure in my assessment, but quite apart from all this, guess what? Trivial as this may seem, they came with what is to me one of the most important yet least available of accessories: a bag. Said bag's nothing special, being nothing more than a drawstring thingy made from soft fabric, but with the cans folded and with the cable wrapped around them I can at least stuff them into it in the knowledge that they'll be protected from knocks and scratches and that the cable won't unwrap itself until the next time I need them. More importantly, I can now chuck them into a larger case full of other stuff with relative impunity. I use an assortment of gear in live situations, and given that a great deal of equipment originally conceived for studio use is now sufficiently small and portable for live deployment, I deploy it, as do many others. Why, then, can't this be reflected in the availability of simple bags and carrying cases for all those sequencers, effects units, controllers, small sound modules and other nonrackmountable widgets that otherwise make portable music technology such fun? Full-sized rack units can at least go into a padded rack case and there are various cases for keyboards and DJ equipment, but that's about it. Over the years I'm come across a few instances where file:///F|/SoS/SoS%2012-2005/soundingoff.htm (1 of 3)11/23/2005 3:05:24 PM

About The Author Roger Thomas is an author, journalist, lecturer and musician in variable proportions. If you would like to air your views in this column, please send your submissions to soundingoff@ soundonsound. com or to the postal address listed in the front of the magazine.

Sounding Off

manufacturers realised this. I bought a Yamaha QY10 when it first came out, and, lo and behold, it came with a little plastic case that fitted it exactly. I've also got a couple of those ubiquitous, cheap mini-mixers, sold under the Soundlab brand among many others, each of which has its own dinky rigid case. Other than these and a very few similar examples, though, I've been forced to use all sorts of weird and wonderful bags, cases and pouches in an effort to protect bits of gear that were otherwise clearly designed with portability in mind. Multitrackers and small mixers? Laptop bags from Maplin. Drum machines and other odd rectangular devices? Zip-up toiletries bags. Microphones? Pencil cases from WH Smith. My Roland drum pads? A briefcaseshaped tool case. My multi-channel headphones and their amplifier? A small padded drum bag. Yamaha's insight seemed to fail them when they brought out the QY-sized MU15 sound module without a case, so that ended up living in a double VHS box. I also find myself ransacking camera shops, DIY stores, Oxfam shops and even office suppliers hunting for that elusive case/bag/cover in whatever shape and size I need. I'm currently looking out for something that will accommodate an Evolution UC16 MIDI controller, its tough metal casing rendered redundant by its 16 unrecessed fragile plastic knobs and its otherwise user-friendly oblong shape. Now, no doubt most users of all these more-or-less delicate bits of kit just leave them safely in the studio, but I'm sure I'm speaking for a pretty sizeable minority who don't. Only recently I've been called upon to set up a multi-keyboard MIDI rig in a local concert hall, provide some Latin percussion in such a small space that digital was the only answer and sort out PA and background music at another event. All of the equipment involved requires some sort of protection so that it can be shunted around safely, as things are far more likely to get damaged in transit than in use. How often do we find ourselves carting stuff around in its original packaging of tatty cardboard and crumbling polystyrene, which is not only scruffy and awkward but which positively bellows 'steal me!' to any passing miscreant? Could I therefore gently suggest to manufacturers that they consider providing even the most basic of bags or cases for all those vulnerable small items of equipment we can't do without? The cost to the manufacturers would be minimal, but I'd personally be quite happy to see an extra couple of quid added to RRPs if only to provide soft drawstring bags like the one that came with my headphones, as these can still make the difference between damage and no damage if the thing takes a wallop. Furthermore, aren't we more likely to upgrade more often if we're confident that we'll get reasonable prices for our undamaged used gear? I'd have thought so. It's got to the point where I'm seriously thinking of enlisting the aid of a competent craftsperson and knocking out a range of suitable bags myself, so if you see someone leaving a newsagent carrying the latest Sound On Sound and a pile of needlework magazines, it'll be me, still wondering what to do with my UC16. Published in SOS December 2005

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Sounding Off

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

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

file:///F|/SoS/SoS%2012-2005/soundingoff.htm (3 of 3)11/23/2005 3:05:24 PM

Studio SOS

In this article:

Studio SOS

Rewiring & Reorganisation Control-room Bass Problems Dave Rogers Published in SOS December 2005 Acoustic Treatment Dave's Comments Print article : Close window Sequencer Tips

People : Studio SOS

Dave Rogers was having trouble with his monitoring, so the SOS team sped over to his home studio in Bristol to sort things out. Paul White & Hugh Robjohns

Dave Rogers left his native Boston on the east coast of the US to live in Bristol, where he has set up a home studio in a small room created by partitioning off the last eight feet of his single garage. This has left him with a near-cuboid space measuring approximately eight by eight by nine feet, where three of the walls are solid and one is a studding 'drywall' partition incorporating a doorway to the rest of the garage. A side door in one of the solid walls provides access. Dave's recording system is based around a Mac Powerbook laptop fitted with 1GB of RAM and connected to the outside world via an M Audio Firewire 410 audio interface that provides one set of MIDI ports. The active monitors are also from M Audio (BX8s) and Dave had these set up on piles of heavy concrete building blocks, a cheap alternative to speaker stands which Paul suggested to him on the phone while arranging the visit. He also has a Novation rackmount synth, a Micro Korg synth and a couple of processor file:///F|/SoS/SoS%2012-2005/studiosos.htm (1 of 7)11/23/2005 3:05:27 PM

Dave had improvised speaker stands using piles of concrete building blocks. However, his original design (left) was rather unstable, so Paul rearranged it into a more stable configuration (right), and added some

Studio SOS

Auralex Mo Pad foam isolation pads for boxes, including a Dbx 266XL compressor, which he had connected to good measure. the aux send and returns of the small Behringer Eurorack MX2004A mixer he was using to feed his synths into the audio interface.

When we arrived, the room was completely untreated, so the acoustic was predictably boxy and ringy, given that there was only the floor covering and the equipment in the room to absorb and scatter the sound energy. We played some commercial tracks over the speakers, and were greeted by a very confused sound with exaggerated presence and high end, no real stereo imaging, and a lot of overhang on the bass notes. As is common with small rooms, the bass end dipped considerably if you listened from the dead centre of the room, which is where Dave's chair normally sat. He'd noticed something was wrong, because his headphone mix always seemed to have a lot more going on at the low end than appeared in his monitors. As usual, we started by trying to optimise the monitoring system prior to treating the room, and it was immediately obvious that the stacked concrete blocks were far from rigid, so we dismantled them and built them up again as shown in the photograph, with two blocks one way, then two the other, capped off with flat blocks on top to form a platform. This was far more rigid, and once we'd added a pair of Auralex's dense foam Mo Pads the speakers were at the correct height with the tweeters pointing directly at Dave's head when he was seated in his monitoring chair. Hugh then checked the EQ settings on the backs of the speakers, which were set flat except for the mid-range presence switch. He turned this off and also dropped the tweeter level by 2dB, as one characteristic of these speakers seems to be an over-pronounced high end. He also reduced the bass extension from the lowest 37Hz setting to 47Hz, which initially seemed more appropriate for the small room.

Rewiring & Reorganisation The result was a slight improvement, but at this stage the room acoustics were still playing havoc with the monitoring. We also noticed that every time David needed to replay anything he had to do a lot of re-patching, so we suggested reorganising his system slightly before proceeding with the acoustic treatment. With small systems such as this, a simple option is to feed the stereo output from the audio interface into the two-track returns on the mixer and then select this two-track return as the mixer's monitoring source. The advantage of this approach is that the mixer's control-room monitor level control can be used to adjust the monitors, whereas previously Dave had fed the monitors directly from the output of his audio interface, which meant that he had to adjust the playback volumes in software. The input channels could then be mixed as normal and fed to the audio interface inputs via the mixer's main stereo output. file:///F|/SoS/SoS%2012-2005/studiosos.htm (2 of 7)11/23/2005 3:05:27 PM

Studio SOS

Dave has his synths connected to the mixer, but he also has a DJ deck setup that could be fed into the mixer for monitoring or recording purposes. For playback or keyboard playing without the computer, the mixer could be switched back to monitoring the main stereo mix rather than the two-track input. We rewired as above using Hugh switched off the mid-range boost Dave's existing cables, all of which setting on the M Audio active monitors, and were unbalanced. There was little hum also tamed their over-prominent high-end or noise in evidence, but using response by reducing the tweeter level a little. balanced cables to feed the active M Audio monitors would have been better. The two-track inputs were on unbalanced phonos anyway, so the existing cables were fine. Next we explained that compressors aren't normally used in effects sendreturn loops unless you're doing certain unconventional mastering tricks, and as Dave was using Logic Pro 7 on his laptop he could always compress within the software using Logic's included plug-ins. However, he liked the sound of the Dbx on certain basssynth sounds, so Paul suggested that he hard-wire the Dbx compressor between his Novation synth (which is the one he usually uses for bass) and the mixer input.

Dave had been monitoring his mixes directly from his audio interface output, but this meant that he needed to re-patch to play anything else back over his studio monitors. By rewiring the audio interface outputs to the mixer's two-track inputs instead, Hugh was able to set up much more flexible monitoring, making better use of the mixer's routing facilities.

One odd occurrence Dave demonstrated to us was that whenever he booted up the computer, the stereo image was obviously offset to one side. This turned out to be due to Audio MIDI Setup in the Mac OS turning down one channel by 2-3dB for no apparent reason. Dave had even gone so far as to completely reinstall the OS, but the problem still remained, and he tells us that it only started after he installed the M Audio Firewire 410. Until a solution is found, he has to go into Audio MIDI Setup and reset the faders every time he starts up the computer.

Control-room Bass Problems After an eclectic lunch of chocolate Hobnobs augmented by real chip-shop fish

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Studio SOS

and chips, we turned our attention to the sound of the room itself. Being realistic, a small room like the one Dave has is never going to make an ideal monitoring environment, but we felt that it should at least allow mixes to be created that don't sound too far out of balance when played on other systems. As we knew we were going to face bass-end problems in such a small room (the smaller the room, the more widely spaced the room modes, and the less even the bass will be), we brought along a couple of Mini Traps kindly donated by Ethan Winer of Real Traps. Ideally you'd need four or more of these traps to really make a serious dent in the problem, but Dave agreed that if he could hear an improvement with two fitted, he'd buy two more to fit himself at a later date. Mini Traps work on a different principle to the acoustic foam panels that Fitting the Real Traps bass traps was a Auralex normally provide for our Studio simple matter of fixing nylon cord through SOS visits: acoustic foam absorbs the holes on the back of each trap's metal sound by frictional loss, and its frame, and then hanging the traps from hooks which had been screwed through the effectiveness at low frequencies is plasterboard into the ceiling joists. directly related to its thickness and its spacing from the wall. On the other hand, the Mini Traps use the principle of lightweight panels between which are sandwiched three inches of damping material. These traps are then fitted across corners (usually between walls and ceilings), the idea being that the bass energy tries to move the panels and the damping material behind absorbs the energy rather than reflecting it back into the room. The easiest way to hang these panels is by using picture-frame wire or nylon cord, and we used the latter for simplicity. However, getting the cord length right to get the panels hanging correctly proved to be quite tricky. We also had to locate the ceiling joists to screw in the hooks needed to support the cord. However, once the panels were up, they looked fine and seemed stable. Another listening test showed that the bass dip in the centre of the room was noticeably less fierce and the bass notes didn't appear to overhang as much. Equally importantly, the levels of bass notes were now more even, though still not perfect. Two further traps would improve this situation, but as it was, the bass performance of the room seemed adequate, provided that any crucial monitoring decisions were made with the chair pushed back behind the centre of the room.

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Studio SOS

Acoustic Treatment Next we used four panels of Auralex foam to tame the remaining ringing and boxiness, following the usual strategy of placing absorbers at each side of the listening position to damp side-toside reflections — these are the ones that have the most detrimental effect on stereo imaging. A further panel was placed on the rear wall to reduce reflections from that surface, and the remaining panel was cut in two and glued to the wall behind the monitors. Ideally we'd also have placed a panel on the ceiling centred just forward of the monitoring position, but we didn't have enough material with us to do this. Dave said that he'd buy a panel as soon as possible and fix this himself. In order to tidy up the monitoring system's stereo imaging, Paul and Dave used contact adhesive to glue acoustic foam panels around the side walls of the room, thereby reducing acoustic reflections reaching the mixing position.

Once the acoustic foam was in place, the difference was immediately obvious. Just speaking in the room showed that it was much better damped, and all that 'broom cupboard' boxiness was gone. Playing the same material over the monitors confirmed that everything now sounded less confused, more controlled, and tighter, with greatly improved imaging. There was still some bass drop-off at the exact centre of the room, but nothing as bad as what we'd heard before putting up the Mini Traps. The consistency of level across different bass notes was also better. Dave seemed very pleased with the sonic improvements, and also with the ergonomic benefits of rewiring the mixer to allow it to control the monitor level. During the day, we'd tried using the monitors set to a higher bass roll-off position, but with all the treatment work complete, we found we got a better correlation between the monitor mix and the headphone mix if we set the speakers back to their full bass extension. We stuck with the flat mid-range setting and the 2dB high cut, and left the environment switch set for half-space operation, which is usually best when the speakers are used close to a wall. For judging mixes, it would still be better to roll the seat backwards a foot or so away from the dead centre of the room, but that's easy enough.

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Dave's Comments "First I would like to say thank you to the team — this has been one of the highlights of my life! I never thought about how the sound would change or even how to go about changing it, but the room now sounds fantastic. The Bass traps do a proper job, and I have already been looking into getting at least two more for the room. I have also taken your advice and changed most of my unbalanced cables to balanced, as well as hard-wiring the compressor. "I feel much more confident with my mixes now. I know where to put the guitar amp and where to sit for mixdown. The Screenset tips you gave me make getting around Logic quick and easy. I love going into the studio with confidence that it's all good."

Sequencer Tips We used up our remaining time going through a few Logic tricks with Dave. He's a big fan of synth bass sounds, so we tried processing some of his patches using Logic's Guitar Amp Pro and Phase Distortion plug-ins, which Paul has found to be very effective in getting dirty Leftfield-type heavy bass sounds. The guitar amp can produce more subtle effects, and by choosing the right speaker-cabinet simulation it's often possible to create more analogue-like sounds. Not only does this sound good, but it also tames the high-frequency harmonics in the sound, so it's often easier to place the bass line in the mix without it trampling all over the mid-range. Paul also demonstrated using Logic's Tremolo plug-in as a beatsync'ed chopper for turning pads or other sustained sounds into rhythmically supportive elements. Dave was fairly new to Logic, so we also covered a couple of basic operation tips, the first and most obvious being to spend the time to create a comprehensive default Song with Screensets arranged to navigate quickly between the most commonly used screens, such as the Arrange window, the Track Mixer, and the Environment. The use of Screensets is particularly important when using a laptop, because of the very limited screen space. However, Screensets are saved with each Song, so you need to work them into your default

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To wrap things up, Paul demonstrated how Dave could make Logic quicker to use by spending some time setting up a suitable default project, with all the MIDI routing, audio tracks, and mixer channels he might need.

Studio SOS

Song if you want them to appear in new Songs you create. Another great space-saver for laptop users is to use the Arrange window's View Channel Strip Only Key Command, so that you can easily toggle between seeing the whole channel strip on the left of the window or all the parameter windows plus just a part of the channel strip. Our final tip was to use the Escape key to bring up the toolbox at the current cursor position, as this saves having to move over to the left of the screen every time you wish to select a new tool. Hitting Escape a further couple of times brings you back to the original tool. Paul also recommended Dave get the version 7.1 upgrade, as it includes a few further improvements. Upgrading Mac OS to Tiger would probably also be a good move as well in his experience, as Logic is much more stable under Tiger than under the previous versions of Mac OS X. Before we left, Dave asked our advice on improving the entrance door. As daylight was clearly visible around the edge, and it was a lightweight door, it leaked quite a lot of sound. We suggested adding a layer of half-inch plywood to the inside, with Rockwool filling any spaces, then fitting a compression latch to pull the door hard against the frame when closed. The door frame could then be modified with neoprene foam seals that touch the door on all four sides to make it airtight. While a single door will never be as good as two spaced doors for isolating sound, the difference should be very worthwhile. It may also be worth damping the door leading through to the garage, as the light pine panels may resonate when hit with high levels of bass. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Advanced Timing Correction In Pro Tools

In this article:

Creating A Tempo Map Sharpening Up Pro Tools News A Different Problem

Advanced Timing Correction In Pro Tools Pro Tools Notes & Technique Published in SOS December 2005 Print article : Close window

Technique : Pro Tools Notes

We all know that Beat Detective can be used to fix up dodgy drumming. But how about creating a tempo map from a freely played keyboard part? Or replacing a piano track with note-for-note accuracy? You can achieve amazing results when you know how... Mike Thornton

I have now worked on numerous tracks that would at best have been a nightmare, and at worst wouldn't have made it to the client's CD, without the help of Pro Tools's Beat Detective feature. Its most obvious use is taking freely recorded drum parts and making them conform to a grid or some other timing reference, but it can also be used on all sorts of other material, and this month I'm going to share some of the more advanced tips and experiences I have gained using Beat Detective in anger. The good news is that everything I am going to describe is possible on both TDM and LE versions of Pro Tools: Beat Detective used to be TDM-only, but most of its features were made available on the LE version from v6.7 onwards. The first example I'm going to use came from a client's interpretation of the song 'Breath Of Heaven' (made famous by Amy Grant). Initially, I

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Here, I've selected four bars of MIDI for Beat Detective to generate a tempo map from.

Beat Detective shows me where it thinks the bars and beats fall within the selection. Adjusting the Sensitivity control helps to get it right.

Advanced Timing Correction In Pro Tools

recorded the bass guitar, keyboards as both audio and MIDI and a guide vocal in one pass. We didn't want to use a click, as the musicians wanted to be able to 'feel' the timing and incorporate ritandos and so forth, so we laid down a couple of takes straight into Pro Tools until we were basically happy with the feel of it. Then we patched up a few keyboard mistakes and I sent the musicians home. I recorded the MIDI output from the keyboard as well as the audio, as I thought it might help in the next phase of the process. This was to create a tempo map of the live playing so I could get Pro Tools to create a click track ready for the following day, when we would be recording a 13-piece string section layered up three times.

Analysing the audio output from my keyboard, rather than the MIDI notes, produces slightly different results.

Creating A Tempo Map Beat Detective has been designed to be extremely 'intelligent' and so isn't easily fooled, as we will see. It is best to put it to work on small sections, rather than a whole song at a time, especially if there are large changes in tempo. I tend to work in four-bar sections unless there is a particularly easy section, in which case I will jump up to eight; for particularly complex sections I'll go down to two or even one bar at a time, which doesn't make for a very fast process but does make for a very accurate tempo map. I began by working on the MIDI part: as you can see in the screen shot, I have selected the first four bars on the MIDI track and opened the Beat Detective window (Windows / Show Beat Detective or Command+8 on the numeric pad on a Mac or Ctrl+8 on a Windows system). I then enter the start and end bar and beat numbers as well as time signature in the Selection file:///F|/SoS/SoS%2012-2005/protoolsnotes.htm (2 of 7)11/23/2005 3:05:36 PM

Beat Detective doesn't cope very well with ritandos in the middle of a section (above); changing the selection so that the ritando is at the end enables it to get things right (below).

Don't expect Beat Detective to deal with changes of time signatures within a section.

Advanced Timing Correction In Pro Tools

section in the middle, set MIDI (in this case as we are analysing a MIDI track) from the drop-down menu in the Operation section of the Beat Detective window and then click the Analyse button. As I adjust the Sensitivity control, bar and beat lines appear, and some of the lines will move as the Sensitivity is increased (see screen overleaf). To make sure Beat Detective is putting the bar and beat lines in the correct places, it's much easier if you either Above and left: This particular section is a know the piece very well, or have a problem for Beat Detective: it can't find the copy of the music, so you can see correct bar and beat positions. However, it's what the notes are and so where the possible to use the Grabber tool on the audio they should be. Having satisfied track to move the bar and beat lines to the correct positions. yourself that they are correct, click on the Generate button. If there are any tick-based tracks (which includes all MIDI tracks) Pro Tools will ask you whether you want them to move or not. In this case I don't want the tick-based tracks to move, as I want the MIDI track to remain in sync with the audio tracks. Click OK to dismiss the window and you will see that Pro Tools will have inserted various tempo changes and the correct bar lines in the timeline section of the window. Other functions also affect the way Beat Detective interprets the selection: changing the Contains drop-down menu to quarter notes instead of eighth notes changes how Beat Detective 'sees' this section and in this case brings up errors. Changing the Analysis drop-down menu to other options like Loudest Note can also produce errors. Make sure you click the Analyse button again when you have changed some of these settings to update Beat Detective's analysis of the selection. Now let's take a look at the same selection, but this time analyse the keyboard's audio output. This time, I have Audio selected in the Operation section of the Beat Detective window (right); note that the Analysis drop-down menu now offers High and Low Emphasis. You should normally use High Emphasis unless you are analysing low-frequency audio like a kick drum or bass guitar track. If you compare the results of analysing the audio and the MIDI (top), they are very similar but not exactly the same. I found that there were some phrases where the analysis of the MIDI was better and other phrases where the reverse was true.

Sharpening Up There are a number of additional things you can do to ensure that you get the most accurate analysis: file:///F|/SoS/SoS%2012-2005/protoolsnotes.htm (3 of 7)11/23/2005 3:05:36 PM

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Make sure that when you highlight a selection, you highlight from just before the first note of the selection and highlight on past to just beyond the first note in what will be your next selection. This helps Beat Detective to know where the exact starts and finishes are and is especially important during ritandos, where the tempo can slow up a lot. You can use the Capture Selection button if the selection is close to the correct number of bars, but again, if the selection includes a ritando, be ready to change the end point; make sure you press the Enter key after changing it or Beat Detective won't recognise the new end point. Don't just depend on your eyes. Having made the selection, play it to make sure you have the correct selection. It is most important you don't make a mistake here as any error will mean everything after it in the song will be wrong! Always be extra sure you have the correct selection, and that Beat Detective analyses it correctly, before hitting the Generate button. To my cost, I've found that it is not possible to go back and simply correct the error, as this doesn't fix all the 'knock on' effects. You will have to redo the entire analysis from the point where you made the mistake. Don't have a ritando in the middle of a selection, as the chances are that Beat Detective won't be able to make sense of it. In the penultimate screen on the previous page, bar 31 should be the chord in the middle of the section, but it can't work it out! As soon as I select only two bars, so the end of the selection is also the end of the ritando, Beat Detective is able to resolve it perfectly. The time signature is important. In this piece there is a four-bar selection. At bar 52 there is one 3/2 bar, then two 4/4 bars; then bar 56 is another 3/2 bar. Beat Detective is totally confused with the time signature set to 4/4, as the screen (top) shows. However, if I set Beat Detective to analyse just bar 52 and tell it to expect a 3/2 bar, hey presto, it works! There are times when you need to give Beat Detective manual help. Take a look at the middle of the three screens to the right. It is a two-bar piano selection. I had first tried a four-bar selection and it wasn't working. It was putting all the notes in the wrong place. Next, I tried the same two-bar selection on the audio track instead, and although it could find the bar lines fine, it still couldn't work out where to put the beats. In these situations, if you are working on an audio track, you can get hold of the beat marks with the Grabber Tool and help Beat Detective to get the bar and beat lines in the correct places (right).

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Advanced Timing Correction In Pro Tools

Pro Tools News * Firewire 800 drives qualified on Tiger Firewire 800 drives with the Oxford 912 interface have been officially qualified by Digidesign on Mac OS 10.4 with Pro Tools HD and Pro Tools LE systems running Pro Tools 6.9.2 or higher. Digidesign have not tested Mac OS 10.3 or Windows with FW800 and the degree of this support is very specific. Digidesign have confirmed that mixing FW800 and FW400 on the same computer is not approved, as they have seen numerous problems when devices are mixed on the same buss. However, following comments on the User Conference, Digidesign are approving the use of the Canopus DV video boxes, which are FW400 of course, for use with FW800 drives. * iLok transfer trick You may have noticed that since Digidesign have released the Massive Pack 4 bundle there have been loads of plug-ins available for sale on eBay and the like. As I stated last month, you should proceed with caution when buying and selling plug-ins, especially those included in the Massive Pack 4 bundle, as they cannot be transferred. Those who already owned some of the individual plug-ins and then bought the bundle could sell off their original copies, but be aware that if you sell your non-bundled version you may compromise your upgrade options, as most if not all of the bundled versions have restrictions on upgrades later without spending more money. So I am afraid there is no such thing as a free lunch. If you do choose to sell your non-bundled plug-ins, it is going to cost someone $25 per plug-in to transfer the licence from one iLok account to another. However there is a workaround that is worth considering if you have a number of plug-ins you are selling or buying from one source. The buyer supplies the seller with a blank iLok, which the seller synchronises to their iLok account and transfers the licences for the plug-ins they are selling. The seller then removes it from their account within the seven-day grace period offered by PACE to allow for mistakes and sends it to the buyer, who synchronises it to their iLok account. Hey presto, it's all done for the cost of one blank iLok. I can't guarantee that this loophole will stay open for ever, and you should also be aware that not all plug-in manufacturers allow licence transfers anyway, so if you are buying or selling iLok licences, be very careful as there are a large number of pitfalls to drop in. * M Box 2 and native Core Audio By now you will have heard all about the new M Box 2, but here is one neat thing you may not be aware of: it is a native Core Audio device, which means it does not need the Core Audio Manager Application, and will allow you to run Pro Tools LE and Core Audio applications simultaneously! However remember that the original M Box, Digi 002 and HD systems will never work this way and will always require the Core Audio Manager. I suspect this is a sign of things to come in future Digidesign hardware. * Dial CS for updates There have been more Pro Tools 'cs' updates again this month. I suggest you either check the User Conference at http://duc.digidesign.com/ or go to the 'cs' page in the support area of Digidesign's web site at www.digidesign.com/ download/cs/ regularly.

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A Different Problem As it happens, the second example I'm going to focus on was another Amy Grant cover I recorded for the same client two years ago. In this case the song had been tracked and mixed, but then the client wanted to replace the original keyboard part with a piano part that contained all the detailed fills and so on that hadn't been played on the keyboard part. However when the pianist came to play the new piano part she understandably found it very difficult to play exactly in time with the original, especially as there was no click track to work to. We worked as hard to get the piano part as close as possible and then I sent them home and set about using Beat Detective to match the two. Firstly I had to create a tempo map of the song. I didn't have a MIDI track to help me on this one (I learnt that for next time — it's easier to analyse the MIDI data!), but this song did have drums, so I used the same technique as described in the first example to create a tempo map from the kit. When the kit wasn't playing then I used the keyboard part. However, what was really needed in this example was to 'quantise' the new piano audio to the tempo map created from the original keyboard and kit parts. This is the other job for which Beat Detective is designed.

The other main function of Beat Detective is its ability to 'chop up' an audio part and move the sections around so that they line up with the grid or with a tempo map. Here, I've analysed a two-bar piano part to find the start of each note (above), before hitting Separate (below) to divide it into separate regions.

Hitting Conform moves the new regions to the appropriate positions on the tempo map.

Moving individual notes around invariably creates gaps and overlaps. Beat Detective's Edit Smoothing function automatically extends regions to fill gaps and creates crossfades where needed.

Again, this should be done in small sections. As you can see in the topmost screen to the right, I have selected a two-bar phrase and analysed it to find the start of each note. Then I select the Region Separation option in the Operations section of the Beat Detective window and Analyse it, adjusting the Sensitivity control so Beat Detective picks up the correct note edges. When I am satisfied, I then hit the Separate button, and Pro Tools will file:///F|/SoS/SoS%2012-2005/protoolsnotes.htm (6 of 7)11/23/2005 3:05:36 PM

Advanced Timing Correction In Pro Tools

create regions for each note (right). I now select the Region Conform option in the Operations section, and you will see that you have options as to how Beat Detective will move these regions to fit the tempo map. Standard will lock the new regions tight to the tempo map, but you also have the option to impose groove and swing in the way the regions will be positioned. On hitting the Conform button, Pro Tools will move each one of these regions to line up with the tempo map. You will notice that there are gaps in places where the regions been moved, and the final stage in the process is to sort those gaps out and 'smooth over' the edits. To do this, select the final option in the Operations section of the Beat Detective window: Edit Smoothing. This will extend regions intelligently and apply crossfades to all the edits. This example was way more demanding than the usual applications of Beat Detective like tightening up a drummer's parts or reining in a wayward bass guitar, and it certainly saved the day for my client! It just goes to show that Beat Detective is an extremely powerful tool in our arsenal to help produce music of the highest quality — provided you make sure its interpretation of your music is the correct one. Otherwise, you will find the errors just get compounded. Enjoy! Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Audio Interface Manufacturers' Round Table

In this article:

Audio Interface Manufacturers' Round Table

Support For PCI Express PC Musician The Round Table Panel The Death Of The PCI Card Published in SOS December 2005 Typical Customer Problems Print article : Close window Firewire Versus USB 2.0 Technique : PC Musician Musical Networks & mLAN Firewire Hot-plugging PC Chip-set Compatibility Final Thoughts

With interface standards and user requirements changing all the time, the audio interface marketplace is a volatile one. We catch up with representatives of eight leading manufacturers for the inside track on the future of audio I/O hardware. Martin Walker

Back in SOS September 2004, we gathered together the thoughts of music PC manufacturers in a 'round table' discussion, which proved very popular with SOS readers and with manufacturers. The latter were finally able to explain why they chose the components they did for their PCs, how they managed to keep them quiet and cool, and what steps they took to make sure they were as reliable as possible. In our latest round table, we turn to audio interface manufacturers, who are the best people to explain why they choose to support the formats they do in their current product ranges and what formats they are likely to support in the future. After all, this is a time of great change, and there are uncertainties about how long the well-established PCI slot will continue to appear on new PC motherboards, whether or not PCI Express products will take over, or whether we'll all make the move to USB and Firewire audio interfaces. Given the almost inevitable compatibility problems caused by the huge variety of available PC motherboard chipsets, we also hoped to find out more about how manufacturers test their new products before release. Representatives from Echo Audio, Edirol, ESI, M-Audio, MOTU, RME, Terratec and Yamaha were kind enough to agree to answer our questions.

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Audio Interface Manufacturers' Round Table

Support For PCI Express PCs featuring Intel's PCI Express slots have now been available for over a year, yet not a single PCI Express audio interface has yet been released. What do you think of this new format's audio capabilities and do you anticipate developing new products that support PCI Express? Matthias Carstens, RME (Matthias): "A PCI Express core is much more complicated than a PCI core, so it makes no sense for the pro-audio industry to invest a year or more in designing one, when a complete solution will be available sooner or later from specialised sources. PCI is everything one needs for 'normal' usage. PCI Express is only helpful for professional multitrack users, who exceed the typical PCI limits. For example, when using multiple HDSP MADI cards (each with 64 I/Os), PCI Express is expected to push the limits significantly. Therefore it is no surprise that we do have plans to have the MADI card ported to PCI Express, but no date so far. "It might be interesting to note that the first PCI Express Firewire cards are available. First tests show that everything works as usual. This is a good sign, as a complete disaster (crackling all over the place, despite the high transfer rate) would have surprised nobody in the audio world. Further tests with multiple Firefaces running at 192kHz will be necessary to check the limits of PCI Express audio operation. If this works better than before (so far, all Firewire is PCI-based), more audio will surely find its way to this new platform even faster."

