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Shaped By Technology W
hen sequencers first appeared on the scene, I was concerned that they would distract musicians, specifically keyboard players, away from learning to perform to the best of their ability. After all, MIDI performances could be tidied up after the event, or even typed in as a series of notes and velocities without a music keyboard ever being involved, so why learn to play ‘properly’? The jury is still out on whether musicianship suffered or not, but what can’t be denied is that the sequencer spawned new genres of music based on rigid timing, hard quantisation and sampling which still exert a strong hold over the pop music scene today. This is perhaps the most powerful example of technology influencing the musical art that we’ve seen to date. There have been other ramifications of computer recording — some beneficial, others not. For example, many of today’s drummers can now play to a click track or guide groove, not only staying perfectly in time but also being able to adopt the groove of the guide part. This ability is clearly a benefit in some situations, just as long as it doesn’t leave the drummer unable to play without a click being present. On the other hand, it can be argued that playing to a pre-recorded click or groove robs the drummer of the ability to inject subtle tempo shifts into the music, something that a good drummer would always do subconsciously — for example, to push a chorus or stretch a fill. However, even I was surprised at the latest example of technology affecting performance. During one of our Studio SOS visits a few months back (the results of which you can see in this very issue), Hugh and I listened to a demo recording on which the studio owner’s daughter had provided the vocals, and we both thought that he’d been too heavy-handed with Auto-Tune or some similar pitch-correction plug-in. He told us that he hadn’t used any pitch-correction processing at all, that was just the way she sung. It turns out that she’d grown up on a diet of pop music in which heavy pitch-correction was the norm, and she’d learned to sing by emulating what she heard on record. I mentioned this to UK producer Steve Levine when we met to sit on a panel earlier this year and he said he’d also come across this development, specifically female vocalists who had learned to pitch very precisely and to move cleanly from one note to another without the normal glides and slides you hear in a typical unprocessed vocal. Which leap of technology will be the next to have a big influence on music making is anybody’s guess, but wouldn’t it be ironic if a software company came up with a plug-in specifically designed to Paul White Editor In Chief

put back the very human vocal artifacts that a generation of Auto-Tune indoctrinated singers have trained out of their own voices? And if that became as over-used as pitch correction clearly has, maybe we’d see a new generation of vocalists grow up with even stranger vocal characteristics. Say what you like about tape, at least it had relatively little influence over the types and styles of music recorded on it!

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The contents of this publication 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 Publisher. Great care is taken to ensure accuracy in the preparation of this publication but neither Sound On Sound Limited nor the Editor can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the Publisher or Editor. The Publisher accepts no responsibility for the return of unsolicited manuscripts, photographs, or artwork. © Copyright 2012 Sound On Sound Limited. Incorporating Music Software magazine, Recording Musician magazine, Sound On Stage magazine, SPL magazine, Sound Pro magazine and Performing Musician magazine. All rights reserved. Where prices are given in US Dollars, these exclude local taxes. Review products tested are UK versions. Price comparisons pertain to the UK market and may or may not apply in other territories. SOS recognises all trademarks.

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w w w . s o u n d o n s o u n d . c o m / January 2012


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Neumann TLM 102

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January 2012 / issue 3 / volume 27

18 Off The Record
Economic uncertainty and the rise of digital distribution have taken their toll on many major studios, but there is still a future for them.

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124 Spotlight
Looking for a handheld audio recorder? Check out our overview of a selection of current models.

126 Studio SOS
The SOS team travel to Nottingham, where optimising vocal recordings and tackling some troublesome mixes are on the agenda.

132 Mix Rescue
This month, we aim for more punch, depth and clarity in a quirky urban track.

The single ‘Downtown’ gave Petula Clark a worldwide hit and rejuvenated her career. Presiding over the session was engineer Ray Prickett, who tells us how it happened...

138 Practical Noise Reduction
Whether it’s tape hiss, guitar hum or a slamming door that’s the problem, tools and techniques are available to rescue your noisy recordings.


166 Mandy Parnell:

Mastering Björk’s Biophilia

The mastering engineer’s role is changing as artists explore new formats — and what works for the iPad might not work on CD.

170 Jake Gosling:

Producing Ed Sheeran

Ed Sheeran’s phenomenal success depended on hard work, a few lucky breaks, and the talents of long-term co-writer and producer Jake Gosling.

175 Loudspeaker Orchestras
Many musicians spend their lives trying to cram hundreds of tracks into just two speakers — but performing with an NSML system presents exactly the opposite challenge.

180 Inside Track: Randy Staub
The multitracks for Evanescence’s third album were so big that they required two maxed-out Pro Tools rigs to play back!

187 Notes From The Deadline
We all have musical skeletons in our closets, but at least media composers don’t have to wheel them out every night at Wembley Arena.


188 The Mix Review
Our tame engineer examines the mixes of tracks by Rizzle Kick, Foo Fighters, Christina Perri and Nicki Minaj.

144 Learn to edit audio in Pro Tools with the QWERTY keyboard. 146 Make the most of Live’s Grain Delay effect. 148 Improve project navigation in Cubase via keyboard shortcuts. 150 Get to grips with vinyl slow-down effects in Logic. 152 Explore creative phase-switching in Sonar X1. 154 Explore the new features and functions of Reason 6. 156 Use the MachFive 3 software sampler with Digital Performer.

192 Playback
The SOS team break their New Year’s resolutions almost immediately by gorging themselves on readers’ demos.

Win! A full studio setup from

Samson & Zoom


158 Record your own impulse responses in Reaper.

30 34 40 44 48 56 64 70 72 74 80 84 84 86 88 96 100 104


SE Munro The Egg
Monitoring System Universal Audio Ampex Tape Simulation Plug-in For UAD2 Yamaha MOX6 Synthesizer Workstation JZ Microphones Vintage 11 Large-diaphragm Condenser Mic Bettermaker EQ230P Digitally Controlled Analogue Equaliser Acustica Audio Nebula 3.5 Dynamic Convolution Plug-in Best Service Klanghaus Sample Library Teenage Engineering OP1 Portable Synthesizer Soundhack PVOC Effects Plug-ins Blackstar HTI 1W Valve Guitar Amplifier Native Instruments Kontakt 5 Software Sampler Magneto Labs VariOhm Microphone Impedance Converter Fostex HPP1 DAC & Headphone Amp for iOS IK Multimedia iRig MIDI MIDI Interface For iOS TK Audio BC1 VCA Compressor Limiter Steinberg Sequel III Loop-based Music Production Software AEA KU4 Supercardioid Ribbon Microphone Arturia Analog Laboratory Soft Synth & Controller. Coleman Audio QS8 Monitor Controller PSP Noble Q EQ Plug-in Sony MDR 7520 Studio Headphones Propellerhead Balance USB Audio Interface Electro-Harmonix Ravish Sitar-emulation Effect Pedal. Kinman P90HX Noiseless Guitar Pickups. Zaor Stand Monitor Speaker Stands. Turbosound Mi0 Active Loudspeaker Presonus Studio Live Digital Mixing Console Phonak Audéo PFE 232 In-ear Monitors Sample Libraries Libraries from Ueberschall, Big Fish Audio, Toontrack & Soniccouture.

The other mics in JZ’s Vintage range are based on classic designs of the past — but they hope this one will become a classic in its own right.


108 Sony’s MDR range of

closed-back cans has been popular in studios for years. Discover how the latest top-of-the-line model compares.

44 With an all-analogue signal
path and total recall courtesy of digital control, this EQ promises a lot. Find out if it delivers...

106 108 110 114 114 115

8 194 196 197 202
News Apple Notes PC Notes Q&A Sounding Off

116 118 122 190

The toy-like exterior of Teenage Engineering’s OP1 belies the versatile synth, sequencer, sampler and recorder hidden within. We retreat to our bedroom and put it though its paces.


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Monotron Delay and Duo analogue synths from Korg

Monotrons for everyone
On, the far right are the controls for delay time and feedback. Korg describe the delay as “analogue-like” (it is, in fact, digital), and say that adjusting the delay time while sound is passing through the unit results in the kind of


ack in 2010, Korg released the Monotron, their first all-analogue synthesizer in a quarter of a century, to great acclaim. They followed this up with the more fully featured Monotribe, which incorporated analogue drums and a sequencer, and they’ve now released two more analogue playthings: the Monotron Delay and Monotron Duo. Both are derived from the original Monotron, and respectively add a built-in delay and a second oscillator to the original. The two new instruments are based around the now-familiar ribbon controller, although the Monotron Delay dispenses with the Pitch control of the original. Its LFO is hard-wired to modulate the oscillator’s pitch, unlike on the original, where it could also optionally modulate the filter’s cutoff frequency. The Monotron Delay does, however, offer a choice of LFO waveforms: triangle or square. The filter is based on that found on Korg’s classic MS20, and although its frequency can be adjusted via a knob, the amount of resonance is fixed.

pitch fluctuations produced by a tape echo. The Monotron Duo, meanwhile, has two oscillators, each with its own pitch control. Accompanying these is a knob marked X-Mod, which cross-modulates the frequency of the second oscillator with that of the first — a technique Korg first used in their Mono/Poly synth. Another

feature unique to the Monotron Duo is the Scale Select switch on the back. This has three modes (chromatic, major and minor), and when engaged, it ‘locks’ the pitch you play on the ribbon to a valid note (the nearest semitone, for example, when in chromatic mode). Like the original, the new Monotrons run on two AAA batteries, and have an internal speaker as well as a headphone output. Both newcomers feature a mini-jack aux input, which lets you process external sounds through their analogue filters — and, in the case of the Monotron Delay, apply echo to them as well. The Monotron and the Monotribe have both been popular among DIY fans, who have managed to upgrade their analogue toys with features such as MIDI and CV control, and filter key tracking — and we expect the same to be true of these latest additions to the ‘Mono’ range. The Monotron Delay and Monotron Duo should be available from late January. Pricing was yet to be determined at the time of writing. Korg USA +1 631 390 8737


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Radial take control
MC3 monitor controller
anadian hardware makers Radial have released a studio monitor controller called the MC3. Built into the same folded-metal chassis as their popular DI boxes, the MC3 has two inputs (for a stereo line-level source), and six line outputs: two pairs for two sets of stereo monitors, plus a subwoofer out, and an aux out (these last two being mono-summed versions of the stereo input). The MC3’s top panel houses trim pots for setting the subwoofer output’s level and polarity, the maximum levels for the monitor A and B output pairs, and the attenuation imposed by the dim switch on the front. Next to the dim switch are buttons for mono summing and subwoofer mute, and a pot for monitor level. The two sets of monitors can be turned on and off independently (so you can switch them both on or both off, as well as simply switching between pairs of speakers). Finally, there’s a headphone-monitoring section to the right of the unit, with two quarter-inch sockets and one 3.5mm socket. These headphone outs are all controlled by a single headphone level knob. All switching and attenuation is performed passively, according to Radial, which means that the unit’s circuitry should have no audible effect on


line-level signals passing through it. The MC3 derives its power from an included 15V DC power supply. Pricing for Radial’s new monitor controller was yet to be announced at the time of writing. Radial +1 604 942 1001

Ace of Grace

Premium hardware makers Grace Design ( ) have released two new single-channel optical compressors: the M102 and M502. The former is a 1U-high, half-rack-width processor, while the latter is an API Lunchbox-compatible module, though both use identical circuitry and have almost exactly the same set of features. Employing feedback topologies, the M102 and M502 are said to be neutral and transparent, “regardless of how much you squeeze”. Their compression ratios are continuously variable between 1:1 and 12:1, with attack and release times from 3-200 ms and 30-3000 ms respectively. Both have 10-LED gain-reduction metering, an input-peak indicator and a bypass switch. They also have external side-chain inputs, and can be used in stereo linked mode. On the M102, the side-chain and linking features are implemented using jack sockets on the rear, while on the M502 these options are down to Grace’s decision to use Radial’s Workhorse standards (the Workhorse is an advanced version of API’s Lunchbox, which accommodates an extra input per module compared with the API design). The M502 can additionally make use of the Workhorse’s bussing abilities. Pricing for the M102 and M502 wasn’t available at the time of writing, but more information should be available on Grace’s web site by the time you read this.

w w w . s o u n d o n s o u n d . c o m / January 2012


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Izotope’s finishing touch
Ozone 5 mastering suite announced
merican software developers Izotope have announced a major update to their mastering plug-in suite, Ozone. Version 5, like its predecessors, is a ‘compound’ plug-in comprising six separate processors: a parametric EQ, a mastering reverb, a limiter, a dynamics processor, an enhancer and a stereo-width processor (the last three offering multi-band operation). All of these modules have been refined and enhanced, in terms of both sound and usability, so not only are the algorithms improved, but the interfaces are easier to navigate too. “Ozone 5 is the biggest update we’ve ever offered,” says Izotope’s Nick Dika, “with improvements that will appeal to Ozone experts and new users alike”. For the first time, the Ozone suite will be available in two different versions: the ‘normal’ version, which has all the features of version 4 (plus the aforementioned sonic and ergonomic enhancements), and an Advanced iteration. The latter is aimed at more experienced engineers, and, as such, gives the user access to otherwise hidden parameters. The maximiser has various stereo-linking features and a transient-recovery control, for example, and a mono-compatible stereo-synthesis function is also available. Metering is also enhanced in the Advanced


version of Ozone 5, so you can view a 3D spectrogram of the material you’re working on, as well as loudness metering and a vectorscope, with stereo balance and correlation metering. Finally, Ozone 5 Advanced offers all of Ozone’s constituent processors as individual plug-ins, so you can use one processor without taxing your CPU with all the others (while mixing, for example).

The standard version of Ozone 5 carries a price of $249, while Ozone 5 Advanced will set you back $999. Upgrade paths from previous versions of Ozone are available, and if you bought Ozone 4 after 1st October, you’ll be eligible for a free upgrade to version 5. M-Audio US +1 626 633 9050

Brainworx reach new heights
Vertigo VSC2 compressor plug-in


erman plug-in coders Brainworx have released a new software compressor called the VSC2, a modelled plug-in based on the VSC2 Quad Discrete Compressor by Vertigo Sound. It’s so named because it uses four VCAs (voltage-controlled amplifiers) for the gain reduction, and is built entirely from discrete components. Brainworx say they have recreated the original so accurately that they can’t discern between the sound of the software and the real thing! All the expected controls are accessible, including threshold, ratio, attack, release and make-up gain, as well as high-pass

filtering for the internal side-chain signal (at either 60Hz or 90Hz). It can operate in stereo or dual-mono modes, and Brainworx have also made a single-channel version available, for processing mono sources. The VSC2 plug-in can be bought directly from the Brainworx web site and costs €249. If it’s as accurate as they claim, this represents something of a bargain: the hardware unit normally sells for over $6000! Check out the link below for a video of the plug-in in action. Brainworx +49 (0)217 4671 8935


January 2012 / w w w . s o u n d o n s o u n d . c o m

NAMM booth #5900
© 2012 CASIO AMERICA, INC. All rights reserved.


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MXR and Little Labs plug-ins from UA
Classic flanger and VOG bass processor released
wo new additions to Universal Audio’s range of DSP-powered plug-ins are the MXR Flanger/Doubler and the Little Labs VOG. As has been the trend with UAD plug-ins lately, both are models of existing hardware, and have all the features of their real-life counterparts. The MXR Flanger/Doubler, a rackmount processor released in the early ‘80s, was an analogue unit that offered delay and flange effects. It was popular with guitarists, and can be heard on tracks featuring Randy Rhoads and Pantera’s Dimebag Darrell. It used ‘bucket brigade’ chips to provide the delay used in both flanger and doubler modes. (Both modes are virtually identical, in fact, with the difference between them being that when set to Doubler, the delays are longer.) Its effects are created by running a dry signal in parallel with a delayed version of itself, and modulating the duration of the delay. There’s a Mix control, for setting the relative wet and dry signal levels, and a knob marked Regen governs the delay feedback. The Invert switch reverses the polarity of the


delayed signal relative to the dry, resulting in a slightly different sound when the two are summed. The Sweep section comprises Width and Speed controls, and these determine the amount and rate of delay-time modulation, respectively. When both are set to their lowest positions, there is no modulation, and the control labelled Manual sets the delay amount. This can be used to create ‘static’ flanger effects, or it can be automated from within your DAW. Adding to the original’s features are buttons marked Sync and Dual/Single. The first of these locks the Speed parameter to your project’s tempo, for synchronised modulation, while the second inverts the polarity of the left delayed signal relative to the right (when the plug-in is used on a stereo channel), for ultra-wide flanging effects. Next up is the Little Labs VOG plug-in, which is based on the API 500-series Voice Of God module we reviewed in the November 2011 issue ( articles/littlelabs-vog.htm). Described

as a ‘bass resonance processor’, it has but two knobs, marked Amplitude and Frequency. Essentially, it’s a high-pass filter with a resonant peak at the turnover frequency, which lets you perform narrow bass boosts while cutting unnecessary super-low frequencies. In combination with the Frequency control, the two Center buttons let you set the turnover anywhere between 40Hz and 200Hz, while the Amplitude knob makes the filter progressively more resonant as it is turned up. This last control also affects the steepness of the filter, which can cut up to 24dB per octave. Universal Audio describe the Little Labs VOG as a “magnifying glass for the bottom end of your mixes”, touting its usefulness on kick drums, bass, toms and vocals. These new plug-ins require a UAD2 card to run, as well as version 6.1 of the UAD software. The MXR Flanger/Doubler costs $199, while the Little Labs VOG carries a price of $149, and both can be bought directly from Universal Audio’s online shop. Universal Audio +1 877 698 2834

Alesis SamplePad
Play your own drum samples
lesis’ latest creation is the SamplePad: an affordable electronic drum pad that allows the user to trigger samples uploaded to it via an SD card slot. In addition, it has 25 of its own percussion and drum sounds. There are four velocity-sensitive pads on top, each of which can trigger its own sound, and a jack socket on the rear panel can accommodate an extra pad or a foot pedal. The user can tune any onboard samples and add reverb, and a MIDI output lets you use the SamplePad to trigger external instruments, including computer-hosted hosted soft synths. Audio from the SamplePad outputs from either the left and right jack sockets or a headphone socket. Available from the beginning of 2012, the SamplePad is expected to sell for around $199. Alesis +1 401 658 5760



January 2012 / w w w . s o u n d o n s o u n d . c o m


W W W . S O U N D O N S O U N D . C O M / N E W S

Yamaha update 01V96
Digital desk and USB interface combined


amaha’s 01V96 digital console has just been given a major update. The new version is called the 01V96i, and it promises improved sonics as well as a range of advanced new features. Its headphone amps have been upgraded, and the 01V96i now comes with Yamaha’s VCM (Virtual Circuit Modelling) plug-ins as standard. Probably the most significant new feature, however, is the inclusion of a USB2 audio interface. Using this, it will be possible to record up to 16 channels of audio to a connected computer, eliminating the need for bulky looms and external interfaces when recording concerts. The interface also works in the other direction, allowing you to play back 16 channels from a computer. This means you can use the 01V96i’s effects when mixing from a computer, or even perform ‘virtual’ soundchecks without the artist needing to be present. Best of all, though, is the fact that the 01V96i will cost exactly the same as the previous version ($2699). It’s expected to be available from the end of December. Yamaha +1 714 522 9011

Respecting boundaries
Maier’s boundary mic converter


oundary mics are perhaps under-represented in the pages of SOS, though they have their uses in music recording, with many high-profile engineers employing them routinely. Perhaps one of the reasons why they don’t grace these pages as much as they might is that there aren’t many high-quality models available, although it is possible to use a ‘normal’ pressure-operated mic (such as an omnidirectional condenser) in a similar way, simply by laying it on the floor. There, it will be in the area where the greatest fluctuations in air pressure occur, meaning it will be extra-responsive, significantly raising the level of wanted sound, as well as eliminating unwanted reflections from the floor. Unfortunately, the above means putting one of your prized mics on the floor for all and sundry to step on and trip up over, but German company Maier Sound Design have come up with a neat

solution to this problem. The Turtle is described by its makers as a ‘boundary converter’, and it incorporates both a shockmount and a protective shield. The shockmount uses Rycote’s innovative ‘lyre’ system, to hold mics securely while minimising mechanical coupling between them and the floor. The shield, meanwhile, prevents the mic from being stepped on! Maier’s Turtle is made to accommodate pencil-style mics with a diameter between 19 and 21mm, and it ships in a protective wooden box. You can get them in unobtrusive black, or in red or yellow, if you want them to stand out. They’re available to buy from Circle Sound, at a price of £237.60 each . Circle Sound +44 (0)1869 240051


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ORCHESTRAL ESSENTIALS holds the carefully selected essentials from the complete ProjectSAM catalog, including the Symphobia series, True Strike series and Orchestral Brass Classic.


€ 349 / $449

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Cloudlifter Z
Back in November, we reviewed the Cloudlifter in-line mic preamps from Arizonian company Cloud Microphones ( sos/nov11/articles/cloudlifter.htm). These simple devices use phantom power to provide 20dB of gain to low-output mics (such as passive ribbons), and we found them to be highly effective at negating the shortcomings of low-end preamps. Now the company have released the Cloudlifter Z, another compact box that performs a similar function, while adding a few useful extra features. The Cloudlifter Z offers two different gain settings: +25dB and +12dB, selected via a toggle switch. It also has a switchable high-pass filter and, usefully, a range of seven different input impedances (150Ω, 350Ω, 1.5kΩ, 3kΩ, 7kΩ, 11kΩ and 15kΩ). The latter feature is a thoughtful inclusion, given that the product is likely to be of most use with low-output dynamic and ribbon mics, which tend to be particularly sensitive to preamp impedances. Expected to be available later this month, the Cloudlifter Z will be selling for around $299. For more information, visit

Zplane’s PPMulator metering plug-in, which we reviewed in November ( articles/ppmulator.htm), has been updated to version 3.0.8. The update brings with it compatibility with 64-bit hosts, increased maximum duration (to 24 hours) for loudness measuring, and an extra fall-back option for all metering modes. The update is free to current owners of PPMulator+ and PPMulator XL. Ueberschall have expanded their Elastik-compatible sample packs with the addition of Funk & Soul. Featuring guitars, drum loops and bass lines, among other instruments, it is said to “keep the authentic feeling of funk and soul music”. It contains 2.6GB of audio (over 1200 individual samples), and costs just €149. Washington-based mic manufacturers Cascade have announced a new ribbon

model, called the Knuckle Head. Its basket assembly (which houses the ribbon) is suspended in a metal hoop by eight springs, giving it an attractive vintage look. Indeed, many of its attributes are pretty classic for a ribbon mic: it’s passive, and so doesn’t require phantom power; and the ribbon is suspended exactly halfway between the front and back grilles, so it should sound identical on both sides (Cascade point out that this also makes it suitable for use as a Mid mic in a Mid/Side array). The Knuckle Head has a frequency response of 30Hz to 18kHz (±3dB), and a surprisingly high SPL-handling capability of 165dB (at 1kHz). It ships in an aluminium case with a mic clip and a cleaning cloth, and carries an appealing US price of $225. When we reviewed the Sontronics DM mic series last month, one of our only criticisms was that the included mic clips could sometimes feel a little loose. Happily, Sontronics have replaced these clips with a newer, sturdier type. Even better, they’re offering to replace

the older type of clip free of charge, to customers who bought any of the DM mics with the older clips after June 2011. For more information, check out the Sontronics web site. The Earthworks SR40V high-end stage vocal microphone now ships in a rugged carrying case, it has been announced. When we reviewed the mic last month, we noted that the supplied wooden box was perhaps not best suited to life on the road, but the new case (pictured below) looks as if it’ll withstand the rigours of gigging very well.


January 2012 / w w w . s o u n d o n s o u n d . c o m


Off The Record
Economic uncertainty and the rise of digital distribution have taken their toll on many major studios, but there is still a future for them.

Music & Recording Industry News

here is an old elementary school joke, in which a person, when asked why he continues to hit himself in the head with a hammer, replies “because it feels so good when I stop.” Proprietors of the large, multi-room recording facilities that once dominated the music-production business are feeling some relief too, lately, if only because the relentless hammering that they have had to endure for the last decade has finally subsided a little. And now that the smoke has cleared, some of them are looking forward to going to work in the morning again. The reasons for so many large music studios shutting down in the last decade are both economic and cultural. The shift to digital distribution of music, legitimate and otherwise, damaged the traditional revenue sources for those studios (the major record labels). Then that same technology enabled many record makers to take their work into their own studios, while a shift towards an ‘indie’ culture also made it cool to do so. The knock-on effects were as predictable as they were devastating to this class of facility, causing several dozen studios to drop off the radar during the last decade. But that same Malthusian dynamic is also responsible for an increase in work for the survivors. “It’s simple supply and demand,” says Ellis Sorkin, owner of Studio Referral Service, a studio-booking agency in Los Angeles. “There’s less work that can afford to go into large studios, but now there are even fewer rooms out there for it to go to. That’s driving demand up a bit. Not sure if I’d call it a comeback, but the survivors are getting more of what work there is.” The roll call of losses is lengthy and notable. New York lost facilities including


Clinton Studios, Chung King, Sorcerer, Battery Sound and Sony Music Studios, while in Los Angeles, O’Henry and Cherokee also expired. Sorkin estimates that New York City had the largest losses — as much as 40 percent more than LA, due mainly to Manhattan’s higher real-estate costs, which also fuelled the exodus across the East River to Brooklyn in that same time frame. Of the large multi-room facilities that managed to survive the decade, many of them had hip-hop’s blingy culture to thank for their survival, with producer teams like the Neptunes routinely booking out entire recording complexes in New York and LA, in as much of an advertisement of their commercial prowess as the fleets of Lincoln Navigators that filled the studios’ parking lots. Even that trend has tailed off somewhat for the big rooms, as record

“The travails of the past decade have proved that there is still a place for the big ‘Battlestar Galactica’ studio...”
budgets continue to contract. However, macro-economic dynamics have brought some relief, too. Consolidation among the labels — Warner’s and EMI’s music groups were both sold last year — has reduced overhead costs, and those labels can more easily find the Dr Luke-level money they need to keep the likes of Lady Gaga, Katy Perry and other slickly packaged, high-sales machines running smoothly. And those are the cohorts of artists and producers that are also helping to stabilise the business a bit for the large studios. David Amlen’s MSR Studios in Manhattan are a testament to consolidation: his three-room Sound On Sound facility had merged with the six-room Right Track in the middle of the

last decade to form Legacy Studios, until he settled out with the remaining partners last year and renamed it. Amlen says that even though rates and revenues are not rising, stabilisation is a welcome visitor after a combination of digital destruction and Wall Street mayhem. “There’s still a desire on the part of major artists and producers to use big studio facilities in New York and LA,” he says. “These are the only places that can offer the infrastructure that they want. The big question remains, though: how are they going to pay for it?” Jeff Greenberg, president of Village Recorders in Los Angeles, says that the idea of the multi-room facility as a dying breed is a fallacy. “We provide services like surround mixing, and, more importantly, we’re a place where professionals can be with other professionals. That’s something that got lost when music recording went into people’s homes.” If it’s true that what doesn’t kill you makes you stronger, then the travails of the past decade have proved that there is still a place for the big ‘Battlestar Galactica’ studio in modern record production. But this period underscored the need for fewer of them than in the music industry’s pre-digital heyday, and although some new members of the class — such as Germano Studios in New York — have arrived, the net head-count is still lower than it was. In the current economy, no one’s talking about green shoots any more — we read the tea leaves instead, in search of ‘the new normal,’ whatever that may be. But the large-facility studio model, battered though it may be, is proving that its combination of technological diversity, industrial sophistication and high-level service can remain a key part of music making in an uncertain economic future.


January 2012 / w w w . s o u n d o n s o u n d . c o m

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SE Electronics Munro Egg 150
Monitoring System


January 2012 / w w w . s o u n d o n s o u n d . c o m

SE Electronics have enlisted the help of renowned acoustician Andy Munro to design these striking studio monitors. Does their unique approach to speaker design pay off in the real world?

n an ever-changing world, the one thing we can probably always rely on is the fact that, at some point, the ‘music’ (in whatever electrical form it might be) has to be turned into acoustic sound waves for us to hear — and in most cases that means some form of monitoring loudspeaker. The Sound On Sound Monitors & Headphones Smart Guide catalogues 185 small and medium-sized monitors, and there are probably even more currently available on the global market, each with different strengths and weaknesses.


SE Electronics Munro Egg 150 $2595
• Distinctive styling with very practical and audible benefits. • Very accurate, neutral, revealing and precise sound character. • Remarkably extended LF. • Deep, wide and stable stereo imaging. • Astonishing volume available. • Bright ‘aiming’ LED on baffle to align the monitors — with an ‘off ’ switch! • Limited room-tuning EQ controls. • Front-panel mid-range EQ modes. • Built-in source selection and volume controls. • Market-leading warranty and auditioning services.

• Having to run speaker cables back to a central amplifier chassis may seem like a backward step. • The base doesn’t allow the Eggs to be tilted upwards.

An innovative and very distinctive monitor speaker, applying scientific principles to audible effect. The Egg 150 delivers a remarkably high standard of sound quality in a compact space with no significant vices.

And that’s an important point: none could be said to be ‘perfect’, and it’s very hard even to point at a high-end professional monitor that could approach true perfection in every respect! The plain fact is that loudspeaker monitoring remains the weakest link in the audio chain by a considerable margin, producing far more distortion and unwanted response irregularities than anything else. Although it’s true to say that small and incremental advances are still being made, fundamental loudspeaker science has barely changed in well over 50 years. The differences between the countless monitor speakers basically come down to slightly different design compromises and priorities, with the end users choosing one model over another largely on the basis of personal preference rather than technical achievement. Amplifier technology is mature, and even low-cost systems can deliver extremely good quality. Loudspeaker drive units, too, have reached something of a quality plateau: yes, a bigger Cabinet shape and frequency response: These budget buys a fractionally more capable graphs demonstrate the effect a loudspeaker’s driver, but even budget units perform cabinet shape has on its frequency response. acceptably. However, the most influential aspect of a loudspeaker design is, arguably, the cabinet: the big wooden ‘domestic manager’ can place a flower vase box that holds everything together. and a photo-frame on the top to make it Although constrained by the size and look less industrial, and in a studio we often budget restrictions imposed by the intended place all manner of technical studio debris market, cabinet design plays a huge role in on top! This might be very convenient, but determining the overall sound quality and is not necessarily the best way of building character of the loudspeaker. a loudspeaker cabinet. There are several different cabinet The acoustic effects of different shapes operating principles available to of loudspeaker cabinets have been known a loudspeaker designer, such as sealed about empirically since at least the early cabinets, vented or ported cabinets (with 1940s, but it was really the academic work the option of passive radiators instead of HF Olson that properly documented of open ports), and the so-called (but what was going on, in a paper he published not really in the true engineering sense) in the Journal of the AES in 1969. This ‘transmission-line’ cabinets. Each approach work revealed very clearly that cubic and has different strengths and weaknesses, and rectangular cabinets had a very damaging each manufacturer tries to optimise those effect on the overall frequency response, in creating a well-balanced final product, whereas cabinets with rounded or deeply albeit with varying degrees of success! angled front-baffle edges performed One thing that almost all cabinet designs considerably better. A spherical cabinet share, though, is that they are almost all delivered an almost perfect frequency rectangular cuboid in shape... response. (The ‘Cabinet shape and frequency response’ diagram shows the Thinking Outside The Box frequency responses of various different Rectangular boxes are relatively easy to cabinet shapes.) construct, relatively efficient in terms of The physics of the situation is essentially enclosed volume, and relatively easy to that the sound wave generated by live with. If you place a rectangular box a loudspeaker driver radiates outwards in on a flat surface, it won’t fall over or roll a hemispherical wave, travelling sideways away, for example! In a hi-fi application, the across the baffle surface and out into the

w w w . s o u n d o n s o u n d . c o m / January 2012



room. However, when the sound waves reach the baffle edge of a cuboid cabinet, they encounter a pressure discontinuity. There is nothing for the sound waves to press against any more, and that step change causes severe diffraction. In effect, the sharp cabinet edge forms a secondary source of sound-wave radiation, and sound waves from that ‘virtual’ source interfere with those from the loudspeaker driver itself, resulting in comb filtering, directional beaming and an uneven response. The precise frequencies affected and the strength of the interference effects depend on the relative distances between the driver and the various baffle edges. Not surprisingly, Olson’s work revealed that chamfering or rounding the front baffle edges helps to reduce these interference effects by softening the transition and severity of the pressure discontinuity at the cabinet edge — and that’s why most modern loudspeaker cabinets have rounded edges to varying degrees. But the best performance was obtained with a spherical cabinet, since there are obviously absolutely no hard edges, and thus no step-change discontinuities. However, a spherical cabinet presents other practical problems, not least being how to stop the speaker from rolling off the console meter-bridge! On a more serious note, a sphere has only one dimension and thus has a very strong resonant frequency. A better compromise, combining the soft baffle edges of a sphere but with a broad spread of internal resonant frequencies, is the ovoid or egg shape. And that’s where SE’s new monitors enter the picture.

Inside the Egg: The Egg’s ‘Monocoque’ construction. The internal bracing ribs help to control cabinet resonances, while adding to the speaker’s strength without significantly increasing its weight.