Matthias Carstens, RME: "A PCI Express core is much more complicated than a PCI core, so it makes no sense for the proaudio industry to invest a year or more in designing one, when a complete solution will be available sooner or later from specialised sources."

Claus Riethmueller, ESI (Claus): "PCI Express is at least as advanced and flexible as PCI or PCI-X [extended]. However, it's not compatible, making development more complex for hardware vendors at this moment. In any case, PCI Express is certainly on the agenda of ESI Professional for future developments." Milo Street, Echo (Milo): "We're still evaluating PCI Express and probably will be developing products to support it in the future. One potential advantage over PCI relates to quality of service and the ability to allocate bandwidth. This could potentially allow lower latencies than PCI, which is already better than Firewire or USB." Bret Costin, M-Audio (Bret): "PCI Express promises increased bandwidth but our customers are currently well-served by our Firewire-, USB- and PCI-based products. Few of today's computers have spare PCI Express slots available for use with audio, and audio-chip manufacturer support for PCI Express appears to file:///F|/SoS/SoS%2012-2005/pcmusician.htm (2 of 14)11/23/2005 3:05:41 PM

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be non-existent at this time." Phil Palmer, Edirol (Phil): "We currently have no plans for any PCI Express devices. Edirol/Roland have led the development of USB interfaces on PC and Mac and we work closely with Apple on Firewire products. We feel that concentrating on these core technologies is the best way to bring innovative new products to market. The PCI Express protocol is still very new and, like all high-speed serial technologies, may be more suited initially to the sort of continuous unidirectional transfer that characterises disk controllers and graphics cards." Mario Michel, Terratec (Mario): "Terratec Producer's PCI audio systems are always based on dedicated PCI audio controller chips such as the VIA1712(24). Until now we are not aware of a standard PCI Express audio controller chip, so we cannot plan in that direction. Anyway, PCI Express is mainly useful for huge quantities of audio channels (for example, MADI) and we do not plan such a device in the near future." Peter Peck, Yamaha (Peter): "Yamaha cannot comment on any new development areas we are investigating. Our development currently focuses around the mLAN products we have (and are still continuing to release), as the requirements for our users are more than met by the capabilities of the IEEE1394 buss. Currently, there is no immediate need to develop support for PCI Express when we can already achieve channel input and output counts over mLAN which exceed even the most demanding of audio situations. However... never say never!"

Claus Riethmueller, ESI Professional: "PCI allows extremely affordable solutions that are not possible via USB or Firewire at the moment with similar pricing or the same level of audio quality."

Jim Cooper, MOTU (Jim): "As leading audio interface manufacturers, MOTU take a serious look at all new I/O technologies."

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The Round Table Panel Milo Street, Chief Technical Officer, Echo Digital Audio (www.echoaudio.com)

Echo Digital Audio manufacture a complete range of multi-channel, DSP-based audio interfaces using PCI, Cardbus and Firewire technologies. Phil Palmer, Product Specialist, Edirol Europe (www.edirol.co.uk)

Edirol cover all types of audio interface for Mac and PC, using USB 1.1, USB 2.0 and Firewire, starting from the straightforward UA1EX all the way up to 10-in/10out 24bit/192kHz rackmount devices such as the UA1000, UA101 and FA101. Claus Riethmueller, Head Of Sales & Marketing, ESI Professional (www.esi-pro.com)

ESI offer a full range of products for the home recording and professional project studio market, including PCI audio interfaces and USB devices for the digital home DJ market. Bret Costin, VP of Engineering, M-Audio (www.m-audio.com)

M-Audio offer more than 20 interfaces of varying configurations, from entry level to professional, using PCI, USB and Firewire technology. Jim Cooper, Director of Marketing, MOTU, Inc. (www.motu.com)

MOTU are leading developers of PCI and Firewire audio interface products. Their Firewire interfaces include the 828mkII, 896HD and the buss-powered Traveler. Their PCI line consists of the PCI-424 system, to which up to four breakout interfaces can be connected for a maximum of 96 simultaneous channels in and out. Matthias Carstens, President, RME (www.rme-audio.com)

RME design and develop professional digital audio interface solutions. The company's range of PCI cards extends from the all-in-one HDSP 9632 (featuring analogue, ADAT, SPDIF and MIDI interfacing) up to the 64-channel HDSP MADI card. Using Cardbus and various external breakout boxes, RME also offer professional portable laptop-based multitrack solutions. Mario Michel, Product Management, Terratec Producer (http://audioen.terratec.net)

Terratec Producer is part of Terratec Electronic, which is mainly a manufacturer of PC and Mac consumer retail audio and video products. The company have continuously expanded their product range, which now includes professional audio systems and soundcards including the DMX, EWX, EWS and Phase series. Peter Peck, Marketing Manager, Music Production, Yamaha-Kemble (www.yamahamusic.co.uk)

Yamaha Corporation manufacture a complete line of professional audio and musical instrument products for sound reinforcement, recording, post-production and live performance applications. Yamaha mLAN technology provides the ability to route digital audio/music from one device to another without physically changing the cable connection, and mLAN hardware and software devices work individually, or cohesively with other computers or mLAN devices, in conjunction with built-in IEEE 1394 Firewire ports.

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The Death Of The PCI Card With the introduction of PCI Express and the popularity of both USB and Firewire audio interfaces, many musicians are beginning to view the PCI soundcard as an endangered species. How long do you think it will be before the PCI audio interface dies out altogether, like the ISA standard before it? Claus: "At the moment, PCI and PCI-X are providing the most cost-effective audio solutions, either in the high end when a lot of I/O channels are required (like our MaXiO range of products) or for the entry-level market (with products such as Juli@ or ESP1010). PCI allows extremely affordable solutions that are not possible via USB or Firewire at the moment with similar pricing or the same level of audio quality. Simply because of that, we will see PCI audio devices for quite some time into the future. Ultimately, PCI Express will replace PCI and will establish itself as an important alternative to Firewire and ultimately USB."

Phil Palmer, Edirol Europe: "We have worked very closely with both Microsoft and Apple to ensure that USB 2.0 support within the OS works perfectly, and this may be why we are the only manufacturer that offers USB 1.1, USB 2.0 and Firewire products on both platforms."

Jim: "MOTU's current PCI-based systems still have a performance advantage over Firewire and USB products, even second-generation Firewire B (800Mbit) and USB 2.0 (480MBit) products. And our sales reflect this. MOTU PCI systems are still very attractive to many users — typically high-end customers who need the highest quality A-D/D-A money can buy, large channel counts, varied I/O formats, very low latency and the large-scale, interinterface matrix mixing offered by our PCI424 family. We believe our PCI424 system is the very best native system that money can buy."

Bret: "It will likely be a few years longer. Soundcard performance was quite different between ISA and PCI, as the latter offered a serious advantage over ISA. The improvements today are more incremental, and as a result the push is not as aggressive to adopt these new technologies." Mario: "Our development focus is USB 1.1/2.0, as well as IEEE1394 Firewire 400/800. We do not plan any new PCI audio systems in the near future and will continue to update the current PCI system drivers and software for a long time. We will sell our PCI-based systems for as long as our customers are willing to buy them, and I'm sure that will be the case for the next two to three years."

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Phil: "This is really difficult to foresee, and I suspect that they will be around (certainly at the lowest price points) until PC manufacturers stop including PCI slots in their designs." Matthias: "At least five more years. IMHO." Milo: "The audio advantages of PCI Express over PCI aren't as great as they were for PCI over ISA. Also, it could be a while before PCI slots are no longer supplied on motherboards (it took several years for ISA to go away), so a PCI audio interface purchased today should be usable for quite some time. However, I would expect most manufacturers to eventually either migrate to PCI Express or solely support the serial interfaces." Peter: "In my experience, musicians simply like the flexibility of external devices — being able to move their hardware from computer to computer as their systems naturally evolve and not having to open their PCs. Additionally, with the increased use of the laptop in music production, external devices are increasingly more desirable to the customer. This flexibility allows customers to keep their external devices longer than a card-based product, thus increasing its 'value for money' and lifespan. That demand for product longevity and flexibility has reduced the emphasis on the PCI buss, and I suspect that this is another factor that makes it appear that PCI is 'endangered'."

Typical Customer Problems What are the most typical interface problems reported by your customers, could they avoid them and, if so, how? Phil: "The two commonest problems would have to be installation issues and noise in the audio signal. The first issue could be avoided with some reading of the manual — clichéd as that statement is. In the vast majority of cases, our support staff merely run through the procedures in the manual and this solves the problem. That, plus a little care over what software and hardware is installed on the customer's computers, probably covers 50 to 60 percent of calls. "The second issue is one that has been covered (extensively and well) in previous issues of SOS and primarily relates to laptop use with buss-powered interfaces. This is almost always due to the design of the laptop and its power supply, whereby multiple double-insulated circuits connected by the audio interface end up with noise on the audio input (ie. a voltage difference between grounds). If you must use a laptop for your DAW, the simplest way to avoid this problem is by buying one from those specialist suppliers who advertise in SOS, that has been built for audio use." Jim: "All MOTU products are accompanied by a thoroughly written, well-indexed file:///F|/SoS/SoS%2012-2005/pcmusician.htm (6 of 14)11/23/2005 3:05:41 PM

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printed manual for both Mac and PC. As you might expect, we find that the vast majority of customer issues could be resolved by reading and understanding the information in the manual. For further information, users can check our technical support database at www.motu.com. We also recommend that users always check for the latest drivers and other software updates at www.motu.com. Between the manual, the tech support database and the latest software updates, most users experience quick and easy resolution of any issues that arise." Bret: "The single biggest issue we're aware of from our customers is system configuration. When a computer is not configured correctly, the user can experience errors such as IRQ conflicts, devices not being recognised, pops and clicks in audio, and so on. Our large technical support team spend much of their time helping our customers to ensure that their computers are properly optimised for audio." Mario: "If a customer is asking for support he mostly just needs some help to set up his complete hardware and software system (BIOS, OS and audio application settings). We try to avoid these problems by offering as much information as possible in the manual." Peter: "Just general PC issues really, such as non-standard setups, illegal software cracks and general PC maintenance issues."

Firewire Versus USB 2.0 Now that both Macs and PCs have USB 2.0 ports, it's perhaps surprising that so few USB 2.0 audio interfaces have been released. Does Firewire offer you inherent practical advantages over USB 2.0, or are there other reasons for USB 2.0's limited support? Phil: "We have worked very closely with both Microsoft and Apple to ensure that USB 2.0 support within the OS works perfectly, and this may be why we are the only manufacturer that offers USB 1.1, USB 2.0 and Firewire products that work on both platforms. Now that the OS issues have been sorted out, there are no inherent advantages in Firewire. Of course, the other manufacturers may need time to catch up..." Mario Michel, Terratec Producer: "We do not plan any new PCI audio systems in the near future and will continue to update the current PCI system drivers and software for a long time. We will sell our PCI-based systems for as long as our customers are willing to buy them, and I'm sure that will be the case for the next two to three years."

Claus: "In short, Firewire has no real practical advantages compared to a properly designed USB 2.0 audio interface. However, developing a USB 2.0 audio device that provides the stability required in the pro audio world is not as simple as it might look — in fact, considering the number of components available for development, it is definitely more difficult compared to the design of file:///F|/SoS/SoS%2012-2005/pcmusician.htm (7 of 14)11/23/2005 3:05:41 PM

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Firewire audio devices, at this moment. However, once development is completed, USB 2.0 devices can be produced in a more cost-effective way than Firewire devices, and that means that the number of USB 2.0 audio devices on the market will increase in the future. USB 2.0 has the potential to replace Firewire completely in the long term. It's happening already for non-audio peripherals and it will happen for audio interfaces as well." Mario: "Right now there are lots of USB 1.1 audio devices in the market because dedicated USB 1.1 audio controller chips such as the TI TAS1020 or TUSB3200A are available. It seems that chip manufacturers are not willing to produce chips just for the MI/pro-audio market but only for the bigger quantity PC/ Mac consumer retail business. Here good driver support from the OS suppliers is necessary (ie. Windows class-compliant or Mac Core Audio support). For Mac and Windows it took quite some time to get working OS-based USB 1.1 driver support. USB 2.0 class-compliant driver support with XP is available now but is still not for Mac OS X Core Audio. The reason might be that there still are no dedicated USB 2.0 audio controllers available. "At the moment, Firewire and USB 2.0 audio devices need a lot of development work, because the USB 2.0 and Firewire audio controllers do have to be built by using standard (not audio dedicated) components. First, such a solution increases the price, and USB still indicates a lower price. On the other hand, to start twin developments at the same time needs a lot of resources. After having Firewire products in the market we're now developing USB 2.0 products." Milo: "Firewire was designed from the start as a media interface, and industry standards were adopted early on for streaming audio and video. Apple have done a good job of formalising this and include multi-channel Firewire audio support in OS X, with no need to develop specialised drivers for each piece of Firewire gear. Streaming media support and high-speed data rates were added to USB as an afterthought and the standards lag accordingly. Also, there are currently no turnkey solutions, such as BridgeCo controller chips for USB 2.0, forcing manufacturers to 'roll their own'." Jim: "From a performance standpoint, Firewire A (400Mbit) and USB 2.0 (480Mbit) are fairly equal. Firewire B ups the ante, with 800Mbit performance. USB 2.0 is very much a viable format for audio I/O, especially for Windows, where it historically has been most pervasive, and I think the marketplace is likely to see more USB 2.0 audio interface products emerge as time goes on. I think the inequality in market presence between Firewire and USB 2.0 is due to the fact that USB 2.0 is relatively new compared to 400Mbit Firewire." Peter: "Firewire connectivity is essential for mLAN communication. Why file:///F|/SoS/SoS%2012-2005/pcmusician.htm (8 of 14)11/23/2005 3:05:41 PM

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Firewire? Because it allows communication without a computer in the network. We can now transmit digital audio, clock and MIDI data from instrument to instrument and device to device without the need for a host computer — a feature that is becoming increasingly important as live performance setups become more technologically advanced. Unfortunately, USB still carries the image it collected in v1.1, that it cannot handle high volumes of data in a reliable manner. Firewire also gives the advantage of still being scaled upwards, with 800MBPS now commonplace on new Macintosh computers." Bret: "There is currently no cross-platform audio-class driver support for USB 2.0, and M-Audio's Firewire product line is flexible in terms of being busspowered and therefore highly mobile. Another issue is that USB 2.0 audio devices are resource intensive compared to Firewire audio devices: an 8x8 USB 2.0 interface uses a lot more CPU resources than a similar 8x8 Firewire interface." Matthias: "Of course [Firewire offers advantages]. The whole format and underlying technology is better suited to streaming audio at low latency with low CPU load. Therefore USB 2 never had and will never have a bright future as an audio interface format."

Musical Networks & mLAN Yamaha's mLAN protocol is now available for both Windows and Mac computers and provides connection and control of a musical network via Firewire cables, but until recently support was restricted to devices equipped with Yamaha's own Firewire controller chips. Now that BridgeCo controller chips are also supported, and that over 100 manufacturers are part of the mLAN Alliance, are lots more mLAN-compatible products going to be found in recording studios worldwide? Peter: "Yes. During August, Yamaha released more mLAN-compliant products (the S90ES Synthesizer and the AW2400 Digital Audio Workstation), so the network continues to expand. Implementation of mLAN into existing hardware is also becoming a reality, with expansion boards for the Yamaha digital mixers. The mLAN protocol continues to evolve, allowing different types of device (synths, mixers, DAWs and so on) into the network. Various companies, including CME, have also recently announced mLAN support for their products, which, with the new chipsets, makes mLAN even more tangible for many users." Mario: "Hopefully! I cannot speak for our competitors but we already have file:///F|/SoS/SoS%2012-2005/pcmusician.htm (9 of 14)11/23/2005 3:05:41 PM

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Firewire mLAN support for one Terratec Producer product (Phase 24 FW). The first candidate software will be released end of September." Claus: "No." Jim: "MOTU's Firewire products have always been focused on providing high-quality, highperformance audio I/O to and from the computer (both Mac and PC). This focus allowed us to bring the very first Firewire audio interface (the 828) to market in 2001. However, we've been keeping a close eye on mLAN."

Milo Street, Echo Digital Audio: "We're still evaluating PCI Express and probably will be developing products to support it in the future. (It) could potentially allow lower latencies than PCI, which is already better than Firewire or USB."

Phil: "I suspect that there will not be lots of mLAN-compatible products released. The protocol is yet another layer on top of the 1394 standard and therefore adds to the complexity of full OS support from hardware manufacturers."

Bret: "We see mLAN as another ill-fated additional standard that never offered much more than class-compliant drivers. Apple and Microsoft are committed and required to update their class drivers, while mLAN is not maintainable at that same operating system level, limiting its reach and potential effectiveness." Matthias: "We don't think so."

Firewire Hot-plugging Do you think that the popularity of Firewire audio interfaces has been affected by the hot-plugging problems that have fried some musicians' Firewire ports, motherboards and peripherals, and what, if anything, are you doing to minimise the risks on your own Firewire products? Milo: "Not especially, since hot-swapping is a convenience most users can live without. Our Firewire products have their own internal power supplies and don't rely on cable power, which I assume has been the primary source of the problems. We haven't had any issues with hot-plugging our boxes." Phil: "I think that this may have affected the popularity of some manufacturers' products rather than Firewire interfaces in general. Our products have circuit protection built in, to minimise any chance of failure, and we've never had any damage reported, to the best of my knowledge."

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Claus: "We had no serious problems of this kind with our Firewire devices." Bret: "M-Audio products adhere rigidly to the Firewire industry standard and pass stringent internal testing. Beyond that, we have designed an extra robustness into our Firewire products. MAudio are being pro-active in investigating any issues that may adversely affect our customers. We educate users on the risks of hot-plugging and steps to avoid problems with inserts inside our packages and on our web site." Jim: "MOTU Firewire interfaces are carefully engineered and rigorously tested. We are well aware of the potential hazards of hot-plugging the Firewire connection and have taken all necessary precautions in our design and manufacturing to prevent this type of problem. We have thousands of satisfied customers who use their MOTU Firewire interface every day without incident."

Peter Peck, YamahaKemble:"The mLAN protocol continues to evolve, allowing different types of devices (synths, mixers, DAWs and so on) into the network. Various companies have also recently announced mLAN support for their products which, with the new chip-sets, makes mLAN even more tangible for many users."

Mario: "Maybe! With Terratec Producer Firewire systems I never heard of such a problem and we've never had any customer feedback in that direction." Matthias: "Not at all. The Fireface does not use buss power. It needs too much current, so this option was removed early in the development process. When we became aware of the power spike problems we also removed the power connections between the (hub) sockets in the Fireface, so Firewire power is not passed through to other devices." Peter: "No. We do not believe such issues have affected Firewire popularity, and believe this is just journalistic scaremongering. It is still the easiest and most versatile communication protocol. We can honestly say we have never seen an mLAN device fry a computer's ports."

PC Chip-set Compatibility What range of PC system chip-sets do you test your audio interfaces with before release, so that they are compatible with the vast majority of hardware? Given the almost infinite number of combinations of PC

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components, is it inevitable that a few systems will always end up incompatible with some product? Milo: "We test on a wide range of in-house PCs and Macs, from low-end machines to multi-processor systems, as well as beta-testers' machines external to Echo. But incompatibilities will inevitably arise. As an example, one particular Southbridge chip recently gave us problems due to incorrect BIOS settings programmed by the motherboard manufacturer. The chip-set worked properly in some systems but not others. These things can be tough to track down." Phil: "We utilise two phases of testing. First, there Jim Cooper, MOTU: "USB are official test sites in Japan maintained by PC 2.0 is very much a viable vendors (including Apple) that allow us to test a wide format for audio I/O, range of current retail models. This would probably especially for Windows, mean that between 50 and 60 models of computer where it historically has would be tested with each of our products. Second, been most pervasive, and I think the marketplace is we assemble new PCs in our labs when any major likely to see more USB 2.0 new chip-set release is made by Intel, VIA, SiS, ALi audio interface products and so on, and use these for testing and emerge as time goes on." development. The combination of these two methods means that a very large number of machines is tested with every product that we produce. That's probably the reason why we rarely have compatibility issues, although it is impossible to give guarantees, due to the enormous variability of the software/hardware combination." Claus: "In principle, it is inevitable. However, with proper testing there are actually very few cases of incompatibility that cannot be resolved these days. This is different to the situation a few years ago, where the performance and compatibility differences between the various main board chip-sets caused a lot of work for engineering and technical support. At the moment, it is enough to make sure to test every modern chip-set (or Firewire/USB controller chip for specific devices) with two to three different main boards (or controllers) from different vendors during the development of the product to ensure a very high rate of compatibility. After the product has been released, the tests continue, depending on the input from technical support." Matthias: "We test with lots of computers that we gathered over the years. Plus newer models, of course. RME's big advantage is that all our devices use the same core technology. All our PCI cards use the same PCI core and thus show the exact same compatibility range. That makes testing and a prediction of compatibility much easier. Thanks to our technology, we were able to implement several workarounds for performance and functional flaws of some chip-sets. Sometimes there was no workaround, but then the manufacturer came up with a driver fix. In this way RME's PCI and Cardbus cards have become exemplary regarding performance and compatibility."

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Bret: "Over the years, M-Audio have developed a wide compatibility test suite, and we are pro-active in acquiring newgeneration systems, motherboards and chip-sets to test our product line with. Our experience has been that by using this test environment we have only rarely seen an incompatible system in the field." Jim: "MOTU thoroughly test our products before we bring them to market, with our Bret Costin, M-Audio: "There is currently extensive in-house test bed as well as an no cross-platform audio-class driver extensive outside beta-testing pool. Over support for USB 2.0. Another issue is the last 20 years, we've learned how to that USB 2.0 audio devices are resource cover all the bases to ensure that our intensive compared to Firewire audio products are ready for the marketplace. devices." While individual incompatibilities invariably arise in the PC world, they are infrequent enough that we have been able to promptly address them." Mario: "Terratec Producer is part of Terratec Electronic, which is mainly a manufacturer of PC/Mac consumer retail audio and video products. Therefore we have access to a very well-equipped test department and can make these compatibility tests with nearly all main board chip-sets available. Because of that we can reduce incompatibility problems to a minimum." Peter: "With our R&D teams around the world, we work very closely with all the major computer developers. As a small example, we have programming staff regularly posted within Apple in Cupertino, working alongside their Core Audio developers to make sure that the mLAN implementation works consistently within the Apple operating system. We also have worked closely with AMD and other PC manufacturers to guarantee a wide range of compatibility. I believe that this is a pretty unique situation and shows how closely we work with these companies to make sure that our hardware and software solutions perform correctly."

Final Thoughts So there we are. It looks as if PCI Express will eventually prove popular for audio interfaces, but only once mainstream manufacturers develop suitable interface chips that can be pressed into service by the specialist audio community, and we may end up with even lower audio latencies as a result. However, some manufacturers look likely to stick with Firewire-based interfaces, especially as new and faster variants of this standard appear.

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Most companies seem to think that PCI slots will be with us for at least another two years, and possibly five or more, but USB 2.0 audio interfaces face a more mixed reception, as does support for Yamaha's mLAN protocol. While a few fried Firewire chips have been found around the world, most of the audio interface manufacturers either avoid self-powered devices altogether or include measures that protect their products, which is reassuring. Finally, the old belief that manufacturers only test their products on computers that have Intel chip-sets is finally disproved, with some companies utilising 60 or more different PCs to ensure the widest compatibility. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

In this article:

Learning Curve Before The Fact Shorter And Sweeter Portable Pixies Going Downtown Carried Away To The Mix

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven' Producer/Engineer: Gil Norton Published in SOS December 2005 Print article : Close window

Technique : Recording/Mixing

With their oblique, short and often brutally noisy songs, the Pixies reinvented rock music at the turn of the '90s, and influenced almost everyone who picked up a guitar in the following decade. Producer and engineer Gil Norton helped them to shape their breakthrough single. Richard Buskin

Punk, heavy metal, surf rock, pure pop, screaming guitars, manic vocals, abrupt stop-starts, short song structures, strange, often disturbing lyrics about almost everything under the sun: from their 1986 inception to their 1992 disbandment, the Pixies combined all these elements to stunning effect, and in so doing paved the way for the indie/ alternative/grunge explosion of the early 1990s. Formed in Boston by guitarists Charles Thompson and Joey Santiago, and swiftly augmented by bassist Kim Deal and drummer David Lovering, the quartet signed with the English 4AD record label in 1987 and released their uniquely eclectic debut album, Surfer Rosa, the following year. While this earned the Pixies critical plaudits and some commercial success in the UK, along with plenty of American college radio exposure and a Stateside deal with Elektra Records, the band members surprisingly opted to part ways with engineer Steve Albini before embarking on their much-anticipated follow-up, entitled Doolittle. Albini had played an intrinsic role in crafting the band's rasping, guitar-dominated sound, but Thompson, who was now going by the name of Black Francis, was keen to experiment with new ideas, and so Gil Norton was recruited to take his place for the Doolittle album. file:///F|/SoS/SoS%2012-2005/classictracks.htm (1 of 10)11/23/2005 3:05:45 PM

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

Learning Curve Norton's engineering career had begun in 1980 at Amazon Studios in his native Liverpool, where he'd worked freelance with Echo & the Bunnymen, China Crisis, OMD and members of Teardrop Explodes, Dead Or Alive and Icicle Works, often on demo sessions. "It was a good education," he says. "We'd start at 10 in the morning and have three songs finished by six in the evening. That's how we did things, and there'd be a different band each day. For me, just learning to mix was very educational, because we were working eight-track — the kit went down in stereo, and then you'd have a bass track, a guitar rhythm track, a keyboard track and maybe a bit of percussion, and all that went down to a stereo pair. You'd have to pre-mix that down, and then you did vocals and any little bits of glitter on top of that, so you had to learn how to balance things. There were no second chances. You were pre-mixing straight away. The kit was the kit — it wasn't like the bass drum or snare were separate — and after that the entire backing track was done. You'd put the vocals on, along with the harmonies, keyboards, bits of string and whatever else you'd want to do, and you were mixing as you went along, which I quite liked and I still like doing. Especially with Pro Tools, which I use a lot now."

The Pixies: from left, David Lovering, Charles 'Black Francis' Thompson, Kim Deal and Joey Santiago.

By 1983, Norton was also producing, and soon he was being managed by John Reed, who takes care of his career to this day. Work with Throwing Muses in Boston led to Norton watching their support act, the Pixies, perform at a hip local punk club named The Rat (formerly the Rathskeller). "They blew my brains away," he recalls. "The first time I saw them, Kim wasn't involved. One of her family members was ill, so she didn't play bass that night, but the other guys were amazing and I was just totally knocked out by the whole vibe and the energy. You see, I grew up with punk music — I was 16 in 1977 — which was basically high-energy pop music, and that had all disappeared in England by the time we got to the 1980s. We had the New Romantics and then we were into electronic music, and the high-energy rock and roll bands had either disappeared or weren't doing all that well. So, to go to America and see a band that had energy and a vibe, playing guitars and singing and coming out with melodies as well as plenty of attitude — it was quite startling, really. And it was also a bit anti-climactic then seeing Throwing Muses after I had seen the Pixies — it was a whole different vibe." file:///F|/SoS/SoS%2012-2005/classictracks.htm (2 of 10)11/23/2005 3:05:45 PM

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

When the Surfer Rosa album failed to produce an obvious single, Gil Norton was approached to redo 'Gigantic', the Black Francis/Kim Deal composition about sex with a well-hung guy, featuring a rare lead vocal by Deal (credited on the record as Mrs John Murphy). Norton happily obliged, with Al Clay taking care of the engineering, and when Clay then got involved with the recording of the Rain Man movie soundtrack, Norton both engineered and produced Doolittle. This still featured abrasive sounds, disjointed songs and bizarre lyrics, but it also boasted a degree of polish and textural sophistication that helped broaden the Pixies' appeal and made them counterculture heroes before the likes of Nirvana came along.

Before The Fact "The way I work with bands is I do a lot of pre-production, and I make sure everybody knows what they are doing," Norton explains. "I think you've really got to understand how you're recording, why you're recording and what is important about a specific performance in terms of what is good and what isn't. I try to get a band to think about that as much as possible, because I can tell them that something's good or bad, but if they don't understand that then they're not going to do anything different, because they're just thinking that everything's great. So, there's a lot of playing with arrangements during pre-production, and by the time we hit the studio everyone's got an understanding of what they're supposed to be doing. Whether or not we achieve that is a different matter. At least it isn't confusing. I don't like my approach to be confusing to artists. "In that sense, I think the band understood more about what they were doing when we did Doolittle, just because we had worked hard in the pre-production area and we had the arrangements sorted out. We never messed around or overdubbed a lot — the band played live, and while certain things such as vocals were overdubbed, pretty much everything was done in that vein." Pre-production took place in a rehearsal room normally used by singer-songwriter Juliana Hatfield, with the band set up in a circle. The Doolittle songs had mostly already been demoed with Gary Smith, the producer who had discovered the band in 1986 and taken them into his Fort Apache Studios to record what became the eight-song Come On Pilgrim mini-album. "Some of the songs on Doolittle were newish and others they'd had for a while," Norton says. "For instance, they'd had 'Here Comes Your Man' for quite some time, and the version that appeared on the album was the third time they had recorded it. I listened to the different versions and came up file:///F|/SoS/SoS%2012-2005/classictracks.htm (3 of 10)11/23/2005 3:05:45 PM

Photo courtesy Gil Norton

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

with that arrangement of the song.

Gil Norton and engineer Al Clay with Theremin player Robert Brunner, in a photo taken during the recording of the Pixies' later Bossanova album.

"I remember thinking that I didn't want to go straight into a rehearsal room with the whole band. So, having been put up in a really nice flat in Boston, I asked Charles [Black Francis] to visit me there with an acoustic guitar so we could talk about the songs and work out some arrangements. That way we could go back to the band and say 'Look, this is basically what we're thinking,' and fine-tune everything from there rather than have everyone immediately just blasting away. That would have been a bit insane. So, Charles came over and we had two days where we just went through the demos, and that was when we really got to know each other properly in an artistic sense. "Charles would have all of these little ditties — minute-and-a-half songs consisting of verse, chorus, verse, beat-beat-beat-bang, out, we're finished. I would go 'Uh, this is really short. Can we double this bit and can we do this again?' and he'd say 'Why? Within that minute and a half I've said everything I'm gonna say.' We had this ongoing thing when we first worked together, where I'd be trying to transform the raw material into song arrangements and he'd just go 'Look, I'm not going to play that twice.'" Hence the need for 15 songs on the Doolittle album, only three of which make it past the three-minute mark. "We'd usually work on 22 or 23 songs for a Pixies album, and some of them would end up as 'B' sides and others would be scrapped halfway through," Norton states. That meant a lot more pressure on Black Francis to come up with a variety of material... "Yes, but I now try to approach things from the punter's point of view, and when I'm in the rehearsal room with any band and they're playing a song there's a certain point at which I think 'I'm bored now. Do something else.' Whenever I get to that point, something's got to happen, something's got to give. You know, a verse is 16 bars long and it should only be eight, or the bridge is eight bars and it should be four. Then there are times when there's no intro and I'm being set up for the mood I'm gonna get into, or I'm coming out of the chorus and I'm just thrown straight back into the second verse, where it would be better if I had a brief musical interlude just to get away from the vocal. "I'm still producing from the point of view of what I would like to listen to, and I think that's an important thing to do. Some people don't care, and that's fine, too. But I think that if you're aware of your market, where you're trying to aim things and who's going to be listening to this, it's important to remain within their attention parameters."

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CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

Shorter And Sweeter As the Pixies' chief songwriter, Charles Thompson was very insistent that their songs should not outstay their welcome, a point which led to much discussion with their producer. "As a producer, the whole [pre-production] process with Charles was very educational for me," says Gil Norton. "It made me think about why you want an artist to do things — if you do something twice, can you make it different? Can it grow? What can we do with it? How can we approach the whole element of dynamics, and what can we introduce to make it better and not have it sound like we're just doing the same thing over and over? I think things should develop, and that's an approach I've taken throughout the rest of my career up until now — 'OK, I've done that on the first verse. What's going to happen on the second verse? What's going to happen in the middle? Is a harmony going to come in or should it remain the same?' I think there are a lot of questions you have to ask yourself to make sure you come up with the right result, and in that pre-production area there are lots of things that you can try out as well. "I mean, sometimes it is good to just repeat exactly the same thing — it's more hypnotic that way, and there aren't as many things being forced on the listener and interrupting the vibe. There are no rules and regulations as to what you should do. However, I remember the second afternoon I spent with Charles, after we'd gone through this process of me constantly trying to lengthen the songs from a minute and a half and provide them with more complex arrangements, he said 'Let's go for a walk,' so that's what we did, and we went into a big music store where he picked up a copy of Buddy Holly's Greatest Hits and handed it to me. He said 'Gil, look at the times on these songs.' And when I looked at them, they were nearly all under two minutes. "Very few Buddy Holly songs were over two minutes, and that was an amazing thing for Charles to do, really, because how could I argue with him? Some of the best, most classic songs that anybody remembers are the Buddy Holly songs, and they were short and sweet, bang-bang-bang. That was very educational for me on so many levels, and it increased Charles's trust in me when he could see I was taking that on board."

Portable Pixies Following two days of routining the material and familiarising themselves with one another, Gil Norton and Black Francis went into the rehearsal studio with the rest of the band and worked on the drums, bass line and guitar parts for each of the numbers. "The most important thing with the Pixies — and something I still try to retain — was the word 'portable'," Norton says. "A song had to be portable. They didn't want 15 guitar parts on there, because they couldn't play them live. So, as a producer that was what I had to stick to — I had bass, drums and two guitars, and then the vocals. Those were the parameters I was working within, and that made it good, because at least on that level I had to restrict what I was going to do. There weren't going to be a lot of guitar overdubs or loads of vocals. Whatever we recorded, the band had to be able to reproduce live. And that was a

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CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

concern with 'Monkey Gone To Heaven', because we went outside the usual parameters by putting strings on there. We weren't ever going to do that on a Pixies song." If one song best encapsulates the Pixies' aforementioned melding of pop, punk and surf sensibilities with Black Francis's frenzied singing and Kim Deal's more sensuous backing vocals, it's this characteristically offbeat paean to ecological disaster. "The whole idea of adding strings occurred during the recording process," Gil Norton continues. "I love strings. They were part of getting back to where I came from. At college I was classically trained in music, and for a long time I used to play trumpet in a youth orchestra, so I grew up with sonics around me as much as I grew up with rock & roll. I've always liked those textures. I do like strings and I do like different instrumentations, but I don't want to inflict these on rock & roll bands. However, with 'Monkey Gone To Heaven' Kim was playing a grand piano in the studio, picking the strings with a plectrum while the track was playing: 'Ding-ding-ding-ding-ding-ding-ding-ding-ding-ding...' I just thought this sounded great, so I broached the idea and we ended up putting that on the chorus. Then I thought 'Let's add strings to that, they'll sound fantastic,' and that's how the whole string thing came about. "Since we had under three weeks to record, most of Doolittle was a song a day, and we managed to keep to that except for 'Monkey Gone To Heaven'. It was a case of 'Oh, it would be great just to try putting some strings on that,' and because we didn't have enough time in Boston, we had to wait until we got to the Carriage House in Connecticut."

Going Downtown Before that, the recording sessions proper took place at Downtown Recorders, where the control room housed an MCI 636-28 console and MCI JH24 tape machine, in addition to a quality assortment of vintage microphones. "They had nice equipment there and the assistants were really, really cool," recalls Norton. "One of them was Dave Snider, who was really helpful and knew his way around the studio. When you're an engineer, you don't know your way around a studio and you've got a limited amount of time, it always makes life easier to have someone cool working with you. "There were certain starting points I always had when it came to miking. For instance, I would definitely have a [AKG] D12 on the bass drum, although the snare sound on 'Monkey Gone To Heaven' was a toy snare, a little five-inch Premier snare that you couldn't do anything with. You couldn't tune it, it just came file:///F|/SoS/SoS%2012-2005/classictracks.htm (6 of 10)11/23/2005 3:05:45 PM

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

as it was and produced that sound, and that's the sound you hear on 'Monkey Gone To Heaven'. I just loved it. I thought it was great. It had an innocence about it. These days I mic snares top and bottom, but back then I never did that — normally I just used a Shure SM57 or an AKG 414, while for the toms I'd use a Sennheiser 421, a [Neumann] KM84 on the hi-hat, a D12 on the kick, and a 414 or Neumann Photo courtesy Mitch Benoff U87 overhead — if I wanted a bit more Downtown Recorders in Boston, where the of a hard-sounding cymbal, I'd use a Doolittle album was tracked. 414 fairly close, whereas for more of a room sound I'd use the 87 a little bit higher up." With the rhythm section set up in a live formation, Dave Lovering's kit was positioned at the far end of the studio. Standing nearby, Kim Deal's bass was DI'd and miked with a U47 on the cabinet, while the guitars were amped with Marshalls or Peaveys — sometimes a combination of the two, split and then mixed together — and miked with 57s or 414s. "For 'Hey' everything was done live, including the lead vocal, and that's because otherwise it would have felt disconnected," Norton remarks. "The song's emotion meant that it needed a performance, and I don't think it would have felt very good if it wasn't done in a live situation. Normally, depending on whether or not he'd finished the lyrics, Charles would do some sort of guide vocal or some sort of reference, but on 'Hey' he did the real thing and I put him in a cupboard at the back of the room for separation. I can still picture him with his left arm up around his head, playing his guitar while he was singing the song. It's one of my favourite songs on the whole album just because of that, and the fact that Joey got to the end of his solo without making a mistake was phenomenal. I remember halfway through it, saying 'Go on, Joey! You can do it, mate!' "Charles was always very easy to record. I'd just stick a mic in front of him — sometimes it could be a 58 and sometimes it could be an 87 and then sometimes we used a 47. It depended on whether he wanted to hold it or whether he wanted to scream his head off. Normally, we'd record three or four takes and then just comp. Sometimes, one take would be enough and it would be downhill from there, whereas at other times takes three or four would normally be it, and then there might be another line that we would get from take two just for freshness. Really it was a case of running through it a few times to get the mood, and then once you were there, you were there."

Carried Away "What excited me about the Pixies — and Doolittle was a stand-out in this file:///F|/SoS/SoS%2012-2005/classictracks.htm (7 of 10)11/23/2005 3:05:45 PM

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

respect — was that it was always like going on a rollercoaster ride with them. You sort of got on the rollercoaster and there was no way you were getting off until it stopped. They had so much personality, and Charles was such a character, ruling through all his ideas — for me it was always great to have so much material to work with and so many different types of songs. You know, it all made sense, it always sounded like the Pixies, but there were never, say, 12 or 15 tracks of just rock songs. There'd be quick little country songs or jazzy-type numbers, and stylistically the record would go through quite a few different moods, and that's why it was like a rollercoaster — you could go left, right, up or down at any given moment. "When I began working with the Pixies, I was never really given much information as to what they were looking for in terms of the sound or the direction. After I'd seen them live, they just wanted me to reproduce what they'd do to the best of my ability, and that's part of the job of a producer. It's a very ambiguous role, really, but it's definitely to bring out the best in a band, and on 'Monkey Gone To Heaven' I just wanted to capture the Carriage House Studios in Connecticut, song's innocence and angelic beauty. where the Doolittle album was mixed. That's why I wanted to use the strings — it had to be quite powerful, but there also had to be a purity to the power. What with the guitars and the dynamics, the song started with a mood and an impact, and then the guitars dropped out on the verses to make lots of room for Charles to start telling his story. "Although I thought strings would be great on the song, there was this whole thing about keeping it 'portable', we didn't have the budget to record strings, and I also wondered if they would make the sound really un-cool. But then I thought about the song's innocence, and I knew that if we just got a quartet in there would be a certain innocence. Rather than sounding pompous it would have that chamber thing about it, and it would be a bit more in keeping with the live performance. So, I just used two violins and two cellos — I wanted plucked strings, not to be overpowering but just to be in there as a texture. If you listen to the song, they're not a big feature. The cellos sort of go along more with what the bass line is doing, and the song works with or without the strings, but I think they're a nice enhancement. That was an important part of being a 'portable' song." Not that Gil Norton had an arrangement when the string session took place at the Carriage House Studio in Stamford, Connecticut, with violinists Karen Karlsrud and Corine Metter, and cellists Arthur Fiacco and Ann Rorich. "They were from a local orchestra and they were really cool," remarks Norton, who in addition to producing and engineering the Pixies' two subsequent albums,

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CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

Bossanova and Trompe Le Monde, has since worked with artists such as James, Del Amitri, Counting Crows, Foo Fighters and, most recently, Gomez. "They had just done a show and they were still in their tuxes and gowns. I sang to Arthur what I wanted to be played and he just sketched it out for me. Then we finetuned it, because without an arrangement we needed to find out if it was going to work or not, and the musicians began playing, and within two hours we had the bits that we wanted."

To The Mix "Having engineered the whole record, towards the end I just thought it would be great to get a fresh pair of ears on this as well, because sometimes when you're producing and engineering it's nice to get a fresh perspective on the mix. So, I asked Steven Haigler to come in and help me with it, using the SSL at the Carriage House. At that point, I was used to mixing on SSL — I didn't mind what we recorded on, although I still don't really like recording on SSLs. I'd much rather be on a Neve or a Trident... or an MCI, or whatever, just to get another character in there, and then I do like mixing on SSL. To me, it's just like driving a car. It's very easy, you sort of know where you are, and I also quite like that sort of sound it gives at the end of it. If you've got all the bottom end and you've got what you want, it finishes it off nicely." While Steve Albini had captured the hard edges of the Pixies' sound in a fairly uncompromising way, Gil Norton and Steve Haigler retained some of this edge while using reverb and compression to smooth things out and place a little more emphasis on the band's pop sensibilities. These, after all, were sensibilities that Norton himself shared. "I love pop music," he says. "People sometimes turn their noses up when they hear the words 'pop music', but popular music is what we're doing, and if you don't want people to like it then you should just do your own little thing and play it for yourself in your bedroom. Once you get to the point where you're putting things out for the world to hear, the reason to do that is hopefully people will get what you're doing and like it.

Photo: Fran Norton Gil Norton today.

"I love a hook. I'm a glutton for it, and I love '60s pop music. I just think some of the best songs ever were in so many different genres of music during that era. It's what I grew up with, and at the end of the day I'm a pop producer. That's what I want to do, and that term 'alternative producer' always used to freak me out, really, because I never set myself up to be an alternative record producer. I just want to produce good records, and I never knew what it meant to be alternative file:///F|/SoS/SoS%2012-2005/classictracks.htm (9 of 10)11/23/2005 3:05:45 PM

CLASSIC TRACKS: The Pixies 'Monkey Gone To Heaven'

to whatever. All I wanted was for the Pixies to be the biggest band in the universe. I don't think we went into any of the stuff that we did thinking 'We want to be quirky and arty and not have anyone like us.' "One thing I do is pay a lot of attention to detail. It's important to me to get these little things right, the things that other people might not think are important, and I think that sort of excites Charles in a certain way. He's got quite a weird mathematical brain and he likes things that excite him. He likes detail and he likes things that sound simple but are not. So, in a working situation we got on well together. I helped him bring out his pop sensibility to a certain extent, and he helped me on a quirky level as to where and how you choose to do things; how to avoid doing the obvious but do what you normally wouldn't do and make things more interesting. "I think within every genre of music the best songs can be played on an acoustic guitar and they've got a great melody. That was the case with Doolittle. It was routined on an acoustic guitar and all of the songs work on an acoustic guitar. I think that's the way to start, and then how you shape things after that is the art of making a record, really, or being a good band. The song has to work on its own, it has to stand up, and you have to be able to play it. You can't rely on bells and whistles to make things work, it has to be already there within the structure of the song, and that was certainly the case with 'Monkey Gone To Heaven'. "Charles hated choruses, he didn't want anything to have a chorus on it, and we had to be very subtle about anything we used. Charles liked organic, he didn't like trickery, and I don't think anything too obvious would have gone down very well. If Charles had heard a big delay or something, he wouldn't have liked it, although he didn't mind a bit of reverb, so we got away with that. It just depended on what we were trying to do, really. And I quite liked that as well, because it kept things honest to a certain extent. I mean, there isn't a lot of trickery on that record — the odd little delay or a bit of plate and a little bit of spring, because I like spring reverbs, but there wasn't a lot of anything really. We tried to keep it simple. Simple did the trick." Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Dual-core/Dual-processor G5s

In this article:

Dual-core/Dual-processor G5s

Four Heads Are Better Than Apple Notes One Published in SOS December 2005 Waiting For The Express The New iPod: Must Be Print article : Close window Seen? Technique : Apple Notes One More Thing...

Christmas came slightly early this year for Mac enthusiasts, with significant product announcements, including new dual-core, dual-processor Power Mac G5s. But just what do the new high-spec computers mean for musicians? Mark Wherry

Ever since Apple announced the transition to Intel processors earlier in the year, there's been a great deal of speculation concerning what would happen to the company's Pro line of Macintosh computers, namely the Power Mac and Power Book. Would Pro customers have to wait until the middle of next year to see a significant leap forward in Mac architecture, when Intel-based Macs are scheduled to begin shipping? Or would Apple integrate newer Power PC processors and technology into these products to keep Pro customers happy — and, perhaps more importantly, keep them buying new Macs?

Four Heads Are Better Than One If you've read anything in the technology or computer press recently, you can't fail to have heard about dual-core processors. A dual-core processor basically combines two independent processor cores and caches on a single chip, giving you pretty much the performance of a dual-processor system in a computer with only one physical processor chip. Of course, Apple, like many other computer companies, have been supplying systems with two processor cores on separate chips for some time, so you might be wondering what the big deal about having the two cores on one chip is. Indeed, for two-core systems, there isn't much difference. Where this is a big deal is for dual-processor systems where each processor is also dual-core: suddenly you double the number of available processor cores again, to four, and this gives a huge performance boost to applications written to take advantage of multiple processor cores.

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Dual-core/Dual-processor G5s

AMD have been making a great deal of noise about their dual-core, dualprocessor capable Opterons this year, and Intel had just released dual-core, dualprocessor-capable Xeon processors as I was writing this article. Both companies had been shipping dual-core, singleprocessor chips for most of this year already. IBM's Power PC 970 processor is better known to Mac users as the Power PC G5, and after IBM announced the dual-core 970MP processor earlier in the year, many people speculated as to whether dual-core Photos courtesy of Apple. Power Macs were also on the horizon, The new Power Mac G5 Quad is a incorporating these new chips. seriously powerful beast, featuring four 2.5GHz G5 processor cores, the ability Accordingly, on October 19th at an Apple Special Event in New York, the day before to host 16GB RAM, and PCI Express slots for graphics and other expansion the Photo Plus Expo, Apple VP David cards. Moody unveiled a new range of three dualcore Power Macs. The first two models feature single-processor, dual-core configurations at 2GHz and 2.3GHz, priced at £1399 and £1749 respectively, while the new high-end model offers a dualprocessor, dual-core system at 2.5GHz for £2299. However, rather than simply replacing the previous single-core models with dualcore processors, the new dual-core line-up represents perhaps the biggest architectural change to the Power Mac G5 since its original release over two years ago. Compared to the previous generation of Power Macs, the new models offer a 1MB L2 cache per processor instead of 512k (although on a per-core basis this is basically the same). The new Power Macs support 533MHz DDR2 memory (PC2-4200), whereas previous models supported 400MHz DDR (PC3200) memory, although all models still come with 512MB RAM as the standard configuration, which is perhaps a little low for the quad-core model. On the plus side, though, you can now expand the memory of each Power Mac to 16GB (as opposed to 8GB in previous models), which should be pretty exciting once more samplers and other music programs begin to make greater use of 64-bit memory addressing. The low-end Power Mac has a 160GB 7200RPM SATA drive, while the other two models have a 250GB drive, and all models feature a dual-layer compatible 16speed Super Drive. This is all the same as before, and the connectivity is also pretty similar, with the Power Mac G5 offering one Firewire 800 port; two Firewire 400 ports (one on the front and one on the back); four USB 2 ports (one on the front and three on the back, which is one more than before); two USB 1.1 ports on the keyboard (a shame, because a USB 2.0-compatible hub in the keyboard would be nice); and optical and analogue audio input and output ports. Apple also supply a Mighty Mouse as standard now, instead of the older Apple Pro

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Dual-core/Dual-processor G5s

Mouse. For networking, Bluetooth 2+EDR (Enhanced Data Rate) and Airport Extreme are build-to-order options, and one particularly welcome change is the inclusion of a second Gigabit Ethernet port. In addition to simply providing more network bandwidth, this could also come in handy for connecting Ethernet-based control surfaces to your Mac at the same time as a general data network, preventing the control surface packets from being scheduled in amongst a data load.

Waiting For The Express As is often the case with new Power Macs, one of the most drastic architectural changes concerns expansion slots. Instead of AGP, PCI or PCI-X, as featured in previous Power Macs, Apple have instead completely adopted the newer PCI Express standard for graphics and all other expansion cards. Whereas PCI-X (as used in the original Power Mac G5) was an evolution of PCI (Peripheral Component Interconnect), offering backwards compatibility with PCI in terms of both hardware and software, PCI Express is a completely new hardware implementation that uses a high-speed, 2.5GB/s serial transmission and has a level of software compatibility with the original PCI-buss architecture. What all of this means is that you can't use any existing PCI or PCI-X cards in the new Power Macs, which also means that anyone with such an audio card or DSP accelerator is going to be out of luck. This could affect Pro Tools users most significantly, since it makes it impossible to run Pro Tools HD on a new Power Mac G5, whereas at least there are USB or Firewire solutions for Pro Tools LE and other music and audio software, such as Logic, Cubase and Digital Performer. It is, perhaps, a shame that Apple didn't opt to use PCI Express for the graphics, with one other PCI Express slot for expansion, alongside two PCI-X slots: a mixture of PCI Express and PCI-X slots has been pretty common this year in the Intel and AMD world, for example. This could have been useful for musicians, since there are currently no PCI Express audio or DSP cards (specifically for music and audio) on the market. On the other hand, a move to an all-PCI Express system is inevitable in the coming year for all computer platforms [see our audio interface manufacturers' round table feature starting on page 130 of this issue], so both users and developers are going to have to migrate at some point. Fortunately, given the music and audio world's reliance on PCI-based hardware at the moment, Apple have decided to keep the previous high-end model of the last generation of G5s in its product inventory. So you can still buy a single-core, dual-2.7GHz Power Mac G5 at a reduced price of £1949. The graphics cards supplied in the new Power Macs have also been updated with an all-Nvidia line-up, offering one single-link DVI port and one dual-link DVI port (for a 30-inch display) as standard, meaning that every Mac now supports one 30-inch display straight out of the box. The entry-level model features a file:///F|/SoS/SoS%2012-2005/applenotes.htm (3 of 5)11/23/2005 3:05:48 PM

Dual-core/Dual-processor G5s

Geforce 6600LE card with 128MB of video memory, while the other two models include a full Geforce 6600 card with 256MB memory. Two other build-to-order Nvidia graphics cards are also available: the 7800GT, featuring 256MB memory, which offers increased memory bandwidth with a 256bit memory interface (the standard cards have a 128-bit interface), and the Quadro FX 4500, which features 512MB memory, a 256-bit memory interface, two dual-link DVI ports (for connecting two 30-inch displays simultaneously) and a stereo 3D graphics port that enables 3D goggles to be attached to the Mac, for those working with advanced visualisations. Probably not too useful for just tweaking plug-ins! The 4500 costs around an extra £1100, but pricing for the 7800GT wasn't available as this article went to press.

The New iPod: Must Be Seen? A brand-new, video-capable iPod was announced by Apple CEO Steve Jobs at another 'Special Event' the week before the new Power Macs and Powerbooks were revealed.

Like the iPod Nano, the new video-capable iPod is also available in black. Unlike the Nano, the new iPod features up to a 60GB hard drive, with battery life of 20 hours when playing back music.

The new iPod replaces the previous iPod, formerly known as the iPod Photo, and, in addition to being thinner than the previous generation, offers a 320 x 240 colour display for watching MPEG4 and H.264 video. In terms of video content for actually watching on your iPod, Apple are now selling music videos from the iTunes Music Store, although the real highlight, at least for US-based iPod owners, is the deal Apple have struck with Disney/ABC to offer five TV shows for purchasing via the iTunes Music store. You can now download episodes of Lost, Desperate Housewives, Night Stalker, The Suite Life and That's So Raven for $1.99 each. In addition to previous seasons being available to purchase, the latest episode will always be available for purchase and download the day after it has been broadcast. The new iPod is available in a 30GB model for £219, with a 60GB version costing £299. Both models are available in black or white.

One More Thing... Unveiling the new Power Macs ahead of the Photo Plus Expo was only part of the news from the most recent Apple Special Event, and the big reason for holding this event before a major photography show was to announce Aperture, a new pro application aimed at improving the workflow of photographers,

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especially those working with RAW-format images, that costs £349 and will be released in November. However, Apple also revamped the somewhat ageing Powerbook line. Battery life has been improved to approximately five hours in the 12-inch Powerbook and 5.5 hours in the 15- and 17-inch models. Perhaps more significantly, the 15- and 17-inch models have new displays with improved resolution. The 15-inch goes from 1280 x 854 to 1440 x 900 (the same resolution as the old 17-inch Powerbook display), and the new 17-inch Powerbook offers 1680 x 1050 pixels (the same resolution as the current 20-inch Cinema Display). All 12-, 15- and 17-inch Powerbooks now feature an eight-speed Super Drive as standard and the 15-inch and 17-inch models now offer dual-layer support as well. The 15-inch model has also been upgraded to offer a dual-link DVI port as standard. Other specifications remain unchanged from the previous line-up. The 12-, 15- and 17-inch Powerbooks retail for £1099, £1399 and £1749 respectively. Apple also unveiled new iMac models this month alongside the new iPod (see box, left). The new iMac is thinner, includes a built-in iSight camera for using iChat AV, and offers a new feature called Front Row, which includes a small remote control enabling you to use the iMac's media functionality (iTunes, iMovie, playing DVDs and so on...) from your armchair, via a brand new interface. This is Apple's first attempt at competing with Microsoft's Windows Media Center products and it will be interesting to see how this direction is developed. The new iMacs feature 512MB 533MHz DDR2 memory and use PCI Express for graphics, with an ATI X600XT offering 128MB video memory. They come with a Mighty Mouse and both the 17- and 20-inch models now provide an eight-speed dual-layer Super Drive as standard. The 17-inch model has a 1.9GHz G5 processor and retails for £899, while the 20-inch model has a 2.1GHz G5 processor and costs £1199. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Granular Synthesis

In this article:

Granular Synthesis

Granular Synthesis 101 Warping Time & Pitch With How It Works & Ways To Published in SOS December 2005 Granular Synthesis Granular Samplers & Synths Print article : Close window Drums & Transients Technique : Synthesis Sound-design Tools Advanced Possibilities

Use It

Granular synthesis is the core technology behind the latest time-stretching and pitch-shifting algorithms, but it can also be used to generate extraordinary evolving soundscapes. We explain how the process works and show you how to get the best from the software that uses it. Simon Price

The majority of software instruments use variations on the synthesis method known as subtractive synthesis. This is the sound generation method where you start with simple (yet harmonically rich) waveforms such as triangle, square, and sawtooth waves, then use volume envelopes, filters, filter envelopes, and LFOs (Low Frequency Oscillators) to sculpt the starting sound into something more musical. The reasons why subtractive synthesis is so dominant are both historical and practical. The historical reason is that most of the synthesizers that shaped the development of electronic music production (the classic analogue Moogs, ARPs, Korgs, and so on) used this scheme. Hence subtractive software synthesizers are commonly known as 'virtual analogue' instruments. These instruments are what musicians are accustomed to using, and they make characteristic sounds that have become part of the common musical sound repertoire. On a practical level, these synths are relatively easy to learn, and can be modelled in software without using a huge amount of processing power. It is probably for this last reason that subtractive synths, and straightforward sampleplayback instruments, have taken such a lead in desktop music. However, as computers have become much faster, digital signal processing techniques that were once the preserve of academic labs and telephone companies are finding a strong foothold in music software.

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Granular Synthesis

The technique known rather grandly as granular synthesis is an extremely powerful audio manipulation system that makes it possible to adjust the speed, pitch, and formant characteristics of audio samples independently of one another, and all in real time if your computer is fast enough. However, granular synthesis principles can also create new and often spectacular shifting sounds using very basic means. Acid, now at version 5 and under the care of Sony, was the first to make realtime pitch and time adjustment well known, and nowadays most people into computer music will have played with Ableton's tempo-warping Live software, not to mention Apple's Garage Band. Celemony's Melodyne is now arguably the purest and most sophisticated package for editing audio using granular synthesis, managing to carve out a niche alongside the mighty Auto-Tune. Native Instruments Reaktor has always had this technology right at its heart, but focuses on the creative sound-design possibilities of the granular approach. NI's work in this area has led to the powerful time, pitch, and timbre manipulation in their Kontakt, Intakt, and Absynth packages, finally blurring the line between samplers and synthesizers. Propellerhead's Reason package also contains a granular synth Malström, and even Fruity Loops Studio has the Granulizer. The aim of this article is to explain the basics of how granular synthesis works (for those with an interest in these things), and also to describe some examples of it in action. For those who use software with granular synthesis technology under the bonnet — whether it is for time/pitch manipulation or sound generation — understanding how it works should shed some light on how to approach some of the more esoteric parameters like Grain Size or Smoothing.

Granular Synthesis 101

The top waveform shows a very short clip of a vocal recording, while the one below it has been time-stretched without a drop in pitch, repeating the wave pattern to achieve the extra length. The third track shows the audio transposed up seven semitones — increasing the pitch squashes the waveform, so it has again been looped to preserve the clip's length.

Have you ever wondered how some audio-editing programs and plug-ins can manipulate the tempo and pitch of audio independently? Normally, of course, the laws of physics tie these two parameters together: slow audio down and the pitch drops proportionally. The screenshot on the facing page shows some very close-up views of audio waveforms in Pro Tools. The top track is a very short section of a female vocal recording. The point we're looking at is part of the 'ooo' sound in the word 'you'. The second track has the same audio clip that's been slowed down dramatically using Pro Tools' built in Time Stretch plug-in. Notice how the waveform itself has not been stretched — this would cause a drop in pitch, because pitch is inversely proportional to file:///F|/SoS/SoS%2012-2005/granularworkshop.htm (2 of 10)11/23/2005 3:05:51 PM

Granular Synthesis

wavelength. Instead, the Time Stretch algorithm has detected a repeating wave pattern, and simply looped it to achieve the extra length. The third track shows the original clip transposed up seven semitones by the Pitch Shift plug-in. The original waveform has been squashed horizontally (in time) to achieve an increase in pitch, so again the algorithm has had to loop the waveform, this time in order to preserve the length. This scheme works because, although most sounds sound to our ears like they change and develop quickly, when you zoom in and look at even the most complex waveforms (like speech) you see that, in fact, many parts of harmonic and vocal sounds consist of steady periods of a repeating waveform, with short transitions in between. A little experiment makes this clearer: try saying your name really slowly and listen to the sound you make. For me that goes something like, 'sss-aah-eee-mmm-nnn'. Your voice moves from one consistent steady sound to another, except for when you get to hard consonants like 'k' or 't' (see the 'Drums & Transients' box for more on this). At the waveform level, steady sounds appear as many cycles of the same small wave shape. So, if I recorded myself saying my name at normal speed into Pro Tools, I could zoom in and painstakingly loop the waveforms during each section of the word (crossfading the edit points), and end up with something that sounded similar to me saying the word slowly, but at the same pitch. Conversely, I could speed myself up by deleting some of the cycles from each portion of the word. This is the basis of how time-stretching works. Now, say that I took a few cycles of waveform from each sound in my name, and mapped them as loops onto keys in a sampler. One key would give a steady 'sssssss', another a steady 'ahhhhh', and so on. Now if I pressed each key in rapid succession it would (roughly) re-synthesize the original recording using these 'grains' of sound. If I played the sequence of keys faster, the word would be reconstructed faster, but the pitch would stay the same. Also, I could push the pitch-bend wheel to pitch up the samples, but still play the key sequence at any speed I liked. What's more, I could play back the sequence in any order, and even make the sounds overlap by holding down more than one key at a time, generating an entirely new and more complicated sound. This is how granular synthesis works. Granular synthesis is a catch-all term for a number of different audio systems

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Native Instruments Intakt uses granular synthesis in its Time Machine mode (left), providing a choice of preset grain sizes to

Granular Synthesis

suit different types of audio. Ableton Live that work by using tiny snippets of has a similar range of presets in the Warp sound that can be manipulated section of its sample editor pane (right) — individually and are recombined to it's currently set to the Beats preset. generate the final output. The majority of granular systems available use audio files/samples as their raw material. Samples are sliced up (behind the scenes) into a series of tiny sections, each usually between one 100th and one 10th of a second in duration. Each slice is known as a 'grain', and a sequence of grains is called a 'graintable'. If the software made up a graintable which played back all the grains extracted from a given sample in their original sequence and at the original speed, then you'd hear the original sample reproduced. If the software played the sequence back more slowly, gaps would appear between the slices, so the current slice in the graintable is usually looped. Played back more quickly, each grain overlaps with the next one, or some grains get skipped depending on how the software works. To avoid clicks and glitches, each grain is faded in and out with a volume envelope, a process known as 'smoothing'.