Hatching The Egg
If the concept of spherical and egg-shaped loudspeaker cabinets has been around for 40 years or more, why has no-one done anything about it until now? Well, one reason is that people are used to rectangular cabinets, and another is that it is quite difficult to make an egg-shaped

cabinet in a commercially viable way. It’s not practical to use wood, and while a metal casting is possible, it is also quite expensive — although the current Genelec range has gone some way down this road. SE’s approach has been to use a heavily engineered plastic cabinet, which, although expensive to develop, is relatively cost-effective to build in quantity. And so we welcome the SE Munro Egg 150 monitoring system, which is the first in a planned series of related monitoring products. As the title suggests, the development of this innovative design has been guided by the highly regarded and enormously experienced acoustician, Andy Munro, along with SE’s James Ishmaev-Young and Siwei Zou. The fundamental engineering concept of the SE Egg monitor is to replace the familiar rectangular cuboid cabinet with a far more strongly curved, egg-like enclosure, with the aim of virtually eliminating both edge diffraction on the outside and strong resonant effects on the inside. It’s a beautifully simple and attractive idea but, as is always the case, turning the concept into engineering reality is far from trivial and it has taken the team over two years to perfect.

Egg Boxes
The Egg 150 Monitoring System is exactly that: a fully integrated, active, two-way monitoring system, with everything packaged in one enormous box for shipping. Indeed, the product’s marketing

tag-line is ‘AIMS’, which stands for ‘Active Integrated Monitoring System’. Inside the outer shipping carton are several more separate boxes, containing two egg-shaped loudspeakers (each weighing about 5kg), two base plates (1.5kg each), a power amplifier and control unit (another 8kg), and the associated three-metre speaker cables, terminated in Neutrik Speakon connectors. The Egg speaker cabinets have an attractive and very tactile matte-black, rubberised surface — the same as is used on many of SE’s high-end microphones — and the Egg dimensions are 465 x 289 x 258 mm (HxWxD) when mounted on the supplied base plate. The base-plate area is slightly smaller than the cabinet, measuring 245mm deep by 220mm wide, and the design is such that the Egg cabinets can be tilted downwards over a useful range, but not upwards. Apparently, SE decided that their speakers would always be mounted on console meter-bridges or tall speaker stands behind a work surface, and so only a downward tilt option would be required. I suspect that there is also a balance issue here, and that to enable a tilt-up action, the base plate would have had to project behind the speaker itself. The very unusual and visually most distinctive Egg cabinets have an internal volume of 14 litres and are ported, with the vent tuned to 51Hz and firing downwards at the front below the bass driver. The cabinet is actually formed from a very strong and acoustically inert type of plastic, moulded in three main sections and screwed together to form a unique ‘monocoque shell’ construction (see ‘Inside the Egg’ picture). The final mould design was arrived at after a lot of complex mathematical modelling

The 2U-high amplifier has two pairs of inputs, each with its own volume control, plus a headphone socket and a three-way switch for selecting between Soft, Hard and normal mid-range tonalities.


January 2012 / w w w . s o u n d o n s o u n d . c o m

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and exhaustive testing to achieve the ideal chassis thickness and balance. Perhaps the most critical aspect of the design — and something that apparently took a long time to fully optimise — is the use of internal bracing ribs as part of the moulding. These establish the overall shape, control any shell resonances, and give it remarkable strength for relatively little weight. A very useful facility included in the Egg 150 speakers (something I first came across on an M&K monitor speaker several years ago, but which few other manufacturers have copied) is a deeply recessed ‘aiming’ LED on the front baffle. Since the LED is sunk deeply into the baffle it can only be seen when directly on axis — both vertically and horizontally — and it therefore provides a very precise means of aligning the speakers to the intended sweet spot. However, who wants to have their retinas burned out by searing blue LEDs while mixing? Thankfully, SE have recognised this and provided an off switch on the main amplifier unit, so that once the system has been installed the lights can be extinguished. Of course, loudspeaker drive units are generally built to be mounted on flat baffles, and as a result SE have had to flatten one side of the Egg cabinet to accommodate two traditional drive units. This inevitably compromises the edge-diffraction performance slightly, because it distorts the egg shape, but the junction between the main Egg cabinet and the flattened baffle surface is as smoothly rounded as possible, and the effect seems negligible. Both drive units are made by Monacor, and are previously discontinued models now revised to SE’s own specifications. The bass driver is a 165mm polypropylene unit, rated conservatively at 50W (RMS), while the tweeter is a 25mm soft-dome unit with a neodymium magnet, rated at 40W (RMS). In effect, each physical loudspeaker cabinet is a passive box, with the two drive units wired individually back to the central system amplifier unit via an integral

Try Before You Buy
SE Electronics have become well known for their policy on allowing potential customers to borrow their mid-priced and high-end microphones on a try-before-you-buy scheme, as well as on their no-quibbles warranty policy. The same facilities are being extended to the Egg 150 monitors, providing enormous peace of mind both to potential purchasers who might not be able to find a retailer with a set for auditioning, and for existing users who might not have additional monitors available should the Eggs fail at any point in the future. Essentially, SE are providing a five-year manufacturing defect warranty, together with a three-year ‘no-downtime’ repair warranty, and a free auditioning loan service. There’s some small-print, of course, and the full terms and conditions are on the company’s web site, but it is a genuine and credible warranty. The way the loan system works is that if you are interested in the Egg 150s, you can contact your local SE distributor and they will ship a system to you to try, in the familiar comfort of your own working environment. If you like the system, you can buy it, and if you don’t, you can ship it back — all at SE’s expense. No risk, no pressure and no hassle... But I’d be surprised if you wanted to send them back! The ‘no-downtime’ warranty works in a similar way. Should the Egg 150 system fail because of a manufacturing defect, SE will ship a complete new system to stand in for your own system, while the latter is returned to SE for repair. Once fixed, it is returned and the loan system reclaimed. Again, minimal downtime and minimal hassle. Of course, the reason the entire system has to be shipped back — both speakers and the amplifier unit — is because the amplifier channels are finely matched to the drivers, and any repair or replacement will require complete realignment of the whole system to maintain the original factory specifications and tolerances. Clearly, the free customer-audition service and the free repair-loan service are expensive things to provide — both in terms of the service inventory and the courier costs. Few, if any, other manufacturers offer anything similar for this market sector, and this is a strong statement on SE’s part of the belief they have in the quality of their products.

Speakon connector. However, it should be pointed out that the Speakon wiring is non-standard, so don’t try running the Eggs from any other generic Speakon-equipped power amplifier!

Control Unit
Although the Egg 150 monitoring system is fully active, the amplifiers aren’t physically integrated into the speaker cabinets, as is the case with most small and medium-sized active speakers. Such an approach wouldn’t have been practical because of the curvaceous cabinet shape, and the impact it would have had on the internal volume. One option might have been to build the amplifier chassis into the speaker base in some way, but instead, SE have chosen to house all the electronics in a separate, rackmountable unit (removable rack ears are included). This 2U amplifier unit measures 88 x 420 x 300mm (HxWxD), and each of the four drive units is powered from its own

Connectivity is provided by XLR, RCA and Speakon sockets for the main input, auxiliary input and speaker outputs, respectively. Trim controls for the HF and LF drivers on each speaker are accessible, and the blue ‘aiming’ LEDs on the Eggs can be switched off once the speakers are correctly positioned.

50W amplifier, with signals derived from an analogue crossover stage that splits the audio spectrum at 2.1kHz. Some potential purchasers might be put off by the idea of returning to the old ways of chunky speaker cables and central power amps, but I suspect that this configuration won’t make any practical difference to most. The amplifiers are ‘chip amps’ running on ±35V rails, but appear to be of high quality with high slew rates and ultra-low distortion. The internal construction places the amp chips right next to both the linear power supply’s outputs and the rear-panel Speakon connectors. In effect, the amps modulate the power passing from the PSU to the speakers with the shortest possible connections to ensure the most precise control. It’s a classic design approach and clearly works well here. As part of the factory quality-control testing process, all four amplifier channel gains are matched to their respective speaker drive units, to tolerances of ±0.25dB — which is extremely tight. This is why the entire system is shipped as a single package, and why the apparently identical Egg speakers are clearly labelled specifically for left or right connection to the amplifier unit. The amplifier unit is rather more than just a set of power amps and a crossover in a shiny box. SE have built in some basic monitor-control facilities as well, although I’m slightly disappointed that they didn’t take this further — but perhaps the company have chosen to restrict the


January 2012 / w w w . s o u n d o n s o u n d . c o m


facilities provided here so that the promised larger Egg system can include more comprehensive facilities for the inevitably larger price tag. Nevertheless, the amplifier unit features two small knobs and two large knobs, together with a lovely, blue-lit power on/off button. The small right-hand control switches between the main and auxiliary inputs (main being connected via XLRs and auxiliary via RCA phono sockets, all on the rear panel). These two input sources have independent stereo level controls, which are the two large knobs towards the outer edges of the front panel. This arrangement makes it very easy to level-match the two sources: a great help when comparing the mix from a DAW with a reference track from a CD player, for example. The fourth knob is a mid-band equaliser, with three options. Originally it was intended to emulate the Yamaha NS10’s peaky response, but during the final pre-production auditions it was decided that the EQ was too harsh and not really as useful as hoped. So, after some further tweaking, the mid-range EQ options are more subtle than originally intended, but actually all the more useful for that. The control’s centre position leaves the system’s frequency response as flat as Andy Munro designed it to be. Rotating the knob left or right selects either ‘Soft’ or ‘Hard’ modes, in which the mid-range response is reduced or raised by about 1.5dB, respectively. The idea of these options is essentially to provide either a more ‘easy listening’, hi-fi-style mode with the classic ‘smile’ response curve, or to provide a ‘shouty’, mid-forward and strongly detailed character that exposes the critical mid-range region of a mix. A pair of red LEDs on the front panel warns when the amplifiers are close to clipping. It seems a missed opportunity not to have included ‘mono’ and ‘dim’ buttons, and the lack of any scale around the volume controls makes accurate setting and resetting of the listening volume more difficult than it should be. But these are minor niggles in the grand scheme of things. On the rear panel, alongside the IEC mains inlet, the two sets of input connectors, the Speakon output sockets, and the

aiming-LED off switch, are four level-trim potentiometers for the two HF and two LF outputs. These are provided to enable the system’s response to be tuned to the room, if necessary, with the ability to reduce the bass in compensation for boundary proximity, and to boost or cut the treble as necessary to match the room’s acoustics. As I mentioned previously, the Egg 150 system employs standard four-pole Speakon connectors for the speaker cable, but only three of the four internal wires carry the audio signals. The HF and LF amplifier outputs share a common return wire, leaving the fourth wire to carry the power for the switchable aiming LEDs

Graphic Detail
In the online version of this article ( articles/seegg150.htm), you’ll find three frequency-response graphs for the Munro Egg 150 speakers. The graphs show the on- and off-axis frequency response for a single speaker; the range available on the HF, LF and Mid trim controls; and the responses of the left and right speakers in a pair compared.

Listening Eggsperiences
Having unpacked the enormous Egg 150 system carton, and all the smaller inner cartons, I placed the two Egg speakers in

The stand built in to the Egg speakers allows them to be angled downwards by a maximum of 15 degrees.

I’ve not come across any other egg-shaped speakers, although the current Genelec range does go quite some way in a similar direction. I can’t think of any other active monitors in this market sector that include a basic monitor controller either.

their allotted positions on a pair of Zaor height-adjustable speaker stands, well clear of side and rear walls, with the amplifier unit on the work surface in front of them. I hooked an HHB UDP89 multi-format disc player directly to the amplifier chassis ‘main’ inputs as a reference source, and started to work through my usual collection of reference CDs. The first thing I noticed on firing up the speakers was just how big they sounded. For such modestly sized units, the bass response is extraordinary, in terms of both the low-frequency extension and the speed

and dynamics of bass instruments. There seems to be virtually no port resonance and no ‘hangover’: bass notes start and stop extremely cleanly and quickly. If only all ported cabinets could achieve as much! Andy Munro suggests that the egg-shaped cabinet has a complete absence of strong internal resonances, and that plays a big part in helping the port output to integrate almost perfectly with that of the LF driver. The result is minimal time smearing and an excellent transient response — and both are very audible. The next thing I noticed (accidentally) was just how loud these monitors can go. Clearly, the drive units are quite efficient and the power amps are conservatively rated, but even the heaviest rocker won’t find anything to complain about in terms of volume here! The Egg 150 monitors are intended as midfields, but their compact size makes them usable as nearfields as well, while their power handling would probably enable them to serve as main monitors in moderately sized rooms! And even when the red warning LEDs start to flash, there’s no obvious distortion or compression to degrade the performance. I found the stereo imaging to be very precise and completely stable, with a strong centre image and a superb impression of depth, as well as spaciousness, on well-recorded material. The subtle room tones and reverberation of old jazz recordings was very audible — something that lesser speakers fail to extract — and the overall tonal balance was spot-on, to my ears. The Eggs exhibited excellent and seamless integration, from the surprisingly deep lows, right through the mid-range, and on to the high end, and I didn’t feel any need to tweak the balance at all. These initial impressions were obtained with the front panel mid-range EQ control in the ‘off’ position, but turning it to the ‘Hard’ position brought mid-range instruments and voices forward quite dramatically, adding a certain impact and urgency to the sound,


January 2012 / w w w . s o u n d o n s o u n d . c o m

and making subtle level differences a little more obvious and demanding of attention. Conversely, the ‘Soft’ mode instilled a far more laid-back effect, which was much easier on the ear, to the point of blandness. Of course, such a facility could be a dangerous thing in a studio monitoring situation if used unthinkingly, but it’s handy if you just want to kick back and enjoy your music on a spiritual level, instead of analysing it on an intellectual one! The raison d’être of every true monitor speaker, of course, is to reveal and expose information about the individual sources and the way they interact when mixed together. Monitor speakers aren’t supposed to sound ‘nice’ — they are supposed to reveal technical and aesthetic flaws (when they’re there), and I have to say that the Egg monitors do a pretty good job of that. The bass is fast and tight, revealing the true character of any kick-drum EQ, as well as the timing relationship between the bass and kick. When the kick drum is thumped, you hear just the thump, and not the extended ‘boom’ that so many lesser speakers produce in the hope of appearing more powerful and impressive than they really are! Such tricks don’t help when trying to fine-tune a mix. The critical mid-range region is crystal clear, and can be made even more revealing and insistent by using the ‘Hard’ mode, if required, and although my delicately rounded BBC ears preferred the ‘off’ option most of the time, even I would admit that it is a useful facility to have! The high end is open and spacious, without any edge or grittiness, and with extended listening I found little evidence of fatigue, which indicates very low distortion levels — something that is also supported by the fact that I found myself listening at far higher levels than I thought on several occasions.

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The Eggs Factor
Overall, the Egg monitors delivered far more than I was expecting. The bandwidth, especially at the bass end, was far greater than a monitor of this size would normally deliver, but without any hint of over-inflated port resonances to bolster the performance. Indeed, bass clarity, precision and speed are some of the strengths of this unique design. The system is also capable of much more volume than any sane user will need, and the clarity and ability to hear into a mix is excellent. There is absolutely no doubt that these are very good monitor speakers indeed, and certainly worthy of the title. The design takes an age-old idea and implements it extremely well, to reach the promised gold at the end of the rainbow. To some, the list price of these monitors might seem high, but the Egg 150 monitoring system stands direct comparison with its peers extremely well, and might even embarrass some! But if the asking price is a little more than your current budget can stand, I’d advise being patient, as the smaller Egg 100 monitors, which appear to share the same attributes, are in the advanced stages of production and should be revealed early next year. In the meantime, I’d urge those seriously $ $2595 per pair. contemplating T Fingerprint Audio a monitoring +1 512 847 5696. upgrade to take E SE up on their W free auditioning W option.

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w w w . s o u n d o n s o u n d . c o m / January 2012



UAD Ampex ATR102
Tape Simulation Plug-in For UAD2
Having already given us a virtual studio multitrack tape recorder, the boffins at Universal Audio turn their attention to mastering machines.

ntroduced to the market back in 1976, the two-channel Ampex ATR102 analogue tape recorder is a mastering classic and, after the Studer A800, is the second tape machine to be faithfully modelled by Universal Audio. However, comparing this plug-in with any given original would only tell part of the story, as ATR102s were often modified to run at different tape widths up to one inch, and may have been further modded to remove their audio transformers. Likewise, there are adjustments on this plug-in that go far beyond what the original hardware would have offered, including variable wow and flutter and tape hiss that would, at the higher settings, make even a cassette dictaphone look good. Amongst the ATR102’s attributes is its ability to run almost totally clean or to be pushed into very obvious distortion and, as with so many analogue machines, it has that mysterious ability to glue the elements of a mix together in a way that simple digital mixing never seems to manage.


Universal Audio Ampex ATR102 $349
• Extremely detailed model of the original machine. • Captures the essence of the tape sound very accurately. • Additional control range allows for the creation of special effects.

‘Flipping the lid’ gives you access to detailed calibration settings.

• I was hoping for a varispeed knob so I could mess around with tape flanging.

Once again, the UA team have shown their talent for meticulous modelling. This plug-in comes so close to the sound of a real high-end tape machine that any remaining differences are not going to affect your record sales!

UA’s model of the ATR102 electronic signal path includes the audio transformers, plus a choice of repro or sync head playback, along with multiple tape brands, tape speed, head configurations and calibration levels. What once were undesirable artifacts, such as hiss, hum, wow and flutter (which were extremely low on the original) and crosstalk between channels can be used creatively, and even

the input stage has been designed to reproduce the saturation characteristics of the original hardware circuitry when overdriven.

Controls on the plug-in enable the user to choose between input, sync and repro signal sources, four different tape speeds (3.75, 7.5, 15 and 30 ips), NAB, CCIR or


January 2012 / w w w . s o u n d o n s o u n d . c o m

AES EQ, the last being the only option at 30ips, and seven tape types, including the ubiquitous Ampex 456, of course. Note that at the two lower tape speeds, only quarter-inch tape is available. A nice touch is that when NAB EQ is selected, any mains hum you dial in will be at 60Hz, while CCIR gives 50Hz hum. At 30ips, where the EQ is automatically set to AES, you can still use the two EQ buttons to select the hum frequency you’d like! An Input Gain knob sets the record level in conjunction with the Cal(ibration) button settings, while the heads can be switched to quarter-inch, half-inch or one inch without ever having to go near an alignment tape. Biasing and calibration controls allow auto or manual settings and of course crosstalk, noise, hum and wow and flutter can all be tweaked to be orders of magnitude worse than on the original, for the creation of special effects. Most of these geeky adjustments are accessed by pressing the Open button on the main panel, which lets you get at the calibration, biasing and artifact setting knobs. Tape machines were generally set up to match the input signal level to a desired magnetic flux suited to the type of tape in use. Here, there’s an automatic calibration feature that matches the circuit settings to the tape and tape speed, offering four different settings, but the plug-in also provides manual calibration tools, including a tone generator, a distortion meter and a set of virtual alignment tapes — so even if you’ve never experienced the delights of aligning the real thing, you can practise here without fear of messing anything up. Those long winter evenings will simply fly by! An inbuilt tape-delay feature with direct and delay level adjustment also makes it easy to set up ADT (automatic double tracking) effects. Note that when the plug-in is used mono-in/stereo-out, the input signal is sent to both channels, which may then be adjusted separately. A Link button gangs all the controls between the two channels, with the left channel controls serving as the masters. By default, the output signal is sourced from the repro head, but you can switch to sync, which emulates the sound of playback via the record head and its electronics. This would normally only be used in the studio to keep overdubs in sync, because the frequency response isn’t as accurate as from the repro head, but it can be an interesting special effect. Input mode, meanwhile, gives you the sound of the machine’s electronics but without the tape — as you’d hear in real

The output of the plug-in can be monitored through the repro head, the sync head or just the input electronics.

life with the recorder parked in live monitoring mode. There’s also a mode labelled Thru, which is basically the same as bypassing the plug-in, except that in Thru mode the meters still function.

In Use
The easiest way to start using the plug-in is either to explore the factory presets, or choose a tape type, speed and EQ curve, then adjust the input gain to get as much level onto ‘tape’ as you like. The plug-in gain structure is set so that a -12dBFS input produces a 0dBFS level at the meters with the Repro level set against the red calibration marker. Separate controls are available for record and replay level, and

the meters can be switched to VU or peak reading. A table of the manufacturers’ calibration settings for each tape type is provided in the user manual, though you can switch to any of the four options if you prefer to experiment. Using the default ‘correctly set up’ machine at tape speeds of 15 or 30ips

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11/22/2011 4:01:18 31 PM w w w . s o u n d o n s o u n d . c o m / January 2012

U A D A M P E X AT R 1 0 2

and pushing the meters into the red but not banging them against the end stops produces the familiar sense of a more homogeneous mix, with subtly smoother highs and fatter lows; the amount of coloration is really very mild. If you want to exploit the ‘head bump’ to firm up the bass end, 15ips is the usual choice. As you’d expect, the lower the tape speed, the more obvious the tonal coloration, with 3.75ips sounding distinctly less pristine than the 15 or 30ips ‘mastered’ sound with which we’re more familiar. Similarly, the higher the record level above 0VU, the dirtier the sound gets, and each brand of tape has its own subtle character. As you increase the input level, the sound at first gets fatter and slightly compressed-sounding, then as you go further it makes the transition through grit to absolute filth! While I wouldn’t subject a mix to anything close to the filth setting, it might just work for sorting out that weedy snare-drum track.

While the obvious applications of this plug-in are for mastering and individual track or group processing, the extra control range allows plenty of scope for ‘abusive’ special effects such as highly distorted cassette-tape emulation with generous helpings of wow and flutter. The tape-echo function is also brilliant for creating traditional tape slapback delay — the maximum delay time available is one second, adjustable in one millisecond steps. Add a hint of wow and flutter to the delay and the result is pure magic. I was also keen to see if I could coax any tape-flanging effects from the unit; it turns out that you can, but only to a limited extent. If you use the tape delay setting with the delay time dialled to zero with equal levels of dry and delayed sound, then crank the Wow control way up and the Flutter to around one third, you get a very plausible mild tape-flange effect. I was hoping I could use the mouse to apply hand pressure to

Perhaps the closest rival is Waves’ Kramer Master Tape, which is modelled on a different Ampex machine, while the UAD range also includes the Studer A800 plug-in, although that, of course, emulates a multitrack tape deck. Alternatively, you might prefer to buy a used Revox or Tascam two-track machine and bounce your mixes to actual tape.

the tape reels to create speed variations the old-school way, but sadly there is no option to do this. Nor is there any varispeed, which would also have been perfect for controlling flanging. Unsurprisingly, the ATR102 is a fairly DSP-hungry beast: it isn’t available at all for the older UAD1 card, but on my quad-core UAD2 card at 44.1kHz, I could have 16 of the things whirring away and taking up 71 percent of the board’s capacity.

As with most UAD plug-ins, this one has been modelled to a degree of accuracy that some might call obsessive excess. It comes very close to the sound of a top-flight tape machine, but has enough user adjustment to make it sound as nasty as you need — just screw up the bias, add some wow and flutter and you have a charity-shop cassette! The alignment options could form the basis of a very usable educational tool, while the tape delay section recreates those warm tape slapbacks perfectly. The delay time even defaults to the correct value for the selected tape speed, although you can change it to whatever you like. There are also subtle tonal differences between the tape types; I usually ended up back with good old Ampex 456. At least you know you’re never going to suffer from ‘sticky shed’ syndrome (in which old tapes shed their oxide) when working with a plug-in! On a practical level, my mixes definitely sounded more integrated and a touch more ‘cuddly’ when processed via the higher tape speeds, with the 15ips head-bump adding some welcome low-end support. The more overdriven settings work best on individual mix tracks such as drums and electric guitar. If you’ve always dreamed of mastering to tape but could do without the cost and the hassle, the ATR102 plug-in for the UAD card is about as close as you’re going to get. $ T E W
$349. Universal Audio +1 877 698 2834.

The plug-in includes a number of ‘celebrity’ presets from well-known engineers and producers.


January 2012 / w w w . s o u n d o n s o u n d . c o m


Recognized with the coveted Technical GRAMMY Award® “for continual mastery and innovation,” JBL remains passionately focused on a single goal: Helping you produce mixes that shine. The LSR Series Studio Monitors are a clear example of this continued commitment to excellence. Blending innovations like JBL engineered transducers for superior sonic quality, Linear Spatial Reference design and Room Mode Correction for greater accuracy, even in acoustically less-than-perfect rooms. With three professional monitor lines, the flagship LSR6300 Series, the revolutionary LSR4300 Series, and our most affordable LSR2300 Series, JBL has a model within reach of any studio’s budget. No matter which system you choose, your mixes will hit their mark.

Find out more:
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From left to right: LSR4326P, LSR6328P, LSR2325P, MSC1 Monitor System Controller
© 2011 Harman International Industries, Incorporated


Yamaha MOX6

he MOX6 and MOX8 synthesizers are a subset of Yamaha’s popular Motif range, borrowing the majority of their design from the Motif XS. Although having no onboard sampling or audio recording facilities of their own, they are designed to serve as the heart of a compact audio/MIDI music production system. Simply connect one via USB to a DAW-equipped laptop, add a microphone and a pair of headphones, and you have a keyboard-based recording setup that’s light enough for one person to carry around. Both MOX models are of comparatively lightweight construction, so they’re far more portable than your typical synth workstation. The 61-note MOX6 on review here, at a mere 7kg, can be lifted easily with just one hand. The 88-note MOX8 weighs in at 14.8kg, which may prove to be more than just a handful for some, but is still remarkably light for a weighted action keyboard. True


Synthesizer Workstation
Yamaha’s MOX range offers unusually tight integration with your DAW. Is this the way forward for synthesizer workstations?
portability would, of course, mean the option of battery power, however, the MOX requires mains power for its external 12V PSU, so producing smash hits in a rowing boat is probably not an option. The internal synth architecture of the MOX is essentially the same as the Motif XS — the two instruments even share the same sample ROM and Preset Voices. There are changes and additions to some menus, and occasional operational variances, but since none of them represent a drastic departure from the overall functionality of a Motif XS, this review will concentrate on the more immediate differences. If you are unfamiliar with the XS, the review in the October 2007 issue of SOS explains its features in some detail.

Yamaha have kept the cost, size and weight of the MOX6 to a minimum in several ways. Most obviously, the construction is almost entirely of plastic. Despite this, it feels sturdy enough, without too much of the plasticky


January 2012 / w w w . s o u n d o n s o u n d . c o m

‘creaking’ feeling you might expect. The synth-action keyboard has a shortened front-to-back scale and no aftertouch, there is no ribbon controller, the monochrome LCD display has the same 240 x 60 pixel dimensions as that of the Motif ES, and the real-time controllers comprise only eight knobs and no sliders. Most of the controls from the display to the right-hand end of the panel will be familiar to XS users; the most significant changes are all on the left-hand end. The functions of the eight control knobs vary slightly from the XS: there are fewer arpeggio and EQ parameters, replaced by chorus and reverb preset selectors, portamento time and volume. The volume knob serves as a substitute for the missing level faders; Part levels can be adjusted here, but only one Part at a time. A pair

of long-overdue transpose buttons has also been included. The AF1 and AF2 buttons can now be controlled via a footswitch (hooray!), so you no longer have to remove a hand from the keyboard when playing voices that feature sound variations assigned to these buttons. The audio input gain knob has moved from the rear panel to the top, with its own on/off switch. New to the MOX are an eightsegment LED meter and a DAW level slider. The meter can be switched to display either the audio input level or the signal level at the MOX’s output (ie. the sum of the internal synth and incoming DAW audio, more on which later). The rear panel largely resembles that of the XS, but with one stereo output, one assignable foot-controller jack, and no S/PDIF output, Ethernet

The MOX6’s back panel includes a pair of USB ports, MIDI In, Out and Thru sockets and (all on quarter-inch jack sockets) three footswitch inputs, a pair of stereo outputs, a pair of stereo inputs and a headphone port.

YA M A H A M O X 6

connection, or mLAN expansion slot. The USB To Host socket makes the latter three connections redundant, since it handles all the audio and MIDI communication between MOX and computer.

Performance Creator
Performances consist of up to four parts, which can be layered and/or split, with independent arpeggiators available for each Part. Many MOX preset Performances include arpeggiators to provide rhythmical synth textures, automated bass parts and drum patterns, and can be useful sources of songwriting inspiration. Performance Creator is a new fast-track tool for making your own multi-layered Performances. Using any Voice as the starting point, pressing Layer invokes the Category Search window. Choose the desired sound category and Voice, press Enter, and that Voice is automatically layered with the first Voice, assigned to Part two. Adding a bass with a keyboard split to the third Part is just as easy: pressing Split invokes the Category Search window again, which now includes split point and upper/lower parameters. Choose whether you want the new Voice to be above or below the split point, and

If you’re looking for alternatives to the MOX purely as a stand-alone synth, the Motif XS and XF are obvious choices, as they share much of the same functionality and all their synthesis architecture. Beyond Yamaha, synths such as the Korg M50, Roland Fantom G-series and Kurzweil PC3 offer similar synth/sequencer packages. However, the MOX is currently unique in its compact approach to recording audio and MIDI combined with DAW control and VST instrument compatibility.

There are a few economic concessions to the MOX’s internal workings compared to the XS. The biggest cutbacks are the absence of any sampling facilities and the halving of the maximum polyphony to 64 notes. The number of Insert effects available in song and mixer (ie. multitimbral) modes has also been reduced from eight to three. While these limitations obviously restrict how much the MOX can do at once, the whole point of the MOX is that tracks can be rendered as audio direct to your DAW, and their MIDI tracks archived. Insert effects can be reassigned as and when instruments require them, and those instruments rendered to audio. It’s not all about cutbacks, though: 87 new arpeggios bring the total to 6720, the sequencer’s capacity has been almost doubled, to 226,000 notes, and the Favourites category has its own dedicated button for quick recall of frequently used Voices.

you’re done. Pressing Drum Assign adds a drum kit with an arpeggiated drum pattern automatically activated (drums are always given to Part four). If you want to change the drum pattern, just hit Arp Edit and select a style and pattern from the vast number available.

DAW Audio Recording
The USB To Host connection enables MIDI and audio data communication between the MOX and a computer. The MOX functions both as a synth

Software Support, Cubase & VST
As with previous Motifs, Yamaha’s Studio Connection concept brings full integration with its ‘Total Recall’ environment to the MOX, when used in conjunction with a Steinberg DAW such as Cubase. To get the full benefit, several pieces of software in addition to the Yamaha USB driver are required: firstly, Cubase or one of its stablemates (e.g Cubase AI5), MOX6/MOX8 Remote Tools, Studio Manager and MOX6/MOX8 Editor VST. All can be downloaded from www. Remote Tools comprises two items. The first is the MOX6/MOX8 Extensions. This enables Cubase to detect an MOX synth, enable the ASIO driver and configure all the I/O ports, and provides integration of the MOX as a remote controller for Cubase and installed VST instrument plug-ins. Secondly, the Remote Editor allows you to edit the existing synth remote-control templates and to create your own. Steinberg DAW users can create templates that include DAW and synth control, while non-Steinberg users can only create synth templates. Nevertheless, the ability for all users to customise the MOX to control a selection of their favourite VST instrument plug-ins is not to be sneezed at. Studio Manager acts as host and ‘organiser’ for a range of software editing programs for Studio Connections compatible devices — ie. most Yamaha equipment — and can be accessed from within Cubase. The MOX can be run as if it were a VST instrument within Cubase, and MOX6/MOX8 Editor VST is intended specifically for this application. The obvious advantage of running the MOX as a VST intrument is Total Recall — all its current settings are stored with a project. Any edits made on the MOX’s own panel update the on-screen editor accordingly, and vice versa. Without Cubase, it’s still possible to enjoy a pretty well-integrated MOX by installing Studio Manager and the non-VST version of the MOX6/MOX8 Editor. The Editor is accessed from within Studio Manager, both running as stand-alone programs alongside your DAW. It just means you have to manually start them up and manually save their settings, rather than having the ‘symbiotic’ VST integration offered by Cubase. Not wanting to miss a trick, Yamaha bundle Cubase AI5 with the MOX for free, together with free downloads of Steinberg’s Prologue synth and YC3B Hammond clone, so the joys of full integration can be experienced by all. Well, nearly all. AI5 must be registered online, even before you can use it as a time-limited demo, then activated to make its use permanent. However, registration and activation can only be done via the included e-licensing software, which requires the computer on which you’ve installed the software to be online — there is no facility for doing it from a different computer. Although it’s unlikely for a laptop not to be web enabled, there are still people out there who choose to keep their main studio computer safely isolated from the web. Surely it would be preferable to be offered alternative methods of activating software, so you can keep your studio setup web-free?

Yamaha MOX6 $1199
• Motif XS-style synthesis in a lightweight package. • Audio recording and playback using a computer connected via a single USB cable. • Full DAW and VSTi integration when used in tandem with a Steinberg DAW. • An impressive amount of synth and DAW control even with non-Steinberg software.

• To get the full complement of DAW and VSTi integration, you need to be running in tandem with a Steinberg DAW. • Portability reliant on having a mains supply available.

Despite its need for mains power, the MOX6 makes an ideal centrepiece for a portable recording setup. Although light enough to carry around in a gig bag with little effort, it’s still a serious Motif-style workstation keyboard in its own right. It should prove valuable not only as a main gigging and recording instrument, but also as a means of capturing instrumental and vocal ideas into a DAW-equipped laptop, without the need for additional audio or MIDI hardware.