Warping Time & Pitch With Granular Synthesis Native Instruments' Intakt is a loop playback and manipulation tool with three 'audio engines'. As well as basic sample playback and a beat-slicing mode (handling each rhythmic event in Propellerhead Reason's Malström lets you select different preset samples as your initial a loop as a separate sample), Intakt graintable, and you can set the starting has a granular Time Machine mode. position for playback using the Index slider. To the left of the waveform display you The speed at which Malström plays the get two knobs, both marked Tempo. graintable is determined by the Motion knob. The smaller one on the right gives you manual control over the speed of playback of the sample, and is great for experimenting with how granular timestretching works and sounds. While playing the sample, if you turn this knob anticlockwise the playback gradually slows down, but maintaining the original pitch. If you go to extremes, you should be able to hear what is happening — at about five percent of the original speed you can clearly hear the playback graintable stepping from one grain to the next, each grain being looped until the next one takes over. To the right of the waveform are some controls that show up in different guises in most granular synthesis-based software. The first is the Grain Size control, which is a pop-up list of options in Intakt. Grain Size is the length of each slice of sound, determining how finely the original sample is chopped up. In Intakt, the list gives suggestions for which grain size to use to obtain the most transparent results for different types of material. There are similar parameters in the Warp section of Ableton's Live software — again, rather than a continuous Grain Size control, a list of options is provided: Beats, Tones, and Textures.

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Granular Synthesis

Granular Samplers & Synths Tools such as Intakt, Melodyne, and Live use granular synthesis to edit and match the tempos, timings, and keys of recorded audio clips. A whole other breed of products uses granulated samples as the source sounds for instruments. The likes of Absynth, Malström, and Kontakt all use the familiar synth/sampler instrument structure, with sound generators being modulated and filtered. The difference is that they can all swap the usual sound-generation stage of oscillators or samples for granular synthesis engines. A detailed analysis of how this works in Malström can be found in the Reason Notes column in SOS August 2005. The same principle applies to other current granular synths. When used in granular mode, each sound source in the instrument is a granulated sample: a graintable. Some instruments allow the user to load their own sample (for example Reaktor, Kontakt, and Absynth), while others provide preset waveforms (Malström). Looking at Reason Malström, in the Oscillator A section (the soundgenerating module of the synth), there is a small pop-up window which selects the sample (graintable) that is to be used as the starting point. In the screenshot I've chosen Ambient Chord 2. The other parameters on the Oscillator A module should now begin to make sense. The Index slider sets the starting position for playback in the graintable, and the entire sample is mapped out along this slider. The Because you can load any sample into Motion control simply sets the speed at Native Instruments Kontakt for granular processing, manual Grain Size and which Malström sweeps through the Smoothing controls are provided so that you graintable, and the main pitch settings can optimise the processing to different transpose the sample — speed and sounds. pitch changes are, of course, independent. Finally, the Shift knob provides independent control over the formant characteristics of the sound. With all these controls at their zero positions, Malström behaves like an ordinary sampler, with the significant advantage that playing up and down the keyboard does not speed up and slow down the sample: it's like having a multisample map, but without having to have more than one sample. Beyond this, there is a huge amount of flexibility, and you can quickly move away from the starting point to make radically different sounds. All the controls can be modulated with Malström's LFOs, and it's the sweeping of the parameters that gives granular synthesizers their characteristically rich and 'alive' sound. Something you can do with Malström is modulate the Index control, or sample position. As we'll see when we look at Reaktor, this is one of the most valuable tools for creating deep granular sounds and atmospheres. Playing around with the graintable position file:///F|/SoS/SoS%2012-2005/granularworkshop.htm (5 of 10)11/23/2005 3:05:51 PM

Granular Synthesis

and playback characteristics means that one sample can provide the material to generate a huge variety of unexpected results. Kontakt cannot modulate the sample position (although you can create loop points), but it does give control over some other parameters that are preset in Malström's graintables. Specifically, it features controls for Grain Size and The Density setting in Native Instruments Smoothing. In Malström, with it's Absynth lets each grain overlay those following it, which often creates phasey preloaded list of graintables, the grain metallic sounds. size has been preset to provide the most transparent response. Because Kontakt can load any audio file as its starting point, the user must set the grain size. This means that you can forget about transparency if you wish, and go for a more grungy sound. You can also modulate the grain size via an LFO or envelope. The Smoothing parameter, to recap, is the volume envelope applied to each individual grain, so it's effectively a fade-in/out control. Again, you can set this to produce a nice even response, or go for a special effect. The last synth I want to look at is Absynth, because it features yet another parameter, leading us towards the full implementation of granular synthesis found in Reaktor. Most of the parameters in Absynth 2 should now be familiar, but a Density setting has also been added. This sets the number of grains that can be playing back at once, which in Absynth's case can be between one and eight. All the examples we've looked at before can be likened to having a single 'play head' sweeping around the graintable in a mostly linear fashion. However, granular synthesis gets really interesting as a sound-design tool when you start firing off multiple grains simultaneously, and not necessarily in sequential order. Absynth doesn't go quite this far: its Density control just provides for varying grain overlap, which means that you can have several neighbouring grains firing at once as the graintable is played. This smooths out and thickens the sound, but inevitably adds a metallic or phasey characteristic, as you are overlapping a series of similar-sounding grains with a tiny delay between them.

Drums & Transients Percussive sounds and drum loops pose some fairly major challenges to granular synthesis engines, especially when you're wanting to time-stretch samples to slow them down. Granular time-stretching relies on the fact that a lot of what we hear consists of repeated cycles of small waveforms, but transients (like drum hits and hard consonants in vocals/speech) are quite different. These parts of a sound are typically short, complicated, rapidly changing waveforms. When a sample is split into grains, the transients may fall within a whole grain, or split across several, depending on the grain size used. Neither of these situations is welcome, because when the graintable is played back slowly grains are moved apart and looped. You will probably have heard the problem this causes: drums that have been slowed down by time-stretching start to sound flammy. The same goes for vocals, with the hard consonants st-t-t-uutt-t-t-ering. file:///F|/SoS/SoS%2012-2005/granularworkshop.htm (6 of 10)11/23/2005 3:05:51 PM

Granular Synthesis

Systems that don't have any way of compensating for this problem have a very limited range within which a sample can be slowed down. If you load a drum loop into Intakt, you can slow it down and listen for when the problem starts causing noticeable degradation of the sound. Short, sharp sections of the waveform, such as rim-shots, present a particularly tough test, especially if the grain size is set manually without any intelligent analysis. My ears can detect a drum loop's rim-shot starting to 'break up' into two peaks at just two to three percent slower than original speed, and ordinary snare drums start to flam at about four to five percent down.

The Transient Size (TRS) control and Transient Copy (TRC) switch are provided to help avoid the flamming of percussive sounds when you're slowing down sampled loops using Intakt's Time Machine mode.

There are a number of ways in which the designers of a granular synthesis or time-stretching system can improve on this situation, two of which are present in Intakt. The first is to have the software analyse the sample and choose variable grain sizes. In other words, instead of relying on a user-defined grain size, the software tries to chop up each part of the sample in the most efficient way. The software makes distinctions between areas that change rapidly, and those that are more steady tones. This at least ensures that transients are not split across more than one grain. In Intakt this option is the default, with the user being able to change to a fixed Grain Size if desired. The second way that transient handling can be improved is to use a transient detection system to ensure that the transients are preserved in their original state, at whatever playback speed (as they would in real life drumming or vocal performances). This means that not only must they be contained within one chunk (one grain), but that they should only be played back once instead of being looped at slower speeds. Intakt and Kontakt do something like this when you engage the TRC (Transient Copy) button. The software detects peaks and sudden changes, and interprets these as transients. A second control, TRS (Transient Size) is set manually and determines a length for these sections. During playback, the original transient sections are overlaid on the loop, with their position staying correct relative to the rest of the sample. Ableton Live has similar functionality, although it doesn't use transient detection. Time-stretching and granular settings are chosen from the sample editor window's Warp pane, and setting the audio type here to Beats tells the software to try to preserve transients. Instead of detecting these, Live uses time divisions set by the user in the Transients field, and has to assume that the drum hits land close to these. Anyone familiar with beat-slicing software, such as Propellerhead Recycle, may have spotted that this system is a best-of-both-worlds mix of techniques, preserving the original hits (as with beat slicing) but filling the gaps with timestretched material. Another problem shared with beat slicing is that decays and reverb tails are difficult to keep sounding natural. Where available, a mixture of small grain size and large transient 'windows' often works best with drums. Without transient compensation, larger grain sizes will probably be better.

Sound-design Tools Despite all the sampling and synthesis flexibility afforded by the applications

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Granular Synthesis

we've looked at so far, when most people think about granular synthesis they probably think of the rich shifting soundscapes generated by certain Reaktor patches. It's perfectly possible to build synths in Reaktor similar to those we've already looked at. For example, Triptonizer is not a million miles away from Malström, except that it uses envelopes to control the movement of sample position, formant, and so forth. However, as with the synths we've covered, this kind of instrument generally sweeps fairly uniformly through a graintable. For the more weird and wonderful sounds, we want to be layering up clusters of grains, introducing randomness, and getting away from thinking about the samples as a whole. The result is a composite sound known as a 'graincloud'. Reaktor has a straightforward sample synth module, and a Pitch Former (which is similar but moulds the results into a definite pitched sound), but it also has a module called Grain Cloud. If you don't have Reaktor, you can download the demo version and check out the factory instruments Grainstates and Travelizer to get an instant idea of what this module can do. Most of Travelizer's front-panel options should now make sense. The large XY controller sets the sample position and the grain size. The waveform display has two vertical lines that indicate the current playback position and the grain size (Length). The panels to the left allow modulation of the pitch Native Instruments Reaktor offers extremely and graintable position, and there's a powerful granular synthesis facilities, as familiar Smoothing control. So what showcased in its Travelizer instrument. sets this apart from, say, Malström or Kontakt? Firstly, the Grain Cloud module at the heart of this instrument has a parameter called Distance which sets the rate at which grains are triggered. This means that, as the current playback position moves around the graintable, you can fire off as many or as few grains as you want. The Grain Cloud module can overlap up to 1000 grains at once, so the output signal is the composite of many tiny portions of the sampled waveform. The final ingredient is the inclusion of Jitter inputs on Grain Cloud. These allow you to add varying degrees of random 'jumpiness' to several of the main parameters, namely Pitch, Position, Length, Distance, and Pan. Now, begin to imagine how things come to life when combining all these things: grains of sound are fired off from across the original sample, some are clustered in small recognisable sequences, while others are thrown in at random. The length of the grains and rate at which they appear and disappear is chaotic, and they smear out across the stereo field, overlap, and become a boiling swarm. A soundscape builds up that's like nothing you've heard before, yet the chaos and movement tricks your brain into thinking it might somehow be natural and not a synth. From this point you can mould and constrain the sound with all the familiar tools — filters, envelopes, and effects — to create a playable musical instrument, or just enjoy it for what it is.

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Granular Synthesis

Advanced Possibilities Most of what we've looked at is the brand of granular synthesis that uses a chopped-up audio sample as the source of sound grains. This is because the large majority of music products available that employ granular synthesis work this way. However, this is only a partial view of what can be done. For a start, it's perfectly possible for software to use a live audio input instead of an audio file. Computers are fast enough to chop a signal into grains on the fly, then synthesize and mess with them, all in real time. This is how granular synthesis-based effects, such as Spektral Delay, KTGranulator, and many Pluggo plug-ins, work. Most realtime pitch-shifters and vocal processors are likely also to be taking a granular approach.

Here you can see the internal construction of the Travelizer instrument, centred around Reaktor's Grain Cloud module. The window on the left selects samples from your computer's hard drive for granular processing. The left-hand side of the Grain Cloud module has a long list of nodes, with cables attached from various controllers and other modules. This represents all the parameters that can be controlled and modulated, and is the key to the extraordinary variety of sounds the unit can generate.

Mentioning Spektral Delay raises the topic of other methods of granular synthesis that have rarely seen the light of day. Everything we've looked at so far uses grains based in the time domain, but it's also possible to split up sounds by frequency and then resynthesize them, as Spektral Delay does. The next logical step will be for synths to do away with sampled or digitised audio sources altogether, and synthesise their own grains from scratch. This would be like a two-stage synthesis process, with the first stage generating an array of grains and envelopes, each probably one cycle in length (and known as a 'wavelet'), which would then be synthesized by the second stage. Something close to this could probably be built in Reaktor, using the Grain Delay module, so if you get a few months off, there's a challenge! Granular synthesis is likely to find its way into many more instruments in the future, and is perfect for those days when you're bored of the same old array of re-created analogue sounds. Not only do granular synths create dynamic, organic sounds, they have an untamed quality and often produce unexpected treats that turn into song ideas. In fact, if you produce ambient or film music, a decent granular synth can do half your job for you! Published in SOS December 2005

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Granular Synthesis

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

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

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Making A Living From Music For Picture

In this article:

Making A Living From Music For Picture

Who Am I To Tell You All Part 1 This? Published in SOS December 2005 Married To The Job Gear Doesn't Grow On Trees Print article : Close window Working For Nowt Technique : Composing/Arranging Money Matters Family Affairs Feedback Welcome? Support Issues Still Up For It? Writing music for picture seems

like the ideal career. You get to work in your studio for a living, you can earn good money, and there's so much potential work: action films, travel and nature documentaries, romantic comedies, cartoons, low-budget sci-fi, even breakfast cereal ads. But how do you break into this lucrative world? As we find out in the first part of this new series, the first thing you need is determination... Hilgrove Kenrick

I'm sure this isn't just true of those who write music to accompany moving images for a living, but I admit that if I'd known what I was getting myself into when I decided to try to make my career in this field, I'd have given up on the idea (and, in my case, gone straight back into IT consultancy). Over the years I've been writing music for picture, I've been asked many times how to break into this line of work, and always, my simple advice is 'don't do it!', simply because it would take far too long to convey the finer detail of just how awful — but also just how rewarding — such a career choice can be. Over the course of this series, I hope to be able to make some of these finer points clear, so that you can make your mind up for yourself whether it's for you. You'll find plenty of tips and tricks of the trade here, some straightforward, some technical and even some to do with your emotional attitude to this kind of work — in my experience, this is an important aspect of the job that is often overlooked. Of course, this series isn't designed to provide exhaustive advice on how to be successful in this field, nor is that even possible — there is no one path to achieve that, any more than there is in the music business. But after reading what I have to say, you might at least have better advice available to you than I ever did. When I was asked to write this, friends joked that I'd be telling everyone how to file:///F|/SoS/SoS%2012-2005/musicforpicture.htm (1 of 8)11/23/2005 3:05:54 PM

Making A Living From Music For Picture

do it, and thus do myself out of a job. Well, not really. You might think that anyone can do it — like driving a car. In response, I'd say that while lots of people drive cars, few of them earn as much money as Michael Schumacher from doing it, and it's similar with music-for-picture work. Just as the man in the Ferrari shirt got where he is today by being blessed with instinct, drive, ambition, confidence, self-belief, natural ability and a fair amount of luck — qualities that not everyone possesses — not everyone is destined to make it as a wealthy composer of instantly recognisable film themes. Commitment is a word that sends many musicians running, but if you're not putting in your all in this job, you're not trying hard enough. You'll frequently be called upon to put your heart, soul and bank balance on the line simultaneously, and then expect to lay your sanity next to them. I would be doing you all a disservice if I didn't tell the truth, so that's what you'll get here. But then nothing in this industry is easy, or everyone would be doing it; there will always be a cost, be it personal or financial.

Who Am I To Tell You All This? What gives me the right to tell you the best way to forge a career in music for picture? Well, as you won't be surprised to learn, that's been my chosen path for the last few years, and much of the information and many of the stories that you'll find in this article are based on personal experience, although some of them also come from colleagues and friends, many of whom are far better qualified than I. I might not be as high up the ladder as some of them, but I have been fortunate to make it further than many. I was very lucky in my first few years to bag two film scores before I'd done anything for TV. These breaks had little do to with skill, however, more youthful naïveté — and I made some huge mistakes and other errors of judgement that with hindsight could have been avoided. Ill-health pushed me out for a couple of years, and circumstances demanded I find a sensible job. IT consultancy beckoned, and I joined the massed ranks. After another couple of years of enjoying the money, I realised I was hating every moment of it — Ionian and Aolian modes are one thing, TCP/IP and IPX/SPX are quite another. At around that time, though, I was lucky enough to meet a lady (now my wife) who was adamant that you should follow your dreams. She'd been on the corporate treadmill for years and was likewise heartily sick of it. With her support, I was able to once again turn my back on IT and return to what I really wanted to do. I might not have started working with Spielberg yet, but I've come further both professionally and creatively in the past three years than in the six that went before. From a battered showreel and outmoded equipment, I am now once again up at the cutting edge, and working with three production companies. The beneficial effects of emotional, as well as financial support should never be underestimated in this line of work!

Married To The Job Firstly, it's almost impossible to pursue a career in TV while holding down a regular job. There simply aren't enough hours in the day, and if you do get the file:///F|/SoS/SoS%2012-2005/musicforpicture.htm (2 of 8)11/23/2005 3:05:54 PM

Making A Living From Music For Picture

chance at a pitch or even a commission, you'll have to come up with the goods in a ridiculously short space of time. There will be no time to go to work; you'll have to drop everything and work instantly. Writing music for picture is not like being in a band — you don't gig at night and work by day, and moreover, you certainly won't have six or 12 months to deliver a finished album. It's sometimes more like 12 hours! If you think I'm exaggerating, here's a real-world example. I got wind of an upcoming TV series last year, and when it went into pre-production, I approached the directors and the producer and put myself forward for consideration. I was asked to submit a showreel, plus a couple of ideas for the series. Everything went silent for six months, the shooting date came and went, and I assumed I had been unsuccessful. Then out of the blue, I got a call telling me a draft was on the way, and asking if I could write several cues to pitch for it — by the following Monday.

Of course, you'll need some gear to create all this music with, and there'll be more on what exactly you're likely to need later in this series. But not everybody's music-for-picture setup can be as sleek and lovely as that of well-established music-for-film composer Barrington Pheloung (above, as it looked in 1999)...

I'm used to this kind of thing, but this one was a shock nevertheless — it was Friday, and eight days before my wife had given birth to our first child, so to put it mildly, my mind wasn't on the job and I was exhausted. However, this is the life I have chosen, so my wife was literally left holding the baby, and I shut myself in the studio for four days. In that time, I had a grand total of six hours' sleep, but on the following Monday, the pitch was on the director's desk as promised. You might think that doesn't sound too bad — we've all stayed up through the night working on mixes we desperately wanted to finish, haven't we? However, bear in mind this was purely for the pitch — I hadn't signed any contracts, and there was no guarantee of getting the commission. And of course, no-one ever gets paid to pitch. My wife was off work caring for the baby, too. It's not exactly a life replete with financial security... Here's another scenario, one all too familiar to me. You're beginning to make a name for yourself and the work is coming thick and fast. Ther next few months look good, because Commission A is in February, B is in March, and C is in April. But then Commission Z runs over from January, and you have work like lightning to get A done on time. And then B is delayed, but C is brought forward. Come the end of March, you have two production teams breathing down your neck at the same time... and the money from Commission Z still hasn't come through — much less A, B, or C! With time pressures like these, you can see why hanging on to a day job is next file:///F|/SoS/SoS%2012-2005/musicforpicture.htm (3 of 8)11/23/2005 3:05:54 PM

Making A Living From Music For Picture

to impossible. But what about the financial pressures? If you're like most of us, you need the money from your day job to survive. How do you pay the bills when no one is paying you? The days of understanding and supportive bankers are long gone, so don't expect understanding when you ask for help from them. It's something of a catch-22 situation — you can't get the work without being ... the author's rather less beautiful setup is more like the kind of thing you're likely to be prepared to commit the time most working with as you try to get your career off people devote to their nine-to-five, but the ground! without the salary from a regular job, you won't be able to support yourself and your dependents, if you have any, much less buy gear or support a studio. I'm afraid there's no easy answer to this one, and what suits one person won't work for another. What's more, it's hard to say "I'll try it out, and if I don't make it in six months, I'll go back to my old job." You'll be forever telling yourself that the big break is just around the corner. And even when you get work, it doesn't always lead onwards and upwards. I got lucky and scored my first film at the tender age of 20 — but then I didn't get the chance to do another for two years! One possible middle way is to take one job — a part-time one — to fund the other. For example, if you're a trained musician, you could take a peripatetic teaching job. For those like me, with a background in IT, a sales job in a PC shop (even PC World!) or even better, a part-time post as an IT consultant could give you a little something to keep the roof over your head. And you may need to tighten your belt — give up smoking, and have fewer takeaways and nights out on the town. Whatever your vice, the money has to come from somewhere.

Gear Doesn't Grow On Trees And of course, if you're reading this magazine, it's unlikely that you compose on a ukelele and an ocarina, with a stack of manuscript paper and a pencil. What about your studio gear? Do you have the right gear to compose film and TV music? You're probably going to have to update a few things, and of course that costs money too. For example, if you're called upon to create the soundtrack to a documentary about life on a remote Pacific atoll, you're unlikely to have the director's required indigenous instruments to hand (or the knowledge of how to play them). You're going to need sample libraries, and good ones at that — and they don't come cheap. And supposing someone sends you a draft of the documentary on VHS — are you going to be able to lock that to your sequencer? The chances are you'll need a synchroniser and all the cabling to go with it. This kind of expense soon adds up.

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Making A Living From Music For Picture

Working For Nowt To add further to your financial woes, when you're trying to get a career in film and TV music off the ground, you'll often find people asking you to work for nothing. But before you turn such an offer down in disdain, stop and think. At this stage, getting something on the CV is more important than the potential reward. The chances are that if there's no budget for music or anything else, it's probably a student film or charity corporate video. Ask yourself one simple question — what's in it for you? If no one is ever going to see it outside the Student Union, and you have nothing better to do, do it. After all, it's good practice. And if the film's going to a festival, and a big one at that, then you'd be mad to turn down the free publicity. On the other hand, if the filmmakers have the funds to get it that far and they're touting shiny cameras, you might want to enquire why they expect you to work for love alone... Whatever you decide, make sure you get something out of it, be it practice, exposure or expenses, if nothing else. Also, look at the number of film and TV directors who always work with the same composer. The simple reason is they're mates and work well together, and the best way to forge such a relationship is to get in there early. If the budding Spielberg goes places, the chances are that he could take you along with him.

Money Matters When you hear of some of the fees commanded by top composers for music-forpicture work, it may sound like easy money, and conjure up fantastic ideas of what a lazy, well-paid life they have, but when you break it all down, you begin to realise that there are some philosophy students getting better wages at fast-food outlets. To prove what I say, here's a breakdown of a typical project, based mostly on experience and fees from real jobs. Imagine you've been commissioned (a whole job in itself, and a part of the process we'll look at in more detail later in this series) to write the music for a TV series involving sailing, set on rivers and at sea. It comprises four episodes of 56 minutes each, with 28 minutes of music per episode, including the series theme and cutaways. That's 112 minutes of music in total. Your budget is £2500 per episode — but you only have four weeks to deliver the goods. This kind of timescale is entirely typical — in fact, if anything, it's rather more time than you might be given! EPISODE ONE With one week to get this part finished, and by limiting yourself to an hour's sleep a night, you work 161 out of a possible 168 hours, and you need to buy two sample libraries of suitably nautical musical instruments and sounds at £350 and £500 respectively. EPISODE TWO

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Making A Living From Music For Picture

You're getting the hang of the job, but it's still hard going grabbing two hours' sleep a night. You work 154 hours this week, and you need to hire a session guitarist for 20 hours to play some parts in the series theme and incidental music. That sets you back £700. EPISODE THREE You're in full swing now, but schedules are pressing, so you work 154 hours this week. EPISODE FOUR Disaster — your computer blows a fuse, the PSU dies and takes your dual hard drive array with it. Somnambulism gets you through the week — you work 166 hours. Oh, and the director loved the guitar and wants more, even though you thought you were finished with the session guitarist. You hastily rejig your score, write new parts for guitar, and call your man back in. A new PSU costs you £60, replacing your two hard drives and controller sets you back £470, and the guitarist is a painful £1400. FINAL ANALYSIS Congratulations — you're done and dusted on time, and now there's a cheque for £10,000 on its way to you. But if you break down the time and money you've spent on the project, it doesn't look quite so rosy. You've earned 10 grand, but spent £530 on new equipment, £850 on necessary samples, and £2100 on a session guitarist. That leaves £6520. And if you divide that by the total number of hours you worked (635), you realise that you've earned just over a tenner an hour. Or, put another way, just over £58 per minute of music. Doesn't look so great, now, does it? And that's before you've hired an arranger or psychiatrist to get you through it! That's also assuming a few hours' sleep a night, and sometimes you won't even get that. But surely these figures aren't typical? Well, obviously, not every job is paid like this — rates vary with your experience and according to what the programmemakers think they can get away with. If you want a rule of thumb, the chances are that if the job is important, high-profile and will do wonders for your career whilst being a monumental undertaking, you'll be paid in play-dough and string. The mad world of the media being what it is, things can also work out the other way, whereby you land a job bodging together hackneyed rubbish that's only ever shown on Serengeti TV at 2 o'clock in the morning, and are showered with gold for it — but please, don't count on it!

Family Affairs It's not just the financial aspect to this life that you may find unsettling. If you have family or other dependants, what about them? Opportunities for sleep are few and far between when you're struggling to get a commission finished on time, and when you come out the end of it, you'll have a mountain of chores and duties to fulfil before you can put your feet up. And if there are children to care for, how is your other half going to cope while you're locked away trying to work out why your sequencer's lost sync with the video for the third time that day? When you've finished, and all you want to do is stagger upstairs and collapse into bed, they'll be needing help, as they'll have done nothing but wash, cook, clean and child-mind (and possibly hold down a job of their own) since you file:///F|/SoS/SoS%2012-2005/musicforpicture.htm (6 of 8)11/23/2005 3:05:54 PM

Making A Living From Music For Picture

disappeared into your studio. Imagine another scenario. You've booked a holiday somewhere, and the night before, or the morning of departure, a possible job comes up. Do you take it or leave it? Logic would dictate you step on the plane, but if you turn the client down, they may never ask you again. Can you afford to take that risk? Fix in your mind your reasons for trying to make a career of this — is it for love of music, love of money, or is it just a job? Frankly if it's any of those three, I'd advise you to stop wasting your time. You'll be asked to write styles you hate, and the money is risible until you're four rungs up the ladder. And if it's an ordinary job you want, why are you still reading this article? Whichever way you look at it, it's got to be worth it. Speaking personally, I could no longer conceive of doing anything else — but that doesn't mean there aren't days when I question what it's all for, and if it's really worth it!

Feedback Welcome? Now, have you ever played back one of your own compositions and sat back thinking "my goodness, I'm a genius!" only to feel utterly deflated when If you're asked to write music to accompany someone picks holes in it? Out in the documentaries about far-flung corners of the Earth at the beginning of your career, it goes world of film and TV composition, it's a without saying that you won't have the hundred times worse. You won't budget to record nose-flautists of the usually have the time to polish tracks Kalahari in their natural habitat. Fortunately, as much as you'd like — it often feels that's what 'world' and 'ethnic' sample as though there's always a courier collections are made for! You may be seeing a lot of discs like the ones shown here... waiting on the doorstep to take your finished masters to the dubbing stage. And worse still, you'll rarely, if ever, get any feedback. Granted, if you get the commission, then clearly you were up to the job. However, I can count on the fingers of one hand the times I've had feedback on a showreel — usually it's "yes, here's the job", or absolute silence. If you're a sensitive type, this silence can be soul-destroying. You never know if it was something you did or didn't do, or because of other factors. These might include someone being better than you, the company liking your work but needing something in a slightly different style, which someone else was able to provide, or the director's cousin's sister's former roommate pitching at the same time as you! It might even be down to your CD not having turned up in the right place at the right time. Of all these possibilities, only one can be ruled out to everyone's satisfaction — Royal Mail Special Delivery was created for a purpose! The rest is up to you to overcome... file:///F|/SoS/SoS%2012-2005/musicforpicture.htm (7 of 8)11/23/2005 3:05:54 PM

Making A Living From Music For Picture

Support Issues It's a life of ups and downs. Just imagine; you've canned your day job to make a go of music, stuck your neck on the line, got a pitch and then heard nothing. The company doesn't answer the phone, and the only calls are from the bank saying they won't extend your overdraft. It's got to hurt, and it does. Who's there to catch you when you fall? Emotional support is crucial, and you'll get nowhere without it. To mangle the famous phrase, behind every successful composer is a caring friend — and it doesn't matter if that person is your significant other, a parent, or just a mate. If you know someone who's trying to make it in the same game, they can be the best help of all — when you witter on about ego-crazed directors, equipment woes or composers' block, they'll understand better than anyone! I'd love to say that it gets better with time. Well it does and it doesn't. If you're prone to worry, that trait is only going to get worse, and only care and support will keep you on your feet. On the other hand, if you're so laid back you're horizontal, then you don't have a hope of seeing it through — quit now, while you're ahead.

Still Up For It? Thus far, I have hopefully scared and excited you in equal measure, and with a bit of luck, some of this will serve to keep your feet on the ground over the coming months. Still up for it? What part of "Don't do it!" didn't you understand? Still with me? Then read on next month... Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Mixing Live Recordings In Logic

In this article:

Recording Gigs Into Logic Logic News Mix Processing Copying Mix Settings & Automating Finishing Off Have Your Say!

Mixing Live Recordings In Logic Logic Notes & Techniques Published in SOS December 2005 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

Mixing live band recordings within Logic presents a unique set of challenges, so we show you how to get great results with the minimum hassle. Paul White

Whenever I record live gigs of any complexity, I try to use my Alesis HD24 hard disk recorder, then transfer the files into Logic for editing and mixing. This is simply a personal preference, as hardware always feels more solid at the crucial recording stage, where you simply can't afford to have an 'Unexpectedly Quit' incident during a one-off performance. Invariably this means having long files to deal with, and if you're importing these via Firewire rather than playing them across in real time, there's no simple way to shorten the files prior to import. However, if you can stop and start the recorder between songs and switch to a new song file, it can help break the performance up into more manageable chunks — it all depends on how much time you get between songs. As you may have already noticed, Logic imposes a limit on the length of each Song file, expressed in terms of bars and beats, so if your files won't fit, your only recourse is to reduce Logic's tempo until they do — this doesn't affect the way the audio plays back, of course. If you're going to adjust the tempo after importing, that does mean that all your files need to be continuous and to start at the same time — which is the case when importing from an HD24 via Fireport. If they're not, set your tempo first and check that the Song end point is at a time greater than the length of the audio you're about to import — then leave the tempo alone.

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Recording Gigs Into Logic Should you decide to record entirely within Logic, which is perfectly practical given a large enough audio interface, it is important to set the tempo low enough that the whole set will fit, so that the recording won't grind to a halt prematurely. However, it is equally important not to set the tempo any lower than it needs to be, otherwise page scrolling may not be as smooth as you'd like, given that scrolling takes place after whole bars, and if half a verse of music fits into a single low-tempo Logic bar, navigation is going to be rather coarse. Editing files of up to an hour in length is clearly very cumbersome, so my first task is usually to cut the raw recording into song-length chunks, saving each chunk as a new Logic Song and setting suitable start and end bar locations. It is important that you do cut up the file, rather than simply dialling in the required start and end points, as Logic will otherwise save the entire length of the audio file for each project, even though you're only using a small part of it. If you wish to retain the live feel, it's also important to retain the audience ambience and (hopefully) applause between songs. It's generally easy to crossfade audience ambience if you need to shorten the gaps.

If a player switches instrument partway through a track — for example here the flute player switches to tambourine — it's worth keeping the two sections of audio on separate tracks, feeding separate Audio objects, so that you can process them independently.

Ideally, everything will always be on the same track for every song, but this isn't always the case. For example, the same track may have an acoustic guitar on one song and a flute on another if the player switched instruments, so the simplest way to proceed is to create a track layout that can be used for all the songs, with just one instrument per track. You can then drag the files between tracks if they don't line up when you first import them. This invariably means some tracks will be empty, but it doesn't matter as long as the same instrument is on the same track all the way through the mixing project.

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Logic News Apple have released a free Logic 7.1.1 update, which addresses a variety of longstanding complaints from Logic users. OMF2 is now supported, which will please many Pro Tools users, and the annoying limit of 25 minutes per audio file when exporting XML has been removed. A widely reported problem was that a CD burned directly from Logic would, in some circumstances, end up full of static noise. Although my record collection contains a few albums that sound like this intentionally, for most users it's been an annoying problem which I can confirm is now fixed. The TDM usage meter (which was broken in Tiger) now works again, and DAE users can also, finally, bounce files in WAV format. Global tracks were a really neat addition to Logic 7, and it's nice to see that the Global Tempo track now properly displays time-signatures other than four/four. This will be welcomed by those of us who revel in using odd time-signatures, as will the fact that Apple Loops now work with compound times. The update also contains several plug-in bug-fixes, and Ultrabeat and Sculpture parameters now display correctly in the automation pull-down menu. There's a personal 'hurrah' from me, as Apple have fixed the automation feature whereby you can highlight an area on a track while holding down the Shift, Alt, and Apple keys to create four automation nodes — in earlier versions only three nodes were created. Last, but not least, some of you may be relieved to know that EXS24 now supports more than 32767 sampler instruments... Both Logic Pro and Logic Express have been updated, and more details can be found at www.apple.com/ support/downloads. A week or so before the Logic update, Apple also released a welcome bug-fix for Waveburner. One of the biggest problems for me was the handling of plug-in presets, and this now seems to have been fixed. I'm getting fewer coasters too, but I still don't feel confident enough to send out a Waveburner CD-R without a good critical listening first. Unfortunately, one of the features that wasn't added to Waveburner v1.1.1 is plug-in delay compensation, which means that the Song Position Line is still out of sync with the audio playback and waveform display. Stephen Bennett

Mix Processing Hopefully, most of the Songs will require similar treatments, so my usual strategy is to first set up a mix in just one Song as best I can, with all the necessary plugins in place. If additional overall plug-ins, such as a limiter, are needed on the main stereo mix, I add those here too. I usually have a couple of reverb sends, plus compression and EQ for those instrument and voice tracks that need it. In the case of DI'd acoustic guitar and similar instruments I have occasionally had the player come into the studio to record a short sample of the instrument using a microphone, so that I can use the Match EQ plug-in to make the DI'd sound closer to the miked sound. Where sounds have become indistinct through compromised miking, the Exciter plug-in can add a useful amount of definition, but you need to use it sparingly to avoid harshness. Rolling off unnecessary low end using an 18dB/octave low-cut file:///F|/SoS/SoS%2012-2005/logicnotes.htm (3 of 6)11/23/2005 3:05:57 PM

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filter may also be beneficial in improving separation. If you have a TC Electronic Powercore system, the included Character plug-in can also be very useful in adding focus and definition to instruments in a rather more wide-ranging way that can simple high-end enhancement. The Audio Configuration window shows all Gates are rarely of much use in mixing the audio objects available to your current the miked portions of live recordings Song. One extremely useful feature of this given the amount of spill that tends to window is that you can copy and paste entire audio setups between different Songs — an occur, but there are still some useful invaluable time-saving measure if you're plug-ins that can help clean things up. For example, hums and lighting buzzes working on a number of Songs with the same instrumentation. can be much improved by tuning a series of very narrow parametric cut filters to multiples of the fundamental hum frequency (50Hz or 60Hz depending on your location). Automating high-cut filters can also be a useful way of cleaning up decaying notes before pauses, as most acoustic instrument sounds naturally lose high end more quickly than low end as the notes decay. By getting the filter frequency to fall to, say, 700Hz over the duration of the final note, you'll lose any high-frequency noise or spill, but without compromising the natural sound of the instrument.

There are some exceptional situations where gates can come in useful, typically for loud close-miked sounds that don't suffer from much spill. Electric guitars and basses are good cases in point, because, although spill is rarely an issue, amplifier noise can be obtrusive and responds well to gating, though the sound will be more natural if you set the gate to attenuate by a few decibels rather than shutting off the signal altogether. Gates tend to be less suitable where delay and reverb effects are being used, and even extending the gate's release time is unlikely to be entirely successful, as the gate is bound to compromise the effect tail to some extent. You can always add a little more reverb when you mix to help disguise the effect of gating, but there's little you can do about delay, so using denoising software is probably a better solution. At the time of writing, Logic doesn't have an effective denoiser, but there are several third-party options available, most of which work from a noise-only fingerprint. Alternatively, if you have any say in the way the gig's set up, you can suggest that the guitar player use a gate before his delay and reverb effects, as this tends to sound far more natural than trying to gate afterwards.

Copying Mix Settings & Automating Once you've gone to all the trouble of setting up everything for the first Song, you have the option of saving a copy of that Song as a template for later use. file:///F|/SoS/SoS%2012-2005/logicnotes.htm (4 of 6)11/23/2005 3:05:57 PM

Mixing Live Recordings In Logic

However, rather than importing all the audio files from the individual Song folders into an empty version of your template, you can simply go to the Audio Configuration page, Select All, Copy, then Paste the configuration into the Audio Configuration page of the next Song. What this does is copy over all the plug-ins and settings just as they were in the first Song, which saves a lot of messing around. This is why I feel it is so important to keep all the instruments on the same tracks throughout — this way you can be sure that the plug-ins you copy over will be applied to the correct tracks without any further juggling on your behalf. One thing you can't set up for all your Songs, however, is automation, but level automation is a huge help when mixing live material, as it allows you to turn down tracks when they're not in use, or tweak the levels of odd notes that are too loud or too quiet. It also helps you ensure that each track starts cleanly and ends smoothly. Of course The Character plug-in available for TC audience ambience will also be Electronic Powercore is very useful at the mixdown stage for improving the overall affected on those tracks you need to automate, which is why it helps to have definition of individual instruments. separate audience mics. Where this isn't possible, you can often copy and paste audience noise from other locations in the recording and then layer it over the song transitions in a natural way so as to disguise any underlying ambience fluctuations caused by level automation.

Finishing Off Once you've got the tracks as clean as possible, the mixdown stage is really no different to that of any other project, though it is sometimes necessary to cheat a little with live recordings to get them to sound tighter and better than they were originally played. Endings are particularly important, and I've often had to cut and slide the last drum hit or bass note of a song to get it to line up with the other instruments. How far you go with this is up to you and the artists involved. As you've probably gathered, the secret to hassle-free mixing of live material comes in establishing a system that avoids you having to do the same thing many times. Copying configurations to carry over plug-in allocations and settings is extremely useful, as is setting up a default Song with consistent track allocations and suitable send effects already configured. Inevitably some treatments will have to be devised on a Song-by-Song basis, but you'll be surprised how much stays the same once you have the plug-ins set up for the first Song.

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Mixing Live Recordings In Logic

Have Your Say! If you want to suggest changes or improvements to Logic, then here's your chance! The Apple development team are inviting SOS readers to send in their suggestions of what they'd most like added or changed in Logic. Email your top five suggestions (in order of preference) to [email protected], and we'll forward your lists on to the Logic team. We'll be asking them for feedback on which changes users deem most important and how these might be addressed. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

file:///F|/SoS/SoS%2012-2005/logicnotes.htm (6 of 6)11/23/2005 3:05:57 PM

PC Notes

In this article:

PC Notes

PC Snippets The Perfect Audio Interface? XP x64 News, PC Tips Soundfonts Live Again With Published in SOS December 2005 Synthfont Print article : Close window

& Updates

Technique : PC Notes

The 64-bit Windows XP x64 edition is on the shelves, but musicians should stick with their trusty 32-bit OS for the moment. PC Notes explains why, as well as offering some constructive soundcard feature suggestions to manufacturers. Martin Walker

Windows XP Professional x64 Edition is now freely available for between £90 and £100 from various UK suppliers. As with previous OEM releases, however, Microsoft stipulate that you have to buy it either with a "fully assembled computer system" or with a "non-peripheral computer hardware component" — which basically means something like a RAM upgrade, motherboard, hard drive or CPU. There's been plenty of anticipation about Windows 64-bit computing, but in practice it's still proving a far from pleasurable experience for many users, due to the continued lack of 64-bit drivers for so many hardware items. By way of example, Dell are selling some of their PC systems with 64-bit Windows preinstalled, but you get the following warning when you try to buy them on-line: "Peripherals you currently own or plan to purchase in the near future (cameras, printers, MP3 players, handheld devices) will most likely NOT work on a system purchased with Windows XP Professional x64, and some software applications may not work on the x64 operating system." Windows XP Pro x64 is clearly not a mainstream product, nor for the faint of heart, and one mainstream PC review stated that "only software developers and high-end workstation users will see real benefits from Windows XP Professional x64 Edition; everyone else should stick with 32-bit Windows XP instead." This is hardly surprising, given that the 64-bit version has basically the same look and feel as its 32-bit cousin, and provides roughly equivalent performance when running 32-bit applications using its WOW64 (Windows On Windows 64) translation layer. For once, even gamers aren't too impressed, because until 64bit game versions appear there's little practical benefit from the move, and a few file:///F|/SoS/SoS%2012-2005/pcnotes.htm (1 of 5)11/23/2005 3:06:00 PM

PC Notes

32-bit games won't even run on Windows x64 at the moment. Unfortunately, musicians are one of the groups who can definitely benefit from a totally 64-bit PC once everything is compatible, because having a fully 64bit system can boost the performance of typical music applications by 30 percent or more, and for anyone who It may be available, but is it yet desirable? needs more RAM to hold samples, a Catch up with the current Windows x64 fully 64-bit system removes the 4GB situation in the main text. ceiling of 32-bit systems. However, while some music developers have been super-keen to 'go 64-bit', there others who haven't committed themselves yet. Leading-edge users have been quick to voice their opinion that no company should have released any product requiring Windows drivers over the last six months without including 64-bit versions. However, it's possible to see it from the other side: a tiny percentage of users have so far moved over to x64, so smaller companies must find it hard to justify the cost of developing new drivers that they will have to supply as free updates to such a small number of customers. On the corporate front, some businesses are also avoiding the move to 64-bit Windows because their favourite security applications, such as virus checkers, won't work with it yet. Conversely, while some developers have released 64-bit updates for the corporate versions of their applications, they won't say whether or not 64-bit versions for home and small business users are planned. Some early adopters have already given up, reformatted their hard drives and reinstalled their 32-bit version of Windows XP, just so that they can continue to use their printers, scanners and soundcards. Meanwhile, musicians who can't wait for the dust to settle are better off running a dual-boot setup with both 32-bit and 64-bit versions of Windows: they can then get up to speed with all the new stuff without losing the use of their favourite hardware or software. The 64-bit Windows revolution may have happened, but I suspect that the majority of users may wait until Windows' 64-bit 'Vista' operating system is released before taking the plunge.

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PC Snippets With many PCs now relegated to 'under the desk' status, where their front-panel indicators may end up difficult to see unless you're a contortionist, Lonewolf's tiny Hard Disk Indicator utility may prove very useful. It adds an 'LED' to your system tray, displaying drive load and save activity. Read and write activity on up to five drives or partitions (including CD or DVD drives) can be monitored on a single LED, or you can launch several instances of the utility, each monitoring a different one. You can also choose between six colours of LED, and control the duration of each 'flash', so that you won't miss it. This is certainly handy for spotting unexpected activity on a particular drive, monitoring background tasks (such as defragmentation kicking in) or keeping an eye on the progress of software downloads. Another handy utility in this freeware range is Indicators, which displays the current CPU load and free memory, in terms of percentage, on your System Tray — just the job when trying to shoehorn yet another RAM-based instrument into your soft sampler. You can download both the utilities I've mentioned from www.lonewolf.gr/ software. I mentioned Rightmark's CPU Clock utility in September's PC Musician feature, to help anyone with unexplained laptop audio problems to find out if such problems were related to CPU throttling (automatically switching processor frequency to save power, which can sometimes cause audio clicks and pops during the transition). However, it seems that there's another possible culprit for these gremlins. SOS Forum contributor Olafmol managed to cure an annoying problem with CPU spikes and audio drop-outs with his laptop and PCMCIA audio interface by disabling Microsoft's Smart Battery devices inside Device Manager. Sometimes these smart devices aren't as clever as they think they are! Just two months after Apple announced that their Mac OS X operating system will be able to run on Intel's x86 architecture from 2006, but only on x86 chips used in Appledeveloped hardware, hackers have posted instructions on several web sites for how to bypass the security chip and run the developer's version on any PC. The hack isn't legal, since you need to download two modified files to get around the security technology, but it proves just how keen many PC users are to try Mac OS X.

The Perfect Audio Interface? I know that a lot of audio interface manufacturers must read this column, at least on an occasional basis. So here are a couple of feature suggestions for future products that I hope someone will act on. First, and most important, over the last year I've heard of lots of musicians plugging their new audio interfaces into their PC laptops and experiencing the annoyance of a host of background noises related to hard-drive activity, mouse movements and graphic redraws. These problems are all related to ground loops, and solving them often requires the user to place a DI (Direct Inject) box between the audio interface's output and the amplifier or mixer it's plugged into. Suitable stereo DI boxes start at about £20, but those with full galvanic separation (ie. using an output transformer) to guarantee that the problem will be cured tend to be more expensive — and if you've bought an interface with eight outputs, you'll need a DI box on each one, which is not an elegant or cheap solution. So the market is crying out for a manufacturer to launch a reasonably priced USB or Firewire audio interface with file:///F|/SoS/SoS%2012-2005/pcnotes.htm (3 of 5)11/23/2005 3:06:00 PM

PC Notes

transformer-coupled outputs. The second feature that I feel many musicians are crying out for is an analogue monitoring output with a hardware level control, so that the interface can be plugged directly into active monitor speakers. More and more musicians are finding that they don't need a full-blown analogue or digital mixer to handle their inputs and outputs, since these are all directly connected to the interface. So the sensible solution is to connect the main output of the interface directly to the power amp and speakers, which has the added advantage of keeping the signal path simple, After all, most small mid-priced mixers (£500 to £1000) colour the signals passing through to a small extent, while budget ones can make even more of a difference, as you'll soon hear if you temporarily bypass them. The problem with direct connection at the moment is that the only control you have over output level is generally from the software Control Panel utility of the interface. Such level controls are almost always in the digital domain and therefore degrade the audio signal once you turn them down. You also run the risk of this software control accidentally defaulting to its 0dB level and damaging your amplifier, speakers or (worse) ears. Many interfaces already provide a headphone output with an analogue level control, but these are not only unbalanced (giving rise to possible ground-loop problems again, if plugged into mains-powered speakers), they also often use a lower quality output stage than the main one, optimised for low-impedance headphones. You can, of course, buy dedicated DAW Controllers with large knobs, extra talkback functions, a clutch of switched stereo inputs (for adding CD players and the like) and switched studio monitor outputs for listening to the mix on several speakers, but many musicians don't need all these, or the extra expense. No, what we want is a handy, high-quality analogue output-level control on the interface monitor outputs, and preferably a stereo/mono button as well. These would cost very little to implement, but I guarantee that they would be a big selling feature. Over to the manufacturers...

Soundfonts Live Again With Synthfont Soundfonts... now there's a word that should conjure up fond memories for many PC musicians. Having a hardware-based, multitimbral sampler on your Creative Labs Soundblaster soundcard, courtesy of Soundfonts, was incredibly useful, and many musicians were disappointed when they upgraded to more exotic audio interfaces and found they could no longer use their collection of Soundfont sample collections. Admittedly, Microsoft do include their GS Wavetable synth with Windows, for basic playback of MIDI songs, but this is no substitute for being able to use your own collections of samples. Well, fret no longer: Synthfont is a freeware tool, written by Kenneth Rundt, that emulates Creative's sampler in software, so that you can play back all your old MIDI files that used Soundfonts. The stand-alone version offers a huge range of different windows, including a Files/Explorer window, a Piano Roll, and an Event display, and offers control over SF Layers, Splits and Waves, with various basic

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PC Notes

recording and editing functions so that you can enhance the end result. Even better, you can incorporate VST plug-ins into your old songs to beef up the Soundfont instruments, or replace them altogether with VST Instruments, and there's a separate prototype Soundfont Editor to try out as well. For those who have lots of Soundfonts but prefer to use a VSTcompatible sequencer, Kenneth offers the freeware VSTSynthFont, Got any Soundfonts lying around that you a VST Instrument into which you haven't been able to hear since you load Soundfonts and use as a abandoned an old Soundblaster soundcard? Try the stand-alone Synthfont (and the multitimbral sampler (just drag the VSTSynthFont Instrument) for quickly DLL file into your VST plug-ins accessing those sounds. folder). His web site also includes links to hundreds of free Soundfonts. Don't assume that the Soundfont standard implies low-quality sounds. In my time I've reviewed several SF libraries that were also released in the 'more professional' Akai format, and have imported some of them to use within Gigastudio. Point your browsers at www.synthfont.com/. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

file:///F|/SoS/SoS%2012-2005/pcnotes.htm (5 of 5)11/23/2005 3:06:00 PM

Streamlining Your Workflow In Ableton Live

In this article:

Getting Started Audio & MIDI Routing Let's Go To Work Live News Making Arrangements

Streamlining Your Workflow In Ableton Live Ableton Live Notes & Techniques Published in SOS December 2005 Print article : Close window

Technique : Ableton Live Notes

We begin this new series on using Ableton Live by examining how you can increase your productivity whilst using Live as a writing tool. Ingo Vauk

Welcome to SOS's new regular workshop for Live users. As we are at the start of exploring this exciting software, I thought it a good idea to look at the beginning of the creative process. Over the coming issues, we'll cover the innovative concepts governing work within Live, and look at techniques for streamlining working practices and setting up an efficient system.

Getting Started In this first article, we'll cover the basics, running through a typical writing session in Live using real instruments, starting with some sketchy ideas in the Session View window and taking them through to the creation of a finished arrangement ready for mixing in the Arrangement View. To put you in the picture, my studio is based around my computer and a Digidesign Digi 001 interface. I also have a lovely old Wurlitzer piano, a couple of nice microphones, some guitars, a few synths (soft and hard) and more samples than I will use in three lifetimes. I have a microphone, the Wurlitzer and a Roland Jupiter 8 permanently file:///F|/SoS/SoS%2012-2005/livenotes.htm (1 of 6)11/23/2005 3:06:03 PM

Streamlining Your Workflow In Ableton Live

connected to the Digi 001's inputs, and I use the monitor outputs to feed my speakers. This allows me to sit down and start work almost immediately and, because of the 001's direct outputs, I don't have to worry about latency from the external instruments. The final part of the setup is a MIDI master keyboard, which I use to play in musical parts and also to trigger buttons in Live's Session View. One of the great things about Live is that it gives you almost instant access to your sounds. Rather than have to go through the usual sequencer rigmarole Key Map Mode shows how the Clip triggers of importing audio files, having were assigned in the example Session. browsed through endless windows to find them, it's easy to save certain browser settings in your default Live Set (Live's term for a project), so that you can navigate quickly to the folders that hold your audio files. This means that you're never more than three clicks away from the next sound, regardless of its nature. Audio files and virtual instruments can be dragged onto the Session view and will sort themselves out in the mixer environment, automatically creating the required MIDI or audio tracks as you go along. For new recordings, I have created a few blank audio-input Tracks, routed from the Digi 001, with their outputs muted, as I don't want to mix the latency-delayed signal with the direct one coming from the 001's monitor output. The MIDI preferences are set to enable me to record both MIDI note data and also controller information from hardware control surfaces, which Live's manual refers to as Remote control surfaces. By sitting down and thinking about what you most often need when you start using Live, and then setting up your default Live screen accordingly, you can avoid the messing about that so often gets in your way when you've had an idea and want to start work on it fast.

Audio & MIDI Routing When I first started using Live, it took me a little while to get used to the routing methods, especially in the MIDI domain. Because of the innovative way in which Ableton approach sequencing, the terminology can also be slightly confusing. A Track in Live terms is really a channel on the mixer that may just be there in the role of a normal mixer channel, simply processing the audio it gets from its input and sending it to the output. Alternatively, it can actually hold the sound source either as a Clip or in the Arrangement. The MIDI data is taken care of in similar fashion, and can be turned into sound by a plug-in within the channel, or can be routed to another channel or external device from the output. So in the following text, you can always substitute 'track' for 'channel' and vice versa. You can specify the MIDI In source on a channel-by-channel basis to be any external MIDI device connected to the interface you're using, the computer keyboard, or the output from an existing MIDI track. To use several external MIDI file:///F|/SoS/SoS%2012-2005/livenotes.htm (2 of 6)11/23/2005 3:06:03 PM

Streamlining Your Workflow In Ableton Live

controllers at the same time, each channel/track can be assigned exclusively to an input, so that you are permanently monitoring the relevant MIDI master for the sound. Working alone and with a fixed setup, it usually suffices to set all inputs to 'All Ins' to start with, and then fine-tune your settings as you go. The MIDI output routing works along similar lines. On MIDI channels that hold a plug-in instrument, the MIDI output naturally gets routed to that instrument. In MIDI channels that are there purely for sequencing purposes, the MIDI Out can be routed to any of the physical outputs, or any other MIDI track. In this case, you have the option to send the data to the track input (one way of bouncing together MIDI data from multiple takes) or to the input of another plug-in instrument.

MIDI routing shows how MIDI from one track's output can be routed to the Instrument of another.

As with the MIDI routing, the audio signal flow in Live is slightly different to what you might be used to. Audio Tracks can source their inputs from your audio interface or any other audio track or buss within Live. This gets rid of the need to have specific group or buss channels, since any audio track can be configured to do that job. Similarly, the output of a track can be routed to any physical out, or any other audio track. The 'Sends Only' option will lift the audio from the mix buss but still send the audio via the Aux busses — very useful for feeding dedicated effects channels.

I usually set up a couple of general effects on the auxilliary sends and returns in order to cut down on plug-in use early on in the process. With the wealth of processing available from within Live and third-party products it is worth keeping processing economy in the back of your mind. It is very easy to put a couple of plug-ins on every sound, and find that you are running out of power long before you're done. So leaving some of the processing until later and using just your standard reverb on a send will help here. Alternatively, you can resample sounds immediately and disregard the processing chain that led to it. With Ableton's ingenious file-management system, it is never a problem to retrieve a sound at a later point, if you want to redo a section.

Let's Go To Work To begin, I typically select a loop I want to use to play to, and drag it from the browser into the first empty space on the right of my recording Tracks. I then press the Play button on the resulting Clip. After adjusting the master tempo of the loop, I un-mute the channel to which my Wurlitzer is routed, and set it to input in order to see the level I'm putting 'to tape'. It is good practice to listen 'through' the software at this stage, to make sure you're getting a clean signal without any clipping or similar nasties. Once I'm happy with this, and I've muted the latency-delayed signal again, I arm the track for recording (using the Arm Session Recording button at the bottom of file:///F|/SoS/SoS%2012-2005/livenotes.htm (3 of 6)11/23/2005 3:06:03 PM

Streamlining Your Workflow In Ableton Live

the mixer channel) and finally I assign my sustain pedal to trigger the Play/ Record button of the first empty slot in my Wurlitzer track. This means that I can hit the pedal when I want to start recording, and the drop-in point will be on the next global quantise value set in the transport bar (the next downbeat, in this case). When I want to record a piano part, I hit the pedal, and Live drops into record on the following downbeat. I record a few bars of the idea, and then stop the sequencer. If I'm worried about my timing at this point, I can adjust it by dragging the Warp Markers in the Clip display. You can see from this how quick it can be to record your ideas in Live. After recording a few more parts to build on the rhythm track, I'll typically spend some time creating variations and key changes using Live's various transposition, re-triggering and general mangling tools. Rest assured that we will look into these methods in depth in the months to come. Here we can see part of the Arrangement that resulted from carrying out a global recording of the Session View.

Once you've got enough raw material in the form of multiple Clips, you can think about creating an Arrangement with them. If you wish, you can try your hand at putting the Clips together as Scenes first. A Scene is a horizontal row of Clips that stretches across the Session view, and can be triggered as one entity by clicking on the Scene Launch button on the right.

To create a Scene from Clips, you trigger the Clips you want to use, set them running, and then select 'Capture and Insert Scene' from the Insert menu, or hit the 'I' key while holding down Control and Shift on a PC (Command/Apple and Shift on a Mac). Using this method, it is possible to build up a number of different Clip combinations very rapidly and try out sections of music on the fly. You can label these Scenes (for example 'Verse', 'Bridge', or 'Chorus') by selecting the Scene button and hitting the 'E' key while holding down the Control key (PC) or Command/Apple (Mac). If you're only using a few Clips as your raw material, you might not need to create Scenes, but could simply trigger Clips in real time as and when you want to hear them. Live allows you to assign so-called Clip triggers to most keys on the computer keyboard, and also assign keys to stop playback of individual Clips at any time. You do this in Key Map Mode, which you enter by hitting the 'K' key while holding down the Control key on a PC, or, once again, the Command/Apple key on a Mac. You then simply click on the function you want to assign a key to, and hit the required trigger key on your keyboard. I tend to use a Scene-based method of creating Arrangements when the source

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Clips contain a lot of fine detail, or when I want to create more traditional song structures, but when I want to create a more free-flowing, club-style track, I tend to build an Arrangement by just triggering Clips from individual keys. Whether you use key or Scene triggers is down to personal preference and the type of material you are working on. You can even mix and match both approaches if that suits you best.

Live News Ableton have released Live v5.0.2, which incorporates a number of bug fixes, including resolving certain conditions that previously led to crashes when using copy and paste. You can find full details of the improvements within the tutorial section of the update software. Coldcut will be making files from their new single available to the Ableton Live on-line community, including remix-ready stems and a vocal rant by John Spencer . Keep a look out in the Ableton artist section on www.ableton.com. Quick tip: a new feature in Live 5 allows you to move multiple Warp Markers at the same time. Simply select a Warp Marker and then hold down Shift to select multiple markers. Alternatively, you can select them all by holding down Control and hitting the 'A' key (on a PC) or holding down Command/Apple and hitting 'E' (on a Mac). You can then move these markers incrementally by using the right and left arrow keys. This can be especially handy if you're working with a long track that hasn't been correctly analysed by Ableton's new Auto-Warp feature.

Making Arrangements Essentially an Arrangement is created by recording the locations of the Clip triggers you create in Live's Session View into the timeline in the Arrangement View. To do this, you hit the Global Record button in the main transport bar at the top of the screen, and all your subsequent actions are then recorded into the arrange view. If you are using frozen tracks to conserve CPU power you can still do this. However, once you finish the pass, the track freeze will be lost, and will have to be repeated in the arrangement. The example screen above was done in one take in real time. As you can see, a fair degree of complexity is possible using just this simple setup. Once you're happy with the result, you can change to Arrange View and start finetuning your Arrangement. There might be some late or early triggers to correct, or you might want to add some additional Clips manually. This can be done either by dragging Clips from the Browser into the Arrangement, or by copying them from the Session View and pasting them into the Arrangement.

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Streamlining Your Workflow In Ableton Live

Now is the perfect time to add details such as fills or variations that weren't part of the Clips coming from your Session View performance. You can create new Tracks or drop in to existing ones — this is done by arming the track and then dropping into Global Record mode. Obviously Live allows you to cut and paste in the same way any other DAW does, so you might rework the arrangement by picking out the best bits from the performance. The art, as usual, lies in knowing when to stop. Having said that, the beauty of this way of working is that you can throw together very different arrangements in no time at all. Since I started using Live, I've found that I've changed my way of working from trying to get the perfect track straight away to creating many different ones and combining them into one. At this stage, I often find that my CPU is stretched to the limits, so I use the available tools to free up capacity. Since I don't want to restrict my mixing options too much, I tend not to 'Render to Disk' until the actual mixing is taking place, but a careful combination of consolidating and freezing tracks can unlock considerable processing reserves. This is a matter of personal preference, but I still like to keep the mixing as a separate stage of working, mainly because I work a lot in different places and I prefer not to mix until I'm in an environment where I'm comfortable with the monitoring. More on that subject in the near future, when I shall be looking at mixing in Live. Next month, I'll be approaching the session from another angle: recording an Arrangement and then deconstructing it into Clips for remixing and arranging. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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The Lost Art Of Sampling

In this article:

The Lost Art Of Sampling

A Little History: Samplers Or Part 5 Synths? Published in SOS December 2005 Hidden Synths Envelopes Print article : Close window Filters Technique : Sampling Amplifiers LFOs Putting It All Together Next Month

Nearly all modern samplers have powerful synth engines concealed inside them — and sometimes they're so well hidden that their users are unaware of their existence. But then why would you want a synth in your sampler? Let's find out... Steve Howell

Last month, we concluded by looking at the process of keygrouping and saw how easy it is to assign pitched instrument samples across a range of notes in a modern sampler, so that they can be correctly played by a MIDI controller sending the right trigger notes. Once you've got the hang of that, it's the same procedure, whether you have two samples or 20, whether you're mapping one sample across several notes or mapping one sample to each key. However, mapping drum and percussion samples is slightly different. As mentioned last month, you're unlikely to want to pitch, say, a cabasa sound over the whole range of a keyboard to the exclusion of anything else. It's much more likely that you'll want different drum and percussion sounds on each key, so that you can play in rhythm parts comprising multiple drum sounds in one pass. It's easy to achieve this kind of mapping, especially on modern samplers which allow you to drag and drop samples onto representations of a keyboard to assign them — you simply drag (say) your bass-drum sample onto C1, your snare onto D1, rimshot onto E1, floor tom onto F1, low tom onto G1, and so on. However, depending on how your drum map is set up, you may find that some of your drum sounds play back very slowly after you've assigned your sounds in this way, while others sound about right and still others seem ridiculously sped up. This file:///F|/SoS/SoS%2012-2005/lostscience.htm (1 of 9)11/23/2005 3:06:09 PM

The Lost Art Of Sampling

can happen because drum sounds are often nominally assigned a root-note 'pitch' of C3 (Middle 'C') when they are sampled. If you place, for example, a crash cymbal on C1 which has nominally been assigned a pitch of C3, and if your sampler still thinks it's dealing with pitched instrument samples, it will slow down the playback of the cymbal sample in an effort to create a low 'C' note from what it thinks is a sample played at middle 'C'. Result — a seriously slow-sounding cymbal! Fortunately, there's a way around this. Most samplers have a function called something like 'Keyboard Tracking' in their sample-mapping screens (as always, the exact name varies from sampler to sampler), and deactivating this in mappings designed to deal with multiple drum samples should fix the problem. Samples are then played back at the same speed irrespective of where they are placed on the keyboard, which is perfect for drum and percussion mapping. Some samplers even have dedicated drum-mapping options which don't even offer the option of pitching the samples across the keyboard. You are of course free to assign different drums to any key of your choice, but you may find it helps you to standardise, so that bass drums, for example, are always found on the same key (or set of keys), snares on another, and so on. This way, you're not always trying to remember what you've assigned where! Taking this idea a stage further, you can assign your drums according to the General MIDI drum map convention, which rigidly ties certain sounds to certain keys. This could be useful if you're in the habit of sharing sampled drum kits with other players. Having said that, you may prefer to 'roll your own' — not all drum kits fall comfortably within the GM specification, which requires six toms and seven cymbals, not to mention shedloads of esoteric percussion! Lastly, it might be worth your while to make yourself a 'template' Keymap for drums that you can use over and over again, especially if you tend to make up a lot of drum kits. On my Akai S5000, I simply load a drum Program without the samples and I can then replace those empty Keygroups with my new samples. If your sampler also allows this, it can save an enormous amount of time. My own drum-mapping template is shown at the top of the next page by way of example.