January 2012 / w w w . s o u n d o n s o u n d . c o m

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YA M A H A M O X 6

The MOX Editor in Song (multitimbral) editing mode. In Cubase, the VST version of this editor is invoked simply by pressing the VSTi Window button on the MOX.

and a soundcard/audio interface, so no additional hardware is required. It’s capable of transmitting up to four simultaneous channels (as two stereo pairs) and receiving two channels (your DAW’s stereo output). Audio routing within the MOX is straightforward: there’s an option to transmit just the first audio pair, or both together. Generally, signals from the MOX’s audio input travel down the Stereo 1 pair, while synth sounds appear on Stereo 2, making it possible to record a vocal and the MOX’s performance together, onto separate tracks of the DAW. Each MOX sequencer Part can also be routed to either stereo pair, so by hard panning instruments you can record up to four mono parts at once to separate audio tracks. The MOX also acts as a remote control for your DAW, the level of functionality depending on which DAW program you’re using (see the ‘DAW Remote Control’ section elsewhere). And as you’d expect, the MOX speaks fluent MIDI, handling all 16 channels in both directions. To access

all these features, it’s necessary to install the latest Yamaha USB driver on your computer. Once installed, the MOX will appear in the DAW’s device lists: four MIDI inputs and outputs, two stereo audio inputs with mono L/R options for each stereo pair, and one stereo output. MIDI note data is handled by MIDI port 1, while DAW remote-control data is addressed over MIDI port 2. The Yamaha driver has no problem co-existing with other MIDI drivers, but you may well find it necessary to deactivate other audio drivers your DAW uses. This was the case for me in Sonar, where the Yamaha audio drivers were unavailable until my usual soundcard’s drivers had been

deactivated. After that, everything went swimmingly — audio could be recorded from both MOX audio ports, and Sonar’s audio output appeared in all its glory at the MOX’s stereo output jacks. Different audio routings and MIDI settings may be needed according to the task in hand, so the MOX provides instant access to six Quick Setups pre-configured to the

The Remote Editor allows editing of the MOX’s 50 control templates, and the creation of custom templates for controlling your own soft synths using the knobs on the MOX.


January 2012 / w w w . s o u n d o n s o u n d . c o m

most likely scenarios. Any of these can be customised to your own requirements and stored to any of the six Quick Setups, saving time and menu-surfing when you’re on a roll. A full description of the Quick Setup configurations can be found in the MOX support section at www.motifator. com/index.php/support/mox_articles. The MOX’s audio sample rate and bit depth, incidentally, is fixed at 44.1kHz, 24-bit. Tests with varying ASIO buffer sizes showed that 256 samples gives a fairly respectable 5.8ms latency for running other plug-in synths without clicks or drop-outs. The workstation can monitor its own audio directly with zero latency.

Whether using Cubase or not, anyone can create templates for controlling soft synths, using the Remote Editor. Here’s one I made earlier for M-Tron Pro.

for which I have the software, everything worked exactly as intended. Knobs 5 to 8 give Cubase users access to additional remote functions, the specifics of which depend on the selected template.

True portability implies something you could easily use on a bus or a plane, which clearly isn’t possible in the case of the MOX. However, the MOX6 is certainly light enough to carry around in a gig bag along with a laptop without too much hassle, making an ideal combo for hotel-room writing sessions, recording band rehearsals, or capturing moments of inspiration just about anywhere there’s a mains supply to hand. The MOX is, of course, a highly capable synth in its own right, equally at home as a principal part of an on-stage keyboard rig or in any recording situation — and one your roadie will appreciate, too. $ MOX6 $1199, MOX8 $1999. W

DAW Remote Control
The MOX can remotely control DAW functions, and includes remote-control templates for Cubase, Sonar, Logic Pro and Digital Performer. Users of non-Steinberg DAWs must manually configure their DAW’s control-surface settings to match, typically using the basic Mackie Control template. If you’re using Cubase with the MOX6/MOX8 Remote

Tools software installed, the MOX will automatically configure Cubase’s remote control setup for you. I set it up to work with Sonar, which responded correctly to the MOX’s transport control functions, with the exception of the ‘return to zero’ button and the AI data dial. The MOX also has 50 templates for controlling a range of popular soft synths via its control knobs. The functions of each knob vary according to the selected synth template, and cycle around three rows of parameters using the knob ‘function’ buttons. All eight knobs are operational if you’re using Cubase, but only knobs 1 to 4 will work with non-Steinberg DAWs. On trying this feature with a number of synth templates

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JZ Microphones

Vintage 11
Large-diaphragm Condenser Microphone

The other mics in JZ’s Vintage range are based on classic designs of the past — but they hope this one will become a classic in its own right...
he new Vintage 11 from Latvian company JZ Microphones is not a ‘reborn’ classic, nor is it an off-the-wall design like their Black Hole, which we reviewed back in 2008 (www. jzblackhole.htm). Rather, it is designer Juris Zarins’ attempt to create a new microphone that he believes deserves classic status alongside more familiar names. The number 11 comes from the year 2011, when this model was created, but despite their lofty claims, the Vintage 11 has a selling price of just $699 — admittedly, not a budget mic, but then certainly not what you’d expect to pay for a high-end studio mic either. The flattened shape of the mic body and the gold JZ logo, combined with


the glossy black-paint finish, is oddly reminiscent of something you might find on the shelves of a cosmetics store, but there’s no denying that the mic is very nicely put together: its weight and standard of finish inspire confidence. Apparently, the body of the Vintage 11 is made using both aluminium and plastic, with a two-layer, brass-mesh

“There’s no denying that the mic is very nicely put together: its weight and standard of finish inspire confidence.”
grille similar to that used for the Black Hole microphone. The capsule sits atop a post to keep it well away from the body metalwork. There is logic to the wide, flat basket design, which has only one vertical support in the cardioid capsule’s rear ‘dead zone’. The lack of vertical supports at the front and sides minimises reflections, which means that the basket has only a negligible effect on the capsule’s performance.

JZ Microphones Vintage 11 $699
• Smooth and classy sound. • Distinctive styling. • Low self-noise.

• No pad or low-cut switches, but at the price, that’s not a deal-breaker.

Aural Specs
Technically, this large-diaphragm cardioid condenser microphone is distinguished by exceptionally low self-noise (6.5dB, A-weighted) and a 20Hz to 20kHz frequency response. Its dual-backplate capsule is six microns thick, is edge terminated, and has

The V11 is a versatile mic that delivers a high-quality sound, albeit with that built-in bass bump that may need taming in some applications. For me, its smooth upper-mids, lack of obvious coloration and low noise are its strongest attributes.


January 2012 / w w w . s o u n d o n s o u n d . c o m

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© 2011 Avid Technology, Inc. All rights reserved. Product features, specifications, system requirements and availability are subject to change without notice. Avid, M-Audio, Mbox, Venom, and Pro Tools are trademarks or registered trademarks of Avid Technology, Inc. or its subsidiaries in the United States and/or other countries. All other trademarks contained herein are the property of their respective owners.

J Z M I C R O P H O N E S V I N TA G E 1 1

a diameter of 27mm. The onboard, transformerless electronics manage a sensitivity of 22 mV/Pa (1kHz into 1kΩ), and a maximum SPL, for 0.5 percent THD, of 134dB. This equates to a dynamic range of 128dB. Standard phantom power at 48 Volts is required for operation. There’s little information on the capsule itself, other than to say that it evolved from the company’s GDC67 capsule, and uses their ‘Gold Drop’ sputtering technique to apply the diaphragm coating but, again, there are no specific details of what this process entails. Gold-plated pins are used in the output XLR connector to minimise problems due to corrosion or oxidisation, and there are two further threaded holes in the base to accept the included stand mount, which has its pins mounted via rubber buffers to offer a degree of resilience. There are no pad or filter switches on the mic.

Typical frequency response without proximity effect: the HF roll-off above 15kHz and boost around 90Hz contribute to a smooth, warm sound. The proximity effect would further reinforce the bass end.

Test Time
I set up my usual array of instrument and spoken-word tests, but on my first playback the output of the Vintage 11 appeared to be in opposite polarity to the AT4050 mic I was using as a comparative reference. Was it a wrongly wired cable, was there a problem with my AT mic, or was the JZ the culprit? Switching the Vintage 11 for a different mic showed that the two signals were now in phase, so I have to conclude that the review Vintage 11 was wired opposite to the norm. My spoken-word vocal test revealed the Vintage 11 to have a smoother mid range and more pronounced lows than the AT4050, though both mics would benefit from some LF roll-off when used up close. There was no obvious character to the sound other than the bass lift, which could easily be dealt with if it wasn’t needed, so the overall result

was pretty natural and non-fatiguing, with no barking or honking in the upper mid-range. Both mics fared well on hand percussion and on acoustic guitar, with the Vintage 11 again sounding a little smoother than the AT4050, especially on the latter. I’ve always rated the AT4050 as a good guitar-amp mic — something that can’t be said of all large-diaphragm condenser mics. First up, with a raunchy and gritty overdrive sound, both mics captured the raw edge and attitude of

“The Vintage 11 sounds very natural — the highs are all present, but without clawing at your eardrums as they seem to do with some mics.”
the sound, with the Vintage 11 being a little more polite about the highs than the AT. Dropping the gain back to cleanish blues, both mics again scored highly, with the JZ sounding the smoother of the two. The AT4050 doesn’t have a completely flat frequency response, of course. Few large-diaphragm capacitor mics do. However, it is nominally flat to 6 or 7 kHz within a dB or so, with a fairly modest 4dB presence peak centred at around 10kHz. Because of this, you’d expect the AT4050 to have a very slightly more pronounced high end, which is reflected in our audio examples.

The AKG C214 and Neumann TLM102 are similarly priced cardioid condensers, although the former has both filter and pad switches, which the JZ lacks. The Wedge by Violet Design and Sontronics’ Saturn are both cardioid mics with distinctive looks that cost about the same as the Vintage 11.

at your eardrums, as they seem to do with some mics. The output responds well to EQ, which is helped by the very low self-noise, although a low-cut switch on the mic wouldn’t have gone amiss. Having said that, this is a very attractive price for such a well-specified microphone, so if you have to apply your low-cut filtering later in the recording chain, so be it. The same is true of the lack of a pad switch. I was surprised to find that the microphone’s polarity was inverted, but then it was received before the production versions of the mic went on sale. [JZ have confirmed that this won’t be an issue for customers: the internal wires on the review model weren’t colour-coded, but this has now been addressed.] To sum up, I think JZ have managed to give the Vintage 11 a classic character without making it sound exactly like anything else that has gone before, though whether it is indeed destined to join the revered classics, only time will tell. What I can say is that you get a seriously good microphone for the asking price, presented in a mercifully small, foam-lined wooden case along with the stand mount. Because the mic doesn’t have any obvious tonal traits other than the bass bump, it is more likely to suit a range of voice types than a mic with a very obvious tonal flavour.

Other than its deliberate bass lift, the Vintage 11 sounds very natural — the highs are all present, but without clawing

$ T E W

$699 including shipping. JZ Microphones +37 167 246 648.


January 2012 / w w w . s o u n d o n s o u n d . c o m


Bettermaker EQ230P
Stereo Analogue Equaliser


ettermaker are relatively new-on-the-scene manufacturers based in Poland, and currently there’s only one product in their portfolio: a high-end and quite innovative stereo equaliser called the EQ230P. In my book, quality is always more important than quantity (unless we’re talking about pension funds!), so we’ll overlook the limited product range and enjoy the fact that the EQ230P is a really interesting product that took, apparently, 18 months to develop. In essence, what we have here is a dual-channel/stereo equaliser featuring five individual equalisation stages, grouped into three distinct sections. Both channels are controlled through

With an all-analogue signal path and total recall courtesy of digital control, this EQ promises a lot. Does it deliver?
a single set of controls, shared between or allocated to each of the two channels as required. There’s nothing particularly unusual about a five-band stereo equaliser, of course — although the nature of the EQ here is a little unusual and attractive — but the really interesting aspect of the EQ230P is that all of these equaliser sections enjoy full recall facilities. As you’d expect, the recall memory side of things is digital, and to ensure complete isolation between the digital control/memory circuitry and the delicate analogue audio circuitry, the two sides have completely separate power-supply stages and all the digital control signals communicate to and interface with the analogue stages via optical interfaces. Digitally-controlled potentiometers are employed as the variable elements in the analogue filter signal paths!

The EQ230P is a very classy-looking unit and it feels well built. Internally, construction is generally to a high standard, although the circuit boards have the look of 1970s equipment, with relatively sparse PCB tracks and full-size components everywhere. The analogue circuitry is protected from the control and memory sections, and the power supply, by metal screening plates behind the front

Bettermaker EQ230P $4500
• Uniquely comprehensive combination of complementary EQ sections. • Intuitive user control interface. • Individually bypassable EQ sections. • Useful A/B comparison mode. • Instant recall.

• Slightly limited headroom margin. • This level of versatility, musicality and controllability comes at a high price.

The EQ230P is an unusually well equipped and versatile stereo equaliser, combining precise HPF and parametric sections with a musical Pultec-style section — a combination of facilities that make it a very special product. However, Bettermaker have gone further and included full instant recall facilities, which makes this a unique and phenomenally powerful device.


January 2012 / w w w . s o u n d o n s o u n d . c o m








• 3-DVD boxed set available at all good music stores • Downloads and streams direct from the website • Educational License at participating schools and colleges • Live Master Class Training Sessions all over the world

Narration by Billy Bob Thornton.
Distributed in the USA by Hal Leonard • Produced by KEYFAX NewMedia of Santa Cruz, California 1-800-752-2780


panel and dividing the unit front to back. There are real inductors in there, and lots of sealed relays for signal switching, too. The rear panel carries a standard IEC mains inlet (single voltage), with XLRs for the audio ins and outs, the outputs having two paralleled XLR connectors. A ground-lift switch is also provided to separate the analogue signal ground and chassis safety earth. Although not currently implemented, there are plans to include a USB interface mode, which would allow the EQ230P to be controlled from a DAW plug-in — and that really would be an attractive option. The control layout is logical enough, divided into three clear sections on the crisp, clean, white-on-black panel. Starting in the top left-hand corner, a large silver power-on push-button illuminates with a central red LED. An adjacent pair of clip LEDs warns if either channel is getting too hot, and monitors the levels between every stage to ensure optimum gain structures through the unit. It’s important to note, however, that the maximum acceptable balanced input signal to the EQ230P is +18dBu, which is about 6dB lower than most high-end analogue products. The maximum balanced output level is also restricted to +18dBu. In practice, this means you could run into trouble if feeding the Bettermaker from a high-end D-A outputting full level audio.

To the left of this power button are six smaller illuminated push-buttons, which engage the various individual filter sections (high-pass filter, two parametric stages, and a Pultec-style section), as well as offering an A/B comparison function and a hardwired global bypass mode. The high-pass filter is controlled with a single rotary to adjust the turnover frequency between 18 and 200Hz (see test plots on the SOS web site, as

The clean-cut front panel hosts a number of recallable digital controls for this analogue EQ.

“There’s nothing about the sound that I don’t like, and even the user interface works well.”
detailed in the ‘Technical Specifications’ box). A ring of ‘hidden until lit’ LEDs indicates the current turnover frequency of this simple 12dB/octave (second order) HPF design. When bypassed, the unit’s bandwidth extends down to about 3Hz (-3dB). Interestingly, all the rotary controls are speed-sensitive; slow turns provide high resolution, while quick spins traverse the control’s range much more quickly. This is very intuitive to use and makes operation both fast and precise. The centre-panel section accommodates the two parametric EQs,

Technical Specifications
The EQ230P has a very wide bandwidth which, with all EQ sections switched in but set flat, extends between 9Hz and 40kHz (at the -0.5dB points). The bandwidth is a little wider with the EQ sections bypassed. Crosstalk is also impressive, measuring over 70dB at 10kHz with everything switched in. The signal-to-noise ratio is comfortably over 100dB, and distortion is extremely low too, although my THD+N measurements came out fractionally higher than the published specs (0.007 percent instead of 0.004 percent — nothing to worry about there!). Power consumption is a modest 15 Watts. Should you want more details, you can find some Audio Precision Analyzer test plots on the SOS web site at articles/bettermakermedia.htm

each equipped with Level and Frequency rotary controls, plus increment and decrement push-buttons to adjust the bandwidth. Both sections can boost or attenuate using symmetrical constant-Q bell-curves over a ±15dB range, with bandwidths ranging from 0.2 to three octaves (Q range of 7 to 0.4). However, the two sections have different adjustable centre-frequency ranges, overlapping nicely, with 45Hz to 1kHz for the first parametric section and 650Hz to 15kHz for the second (test plots are available on the SOS web site). Again, the rotary control settings are indicated by white or red LEDs around their periphery. The third panel section is the ‘P-section’ — a set of passive shelf equalisers ‘inspired by’ the famous Pultec design, complete with separate boost and attenuation knobs. The low frequency section can be switched to turn over at 20, 30, 60 or 100Hz, using another pair of up-down push-buttons, and separate rotary controls are provided for boost and attenuation. This classic arrangement obviously allows both boost and attenuation to be applied simultaneously, and although this sounds mad, the result is actually very useful, producing a characteristic dip in the response just above the main shelf, which works extremely well in a musical context. The high-frequency section has three rotary controls and three buttons. Again, separate boost and attenuation controls are provided, along with an adjustable bandwidth facility, while the push-buttons allow the turnover frequencies of the boost and attenuation sections to be adjusted separately. The HF attenuation turnover options are set with a single button cycling through 5, 10 and 20 kHz options, while the HF boost can be tuned to 3, 4, 5, 6, 10, 12 or 16kHz, controlled


January 2012 / w w w . s o u n d o n s o u n d . c o m

display says ‘yes’ and push the button again. Quick, easy and intuitive, whilst also being pretty safe and foolproof! The A/B comparison function I mentioned earlier allows the currently selected memory settings to be compared with the live front-panel settings.

Few analogue equalisers have memory recall facilities, but two high-end units that do are the AMS Neve 8803 Dual Channel Equaliser and the Solid State Logic X-Rack fitted with Stereo EQ and E-Series EQ modules. However, in both cases, recalling settings relies upon the user matching the physical rotary control positions. In the case of the AMS Neve, that means matching the settings displayed on the recall software running on a computer. The SSL system employs a bi-colour LED above each rotary control, which shows red or green to indicate when the rotary control is currently set too high or too low, respectively. Both systems are considerably less expensive than the Bettermaker, but neither has such comprehensive EQ facilities, or such instant ‘recallability’.

In Use
The EQ230P is a very interesting equaliser. Each section produces text-book accurate curves with exquisitely precise control over sensibly designed ranges. More importantly, the unit sounds extremely musical and involving, facilitating anything from gentle tonal shaping to surgically precise corrective treatment — or even both at the same time, thanks to the two separate parametric sections, in addition to a full ‘P-style’ equaliser. I found the high-pass filter and two parametric sections sounded transparent, fast and clean — ultra-modern and precise — while the Pultec-derived section (which is inductor-based) provided a subtly warmer character, with an almost tangible sense of body and weight, but without the dynamic ‘sag’ that can afflict classic Pultec designs when driven hard,

by the two remaining up-down buttons. The final element of the front panel is the digital memory section, comprising a rotary encoder with a press-switch action to select the required memory, and a separate button to select the stereo link mode or to access each channel’s setting individually. Slightly strangely, if the unit has different settings on the two channels and the stereo-linked mode is then selected, both channels assume channel two’s current settings. This isn’t really a problem as such, although intuitively one might have expected channel one’s settings to dominate. The memory section can accommodate

required, while retaining the ability to access different EQ settings for different instruments or from different production sessions instantly and completely accurately. The promised USB-based plug-in interface would enhance this aspect even further, of course, especially if EQ settings could be stored on the computer and transferred to and from the EQ230P’s memories. There aren’t many stand-alone outboard

The rear panel includes a cut-out for a USB port, which will add functionality to the design in the future.

up to 399 user presets, and selecting a preset memory is as simple as rotating the knob to find the required location and then pressing the knob to load the settings. Memory locations with saved presets are indicated by a full stop after the number; empty memory locations don’t have the dot. The Bettermaker EQ230P ships from the factory with a few starter-EQ settings pre-loaded into the memory, including examples of typical vocal, mastering, kick drum, guitars, and more, to use as starting points and to demonstrate the versatility of the unit. Saving current settings is similar to selecting presets: select a memory location, hold the knob in for two seconds, then rotate the knob until the

and without the iron veil of input and output transformers. This unit really does embody the best of all worlds: precise, transparent correction and warm, musical creativity. There’s nothing about the sound that I don’t like, and even the user interface works well. If I have to be critical — and I do; it’s part of the job specification — then I’d point at the slightly restricted headroom margin, which might be a concern in some situations. But even that can be easily managed with careful gain structures and equipment alignment. The major selling point of the EQ230P, though, in addition to it being a superb sounding and comprehensively equipped analogue equaliser in its own right, is the memory recall functionality. This makes it extremely attractive for use in DAW-based work where an analogue character is

recallable equalisers, and none with the versatility of the Bettermaker, which goes a long way to justifying the pretty high price tag! It would serve equally well in a busy DAW-based production situation or in a mastering room, and there aren’t many devices that could claim that. I’m told that the Bettermaker boffins can also provide variations on the EQ230P to special order, with a Mid-Side version, customised frequency ranges, and even personalised logos! Clearly, we’re going to be hearing a lot more from this pro-active Polish company, and I’m looking forward to it. $ T E W W
$4500. ZenPro Audio +1 803 937 6012.

w w w . s o u n d o n s o u n d . c o m / January 2012



Dynamic Convolution Plug-in For Mac & PC
Dynamic convolution can be used to ‘sample’ any piece of audio gear. Have Acustica succeeded in giving this advanced technology a friendly face?

Acustica Nebula Pro 3.5

hen I last looked at Acustica Audio’s Nebula 3 Pro, in SOS February 2008 (www. nebula3.htm), I found myself impressed by its potential. This ‘hardware capture’ plug-in was said to accurately reproduce the dynamics, saturation and signature sounds of real-world hardware ranging from EQs, preamps, mics, tape machines and reverbs, through to dynamic effects such as compression and tremolo, and even ‘time-variant’ treatments such as chorus, flanging and phasing. Many of the sounds in its 6GB bundled library were of very high quality (particularly the preamps and reverbs), but I was less impressed by the confusing interface, the tape/compression effects, the high CPU overheads, and the very confusing web site. Fortunately, a lot has changed during the last three years (including a much easier-to-navigate web site!), so as Nebula Pro reaches version 3.5, it’s time to bring ourselves up to date. Acustica Audio have continued to concentrate on what they do best — enhancing their unique Volterra Kernel engine in a host of different ways and extending the options so that yet more diverse gear can be ever more accurately captured — but they now rely almost exclusively on talented third-party developers to release libraries of retro analogue gear, vintage tubes and equalisers, classic consoles, tape machines and other mouth-watering goodies. (In


CDSoundMaster’s Otari MTR10 running Ampex 499 tape at 15ips exemplifies the extra warmth and added high-end sheen you can gain from a well-calibrated tape machine.

the boxes that accompany this review, I’ve looked at five of my favourites.)

New Features
Nebula 3 Pro features an entirely rewritten CORE II Engine, and it’s now available in 64-bit and 32-bit versions for both PC and Mac. Written in assembly language, it now offers faster processing, greater efficiency, twice as many kernels (to

capture yet finer nuances from the original hardware), side-chain input options for all its compressors, and nested selection menus with extensive sub-categories, for much easier navigation through the hundreds of programs on offer. Smaller but nevertheless welcome improvements include various new metering options to help you get your input signals to the most appropriate level, and a Trim control


January 2012 / w w w . s o u n d o n s o u n d . c o m



Steve Stevens Steve Stevens Grammy award winning guitarist for Grammy award winning guitarist for Billy Idol “The AKG 214 is nothing short Bill Idol


of a revelation for my live guitar “The AKG 214 is nothing short of a sound. No more adding EQ at the revelation for my live guitar sound. console, this mic translates exNo more adding EQ at the console, actly what is coming out of my this mic translates ex-actly what speaker cabs and does so in a huge 3D way. The peace is coming out that of my speaker of mind it has given me, knowing my tone is cabs kicking and does so in a huge 3D way. The peace of ass out in the house is worth it’s weight in mind gold.“it has given me, knowing that my tone is kicking ass out in the house is worth it’s weight in gold.“

Martin Strayer Sound Strayer engineer Joe Cocker, Dixie Chicks Martin Cocker, Dixie Chicks Sound engineer Joe“I’ve been doing vocals

and acoustic guitar with the “I’ve been doing and it’s C 214 here at myvocals studio and acoustic guitar with the on fantastic! I also love this mic C 214 at my studio it’s our here electric guitar! It and is so true, fantastic! I also love this mic on clear, and beefy at the same time... it just reproduces our electric guitar! is so the sound coming out of the speaker so It well, so true, crystalclear, and beefy at the same time... it justtoreproduces clear and punchy. Truly a great addition AKG’s microthe soundline.“ coming out of the speaker so well, so crystalphone clear and punchy. Truly a great addition to AKG’s microphone line.“

Bryon Tate
FOH for Trey Songz Bryon Tate FOH for Trey Songz“After putting the AKG 214’s on

our guitar cabinet, I was immediately impressed with the clarity “After putting the AKG 214’s on definition compared to preourand guitar cabinet, I was immevious microphone The diately impressed with choices. the clarity guitar instantly stoodand out definition in my mixcompared without being harsh to preor thin. I no longer want to use anything else.“ vious microphone choices. The guitar instantly stood out in my mix without being harsh or thin. I no longer want to use anything else.“

Dave Natale

Dave Natale am very impressed with the FOH engineer Tina“I Turner world tour 09 AKG 214’s. ... they are now my
standard. The older mic’s “I new am very impressed with the I use are becoming harder AKG 214’s. ... they are now myand harder to find andolder it is mic’s great to new standard. The find alternatives. This is truly the first microphone, that I I use are becoming harder and have ever used flat with no EQ needed.“ harder to find and it is great to find alternatives. This is truly the first microphone, that I have ever used flat with no EQ needed.“

FOH engineer Tina Turner world tour 09

From to thethe Studio Stage! to the Stage!

C 214 C 214 From the Studio



CDSoundMaster: Reel Too Real & Tape Booster+
CDSoundMaster (http://cdsoundmaster. com) offer an extensive Nebula library, but I decided to highlight their $99 R2R (Reel Too Real) suite, described as ‘The Essential Analog Tape Collection’. It offers a huge 170 ‘virtual tape machines’ captured from eight different hardware models of various qualities and vintages, ranging from a 1950’s Wollensak 1515 running at 3.75 inches per second through domestic models like Akai’s 4000DS MKII, classic machines such as the Revox B77 Pro and Otari MTR10, right up to a 24-track Studer A800 MKIII sampled at both 15 and 30 inches per second. The variety of clean tape ‘colours’ on offer is wonderfully wide, from gentle mastering through the low-end warmth and top-end sheen of the Studer at 15ips, all the way to the telephone voice-like FX results of the early machines. The $39 Tape Booster+ library can be used on its own to offer natural-sounding tape saturation effects, but its 44 programs were primarily designed to offer greater realism in a second instance of Nebula following R2R. It has none of the frequency-response quirks of tape machines, instead adding a range of perceived volume increases up to 8dB using ‘drive’ derived from extra harmonic content (from subtle up to nearly 10 percent according to my measurements). This proved great for adding thickness and richness to drums in particular, but the combination of the two was even better, to me sounding as good as various DSP alternatives.

option for libraries that allows input and output levels to be automatically changed in opposite directions, so you can more easily hear how Nebula effects change your audio as you drive them harder. The engine is also expanding in other directions. A Server version lets the user spread its CPU/RAM load between multiple networked computers, while there’s also a Local Server version: both

extend the RAM ceiling beyond 1.2GB per Nebula 3 plug-in and offer a low RAM usage mode, so that all instances of the same preset use the same RAM. This is ideal if you want to run 50 instances of a console EQ across all the tracks of a complex mix, for instance. For the casual user, there’s also the Acqua interface, which allows third-party developers to to create stand-alone

AlexB: Preamp Color Suite
AlexB ( also offers a huge range of programs, some of which (his latest 4KD and MWD compressors, for instance) push the boundaries of what’s currently capable with Nebula. For me, the sumptuous Preamp Color Suite quickly became a favourite, offering emulations of 39 different highly regarded solid-state and tube preamps. Of course, the names have been disguised, but most enthusiasts will quickly recognise what’s being modelled from names such as ‘A-Meck’, ‘Portsilk’ and ‘VocBocs’. Some are offered without their input transformers or with them for a little more saturation and low-end roll-off, while others offer several captures with different front-panel filter/EQ settings. Nebula is exceedingly good at capturing the sound of preamps, and this is a beautifully recorded collection offering a surprising amount of tonal variation between the various devices, from the hardness of ‘AN81’ to the warm bottom end of ‘MTP std’. As expected, the tube models offer more character, and with the Telefunken V72 you even get both solid-state and tube versions. While you can tweak the sound with Nebula’s Drive control, some preamps are also supplied in clean and driven versions for more real-world accuracy. I was surprised by the amount of extra ‘snap’ between the ‘Focus8 Cln’ and ‘Drv’ versions. I’ve ended up using PCS a lot just recently, and at just €20 it’s a steal!

plug-ins running the Nebula 3 Pro engine hidden beneath their own GUI design. More radically, Acustica Audio now offer the entire Nebula 3 Pro engine running in CUDA format on most recent nVIDIA graphics cards, for those who want to offload some of the Engine’s processing overheads. This only currently benefits reverbs and some EQs and, like DSP hardware, hikes up the total audio latency, but there’s already plenty of potential here for the future.

Getting Up To Speed
Until very recently, Nebula was shipped with very conservative default settings to suit those with older and slower computers, which meant very high latency and sluggish level meters — not a recipe to win over new users owning more typical machines! However, I’m pleased to report that during the course of this review I finally managed to persuade the developers to change the

Acustica Nebula Pro 3.5 €139
• V3.5 offers lower latency and even better audio quality than previous versions. • Many of the latest third-party libraries sound truly stunning. • Installing and using Nebula is now considerably easier.

• Interface still a bit clumsy. • Occasionally rather heavy on CPU. • Bundled library could still do with a cull to raise the overall standard.

Nebula now offers some beautiful high-end analogue sounds via its third-party libraries, and while some may not get on with its idiosyncratic interface and working methods, it’s fabulous value for money for those prepared to put in a little effort.

AlexB’s Preamp Color Suite samples a host of desirable preamps, including this classic Telefunken V72 tube model.


January 2012 / w w w . s o u n d o n s o u n d . c o m

performance for the 1%, priced for the 99%.
m101 – the original high performance single channel mic preampli er. Huge bandwidth and headroom to capture all the details of your source. Our exclusive ribbon mic mode makes it the rst choice for gain hungry ribbons.

m501 – air and detail for the 500 series rack. Our signature transparent mic preampli er circuitry is the perfect palate cleanser for all the high calorie treats in your lunchbox.

m502 – the same circuit as the m102, but for the 500 series rack. A feedback opto compressor design that not only provides a wide range of dynamic control, but remains inherently neutral, open and musical.

m102 – open, musical optical compressor. All your tracks can now have glorious, easy to use dynamic control which regardless of how much you squeeze, will always end up with their sonic essence perfectly intact.

m103 – the complete input solution. Our transparent, musical microphone preampli er coupled to a silky smooth 3 band EQ and a sweet, open optical compressor. The m103 transforms ordinary tracks into beautiful recordings.


Cupwise: Tube FM1
Cupwise ( specialise in extensively sampled collections of unusual gear, and I was particularly taken by their Tube FM1 library. Captured from six 1950s table-top valve radios and stereo tuners by both DI’ing the input signal and transmitting it directly to the radio aerials, the frequency responses are widely varied (as you might expect from such vintage items), so you get a rich variety of tonal variation. Distortion levels are also much higher than modern hi-fi equipment, typically offering at least several percent of second and third harmonics for softer highs, and plenty of added warmth at the bottom end. The Nebula Drive control is also very effective over a wide range if you desire more subtle or more extreme distortion levels that can be pushed well over 100 percent. The radios are presented in different versions, some with carefully extended frequency responses for less radical tone-bending, and all are available in various options with kernel numbers from two (for lower CPU overheads) to 10 (for greater upper-harmonic realism). You’ll have to look elsewhere for subtlety, but at just $16, Tube FM1 offers a huge range of radically different vintage timbres, highly suitable for adding loads of vintage character to both digital and analogue sounds, and is superb value for money.

With radical frequency responses and high harmonic levels from a collection of 1950s radios, the Cupwise Tube FM1 collection is great for special-effect treatments.

few settings that mattered, so you no longer have to perform arcane tweaks to bring it up to speed. Two versions of the plug-in are supplied: Nebula 3 and Nebula 3 Reverb. The standard version is optimised for low latency and, with the new default settings, offers a very snappy 5.9ms latency when used in 44.1kHz projects and just 2.7ms at 96kHz. The other version is intended for presets requiring longer kernels, such as reverbs, but should always be used if you can cope with its higher

latency, since it offers potentially better audio quality with a smoother frequency response, and lower CPU overheads. Moreover, third-party developers always recommend its use to get the best out of their libraries. The ‘Reverb’ name is thus a bit of a misnomer, retained only for compatibility with existing projects. I’m also pleased to report that the days of its 374ms ‘wading through treacle’ latency are gone. On my PC, the new default settings produced very acceptable latency of 26ms at 44.1kHz, and 11.9ms at 96kHz.

The bundled Commercial Library is now cross-platform and up to version 4, although the below-par presets I commented on before are still there, just in case anyone still wants them. Personally, I think they should be relegated to a legacy library to increase the initial wow-factor of the others.