A Little History: Samplers Or Synths? Samplers didn't always contain comprehensive synth engines for sound processing, as they do today — but then when the Fairlight was launched in the late 1970s, there was no such thing as a sample-and-synthesis synth engine! The original Fairlight had no synthesis facilities to speak of for processing samples (although it was a fairly powerul digital synth in its own right) and it had no filters. The first company to truly blur the distinction between synth and sampler was Emu, who rapidly pulled ahead of the competition in terms of the sampleprocessing facilities their products offered. The first Emu sampler, 1980's Emulator, was fairly basic, but more adventurous players, such as Michael Boddicker (session keyboard player and programmer for Michael Jackson, amongst others) had their Emulators retrofitted with analogue filters so that file:///F|/SoS/SoS%2012-2005/lostscience.htm (2 of 9)11/23/2005 3:06:09 PM

The Lost Art Of Sampling

samples could be used like oscillator waveforms in a synth. Emu were quick to pick up on this, and their next model, the Emulator 2 (or EII) came with a fairly comprehensive array of synth functions built in, including filters, envelopes and LFOs. It wasn't sophisticated compared to today's offerings (and you certainly paid for it too... the EII cost around £5000 when it was released), but it was quite something in the early 1980s. Uniquely amongst samplers, the EII's synth-like features were all analogue, reflecting Emu's history up to that point — until 1980 they had been predominantly a manufacturer of modular and semi-modular analogue synths. By 1986, Roland had entered the sampler market, and their samplers rapidly began to offer a full range of synth-like editing facilities. After the release of their D50 sample-based synth workstation in 1987, the distinction between synth and sampler became very hazy — on the one hand, you had Roland's DRoland's S-series samplers (of which the series workstation synths, based S760 is shown above) featured a around samples held in ROM which comprehensive sample-processing engine could be put through a complete like that used in their sample-based synths. synth engine, and on the other, you Later, Emu added even more comprehensive had Roland's S-series samplers, filtering to their range of samplers from the which allowed you to put samples of Emulator IV (top) onwards, based on the Zplane filters introduced in their 1993 your choice held in RAM through a Morpheus synth. These days, though, such complete synth engine! The Roland extensive facilities are commonplace on S-series offered low-pass, bandsoftware samplers like Native Instruments' pass and high-pass filters. Roland Kontakt 2. weren't alone — at around the same time, other manufacturers, such as Ensoniq and Yamaha, were making similarly well-equipped samplers. Ensoniq, for example, included a flexible (though non-resonant) filter section in their Mirage (1985). Curiously, given their later dominance of the hardware sampler market, Akai's offerings during this period were crude in comparison. Their first sampler, 1985's S612, didn't have any filters, and their later S900 (1986) was also somewhat lacking in that department. Whilst the S900 did have filters, they were simply static affairs used as noise-reduction devices. Towards the end of the '80s, as sampling became widely affordable, Akai finally began to offer dynamic (sweepable) filters in their S1000 (1988), but these too were limited; they had no resonance and they were only controllable from an ADSR (you couldn't even assign LFOs to them). It wasn't until Akai released the S3000 in 1993 that we saw resonance on an Akai sampler's low-pass filter. But a significant addition on the S3000 was APM or 'Assignable Program Modulation' where any controller could be assigned to (almost) any destination, allowing LFOs and envelopes to sweep filters and modulate pan positions. LFOs could control other LFOs (as could envelopes)... and so on. This brought many concepts hitherto only seen on modular synthesizers to the digital sampler and so allowed you to gain a lot more mileage out of a simple sample than ever before. In fact, by just using the stock sine, square, pulse and sawtooth waveform samples that were permanantly held in in the S3000's memory, you could coax some decent analogue-style synth sounds out of an Akai S3000 without ever going near its sampling facilities! As explained in the first part of this series, by the early 1990s, most other players

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The Lost Art Of Sampling

had dropped out of the hardware sampler market, leaving Akai and Emu to battle it out with each other. By 1994 Emu brought their offerings up to snuff by incorporating the many different digital 'Z-plane' filter types (introduced in their 1993 Morpheus synth) into the Emulator series, starting with the Emulator IV and its many derivatives. Akai responded by introducing a second multi-mode filter board for the S3000 (and derivatives) and then their S5000 and S6000, released in 1999, offered true multi-mode filters. The arrival of software samplers has taken sample post-processing to another level, affording samplists access to even more filter types, multi-stage envelopes (with as many as 32 stages in some cases!), advanced LFO options, insert effects, global effects, and so on. These features are far in excess of those offered by most hardware synths, let alone hardware samplers. It's another case of technological developments abolishing the former distinctions between different types of equipment. Just as a modern telephone is no longer just a telephone, but can also increasingly be regarded as a personal music player, a personal organiser, or a camera, so it's pointless in many ways to talk of distinctions between software synths and samplers in an age when modern so-called software 'samplers' are, if anything, more synth-like than many hardware synths of former times!

Hidden Synths I've hinted several times during this series at the creative potential for sample manipulation that's offered as standard in all modern samplers, hardware or software. It's often forgotten that modern samplers are nearly always also powerful synths, perhaps because the earliest samplers didn't offer these kinds of facilities (see the box overleaf). Too often, the synth engine in samplers is neglected by users, or only made use of by factory patches or commercial sample libraries. This month, I hope to get you all using it more often. In the third instalment of this series, and last month, I gave a few examples of how you could save on sample memory by taking fewer (or shorter) instrument samples and making up the rest of the notes or note set by employing your sampler's onboard envelopes, modulation routings, and filters. It's now time to look at these features in more detail. When synthesizers were first invented, they existed in the form of a series of modules that were interconnected in different ways (usually by patch cords) to create different sounds — hence the term 'modular synth'. Typically, there were sound sources, sound processors and controllers. The sound sources were typically oscillators — devices that produce a periodic (and hence pitched) waveform (sound). The sound processors were typically filters (devices for removing or enhancing part of the sound source and hence

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A typical drum map, showing the different drum samples assigned to different MIDI notes across part of a keyboard.

The Lost Art Of Sampling

adjusting its basic tone) and amplifiers (devices that regulated or set the amplitude or level of either the oscillators and/or filter, or both). The controllers were just that: devices that could control the sound source(s), the filter(s) and/or the amplifier(s). Typically, these would be things like the keyboard (for controlling the pitch of the sound source), envelopes (for 'opening' and 'closing' the filter and amplifier in a repeatable, controllable way) and Low Frequency Oscillators (or LFOs — for altering pitch, tone, and/or amplitude in a repeatable way). A modern sampler is very similar in structure, although of course the 'modules' are now pieces of software in an operating system, and the patch cords are modulation routings. The only major difference is that in a sampler, there are no oscillators — instead, the sound sources are your samples, which can be of any sound you like, synthetic or otherwise. I would argue that a sampler is the ultimate synthesizer for this very reason, because instead of relying on a restricted set of predefined oscillator waveforms determined by the manufacturer, you can take and process any sound to use as the basis of your sonic creation! This would have been the stuff of dreams, fantasies and science fiction when the early sonic pioneers embarked upon their quest for new musical forms, sounds and textures. Today, of course, we all take it for granted! As we have seen, the synth facilities in modern samplers can be used to save sample RAM, and they can also be used creatively. At its simplest level, you might do this by mapping the velocity with which the sample is triggered to control the playback level or amplitude of the sample, rather than taking many RAM-hungry samples at different loudnesses. But as we saw in the third instalment of this series (see SOS October 2005, or www.soundonsound.com/ sos/oct05/articles/lostscience.htm) acoustic sounds can exhibit many other characteristics, and these can be recreated in a fairly natural-sounding way using your sampler's filters and envelopes, rather than by taking lots of memory-hungry samples. It's outside the scope of this series to explain the component parts of a synth in great detail — SOS has done it many times, perhaps most comprehensively in our long-running series Synth Secrets, which appeared for over five years (links to all the instalments can be found on-line at www.soundonsound.com/sos/ allsynthsecrets.htm). Nevertheless, it's worth briefly recapping a couple of bits of synth architecture that are particularly relevant to manipulating samples.

Envelopes Every sound has an 'envelope', or a way of representing what happens to it over time. A piano starts instantly and dies away slowly over time (as do many other acoustic instruments). A violin starts sounding slowly but sustains at a (relatively) constant level as the intrument is bowed, and then the sound dies away quite quickly after the bowing stops. A woodwind instrument will typically have an initial burst of sound, but will sustain at a relatively constant level for as long as the

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The Lost Art Of Sampling

player can blow and, once the player has run out of puff, will stop quite quickly. Brass instruments follow a similar pattern. These characteristics are known as the sound's 'envelope' and every sampler offers some form of envelope shaper to reproduce these characteristics electronically. Nearly all synths and samplers offer envelopes of the simple ADSR type (Attack, Decay, Sustain, Release), which was created as a simple way to mimic the many real-life sounds which start quietly and build to a crescendo (the Attack phase of the sound), then Decay to a constant level (the Sustain portion), before dying away gradually when they stop sounding (the Release phase). However, more modern synths and software samplers sometimes offer more complex enveloping facilities, usually based on multiple levels, with different rate parameters governing the speed at which the sound passes from level to level. Envelopes are extremely flexible tools for sound-shaping, as they can not only be applied to loudness or level, but to pretty much any synth engine parameter. For example, when applied to pitch, they can create regular, controllable rises and falls in pitch, and when applied to pan parameters, they can move sounds around the stereo spectrum. They can also cause the brightness of the sound to grow or diminish if they are applied to the synth engine's filters. Which brings me neatly to...

Filters As their name suggests, filters remove certain components in the source sound whilst allowing others to pass through unaffected. Low-pass filters remove highfrequency content whilst allowing lower-frequency components to pass through unaffected. High-pass filters do the opposite. Band-pass filters, logically enough, allow a band of frequencies to pass through unaffected whilst removing frequency components either side of that band, and the opposite of this is the band-reject filter, also known as a notch filter. Filters can also be severe or more forgiving in what they remove from a sound. This is determined by the steepness of the filter's 'roll-off', which is measured by the amount of sound attenuated by the filter over the course Different filter roll-off characteristics. of an octave's worth of change in frequencies. The steeper the roll-off, the more sound is attenuated in the course of one octave, and the more severe the effect of the filter. You often encounter six dB-per-octave, 12dB-per-octave, 18dB-per-octave, 24dB-per-octave, and even 36dB-per-octave types. For ease of writing, these types are often known as one-, two-, three-, four- and six-pole filters respectively. Many, if not most, samplers (hard or soft) now feature these different filter types as standard, allowing a great deal of tonal variation and sonic mangling to occur. file:///F|/SoS/SoS%2012-2005/lostscience.htm (6 of 9)11/23/2005 3:06:09 PM

The Lost Art Of Sampling

Most also offer resonance on their filters, and as on synths with these filters, these enable you to emphasise certain frequencies while filtering others, albeit often at the expense of making the sound somewhat synthetic after it has passed through the filter. When processing samples of 'real' instruments, the low-pass filter is usually the most useful, as it mimics what happens in nature. High-frequency content tends to decay as a sound gets quieter, and soft sounds tend to have less highfrequency content. Low-pass filters are therefore ideal to replicate the natural effects of decay over time, or of playing an instrument softly.

Amplifiers These are relatively innocuous little devices whose sole job is to regulate the level of the sound that plays back. They are usually associated with an envelope that allows you to control how the amplifier opens and closes over the course of a note and again, this can be very useful for artifically restoring the natural 'shape' of an instrument (especially if it has been looped, as discussed last month). However, by controlling an amplifier with trigger velocity, you can simulate the natural response of most instruments that are quieter when played softly, and louder when played hard.

LFOs LFO stands for Low-frequency Oscillator, and, as their name suggests, these generate low-frequency, periodic waveforms. These waveforms can then be used to control such parameters as pitch, tone, level and panning. Like envelopes, LFOs can usually be assigned to most synth parameters, and are very flexible tools for altering samples. When assigned to pitch, they cause repeated rises and falls in pitch, or vibrato, and when applied to level, they allow you to create tremolo effects. You can also use them to make sounds gradually brighter and duller by applying them to the filter. Many LFOs offer the facility to be synchronised to incoming MIDI clock, so that their effects run in time with the tempo of your track.

Putting It All Together You may well be thinking that you don't need these synth-like facilities, because you bought your sampler to play 'acoustic' sounds such as piano, strings, guitar, and so on. But think again. Even with such sounds, you can creatively exploit these parts of your sampler's sound engine to bring life to your samples. Here are a few examples...

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You sample a piano at full whack at every minor third across its range, and map the samples out accordingly. If you apply trigger velocity to amplitude, you afford yourself some degree of control over dynamics, but you will only be hearing a bright sample of a piano played at full tilt at different loudnesses. By backing off the lowpass filter cutoff a little and assigning trigger velocity to control tone as well, you can create softer-sounding versions when playing more softly — and all from the same sample. This trick could be applied to all manner of sounds, including guitar, basses, or drums. If a decaying sound you've sampled descends into noise, try applying an envelope (either to sample level and/or the filter) to remove this, setting decay and release times to match the sound's natural decay/release.

A traditional standard Attack/Decay/Sustain/ Release envelope (left), and (below) an example of a more modern multi-stage envelope based on rates and levels (here, four rates and four levels). The second of these is more useful, for example, when trying to emulate the complex envelope characteristics of instruments that cannot be replicated with the simple ADSR (like sforzando brass), but they can be trickier to set up.

If you're sampling a violin or string section, you could try recording them completely without vibrato and with the player(s) bowing at full tilt. You then have samples that can be used to generate string sounds with and without vibrato; you can add the vibrato yourself artificially, using an LFO assigned to modulate pitch. Purists will argue that this doesn't create quite the same effect as 'real' vibrato, but applied carefully it can be convincing enough. Recording without vibrato will also make the samples easier to loop. As I mentioned in the third part of this series, when discussing synth sampling, the trick here is to try to take samples of real instruments with as few 'performance characteristics' as possible. The effect of the latter can often be added to a 'vanilla' sample of the instrument later, by using the sampler's synth engine. Not only will sampling a violin with real vibrato be harder to loop, it will also require many more samples to be taken, because if you repitch a sample with vibrato, the vibrato rate will increase as you play further up the keyboard, and decrease as you play further down. If, on the other hand, you add the vibrato in your sampler, you can use the same basic sample for many pitches, and determine the vibrato rate yourself completely independently of the sample's pitch. Instead of 10 long multisamples, you might be able to get away with just four or five (or fewer) shorter samples, thus saving memory, enabling faster load/ save times, economising on storage and/or potentially reducing CPU strain — and that has to be worth thinking about.

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Next Month Next month, we will look at some of the more subtle details of your sampler's synth engine, and how you can make use of it to achieve subtler effects, such as minimising abrupt tonal changes between different Keygroups. Until then... Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Tuning Drum Loops In Reason

In this article:

Sonic Seasonings The Knobs You'll Be Tweaking Drum Major The Rest Of The Kit Reason News Automating Pitch Changes Going Loopy

Tuning Drum Loops In Reason Reason Notes & Techniques Published in SOS December 2005 Print article : Close window

Technique : Reason Notes

Can't quite get your Reason rhythm section kicking with the rest of the track? If you've never considered tuning your drum samples and loops to help create a tight and harmonious mix, now may be the time to try it... Derek Johnson

Since Reason's inception, there has been much discussion amongst users about whether tracks could be taken from start to finish, including final sweetening, without moving outside the software's environment. Obviously, adding new audio has always required external help. But although mastering processes could be replicated to a certain extent with a little ingenuity, it was the release of version 3's MClass mastering suite that made staying exclusively inside Reason a really workable idea. See Simon Price's 'Mastering Your Mixes' feature last month for more on this aspect of Reason.

Sonic Seasonings

Tuning drums can make a big difference to the sound of a mix, and can be automated throughout a song. In this example, you can see two Redrum samples having their pitch changed automatically; pitch stays at one value until bar 32, changes, then moves back to the original value after a further 16 bars. The changes have been exaggerated for illustration: most alterations wouldn't be this drastic. Note the green highlight around pitch parameters that have been automated.

But there may be some sessions where even the MClass sheen, on top of the usual refining, perfecting and polishing of samples, patches and

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arrangement, can leave some of us wanting more. But what? I'd like to suggest that it might be tuning. You don't need intimate knowledge of the inner workings of the Western music tradition in order to make music with software such as Reason. Many of us just stack bits together, trust to instinct and do our best. And it often works. But when figuring out the 'vertical harmony' of a piece — sorting out which lead or pad notes work with our funky bass line, which produce interesting dissonances and which just don't work at all — how many of us include drum sounds in the equation? And how many of us pay as much attention as we should to how textural material or breakbeats fit? We won't go into a discussion about those styles — usually urban and/or cutting edge — that mash beats, breaks and loops together, hope for the best and occasionally get it. We'll take it for granted that, no matter what source material you're using, you'd like the final result to have a sense of unity. If there's to be any distracting nastiness, let it be intentional! Before we move on, note that a lot of what I'll say should be taken loosely: though there are intrinsic pitches in all sampled sounds (you can play tunes with sampled rain, for example), we don't always need to identify that pitch absolutely. Our aim is to find tunings for what are, strictly speaking, non-pitched sounds, that are in harmony with the rest of a song. This can have quite a significant effect on the impact of your final mixes, and might well help you achieve your goal — a mix that feels subliminally 'right' — more quickly.

The Knobs You'll Be Tweaking Each drum-voice channel in Redrum has a 'pitch' parameter, with a range of an octave below to an octave above the central basic value. The scale, -64 to 0 (no The tuning controls of change) to +63, isn't Redrum and NN19. Shown concordant with absolute above are the pitchfine-tuning but is certainly envelope controls of close enough for our Redrum voices six and purpose. Channels 6 and 7 seven. NN19 has three have an adjustable 'bend' pitch controls, although the option. Originally intended cent-calibrated 'fine' knob for the creation of syn-drum will probably be used most swoops, talking-drum effects in this technique. and the like, one could use this to slide from one concordant pitch to the central pitch, if desired. Other devices that might be used to create drum sounds — samples could be loaded into NNXT or NN19, and syndrum sounds created with Subtractor, for example — have a three-way combination of octave and semitone transposition, plus accurate cent-based (1/100th of a semitone) fine-tuning.

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Tuning Drum Loops In Reason

Drum Major You may think of percussion as just producing some kind of thud, swish or crash, but listen closely and there's a dominant pitch in there somewhere. Load any drum sample from a factory Refill into NN19 and play it from your MIDI keyboard. You'll generate melodies, and it will always be possible to tune the sample to bring it into line with other musical material. This is our first goal: to bring the drum samples loaded into Redrum (or NNXT or NN19, if you use them for drum playback) into a harmonic groove with the rest of the track. If you've never used anything more than EQ, level and panning to fix up your drums, you may be in for a surprise. Remember that real drummers tune their kits to suit a given circumstance, especially in the studio. And as for the world of non-pop hand-played percussion instruments, pitch is part of their raison d'être and they will be generally be fine-tuned to suit their context. Let's start at the bottom. The kick drum will probably be the subject of the most valuable tuning decision you can make, so we'll start by getting it to 'sit' with a bass line — and not just rhythmically. Obviously, you won't be tuning a kick drum to match every note in a bass line, but it can help to centre the groove and the feel of the finished track if the kick is at least pitched to something like the root of a track's overall key. Solo your bass line and kick drum. Set up loop points for a section of song that has the bass line playing a lot of root pitches; if your track tends to groove along in E-flat (for example) for most of its verse or chorus, pick a bar or two where the bass line consists mostly of E-flat notes. Now tune the kick drum to this pitch. You don't have to tune it madly up or down to find an exact E-flat if the Manually recording a change in pitch of -15 to -12 for a Redrum drum voice resulted in result doesn't feel right, or if you find this messy transition at bar 32. yourself tuning too high or too low. The object of the exercise is to find a pitch for the kick that doesn't clash with the bass line. If you can't hear something that sounds right, borrow a trick from acoustic instruments: brass and string players often pull slides out or slacken strings and then slowly move them back to pitch. Move the tuning knob way up or down (the top, central position always corresponds to the root pitch of the sample) file:///F|/SoS/SoS%2012-2005/reasonnotes.htm (3 of 8)11/23/2005 3:06:13 PM

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and then slowly move it in the other direction, stopping when it feels right. Once you've found a tuning value that works, add the rest of the track and listen for any obvious clashes. Most kick sounds won't cause a problem with the higher-frequency elements of your mix, but it's worth checking. If your track changes its harmonic centre, re-tune the kick drum for each section, using Reason's automation facility. In all probability, the change won't be that much, since we're not necessarily transposing the kicks to every root pitch, remember. We're just looking for a value that fits. With the full mix going, a small change in tuning won't be noticed, save perhaps as a subliminal 'rightness' as the new section kicks in. If the only pitch that works 100 percent for a given section is very far removed from the section on either side, a compromise is in order — find a tuning that works for two sections, even if neither is ideal. If this seems like hard work and you've got RAM and CPU overhead to spare, there's another option. Finish your drum-kit creation and pattern programming, then make a copy of your Redrum, or as many copies as you have pitch changes to make. Tune each Redrum as required and then automate the Pattern Enable switches on the Redrums to start them playing when required. You'll need more Remix mixer channels, though, for the extra Redrums. Finally, if you have simple percussion needs, load multiple kicks into a patch, tuning each for the different sections and making the changes with changes of pattern.

The Rest Of The Kit Snare works well if it's tuned to the kick. Again, you'll be listening for feel, so you're not necessarily matching the kick's pitch exactly; a fourth or a fifth above might work. You'll hear a fourth if you play an 'F' above a 'C' on your master keyboard and a fifth if you play a 'G'; if you don't know what 'C', 'F' and 'G' are, you may need a different article! The snare tuning should be cross-referenced to the track, since you're trying to achieve harmony as part of the mix, not just the drum kit. Be prepared for more compromise as you find a pitch that works well against the kick and the mix. The same goes for hi-hats or cymbals: try to tune them within the kit first, and then 'massage' your result in the context of the overall mix. It can help to solo melodic and higher accompaniment textures during the hi-hat tuning process. Toms, if you use them, are another issue. Say you have high, medium and lowpitched toms: you might try to match the low tom to a pitch related to the kick, though something that sounds a fourth or fifth higher might work better. The mid, in an acoustic kit, would be tuned another fourth or fifth higher than that, and the high tom the same again. If there's a lot of tom work, you might want to run through any fills or other tom parts to make sure that this 'circle of fourths or fifths' rule of thumb doesn't create any major clashes.

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Tuning Drum Loops In Reason

Another subtle option is to tune the toms, especially if your kit has more than three, to significant pitches in the overall track's bass line or lead part. Modify your fills accordingly. You may not want to go the whole rototom or '70s disco syn-drum hog, but it's another thing to consider when you're trying to create an overall feel for your track. Claps can be treated like snares, while claves and rim-shots (which have more A moment with the line tool cleans it up nicely. of a definite pitch element) will be even easier to match to the whole track. Don't forget automation as a way to introduce dynamic pitch changes if the musical context of your mix would benefit. Samples of Latin or other hand percussion — congas, bongos, djembes — can be matched in a similar way to toms. It has occurred to me, before you say anything, that many instruments in the hand-percussion world can't be tuned, and that the same goes for hi hats and cymbals. I could answer that well-equipped percussionists have examples of different sizes that can be brought out for different occasions, but, actually, being able to tune what in the real world is essentially untuneable is one of the perks of working with electronic music.

Reason News Following on from last month's SOS review of Propellerhead's excellent Reason Drum Kits Refill is news of RDK 2.0. The price remains the same, £79, but the collection has been expanded and completely rebuilt to use Reason v3.0 features such as the Combinator and the MClass mastering tools. Hi-hat mapping has also been improved, but the biggest addition is the 13-strong collection of 'producer kits'. Developed in conjunction with a number of name producers and engineers, these kits use RDK's samples and Reason effects to reproduce each professional's 'signature sound'. If you're already a Reason Drum Kits user, all is not lost: you can buy an update disk for £20. It might seem like a gimmick, but how about buying yourself a branded Reason USB drive? A mere £27 is good value for the 256Mb capacity of this tiny device and the facility to easily move even large Reason files around. And if you need a

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Tuning Drum Loops In Reason

further incentive to make you take the plunge, there is one: a free Sonic Reality 64Mb Refill is loaded on each drive. Visit the shop at www.propellerheads.se. More than 40 MIDI hardware controllers are supported by Reason 3.0's new Remote Protocol. Trouble is, any new hardware that would benefit from the tight integration with Reason that the protocol offers would have to wait for an update to Reason before being supported. Until now, because with with the release of Propellerhead's Remote Protocol SDK, hardware developers will be able to create their own Remote codecs without waiting for a new version of Reason to be released.

Automating Pitch Changes Automated parameter changes in Reason can be recorded on the fly or drawn in manually (using the pencil tool) in the linear sequencer. For our purposes, a mixture of the two techniques will be the best option. And don't worry: it takes longer to describe this stuff in print than it does to actually do it! First, take a couple of runs through the song to work out which tuning values you require for which samples, and when they need to occur. The tool-tip parameter readout that pops up when you mouse or change a parameter is invaluable here. Changes that happen right on bar lines will make life easier, but don't worry if changes occur on any beats or sub-division thereof. We'll be using the 'snap' facility (enabled by clicking the magnet icon on the sequencer's tool bar) when drawing or editing the automation data. Set snap to the most appropriate value: don't select 16th note if all changes you plan to draw happen on a bar line. Now, Alt-mouse click a parameter to be automated — say 'pitch' on drum voice one of a Redrum. Doing this automatically switches the sequencer into Edit Mode and creates a controller lane for drum one's pitch parameter in Redrum's sequencer track. The lane is immediately available for editing. Disable any other edit lanes (using the icons in the tool bar) that might be cluttering up the screen. Also zoom in to maximum lane height (using the '+' magnifying glass to the right of the lane) since this will make it easier to manipulate controller data, whether drawing it from scratch or editing some that already exists. There's a quick way to register the initial pitch (or any other parameter) value. First, move the pitch knob to the value you require at the start of the song (remember those tool tips). Now fast-forward to a bar or two before the bar where you'd like the next pitch change to occur (if your first pitch-value change happens at bar 32, say, fast-forward to bar 30, or double-click the position display and key in '30'). We're doing this because Reason lacks a count-in facility. Enable the Redrum track for recording by clicking on the MIDI icon to the left. file:///F|/SoS/SoS%2012-2005/reasonnotes.htm (6 of 8)11/23/2005 3:06:13 PM

Tuning Drum Loops In Reason

Now click 'record' and 'play' and after the 'count in', and as quickly as you can, tweak the pitch knob with your mouse to the next desired value (watch the tool tip readout). Stay in record until the bar where the pitch should return to its original, or change to another, value; you don't have to hold the new parameter value with your mouse.

Here's the linear sequencer's tool bar: the main edit lanes are selected by the group of icons to the right of the edit-window selector button (far left); only the controller lane icon is highlighted (it's blue) since other lanes will make our working window too busy. Note that the line tool is highlighed, and that the snap facilitiy — see the highlighted magnet icon — is set to one-bar resolution.

Rewind to the start of the song and you'll see that initial pitch value recorded as controller data, but when you reach the point where you made the change, the controller data will look a little messy (see screens on previous page). Don't worry: this blip can be smoothed out with the pencil or line tool. The line tool is particularly useful here: place its crosshair at the leading edge of the controller data and drag in a straight line. With the snap value set appropriately, you just need to keep your mouse hand steady until the nearest snap value: the line goes jagged if you veer from the straight and narrow. Further pitch changes can be made in the same way: fast-forward to a bar or two before the point where you want the change, record it, stay in record until the parameter should change back, and edit any wobbly bits of controller data later. When smoothing out serious data blips, or drawing parameter changes from scratch, you'll discover that you won't know what value you're drawing. Reason's controller lane lacks tool tips and individual parameter step gradations on its Xaxis. After drawing a change, position your mouse over the corresponding knob to use its tool tip to confirm the value you've just drawn. If the value is right, carry on drawing a straight line. Again, the line tool is the perfect choice in these situations, and an appropriate snap value ensures clean starts and finishes to the controller-data changes.

Going Loopy You know where we're headed now: if you add extra sample material in REX or other formats to your track, you may also want to make sure its central pitch feel matches the track. Listen closely and you'll find that even the most abstract textural loop will have a pitch that you can tweak so that it will sit better in your mix. The same is certainly true of a lot of breakbeats or drum loops. Tuning within Dr: Rex is a doddle, since the sliced-up REX format means that there are no implications for length or temporal relationships once you've manipulated a loop or its individual slices. In general, follow the same methods discussed above, bearing in mind that finding a pitch that works may take closer listening. If the file:///F|/SoS/SoS%2012-2005/reasonnotes.htm (7 of 8)11/23/2005 3:06:13 PM

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loop is just drums, it may even be possible to rework the pitches of each slice in the same way as you would each drum sample in a Redrum kit. Doing so will be less straightforward than with other Reason devices, however. While it is possible to transpose each slice up or down over a huge range in semitone steps, the fine-tune parameter is global for the whole loop, so if you need to tweak the pitch of several slices more finely than a semitone you'll have to use automation to make the changes happen when needed. This isn't a huge problem in Reason, but it will inevitably be more fiddly than the section-based tuning of drums discussed earlier. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Using Digital Performer With External Hardware

In this article:

Monitoring Input Monitoring Mode Monitoring Using Hardware Hardware Synths & DP Real-world Monitoring Digital Performer News External Effects

Using Digital Performer With External Hardware Digital Performer Notes & Techniques Published in SOS December 2005 Print article : Close window

Technique : Digital Performer Notes

Few of us use our software sequencers in isolation — we all need associated hardware, such as monitors, external effects, and favourite MIDI synths. This month we take a look at using DP with such hardware. Robin Bigwood

While computer-based audio and music production is becoming increasingly 'virtual' in nature, there can't be many DP users who don't have some hardware, whether mixers, synths or effects units. The latest versions of DP have plenty of facilities for integrating with such hardware, and this month we look in depth at some of the most important of these.

Monitoring No matter how simple or complex your setup, establishing a flexible and reliable approach to monitoring during recording is absolutely crucial.

Good monitoring during recording is essential. In Digital Performer, flexible monitor mixes can be achieved in various ways. In the screen above, a monitor mix is being sent to musicians via the mixer's aux send section. The Masterverb plug-in is being used in this instance to give the vocalist some reverb in the headphones, but the effect will not be recorded.

With Digital Performer running under the MOTU Audio System (its 'native' audio mode), monitoring behaviour is controlled by the Audio Patch Thru feature. When this is turned on, by engaging the little 'headphones' button in the Audio Monitor window (or, in DP 4.6, in the Consolidated Window's title bar) DP routes the inputs of any record-enabled tracks to their outputs. So if, say, you have a mic patched into a mono audio track in DP and the track's output is set to use the file:///F|/SoS/SoS%2012-2005/performernotes.htm (1 of 8)11/23/2005 3:06:26 PM

Using Digital Performer With External Hardware

stereo output that drives your studio speakers or headphones, you'll hear the mic signal as long as the track is record-enabled and Audio Patch Thru is turned on. Turn off Audio Patch Thru, or take off record-enable status, and you'll no longer be able to hear your input. Audio Patch Thru is, then, a very straightforward system for letting DP handle your monitoring, but it can be configured in various ways to allow for different monitoring requirements. Go Setup menu / Configure Audio System / Input Monitoring Mode and you get a dialogue box that controls how Audio Patch Thru behaves. To begin with, we'll just consider the top two options: 'Direct hardware playthrough' and 'Monitor record-enabled tracks through effects'.

Clicking the 'headphones' icon, shown here nestling amongst the other buttons in the Consolidated Window's title bar, turns Audio Patch Thru on and off.

As with so many things in the computer-based studio, the shadow of latency looms over one of these two choices. If you select 'Direct hardware playthrough', DP will work Audio Patch Thru so that record-enabled track inputs are routed to outputs via the shortest possible route — often a brief round-trip from your audio interface, into the Mac and then straight back to your interface — with the result that monitoring is subject to little or no latency, and the musician hears what he or she is performing instantaneously. Select 'Monitor record-enabled tracks through effects', on the other hand, and the round trip from track input to output now not only goes via your Mac, but passes through the MOTU Audio System itself. This allows the possibility of adding MAS effects (such as reverb) in real time to the monitor signal, but also imposes a delay commensurate with the Buffer Size set in the Configure Hardware Driver dialogue box. How serious this is depends firstly on the Buffer Size itself, and secondly on what you're recording. Keyboard players and guitarists frequently seem oblivious to to the delay associated with a 512-sample Buffer Size, whereas singers and drummers may well still not be happy with the few milliseconds latency of a 128-sample setting. Best practice here would seem to be to use 'Direct hardware playthrough' when latency is unacceptable and 'Monitor record-enabled tracks through effects', coupled with the lowest Buffer Size you can get away with (remembering that low sizes lead to much greater processor usage), when you need to use DP's effects processing on your monitor signal.

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Input Monitoring Mode Vying for the award of 'most confusing feature in Digital Performer' the Input Monitoring Mode dialogue box offers various options that control some of the finer points of input-monitoring behaviour. The main point of this dialogue box is to control the way in which you hear any audio inputs on recordenabled tracks, and pre-existing soundbites in those tracks, during playback and recording. Additionally, if you're running a MOTU audio interface, two extra options appear, to control whether audio signals coming into DP are routed straight back out of your audio hardware, for minimal latency, or whether they pass through DP and MAS, picking up a little latency but allowing DP's effects to be used on them. To cut a long story a bit shorter, these various options don't always result in the behaviour that you'd The Input Monitoring Mode dialogue with expect, so rather than trying to four options is the one you see if you're using make some sweeping (but probably a separate audio interface. The simpler even more confusing) statements dialogue is the one that appears if you're about what's going on, I'm going to using the Mac's built-in hardware. give a run-down of what the four possible combinations mean. If you're not using MOTU hardware, by the way, you're effectively monitoring recordenabled tracks through effects, so it's the bottom two choices that apply to you: Direct hardware playthrough + Only during recording (and punched in)

Entirely contrary to what you'd reasonably expect, input monitoring is always active with this combination, when DP's transport is stopped, is recording (or, indeed, punched in or out) and during playback. Soundbites in tracks play back normally. * Direct hardware playthrough + Always Again, input monitoring is always active, but now existing audio in tracks can't be heard, regardless of whether you're in playback or recording. Monitor record-enabled tracks through effects + Only during recording (and punched in)

This makes more sense. Now you get to hear your input when DP is stopped, and also when it's recording (or when it punches in, during Auto Record). During playback, though, the input signal is muted and you hear the track instead. Monitor record-enabled tracks through effects + Always

Just as with its Direct hardware playthrough counterpart, now you always hear your input signal, but you never hear audio already in the track.

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What does this all mean? The first combination, as described above, may well be the best for recording drop-ins, as the musician can hear existing audio in the runup to the section they're about to record, as well as themselves playing. Switching to the Always option might be better if you're doing an entirely new take of a section, as you'll never hear existing audio. Of the two 'through effects' modes, the first is by far the most logical, and is probably the best choice for general recording. However, the Always option can work well as long as you're good at remembering to disengage record-enable after your takes! I'm probably not alone in thinking that a request for rationalisation of all this could be a good thing to get off to Click here to email.

Monitoring Using Hardware Of course, there are other solutions. One way is never to use Audio Patch Thru at all, but to set up all your monitoring on an external analogue or digital mixer. It's then easy to route signals to DP for recording (perhaps via the mixer's group outputs) while providing a separate monitor mix to the musician, along with any reverb or other effects, courtesy of an external effects unit. Then there are audio interfaces, like many of MOTU's own models, that have zero-latency monitoring abilities. If you have one of these, you can turn off Audio Patch Thru and set up all the monitor routing on your interface. This is frequently done by using a separate, dedicated application, such as the Firewire Cuemix Console that controls MOTU's Cuemix Plusequipped interfaces. Some interfaces may even have some built-in DSP, for providing a little reverb (for example) on monitor signals, but if yours doesn't, you can still provide some reverb without using any additional hardware. The key to this is combining the interface's zerolatency monitoring with DP's 'Monitor record enabled tracks through effects' Audio Patch Thru mode, and it works very well. 1. First, set up the (zero-latency) monitoring you require on your interface. 2. Ensure that DP's Input Monitoring Mode is set to 'Monitor record enabled tracks through effects', before configuring and record-enabling the track you want to record to.

By selecting hardware outputs in the Mixing Board's sends it's possible to mimic hardware mixer practice and set up multiple independent monitor mixes. Here, a vocalist and guitarist are being provided with different monitor mixes courtesy of two output pairs of a multi-channel audio interface.

3. In DP's Mixing Board, place a reverb or other suitable effect in one of your record-enabled track's insert slots and set its wet/dry mix to 100 percent wet. You file:///F|/SoS/SoS%2012-2005/performernotes.htm (4 of 8)11/23/2005 3:06:26 PM

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only want it to supply the effected sound, since all the dry monitor signal is handled by your interface. 4. You can now control the level of the effect by adjusting the track's level fader. Remember that the insert slots on an audio track are 'post-record', so you won't be recording this reverb. Also, the track fader position has no effect on the level being recorded to disk, so you can adjust this with no worries. You can now combine the benefits of zero-latency monitoring, using your interface, with DP's real-time effects-processing capabilities. The reverb signal coming from DP is, of course, beset with latency, but since it's in addition to your latency-free signal it's virtually unnoticeable and just comes over as a reverb with a touch of pre-delay.

Hardware Synths & DP DP uses the setup you've defined in OS X's Audio MIDI Setup application to determine which MIDI devices it can use, so in order for it to communicate correctly with any hardware synth or other devices in your studio, it's essential to get things properly configured here. Actually, compared to OS9 FreeMIDI and OMS systems, there's not that much tweaking you can do in Audio MIDI Setup, besides selecting a Manufacturer and Model for your MIDI devices, choosing the channels they transmit and receive Cherrypicker is the application for you if you on, then drawing in the symbolic ever want to modify OS X/DP patchlists, or connections between them and your get down and dirty with your synth's MIDI MIDI interface. All the 'under the implementation. bonnet' MIDI settings, as well as patch list information, comes from the 'middev' and 'midnam' files in [hard drive]/Library/Audio/MIDI Devices/MOTU that you never need to deal with directly. Audio MIDI Setup's simplicity is all very well as long as you're using mainstream synths and factory presets, but what if you have a more unusual, unsupported synth, or have spent time perfecting your own banks of patches, or have even loaded in banks from elsewhere or from expansion cards? Well, if you want custom patch lists, or need to send precise bank-change and program-change messages to an unusual synth, check out Cherrypicker. This is an all-singing, alldancing application for creating, converting, adapting and saving patch lists and other MIDI-related information in OS X. It can open patch lists in a variety of different formats (even those from OS 9 days, and from the PC platform) and also translate between the slightly varying 'midnam' patch list formats used by DP, Pro Tools and Cubase. It can also edit and create 'middev' files, which are the files DP uses to determine the basic capabilities and compatibility of individual MIDI file:///F|/SoS/SoS%2012-2005/performernotes.htm (5 of 8)11/23/2005 3:06:26 PM

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devices. If you really want to take control of your MIDI setup, or solve an annoying bank-select problem (for example), the donation-ware Cherrypicker is a must. It's downloadable from www.alterspective.com/cherrypicker and has superb documentation.

Real-world Monitoring No matter which technique you employ, it's worth considering what the requirements of a good monitor feed are. First, you need to be able to supply a wide overall range of monitoring signal level. Drummers often require an extremely loud click track, for example. You should also be able to simply adjust the mix of backing track (or click track) and live signal in the musicians' headphones. Finally, some means of setting up multiple individual monitor mixes is a real advantage, especially if you regularly work with more than one musician at a time. It's these last two requirements that can be challenging when you're not incorporating an external hardware mixer or sophisticated audio interface into your recording signal chain (although they can remain challenging even if you are!). Here are a few things to think about that might help in some situations: Creating a Master Fader Track in your project means that you instantly get an overall volume control for your existing 'temp' tracks, which is independent of any live monitoring you've got set up on an external hardware mixer, an audio interface, or using DP's 'Direct hardware playthrough' mode. You can use it to balance the level of your backing track (or DP's click, if it's routed to the same output pair) relative to your live monitor signal. If you use 'Monitor record-enabled tracks through effects' mode, Master Faders are less useful for controlling the relative levels of backing track and live signal, since they control both simultaneously. In these cases, you could consider routing all your backing tracks through an Aux track, which restores independent level control. If that doesn't appeal, set up a Track Group that links all the faders of the tracks used in your backing track. Dragging one then moves the whole lot. To set up several simultaneous monitor mixes, you really need either a hardware mixer with multiple auxes (or even groups) that supply separate headphone feeds, or a multi-channel audio interface with sophisticated zerolatency bussing abilities. To go a stage further and provide independent mixes of your backing track (ie. plenty of drums and bass for a guitarist, but more piano for a vocalist, with both takes happening at the same time), you really need to emulate hardware mixer practice and set up monitor mixes on auxes in DP's Mixing Board. You might choose one of your interface's output pairs in all the topmost aux send 'slots', and a different pair in a lower slot. This way, you can have different mixes going to each musician, and a separate one (on the main Mixing Board faders) for yourself.

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Digital Performer News In early October, Universal Audio released version 4.0.0 of the plugins for their UAD1 Powered Plug-in audio card. The update fixes a couple of key Digital Performer issues associated with mono in/ stereo out usage and crashes caused by deleting and reinstantiating automated UAD1 plugins. There's also a tasty new Roland Dimension D chorus emulation. The 35MB download is at www.uaudio. com and is free for UAD1 owners.

PSP Audioware's new offering continues the PSP tradition of plug-ins that are immensely powerful and sound beautiful, but are surprisingly easy to get to grips with.

MOTU recently announced an update to their Symphonic Instrument orchestral sound library plug-in. Possibly available by now, version 1.1 adds disk streaming and multiple outputs to the existing feature set of the plug-in. Both are welcome developments, with disk streaming significantly reducing the amount of RAM a multi-part Symphonic Instrument setup uses, and multiple outputs allowing submixing of groups of parts, as well as the flexible use of third-party reverbs. The 1.1 update will be free for registered Symphonic Instrument users. A couple of interesting new DP-compatible Audio Unit plug-ins have appeared recently, and are well worth checking out. PSP Audioware have come up trumps again with the PSP608 Multidelay, a multi-channel delay with up to eight seconds of delay time and extremely flexible filtering, modulation, saturation and reverb facilities. As with the long-standing PSP84 and Nitro plug-ins, the sounds that emanate from this thing have to be heard to be believed! PSP608 costs a very reasonable $149, but is cheaper still if you own some of PSP's other plug-ins and take advantage of their special launch offers at www.pspaudioware.com. Cheaper again (actually, it's free) is Solid State Logic's LMC1 compressor. It's an emulation of the Listen Mic Compressor built into 1980s SSL E-series desks, which was originally designed to aid communication between studio and control room but, almost by accident, ended up being the foundation of Phil Collins' drum sound. Grab it from www.solid-state-logic.com.

External Effects If you've a really great, characterful or much-loved hardware effects unit — a reverb or compressor, for example — you might consider patching it into your DP setup in a way that allows you to use it within DP just as you would a plug-in. Doing this is fairly easy if you have a multi-channel audio interface. If you're connecting up a reverb or delay, or any other effect that you might want to access from several tracks, you can simply patch its input(s) to one (or a pair) of

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your interface's outputs. Sending signals to it is then as easy as choosing the same output(s) in one of DP's sends and turning up the send level. If you have your effects unit's outputs patched into a mixer, or into the inputs of a zerolatency audio interface, that might be the end of the story. However, if you want to bring your effect signal back into DP you're immediately facing the latency demon again. If all you want to do is record the effect signal, you're fine, as DP automatically compensates for the delay that the MAS buffer imposes. But if you want to incorporate your effect-unit signal into your DP mix by bringing it in on an Aux track, it will always be heard late, just as if it were a live input being routed through MAS effects. As yet, there's no automatic compensation for this, as there is in the latest versions of Pro Tools, and the only solution you have is to submix every track in your sequence except the hardware effect return via an Aux track equipped with either a simple 100-percent wet delay plug-in, set to match the amount of latency you're suffering, or an instance of MOTU's Buffy plug-in. Both can help the audio to line up better, but neither is an easy or elegant solution. Expect an automatic scheme in a future update of DP, however. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Using MIDI Functions In Sonar 5

In this article:

In-track MIDI MIDI Scale & Zoom In-line Piano Roll View MIDI Step Recording Sonar News: Cakewalk Release Mac-compatible Soft Synth MIDI Effects Make-over New View For Piano Roll More 64-bit Confusion The Piano Roll View & Multiple Tracks Long Live MIDI

Using MIDI Functions In Sonar 5 Sonar Notes & Techniques Published in SOS December 2005 Print article : Close window

Technique : Sonar Notes

Cakewalk have strengthened the MIDI side of Sonar 5 considerably, in recognition of the rise of software synths that benefit from enhanced MIDI controllability. We run through some of the new features and suggest how you might want to use them, as well as rounding up the usual Sonar news and background info. Craig Anderton

As announced last issue, Sonar has hit version 5.0, and while its new synths, REX file support and other topics with a bit more sex appeal might overshadow its MIDI side somewhat, the MIDI facilities of the program have also been overhauled. From layout to effects, there have been substantial changes — so let's talk about how to make the most of these.

In-track MIDI The most fundamental change is integration of MIDI data into track views and the merging of more information into the standard MIDI Piano Roll view. But another pivotal change is that you no longer need to open up a separate window to do effective MIDI editing: it can be done within a MIDI clip in the Track view. So what makes seeing the MIDI data in a track better than seeing it in the usual Piano Roll window? It's all about context — you can see MIDI data in context with other MIDI and audio tracks. This is useful when, for example, you want to file:///F|/SoS/SoS%2012-2005/sonarnotes.htm (1 of 9)11/23/2005 3:06:38 PM

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check whether a MIDI bass note lines up precisely with an audio kick drum hit or is off by a few milliseconds. You no longer need to look carefully to see where on the timeline the note hits, then compare it to where on the timeline the kick hits: just move the tracks next to each other and compare directly. This takes on additional importance because, as we'll see later, controller data can now be shown superimposed on note data, rather than being in a separate, isolated strip. It's also possible to edit controller data in context with other tracks.

MIDI Scale & Zoom The first key to working with Sonar 5's MIDI features is understanding the MIDI scale/zoom function. Load up a sequence with some MIDI data and, in the Track view, expand one of the MIDI tracks to a reasonable height. Note that (similarly to how the audio tracks have a scale for audio level values) the MIDI tracks now show either a keyboard or note numbers that relate to the MIDI data in the track. Here's how to use this new feature. 1. Right-click on the scale and choose either Notes (shows a keyboard with musical note values such as C3, C4, and so on) or 7Bit Values (shows the MIDI note number instead). Note that the option you choose affects all MIDI tracks, not just the one on which you clicked. If the track is zoomed so far out that it's not possible to draw the keyboard graphic, note numbers will be The new scale and zoom control gives a much better view of what's happening within shown even if the Notes option is a MIDI track when you're working in the selected. Zooming in will affect the Track view. note-number display (ie. the notes move further apart) until there's enough space to show the keyboard graphic, at which point it's displayed. 2. To zoom (this is done on individual tracks and is not a global control), click on the MIDI track's scale. Drag up to zoom in or down to zoom out. Note that when 7Bit Values is selected, the notes in the track still zoom, but the scale doesn't change. 3. To 'scroll' up and down to see a higher or lower range of the keyboard, rightclick on the scale and drag up or down. (Incidentally, the standard Piano Roll view doesn't use the same scale/zoom protocol, but retains the method used in Sonar 4: (+) and (-) scroll buttons in the window's lower-right corner. When you're using the scale and zoom options, a particularly handy function in the scale's right-click menu is 'Fit Content' (you can see the option in the screen

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Using MIDI Functions In Sonar 5

shot above). This automatically scales the track so that all data fits within the selected track height.

In-line Piano Roll View In-line Piano Roll View (PRV) takes track integration one step further, as it allows you not only to view the data in context, but edit it. To do this: 1. Click on a Track's PRV button. Also note that there's a separate toolbar for PRV mode, which you can dock or float over the clip (see the screen on the right). Clicking on the PRV button allows you to see 2. The second drop-down menu from the top (just above the volume and pan a MIDI clip in a way very similar to the standard Piano Roll view, except that it's in controls) determines what type of data context with the other tracks. Also note the you'll be entering or editing. You can controller data in the background and the inchoose from Notes/Velocity or one of a line Piano Roll toolbar floating over the clip. list of controllers. To add a new controller, select 'New Value Type' and choose the required type of controller, value and channel.

3. If you want to add notes with the Pencil tool, use the upper-left drop-down menu to select the required note value. 4. The remaining drop-down menu is a 'view filter' that allows showing/hiding of velocity, various controllers, clip outlines and notes. A track with a lot of controllers can get rather confusing, so this makes it easier to focus on a limited number of items for editing.

Another useful feature of PRV is that you can shift-click on one of the scale's keyboard notes to audition it, or shift-click-drag and run your mouse over the keys to hear all the notes as you mouse over them. The PRV toolbar is extremely useful. I dock it along the lower edge of Sonar's main window, along with the other toolbars I use the most. The toolbar consists of buttons for the Select, Draw and Auto Erase tools, along with buttons that show/hide notes, controller data and velocity tails. Two additional buttons select the Fit Content option and turn PRV mode on and off. There is one limitation of working in PRV as opposed to the separate Piano Roll view: scrubbing of MIDI notes is possible only in the standard Piano Roll. In the

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Using MIDI Functions In Sonar 5

Clips pane, the scrub control affects only audio tracks.

MIDI Step Recording 'Real' musicians have always tended to look down on step recording as a crutch for those who aren't capable of playing a part in real time. But step recording has a variety of other uses, such as fast transcribing of sheet music, creation of arpeggios and entering of complex chords. Sonar 5 has made step recording far easier and more efficient. For my way of working, the biggest improvement is the handling of keyboard-shortcut bindings. The bindings are optimised so that you can play the notes you want to step with your left hand and use your right hand with the key This screen shows the Advanced Step bindings to edit the step-recording Recording dialogue, along with a series of characteristics (note lengths, add step notes that were step-recorded. Note how the sizes together, step forward, step spaces between notes correspond to the fullbackward — which you can now do stop symbols inserted between the steps in the pattern section. repeatedly — move one beat forward, and so on). The results of step recording show up immediately in a clip (even if you've enabled PRV mode in a track) and you can use other commands while step recording, including changing tracks. You can edit the key bindings if desired, but I found that the default ones worked just fine. The step-recording dialogue box now has two settings, Basic and Advanced. Advanced mode has several additional features, including the ability to randomise note durations. There are also some more advanced navigation options, such as the ability to link the point where data is inserted to the Now time. Linking to the Now time is a much faster way to jump to particular sections of a tune, compared with using the advance and back options. It's also possible to have Sonar 'learn' a pattern and step accordingly. In other words, if you keep having to enter rests in particular parts of a pattern, Sonar can automate this process. Here's how: 1. Choose the Advanced step-recording dialogue. (This technique isn't possible with the basic dialogue.) 2. Select the desired step size. In the screen shot overleaf, the step size is set to 8th notes. 3. In the Pattern field, enter which steps should have notes (indicated by a file:///F|/SoS/SoS%2012-2005/sonarnotes.htm (4 of 9)11/23/2005 3:06:38 PM

Using MIDI Functions In Sonar 5

number) and which should have a rest (indicated by a '.' — or you can also type 'r'). For example, in the one-measure pattern shown in the screen shot above, there are two eighth notes, followed by an eighth note rest, followed by another eighth note, followed by two eighth note rests, followed by two eighth notes. 4. Start playing. Rests will be inserted automatically between the notes you play, according to the pattern you entered. There are other nice benefits associated with this feature. For example, every pattern you use in a song can be accessed via the pattern drop-down menu. So if, for example, you use one bass timing pattern in a verse and another in a chorus, you can alternate between them when step recording.

Sonar News: Cakewalk Release Mac-compatible Soft Synth Cakewalk's big announcement at the recent Audio Engineering Society (AES) show in New York was the availability of the Dimension Pro software synthesizer (the original version first appeared in Project 5 V2) for PC and — yes — Macintosh. The instrument, which lists for $359, supports Audio Units, DXi and VSTi formats and comes with two DVDs of content. If you look at Project 5 V2 The Dimension Pro is a fine synth in its own comments on Internet forums, the right, but what raised eyebrows at AES was prevailing opinion seems to be that its Mac compatibility. it's worth upgrading Project 5 just to get this synth. Indeed, it's very powerful, not only featuring a ton of presets (1500, to be exact, spread over the two DVDs) for plug-and-play types, but also offering sufficient depth for hardcore tweakers. It employs samples, physical modelling components and 'analogue' sound generation (based on waveguide generators and wavetable oscillators), all of which can work together in various ways, and you can mix and layer up to four stereo parts per program. There's also a Vector Mixer for real-time sound morphing. As for pedigree, the design comes from Rene Ceballos, who's known for the z3ta +, Triangle DXi synth, and Pentagon virtual analogue synth (which is now included with Sonar 5). As Cakewalk now own rgc:audio (Rene's company), I suspect that we'll see other cross-platform instruments appearing before too long. What about a Mac version of Sonar? Cakewalk still don't seem sold on the concept, but I can't help wondering if Apple adopting Intel chips might throw some weight in that direction.

MIDI Effects Make-over

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Using MIDI Functions In Sonar 5

Functionally, Sonar 5's MIDI effects are pretty much the same as those in previous versions of Sonar, although there have been enhancements here and there. However, the look has been very much improved, so you needn't hide anything if you open one up while an important client is looking at the monitor! The new look is reminiscent of Project 5, with more readable parameters. My only disappointment is that the very cool Session Drummer plug-in looks as uncool and user-unfriendly as ever. One warning: If you install Project 5 after Sonar 5 has been installed, don't install the Project 5 MIDI effects. They'll overwrite the Sonar 5 ones with older versions. If you do this by accident, you can always re-install Sonar 5 but tick only the MIDI effects for installation.

New View For Piano Roll Even the Piano Roll view has been updated. The biggest change is the ability to either have controllers and velocity integrated in the same pane as the notes, or segregated in a separate controller pane toward the lower part of the window, as in previous versions of Sonar. You'll also find two drop-down menus like the ones used in the in-track Piano Roll View. One chooses the data that will be displayed in the window (velocity, controllers, notes, clip outlines and so on), while the other facilitates the entering and editing of velocity and controller data. When you're displaying multiple controllers, the window can get extremely You can now adjust velocity for notes in a cluttered. So when you choose one of similar way to how it's done in Project 5, the controllers, it is automatically sent which is more straightforward than in to the 'front', while the other controllers previous versions of Sonar. are shaded lighter and put in the background. This makes it easy to adjust one controller while still seeing it in the right context with other controllers. However, even then you can have too many controllers for comfortable viewing — which is why being able to hide particular controllers is extremely helpful. One of the most useful additions to the Piano Roll view is easy velocity adjustment, as was first introduced in Project 5. Here's the process: 1. Select the Draw tool (the one that looks like a pencil). 2. Mouse over the top edge of the note until seven parallel lines appear to the right of the pencil (see the screen below). 3. Click. The note and its associated velocity stem become brighter, and you'll

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see a readout that shows the note name, note number, velocity and duration. 4. Drag up to increase velocity or down to decrease velocity. The big advantage of this approach is immediately obvious when you want to adjust velocity on individual notes within a chord: you can easily distinguish each note's velocity. However, if you want to change velocities for a lot of notes simultaneously it's still easier to use the controller pane, as you can draw a line over the velocities and have their values conform to the line you drew.

More 64-bit Confusion One of the recurring questions in forums is whether you need to use a 64-bit computer to take advantage of Sonar 5's 64-bit audio-processing engine. We touched on this briefly in last issue's Sonar Notes, but it's worth reiterating that even a 32-bit computer takes full advantage of the 64-bit double-precision audio path. In other words, audio quality will be the same whether you run Sonar on a 32-bit or 64-bit machine. It's not dependent on the operating system either — you'll hear the same results with Windows XP or x64. The difference between 32- and 64-bit computers lies in operating speed, and the ability to address huge amounts of RAM. A 64-bit machine with a 64-bit operating system will be somewhat faster and will be able to address more RAM than a 32bit machine. Also note that you can use 32-bit VST plug-ins in a totally 64-bit environment (processor and OS). Some people have a hard time with this concept because it's counter-intuitive: you'd expect to need plug-ins designed to work with 64-bit systems. But Cakewalk's BitBridge technology lets 32-bit plug-ins 'look like' 64-bit plug-ins, in a similar way to how their SurroundBridge in v4 allowed stereo plugins to function in a surround environment. All clear now?

The Piano Roll View & Multiple Tracks One of the advantages of working with the standard Piano Roll view instead of the in-line version is that you can see multiple MIDI tracks simultaneously and choose a specific one on which you want to work. You might want to see multiple tracks at the same time in the case of keyboard parts where the right-hand and left-hand parts are in separate tracks, drum tracks where different drums and percussion are on separate tracks, and doubled parts where you want some variations. To work with multiple tracks: 1. Select the tracks you want to see in the Piano Roll Track pane. There are two ways to do this. The first one is to select the tracks, either in the Track or Console view, then go View / Piano Roll (or type Alt+5). The selected tracks will be visible in the Piano Roll view. Alternatively, with the Piano Roll view open, type T (for 'Pick Tracks') or click on the Pick Tracks button (the double arrows). In the Pick Tracks dialogue that appears, Control-Click on the tracks you want to file:///F|/SoS/SoS%2012-2005/sonarnotes.htm (7 of 9)11/23/2005 3:06:38 PM

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see, or Shift-click to select multiple contiguous tracks. 2. If you don't see a list of the tracks toward the right of the Piano Roll window, make sure the Show/Hide Track Pane button is on. This is the button to the immediate right of the Pick Tracks button.

With the Track pane showing, you see a list of all tracks whose data is visible in the Piano Roll view. Within the Track pane you can also enable and disable editing for individual tracks.

3. To highlight a track so that it becomes the 'active' track (as indicated by its notes being brighter than the other tracks) and therefore can be edited, you have four options:click on the track name in the Track pane; click on a note that's in the track; select the track in the Track view; or select the track in the Console view. 4. You can now edit the active track at will — enter notes, erase, and so on. Note that each track in the Track Pane has Mute, Solo and Record buttons. Sometimes it's handy to be able to access these functions from within the Piano Roll view rather than having to jump over to the Track or Console view. Another trick is that you can switch all tracks in the Track pane to the next higher or lower number using the drop-down menu to the immediate right of the Pick Tracks button. For example, if the Track pane shows tracks five, eight, nine and 13, selecting Show Previous Track(s) will cause the Track pane to show tracks four, seven, eight and 12, while selecting Show Next Track(s) will cause the Track pane to show tracks six, nine, 10 and 14. Finally, you can Show/Hide track data by clicking on the track's colour button (the leftmost square associated with a track). If the colour is visible, you'll see the notes; click on it to turn it white, in which case the notes are hidden. Note that if some tracks are hidden and some are visible, you can use the invert button (or type 'V') to reverse these. For example, if tracks two and six are hidden while three and 10 are visible, typing 'V' causes tracks two and six to become visible, while tracks three and 10 become hidden.

Long Live MIDI With the surge in virtual instruments driven by MIDI data, MIDI has once again come to the forefront of the sequencing process. By revamping the way Sonar handles, displays, and edits MIDI, Cakewalk have shown that they recognise this trend and have made it a lot easier to work with virtual synths — or any MIDIrelated sequencing. Published in SOS December 2005

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

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

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Working With Video In Cubase SX & SL

In this article:

Working With Video In Cubase SX & SL

Project Time Hit Me Baby, One More Time Cubase Notes & Techniques Published in SOS December 2005 Slave To The Cubase Doing The Time Warp Print article : Close window Cubase News Technique : Cubase Notes Process Tempo

This month we take a look at building tempo maps for writing to picture in Cubase, using Markers, Time Warping and the Process Tempo command. Mark Wherry

Last month we started to investigate working with video in Cubase, looking specifically at the pros and cons of having your video running inside or outside of Cubase, and how to use the built-in video playback features. In this month's article we're going to take a look at the process of actually building tempo maps in Cubase, and you might want to follow the examples discussed with an empty Project on your own system. You won't need any additional video files or musical content for these examples. If you're interested in running Cubase with an external video machine, check out the 'Slave To The Cubase' box for more information.

Project Time

Cubase SX includes many powerful commands for building tempo maps when writing to picture. Here you can see the Time Warp tool being used to create a tempo map based on hitpoints in the video that have been identified by Markers on a Marker track. Notice how the Ruler on the Project window turns dark red when the Time Warp tool is active, and how the tempo changes are illustrated with triangles. The Tempo Editor shows a more graphical overview of the Tempo Events.