Skin Deep
Like many other users, I still find the default Nebula skin a waste of screen space: very few third-party libraries make full use of its eight programmable sliders, and controls 1, 2 and 8 nearly always default to engine settings for kernel attack, release and ‘liquidity’, all of which have already been optimised by the developer for best audio quality and can therefore be largely ignored. Fortunately, there are many smaller but very attractive freeware skins available from the Acustica Audio web site, featuring rotary knobs or even simple number fields beneath the main display window, to save screen space. Using Zabukowski’s freeware Nebula Set-ups utility (www.zabukowski. com/software) it’s also possible, if a little fiddly, to create separate Nebula plug-ins for your compressors, EQs, reverbs,

Analog In The Box: Mammoth EQ
Analog In The Box (www.analoginthebox. com) have some lovely items in their range, but possibly the most impressive is their €15 Mammoth EQ, a clone of a very popular and expensive passive equaliser. Mammoth offers parametric peaking EQs and shelving EQs, plus low- and high-pass filters. Most Nebula EQs offer a single band per program to minimise loading times and processing overheads, but these can be awkward to use when you have to launch two or three instances to tweak multiple bands. For this reason, there’s a very handy selection of ‘combo’ programs in the €10 Mammoth EQ Expander, each offering three different ‘reduced feature’ EQ bands without bandwidth adjustment, which is a great time-saving compromise. Various tests have shown that once you ignore the attractiveness or ease of use of a EQ’s GUI and concentrate solely on matching its frequency response with other EQs, many sound almost identical. The ones that genuinely offer extra character are those that add dynamic harmonic contributions, which Nebula, once again, does with panache. The differences may be subtle, but I was well impressed with the sound of Mammoth, which gave silky-smooth highs with lots of ‘air’ and a warm bottom end without mud.


January 2012 / w w w . s o u n d o n s o u n d . c o m


preamps and so on, each with a different skin and subset of Nebula programs. I found this made using Nebula far easier than wading through a huge number of programs in a single interface.

Rhythm In Mind
Rhythm In Mind ( offer a veritable Aladdin’s Cave of fascinating audio curiosities, with a huge range of single programs at pocket-money prices ranging from just $3. Their speciality is unusual front ends such as those provided by various classic hardware samplers (to add a little grit to your audio), rare filters and other effects that no-one else covers. I enjoyed the three vintage UTC transformer stages, each, at $6, offering a subtly tweaked frequency response and a few percent of mostly third-harmonic saturation down at the low end that proved ideal for adding flavour to drums and bass lines. However, the one that particularly appealed to me was the $10 vintage stereo RCA-Airon Line EQ for its unusual 10kHz program/tilt that reduces levels below this frequency and increases them above, allowing you to simultaneously add air and remove low-end mud, with the added bonus of an API transformer to add a little extra character.

Drive Time
Hitting Nebula with the right input level is crucial if you want to hear the hardware as it was originally ‘captured’. Generally, it’s better to be guided by the individual library developer, as well as using your ears, but Nebula 3 meters are by default set to VU mode, typically with average levels of 0VU equating to -18dB, and peak levels of around -6dBFS. This leaves some headroom, just as you normally would when using hardware. Some libraries do recommend higher levels of 0dbFS for best results, in which case I found it more helpful to change Nebula’s meters to Peak mode and aim for a full-scale reading. Control 7 is invariably Drive, which generally defaults to the input level at which the hardware was originally ‘captured’, leaving you the option of exaggerating or diluting reality with more extreme or subtle levels of harmonics at a particular input level. Otherwise, you simply use Nebula’s input and output controls to adjust how hard you ‘drive’ the effect, just as you would with external hardware, always being careful to back off either of these controls if the ‘overflow’ LED indicates digital clipping. Some libraries also ring nastily at some frequencies if you push them too hard, so leaving at least a few dB of headroom is normally wise. Most third-party libraries require Nebula 3 Pro rather than the Free version for best results, and generally offer a wide range of presets encompassing different drive levels and sample rates. Many have been captured at 24-bit/96kHz for 96kHz projects and even greater realism (albeit with higher CPU overheads), but will convert themselves on loading if your project uses lower sample rates. However, my listening tests suggested that if a developer offers both 44.1kHz and 96kHz library versions, choosing 44.1kHz

This unusual RCA-Airon EQ from Rhythm In Mind is a one-knob wonder, letting you add ‘air’ and remove ‘mud’ simultaneously!

Test Spec
• Acustica Audio Nebula Pro 3.5. • PC with Intel Conroe E6600 2.4GHz dual-core processor, Intel DP965LT motherboard with Intel P965 chip set running 1066MHz system bus, and 2GB Corsair PC2-6400 DDR2 RAM, running Windows XP SP3.

presets for a 44.1kHz/48kHz project often sounded a tad better than automatic downward conversion of the 96kHz ones. Most libraries also offer similar presets with several different numbers of kernels: each additional one adds the contribution of an extra harmonic, at the expense of extra CPU overhead. Nebula currently offers a maximum of 10 kernels, which helps it excel when capturing preamps, EQs and reverbs, but prevents it from offering grungy distortion that requires a wider harmonic series. It also captures a single snapshot of a compressor very well, offering plenty of ‘flavour’ and depth, although because it works with blocks of samples typically around 2ms for compression, it can’t currently offer fast enough attack times to kill the transients on kick drums and bass lines, as some hardware and software compressors can.

Final Thoughts
If you tried Nebula in the past and weren’t convinced, try it again now with some demos of the new third-party libraries reviewed here and, like me, you could become a convert. Getting

into its rather different ‘mind set’ can be tricky at first, but the huge number of high-quality libraries offering the sounds of esoteric audio hardware at pocket-money prices should make it well worth the effort. Due to space restraints, I’ve only been able to sample a few of the most interesting here, but you can find a full list of third-party developers at the Acustica Audio web site ( php?option=com_content&view=article &id=67&Itemid=160). Hopefully, a more informative manual will be completed by the time you read this, to ease newcomers more gently into Nebula’s sometimes quirky world, but it’s one that I feel is most definitely worth exploring. In essence, Nebula 3 Pro offers ‘captured hardware’ effects that in many ways rival those running on dedicated DSP cards, but at far lower cost. In these hard times, that’s a very welcome combination! $ Nebula 3 Pro €139;
Nebula 3 Server €189.



January 2012 / w w w . s o u n d o n s o u n d . c o m



“I chose Kurzweil years ago for its great sound. Kurzweil has allowed me to consistently deliver more sounds, more instruments and more performance flexibility than anything else on the planet at any price. Kurzweil’s non obsolescence has allowed me to bring years of work forward and gives me that competitive edge. Kurzweil provides the “keys” to my long and continued success.” - Rubén Valtierra, Keyboardist for “Weird Al” Yankovic

A division of Jam Industries Ltd

For more information email or call: | 800.431.2609


A German artist’s unique instrument designs forge a new sound world.

Best Service Klanghaus
Sample Library

or most of us, selecting an instrument we fancy owning is simply a matter of pointing at an item in a catalogue and saying ‘I want that one’. Top pros can go one better by asking manufacturers to build them custom instruments, ranging in scope and nuttiness from Pat Metheny’s 42-string Pikasso guitar to Bjork’s ‘gameceleste’ (a hybrid, keyboard-driven metallophone). At a higher level still are the visionaries who construct their own experimental musical designs from scratch: among their ranks we can count Benjamin Franklin (American Founding Father and inventor of the mechanised Glass Harmonica), Russian electronics wiz Lev Termen (Theremin) and our very own Henry Dagg, the ex-BBC sound engineer who dreamed up the Sharpsichord and the ‘toy cat organ’ — look up these names on the net, it’s an entertaining way to spend half an hour or so! Flying the flag for such far-sighted musical creativity is German-born artist-musician Ferdinand Försch. While studying composition, percussion and electronic music in the ’70s, Försch played in rock, jazz, big-band and orchestral settings, and put in a stint with hirsute pop band the Dukes (no, not the XTC psychedelic spin-off), before founding the improvisational Percussion Road ensemble in 1977. From the beginning, our man was interested in exploring the sonic qualities of ‘objets trouvés’ such as brake drums, baking tins and timber. After undergoing a Road to Damascus-style conversion during a brief UK seminar given by the out-of-the-box musical thinker John Cage, Försch decided (in his own words), “to dedicate my life to the research of sound. I built my first instruments, metal drums, in 1982 — and I never stopped.”


Having spent nearly three decades constructing ‘sound machines’, sound sculptures and installations, Försch’s collection of over a hundred instruments has graced concert stages and galleries around the world. In an echo of the aforementioned XTC’s Drums & Wires, they include drums, percussion and stringed instruments, many of them so visually stunning that they qualify as fully-fledged works of art in their own right. A commentator remarked that these instruments look alien, but their other-worldly, futuristic appearance is often tempered by a hint of something ancient — a few of the geometric shapes reminded me of Mayan symbols and Hindu mandalas, as well as evoking the Egyptian influence on Art Deco.

The maker shuns electronics in his live performances, preferring to let his instruments sound acoustically with no processing or sequencing. However, in his capacity as a composer of film and dance music, he does use computers and MIDI, like the rest of us, and it’s reasonable to assume that this led him to the idea of creating a sample library based on his instruments. Försch played all the samples himself and recorded them in his rehearsal studio, after which he spent six months working long hours to program the patches. The resulting 2.9GB collection is called Klanghaus (German for ‘Soundhouse’), taking its name from the maker’s experimental music venue in Hamburg. On a more mundane note, I should add that Klanghaus is formatted exclusively


January 2012 / w w w . s o u n d o n s o u n d . c o m

for Best Service’s Engine sample player (you can see my comments on Engine at articles/world-percussion.htm), software for which is included with the library. The samples ship on a single DVD in a conventional product box (shame, I was hoping for an experimental polyhedron). Product activation is done at Yellow Tools’ site, Yellow Tools being the company that developed Engine for Best Service; newbie Engine users need to register and set up an online account, after which activation is straightforward provided you faithfully follow the instructions set out in Klanghaus’ ‘How to Activate’ document. Stray off the beaten path and you might end up lost in cyberspace.

Percussion maestro Ferdinand Försch performs live in Shanghai.

Arctic Adventure
One of the enjoyable things about reviewing left-field sample libraries is deciphering unfamiliar names — what, for example, is an ‘Arcton’? This mysterious term refers to an instrument that seems to have obsessed Försch since he built the first prototype in 1986: it consists of a flat, narrow length of metal over which are stretched several metal strings, each resting on an intermediate bridge that determines its

Best Service Klanghaus $285
• A beautiful, unique set of percussion, stringed and hybrid instruments created by a visionary designer. • Contains a large range of exotic unpitched and tuned percussion textures, many of them unlike anything you’ll have heard before.

• Arguably not the greatest bang-for-buck, although one should factor in the rarity value of these instruments. • Some of the MIDI performances are very rudimentary.

Unique, arty and fiercely non-standard, this is a great collection of unusual percussion sounds for those looking to add something different and fresh to their mixes. Film composers and neo-classical experimentalists will love it, and its esoteric, colourful and powerful timbres have the potential to enhance many styles, from left-field pop to underground dance. The library focuses on single hits; there are no played loops and some of the MIDI performances are pretty minimal, so be prepared to program your own grooves.

tuning. The nearest equivalent would be a Chinese Gu Zheng or Japanese koto, but this contraption looks far more industrial. Fixed underneath the metal surface is a large, lozenge-shaped, flared plate-metal resonator very similar to that used in the Cristal Baschet (another exotic item for you to look up). The whole shebang is perched on a stand with castors, so you can move it around without doing yourself (or the Arcton) a mischief. Since first developing the instrument, Försch has created bass and tenor Arctons, a double Arcton, and finally the fabulous ‘Triple L’Arcton’ featured in this library, which boasts two resonators, three metal sound boards and a total or 10 or so strings. Played with a drumstick, it sounds something like a giant zither or hammered dulcimer, but the comparison is academic: the Arcton’s steely attack, stentorian tone and hugely resonant, slightly menacing chime place it in a category of its own. You can select full-length, short or ‘rebound’ stick bounces with keyswitches: the ‘short’ option abruptly kills the sustain while allowing the decay of the note to carry on, which produces a fabulous subterranean rumble in the bass register, akin to that of the Close Encounters mothership, or a distant tube train in a tunnel — an unnerving effect crying out to be used in film scores. While the sticked Arcton samples work well for anything from histrionic, Spaghetti Western-style twangy electric guitar themes to crashing, pianistic bass octaves, the bowed version of the instrument is something else again: a grinding, continuous, metallic timbre like a tambura or hurdy-gurdy, it’s ideal for creating big bass drones, and its four-octave range can also accommodate melodies and harmonies. The provision of high-pitched overtones extends the bowed samples’ compass by two further octaves. I particularly enjoyed the eerie ‘flageolet’ harmonic arpeggios; the only drawback is that these high harmonics can get pretty shrill; to mitigate the ‘fingernails on a blackboard’ effect, you can push up

the mod wheel, used throughout Klanghaus as a low-pass filter controller. It would be a challenge for the Musicians’ Union to establish a session rate for this weird and wonderful instrument — the porterage alone would be astronomical! Happily, Klanghaus buyers can now add the Arcton’s steely tones to their mixes without first having to go on bended knee to their bank manager.

Klangrausch Gongs & Chimes
Klangrausch: now there’s a good word. My partner assures me that the noun ‘Rausch’ loosely translates as intoxication, inebriation, a drugged-up state or frenzy, which is a pretty fair assessment of how we SOS reviewers approach our work. Here the term is used to cover a diverse variety of performances on various metal instruments. My frenzied, inebriated key-presses produced doomy, portentous, gong-like bass resonances, the weird pitch-bent groan of a large gong being dipped in water, and fantastic, scary, low-pitched, roaring Arcton noises, all wonderful timbres for a horror-film soundtrack. Best of the bunch were the tremendous ‘String Sculpture’ low, boingy hits. A ‘Polytone’ is also included; no idea

Instrumentation & Loops
Individual Instruments • Triple Arcton • Polytone • Steel Resonator • ‘String Sculpture’ • Water gongs • Metal drums (4) • Water drums (3) • Channel gongs (chimes) • Aluminium bells (chimes) • Metal plates (scrapers and so on) Mixed Instruments • New Percussion Set (mixed drums & percussion) • Prepared Bonus Set (mixed percussion) • Objets Trouvés (junk percussion) Rhythm Loops • Ferdinand Försch Metal Drums rhythms (4) • Ferdinand Försch Prepared Loops (3) • Markus Krause loops (8) • Oliver Morgenroth loops (9) (Numbers in brackets = types of instrument or loop.)

w w w . s o u n d o n s o u n d . c o m / January 2012



what that is, but it sounds great, like a large gong crossed with a bass trombone. Other Klangrausch patches feature speaker-cone-ripping rumbles reminiscent of a large tam tam gong played in a wind tunnel, massive steel resonator strikes, metallic edge hits and bowed, teeth-on-edge, baking-tray screeches (something I’ve yet to see Delia Smith attempt), semi-tuned, waterphone-style spoke twangs, demented circular scrapings on what sounds like a metal salad bowl — you get the idea: a tremendous array of utterly unpredictable, powerful and highly usable metallic textures. After a wild, drugged-up romp through the Klangrausch samples, I was ready for something more chilled-out, so was pleased to discover the library’s ‘Channel Gongs’. These are suspended tubular chimes with a beautiful, somewhat mysterious tone. Their upper notes have a soft, bell-like, clear note, while the pitch of the low register notes is more ambiguous, which all adds to the sense of mystery! A set of keyswitches allows you to select soft beater, hard beater, tremolo or crescendo tremolo hits, as well as a couple of variants where the pitch falls away as the note decays, as if the chimes were being dipped in water — I absolutely loved this effect. The maker has also included splashy water sounds that form a strange sonic symbiosis with the gong-chimes’ metallic tones. Sensibly, the

water has its own separate fader, so you can ‘dry up’ the sound if required. In a similar vein (though much higher in pitch) are ‘Aluminium Bells’, which bear a strong visual resemblance to the ‘mark tree’ instrument used by many Latin percussionists — you’ll have seen it in cutaway shots on countless TV music shows: a row of small metal chimes, gradated in size and suspended from a wooden frame, it’s usually played with a quick sweep up the chimes and is a compulsory overdub for tasteful pop ballads. The Klanghaus version is a little more strident, but no less attractive: as well as performing the obligatory glissandi sweeps, Försch has individually sampled each of the chimes and mapped them out chromatically. Result: a pretty, perfectly-tuned chimes patch that can be layered to great effect with other keyboard textures. Taken together, these diverse instruments constitute a jamboree bag of highly enjoyable, eclectic metal percussion timbres.

Klanghaus’ instruments are unique one-offs, so there is no true alternative. However, libraries such as VSL’s Elements and Soniccouture’s Glassworks feature small collections of unusual, left-field percussion instruments and maintain a similar spirit of experimentalism.

Metal & Water Drums
Recalling Carl Palmer of ELP’s notorious 1970s stainless steel drum kit (which weighed 2.5 tons), my heart sank when I read the words ‘metal drum’. Fortunately, Klanghaus’ version is no monstrosity requiring stages to be reinforced and roadies issued with body-building supplements, but a fairly small instrument

The Klangrausch (‘sound frenzy’) section of the library houses some of its most iconoclastic metal sounds.

that partly adopts the design of a tongue drum (aka logdrum or Gato drum). Försch explains its inception thus: “With the help of a carpenter, I designed a rectangular wooden resonator and fitted a 3mm aluminium panel in which I cut different lengths of tongue. I added holes to the alu-panel and screwed it on... After this first invention, I built this metal drum in different variations: rectangular, eye-shaped, hour-glass shaped, and I discovered: this is fantastic, each form has its distinctive, individual new sound.” This design innovation has certainly created some beautiful tuned-percussion tones: the instrument is played with soft mallets, avoiding the harsh, chinky knock of a hard beater on a metal surface, and the normal earthy thunk of a logdrum is transformed into a purer and more sustaining, metallophone-like pitch. Four specimens were sampled for the main metal drums section of the library: the first is a dead ringer (if you’ll pardon the pun) for the exotic chime of a Javanese bonang, the second has a marimba-like quality, the third sounds something like an African balafon crossed with a logdrum and the fourth has a metallic, ringing, industrial flavour. These are the closest verbal approximations I can offer; the metal drums are truly tonally unique and, unlike mass-produced instruments, each of their notes has a subtly different timbre from its neighbours. Sadly, despite having 10 or so notes, these four instruments are limited to four pitches only, mapped to adjacent white notes in non-linear pitch order right at the bottom of the MIDI note range (which means you need an 88-note keyboard to play them in real time). That’s a pity, because the first metal drum in particular has a gorgeous timbre, which, in my view, deserves the full chromatic treatment. However, I can see that this kind of conventional presentation may not be a priority for the instruments’ inventor. By way of compensation, each metal drum has its own set of MIDI performances, which I’ll describe shortly. Complementing the metal drums are a set of ‘water drums’. These appear to be of the same basic design as the metal


January 2012 / w w w . s o u n d o n s o u n d . c o m


Metal drums and ‘prepared’ multi-percussion sets are grouped together under the cryptic heading ‘Bio Modul’.

drums, with the added attraction of water sloshing around inside. The water noise may be switched off on the instruments’ front panel, and their straight hits are augmented by some fine, atmospheric ‘finger tremolo’ performances. Unlike their non-watery brethren, each water drum is chromatically mapped across a couple of octaves.

As in other Engine-formatted libraries, pre-programmed MIDI performances are supplied, which may be triggered by a single key-press (the ‘trigger keys’ are marked in green on Engine’s on-screen keyboard). The metal drums offer 20 or so basic — and I do mean basic — MIDI performances created by Mr F, all consisting of a simple event that repeats at the top of every 4/4 bar when you hold down its key. These events are limited to, firstly, a single note; secondly, two, three or four reiterations of one note; and thirdly, a phrase consisting of a quick hit on each of the four pitches. Musically ambitious it is not. I’m perplexed that Klanghaus offers its users such rudimentary programming. What’s the point of providing a ‘performance’ of a single note being repeated every four beats? Surely we can manage that ourselves? The maker also created some ‘Prepared Loops’ performed on a fabulous set of big, strong percussion hits whose star turn is an enormous-sounding, low-pitched tar

frame drum (or something that sounds remarkably like it). These also feature the one-note-per-bar basic MIDI patterns, but as you go up the keyboard, a few syncopated mini-phrases are introduced. I’m glad to report that in these patches, the big percussion sounds have been mapped chromatically, so there’s nothing to stop us diving in and creating our own loops. Försch’s MIDI performances are augmented by a bonus collection of MIDI-driven loops programmed by Markus Krause and Oliver Morgenroth. Mr Morgenroth’s contribution is ambient and impressionistic, and the kits that load in with his loops include some lovely ‘alien whales’ sounds. Krause (clearly more at home on the dancefloor) has used imaginative sound combinations to create some fine, toe-tapping performances. Some of his loops reminded me of the sublime rhythm work of Bashiri Johnson, a man who can whip up a killer groove from the humblest of materials. Given the unique nature of some of Klanghaus’s samples, expect to hear the library yielding more top-quality rhythmic material in this vein. (See technical note in the ‘System Requirements’ box.)

for) what interests me. These were bizarre areas, from car parts to buzzsaw blades. I then went on to set up my performances in the conservatory with junk which would be played by three to five percussionists — at that time it caused a scandal.” The timid, insular classical world of the 1970s is thankfully long gone: your average music professor is no longer shocked by the prospect of a man hitting a piece of metal with a stick, and today’s audiences are certainly not averse to listening to junk. (No Simon Cowell jokes, please.) Klanghaus obliges modern listeners by serving up ‘Metal Plates’. These turn out to be wooden-handled metal scrapers, an essential fashion accessory for the trendy decorator. Stick hits on the metal end are a little ho-hum, but the chromatically tuned version makes a great chimes patch, along the lines of the aluminium bells described above. Even better are the ultra-high-pitched bowed scraper samples, which sound almost like an electric violin — they too can be layered beautifully with lower-pitched sounds, and their bat-frequency high notes will cut through the densest of mixes. Rounding off the junk percussion section is a collection of ‘objets trouvés’: 44 miscellaneous boings, biffs, clangs, bashes and thuds, which I assume emanate from radiators, bits of rusty old metal, biscuit tins and other 21st century detritus. (‘Any old iron’, as they used to say in the ’50s.) The advantage of such nondescript sounds is that since none of them are quite colourful enough to leap out and catch the ear, they tend to blend together very well. This means you can sequence them into complex 16th-note grooves without the overall sound picture getting too busy. This is a nice contrast to the dramatic, intense, attention-grabbing sonorities found in the library’s Arcton and Klangrausch sections.

General Observations
In a similar (but more colourful) vein, Försch’s ‘Prepared Bonus Set’ is a nice assembly of short, percussive, largely semi-pitched sounds played on unidentified (and unidentifiable!) instruments, including (I think) some muted string plucks. If you’re looking for something a little less esoteric, the ‘New Percussion Set’ single-hit patch includes a mixture of kick drums, low frame drums, boobam-style

Junk Maelstrom
Asked how he began his search for unconventional percussion, Ferdinand Försch responded: “I went straight to the junkyards and rifled through them (looking


January 2012 / w w w . s o u n d o n s o u n d . c o m

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Saffire PRO 40
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Ferdinand Försch shows off his extensive instrument collection — now, which one shall I hit first?

System Requirements
Klanghaus is formatted exclusively for Best Service’s Engine sample player (based on Yellow Tools’ Independence player). Engine runs stand-alone or as a AU, VST or RTAS plug-in on Mac OS 10.4 or higher and Windows XP, Vista and Windows 7 (32-bit and 64-bit). The minimum recommended systems are G5 or Intel Mac 1.8GHz or Pentium/ Athlon XP 3GHz PC; 1GB RAM; and a SATA 7200rpm hard drive. You also need a DVD drive for installation andnternet connectivity for product activation. Owners of Windows XP systems should take note that older, sub-3GHz PC processors sometimes can’t cope with the CPU demands of the Engine player. As a result of user feedback on this topic, Best Service’s minimum system requirements for Klanghaus Windows buyers has been upgraded from the “1.4GHz processor, 512GB of RAM” printed on the library’s product box to the numbers shown above. Technical note: most of the MIDI performances in the Klanghaus review copy would not play in time on a Windows XP machine; a pronounced delay was audible on the front of all of them, and the delay tended to be a different length each time (anything from a 32nd note to a full quarter note at 120bpm). This occurred in the PC stand-alone and plug-in versions, both over MIDI and when playing the samples by pressing the keys of Engine’s on-screen keyboard. Strangely, the Markus Krause loops were unaffected by this issue, and it didn’t occur at all on a Mac.

toms, China cymbal-like crashes, gongs and various unidentified metal percussion hits: 42 sounds in all, mapped in defiantly non-General MIDI style (although at least the kick drum is on bottom C!) Both of these sets are a nice resource for creating your own Klanghaus grooves. Multi-miking rears its head in this library, as it does in so many others nowadays. The facility is implemented only for the Arcton, which has three faders labelled Mic 1-3 respectively, each with a radically different tone: the first focuses on the mid-range, the second is very toppy, with no bottom end, and the third features only low bass frequencies. There is no apparent difference in listening perspective — the sound variations are purely tonal and quite exaggerated, so it seems likely that the three signals have been radically EQ’d. Although that’s not the way most libraries handle multi-miking, the good news is that if you whack all three up to full volume it creates a good, powerful, full tone with no missing frequencies. All’s well that ends well, though I would have preferred to see some explanation of this feature in the manual. Resonance is a big feature of this library: many of its big metal instruments have so much of it that adding extra reverb would be almost redundant. However, Klanghaus’ smaller drums and hand percussion do benefit from a bit of reverb sweetening; this is done via a nice, built-in convolution reverb which is perfectly matched to these percussive samples. You can turn the reverb up, down or off with a fader on Engine’s ‘Quick Edit’ page (that’s the one with the colour picture reminding you what instrument you’re using). For more detailed tweaking, you can alter the reverb type, length, pre-delay and so on on the player’s Pro Edit page. When using samples, I like to be able to hear their basic sound without any extra

processing. I’m happy that Klanghaus’ instruments load with convolution reverb turned on (it’s a pretty subtle effect, after all), but I was concerned to see that the programmers have added a limiter and level meter on the Pro Edit page. Guys, we don’t need to put a limiter on our samples at source, or look at an internal level meter — those activities are best done outside the player, in the mix environment. I don’t know whether the additional modules add much to Engine’s CPU demands, but it does seem like a case of trying a bit too hard.

Ferdinand Försch says that the relationship between the acoustic and the visual is a central inspiration in his work, so it’s good to see Best Service supporting that idea by supplying an eight-page, DVD-style colour booklet. Most of the booklet pictures are replications of the GUI images that appear on Engine’s screen when you load an instrument, virtual faders and all. The only extras are a superb wide-angle shot of Försch’s elaborate stage setup, a moody cover image of a bowed Arcton and a shot of Mr Försch sharing a joke with Best Service’s Klaus Kandler. I was disappointed that the booklet didn’t make more use of the maker’s large archive of beautiful photos — though I guess we’re unlikely to ever see again the likes of the handsome, 56-page, catalogue-style book that accompanied the 1990s Supreme Beats percussion library, I nevertheless felt that this visually rich collection deserved something more substantial and arty. Thankfully, that minor omission does not impinge upon the central purpose of this project, which is to give the world and his wife (if she’s interested) access to one man’s fascinating, idiosyncratic and highly evolved sound world. Those interested in unusual percussion timbres will find plenty

of inspiring sounds, though you may have to spend a little time getting to know where they are. The documentation is somewhat minimal and it’s easy to overlook some of the great sounds lurking in the corners. Klanghaus effortlessly avoids the mainstream; its non-standard approach should provoke a correspondingly creative response from the artistic community (that’s you, dear reader). This is not a go-to library for banging dance beats, though there’s no reason why its contents can’t be channelled into creating some awesome grooves. In this utilitarian, conformist age, it’s great to hear some genuinely new, experimental sounds, and given the unique nature of the instruments, we should be grateful that their inventor decided to throw open the doors of his Klanghaus. $ T E W W
$285. Best Service +49 (89) 4522 8920.


January 2012 / w w w . s o u n d o n s o u n d . c o m




Portable Synthesizer


The toy-like exterior of Teenage Engineering’s OP1 belies the versatile synth, sequencer, sampler and recorder hidden within. We retreat to our bedroom and put it though its paces...

ometimes a product hits the market well in advance of the technology required to do it justice. Who can forget the first brick-sized mobile phones, or Clive Sinclair’s ‘pedalo’ electric car? However, even the lowliest products deserve a revisit from time to time — for example, Casio’s VL-Tone. This tiny synth, sequencer and calculator was best known for its role in Trio’s hit ‘Da Da Da’ and even though it was rather cheesy, the VL-Tone had two points in its favour: it was small and it was cheap. Fast forward 30 years and there’s a strange feeling of déjà vu surrounding the latest toy-like, but definitely not Fisher Price, synth. Despite originating from Sweden rather than Japan, it’s difficult not to imagine Teenage Engineering’s


OP1, with its mini keyboard and internal speaker, as today’s VL-Tone. However, I’m sure its creators hope for a more substantial legacy, given that they’ve had three decades of technological advances to draw on. The OP1 has eight different synthesizer engines, a choice of three sequencers, drums, effects and a four-track, tape-style recorder function. With a motion sensor, sampler, basic mastering capabilities and even an FM radio, you probably won’t miss the calculator — unless you’re a busking accountant!

Teenage Kicks
Even though I’d seen its pictures on the Teenage Engineering web site, they didn’t prepare me for the physical reality of the OP1. This miniature workstation is shipped in a recycled Paperfoam box, ideal to keep it

safe as it travels around with you. Inside the box you’ll find the OP1, two elastic bands (to keep the box together), a transparent overlay and a USB cable. There’s no sign of a manual, though; for this you need to pay a visit to the Teenage Engineering web site. For a few seconds, I wasn’t quite sure what to think, until I started to handle what proved to be a surprisingly weighty slice of cold aluminium. Any lingering suspicions that this was a toy quickly faded because, despite being just 28 by 10cm, this slender metal object is a thing of beauty. Underneath is a plastic, braille-marked panel that helpfully points to the power switch, mini-USB 2 port and stereo I/O on mini-jacks. Power is supplied by a Li-Ion battery that proudly boasts 16 hours of active life. Charged via USB, this impressive battery also claims two years of stand-by time. The main controls are four encoders and, along with the volume knob, these boost the height to approximately 2cm, while four rubber feet do their best to keep that smooth, grey underbelly scratch-free. Teenage Engineering describe themselves as “young minds working with technology” and anyone doubting their credentials should immediately flick the power switch and marvel at the OLED display. This 320 x 160-pixel screen is fantastic; it’s pin-sharp, intelligently laid out and uses colour and animation meaningfully. On each page, colour is the natural link between on-screen objects and encoders — simple and highly effective.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Initially, you boot into four-track mode with its image of reels and scrolling tape. The transparent overlay describes how to progress from fledgling synth tweaks to a recording in eight easy steps. Following these, you quickly appreciate the layout and feel of the buttons, which are as positive as they are plentiful. You also appreciate the tiny internal speaker, which is perfectly adequate for extended periods of playing around the house, something I did increasingly as the days turned to weeks.

Sampling & Synthesis
It was instantly clear that the OP1 is not a nerdy or techy instrument. Instead it’s slick, occasionally innovative and, above all, uncluttered. There’s a real feeling of restraint, of limiting the tweakable parameters only to the essentials so that you’re never diverted from what matters — making music. Internally, there’s 64MB of RAM and a generous 512MB of flash storage — more than enough to hold a decent collection of audio material and synth patches. USB connectivity ensures that you can back up your work and transfer data across in either direction. In fact, pretty much all file handling is done externally, whether it’s creating and naming folders for user patches or backing up the various types of data. Sometimes, with no computer nearby, I wished I could just name and save my patches or store a ‘reel of tape’ to flash, but this isn’t currently possible. At least everything you do is instantly backed up,

so whenever you feel the urge, picking up from where you left off takes only a matter of seconds. One note shy of two octaves, the fixed-velocity keyboard responds reliably and, though giving little scope for expression, feels durable and solid. The keyboard may be transposed by four octaves either way, the transpose keys doubling as a basic pitch bender when Shifted. With a press of the blue Synthesizer key you’re free to explore the various ‘engines’ on offer. The OP1 isn’t multitimbral, but each engine has six notes of available polyphony and a streamlined user interface that is never daunting. Via eight sound-selection keys you have instant access to patches of your choice based on any of the engines. Bring the Shift key into play and a list of engine types plus available presets within each type is revealed. The engine choices are: Cluster, Digital, Dr Wave, FM, Phase, Pulse, Sampler and String, and each has its own character and supporting graphics. Considering the scope of the synthesis on offer, the factory patches aren’t too spectacular, only a few hinting at the power under the hood. This is where you come in. Checking through each engine in turn, Cluster is revealed as a “multi-layered oscillator cluster” delivering sounds that range from punchy and almost analogue to huge and fuzzy: think ‘supersaw from hell’! The Digital engine employs waveshaping, ring modulation and a control simply named ‘Digitalness’; these all combine to deliver an assault of spiky, crystalline textures or, after some encoder tweaking, noisy, distorted organs. Dr Wave is “Frequency Domain Synthesis” and encompasses tones reminiscent of oscillator sync and formant synthesis, as demonstrated by the patch ‘Talk Box’. The FM engine is more familiar territory, its parameters shifting around on screen as you adjust the frequency, FM amount or routing of its four operators. Then there’s a Phase engine representing Phase Distortion and Pulse, which is two pulse waves whose positions and levels can be adjusted and modulated. Then there’s an engine I couldn’t help pausing on: String. Apparently this is a “waveguide string model”, but whatever the code is doing, it caught my attention as the ideal source of cutting basses and general twanginess, plus shimmering string pads.