One thing to bear in mind if you're running video in Cubase is that the timecode of the video always needs to be locked to the timecode of the Project. Last month we looked at how to set the Project start time (in the Project Setup window) and aligning the Video Event to be in sync with the Project; but quite often this will still need further adjusting, which is to say you probably don't always want the

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beginning of the video to be the beginning of the Project. For example, say the 24-frame video starts at 01:00:00:00, your music starts at 01:00:05:00, and you want the first bar of music to be bar three at 120bpm. It's a good idea to leave a couple of empty bars at the start of your Project in case you need to have an upbeat or make changes later on. Keeping the video where it is, you could achieve this by setting the tempo to 96bpm at the beginning of the Project (bar one) and putting a tempo change to 120bpm at bar three, but this is an awkward solution. It would make it harder to use up-beats, and if you later got real musicians to play your music against a click track, you'd have to manually cut in a click at the right tempo (120bpm) so they had a proper count-in. To help in these situations, Cubase has a neat command called Set Timecode at Cursor (in the Project menu), which does exactly what it Creating a Marker track is a helpful way to says. To set bar three to be identify points in the video that you want to 01:00:05:00 you would set your Project align to musically relevant locations. You can Cursor to bar three, select Set then use the Time Warp tool, as illustrated here, to create appropriate tempo changes Timecode at Cursor, enter the when dragging bars or beats onto Markers. timecode and click OK. Cubase will automatically work out the offset that is required to make the timecode you entered hit the cursor's bar and beat position, so in this case, bar three will now be 01:00:05:00, and you don't have to worry about tempo changes. If you already have content in your Project, before the Set Timecode operation is complete Cubase will prompt you: 'You have modified the timecode offset. Do you want [to] keep the Project content at its timecode positions?' Aside from the missing word, this seemingly confusing question is actually fairly straightforward. Most of the time you'll want to say 'No' to this question: if you have some MIDI and audio Parts on the Project window already at bar three, you probably want to keep them at bar three. Clicking 'Yes' would move your existing objects so their timecode position was preserved, which would place them at musically irrelevant locations. If you're using Cubase with an external video player, you have nothing else to worry about. However, if you're using the built-in video player, you'll need to make sure you readjust the position of the Video Event so that it's in sync with the Project's timecode again. In theory, if the Video Event is the only Event in the Project (and it's already correctly lined up before the timecode adjustment), you could answer 'Yes' to the question. However, if the new timecode is later than the old time, this moves the start point of the Event behind the start of the Project, and Cubase's video player doesn't always seem happy about this. So the best option is probably to readjust the start of the Video Event manually, cropping either the start or the end positions as you would for any other type of Event on the Project window.

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Hit Me Baby, One More Time Once you've got the initial tempo set and the Project start time sorted out, a common task is to identify various hitpoints in the video that you want to tie in with something musically meaningful. One useful technique is to create a Marker track in your Project and add Markers to represent the hitpoints in the video. To create a Marker track, select Project / Add Track / Marker, and you might like to enable Cubase's Divide Track List function by clicking the appropriate button (which looks like an empty rectangular box) at the very top of the Track List, just where the Track List and the Ruler intersect. In this mode, the Event Display and Track List are split into two areas, and by default the Marker track is automatically moved into the upper area. The two areas of the Track List can be vertically scrolled independently of each other (and you can drag the dividing line between the two areas to resize them as you wish) and this makes it possible to keep the Marker track at the top of the Project window, no matter what tracks are visible in the lower part of the Track List. Let's say you have a hitpoint at 01:00:30:18. To create a Marker at this position, first set the Project Cursor to this timecode location. As mentioned last month, when you're working with When you change the start time of the picture you'll probably want to have Project, either in the Project Setup window or with the Set Timecode at Cursor command, Bars+Beats set as your Primary Time Cubase will prompt you to see if you want Display and Timecode as your any existing objects to move so they stay at Secondary Time Display. So to set the the same timecode position. Most of the time Project Cursor to a timecode location, you'll want to say 'No', when working with click in the Secondary Display area on bars and beats. And spot the missing word... the Transport Panel and type in the required timecode value. After this, you can insert a Marker by clicking the Add Marker button on the Marker track itself, or by pressing Insert on Windows-based systems (no default Key Command is assigned on the Mac version), and the Marker is added at the Project Cursor's position. You'll probably want to give the Marker a name to indicate what's going on at the time of that hitpoint, such as 'Man types in Marker name in Cubase', and to do this, first make sure you have Show Marker Names enabled in the Event DisplayMarkers page of the Preferences window. When a Marker is selected on the Project window, its Name ID and Start time are shown in the Event Infoline, and you can enter a name by clicking underneath the name field, typing the name and pressing Return. Marker data can also be edited in the Inspector, so long as the Marker track is selected, and also in the Marker window, which you can open by selecting Project / Markers or pressing Ctrl/Apple+M. In these latter two views, the Marker name is listed in the Description field. The last thing you might want to consider when using the Marker track for

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hitpoints is to switch its timebase from the default Musical option to Linear, which you can do by clicking the illuminated note button on the Marker track so it changes to a non-illuminated clock. Musical timebase is the setting to which all Cubase tracks default, with the exception of the Video track, and this means that Events on a track are stored in relation to their music position in bars and beats. Therefore, if you have an event at bar three and you change the tempo, the event still occurs at bar three, but the exact time at which it occurs in minutes and seconds will have changed depending on whether the new tempo is faster or slower. Linear time, by contrast, stores Events in relation to their absolute position in time, regardless of bars, beats and tempo. If the Marker track was left in Musical time, the Markers representing the hitpoints would drift depending on the tempo, which is absolutely not what you want. Enabling Linear time on the Marker track prevents this.

Slave To The Cubase Dealing with an external video device in Cubase is slightly easier than using the built-in player, even though it offers less integration. As long as your external device is set up with the picture, any dialogue, effects and music tracks, and ready to receive incoming timecode, the amount of setup required in Cubase The MIDI Timecode Destinations group is fairly minimal. In terms of timecode, Cubase is only capable of allows you to set which MIDI ports in your system will output MIDI Timecode from outputting MTC, although many Cubase. MIDI interfaces can translate this into LTC (Linear Timecode, which carries SMPTE timecode data as an audio signal) if needed, and even if yours doesn't, you can use a device like Rosendahl's MIF MIDI Time Code box (www. rosendahl-studiotechnik.de/mif3.html) to do the job. To output timecode to a MIDI port in Cubase, first make sure the correct frame rate and SMPTE start time are set in the Project Setup window, which can be opened by selecting Project / Project Setup or pressing Shift+S (see last month's Cubase Technique article for more information about this). Next, open the Synchronisation Setup window by selecting Transport / Sync Setup or Control/ Command-clicking the Sync button on the Transport window, and enable the appropriate MIDI port to which you want MIDI Time Code to be sent in the MIDI Timecode Destinations group. Underneath this group, you'll notice an option labelled 'MIDI Timecode Follows Project Time', and when this is enabled the MIDI Time Code that Cubase outputs will precisely follow the Project's playback. So when you set up loops or relocate the Project Cursor during playback, the MIDI Timecode will reflect the position of the Project Cursor exactly. If you're experimenting with ideas, such as looping, but would like the timecode to continue as if the Project was still playing in a linear fashion, disable this option, and the timecode will be continuous from the point at which you start playback until you press stop.

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Doing The Time Warp Once you've created some hitpoints, the next step is to come up with a musical structure to incorporate these hitpoints in a way that makes sense. Cubase SX2 introduced the Time Warp tool, allowing you drag bars and beats on to specific linear time positions, and, as you can imagine, combing the Time Warp tool with the Markers is a pretty handy way of building a tempo map in Cubase. As a simple example, the Marker we created in the previous section at 01:00:30:18 happens to fall at beat 15.4.3.0 in the musical timebase. Since it's almost hitting bar 16, you might want to simply move bar 16 so 01:00:30:18 hits this musical location. To do this, select the Time Warp tool, make sure Snap mode is enabled and set Snap to Events. Now, drag the first beat of bar 16 in the Marker track onto the Marker, and because Snap is set to Events, you'll notice that the bar line you're dragging automatically locks to the Marker as the mouse pointer gets close. And, as an aside, if you don't drag within the Marker track area, the bar line won't snap to the Marker. The Time Warp tool works by automatically adjusting the last tempo change in the Project so the bar or beat you're dragging will hit the required position. In the current example, the tempo at the start of the Project is adjusted because we don't Here you can see tempo changes in the have any other tempo changes in the example described in the main text before Project right now, which means the processing with the Process Tempo operation. tempo at the start of the Project will now be 121.008bpm. This means that bar three now hits 01:00:04:23 instead of 01:00:05:00; being one frame out might not matter too much, so you might be happy to live with it. However, what if being one frame out did matter? One possibility to consider is to put bar three back to its original location and put a subtle tempo change halfway through, maybe at bar nine, instead. Bear in mind that normally you could choose a location that made sense musically, instead of an arbitrary position, but this is just intended to illustrate the process. Start by undoing the previous operation where bar 16 was dragged to 01:00:30:18, so that the tempo at the start of the Project is once again 120bpm and bar three hits 01:00:05:00. Next create a Tempo Event at bar three by Shift-clicking bar three in the Event Display (not in the Ruler) with the Time Warp Tool. This, in effect, locks bar three to 01:00:05:00, as it's impossible for the Time Warp tool to affect the tempo behind bar three at this point. Next create another Tempo Event at bar nine; again, this effectively locks all the bars behind bar nine from being affected by tempo changes after bar nine. Now,

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you can drag bar 16 to the Marker again at 01:00:30:18 and the last tempo change will be adjusted — in this case the tempo change at bar nine — so that bar 16 hits the required timecode, leaving all the bars before bar nine (including the all-important bar three) alone. And that's basically all there is to using the Time Warp tool to build tempo maps. One nice thing about the Time Warp tool is that as you hover the mouse around the Event Display, an info box is drawn detailing the current musical time and timecode position of the mouse. This is a quick way to see what the timecode value of a bar or beat is without moving the Project Cursor. Another point is that you'll notice that when the Time Warp tool is selected, Cubase draws the locations of tempo changes in the Ruler, indicated by small triangles. In the same way that Shift-clicking in the Event Display creates a new tempo change, Shiftclicking a Tempo Event in the Ruler deletes it.

Cubase News Following the release of Cubase SX/ SL 3.1, Steinberg have released an update patch for both Mac and Windows users to v3.1.1 (build 944) that can be downloaded from the company's FTP site at ftp.steinberg. net/Download/ in either the 'Cubase_SX_3/3.1.1.944/' or 'Cubase_SL_3/3.1.1.944/' folders. Although there are no real new features in this release, there's a big list of bug fixes, including improved OMF handling. The ability to Ctrl-Tab between open windows is working again, and Cubase will no longer crash when dragging files between the desktop, Pool and Project windows, working with multiple Projects that use Studio Connections, playing back automation data to MIDI Device Panels, moving Parts nested within a Folder, or pressing Ctrl+R to open the Score Editor. So basically, if you found version 3.1 crashing quite a bit, 3.1.1 should help. And good news for SL users: the Track Folding feature that was supposed to be in the 3.1 update was accidentally omitted, but it's been included in the 3.1.1 update. Mac users who had purchased Steinberg's System 4 (www.soundonsound.com/ sos/jul04/articles/cubasesystem4.htm) have been unable to upgrade to Mac OS X Tiger due to the driver for the MI4 USB audio and MIDI interface being incompatible with this latest version of the Mac operating system. Fortunately, Steinberg have now addressed this issue and released a Tiger-compatible version of the MI4 driver, which can be downloaded from www.steinberg.de/ DocSupportDisplay_sb1488.html. Unfortunately, this driver has only been approved for running the MI4 with Cubase SL 3.1, and Steinberg state that it's incompatible with Cubase SL2, which was the version supplied as part of the original System 4 bundle. System 4 is now supplied with Cubase SL 3.1 and the Tiger-compatible driver, and Steinberg recommend that existing users contact their local Steinberg dealer for an upgrade to SL 3.1 at an "extremely attractive price".

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Process Tempo The Time Warp tool is truly a great aid for building tempo maps, but it has a couple of limitations: because only the last tempo change is adjusted when you drag a bar, it means you can only affect one tempo change at a time, and it has to be an immediate (what Cubase would term a Jump Event) change. What if you wanted to adjust several tempo changes simultaneously so that you could affect a sequence of Tempo Events proportionally to hit a cut in the picture? For this, you'll need the Process Tempo command. Forgetting the previously discussed examples, imagine you have a 24fps Project starting at 01:00:00:00 with an initial tempo of 120bpm. At bar five there's a Jump Tempo Event to 110bpm, and at bars 13 and 21 there are two Ramp Tempo Events to 124 and 115 bpm respectively (see the screen on the previous page). Bar 25 currently hits 01:00:48:20, but you just got a new version of the picture and you need to make the music slower as the picture is now a bit longer and bar 25 now needs to hit 01:00:53:02 instead. To do this, open the Tempo Editor and click the Process Tempo button on the Tempo Editor's toolbar. If you don't see the button, right-click an empty space of the Tempo Editor's toolbar and make sure Process Tempo is selected in the list of elements to be displayed on the toolbar. In the Process Tempo window, make sure Time Display Format is set to Process Tempo is a powerful command to Timecode. Next, set the range of the process tempo changes within a given range Project where tempo changes should in order to hit a specific timecode position be processed in the Process Range with a musical location. section. If you select a group of Tempo Events in the Tempo Editor before opening the Process Tempo window, Cubase will automatically set the Process Range according to the selection. However, in this case, set the Start to bar one (1.1.1.0) and the End to bar 25 (25.1.1.0) — the End value should always be the musical location you want to hit with a given timecode position. Next, set the End value in New Range to the timecode value you want to hit, in this case 01:00:53:02, click Process, and then click Close to close the window. You'll notice that Cubase scales the tempo changes; and because there was no Tempo Event at bar 25 (the bar at the end of the range), a tempo change will be automatically inserted to preserve the original tempo at that point. However, if you want to continue with the same tempo during bar 25 as before, simply delete the newly inserted tempo change as this will have no effect on bar 25 itself hitting the correct tempo position. If you check bar 25, you'll see it now hits 01:00:53:02 file:///F|/SoS/SoS%2012-2005/cubasenotes.htm (7 of 8)11/23/2005 3:06:42 PM

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as we intended. And that's all there is to it! Having read both this and last month's article, I hope you now have a better understanding not just of how to use Cubase's videorelated features, but also how to decide which ones suit the task at hand. Whether using video inside or outside of Cubase, and whether you use the Time Warp tool or Process Tempo (or both) for building your tempo maps, all that remains is for you to actually write the cue and sell it to a director! And, unfortunately, there isn't a plug-in that can do that for you just yet. Published in SOS December 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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

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Music Publishing

In this article:

Music Publishing

The Language Of Publishing Everything You Wanted On The Register Published in SOS December 2005 Protection Collecting For Fun And Profit Print article : Close window Do I Need A Publisher? Music Business Shared Ownership Getting Ahead

To Know (But Were Afraid To Ask)

If you want to make money as a songwriter, composer or lyricist, the obvious answer is to find yourself a publisher. But what do music publishers actually do for their clients? Why do you need one, and how can you find the right one? Scott Rubin

As a music publisher, I'm always answering questions about my business. I would say that the most common one is "What exactly does a music publisher do?" The simple answer would be that I work with songwriters, who compose music and/ or lyrics, just as a book publisher would work with an author. Many individuals in the music business, unless they're directly involved in music publishing, have a poor understanding of the particulars of publishing. Couple that with the fact that the laws and procedures vary from country to country and it often becomes quite a daunting task to get a solid understanding of the field. However, in this article, I will attempt to bring some clarity to the complexities of music publishing. Songwriters can use this guide to learn about the basics, from both a US and UK perspective when possible — the laws, registration procedures, collection guidelines and licensing issues differ between the two countries. Whether you're based in the US or the UK, there are plenty of options open when you decide that you are ready to work with a publisher. Before you start to look for a publisher to work with, you should first understand the three basic types of publishers. The first is described as an administrator. Usually an individual or small company, they provide a service, for a small commission, to the songwriter file:///F|/SoS/SoS%2012-2005/allaboutpublishing.htm (1 of 9)11/23/2005 3:07:05 PM

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by handling all aspects of the registration, licensing and collection processes which I'll outline later. However, they do not normally pay advances (which I'll also discuss later in the article) and usually do not offer any creative services. The next level of publishers are called 'independents', a category which includes my company, Reach Global. An independent offers the same administration services as an administrator, but also provides creative services and offers competitive advances to songwriters. Their client lists would usually be made up of mid-level artists plus talented songwriters and producers. Most big stars like Jay Z and Eminem align themselves with the third type of publisher, called a major. Major publishers like Universal, Warner Chappell, Sony, BMG and EMI pay millions of dollars in advances to the songwriters and artists in order to maintain their market share.

The Language Of Publishing As a songwriter, publishers' contractual language will often look foreign to you. Copyrights are assets, just like a piece of land or a house that you own. If you know how complex a mortgage contract is for a home, it's no surprise that a publishing agreement can often be difficult to understand. Don't sweat it; there are a few basic terms that will give you a foundation for understanding the scope of the publishing agreement. Here are a few definitions: Advance/Recouping: This is the financial arrangement within the agreement where the publisher advances money to the songwriter, before the publisher has collected any income. This concept is often used to satisfy the financial needs of the songwriter, as the entire process of registration, licensing and collection can often take a year or more to come to fruition. The songwriter should understand that if the publisher has paid them an advance, they will not receive any royalties until the publisher makes back their money and turns a profit, often known as 'recouping'. Split: This is defined as the share of income that both the publisher and songwriter will be entitled to. Percentages often range from 90/10 in favour of the songwriter to 50/50, usually common when a company invests a large amount of money upfront to a songwriter as an advance against future royalties. Co-publishing Deal: This is a certain type of publishing deal where the publisher and the songwriter are each 50 percent owners of the copyright. They agree to share the income from the copyright on a 50/50 basis. Administration Deal: Without a share of the copyright, the publisher agrees to service the writer by collecting all sources of income and handling all aspects of administration on a commission basis. The range usually goes between 5 and 25 percent but varies from deal to deal. Term: This defines how long the agreement is actually binding, and could be anywhere from six months to forever. Territory: Refers to the 'territories' or countries where the agreement is binding. It is common to have a worldwide agreement, or sometimes to carve out particular countries, such as when a publisher negotiates an exclusive deal solely for Japan or the United Kingdom. MDC: This stands for the 'minimal delivery commitment'. It is the amount of material or songs that a songwriter is required to deliver (for commercial release) during the term of their agreement. The MDC is based on 100 percent ownership of the song. A songwriter who agrees to a MDC of five songs will have to deliver 10 songs if they only write the

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Music Publishing

music and thus get only a 50 percent share of the song. This is a common occurrence since most songwriters specialise in either writing the music or the lyrics but not both.

On The Register What does a music publisher do? What kind of royalties do songwriters earn? The general answer I gave above can be fleshed out by saying that a music publisher is responsible for four basic areas of importance to a songwriter. Those areas are song registration, licensing, royalty collection and creative matters. Let's break those areas down. First off, it all begins with paperwork. The registration process can sometimes be a boring and tedious job that only a publisher could enjoy. During the initial song registration process, the publisher usually informs ASCAP, BMI or PRS about the new song and relays all the relevant information to them. What are ASCAP, BMI and PRS? It is sometimes thought that they are publishers, but they are not: they are 'performing rights societies'. The American Society of Composers, Authors and Publishers (www.ascap.com) and Broadcast Music, Incorporated (www.bmi.com) are American performing rights societies, while the Performing Right Society (www.prs.co.uk) is the equivalent in the United Kingdom. They provide a service by monitoring, collecting and paying out 'performance royalties' to publishers and songwriters. These are royalties that are paid to songwriters and publishers whenever the song is, for example, played on the radio or on TV. Virtually all radio stations and TV networks pay How royalties are collected in the millions of dollars each year to ASCAP, BMI and United Kingdom. PRS for what is called a 'blanket licence', which allows them to broadcast any song they wish, as many times as they like. Once the song is registered by a publisher, the performing rights societies will collect and pay out these performance royalties directly to the publisher and songwriter. Songwriters must choose to affiliate and register their work with only one society in the US; a publisher can help you decide which. In the UK, there is only one performing rights society, the PRS. Performing rights societies will pay equal shares of the performance royalties they receive to the songwriter and to the publisher. Assuming the songwriter has a split of more than 50 percent (see 'The Language Of Publishing' box), the publisher will then pass on the appropriate percentage of their share to the songwriter. Those are, of course, not the only type of royalties a songwriter can earn. There are royalties that get paid when a song is included on an album that is

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commercially released in retail stores or made available legally on-line for downloading. These are called 'mechanical royalties'. How does one collect such royalties? That brings us to the second main area of publishing operations: licensing. This is one of the areas where the procedures are different for the United States and the United Kingdom. In the US, before mechanical royalties can be paid out to the publisher by the record label, you first have to license your song to the record label. When the publisher issues a licence to the record label, it allows the song to be included and sold on a particular album. The record company can then pay mechanical royalties according to how many copies of the album are sold. Your rights and payment rates are set down in this licence, so I don't recommend signing a mechanical licence that is directly sent to a songwriter from a record label without the review of a publisher or lawyer. The procedure is a bit different in the UK. Just as PRS is the Performing Right Society, the UK has a 'mechanical' society called the MCPS (Mechanical Copyright Protection Society, www.mcps.co.uk). The record label which releases the album in the UK does not pay the publisher directly, as they would in the US. Instead, the record label pays the MCPS a certain amount of money, based on sales of the album. Once the publisher registers the song with the MCPS, they can then collect mechanical income from sales in the UK. There is also an organisation called the Harry Fox Agency which deals with the licensing and collection of mechanical royalties. For a commission, the HFA can collect mechanical royalties via record sales throughout the world. Although some publishers and writers rely on the HFA, it is an optional service as publishers have the ability to collect mechanical royalties without the aid of the HFA. Music publishers also must be knowledgeable about new technology and the licensing of additional revenue How licensing works in the United States. streams such as ringtones — songwriters must have a publisher issue a separate licence for ringtone use, and the popularity of ringtones among teenagers leads me to believe that this source of income is not a passing fad but will only get bigger in the years to come. In the US, the publisher will issue a licence to a phone carrier (Verizon Wireless, Nextel or Sprint for example) or a ringtone provider such as Blingtones or Zingy. The publisher would then receive the income directly from the ringtone provider. In the United Kingdom, the income is collected by the collection society rather than directly from the source, just as with mechanical royalties. Generally the income on a ringtone is about 10 cents (for 100 percent of the copyright) in both the US and the UK. With the worldwide explosion of DVD, songwriters should also be aware that a file:///F|/SoS/SoS%2012-2005/allaboutpublishing.htm (4 of 9)11/23/2005 3:07:05 PM

Music Publishing

separate licence is needed for collecting the royalties generated by this popular format. The royalty is usually about 12 cents (for 100 percent of the copyright), though an advance payment fee is also sometimes available through negotiation. The income structure is about the same for both the US and the United Kingdom, though you need to keep in mind that in the US, the royalties can be collected directly from the manufacturer, while the royalties in the UK have to be collected by the society.

Protection An important part of the job of a publisher is to protect songwriters' rights. The most common way to do this is by formally registering your songs for copyright protection. In the US, this is done via the US Copyright Office. In the UK, copyright exists automatically in any written or recorded work, but registering a song with the MCPS is a good way of establishing your claim to have written it in case of dispute — better than sending it to yourself in a Jiffy bag, anyway! As a benefit to the songwriter, a publisher often will play the role of 'bad guy' when working with another party to make sure the songwriters' rights are properly established and granted. In the United States, one of the larger issues in terms of protection comes into play when a publisher attempts to license a song to a record label. Since the record label is the entity paying out the mechanical royalties, they can attempt to pay less than the amount they are required in accordance with US copyright law. (According to copyright law, the current royalty rate you are entitled to receive — called the 'statutory mechanical rate' — is currently 8.5 cents if you wrote 100 percent of the song.) They accomplish this not by breaking the law, but by having you sign documentation that allows the record company to pay you a lower royalty rate or pay royalties on fewer albums. By asking songwriters to accept less money than they are potentially entitled to receive, record companies save millions of dollars in mechanical royalty payments each year. If you work with a publisher, they are experienced enough to not allow the record companies to issue the royalties at a reduced rate, unless the label is actually entitled to according to a prior written agreement (known in industry parlance as a 'controlled composition' clause). The concept of 'statutory mechanical rate' does not apply to the United Kingdom, where rights are a bit easier to manage and protect. However, a UK publisher must still register the song correctly with the mechanical and performance societies to ensure proper and timely payment. Songwriters also need to take care to protect themselves when they sign a publishing deal, often by simply using common sense. Remember the old saying and be careful what you sign. Always have someone with more knowledge than you in publishing, such as a lawyer, look out for your interests. They will guide you to the proper publisher who will protect your rights.

Collecting For Fun And Profit So, through registration and licensing, we've come to the third part of the publisher's role: royalty collection. Song registration and licensing allows the file:///F|/SoS/SoS%2012-2005/allaboutpublishing.htm (5 of 9)11/23/2005 3:07:05 PM

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publisher to collect your proper amount of royalties from all sources. We've already discussed performance and mechanical royalties, but there is another royalty type called 'synchronisation' royalties. These usually occur as a one-off negotiated flat fee whenever music is 'synchronised' to a moving picture. By way of example, sync fees are paid if your song is used in a movie, TV show, commercial or video game. An experienced publisher will negotiate the sync fees and issue the requisite licences. (Future residual royalties are further paid by the performing rights societies for many sync rights.) A music publisher works diligently to collect all of these income streams from the various sources. Lastly, there is the fourth element: creative exploitation of the song. Publishers can often spend a large part of their time attempting to 'pitch' the song to advertising agencies, music supervisors who work in film and TV and video game producers. The creative energies of a publisher can bring untold new opportunities to songwriters both artistically and financially.

Do I Need A Publisher? As a songwriter, how do you decide whether you need a publisher or not? I always tell my clients that if they are doing more work on the administration of their How mechanical royalties songs than on the creative process, then it might be are collected in the US. time to relinquish some control of their catalogue to a publisher. If you are equally as talented with business and the creative arts, you might find it challenging to try and work your own catalogue. However, the administrative processes that I have outlined above will quickly hamper the creative process without a firm grasp of the issues, especially if your songs are played and sold all over the world. For example, if you have a big hit in Japan, can you speak Japanese and make sure you're getting all the money you deserve? I bet not. If you compare the issue to other business concepts like taking on a manager or finding a lawyer to work with, you will probably lean towards working with a publisher. And why not — you should be more focused on the creative process than anything else. The act of actually picking out a publisher to work with should be aided by consultation with an experienced music professional like a manager or musicbusiness lawyer. You shouldn't be afraid to ask questions of your peers: Are you happy with their service? Do they pay royalties on time? What have they done for you in terms of creative exploitation? In addition, you should research the company you are attempting to link up with. Do they have four other songwriters just like you? Are you going to be a forgotten face or a small fish in a big pond? Ask yourself what exactly you're looking for in signing with a publisher.

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Shared Ownership Some songwriters always work alone; others form duos or teams, and it's not uncommon to find half a dozen writers credited on one song. Whether you have a publishing deal or not, there remains a certain amount of responsibility that lies with you, the songwriter. A publisher cannot protect your rights if you erroneously give them away. When you're collaborating with another person or persons, there will hopefully be a wonderful creative vibe throughout the process, but don't neglect to address the business element once the song is complete. Don't be afraid to suggest the shares that you think are appropriate for your contribution to the song. When possible, put in writing the songwriters and their appropriate shares, and have it signed by everyone who participated. This could be done on a simple, handwritten piece of paper if need be; it is better than nothing. That will often avoid problems in licensing and registration, which will certainly affect how and when you get paid. Traditionally, there are common share percentages for certain aspects of songwriting. The writer who composes the music is usually entitled to 50 percent and the writer who delivers the lyrics receives the other 50 percent. In addition, it is also quite common to share equally the percentages, based on the number of songwriters, regardless of their individual contributions. For example, four songwriters on a song would each get 25 percent of the copyright. However, in today's urban and pop music landscape it's common to have at least four to six writers on the song, not counting any samples that might exist in the song. In that case, someone like a manager or publisher should take control of the process by getting parties to all agree on their respective shares.

Getting Ahead That's all very well if you have a track record as a songwriter and can choose to sign with any of a number of publishers. But what if you're not at that stage yet? How should an unknown songwriter go about finding a publisher and getting his or her songs sold? I recommend that as a songwriter, you take as much interest in your career as possible. That means joining societies like NARAS (National Academy of Arts & Sciences — the Grammy people, www.grammy.com) in the US or their equivalents. In addition, attending performance rights society functions hosted by ASCAP, BMI or PRS are a wonderful way to network and expand your relationships. All three of them regularly host songwriter workshops and seminars to assist their members. As a publisher, I get a large amount of unsolicited material into my office. Even if I do not think the material is right for our company, I listen to everything that comes my way. Songwriters often complain that they do not think the material they send gets heard. I can say from personal experience that it is just not true. Most good publishers are always on the look out for the 'next big thing' and we don't know where it's going to come from. As such, we check everything. As a file:///F|/SoS/SoS%2012-2005/allaboutpublishing.htm (7 of 9)11/23/2005 3:07:05 PM

Music Publishing

songwriter, though, there are things you can do to get on the radar of a publisher and things you can do to turn them off to you. When sending out demos, don't worry too much about packaging and graphics, or for the most part, sonic clarity. I don't suggest sending out a very poorly recorded demo that you did on your grandmother's old cassette player, but professional publishers can hear a good song even if it is not mixed properly or contains sounds that are a bit dated. Your concentration should be on the creation of the best material possible, with clear contact information listed on the product, so the publisher can contact you. Professional-sounding phone calls are always acceptable, though in my opinion, most songwriters find it hard to play the line between professionalism In the US, songwriters can choose to sign up and eagerness to be signed. Just like with one of two major performing rights with dating, professional publishers societies, ASCAP and BMI. can smell desperation. If you leave 100 phone messages without any of them being returned, it might be a sign that the publisher doesn't think your music is right for them. Publishers spend a large part of their day speaking professionally to other publishers, record labels, managers, lawyers and so on. What's my point? They aren't afraid to talk on the phone. If they want something from you, they'll be in touch with you so you both can start a business relationship. Another fact that most songwriters fail to understand is that this is a business: this isn't a 'game' or a 'hustle'. It's not personal, either. The quicker you, as a songwriter, have that sink into your head, the better off you'll be. So what if a certain publisher doesn't think your songs are ready for a star to record? There are tons of other publishers and you have to stay positive. Remember that each type of publisher has its pros and cons. What's perfect for one songwriter isn't so for another. Choose the one that fits you best. The important thing is that you should know as much as possible about your career. If that means reading articles like this over and over, so be it. If that means asking questions among your peers, do it! What's the bottom line when it comes to publishing? Just be creative — the right music publisher can handle the rest! Scott Rubin is co-founder and Vice President of Reach Global, Inc., an independent music publishing company based in New York and South Florida. Published in SOS December 2005

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Music Publishing

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

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

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