The OP1’s ‘String’ synthesizer engine.

With four main parameters per engine and no real explanations in the manual, you’re left to dive in and experiment with no preconceptions or baggage — although in the case of the final engine, the sampler, you should at least recognise the terms used! The sampler offers up to six seconds of sample time per patch and, once again, is all about speed and accessibility. A sample may be sourced from the onboard mic, the line input or the built-in FM radio; alternatively, you can feed it with tracks plucked from the recorder. Thanks to the keyboard’s impressive transpose range, a single sample can be taken into some very strange regions, while, for hands-on adjustment, the encoders set the sample start and end points, plus the loop points. Although there’s no built-in normalise function, levels can be boosted after recording, should that be necessary. A small line of LEDs at the side acts as a VU meter, and otherwise there’s remarkably little else to think about. The sampler and synth engines sit in front of common pages for the ADSR envelope, effects and modulation. At any time, you can

Teenage Engineering OP1 €799
• The implementation of the ‘Tape Recorder’ function is excellent. • Straightforward sampling, plus a selection of synthesis engines to explore. • High production values. The screen, in particular, is superb. • A self-contained production or jamming machine that you really can take just about anywhere.

• Not cheap. • MIDI implementation a little thin. • File operations require a computer.

The OP1 draws you in with Furby-like cuteness. Although you will probably wince at the bottom line, it packs a whole lot of slick, creative and approachable technology into a very small and well-designed box.

w w w . s o u n d o n s o u n d . c o m / January 2012



Alongside the power switch a USB 2 port and 3.5mm audio in and out sockets make up all of the OP1’s connections.

replace the current engine without affecting parameters on these screens.

Effects & Drums
The effects are rather idiosyncratic, a delay and spring reverb being the most ‘normal’. There’s a “hacked telephone system” that is so lo-fi it doesn’t just cheapen your audio, it trashes it and throws in strange artifacts of its own. Then there’s a rather bonkers filter whose Frequency, Punch (resonance-ish), Power and Rounds parameters suggest sonic pugilism of an extreme kind. This gritty digital filter varies in effectiveness according to the synth engine loaded; the more harmonics it has to work with, the better. I wouldn’t say the effects are game-changers but they make a worthwhile contribution. Whether it’s the metallic fizz of the spring reverb or the three-dimensional feedback of the Grid effect (think broken flanger or primitive digital delay), they get under your skin — especially when modulated by one of the more unusual LFO implementations of the synthesis world. As well as setting regular cyclic modulation, you can specify the internal three-axis motion sensor as a mod source. Waving the OP1 in the air to generate modulation felt more ‘right’ than I expected, even if it was too easy to hit buttons and accidentally select new patches while doing it. If wavy vibrato doesn’t appeal, how about radio-sourced modulation? Seriously, the internal FM radio isn’t just for sampling and ambience, it’s for modulation too. I long ago gave up trying to persuade analogue modular synth makers to build a voltage-controlled short-wave radio (surely the ideal random noise source?), but hearing the OP1 convinces me I should have persevered. Although not quite as packed

with spacey wibbles as shortwave bands, FM is populated with plenty of bright, clear stations that can be used to modulate, say, the start point of a sample or the damping of the spring reverb. If that isn’t sufficiently wacky for you, further malformation can be introduced by hijacking the microphone or line input. Any of these can act as a modulation source directed at any part of the synthesis engine, making the OP1 far less basic than it first appears. Drum mode incorporates the same effects and LFO capabilities, its kits

The OP1’s Tape Recorder function — a source of almost limitless audio-mangling fun.

constructed from slices of audio sampled on the OP1 itself, or imported via USB. A kit has 24 possible sounds (all of which can play simultaneously), and there’s no reason to limit them to percussion. I got great mileage from synths, vocals, even short loops packed within the maximum 12 seconds of sample time. Each drum is defined by setting start and end pointers within the sample then transposing, reversing or looping as required.

Tape Recorder
Although the synths and the drums are entertaining, it’s the ‘Tape Recorder’ that takes the biscuit, eats the biscuit and then

returns for the whole cookie jar. With a total running time of just six minutes, four mono tracks don’t look too special on paper, by today’s standards anyway. This isn’t the place to discuss whether great music can be born from such limitations, but it is the place to remark that this particular tape-recorder simulation is eerily addictive. Everything from the graphics to the sound of tape scrubbing over tape heads adds to the feel of authenticity. There are wild liberties to be taken too, without fear that the tape might snap! For example, you can switch into reverse at any time, even during recording. Naturally, you can bounce tracks down, loop or repeat sections, and even record while manually winding, for some seriously trippy sound effects. As with real tape, if you record at a faster speed you obtain better-quality results, but my favourite trick was at the other end of the spectrum: slowing it right down, then sampling the results. Monster sampler patches are easily made by overdubbing several synth parts, then lifting them straight from tape into the sampler. Even though each track is mono, you can overdub or cut and paste seemingly forever, capturing performances from the synth, drums or from external sources, such as the iPad I have nearby. If you turn on ‘beat matching’, bar lines scroll along with the tape, as an aid to making smoother loops. This won’t be mistaken for a DAW in a hurry, but there are some frills, such as the built-in metronome and tap tempo — features I never had on my old TEAC four-track! Each track passes through a simple mixer with just level and pan controls; the summed


January 2012 / w w w . s o u n d o n s o u n d . c o m


Experience an innovative new collection of instruments featuring creative and esoteric samples and manipulations of various acoustic guitars. Fractured features 2.3GB of compressed samples and more than one hundred new instruments with numerous variations and effects built into each. Each patch features the organic goodness of the acoustic guitar but stretches its timbre in new directions to create previously unheard musical effects. Fractu Experience Fractured: Prepared Acoustic Guitar.

NAMM Booth 6514


You can record the Tape Recorder’s output to create an ‘album’ of up to six minutes on each side.

the 16-step Pattern sequencer, a more traditional grid-based implementation ideal for polyphonic keyboard or drum parts. Sequence length can be dynamically adjusted during playback, notes rotated, and so on. The step sequencers may be latched their directions can be changed without missing a beat, and all three can drive external instruments.

There is little direct competition; perhaps the iPad is the closest, with its rounded sleekness, portability, and flexible choices of recording, sampling and synthesis apps. However, the OP1 will never distract you from the act of making music by offering a quick game of pinball, plus it has none of the iPad’s annoying restrictions when it comes to transferring data via USB.

result then hits a three-band EQ, a master effect and a final drive section. The master effect is a stereo version of the effects encountered earlier, placed over the entire output. As each instrument passes through the currently selected tape track, this proves to be a way of adding a second effect to synth or drum patches. The mixer even includes a simple compressor, plus a graphic representation of the entire sound path. Having laid down some tracks, you must connect a USB cable to perform a backup or do further production work on them. If you prefer to stay in the box, you can create a stereo master ‘album’ by cutting some virtual vinyl. Two album sides of up to six minutes each can be recorded and the resulting stereo files (in AIFF format) are then ready for transfer to your computer. Even as the vinyl is being ‘cut’, you can add new material alongside the Tape Recorder’s output; the album records whatever it receives.

With no obvious MIDI connections, you’d be forgiven for thinking the OP1 was completely self-contained. While it’s true that there are gaps in its MIDI implementation, there are still goodies to enjoy, thanks to a USB port that isn’t exclusively for backup purposes. In the OP1’s normal operating mode, it receives notes and pitch-bend on a single (currently fixed) MIDI channel. Played via a master keyboard and with the line output connected to studio monitors, the various synth engines started to come alive. I found that chords and heavy velocity produced distortion that had previously gone unheard but it rarely took long to bring this under control. The OP1 doesn’t respond to sustain-pedal information, nor does it sync to (or transmit) MIDI clock, but Teenage Engineering aim to polish the MIDI spec considerably before declaring their opus ‘finished’. I also experienced some odd

even offer relative or absolute operation, which is a rare choice where endless encoders are concerned.

The OP1 makes me think of Keira Knightley — there, I’ve said it! To explain: it is incredibly slender and attractive, yet there’s always a feeling its destiny lies beyond my lowly circles. The price reflects an unflinching approach to quality, but it’s bound to be a significant factor in any purchase decision. Apart from a few aspects of the MIDI implementation and the reliance on a computer for file handling, I couldn’t fault the attention to detail. The OP1’s technologies fit neatly together and offer a level of focus rarely seen in today’s hardware, thanks in no small measure to the welcoming display. Clear graphics reveal each engine far better than long descriptive text, and if each is a relatively uncomplicated digital affair, who’s to say that digital can’t be beautiful too? If you prefer to work with slices of organic reality, the sampler and microphone are never far away. It was the tape-recorder function that really grabbed me, though. I wasn’t expecting such a creative tool, with nuances such as the impression of tape moving at varying speeds across the heads. It offers an unparalleled degree of interaction for a tape recorder, becoming a performance instrument in its own right — and, happily, without all that crappy head-cleaning, tape stretching and oxide dumping that nostalgia fails to mention! Falling somewhere between ‘musical sketchpad’ and ‘full production workstation’, I found Keira to be an elegant companion, ideal to keep nearby should the muse strike, and ready for extended action, thanks to superior battery life. Even though currently out of my league, the OP1 takes portable musical fun to a whole new level. $ T E W
€799. Teenage Engineering +46 8 641 98 53.

Three different sequencers provide everything from simple repetitive patterns to Tenori-On-grade eccentricity. The most unusual is ‘Tombola’, dispensing welcome unpredictability from a rotating tombola tube. Notes are first cast into it by playing the keyboard. You then set their heaviness (loudness), bounciness, and the speed of the tombola’s rotation, before sitting back to enjoy the mayhem. Gaps can be opened in the tube wall, letting old notes spin off to freedom and make way for new ones to be added. Ideal for weird backgrounds and textures. For more regular grooves, there are two step-based sequencers, one of which is fairly similar to the sequencer in Roland’s SH101. First, you enter a sequence of up to 99 notes and then trigger it from the keyboard, applying swing or introducing patterns of gaps as it plays. Lastly, there’s

The OP1’s ‘Tombola’ sequencer is certainly idiosyncratic. More conventional sequencing is on offer elsewhere.

noises in the otherwise quiet output; these were traced to the USB charging process, and to suppress them I was given a preview of a forthcoming operating system featuring a switch that deactivates charging when necessary. There’s one last operating role to mention: that of possibly the most ostentatious nano-controller ever. It could be useful, though; the four encoders can be programmed for MIDI CC transmission and


January 2012 / w w w . s o u n d o n s o u n d . c o m


Soundhack Pvoc Kit

om Erbe’s Soundhack processors have been around since 1991, and are described on the company’s web site as “Essential tools for adventurous musicians and sound designers”. The original Soundhack program has collected many fans, from Trent Reznor to Ry Cooder, and its collection of filters and spectral manipulation techniques has been used on films such as The Matrix and The City Of Lost Children. Since 2000, Erbe has also been spinning off the technology behind Soundhack in various plug-ins. So far, there have been three bundles, all containing plug-ins based around similar themes: the Delay Trio, for example, contains a delay, a pitch delay and the enigmatically titled Bubbler. The Soundhack plug-ins are available for Mac, in VST, AU and RTAS versions, and as VST plug-ins for Windows.


Effects Plug-ins For Mac & PC
Soundhack’s powerful algorithms can provide high-quality pitch-shifting — or sonic mayhem!

Erbe’s new collection is the Pvoc Kit. This is a collection of plug-ins that use the Soundhack phase vocoder and granular synthesis algorithms to stretch time, shift pitch and distort phase. The phase vocoder works by sampling the incoming signal and dividing it into blocks; each block is, in turn, divided into different frequency bands. The number of frequency bands and the number of samples in each block are linked, so there

Soundhack Pvoc Kit $99
• Well thought-out and presented. • Very interesting and easy to use!

is a constant trade-off between frequency response and transient (time) response. The granular synthesis program also divides the sound into small segments or grains, but doesn’t divide it by frequency. Each plug-in has a ‘bands’ control that governs this ratio. At 44100Hz, the sweet spot is between 1024 and 2048 bands, giving you a smooth frequency response and a reasonably fast transient response; in other words, conventional pitch-shifting and time-stretching with relatively few artifacts. But where’s the fun in that? Increasing the band rate above 2048 will give you finer pitch resolution but at the cost of smearing the sound, reducing the resolution of the time transients. This is, as Erbe points out, great for making ambient washes and drone sounds. Setting the rate lower, meanwhile, will allow multiple harmonics to enter individual bands and will increase the amount of cross-modulation and frequency distortion. A world of more aggressive sounds then opens up. The first plug-in of the bundle is +pitchsift, a classic phase-vocoder-based pitch-shifter. Able to shift up or down by

up to four octaves, it has the ability to transform sound beyond recognition. In its ‘sensible’ setting, with bands set between 1024 and 2048 and pitch-shifts less than an octave, it behaves like a normal high-quality processor, although it does have the choice of two modes that give slightly different sounds and — be warned — have very different CPU needs. The first uses a bank of sine-wave oscillators, with each block of samples controlling one oscillator’s amplitude and frequency. The more bands, the greater the CPU drain, so a partial gate control is added, which removes harmonics below a particular threshold level. This harmonic gating can, at extreme settings, reduce the sound to just a small number of sine waves, with interesting results. The other method uses an inverse FFT (Fast Fourier Transform) algorithm to produce a similar result. When you select ‘MIDI vocoder’, the plug-in uses an incoming MIDI note to create a list of harmonics, the number of which is selected by the ‘MIDI harmonics’ control. The pitch-shifted sound is then compared to this list and the frequencies that lie between the

• At high band rates they can become very processor-hungry.

These plug-ins are well-designed and attractive, offering pitch- and time-based effects that are both practical at normal settings and creative as they approach the extremes!


January 2012 / w w w . s o u n d o n s o u n d . c o m

highest and lowest notes are shifted to the nearest harmonic of the MIDI notes being played. This effect varies from a harmoniser to a weird harmonic auto-tuner. The next plug-in is +spiral stretch. This is a multi-layered, real-time stretcher. Once again the sound

up to 11.8 seconds long (at 44.1kHz), either by sampling the input or loading in WAV or AIFF sound files. These can then be altered in many ways, stretched and manipulated by the plug-ins’ algorithms, They can be linked to tempo using the bpm control and altered over MIDI (all the plug-ins

is segmented and then layered, and, again, two methods are used: the phase vocoder and granular synthesis algorithms. Switching between the two can really help you decide which suits the material best. With the granular algorithm, you can also control the grain size and ‘shimmer’. Larger grain sizes will create stuttering sounds, and the shimmer adds a randomness to the grain distribution that’s easier

offer MIDI control of parameters, so are easily manipulated live).

Fun Fun Fun
So do they live up to expectations? Well, yes! The ability to deconstruct sounds and manipulate them is both useful and very enjoyable. The bundle takes very powerful algorithms and breaks them into simple, comprehensible plug-ins with controls that are task-specific and


“Live guitars sound amazing with Royers on the cabinets warm, natural, present and uncolored - and they stand out well in the mix. I use a mix of R-101's and R-121 Lives with Maroon Five. The band loves the natural sound they get in their in-ear monitors.”

to hear than to explain — the name ‘shimmer’ is a good start! The +phasemash plug-in is described as a “collection of simple transformations to phase difference and band assignment”, and is a great little phase-based sound-masher. Press the Scramble button and it randomly reassigns the bands. Take a sound and mash it up! Lastly there’s +pvocloop, which, as you will probably have worked out, is a time-stretching, pitch-shifting looper. It can loop four voices at a time, each

easy to use. Switching between the different modes makes it easy to find better ways of achieving more pleasing results, and often throws up new directions. I think this set has a home in any creative musician’s toolbox. The ability to manipulate sound live, in real time, will also suit these plug-ins to use in the performance arena for musicians and DJs. $ $99. W

Jim Ebdon

FOH Engineer - Maroon 5, Matchbox Twenty, Aerosmith, Annie Lennox


Royer Ribbons

w w w . s o u n d o n s o u n d . c o m / January 2012



Blackstar HT1R Valve Guitar Amplifier
Can a 1W tube amp really deliver a recordable tone? Blackstar think it can...

esigned and built in the UK, Blackstar’s HT1 combos take the concept of their popular HT5 valve amp and shoehorn it into a compact, 1W format. Two versions are available, one with reverb (the HT1R reviewed here) and one without, and both use Blackstar’s ISF (Infinite Shape Feature) EQ circuit to provide a surprisingly wide range of useful tones from just a single control. Separate amp head versions are also available. While a 1W amplifier might seem a touch under-powered, the HT1 can play loud enough to make you want to turn it down. More importantly, it can still sound sweet at lower levels, which makes it well-suited suited to home recording. While most small amps have a class-A output, the HT1 power amplifier is based around a low-powered ‘push-pull’ pull’ stage, using a single ECC82 (12AU7) dual triode. This helps its tonality to mirror what you might expect from a powerful stage amp, but at a much lower level. A solid-state driver stage before the power amp provides the


Blackstar HT1R $299
• Tonally flexible. • Good Brit-style overdrive. • Convincing spring reverb emulation.

• Level changes quite dramatically when the overdrive is switched in.

Because the ideal electric-guitar sound varies from uses to user, there is no best or worst, just something you like or you don’t. The HT1R seems happiest when recreating classic rock and pop sounds and records as well as many larger amplifiers, yet costs little more than a decent overdrive pedal.

necessary phase-inverter function. An ECC83 (12AX7) dual triode fills the role of preamp, in conjunction with another solid-state gain and clipping stage, as used in the popular Blackstar valve-based pedals. The reverb, on the models that feature it, is a very capable electronic emulation of a spring. The closed-back HT1 cabinet is built along the lines of a ‘grown up’ amp, complete with chunky strap handle. The eight-inch speaker delivers a well-balanced sound that, while lacking any really deep lows, largely avoids the boxy, nasal sound that betrays so many smaller combos, but you can also add an 8Ω extension speaker, which mutes the internal speaker when connected. There’s a speaker-emulated output jack for direct recording, and this also mutes the internal speaker. An additional MP3/line input allows outside sources to

be played through the amp, and while the speaker is hardly hi-fi (guitar speakers are deliberately not so, of course), it’s useful for jamming along to pre-recorded material or for practising. There’s no level control on this input, though, so the level must be adjusted at source. The front panel is fairly straightforward, with a gain control adjusting the amount of overdrive and a volume control setting the overall loudness. A ‘select’ button flips between clean and overdrive modes, so can be thought of as a basic version of channel switching, though the level change when you bring in the overdrive can be significant. The result is that the


January 2012 / w w w . s o u n d o n s o u n d . c o m

There are several low-powered valve amps available from the likes of Bugera, Laney, Fender and Epiphone, but each sounds different from the others!

gain control can take you from a nice jangly clean to a classic rock overdrive. My first impression of the ISF (Infinite Shape Feature) EQ control is that it sounds bright when turned anti-clockwise and warmer when fully clockwise, but there’s a lot more too it than that. Anti-clockwise suggests an ‘American’ tonality, whereas moving it slightly clockwise hints at a Vox-like vibe. As you move further clockwise, a more recognisable Brit-rock tone emerges, with a far fuller-sounding low end and less abrasive edge, though perhaps not quite as scooped as some people prefer. The Reverb knob sets the overall level of the reverb effect, which is otherwise preset. This emulates a spring reverb pretty accurately, and the fixed decay time is set to suit most genres. When using the speaker-emulated headphone output,

which can also be used for recording, the reverb is in stereo if a ‘stereo to dual mono’ Y-lead is used to connect to the recording system. With a mono lead connected to the output, it works quite happily, although of course you’ll lose the stereo reverb effect.

Despite its simple controls, this is a surprisingly flexible amplifier, and only its lack of deep lows gives away the fact that it isn’t a much larger beast. So many compact combos sound boxy but this doesn’t. It records well, and is capable of producing a fairly wide range of tones, courtesy of that ingenious ISF EQ knob. Importantly, it has the feel of a valve amp when you play. The eight-inch speaker can sound a hint gritty at some settings, but careful mic positioning or post-recording low-pass EQ helps avoid this. While the speaker-emulated output is perfectly usable, it doesn’t really match what you get from miking the amp, and as this amp sounds so plausible at such low volumes, there’s really no reason not to mic it. Similarly, although the reverb

sounds pretty convincing, making it perfect for practice, most of us have plug-ins that will do a better job, or produce a more appropriate reverb for the song, so that would be a better option when recording. Ultimately, a tiny combo amp like this will never sound quite the same as stage amp that’s allowed to run wild, partly because the smaller speaker tends to enhance certain aspects of the highs while suppressing the punchy lows. A larger extension speaker will get around this limitation, but Blackstar have to be commended for getting this diminutive amp to sound as big as it does. If you need a small amp for recording or practice, the HT1 ticks all the boxes, and it ticks them with style. $ HT1 (combo without reverb) $249; HT1R
(combo with reverb) $299; HT1RH (head with reverb) $249; HT408 (4x8 speaker cabinet) $249. T Korg USA +1 631 390 8737. E W W

w w w . s o u n d o n s o u n d . c o m / January 2012



NI Kontakt 5
Software Sampler
Native Instruments’ flagship soft sampler continues to improve with age. We look at what’s new in version 5.

An instrument open for editing, showing the new Instrument Bus strip. Here, Bus 4 has Transient Master, Solid Bus Comp and Solid-G EQ as insert effects, plus an effect send. The controls for Solid Bus Comp are shown. The Group feeding Bus 4 has the new two-pole Ladder filter inserted.

ative Instrument’s Kontakt has certainly been a trail-blazing success since its inception nine years ago, and has become the preferred format for a multitude of sample-based products. Its versatility, flexibility and ease of use make it a very attractive tool for beginners and serious under-the-hood geeks alike. From simple


one-shot sample playback to the creation of complex, scripted virtual instruments, Kontakt seems to be capable of pretty much anything to do with sampling. If you’re unfamiliar with the software and bemused by all the excitement, you can bone up on the story so far in the following SOS reviews: Kontakt 1 (August 2002); Kontakt 2 (July 2005); Kontakt 3 (January 2008); and Kontakt 4 (February 2010).

Just when you were wondering what more could possibly be added, Kontakt 5 delivers yet another bevy of sample-empowering goodies that you didn’t know you couldn’t live without. The new version remains cosmetically the same as version 4 apart from a minor change: in Instrument Edit mode, the green colour scheme is now replaced by a shade of pale brown that paint manufacturers might call ‘Weimaraner’.

New Filters
Kontakt 5 adds an impressive list of 37 new filters to the original 16 types, bringing the total to 53. New SV (State Variable) filters appear in the LP, HP, BP,

January 2012 / w w w . s o u n d o n s o u n d . c o m

New Filter Types
Low-pass: • SV LP1 • SV LP2 • SV LP4 • Ladder LP1 • Ladder LP2 • Ladder LP3 • Ladder LP4 • AR LP2 • AR LP2/4 • Daft LP High-pass: • SV HP1 • SV HP2 • SV HP4 • Ladder HP1 • Ladder HP2 • Ladder HP3 • Ladder HP4 • AR HP2 • AR HP4 • AR HP2/4 • Daft HP Band-pass: • SV BP2 • SV BP4 • Ladder BP2 • Ladder BP4 • AR BP2 • AR BP4 • AR BP2/4 Peak/Notch Filters: • SV Notch 4 • Ladder Peak • Ladder Notch • Legacy BR4 • Multi Filters • SV Parallel LP/HP • SV Parallel BP/BP • SV Series LP/HP Effects Filters: • Formant I • Formant II

Peak/Notch and Multi categories. NI describe them as the new standard for Kontakt, being cleaner than the legacy filters. The new Ladder filters, which are based on those found in older synths, also offer improved algorithms, together with a High Quality option that applies oversampling at the cost of slightly increased CPU overhead. Especially interesting are the AR (Adaptive Resonance) filters; these adjust the amount of resonance according to the amplitude of the input, acting as a sort of auto-limiter. You can crank the resonance up full and sweep away to your heart’s content, safe in the knowledge that your speakers and eardrums will suffer no harm. In stark contrast, the resonance of the Daft LP and HP filters (borrowed from NI’s Massive soft synth) can be pushed into self-oscillation. By tuning the cutoff frequency to track the keyboard over an evenly tempered scale, you can easily achieve the classic ‘singing filter’ effect of analogue synths. Four Peak/Notch filters are a welcome addition, useful both for fine surgical EQ and for creating
Amongst the new filters are three State Variable Multi types: LP/HP Parallel, BP/ BP Parallel and LP/HP Serial, each with fully adjustable bandwidth separating the cutoff frequencies of the filter pairs. Shown also is Formant 1, whose Size parameter acts like a gender control, changing the size of the ‘vocal tract’.

effects similar to phase-shifting when modulated by an envelope, LFO or the mod wheel, for example. Of the four varieties, SV Notch 4 has a bipolar character: resonance values above 25 percent produce a peak, while lower values produce a notch. Also of special note are two new Formant filters that simulate the vocal tract, allowing you to morph through the vowel sounds all the way from ‘ooo’ to ‘eee’. These will bring a smile of recognition to anyone familiar with the Delay Lama plug-in instrument by AudioNerdz (the animated singing Tibetan Monk with the inscrutably mobile Roger Moore eyebrows). Whereas Delay Lama was monophonic, had one fixed waveform, no envelopes or filters, and sounded (and I say this with the greatest affection) untameably raucous, Kontakt offers all the facilities you need to sculpt any sample into a musically useful sonic chatterbox. The two variations of Formant filter differ in that Formant 1 is the more severe of the two, with a sharper, more focused effect. For a list of the new filters, see the ‘New Filter Types’ box.

NI Kontakt 5 $399
• Studio quality EQ, compressor and Transient Master effects. • Sixteen-way bus routing on each Kontakt instrument. • Improved Time Machine Pro engine. • Pristine sound quality and solid reliability. • Free additional downloadable Retro Machines 2 library.

• It’s totally indispensable. Oh, but that’s a Pro...

Kontakt 5 adds 37 new filters, higher quality time-stretching, two retro drum-machine engines, studio-quality EQ, compressor, transient modelling and tape saturation effect processors, and a highly flexible instrument bussing system to an already world-beating sampler. What’s not to like?

Sampler Engine Enhancements
The sampler engine receives three new playback modes. The most significant of these is Time Machine Pro, a real-time, high quality time-stretching algorithm that gives you independent control over sample playback speeds and tuning with fewer artifacts than the existing two modes (Time Machine and Time Machine 2). Both of those were capable of producing good results, but it could be hit-and-miss, depending on the type of material they had to work with. Looped samples often came off badly (loop points were usually thrown off kilter) and even mild speed variations could sound grainy and distorted, especially when slowing things down. Naturally, I tested Time Machine Pro with a variety of material,

and found the results to be variable but generally impressive. Most complex stereo material (such as full-blown mixes) could take around a speed variation of around 20 percent before acquiring a metallic ring (slower) or a fluttering quality (faster), with sustaining, orchestral music being the most susceptible to artifacts. Percussively oriented material coped a little better with wider speed variations. Nevertheless, the step up in quality from the older Time Machine modes is patently obvious. TM Pro should prove very useful for fine-tuning the duration of sampled special effects without altering their essential character, and since the speed parameter can be sync’ed to the host DAW’s tempo, the speed of pre-sampled musical phrases will always remain in step with any variations in track tempo. TM Pro offers two-, four- or eight-voice polyphony, and also provides a ‘Pro Mode’ option, giving control over the formant character of the time-stretched samples. The new SP1200 Machine and MPC60

w w w . s o u n d o n s o u n d . c o m / January 2012


N I K O N TA K T 5

Machine engines aim to reproduce the sound quality and playback characteristics of two vintage drum machines (made by Emu and Akai respectively). Little clarification is given about these modes except that they degrade the playback quality of samples by replicating the sample rate and frequency range of each drum machine, as well as “changing the way Kontakt changes the pitch and basic handling of the sample playback engine”, according to the PDF manual — a somewhat cabalistic explanation. Curious to discover more, I compared the effect of the two engines on drum sounds. The MPC60 engine reveals a significant high-frequency emphasis; everything sounds super-bright. By contrast, the SP1200 engine’s 12-bit, 26kHz emulation shaves just a gnat’s off the top end. However, I didn’t observe any obvious change in pitch behaviour, as alluded to in the documentation. Comparing their effect on instrument samples, it became apparent that the loop points of looped sounds were being slightly altered — a click here, a buzz there. Even more interesting were the strange additional harmonics creeping in when playing back sinewave samples at certain pitches, notably with the SP1200 engine. I can only assume there’s some sort of aliasing going on, which no doubt gives the SP1200 its signature sound. The SP1200 and MPC60 engines are clearly intended for one-shot drum samples rather than instrumental sounds. Aside from the bright quality of the MPC60, any other significant tonal differences when applied to drums (aliasing notwithstanding) were not overtly obvious to me, despite donning my best bat’s ears. I’ll doubtless be taken to task over this by hordes of angry 12-year-old retro drum-machine aficionados. As is the case with all the non-DFD engines, the entire sample data is loaded into RAM when any of these playback modes is used, so be wary if applying

Kontakt 5 Library
Like its version 4 predecessor, Kontakt 5 comes with a 43GB sample library comprising a generous compendium of instruments in seven categories: Band, Choir, Synth, Urban, Vintage, World and, of course, the essential Orchestral category culled from the famous Vienna Symphonic Library. Further details of Kontakt’s library can be read in the SOS Kontakt 4 review mentioned elsewhere. In addition to this, NI’s Retro Machines Mk2 library is available for free download once Kontakt 5 has been registered online. This chirpy assemblage of analogue synths, antique electronic pianos and string machines complements the existing Synth and Vintage categories, with the inclusion of some natty parameter morphing, chord sequencing and arpeggiator trickery into the bargain. Kontakt 4 owners may wonder if they need to install the version 5 library, because of both libraries’ apparently similar content. I asked NI, who replied that there may be a smidgeon of extra sample data in the K5 library and some instrument patches may have been reworked slightly, although they didn’t cite specific examples. Some people may feel it’s not worth the installation time and another 43GB of arguably redundant disk space to discover what the differences are, if any, and that’s fair enough. The K4 library can still be accessed via K5’s file browser, even though it no longer shows up under K5’s Libraries tab. Just be aware that if you overwrite any K4 patches from within K5, they’ll no longer be compatible with K4. And no, I confess I haven’t inspected all the of 1000plus K5 instruments looking for any differences!

them to all Groups of a RAM-hungry instrument at once.

Studio Quality Effects
Four new effects have joined the Kontakt fold. Three of these are available to buy as separate plug-ins, so it’s a real bonus to have them included as part of Kontakt 5. First up is the Solid-G EQ, presumably

The Solid Bus Comp appears to be modelled on the SSL G-series Bus Compressor and, again, is a vast improvement on the existing Kontakt effect. Controls for Threshold, Attack (0.1ms to 30ms), Ratio (1:1.5 to 1:30), and Release (100ms to 1600ms + Auto) provide similar functionality to the SSL model. Parallel compression is possible

The Solid Bus Compressor, with its handy gain reduction meter on the right. The Mix knob adjusts from a dry to a fully compressed signal, with parallel compression occurring at all settings in between.

modelled on SSL’s G-series console EQ. This four-band parametric design is a great asset, having switchable shelving or bell curves on the lowest and highest frequency bands. Kontakt EQs have always lacked high- and low-end shelving until now, and using wide Q bandwidths on the high and low frequencies as a substitute has always seemed unsatisfactory and lacking in sufficient subtlety. Solid-G fills this gap perfectly and sounds, to my ears, cleaner and more ‘musical’ than the standard Kontakt type, adding gloss, warmth or weight just where it’s needed.

with a simple tweak of the Mix knob, Makeup Gain brings the post-compression level back up, and a handy gain-reduction meter gives you a visual guide to how much you’re pushing things. This compressor sounds good on pretty much anything, and really comes into its own when used with drums. The Transient Master is not a male instructor on a temporary supply-teaching job, but a tool for sculpting the dynamic shape of percussive sounds. There are two principal sculpting controls, Attack and Sustain. Positive values of Attack accentuate transients and negative values soften them. Soggy snares can be given added snap and
The controls for the Solid-G EQ. The buttons below the LF and HF bands toggle between shelf and bell curves; both LMF and HMF bands have variable bandwidth (Q) knobs.


January 2012 / w w w . s o u n d o n s o u n d . c o m

N I K O N TA K T 5

The following software samplers all feature detailed editing and sound-manipulation capabilities. Their included sample libraries vary in size, with Independence having perhaps the largest at 70GB: Emu Emulator X3 (PC only); Apple EXS24 (only for Logic Pro on Mac); Steinberg Halion 4; MOTU MachFive 2; IK Multimedia SampleTank 2; Yellow Tools Independence Pro; Avid Structure (Pro Tools only).

they are aware of this bug, and that it will be fixed in a subsequent service update.

System Requirements
• Windows 7 (latest Service Pack, 32-/64-bit), Intel Core Duo or AMD Athlon 64 CPU, 2GB RAM. • Mac OS 10.6 (latest update) or 10.7, Intel Core Duo CPU, 2GB RAM. • Native 64-bit support for stand-alone and plug-in versions. • 1GB free disk space or 48GB for complete installation. Kontakt 5 works as stand-alone, VST, Audio Units and RTAS (Pro Tools 8 or higher) versions. Although Windows 7 is a quoted requirement, Kontakt has been running on Windows XP (service pack 3) on a 32-bit machine throughout the course of this review, with no problems at all.

Instrument Buses
Perhaps one of the most useful additions to Kontakt, Instrument Buses are to be found nestled between the Amplifier module and the Insert Effects strip. They provide the solution to a perennial problem: how can you easily apply a single set of treatments to a collection of Groups without having to apply them to each Group individually? This is best illustrated with a typical scenario: processing the various elements that make up a drum kit. Take the snare for instance, often represented by numerous articulations — centre hits, edge hits, rim shot, side stick, flam, snares off... you get the picture. Articulations are often assigned to their own Groups, each of which may have its own unique level setting, or possibly an effect of some kind. To adjust the overall level of all the snare Groups, you’d previously have had to insert a Gainer as the last insert effect

vigour, while overly spiky guitars can be tamed to sit less prominently in the mix — without the need to alter their levels. Positive values of Sustain lift the post-transient level, bringing up the body of the sound, while negative values reduce and effectively shorten it. Transient Master is great for rescuing otherwise unusable samples; that plinky piano can be afforded the fullness it lacks, overly ambient drum samples dried out, and polite snares pumped up to in-your-face proportions. The Tape Saturator emulates the effect of over-recording to tape, and

The Transient Master controls. The Smooth button can provide better results with non-percussive material.

is not dissimilar to the Saturator effect from previous Kontakt versions. It adds subtle warmth at low values, increasing to aggressive distortion at higher values with greater amounts of compression. The gentle growl it can impart to organ samples is particularly good. The Warmth control boosts or cuts low frequencies, while the HF Roll-off attenuates frequencies above 12kHz. Oversampling can be applied with the High Quality button. At the time of writing, when Transient Master is applied as a Group insert effect, it doesn’t function correctly. Consequently, it should be applied either as a main Instrument insert effect, or an Instrument Bus effect, where it works perfectly. A quick call to NI confirmed that

of every Group, and adjust them all in unison. Similarly, adding effects such as a compressor and an EQ would involve replicating those effects numerous times across all the Groups — hardly elegant or CPU friendly. All this can be done much more efficiently using Instrument Buses. Sixteen Buses are available to each Kontakt instrument and appear as destinations under each Group’s output assign button. Once all the snare’s Groups have been identified and selected for editing, simply route them to an Instrument Bus. Now the entire Bus’s level and pan can be controlled easily from its own controls. Each of the 16 Buses also has slots for eight insert effects, so you only need to add a single compressor to the snare Bus,

for example, to process all its articulations. Unlike Group insert effects, which operate polyphonically (ie. the effect is calculated for each individual note), Bus insert effects operate monophonically, just as they would if you were using outboard gear. Effect sends can also be added to each Bus’s inserts, so adding reverb to the contents of a Bus is easy. The output from each Bus returns just before the main instrument insert effects by default, but there is also an option to bypass those insert effects, keeping any Bus you choose (and its own effects) entirely separate from the main effects.

Armed with the characterful Solid-G EQ and Solid Bus Comp compressor, the Transient Master and the excellent new Instrument Bussing system, it’s now more feasible than ever to produce high-quality tracks entirely within Kontakt, given an adequately wide-ranging sample library. Sample library developers, too, will welcome Kontakt’s extended Script Processor instruction set, which includes a MIDI file player. These improvements might not get every Kontakt user champing at the bit, but for anyone with a passion for tinkering, tweaking and fine-tuning their sounds, I’d say this is a must-have upgrade. Besides, with increasing numbers of Kontakt-based libraries being written specifically for version 5, upgrading may swiftly become a necessity! $ $399. W

The Time Machine Pro engine with Pro Mode active. Pro Mode allows the spectral envelope (formant) to be adjusted with the Env.Order and Env.Fact knobs. The button at the right selects two-, four- or eight-voice polyphony.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Introducing Native plug-ins from Eventide

Dynamics processor with an attitude!

2016 Stereo Room
Classic stereo reverb

H3000 Factory™

One Alsan Way

Little Ferry, NJ 07643

Eventide and Harmonizer are registered trademarks; Omnipressor and H3000 Factory are trademarks of Eventide Inc. ©2012 Eventide Inc.


Magneto Audio Labs VariOhm
Microphone Impedance Converter

n the days of yore — and we’re talking 40 or 50 years ago, really — microphones used to be designed to operate in a ‘matched impedance’ environment with their preamplifiers, and usually at quite low impedances, such as 30Ω. This mode of connectivity is all about maximising the transfer of power from the source to the destination, and essentially came from the practices employed in the telephone industry upon which pro audio was conceived and developed. The benefit of the impedance-matched interface in the telecoms world at that time was that the power generated by a telephone’s microphone was passed in the most efficient way possible to another telephone’s earpiece in a predominantly passive system. However, eventually common sense prevailed in the professional audio world, and microphone interfaces evolved into a matched-voltage configuration. We’re not interested in passing power around the place — everything is powered and active anyway — and all we need to do is make sure the alternating voltage generated by


If you’re searching for the perfect match of mic and preamp, this useful gadget could help you achieve it...
the microphone, which represents the audio signal, is passed as accurately as possible to the microphone preamplifier. To that end, we generally use output devices with low source impedances, and detect the voltage generated by the source with a relatively high-impedance destination. The general rule of thumb is that the destination should have about 10 times the impedance of the source. Consequently, the vast majority of microphones have output impedances of 150-200Ω, and most microphone preamps have an input impedance of 1500-2400kΩ. tonality and the sense of space and room that they capture. A lot of modern capacitor microphones are equipped with active electronic output stages, of course, such as Neumann’s TLM designs, and in these cases the input impedance of the preamp has little effect on performance. However, quite a lot of capacitor mics — especially the older ones — employ output transformers, as do a lot of dynamic microphones (and those that don’t feed the output straight from the voice coil anyway). In these cases, the input impedance of the preamplifier can have quite a pronounced and audible effect because it determines the ‘loading’ seen by the voltage generator — whether that’s the microphone capsule itself or the impedance-converter electronics — and that often affects how the microphone performs! Typically, changing the loading affects the tonality because it alters the resonance peaks in the coupling

Why Change Impedance?
That’s all fine and dandy in theory… but the real world likes to throw in a few curve-balls now and then. Some long-serving microphones, like the classic Shure SM57, were originally designed to work with 1960s mic preamps presenting a 300-600Ω input impedance, and forcing them to work into higher impedance loads does affect their


January 2012 / w w w . s o u n d o n s o u n d . c o m

transformers, and it often also affects the sense of space that the microphone picks up, almost like a very transparent compressor in some ways. The problem is that the specific effect of changing output and input impedances on microphones is complex and usually quite difficult to predict. Moreover, because different mics and different preamps have different impedances, plugging a mic into a different preamp (or a different mic into the same preamp) can often produce a surprising change in sound character, beyond that which might be expected from the different mic or preamp alone. Of course, experienced recording engineers and producers usually discover this effect for themselves early on in their careers, and that’s why many often have personal favourite combinations of particular preamps which they use with particular mics. It’s also why some equipment tends to sound different to others. For example, although the ‘standard’ input impedance for a mic preamp is between 1500Ω and 2400Ω, most high-end mic preamps tend to have significantly higher input impedances. The Millennia HV3C preamp has an input impedance of nearly 7000Ω, for instance, and the Grace Design m201 is over 8000Ω. AEA’s own ribbon mic preamps present an impedance of over 18,000Ω, and most of the later Rupert Neve-designed preamps present well over 5000Ω. In general, the higher the impedance, the less load on the mic and the better the overall performance — especially in terms of transient distortion, and that seems appropriate for high-end applications. But, as I’ve already mentioned, a lot of classic mics were designed for a different age and a different way of interfacing, and they do sound different when presented with lower impedances. As a result, over the last decade or so it has become increasingly common to find mic preamp designs with switchable input-impedance options. Many end-users find this a useful facility, as it allows some additional adjustment of tonal colour in a very different way to a normal EQ. Again, it’s often hard to predict exactly

Stereo Large Diaphragm Tube


Magneto Audio Labs VariOhm $289
• Entirely passive. • Easy to hook up between mic and preamp. • Useful phantom-power defeat and polarity inversion facilities. • Allows the audible effects of impedance loading to be explored quickly, easily and safely.

• Adding anything between mic and preamp potentially increases noise and distortion.

An old idea made very practical, and in a way that allows it to be used as an easy add-on to a fixed-impedance preamp. And if the impedance variations don’t deliver the sound you’re after, maybe the nice transformers it inserts into the signal path will anyway!

w w w . s o u n d o n s o u n d . c o m / January 2012



As the VariOhm is a passive device, its rear panel isn’t exactly feature-packed!

what will happen when changing the impedance setting, but usually one particular configuration sounds distinctly more appropriate than another for a given microphone and application. Of course, that doesn’t help anyone who has a preamp with a fixed input impedance… and that’s where Magneto Audio Labs come in, with their interesting VariOhm. This is an entirely passive device — no mains or battery power required — that plugs between a microphone and a preamp, and provides the ability to change the input impedance, so that the user can find a combination that sounds appealing.

The VariOhm is a half-rack width, 1U-high black box made from extruded aluminium. The rear panel has two XLR connectors to accept the signal from a microphone and to pass it on (at microphone level) to the microphone preamp. The front panel isn’t much more complicated either! A ‘magic-eye’ push-button provides a hard bypass mode, while two large toggle switches allow the phantom power pass-through to be disabled and the signal polarity to be inverted. (The normal position for both switches is down, providing correct absolute polarity and passing phantom power). The most important control is the central rotary switch, which allows the impedance seen by the microphone to be changed between any of six different settings: 2400Ω, 1200Ω, 600Ω, 300Ω, 200Ω and 50Ω. Operation is as simple as plugging the microphone into the VariOhm, connecting the VariOhm output to the mic preamp, and then setting the controls as required. The

Lots of preamplifiers include variable input-impedance options, but I don’t know of similar stand-alone mic-impedance switchers.

phantom-power switch is a useful facility that provides an additional level of protection for a ribbon mic against phantom power, and the polarity inversion is handy if your preamp doesn’t already have that facility... but in general, I doubt these two switches would be used very much. However, the bypass and impedance switches will be used extensively. I found that I tended to switch quickly between the different settings to see which I favoured most, and then used the bypass switch to provide a ‘sanity check’, by comparing the standard mic-to-preamp sound character with whatever alternative impedance combination I had dialled up with the VariOhm. Of course, adding anything into the signal path between the delicate output of a microphone and the sensitive input of a preamp risks introducing noise and distortion. However, I didn’t notice any significant ill-effects — although it’s sensible to keep the VariOhm away from strong mains transformers to avoid unwanted hum pickup — and the tonal benefits completely outweigh any theoretical technical degradations. Internally, the VariOhm is very simple, but nicely put together. The front-panel controls are mounted on a small, vertical glass-fibre circuit board connected to the main horizontal circuit board on the base of the unit via a couple of short ribbon cables. The circuitry basically comprises two metal-screened microphone transformers, each with multiple taps on their windings, and the few switches I’ve already mentioned. The transformer windings are connected to the front-panel switch in such a way that they can be arranged in various series and parallel combinations to achieve the required impedance loading. It’s that simple — but the devil is in the detail, of course, and Magneto’s Bob Reardon has obviously done a nice job in finding good transformers and putting the VariOhm together. You can see a brief interview

with Bob, talking about the VariOhm, on SOS TV, recorded at the NAMM 2010 exhibition ( news?NewsID=11859) Besides having the ability to present different load impedances to your microphones, the VariOhm is also a microphone transformer in a box! As we all know, transformers tend to add a little character or ‘iron’ to the sound of a microphone, and in this case the transformer sound is pretty subtle — but it’s definitely there. So if you have a modern, transformerless capacitor microphone and a modern, transformerless mic preamp, the overall sound might seem to be slightly sterile. Inserting the VariOhm into the signal path might be just the ticket for putting a little harmonic richness and transformer ‘iron’ back into the sound, and that’s certainly an additional appeal of this device. Given my own tastes in equipment and acoustic recording, I tend to choose microphone preamps that have a relatively high input impedance. However, my Focusrite ISA428 does have switchable impedance modes and I do make use of them, particularly when using dynamic microphones and capacitor mics with output transformers — because doing so often reveals sonically interesting tonal variations and colours that are sometimes musically useful. The VariOhm’s impedance options focus on the bottom end of the impedance spectrum, and in my experience that’s where the strongest colorations and sound characters tend to be found. There’s no doubt that this is an interesting and unusual box that does exactly what it claims, and is well worth checking out. $ T E W W
$289. Sonic Distribution US +1 617 623 5581.


January 2012 / w w w . s o u n d o n s o u n d . c o m


delivers breathtaking cinematic percussion right out of the box. Created by Heavyocity to give your music a tense, epic edge, DAMAGE provides an arsenal of deeply-sampled drums, crushing loops, and smashing one-shots, all recorded in brutal detail. From battered dumpsters and exploding cars to edgy electronics, the entire gamut of modern cinematic sound comes together in this uniquely playable and inspiring instrument. Deep, thrilling and intense – DAMAGE never sounded so good.


Making Music On The Move
IK Multimedia iRig MIDI
MIDI Interface for iOS
K Multimedia’s iRig MIDI is not the only MIDI interface for iOS devices that’s currently available, but it does have some unique features. What sets the iRig MIDI apart from its competitors, such as the MIDI Mobilizer by Line 6 and Yamaha’s iMX1, is the presence of a MIDI Thru socket, and the fact that you can charge your iOS device while the MIDI interface is in place. The interface comes supplied with two 1.6m mini-jack to five-pin MIDI cables for connecting instruments and controllers, as well as a very short mini-USB cable to connect to a charging adaptor. Apparently the short length of the charging cable was determined by Apple, in order to ensure that sufficient charge could be guaranteed to reach the device. The iRig MIDI also features two LED indicators to indicate active MIDI In and Out connections. The iRig MIDI does feel a little flimsy (don’t tread on it) and, as with all interfaces that use the dock connector, it does come with the worry that something might snap if stressed while in use. I also noted a similar gripe to one I had with the Apogee Jam: the interface doesn’t click into place (unlike the


The iRig MIDI offers MIDI In, Out and Thru, as well as charging via USB while the interface is in use.

original iPhone charging cable) and feels as if it might work itself loose in a live situation. Plugging in the iRig MIDI will prompt the user to download the free iRig MIDI Recorder app. This has the ability to record and play back freely-played MIDI sequences and SysEx dumps (remember those?). While it’s not a particularly inspiring app in itself, it will allow you to troubleshoot the device and install firmware updates. The packaging also advertises that SampleTank Free comes free with the iRig MIDI. Well, it’s free anyway, but nonetheless gives you a basic range of sounds to play with if you haven’t bought any other apps yet! The paid version and sound packs that can be purchased in-app are doubtless impressive, but they are certainly not free. Although I can already send MIDI data into the iPad using a compatible USB MIDI keyboard controller — like the latest version of M-Audio’s Oxygen 25, for example — and Apple’s Camera Connection kit, the iRig MIDI allowed me to connect a great variety of legacy MIDI equipment. MIDI Thru and MIDI Out also worked flawlessly with apps that output MIDI data, such as Pianist Pro and Little MIDI Machine. If you’re still using the on-screen keyboard in the GarageBand app, then using an external keyboard to play the built-in instruments will reveal an extra dimension of touch sensitivity you never knew was there! While it isn’t cheap considering the build quality, you’ll find that if you want to use an external MIDI device that doesn’t work via the Apple Camera

Sampletank Free is available for use with iRig MIDI, and has a variety of instruments built in to get you started.

Connection kit, you wish to use MIDI Thru and you’d like to charge your device while using MIDI connectivity, the iRig MIDI is the most sensible choice currently on the market. Mike Watkinson $ $69. W

Fostex HPP1
Portable DAC & Headphone Amplifier for iOS
n the land of digital-to-analogue convertors (DACs), USB is becoming increasingly popular: many of the DACs built during the last few years will have a USB input, and it’d be odd if USB connectivity weren’t at least an optional extra. I think this is a great idea for a number of reasons, not least because it allows the DAC to be used in portable setups, especially if the DAC includes a headphone amp too. Apple’s iOS devices transfer data in a different way to standard USB devices, so plugging a 30-pin to USB Apple dock cable into a standard USB DAC simply won’t work. For this reason, Fostex have released the HPP1, a portable DAC and headphone amp for iOS devices. There’s no denying that the HPP1 is a quality product. It’s lightweight, yet feels nice and solid and is made of aircraft-grade aluminium. It comes with a rugged soft case, as well as a USB cable for charging



January 2012 / w w w . s o u n d o n s o u n d . c o m

the internal battery and a short USB-to-dock connector for plugging in your iOS device. It’s compatible with all the most recent iOS products, but do check the Fostex site to make sure before you buy. The rear panel hosts a filter switch, a three-position gain switch, an analogue line-out and an S/PDIF ‘thru’. The front panel has the volume knob, an analogue line-in, the main headphone output and a USB connection for the iOS device. The volume knob also acts as the power switch, activating with a satisfying click. The DAC can’t charge your iOS device, so you’ll find that while you’re using it, both the HPP1 and sound source will be running out of juice unless you plug the HPP1 into USB power, in which case your iOS hardware will give up on its own. As is becoming more popular with DACs, there are multiple settings for the reconstruction filter. Filter 1 is a steep linear-phase filter, similar to the kind found in most portable devices, while filter 2 uses a more sophisticated minimum-phase design. Sound wise, setting 2 is superior, especially in the high-mid and high frequencies. It’s very revealing, with much more separation between instruments and sounds. Minimum-phase designs like filter 2 exhibit no pre-ringing, unlike linear-phase designs, the audible result being that the timing and separation of sonic elements tends to be clearer and more noticeable. I found the artifacts of Filter 1 more flattering to the music, especially when listening casually to less spacious mixes, as they blur and disguise sonic imperfections. Filter 1 can be comforting, but isn’t as suitable for production or critical listening, as it’s far more detrimental to sound quality than the practically undetectable phase discrepancies of a well-designed minimum-phase filter. As you can probably tell, filter 2 is my desired setting for the most part, and my descriptions that follow are from using the HPP1 with this setting. At one point I had some issues with little pops and clicks in my headphones, which

The Fostex HPP1 is small, light and sturdy, drawing its power from an internal battery that can be charged via USB.

I solved by switching the host device off, then powering it on again. This little glitch only happened once during my review but I’ve seen it mentioned online, and it’s always solved in the same manner, since it’s caused by the host device, not the HPP1. Also, if you’re using the HPP1 with the iPhone it will pick up some GSM interference from phone, GPRS and EDGE transmissions. Listening to some tracks I know well, like ‘Never Going Back Again’ from Fleetwod Mac’s Rumors, was actually quite intimidating when using the HPP1! The attack and weight of each finger-picked note was evident, and every instrument was separated immaculately from the next. Likewise, listening through the HPP1 to recordings I made using Multitrack DAW for iPhone, I could hear the changes I was making with built-in effects much more accurately than when I used the iPhone’s headphone output. The HPP1 has a little emphasis on the low end, but it’s a very smooth emphasis that’s not at all overbearing. Meanwhile, the ultra-highs are also quite smooth, and a tiny bit lacking in sparkle at the very highest registers. These are extremely minor notes, though, and my main point of comparison for the HPP1 was the Becnhmark

DAC1, which carries double the price tag! The sound produced by the HPP1 is totally ‘grain-free’ in the high-mid frequencies, which can be a problematic range for DACs: cymbals and hi-hats won’t be shearing your ears off unless that’s what’s in the recording. The stereo field is wonderfully round and three-dimensional too, and gives a real sense of live performance and space. Just like the Benchmark DAC1 I use for my main monitoring, the HP P1 really does show up over-compressed pop and rock music as being a bit dead and lifeless. It’s very honest, unlike a flattering and enjoyable consumer DAC, which is precisely what most of us interested in pro audio want Despite the excellent conversion quality, there are a couple of cons to the HPP1. The most obvious is the price: it does sound brilliant, and if it had an S/PDIF or standard USB input as well as an iOS input, it could be your main DAC for monitoring, headphone amplification and portable iOS duties. This would make it a bargain of sorts, but as it is it only works with iOS devices. There’s no questioning the quality of the HPP1 as a DAC and headphone amp: it’s top class. The minimum-phase filter setting is pristine and revealing, and the three gain levels allow it to drive headphones of all kinds, from sensitive IEMs to large and inefficient Sennheiser HD650s. Whether or not the HPP1 is right for you will depend on how serious you are about using iOS devices for audio, and I can’t help but be a little frustrated that such a lovely sounding, lightweight and sleek piece of kit is limited to iOS input. It sounds nice enough that if it had a standard USB or S/PDIF input as well, I’d take it everywhere! The HPP1 is a significant investment, designed for those who are very serious about iOS production or listening. As the quality of applications for iOS increases and we head towards the inevitable merging of iOS and Apple OS, more and more users may find such a device handy to have around. J G Harding $ $795. W www.americanmusic

The HPP1 is supplied with a short dock cable for connecting your iOS device to the USB port on the front.
w w w . s o u n d o n s o u n d . c o m / January 2012



TK Audio BC1 MkII
Stereo Compressor-Limiter
The concept behind TK’s BC1 is simple: take a classic bus-compressor design and improve on it.

itting beside me as I write this review is a second-edition (1990) SSL G-series mixing console manual, and in its ‘applications guide’, two quietly tucked-away lines suggest: “finally, before you commit the mix to tape, try the compressor on the main Quad output”. That’s all they thought to include about how to use what’s now considered a classic piece of gear. With a minimum ratio of 2:1, this compressor could be quite savage, so at that time most engineers wouldn’t have considered tickling the meters with more than


TK Audio BC1 MkII
• Includes both an external side-chain input and a switchable side-chain HPF. • Wet/dry blend control. • Greater range of ratios available than on the SSL that inspired it. • Good build quality. • Attractively priced.

2-3dB of gain reduction on the mix bus — which would provide the ‘glue’ for which the device has since become famous. You could patch the compressor into other channels, but you already had a compressor built into each channel, so even the temptation to use it as, say, a drum-bus compressor was less than it might be today, when we’re working without consoles and often spend time searching for the ‘right’ compressor for each job in a mix. SSL now make rackmount and modular versions of this compressor (and very nice they are too), but in recent years there have been plenty of hardware imitations and software models that aim to offer the same sonic result. A select few designers, though, have taken the original design and then tweaked and augmented it, either to improve the technical performance or to add useful functionality that increases the compressor’s versatility. And this is where TK Audio come in, with their BC1 MkII.

In its aesthetics — the clean, uncluttered layout of the 1U front panel, and the white-on-black gain-reduction meter — and in its compression character (of which more later), this stereo compressor-limiter quite clearly tips its hat at the SSL G-series, but designer Thomas Kristiansson has made several thoughtful additions which extend its functionality and make it rather more versatile. As with the BC1 MkI, the MkII includes a built-in switchable (on/off) 6dB/octave

• None that I can think of at this price.

The BC1 is an excellent take on the basic SSL G-series compressor design, and one that’s different from the usual clones out there. Think of it as a well behaved, more controllable G-series bus compressor and you’ll have a good idea of what the BC1 offers.

side-chain high-pass filter, with a turnover frequency of 150Hz. There’s also an external side-chain input, and both features are useful in shaping the side-chain signal that dictates when the gain-reduction circuit operates. More obvious additions to the G-series concept are the wet/dry blend control, which gives you instant access to the world of parallel compression, and two new ratio settings in addition to the 2:1, 4:1 and 10:1 of the SSL: 1.5:1, and ‘hard’. The threshold control runs from -20 to +20 dB, the attack from UF (‘ultrafast’) to 120ms, with eight available settings in total (two more than SSL’s equivalent X-Logic processor), and the release control offers 50, 100, 300 and 600 ms and 1.2s settings, as well as an automatic release. The 1:5:1 ratio and the blend control instantly make this compressor suitable for more subtle applications than the SSL. The lower ratio enables you to bring the threshold down further, so you can gently squeeze things, and the blend means that if you take things too far you can shift the balance towards the dry sound for a more natural result. The ‘hard’ setting, on the other hand, enables you to push the BC1 into aggressive limiting territory. Finally, a switchable L+R option sums the two channels’ side-chain signals to mono before hitting the threshold detectors, which increases sensitivity to centre-panned sources by 6dB.

What’s New In MkII?
The Mk1 version of Kristiansson’s design has been on sale for a while now, and


January 2012 / w w w . s o u n d o n s o u n d . c o m

compressor from an unbalanced insert send/return loop). Like many SSL-inspired designs (including, I believe, the current SSL XLogic version), the gain-reduction element in both the MkI and MkII BC1 designs is based on a pair of THAT Corporation 2181 VCA ICs. These THAT chips are placed in a Dbx 202 emulation circuit, and in the MkII TK have tweaked this circuit, resulting in slightly lower noise figures.

On Test
To test the BC1 MkII, I hooked it up to my DAW and ran several different sources, including single tracks and stereo stems and mixes through it, to see how it performed, and I was reminded of what

of switching in the high-pass filter is a cleaner, and yet more solid, hard-hitting drum sound. The kick sound and the low end of the snare seem to suffer less, and the signal as a whole doesn’t tend to pump in time with the kick so much. It’s a sound I like. My only criticism here is that I’d have liked to see more side-chain filter frequencies available on the front panel, but you always have the external side-chain input for that. Finally, for sanity-checking purposes, the switchable high-pass filter and both the compressor and hard bypass buttons were useful.

The SSL G-series compressor is a tried, tested, and well-loved design, and the BC1 builds on it in a useful way. The

my initial tests were carried out with that version. Just before I was due to write up the review for SOS, though, UK distributors ASAP Europe sent me the newer MkII. The changes aren’t vast, but they’re enough to make a useful difference. There’s a new panel layout, with digits around threshold, make-up gain and blend controls, which makes it easier to recall and compare settings, and the gain-reduction meter responds more speedily than on the first design. TK have thoughtfully added internal jumpers so that you can configure the device for balanced or unbalanced use (the latter will be useful for anyone feeding the

The world isn’t short of SSL-inspired options. Obviously, SSL’s own X-Logic take on their own console bus compressor is a viable alternative, as is their modular X-rack version. Former SSL designer Al Smart has also created his own versions, in the form of the Smart C1 and C2, the latter tweaked in the direction of more ‘attitude’. The Rolls Super Stereo compressor is another option, and DIY fans might be interested in building their own clone, with Gyraf’s GSSL project seeming popular. However, the BC1 isn’t an out-and-out clone: it’s also capable of sounding rather smoother and of more finessed results. If I were to list alternatives on that criterion alone, though, this box would take up the whole magazine.

I’d initially liked about the MkI. Other than the tickling-of-the-meters tactic I described earlier, I’ve usually found the SSL-style bus compressors to be rather blunt tools: they tend either to give me exactly the result I’m looking for, or come nowhere close to it. The wet/dry blend control offered here opened up a whole new world of options. For example, I could let the gain-reduction meter’s needle hit down nearly as far -10dB on a drum bus, to get a satisfyingly hard-hitting compression character, before backing off the blend control to the 12 o’clock position which retained elements of ‘smack’ but also restored some of the natural feel. Used on the mix bus with the 1.5:1 ratio, the BC1 started to deliver on the promise made in the manual of it being “one of the most transparent bus compressors ever made”. In this role, the BC1 comes across as being much more ‘honest’ and ‘polite’ than the SSL, in a rather pleasant way: smoother and better behaved, if you like. Whatever material I was deploying the BC1 on, the switchable side-chain filter seemed useful. It’s one of those no-brainer additions that I’m surprised hasn’t appeared in more software emulations, as it’s so useful. Going back to the drum-bus example, the effect

There’s little to see on the rear panel: just the balanced audio I/O, the IEC power inlet and the external side-chain input.

wet/dry blend control is perhaps the most useful addition, and after a while, I found that I never wanted to have it set fully wet, such that I could only hear the compressed signal. The result just didn’t sound as good to me that way. Is the BC1 one of the most transparent bus-compressors around, as TK claim? I’m not sure ‘transparent’ is exactly the right word, given the compression character inherent in this design. ‘Clean’, ‘fast’ and ‘good’ would be perfectly apt, though, and whatever labels you pin to this device it’s hard to find fault with it: it’s capable of clean, gentle compression; it’s capable of royally spanking a drum bus; and it’s very much at home on the stereo bus in a range of genres. I have to say that the price, given the quality and functionality on offer here, was a pleasant surprise too. If you like that SSL bus compression sound, but are looking for something that can turn its hand to rather more tricks, the BC1 MkII is well worth an audition.

$ W W W


w w w . s o u n d o n s o u n d . c o m / January 2012



Steinberg Sequel 3
Loop-based Music Production Software For Mac & PC

Steinberg’s Sequel is one of many simple but intuitive loop-based music programs. What’s new in version 3 — the sequel to the sequel to Sequel?

teinberg’s Sequel is one of several applications which, like Sony’s Acid, Apple’s GarageBand and Acoustica’s Mixcraft, is equipped with tools including integrated time-stretch and pitch-shifting for manipulating sampled loops, and supplied with pre-made libraries of content to play with. Although targeted at beginners, these packages are deceptively powerful, and capable of producing quite professional-sounding results. Our last encounter with Sequel was back in 2007, when version 1.0 had not long been


released. Since then, the application has been updated, as applications will be, and the current version is 3.0. A fair selection of new features has been added, but the price has been kept low and the emphasis remains on ease of use, with beginners on a budget still forming the target audience.

Sequel Returns
Sequel can be bought as a boxed package or a (large — more than 4GB) download, for Mac OS 10.6 or 10.7, or Windows 7. Earlier operating system versions aren’t mentioned in the minimum system requirements, so it’s safest to assume they won’t work. Sequel is also

slightly more demanding of hardware than before: a dual-core CPU and 2GB of RAM are now considered essential, as is a display resolution of at least 1280 x 800. As in previous versions, Steinberg use their ‘eLicenser’ software to handle copy protection. A control-panel application and device driver are installed, and if you have a Steinberg hardware USB dongles (not supplied), you can store your licence on it rather than on the machine itself. Either way, activation requires an Internet connection and is compulsory. You’re also invited to register online. Note that the MP3 encoder shipped with Sequel is, disappointingly, a demo, limited to 20 starts, after which a separate licence ($15.99) must be purchased if you want to export projects as MP3 files.

Sequel Harder
Version 3.0 adds some new features and streamlines some existing ones, but


January 2012 / w w w . s o u n d o n s o u n d . c o m

the program hasn’t undergone a radical overhaul — which is fine, because it didn’t need one. The user interface is largely unchanged, and remains recognisably similar to previous versions and to its competitors. The main application window is divided into three panes. The Arrange Zone is in the top left, and is equivalent to the arrangement window in any conventional sequencer: audio and MIDI tracks run left to right along a timeline, while a moving marker indicates the playback position. The Media Bay in the top right is a file browser that allows you to explore the content provided (along with any other sound libraries you possess), audition files and add them to the current project. The Multi Zone along the bottom is a tabbed strip in which a mixture of relevant information and parameters can be displayed; the main mixer appears here, along with effects and instrument parameters, the waveform editor for audio parts, editors for MIDI parts, and so on. The narrow strip along the top of the window where the transport controls live is called the Pilot Zone.

Steinberg Sequel 3 $80
• The Beat Page makes drum programming quick and easy. • VST Amp Rack is easy to use and sounds impressive. • Third-party VST3 plug-in support is welcome.

Selecting a MIDI or audio part in the Arrange Zone allows you to access the piano roll or waveform editor, respectively, in the Multi Zone, and to choose insert and/or send effects for that track’s mixer channel. New audio and MIDI parts can be recorded from whatever input hardware you happen to have available (your computer’s audio inputs, say, or an external MIDI keyboard). External MIDI controllers can also be assigned to application and plug-in parameters. Audio files are auditioned simply by clicking their names in the Media Bay. Double-click a file and it will be added to the arrangement on a new audio track, where it can have its start and end points adjusted, be copied and pasted, and so on. MIDI files are handled in pretty much the same way: double-click one and a MIDI track is automatically created with an appropriate instrument assigned. The library of supplied MIDI and audio loops makes it possible to sketch out a simple backing track quickly and easily. Vocals or other instruments can then be overdubbed and edited as required, the relative levels adjusted in the Multi Zone’s mixer, and the finished project exported as a sound file in either WAV, AIFC, AIFF or OGG formats (or MP3, while the MP3 encoder demo lasts).

The new Step Envelopes let you control effect and mixer parameters on a per-slice basis.

or created and edited in step time. MP3 files can now be dragged and dropped into an arrangement, and Sequel will automatically detect the tempo — an aid to mashing up or remixing. Step Envelopes are a new feature of Sequel’s sample editor, and provide an easy way to modulate audio loop playback, via a kind of automation. Five different Step Envelope types exist: Level, Pan, Decay, Pitch and Reverse. Steps are either auto-detected from transients such as single hits within drum loops, or imposed in a quarter-, eighth- or 16th-note grid for less straightforwardly rhythmic sounds. Select the tool corresponding to the desired envelope type and you can either draw in ‘ramps’ (for instance, with the level tool, to create a crude fade-in or fade-out), or assign discrete values for individual steps. You could, for instance, use the reverse tool to reverse every third step. A Randomize button generates random values for each step, which are applied using the currently selected tool; this is a quick and easy way to produce variations on a loop.

Sequel 3D
Beat Tracks are a new track type in Sequel 3, and make use of a new editor, the Beat Page, to edit part data in the Multi View. The Beat Page consists of a simple step sequencer paired with the integrated Groove Agent One drum sampler. The sampler can load up to eight banks of 16 sounds, each one automatically assigned its own ‘lane’ in the step sequencer. Various ready-made ‘pattern banks’ can be selected in the Media Bay. Dragging

Sequel: Resurrection
So far, so similar. But what’s new? Well, the Arranger Track from previous versions of Sequel has been replaced with (or renamed as) the Performance Track, although this still does much the same job. Parts created on the Performance Track are automatically associated with Sequel’s Live Pads, which are accessible in the Multi Zone and can be used to trigger jumps between different sections of your composition. In this way, new arrangements can be improvised as you go, in a manner vaguely reminiscent of Ableton’s Live. These Performances can be recorded and played back in real time,

• MP3 encoder shouldn’t really be an optional extra.

An already capable package receives some new features and polish. Sequel 3 is nothing revolutionary, but continues to provide a good all-in-one solution for people who have been wondering where to start.

Test Spec
• Steinberg Sequel 3.0.0. • PC laptop with 2.1GHz Intel dual-core CPU and 2GB RAM, running Windows 7 SP1.

w w w . s o u n d o n s o u n d . c o m / January 2012



Beat Tracks, and the Beat Page editor shown here, provide simple and effective tools for drum programming.

and dropping a pattern bank into the Arrange Zone automatically creates a new Beat Track. You can also create your own kits by manually loading sound files into the sampler. Each sample can be tuned up or down as required, be reversed, and have amp and filter envelopes applied. The Beat Page sequencer has 16 steps to a bar, and a button to toggle pattern length between one and two bars. Clicking a step activates it, to trigger the sample assigned to that lane. Dragging up or down adjusts the velocity for a hit, whereupon the activated step changes colour to reflect its value. Shift-dragging allows you move all hits on a lane backwards or forwards, in step increments. If fixed 16th-note quantisation is not to your liking, there are various things you can do to loosen it up: the pattern resolution can be switched to triplets; a Swing value can be specified to move alternate hits slightly ahead of or behind the beat; a Flam tool allows you to create a secondary hit just before or just after the activated step; and dragging left or right on an icon beside the sample name allows you to create a small offset, in either direction, for all hits on that lane. Programming drum patterns on the Beat Page — whether with single hits, loops, or a combination of both — couldn’t be much simpler. The supplied sound library includes plenty of usable stuff, and it’s easy to mix it up with sounds from your own collection. The Beat Page arguably favours ease of use over flexibility, but that’s no bad thing in an application targeted at newcomers. Sequel has a built-in sample player for pitched instruments too, derived from Steinberg’s HALion software sampler line and dubbed HALion Sonic SE. It is, as you
The Prologue virtual synth is one of several additions taken from Steinberg’s Cubase DAW.

would expect, a rather simplified version of the larger HALion Sonic product, but useful nonetheless. It’s a sample-playback synth rather than a tool for experimental sound-mangling, but it’s easy to use and works well. The supplied sound library is good, and there are enough tweakable parameters, including tuning and transposition, filter and amp envelopes, and built-in multi-effects, to allow for lots of variation on the presets. The instrument’s Macro Page makes most of the important parameters accessible in one place, and you can save your tweaked patches as ‘.vstpreset’ files. Steinberg’s Prologue, an analogue-style subtractive synthesizer, is also included. It’s a simple but effective synth with three oscillators, a multi-mode resonant filter, four ADSR envelope generators, two LFOs and some built-in effects. The presets are generally usable, if not breathtaking. The supplied documentation is minimal, bordering on

non-existent, but the instrument’s user interface is large and clear enough that even beginners should feel able to try some tentative tweaking.

Escape From The Planet Of The Sequels
A new effect in Sequel 3.0 is VST Amp Rack. This is a nicely implemented guitar-amp simulator, capable of producing an impressive range of recognisably classic guitar sounds. Its user interface is clear and simple, with just six control knobs and a large library of usable presets. Many of these include additional effects such as distortion or reverb, represented as miniature stomp boxes, which are not editable. Both the quality and variety of the sounds on offer is striking, and the clean sounds are every bit as good as the dirty ones, which is not always the case with digital emulations. There are some nice imitations of vintage ‘tweed’ and


January 2012 / w w w . s o u n d o n s o u n d . c o m

‘silverface’ combos, for instance, along with the standard rock and metal stacks. In a nice cosmetic touch, the plug-in’s graphics change to match the amp type in use. VST Amp Rack can be used either to liven up sampled guitar parts from the library, or to record new guitar parts, DI’d into your computer. It’s as easy to use as a real guitar amp, and does a pretty good job of sounding like one (or, rather, like several). All the effects and instruments supplied with Sequel are implemented as VST3 plug-ins, and Sequel now also supports the use of third-party plug-ins as well. This is a step forward (it was a rather puzzling omission from earlier versions) and should allow Sequel users to experiment with a wider variety of effects and instruments. Note, however, that only VST3-format plug-ins are supported; plug-ins in the earlier and still much more widespread VST 2.4 format are not.

VST Amp Rack is a simple but versatile amp simulator.

28 Sequels Later
Sequel 3.0 is a polished package, which does a good job of providing

novice users with the tools they need to get started with computer-based recording and production. The basic building blocks of a composition can be assembled quickly, and simple projects can be brought to fruition without the need for any other tools (except, perhaps, that optional MP3 encoder). There are strong competitors in this field, as mentioned earlier in this review,

offering at least comparable features for around the same price, so it would be wise to check out demo versions of the alternatives before making a decision. However, with version 3.0, Steinberg have ensured that Sequel remains deserving of serious consideration. $ T E W
$79.99; upgrades from v1/2 $29.99. Steinberg +1 877 253 3900.

w w w . s o u n d o n s o u n d . c o m / January 2012




Worth over


January 2012 / w w w . s o u n d o n s o u n d . c o m

Win! A full studio setup from
t may be hard to believe, but Samson Technologies are over 30 years old! Founded in 1980, Samson began developing wireless microphone systems, and have since gone on to produce a broad range of audio equipment, including power amps, mixers and signal processors. Today, Samson Technologies are associated with three well-known brands: Samson, Hartke and Zoom. Hartke specialise in bass amplification, and are best known for pioneering the use of aluminium speaker cones in 1985, with their 410XL cabinet. Their amps have graced the stages of such luminaries as Jack Bruce, Jaco Pastorius and Victor Wooten, to name but a few. Zoom, distributed in the States by Samson Technologies, are famous for their digital guitar pedals, including the 505 and GFX8 multi-effects units. They have now branched out into manufacturing drum machines, samplers and, more recently, hardware audio recorders — the pocket-sized H1, for example, and the R8 multitrack recorder. For this month’s SOS competition, Samson Technologies are giving away a complete studio bundle, comprising

Samson & Zoom
both Samson and Zoom products, with a combined value of over $4000! The collection of prizes begins with the R24, Zoom’s premier multitrack recorder. It can capture eight channels simultaneously, and has 24 record tracks in total. It even incorporates sampler and drum-machine functions. You can read our review of it at articles/zoom-r24.htm. And that’s just the beginning: Samson are also supplying a C01 large-diaphragm condenser, as well as a Samson 7Kit, which is a seven-piece drum-mic set featuring a bass-drum mic, a snare mic, three tom mics and two overheads. The winner will also receive a Samson Q7 supercardioid dynamic mic. To allow the winner to listen back to their recordings, Samson will provide a pair of their Resolv A6 active studio reference monitors and SR850 headphones. A Zoom H2N portable, surround-sound-capable recorder is also included in the prize. This handy recorder is great for capturing field recordings, as is the included Zoom Q3HD (with its accessory pack), which combines a high‐quality stereo recorder with HD video capture. Samson’s Meteor Mic — a desktop microphone with a built-in USB interface for quick and easy computer recording — is also included in the impressive bundle. Rounding off this fantastic suite of prizes is a collection of essential accessories: a set of Samson MS200 monitor stands, a CM40 chromatic tuner/metronome, a PS01 pop filter, an MB1 boom mic stand, two BL3 ultra-light mic stands, an MD5 desktop mic stand, an SP01 Spider shockmount, four Tourtek mic cables and four Tourtek instrument cables. In other words, the prize includes absolutely everything you need to get recording straight away! For a chance to win this gargantuan set of prizes (26 individual products in total), all you have to do is fill out the form below, answer all of the questions, including the tie-breaker, and send the form to us. Alternatively, you can enter online, via the SOS web site. The closing date for entries is 11th February 2012.
Prizes kindly donated by Samson Technologies T Samson Technologies +1 631 784 2200 W


the small print
1. Only one entry per person is permitted. 2. Employees of SOS Publications Ltd, Samson Technologies, and their immediate families are ineligible for entry. 3. No cash alternative is available in lieu of the stated prize. 4. The competition organisers reserve the right to change the specification of the prize offered. 5. The judges’ decision is final and legally binding, and no correspondence will be entered into. 6. No other correspondence is to be included with competition entries. 7. Please ensure that you give your DAYTIME telephone number on your entry form. 8. Prize winners must be prepared to make themselves available in the event that the competition organisers wish to make a personal presentation.

What were the first products that Samson developed? a. Multitrack recorders b. Guitar effects pedals c. Audio interfaces d. Wireless microphones In what year were Samson founded? a. 1978 b. 1980 c. 1995 d. 1999 How many channels can the Zoom R24 record simultaneously? a. 24 b. 32 c. 8 d. 2

Samson & Zoom Tie-breaker
As well as being the name of an audio company, Samson was also a biblical figure, who used his supernatural strength to wrestle a lion. If you had supernatural strength, what would you do with it and why? Answers in 30 words or fewer, please.
.................................................................................................................................................................................................................................... .................................................................................................................................................................................................................................... ....................................................................................................................................................................................................................................

Name: ....................................................................................................... Address: .................................................................................................... .................................................................................................................... Daytime tel. no: ....................................................................................... Email: ........................................................................................................

If you would like to receive more information about Samson Technologies products, please tick or cross this box.

Post your completed entry to: Samson & Zoom Competition January 2012, Sound On Sound USA, 122 Calistoga Road #330, Santa Rosa, CA 95049, USA.

w w w . s o u n d o n s o u n d . c o m / January 2012



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> More product coverage than any other music magazine > In-depth product testing > Software and hardware: we know the importance of both > The best musician, engineer and produce interviews you’ll read anywhere

on the bookstore p rice


> Tutorials and practical ‘how to’ workshops offering no-nonsense tips and techniques > A highly experienced editorial team, second to none in hands-on expertise > Since 1985 Sound On Sound has remained consistently at the forefront of developments in technology shaping the music production industry today
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This lovely mic from AEA is based on a classic RCA design from the ‘40s. It certainly looks the part, but is it worth the asking price?

Supercardioid Ribbon Microphone

hen the ‘ribbon microphone’ comes up in a discussion, most people will automatically assume that the mic in question has a figure-of-eight polar pattern — after all, the vast majority do! Compared with other capsule technologies, the practicalities of building a ribbon microphone make it harder to deliver a variety of polar patterns, and the nature of the ribbon itself naturally pre-disposes it to velocity (pressure-gradient) operation, which means a figure-of-eight polar pattern. To put some numbers on that notion, of the 58 ribbon microphones currently listed on, 55 have a figure-of-eight polar pattern. Two of the remaining three (the Beyerdynamic M160 and M260) are hypercardioids, and the last is the cardioid RCA KU3A — but that particular mic is long since out of production, and is only listed for historical interest. Although not listed on the site, Silvia Classics make a cardioid ribbon mic called the SC5C, which looks similar in design to the long-deleted RCA BK5.


AEA KU4 $4720
PROS However, there is no intrinsic link between the electro-mechanical nature of a microphone, be it electret, capacitor, ribbon or moving-coil, and its acoustical properties (pressure or velocity operation, or a combination of both). We are quite used to capacitor mics being available with omnidirectional, cardioid, hypercardioid or figure-of-eight polar patterns and, in theory at least, a ribbon microphone could be constructed to deliver the same range of pickup patterns if so desired. All of which brings us neatly to the subject of this review: Audio Engineering Associate’s new KU4 ‘unidirectional’ ribbon. And if you’ve noticed a similarity between this new model’s name
• Silky smooth ribbon character with a supercardioid polar pattern. • AEA have taken considerable time and effort to deliver what the KU3A promised. • Looks fabulously imposing!

• Hugely expensive. • Hugely heavy. • Huge.

There is always something special about the sound of a good ribbon mic, and this reincarnation of the KU3A is stunning in every sense. I’ve said this before about AEA’s classic mics but it remains true: they are expensive and revered, like a vintage Bentley, but not easy to justify to the accountants.


January 2012 / w w w . s o u n d o n s o u n d . c o m

and that of the RCA classic I mentioned above, that’s because the KU4 ‘celebrates’ the KU3A. You can read more about the original KU3A microphone in the ‘Blast From The Past’ box.

Ribbon Revival
AEA are very well known specialist ribbon-microphone manufacturers and also have a strong pedigree in repairing and restoring classic ribbon microphones. Simultaneously, they have built quite a reputation for their modern ‘reincarnations’ of some of those original RCA designs, as well as their own bespoke models. I’ve reviewed many in the past, including the A440 (June 2009), the R44C (June 2002), the R92 (April 2007), and the R84 (February 2004), as well as the company’s purpose-designed TRP and RPQ ribbon-mic preamps (April 2007 and October 2008, respectively). The company’s new KU4 microphone is a very close derivative — a modern recreation, if you like — of the original RCA KU3A, and even uses new-old-stock (NOS) RCA ribbon material. The similarity in design is emphasised by the fact that some parts are directly interchangeable with the original model. While surviving samples of the KU3A are rare, it remains a much revered microphone for its sound quality. However, I’m told that finding two examples that sound remotely similar is extremely difficult, partly because of the effects of ageing and partly because manufacturing consistency was notoriously poor. Apparently it has taken AEA’s boffins over two years to develop and fine-tune the design and manufacturing process of the labyrinth chamber in the KU4 to achieve the level of consistency that modern customers require — and to prove the point, KU4s are available as matched pairs for stereo applications, if your bank manager is obliging enough!

and a permanently attached two-metre microphone cable. The cable, which I believe is Accusound Silver Studio Pro, has a vintage-style woven cloth exterior and is terminated in a Switchcraft XLR plug. (The original KU3A employed a large F&E three-pin connector, mounted at an angle on the top rear side, to provide the output signal.) The microphone is covered by a three-year parts and labour warranty, but there is nothing in the microphone that has a limited lifespan, so it should provide decades of service provided it is handled and used with respect.

There is nothing currently in production that really comes close to the sound or styling of the KU4, although other hypercardioid and supercardioid ribbon mics are available in the form of the Beyerdynamic M160 and M260, and the Silvia Classics SC5C. Vintage RCA KU3As are very rare, just as expensive and rather more variable in tonal quality!

Intelligent Design
The KU4 is about the same size as the original KU3A, measuring 324 x 117 x 95mm (HWD), including the mounting bracket. The KU4 appears to be heavier in its mounting bracket than the KU3A, at a massive 2.15kg, so this microphone needs a very substantial microphone stand, and if it is mounted on a boom arm, a serious counterweight will be required too. The KU4’s sensitivity is listed as 2.5mV/ Pa (about 1.5dB lower than the original), maximum SPL is 140dB, and recommended load impedance is 10kΩ or higher. This last point is worth noting, as most mic preamps present an input impedance of between 1kΩ and 2kΩ. In contrast, most ribbons prefer to see considerably higher loads, which is why AEA’s own mic preamps have input impedances of 18kΩ or so! (The original KU3A’s output transformer could be adjusted for 30Ω, 150Ω or 250Ω output impedances). The microphone’s polar pattern is a well-defined supercardioid (the KU3A was claimed to have a cardioid pattern, although most actually had a hypercardioid response), with deep rear minima at 135 and 225 degrees. AEA found that the best compromise between on-axis frequency response and consistency of the off-axis response around the polar pattern was achieved with a supercardioid pattern. As you might expect, the polar pattern narrows across the front quite rapidly for extreme HF sounds (10kHz being 5dB down by ±45 degrees off-axis, for example), but it actually has quite a wide pickup angle across most of the audio spectrum, with the -6dB points being about ±75 degrees. The ribbon itself is relatively large, measuring 2.1mm wide by 28.6mm long, and is made from pure aluminium measuring just 1.8 microns (0.0018mm) thick. The distinctive grille basket and the suspension yoke assembly are completely

interchangeable with those parts of the original KU3A, but the internal mechanical design is entirely different, as AEA have employed more modern construction techniques to ensure both tighter tolerances and a more consistent sound character from unit to unit. The design also makes it practical to replace the pole-piece and ribbon assembly in the field, if necessary. Most KU3As were slung from overhead microphone booms (either fixed or mobile), as is the norm for film and TV studios, and all manner of custom suspensions were employed, providing shockmounting and often allowing the mic to be ‘steered’ as necessary. The KU4 is provided with a much simpler removable mounting-yoke assembly, which has been derived from that used on AEA’s R44 series microphones. It incorporates a flexible ‘cushion’ mount system in the base of the yoke to provide a useful degree of vibration isolation.

In Use
Like most ribbon mics, the KU4 is actually far more robust than most people assume, but it can be severely damaged if used carelessly. The mic will tolerate proper phantom power quite happily, but as a momentary imbalance in the voltage on each side of the balanced output might damage or even destroy the ribbon, the safest advice is to avoid phantom power if possible or, if not, then avoid hot-plugging the mic while phantom is active — and always check that your mic cables don’t have any short or open connections before using them to connect a ribbon mic! A more common and likely cause of damage is to expose the microphone to strong air currents. The KU4 can cope with extremely loud sound pressure levels without any problems at all (140dB SPL produces just 1 percent third-harmonic distortion), but a small gust of wind can stretch the ribbon irreparably, reducing sensitivity and skewing the frequency response forever! For this reason, care must be taken to avoid exposing the microphone to any kind of wind blast at all times. So when not in use, the microphone should either be packed away properly or covered with a bag (the supplied

Opening The Box
The KU4 is supplied inside a zip-closed, soft-cotton twill bag intended to prevent dust and draughts from affecting the ribbon while in storage (the bag can also be used to protect the mic while mounted on a stand but not in use). The bagged mic is protected from knocks when in storage and shipping by a burgundy nylon/Cordura case lined with a sturdy foam insert, and the storage case is designed to encourage the user to store the microphone in a vertical orientation to reduce the risk of ‘ribbon sag’. The KU4 has an integral (but removable) stand adaptor, a 10-page A4-sized manual,

w w w . s o u n d o n s o u n d . c o m / January 2012



cotton bag or a clean polythene bag, for example) to protect it from accidental draughts. When considering where to place a ribbon mic for a given sound source, it is good practice to test the environment for strong air movement with the back of a hand first. If you can feel moving air when the source is playing, consider placing the mic somewhere else or protecting it with a pop screen. Angling the mic at 45 degrees to the sound source also helps to reduce the risk of damage, albeit with a slight change to the tonality. I’ve known ribbon mics to be destroyed by the puff of air from a modest guitar cabinet on the first note of a power chord, so it pays to be cautious! Most pure pressure-gradient (figure-of-eight pattern) ribbons exhibit a strong proximity (bass tip-up) effect but, because of the significant element of pressure operation employed by the KU4 to achieve its supercardioid response, it suffers less proximity effect compared with typical figure-of-eight ribbon mics. Reasonably close placement can therefore be used without excessive bass bloom, if required. As a result, AEA recommend the KU4 for a wide range of applications including low strings, brass and vocals. Like all ribbons, the KU4’s diaphragm and suspension are tuned to resonate at a very low frequency (well below 20Hz), which plays a large part in the characteristically smooth high end and the very naturalistic sound associated with ribbons generally. In contrast, capacitor microphone diaphragms tend to be tuned to a high frequency (typically in the 15-20kHz region) and this inherent HF resonance can result in ringing and an edginess or harshness if not well controlled. Although the KU4’s response tails off quite steeply above 10kHz, it actually rallies again above 16kHz or so, which helps to preserve the transient details of percussive sources very nicely, and retains a sense of air and space without ever sounding abrasive or harsh. I found the KU4 to be a sublime microphone, sharing a similarly expansive character to AEA’s other large ribbon mics, lending voices and instruments a wonderful ‘scale’ and body that is often hard to match with capacitor mics. The polar pattern is controlled extremely well, and the off-axis sound character is remarkably smooth and consistent, putting many modern mics to shame with its minimal coloration! On percussive and stringed instruments the microphone reveals a fast, precise character, with a warm, rich bass response and a silky smooth treble that is easy on the ear but

Blast From The Past
The RCA KU3A was introduced in 1948, and was designed to meet the specific needs of Hollywood sound stages, particularly for use on booms, although it was later very popular as a TV studio mic too. It is occasionally referred to by its manufacturing number, M10001, and it was RCA’s most expensive and rarest ribbon mic. In fact, fewer than 600 were produced in total, which is why they are such valuable collectors’ items today. The operational instructions advised suspending the microphone at 45 degrees to the floor (the ribbon is mounted on the vertical axis, not parallel with the sloping front grille), and the marketing brochure boasted about the microphone’s small size and light weight! By modern standards, the KU3A was both huge and extremely heavy. The KU3A essentially combined the fundamental ribbon design of that other RCA classic, the equally huge and heavy 44BX, with a complex and elaborately constructed acoustical labyrinth housed in the cylindrical body. This labyrinth (basically a folded pipe with internal damping) provided an acoustic delay for sound en route to the rear side of the ribbon diaphragm, and in doing so introduced an element of pressure operation to the capsule, bestowing it with a more cardioid-like polar pattern. A welcome bonus of a cardioid response is a reduced proximity effect compared to a pure velocity or pressure-gradient figure-of-eight mic, and the microphone quickly became a familiar and popular feature of film studio sound stages around the world. The KU3A was remarkably sensitive for a ribbon, at about 3mV/Pa, and the frequency response extended comfortably between 30Hz and 15kHz, with a modest presence boost between 2 and 6 kHz. However, its real claim to fame was the accuracy of the cardioid polar pattern, which was quite remarkable for its time and a substantial improvement on previous techniques to achieve a cardioid polar pattern, most of which employed either two physically separate capsules (usually a moving-coil pressure element and a ribbon velocity element), or one or two parallel ribbon elements with one (or a half of the single ribbon) closed off at the rear for pressure operation. The STC 4033A is a prime example of the former approach, while one of the KU3A’s sister microphones, the 77 (designed in 1932) employed the latter technique. The RCA 77 (in its suffix A and B versions) provided a fixed cardioid pattern by placing half of its ribbon diaphragm in an acoustically terminated chamber to create a pressure element. The suffix C version (introduced in the late 1940s) allowed the two halves of the ribbon to be selected electrically to produce switchable omni, cardioid or figure-of-eight patterns, while the suffix D and DX versions used a mechanical rear shutter to produce omni, cardioid, hyper-cardioid and figure-of-eight polar patterns. However, because the KU3A was designed from the outset as a fixed cardioid-pattern mic, the ribbon motor assembly and the acoustic labyrinth could be optimised to achieve an extremely consistent frequency response both on and off-axis — which was outstanding for the time and is still very impressive today.

still clear and detailed. Nothing seems to capture brass as naturally as a ribbon, and the KU4 certainly doesn’t disappoint here — and it can cope with close placement quite comfortably.

Final Thoughts
I’m a big fan of ribbon microphones generally, but the first to admit that the near-universal figure-of-eight polar pattern isn’t always very convenient. The combination of classic ribbon smoothness with a near-cardioid polar pattern is a very beguiling permutation that enables the use of ribbon mics in situations that are normally inappropriate or too challenging. The KU4’s only practical limitation, really, is due to its vast size and huge weight, but if you have mic stands that are up to the job it’s hard to think of an application that wouldn’t benefit from a KU4! However, it’s all well and good loving the sound of a microphone, but is it a sound buy? At over £5000 in the UK, few home studios will be able to justify the purchase of

a KU4, although people are often prepared to invest that kind of money in boutique ‘homage’ microphones. If you’re lucky enough to find an original KU3A in good condition (or refurbished well), it will set you back the best part of $4000 — and no two examples are likely to sound exactly the same, so the list price of the KU4 is not as mad as it might seem at first. The AEA KU4 is, apparently, a really difficult microphone to make, and is hand-built to look and sound beautiful — and to sound like all the other KU4s, which is a worthy bonus! If your numbers come up on the Lottery this weekend, I would encourage you to invest in a pair of KU4s — and it would be an investment. The rest of us will have to dream, but at least I’ve had the privilege of using one in earnest. $ $4720. T Audio Engineering Associates
+1 800 798 9127.



January 2012 / w w w . s o u n d o n s o u n d . c o m

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Analog Experience

The Laboratory
Aux, no breath control input, no pads and fewer of the other controls), let alone that of the Player, it’s clear that it’s a big step up from its smaller siblings.

hen I first read about the flagship of Arturia’s Analogue Experience (‘AE’) series, I wondered whether there would be any demand for another product based on the company’s existing analogue modelling technology. My review of the curent two products in the series — the Player and the Factory (see Sound On Sound January 2011) — suggested that they provide an interesting set of facilities in a novel fashion at an attractive price, but what of the Laboratory? Could there be room above the Factory for a bigger and better ‘Experience’, or were Arturia attempting to slice the virtual analogue soft synth market just a little too thinly? (As we were going to press, Arturia announced a 61-note version of the Laboratory, but this review will concentrate on the 49-note model.)


The latest addition to Arturia’s Analog Experience series takes their hybrid software/hardware synth concept even further.
Let’s start with the 49-note velocityand aftertouch-sensitive keyboard. This is not just wider than the keyboards supplied with the Player and the Factory; it offers a greater range of facilities and more in the way of physical controls, with nine sliders (two ADSRs and Tempo), 13 knobs, 23 buttons (including 10 snapshot buttons that allow you to recall favourite patches from the keyboard) and four pads, as well as traditional pitch-bend and modulation wheels. In addition to these, there are six buttons that send MMC messages (Start, Stop, Record, Backward, Forward and Loop) for transport control of hardware and software sequencers. Meanwhile, around at the back I was pleased to find five-pin DIN sockets for MIDI In and Out, as well as the expected USB/MIDI socket, plus quarter-inch sockets for no fewer than four forms of control: sustain, expression, ‘Aux’ and breath control. This is very sophisticated for a low-cost system and, if you compare this with the 32-note keyboard that comes with the Factory (which has no MIDI In, no

The Laboratory Sounds
Complementing the hardware, the Laboratory contains the latest generation of Arturia’s Analogue Experience software, with 3500 preset (but editable) sounds based on the company’s Moog, ARP, Roland, SCI and Yamaha soft synths. Once installed, the software appears as VST, AU and RTAS plug-ins, as well as a stand-alone application, and will run on OS X, Windows XP, Vista and 7. Getting it up and running on my MacBook Pro proved to be straightforward, although you need Internet access to register it, and a bit of prior experience with the eLicenser system used by Arturia goes a long way toward making the process as painless as possible. The software seems very similar to that supplied with the Factory and the Player, and it’s worth reading my earlier

Physically Speaking
Like its siblings, the Laboratory combines three elements: a keyboard controller, a large library of editable sounds based on the company’s existing V-series soft synths, and a separate software package that allows users to configure the keyboard for use with other software.


January 2012 / w w w . s o u n d o n s o u n d . c o m

review if you haven’t already done so, because almost everything that I wrote about the smaller systems remains relevant. But while it’s tempting to assume that the Laboratory version is much the same, with just a bit more of this and a smidgen more of that, it would be a mistake, because, in three important ways, the Laboratory is much more powerful than its siblings. Although Analogue Experience sounds are based on Arturia’s V-series soft synths you have (until now) only had access to a limited subset of their editing and performing parameters. So, for example, the Player allows you to adjust the filter, the LFO and the amplitude envelope of a sound that has dozens of other parameters hidden away ‘under the hood’. The Factory is somewhat more flexible, adding access to the effects mixes and four assignable parameters, but it is still unable to edit a sound fully. And, at first sight, the Laboratory is only slightly more advanced, with the addition of a second ADSR contour generator. But if you have any of Arturia’s V-series synths installed on the same computer as the Laboratory, something magical happens. Click on the edit button of an appropriate sound

and you can open the original soft synth within the Laboratory to edit it, and even create completely new sounds. Strangely, Arturia don’t seem to make a big deal of this within its documentation, but I think that it’s a huge step forward. It makes the Laboratory much more than just a tweaker and player of preset sounds; it’s now an über-editor/librarian for all of your Arturia synths, integrating them into a single environment that is much more manageable than invoking each of them individually. The next huge difference is the provision of Scenes. These allow you to combine two sounds; either layered or placed either side of a user-defined split point. There’s also a Multi mode, although this is not the full multitimbral mode that the name implies — it’s a duo-timbral mode that assigns the sounds of your choice to MIDI channels 1 and 2. You can assign sounds to the Upper and Lower parts; select and edit them from the keyboard; transpose, mix and pan them; and assign a ‘Melody’ (actually, one of 180 preset arpeggios) to one of them. While there’s no way to edit these arpeggios, they can be used in conjunction with the library of preset rhythm loops accessed using the pads on the keyboard (or their on-screen equivalents), which makes them rather useful when using the Laboratory ‘DJ-style’ or as a scratch pad for ideas. Ah yes, the pads... Like much else in the Laboratory, these offer more than is immediately obvious. Far from

Test Spec
• Apple MacBook Pro, 2.6GHz Intel Core 2 Duo, 4GB RAM, OS 10.6.8. • Analogue Laboratory v1.2.0 and v1.3.1.

simply allowing you to play notes or tap percussion instruments (which, of course, they do) each provides three modes — gate, trigger and loop — and can act exclusively (or not) with respect to the others. So, for example, you can place up to four complementary rhythm loops under the pads, select ‘loop’ for each, switch ‘exclusive’ off for each, mix their levels to taste, and then play simple or layered rhythms, switching each loop on or off by tapping the appropriate pad. Selecting the loops couldn’t be simpler (you drag and drop them from the list in the pads’ setup page) and you don’t even have to be precise when you play them; adding a new loop to something that’s already playing always results in a synchronised rhythm, no matter how poor your timing might be.

The MIDI Control Centre
The final element in the package is the MIDI Control Centre software installed alongside the Laboratory itself. This allows you to configure the hardware so that every control, when tweaked, sends the MIDI CC# of your choice. You can then store the configuration in the keyboard itself, in effect turning it into a dedicated controller for another synth

Arturia Analog Laboratory $399
• Scenes and rhythms take the Laboratory to another level when compared with the Factory and Player. • You can now access and edit AE sounds using the full V-series soft synths (if you have them installed). • The sound quality remains uncompromised. • It remains good value for money.

• There are still a few small glitches to iron out.

Not only does the Analogue Experience remain a unique product concept, the Laboratory is clearly the most powerful and flexible of the three models in the range. If you have room for its larger, more accomplished keyboard, it’s definitely worth considering in preference to its smaller siblings.

The software side of the Laboratory; the bottom panel mimics the settings made on the controller keyboard.
w w w . s o u n d o n s o u n d . c o m / January 2012



The Laboratory’s rear panel has MIDI In and Out ports, three control inputs on quarter-inch jacks, a USB connection, and an input for an optional 6V power supply.

or software package. Since configurations can be saved on your host computer, you can quickly reconfigure the keyboard for whatever purpose is required, and I particularly like the idea of assigning the nine sliders to act as physical drawbars for a software Hammond organ emulation. The keyboard boasts a 12V DC PSU input, so that you can use it as a stand-alone MIDI controller without connecting it to a computer for USB power. It’s also worth mentioning that it’s a USB/MIDI converter too, and this allowed me to use my Mac to play and control vintage MIDI synths (which have no USB inputs) without the need for a dedicated interface. The only time that this failed to work was when I attempted to play them from the Laboratory’s GUI. I confirmed this with Arturia, who admitted that they had not expected anyone to attempt this.

In Use
Using the Laboratory could not be much simpler. Load the software with the keyboard plugged in, and everything synchronises and is ready for use. Nonetheless, its capabilities range far beyond simply playing its existing sounds. To illustrate this, I loaded the Laboratory as a plug-in within Digital Performer 7, created a MIDI track, routed the input from the keyboard to the plug-in, and selected a string ensemble patch within it. I then loaded a second

Other than the earlier Analogue Experience products, the only previous soft synth supplied with multiple synth models and a dedicated keyboard was, as far as I am aware, the original version of Korg’s Legacy Collection. Currently, therefore, the Analogue Experience is unique. What’s more, the only other semi-preset soft synth that allows you to invoke a selection of more powerful soft synths to edit its wide range of ready-for-use patches is... well, I can’t think of one.

instance, allocated this to a different MIDI track and selected a choral patch. I could now click on either of the MIDI tracks and play the instance of the Laboratory connected to each. Moving on, I created a MIDI Group and attempted to play both instances of the Laboratory simultaneously. This worked perfectly, and I now had a luscious ‘choir and strings’ ensemble under my fingertips. Invoking Scenes on each, I also had access to two duo-timbral synths with two, independent (but synchronised) rhythm sections. This was getting interesting. Six timbres and three rhythm sections proved to be even more so. As for eight (or 10, or 12...) timbres derived from surprisingly accurate imitations of Moogs, ARP 2600s, CS80s, Jupiter 8s and Prophets, whether split, layered or accessible as a complete multitimbral setup... well, I’m sure you get the picture. I even invoked the full V-series GUIs within these setups, and everything functioned as it should. This was good stuff. Regarding the hardware itself, I mentioned to Arturia that a couple of the sliders and the mod-wheel brushed very slightly against the case on the review keyboard. It wasn’t a serious problem, and I doubt that many users would even have cared, but they immediately despatched another one to me. This was much better. On both units, I found the semi-weighted keybed just a little too light for serious playing, but this was not because the action had changed significantly from the keybed that I complemented on the Factory keyboard. It was because my fingers and eyes have different expectations of a keyboard as wide as a Nord Wave’s or a Waldorf XTk’s rather than one that is clearly intended for use as a small USB MIDI controller. To be fair, there is still room for improvement in the Laboratory and, in particular, I would like to see the MIDI channel assignment made more flexible

in Scene mode. Furthermore, the review was not entirely without glitches, but these were either harmless (such as the ‘amazing but easily resolved disappearing GUI trick’ that I could perform at will when I had multiple instances running in DP7) or could be prevented by avoiding arcane ways of doing things. (When all else failed, as it did on only one occasion, a suggestion from Arturia’s support people quickly identified and resolved the problem.) But these are minor niggles. The laboratory does what it promises, and in general does it very well indeed.

For players who love the sounds of classic analogue synths but have no desire to learn how to wring the best out of complex control panels, the Analogue Experience series, with its huge selection of high quality, easily accessible, and tweakable VA sounds, remains unparalleled. Moreover, within this series, the Laboratory — with its enhanced hardware, scenes and rhythms — is very clearly the pick of the bunch. But it’s more than that. When I concluded my review of the Player and Factory I wrote, “Let’s be clear, these are no über-synths that allow you to create outrageous, never-heard-before sounds.” Sure, if you don’t own any of Arturia’s V-series soft synths, this remains true but, with the synths present, the Laboratory also becomes a powerful librarian and editor that centralises access and control over all of the Arturia synths loaded onto your computer. So I now have the answer to the question that I posed at the start of this review. Is there room above the Factory for a bigger and better ‘Experience’? Yes, there is. $ 49-key version $399,
61-note version $499.



January 2012 / w w w . s o u n d o n s o u n d . c o m


Coleman Audio QS8

oleman are American manufacturers of simple but very functional analogue audio switchers, meter units and passive monitor controllers. We’ve already reviewed the M3PH MkII and SR5.1 MkII in the pages of Sound On Sound (SOS March 2006 and April 2007, respectively), but the latest addition to the fold is the QS8 Master Monitor Control unit.


Master Monitor Controller
Monitor controllers aren’t just glorified volume knobs, as this latest offering from Coleman shows...
channel crosstalk varies a little with the attenuator setting, but is typically around -80dB (10kHz, 0dBu test signal). In normal operation, the monitor path can be derived from one of two stereo inputs (Aux 1 or Aux 2), and the selected signal is routed to either the main or alternate speaker outputs, all selected via front-panel push-buttons. The two stereo Aux inputs and the two stereo speaker outputs are all connected via rear-panel XLRs. The QS8’s monitor path controls aren’t laid out in the most logical or intuitive way, perhaps, but work well enough in practice. The passive signal path means that there are no active electronics to add noise, distortion or tonal colouration, of course, and, provided that connecting cable lengths aren’t excessive and the source and destination impedances are within the normal range, this solution is about as accurate and transparent as is possible for a monitor controller. The rest of the QS8’s facilities are all mains powered, and the unit has a captive mains cable that exits from the back of the right-hand panel. This is a common feature of Coleman products, and while it’s not normally an issue, it can make installation in a tight rack frame difficult in some cases. An on-off rocker switch on the front panel powers the unit, although there is no power-on indicator, which is a shame. There is no external voltage selector or mains fuse, and no indication as to the equipment’s voltage rating — something that would theoretically force an instant PAT fail! Starting at the left of the control panel, the built-in headphone amplifier is equipped with a rotary level control and a quarter-inch TRS (stereo) output socket. The headphone output is normally the same as the selected monitor path source, but a push-button allows the artist’s cue-mix output to be auditioned instead, to check what the artist is hearing. Consequently, the headphone facility could serve as a convenient cue source for an artist performing in the control room, if required. The artist’s cue section dominates the centre of the control panel. Its principal source is a stereo balanced cue input (on quarter-inch TRS sockets on the rear panel), the level of which can be adjusted with a rotary level control that provides unity gain when fully clockwise. This signal would

This new model is a multi-function device that serves as a stereo passive monitor controller and also has a built-in engineer’s headphone amp, an integrated artist cue facility, complete with talkback and, rather unusually, an analogue stereo summing mixer that accepts four stereo inputs. All this comes in a smart 1U rackmount case! As with all Coleman monitor controllers, the principal control-room monitor path is entirely passive, with the balanced signal passing through high-quality switches and a rotary switched attenuator that provides an accurate, stepped level control. The lightly detented switch positions provide roughly 2dB steps over the top half of the range, from unity down to -34dB of attenuation. After that, there are 4dB decrements down to -50dB, then a 10dB jump to -60dB and, finally, an ‘off’ position providing 105dB of attenuation. The tracking accuracy is superb across the critical top half of the range, with less than 0.05dB between the two channels. The


January 2012 / w w w . s o u n d o n s o u n d . c o m

The controls are neatly laid out on the front panel, and all the I/O is hosted on the rear.

normally be a rough mix or backing track derived from the DAW to which the artist is recording. Alternatively, a push-button allows the cue input to be replaced with the monitor path’s selected signal source instead. The balanced cue output signal is dispatched via another pair of quarter-inch TRS sockets and is intended to feed an external headphone amplifier. A rotary master output-level control (Cue Mix Level) provides unity gain when at the 12 o’clock position, and +16dB gain when fully clockwise. A second source, labelled Aux 3 (and connected via a pair of XLR sockets on the rear panel) can be combined with the cue input. Again, this source is equipped with a rotary level control (unity gain when fully clockwise), as well as a mono button. This additional cue-mix input is intended for latency-free source monitoring, and the idea is that the artist’s microphone preamp output is split to feed both the Aux 3 input for direct monitoring, and the DAW

Coleman Audio QS8 $1299
• High-quality switched attenuator in main passive monitor path. • A/B source and ‘alt’ speaker switching. • Second input to artist cue system, to facilitate zero-latency source monitoring. • External talkback mic for convenient placement. • Stereo summing mixer capability.

• Summing-section headroom issues on some inputs. • No power-on indication.

A well-equipped passive monitor controller with added value features including artist cue facilities and talkback, and a basic summing mixer thrown in for good measure. The design has some quirks that may limit its application in some cases.

interface for recording. In this way, the artist can hear their own performance without latency, and with the ability to balance their own level against the cue-mix track from the DAW. To aid communication with the artist, the QS8 also includes a talkback facility with a volume control and a non-latching, click-free activation button that routes the talkback signal directly to the Cue Mix output. An external talkback mic is provided in the form of a large plastic disc (roughly 5cm in diameter and 1.5cm thick) with a long cable terminated in a 3.5mm mini-jack. This is plugged into the corresponding front-panel socket and allows the mic to be placed in any convenient position. I found there was plenty of gain available to ensure that talkback messages were clearly audible. Two more 3.5mm sockets on the rear panel are used to accept an optional footswitch for remote talkback operation, and to provide the talkback mic output for use as a slate mic feed if you want to record talkback to a dedicated recorder track. The most unusual feature of the QS8 is that it can be used to sum four stereo inputs into a dedicated balanced stereo mix output. The four QS8 inputs (Aux 1, Aux 2, Aux 3 and cue input) are combined with the resulting mix made available on another pair of XLRs, labelled Mix Output. A Mix button in the control-room source selector section enables the mix signal to be auditioned. The Aux 3 and cue inputs to the summing section are derived after their level controls (and Aux 3’s mono button), enabling these sources to be balanced against Aux 1 and Aux 2 (which both have a fixed gain, of course). However, if Aux 1 or Aux 2 is being monitored in the control room, its signal is removed from the summing bus. The summing output is available at all times and can be used as a second cue-mix section, if required.

The most obvious direct competitor, with a passive control-room monitor path and broadly equivalent artist cue facilities, is the Presonus Central Station, although that model doesn’t have any summing facilities. Another contender with good artist cue facilities, but with a high-quality active monitor chain, is the Audient Centro. The Dangerous Music D-Box has less flexible artist cue facilities, but does include a summing mixer and has built-in D-A conversion for digital sources.

input signals up to +22dBu via the Aux 1 and Aux 2 inputs, although an input level of +24dBu (the default for many pro D-A converters) caused severe clipping distortion. Strangely, the signal-to-noise ratio measured 102dB for the left channel and 114dB for the right for both Aux 1 and Aux 2. Both are good figures, but the disparity is intriguing! Testing the Cue In and Aux 3 inputs, I found that the headroom margin was much lower, with obvious overload distortion for input levels above +12dBu. The signal-to-noise ratio for these two inputs to the summing Mix output was 104dB on both channels. In general, I’m quite a fan of Coleman products, but there’s almost always something that I find a little frustrating. In this case, it’s the absence of any front-panel indication of the mains power status, and the very odd headroom margin issues with the summing system. Nevertheless, the QS8 is an interesting product with some unusual and useful design features. The limited headroom on the Aux 3 and Cue inputs to the mix bus makes the summing element of the design rather less useful than it should have been, but in every other respect this is a very clean and capable performer. The QS8 isn’t a cheap monitor controller option, but it is very clean and transparent, and does pack some handy features into a very compact space. $ T E W
$1299. Coleman Audio +1 516 334 7109.

On the test bench, the summing-section electronics were able to accommodate

w w w . s o u n d o n s o u n d . c o m / January 2012



PSP NobleQ

EQ Plug-in For Mac & PC
PSP’s latest equaliser seeks to combine analogue smoothness with digital precision.

t always seems to me that passive EQs have unmatched smoothness and finesse, while a good active parametric design has finely controllable adjustment, but few designs have both. PSP may have brought that ideal one step closer with their NobleQ and NobleQex. Clearly inspired by hardware passive equalisers such as the Pultec EQ1PA, the PSP NobleQ isn’t an exact copy of anything specific, providing the expected features of passive equalisers


but adding a wider range of frequency adjustment, an adjustable high-pass filter, and the option to switch the high peak and shelf filters to either boost or attenuate. The audio sample rate is doubled within the plug-in to create a smoother, more accurate response, and there’s also a simulated tube make-up gain amplifier with a user-controllable amount of saturation that works so well it could have been sold as a plug-in in its own right. As with most PSP plug-ins, both Windows and Mac OS are supported, as are the VST, AU and RTAS plug-in formats.

Fighting Filters

PSP NobleQ $69
• Musical, analogue sound quality. • Affordable. • Really nice tube emulation.

• Would benefit from a graphical EQ curve display.

If you need a seriously sweet-sounding plug-in that gets very close to what you’d expect from an analogue EQ, this is one of the more affordable options.

To take the basic NobleQ first, the secret to its big-sounding low end is the provision of two low shelving sections: one boosts, while the other attenuates, but at a slightly higher frequency. A Low Shelf Freq pointer knob selects the cutoff frequency for both (although the manual reveals that the actual frequency is slightly higher than indicated). The Boost and Attenuation knobs can then be used together, and interact in a similar way to those on a Pultec EQ, so that as you boost the lows, the lower end of the mid-range is automatically pulled back to prevent it becoming too congested. The centre position on the cut/boost toggle

switch is used to bypass this EQ section, and a further switchable-frequency low-cut filter is available to remove excessive lows. There’s also a dual-filter approach to controlling the highs, with a variable-width peaking filter normally used for boosting and a shelving filter usually used to attenuate. As with the low end, it is the interaction of these two filters that creates the magic, in that the more HF attenuation is applied, the more selective the peaking filter becomes, so that the filter’s centre frequency is boosted more as more shelving cut is applied. There’s also a switchable low-pass filter in this section. Note that extra interim filter frequencies are available between each of the marked positions on the various frequency knobs, so there are more frequency settings than the controls initially indicate. Also on the panel is a master level control, a small adjuster for the valve emulation, to alter the amount of ‘attitube’ (I just thought of that one!) and a three-way switch labelled Valve/Clear/Off. When this is in its Off position, all the filters are bypassed, at the Clear position the filters are on but the valve emulation is bypassed, and in Valve mode you get the filters and the valve. PSP NobleQex, the second plug-in of the set, is basically a PSP NobleQ equipped with a mid-range band-pass filter and something PSP call an adjustable low-shelf dip frequency shift, operated via the Attenuation Shift control in the low section. I asked Antoni at PSP exactly what this did and he said: “It actually shifts the low-shelf attenuation filter up and down by one


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NobleQex adds an extra mid-range equaliser to the basic NobleQ template.

octave”, although the resulting frequency dip is apparently settings-dependent, so you just have to adjust it by ear. There’s nothing unusual about the mid-range section, other than that it has a cut/boost switch rather than the more familiar centre-flat, cut/boost knob found on most modern parametric equalisers. You get separate controls for frequency (switchable in steps from 150Hz to 5kHz), amount of cut or boost and bandwidth, with the option to bypass the section, using the centre switch position, if it’s not required.

In Action
I like how PSP have taken the musical aspects of vintage passive equalisers without feeling the need to recreate their specific flaws or to limit the control range just to be faithful to the originals. The NobleQ is all about vibe, not about emulation, and that really comes across in its smooth, musically expansive tonality. Its tube emulation fattens bass sounds in a very elegant and satisfying

way, seeming also to add definition, while the upper shelf frequency works well on high frequencies such as those in cymbals or acoustic guitars, to add a musical sheen to the sound without becoming abrasive. If I have any criticism of the plug-in, it is one that applies to all similar equalisers, and that is that the user could really benefit from a display showing what the EQ curves actually look like, as the control interactions are sometimes a little too complex to be entirely intuitive. (The counter-argument is that too many people already mix with their eyes, and I’d go along with that, but in the case of analogue equalisers with non-obvious control interactions, I still think a response graph could be very educational.) Sonically, these two equalisers are amongst the best and most

‘analogue-sounding’ EQ plug-ins I’ve used to date, and I’d have no hesitation in using them for mastering, as well as track and group processing when mixing. Even a gentle application makes sounds appear more solid and present, something that is often missing from a digital mix, while the way they are able to smooth the highs without losing presence is extremely appealing. It is often said that a good processor helps glue the various elements of a mix into a cohesive whole, and these equalisers do exactly that. If you need EQ that can polish as well as add weight and finesse to your mixes, this pair of plug-ins from PSP will take a lot of beating. $ $69. W


Sony MDR 7520
Studio Headphones
Sony’s MDR range of closed-back cans has been popular in studios for years, and now includes a new flagship model.

hen choosing headphones for mixing and mastering, most people opt for an open-backed model, as these tend to offer a more natural sound than closed-back designs. The latter tend sonically towards a slight ‘boxiness’, but they offer the potential advantages of good isolation from external sounds and lower levels of leakage — qualities that make closed-back models popular choices for recording artists and recording engineers working on location. In a domestic studio, of course, many people also require a closed-back model, whether to block out ambient noise when mixing, or simply to keep their cohabitants sane! So what makes a good closed-back model for mixing? As well as good isolation and low levels of leakage, what I’m really looking for is a good, fast response across all frequencies, and excellent separation between the different


Sony MDR 7520 $425
• Clean, detailed sound with good separation of different mix elements. • Comfortable. • Good isolation and low leakage. • Low impedance means pretty much any amp will drive these.

elements in the mix. I can live with a slightly coloured frequency response, as that’s something your ears become accustomed to after listening to a range of familiar material, though I prefer a relatively even response. If it’s a model that I want to use for general listening as well — and I’d advocate that as a good way of getting your ears used to the sound — then I’ll look for a model with a low impedance, so that it can be driven by an iPod or similar consumer device. They also need to be comfortable enough to wear them for long periods: that means that your ears need to be able to ‘breathe’ and the headphones need to stay on your head as you move, and that they must do so without pinching.

• Coiled cable won’t be to everyone’s taste. • Inevitable boxiness of closed-back designs. • Not what I’d call inexpensive.

Sony’s MDR 7509s have proved a popular choice. They score highly on isolation and leakage, though personally, I’ve always found them to be a little over-dominant in, and slightly too ‘smoothing’ of, the bass end, almost to the point of sluggishness. They seem to me to work well on acoustic or low-key, laid-back styles, but they do tend to flatter some mixes, hiding a few problems

The MDR 7520 is an excellent pair of closed-back headphones for studio monitoring duties, with a reliable, albeit loud, bass end and excellent separation — and they’re happy being driven by an iPod too, so you’ll be able to get familiar with their sound very quickly!

when working on busy, loud rock music. Sony continue to offer the 7509 but have launched two new models in the range, the less expensive MDR 7510, and the pricier MDR 7520, which is reviewed here. Sony say the 7520 is a completely new model, but I expected it to at least be similar to the MDR 7509, perhaps sharing a ‘family’ sound, but with a few subtle improvements. I couldn’t have been more wrong! Instead, its DNA seems to be shared with a high-end hi-fi model, the MDR Z1000. That’s not a model I’m familiar with, but


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the two certainly look very similar, and the published specifications are almost identical. It may be that they’re slightly differently voiced with frequency-response tweaks, I suppose, but the only major differences appear to be cosmetic — there’s a different finish (I prefer that of the 7520), and the 7520 has a coiled cord, whereas the Z1000 offers both coiled and straight options. The Z1000 is also more expensive! Not having had my hands on the Z1000s, I’ll focus solely on the 7520s. The 7520’s design includes a new 50mm ’high-definition’ driver unit, and Sony say that a liquid-crystal polymer film helps this reproduce frequencies in an astonishingly wide band — from 5Hz to 80kHz — though unfortunately they don’t qualify that with any figures such as ±3dB points, or frequency response plots. The impedance is quoted as 24Ω, which means they’ll work with most consumer, as well as professional, gear. The housing is made of a lightweight magnesium alloy, and the urethane earpads are designed to keep noise out and in, as well as to fit comfortably.

vocals to taste), but the response seemed fast and tight, right across the frequency spectrum. I was able to hear both detailed hi-hat work and what was going on at the bass end very clearly. You’re never going to get that thump of real bass on headphones, as you feel that in your chest as much as hear it with your ears, but the MDR 7520s did as well as I could hope for in a headphone of this type. The only down side is that the level of detail can become a little overwhelming, which is why I rarely like to mix exclusively on good headphones. Counter-intuitively, I find that I sometimes need to be able to ignore little details! Importantly, mix decisions translated pretty well on to other systems, including those about the bass end.

I have few points of criticism. One is that like all closed-back, over-ear headphones, the MDR 7520s impart a certain amount of boxiness to the sound. This was always noticeable, but not offputting, and it never really compromised my decision-making. Also, these are pricier than most of the competition — but I think they’re also rather better, and the price is nothing compared with nearfield monitors of similar quality. I’d much rather rely on these than on cheap nearfields any day! Highly recommended. $ $424.99. W

On Test
I played a range of reference material on the 7520s over the course of a week or two, and when I felt I’d become sufficiently familiar with them, I set about mixing a track, using the 7520s via a Dangerous D-Box headphone amp as my main monitoring system, and occasionally checking back on my usual nearfields and (via the same headphone amp) my Sennheiser HD650 open-backed headphones. Later, I compared the sound of a few tracks played over the 7520s, the 7509s and the Shure SRH940s, which also happened to be in for review at the time. The first thing to say is that the fit of the 7520s (on my head, at least) is good: they’re nice and snug, there’s no pinching, and they fit nicely over the ears rather than pressing on them: very comfortable. The isolation and leakage performance was also decent, but they never felt suffocating, so they’ve hit my criteria bang on. On to the sonics, then. Despite sounding very different from my HD650s, the 7520s ticked a lot of my mixing boxes immediately. Not only was I greeted with excellent separation of different elements in a mix (it was easy to hear into reverb tails, to pick out individual elements in a wall of distorted guitars, and to EQ


w w w . s o u n d o n s o u n d . c o m / January 2012



Propellerhead Balance USB Audio Interface

s all marketing experts and Dragon’s Den (Shark Tank in the US, apparently!) viewers know, every product needs a Unique Selling Point. For audio interface manufacturers it’s usually the particular combination of connectivity, audio and build quality, convenience, portability and cost that each model offers which affords it some uniqueness. It’s much rarer to encounter genuine ‘one-off’ features, but that’s exactly what we find with Propellerhead’s first hardware product, the Balance. I’m going to get on to those Unique Selling Points in a moment, but first of all let’s get all the usual key facts and figures out the way.


We examine the Swedish soft-synth specialists’ first foray into the world of hardware: the Balance.
A simple Direct Monitoring facility routes inputs directly to outputs to achieve nearzero-latency monitoring during tracking, and for when you want to use the Balance independently of your DAW. Used with OS X it’s class-compliant, and doesn’t require drivers, but for Windows users an ASIO driver is supplied. Now for the more unusual features. The Balance has something I’ve never encountered before: front-panel input selector buttons, like you see on many hi-fi amplifiers. These work in conjunction with multiple physical input sockets for each channel: two balanced line level inputs on quarter-inch sockets, a high-impedance unbalanced guitar input with switchable pad on another quarter-inch socket, and a balanced mic input with switchable 48V phantom power on an XLR socket. Clearly the idea is that you can leave multiple input sources permanently connected, and then choose which you record with the input

Propellerhead Balance
• Subjectively good sound quality. • Comes bundled with Reason Essentials (and a Reason 6 upgrade). • Reason-specific Clip Safe button and other features are genuinely useful. • The input selection buttons are handy, letting you leave gear permanently connected

• Expensive if you don’t intend to make use of the bundled software. • No MIDI connectivity, or any digital audio inputs. • Only one headphone output.

Balancing Act
The Balance is a USB2 bus-powered audio interface — in fact it’s only bus-powered, with no mains or battery option. It has a two-in, two-out design and supports up to 96kHz, 24-bit operation. The output channels can simultaneously feed a pair of monitor speakers and headphones, and there are separate volume controls for both.

Propellerhead’s first hardware product is a good sounding and nicely designed audio interface with useful Reason-specific features. But make sure its lack of MIDI and expandability suit your needs before depleting your bank balance!


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buttons. It’s a neat idea. Also uncommon, particularly at this price point, are stepped gain controls: 21 positive notches take each of the two pots from minimum to maximum gain level. The really unique features, though, are all related to the Balance’s integration with Propellerhead’s DAW software, Reason 6. Most notable of these is Clip Safe. Press the front-panel button and a single channel (either left or right, you choose in Reason) is recorded through both channels’ A-D converters, and a pad is engaged on one of them. Then, if your ‘normal’ recording clips, causing nasty digital distortion, you can choose

There are two more Reason-only features. The first is a Meter/Tuner button, which brings up a floating level meter window and guitar tuner display. The second is a Ready LED next to each channel’s gain pot that confirms the channel is routed to a record-enabled Reason track.

Sound Quality
I’m going to resist all temptations towards poetic rambling and simply say that the Balance sounds just fine. The recordings I made during the test period, via all the inputs, were full-sounding and detailed. When compared to my RME Fireface

“Used in conjunction with Reason, in a typical single-user, laptop-based home-studio scenario, the Balance does everything most users will need it to.”
share of premium small interfaces like Apogee’s Duet and RME’s Babyface. I personally don’t see that, as those interfaces are in a higher price bracket again. Nor do the Balance’s published specs point to such an intention; they’re plenty good enough for the roles it’ll

Clip Safe gives Reason 6 and Essentials users a back-up against clipping distortion. When it’s enabled, sections of audio clips that have clipped are shown by red areas. All is not lost — just click the clip’s ‘CS’ button or choose the ‘Heal Clip Safe Clips’ menu item and it’s substituted with a distortion-free version.

(again in Reason) to ‘heal’ the take with the padded version, which will most likely be clip-free. Reason applies the necessary gain boost, in the software, to bring the padded take up to the original desired recording level, and catastrophe is averted. If there’s a down side, it’s that Clip Safe effectively turns the Balance into a single-input interface. But within that limitation the whole process is slick and easy to use. Propellerhead call it a “red-eye tool for recorded audio” and that sums it up nicely.

UFX — a completely different beast, five times the price — they didn’t give much away. I fancied the RME sounded a bit more transparent in the mid-range, and the stereo image more generous, but there wasn’t much in it. And even then, this was not true double-blind, instantaneous-comparison testing, so it’s all a bit subjective. Certainly I could hear no obvious flaws. Because of the Balance’s not-insignificant price tag, there’s been some Internet forum talk about Propellerhead going after the market

typically fulfil during its working life, but not state-of-the-art. Propellerhead’s marketing blurb says the preamps and converters are ‘Hi-end’ and that’s all. Turning to some specifics now, I can report that the Balance’s stepped gains are a mixed blessing. On the bright side, the 40dB or so gain range is not bunched up towards the top of the pot’s travel. The gain change between detents, though, is very variable, with much bigger jumps in the middle of the range. Also, on the review unit it proved impossible to exactly match gain amounts on both

Bundles, Upgrades & Authorisations
The Balance is bundled with Reason Essentials, a Mac OS/Windows DAW. Compared to the full-blown Reason 6 it lacks some of the tastier features and devices, notably audio clip transposition, the compressor/gate section on mixer channels, the instruments Kong, Thor and Malström, and the effects devices Alligator, Pulveriser, The Echo, Neptune and Vocoder BV512. The factory sound bank is smaller too. But it’s still a perfectly serviceable DAW, with unlimited audio and instrument tracks and Propellerhead’s notably good real-time audio time-stretching. For those who already own a version of Reason, or Propellerhead’s previous DAW, Record, buying the Balance also gets you an upgrade to Reason 6, which by itself would cost $169. One less obvious but very welcome feature of the Balance is its ability to double as an Ignition Key dongle. The Ignition Key is Propellerhead’s copy-protection system, and usually comes in the form of a little USB stick, but the Balance has the same functionality built in, and that will make life easier, and use up one less USB socket, for some users.

This render of the Balance’s back panel shows the unit’s stereo outputs, dual stereo line inputs, guitar inputs with pad buttons, and XLR mic inputs with switchable 48V phantom power. At the top left is a USB port to connect the Balance to your computer.

w w w . s o u n d o n s o u n d . c o m / January 2012



channels when they were sent the same test signal; there was always a discrepancy of between 1 and 2dB. Of course, this can be easily remedied in your DAW, but it’s not ideal for stereo recording. Figures for maximum gain are not published anywhere, but an informal comparison to other preamps reveal it’s probably not more than 60dB. Interestingly, exactly the same amount of gain appears to be available through the guitar and line inputs as through the mic XLRs, which suggests they all share the same preamp electronics, but in any case noise performance at all gain amounts seems subjectively excellent. The headphone amp doesn’t appear to be particularly powerful. My Sennheiser HD650s required the headphone volume pot to be turned all the way up for a really healthy level, though my HD25s demanded a little less of it. Finally, a word about the Direct Monitoring option. When engaged, the input level fed to the headphone output appears to be fixed, and curiously, the resulting monitoring level is lower than that which comes back via software (I used Reason 6 during testing). So, when using Direct Monitoring you’ll often have to lower your DAW’s master fader to get a good balance between the input and the DAW guide track. An additional input/playback balance control would have been welcome here.

Propellerhead’s Balance. The wedge shape is modern and funky, underlined by the very cool black rubberised finish and red underside. Build quality feels really good — the rear sockets are secured to the casework and the pots are ultra-smooth and positive. Used in conjunction with Reason, in a typical single-user, laptop-based home-studio scenario, the Balance does everything most users will need it to. It lets you lay down vocals and guitar tracks with the greatest of ease,

System Requirements
• Mac OS: Intel processor with OS 10.6.3 or later. • Windows: Intel Pentium 4/AMD Opteron or better, Windows XP SP3, Vista or Windows 7. • Regardless or your operating system you’ll also need 1GB RAM, about 23GB free hard disk space (for installation and ‘scratch disk’ usage), a DVD drive, a free USB2 port and an Internet connection for registration.

An Equal Music
From a purely visual and tactile point of view, it’s hard not to admire

If you only want to match the Balance’s basic two-in/two-out design spec, you don’t need to spend anything like its asking price. The long-established Tascam US122 MkII is a mere snip at around $119 (street price), has MIDI In and Out and is more portable. Spend only a little more on the likes of Focusrite’s Scarlett 18i6, at $299, and you can comprehensively trounce the Balance’s connectivity with eight analogue inputs, S/PDIF and ADAT expandability and MIDI. But as I mentioned elsewhere, nothing else out there has the Balance’s Reason-specific features. If you want those then the Balance has no competitors.

all backed up with the reassurance of the Clip Safe feature and those other Reason-integration niceties. It takes up very little desk space, and the combination of multiple input sockets and source-selection buttons can help keep your workflow fast and your working environment neat and compact. Take the Reason-specific features out of the equation, though, as users of other DAWs inevitably will, and the Balance doesn’t seem quite so attractive. At its heart, it’s a decidedly standard two-in/two-out interface. It’s got no MIDI connectivity, so forget sequencing your hardware synths. The single headphone output seems stingy — you’ll have to use an additional headphone distribution amp, or at least a splitter, if you want to record a singer in the same room and still monitor proceedings. It’s a shame there’s no ADAT or S/PDIF expandability too. Yes, you can do a lot with two channels, but any meaningful drum kit recording, or tracking of several musicians at once, is pretty much ruled out. And here are two more gripes: the wedge-shaped design is striking, but it prevents the Balance from fitting in a laptop bag very well. And despite its sporting many features that

Pressing the Balance’s Meter/Tuner button calls up this large floating input level meter and tuner window in Reason 6 and Reason Essentials.

seem to lend it to use in the educational field, some sort of security anchor is conspicuous by its absence.

It all boils down to your individual requirements. The Balance is not an especially versatile interface, nor one that will grow with you as your recording projects become more complex and you acquire more outboard gear. What it does have, though, is respectable audio performance and build quality, and a focused feature set that will be perfect for many bedroom-based musicians. The inclusion of Reason Essentials, and a Reason 6 upgrade, is clearly part of the value of the product and absolutely cannot be overlooked. And those Reason-specific features are truly unique. Only you can decide whether, on balance, this is the interface for you. $ T W W
$479. Line 6 +1 818 575 3600.


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PRECISION KEYBOARD - INSTANT MAPPING Want to get hands on with your music software? MIDI keyboards often have knobs and faders. Nobody uses them because mapping and re-mapping is so tedious. Impulse comes with Automap 4 control software. This gives you instant access to your DAW’s mixer, transport and plug-ins. You can re-assign controls with the touch of the ‘learn’ button, and see what is mapped to what – Instant Mapping, Instant Music Making. • Precision keyboard with aftertouch • Automap 4 control software included • DAW/Plug-in control surface • 8 drum pads with Ableton clip launch


For more information: (310) 322 5500


Kinman P90HX Noiseless Guitar Pickups
It seems that Chris Kinman seems has embarked on something of a one-man crusade to create the perfect noiseless pickup. Regular readers will know that he’s already created a noiseless pickup that delivers the look and sound of Strat-style single-coil pickups, and he’s now tackled the Gibson P90. Guitars fitted with P90s can be particularly troublesome in a studio, as their large coils make them very susceptible to hum pickup, but most of the hum-cancelling alternatives change the tonal attributes that draw players to P90s in the first place. Having suffered hum problems with two of my Gibson guitars, I bought one of the new Kinman P90s for my Les Paul Junior and a neck/bridge pair for my SG. They aren’t exactly cheap, but when you consider they incorporate 202 individual parts (only 13 are used in a standard P90, and 23 in other noise-cancelling designs), they don’t seem like bad value! I fitted them to both guitars, and to my ears the tone was at least the equal of the best P90s Gibson has to offer. In fact, I thought the dynamic response was better: I heard sweet highs and warm lows but without any mushiness. Kinman tells me that he tweaked the neck pickup to slightly improve the clarity, and with the SG the ‘both on’ setting yielded excellent results. Importantly, there was absolutely no audible hum! There are a few practicalities to bear in mind. These pickups are deeper than standard P90s (the depth is about the same as Gibson’s own P100 noise-cancelling pickup), because of the hum-cancelling coil, so on some guitars you’ll need to make the pickup cavity deeper. On my Les Paul Junior, I had to use a router and a hand-made template to deepen the cavity by around 4mm, but this went smoothly and the pickup dropped right in. With the SG, though, the cavities were already deep enough. The pickups come with covers (black or cream), short rubber-tube ‘springs’ and thin woodscrews that pass through the pickup for mounting. I used these in the SG, though I had to clip off the tips of the screws after ‘starting’ the holes to avoid the risk of them breaking through the back of the guitar in the neck position. However, in the Les Paul Junior, the pickup cover was a ‘dog ear’ type, so all I had to do was

place some foam rubber under the pickup to act as a spring and then screw the cover down on top. The pickups terminate in short pins that are compatible with Kinman’s DIY ‘no-soldering’ harnesses, so I cut these off, joining the cold and ground wires together. Standard P90 ‘soapbar’ covers fit these pickups, so you can keep your old pickup covers if you prefer. The difference with these pickups is remarkable. You still have the classic P90 sound with the benefit of a little more focus from the neck pickup. And you can record without the fear of hum. That’s what I call a result! Paul White $ £99 each. W

Electro Harmonix Ravish
There’s nobody quite as far ‘out there’ as Electro Harmonix when it comes to stomp-box effects. Their latest offering, the cheekily named Ravish, is being promoted as a sitar emulator for electric guitar, but it is far more customisable than that description suggests and it allows the user to coax some very musical abstract sounds from it. A sitar has between 11 and 17 ‘sympathetic’ strings that aren’t plucked, but instead drone in sympathy with the main strings, and it’s the resonant sympathetic string feature that’s arguably the most important part of this pedal. Electro Harmonix have combined their impressive ‘Freeze’ infinite sustain technology (used in their pedal of the same name) with some clever filtering that allows you to set the scale key, so only notes in the chosen scale excite the sympathetic strings. Up to nine user presets can be stored, and you can select from ready-made major, minor and ‘exotic’ (Indian) scales in any key. You

Sitar Emulation Stomp-box Effect
can also store custom scales of up to 17 notes, so you can include quarter-tones. A subtle, user-adjustable modulation can be added to the sympathetic strings to create a tambura-like drone and, as with the Freeze pedal, pressing a switch allows you to hold the droning sympathetic string sound indefinitely. If not held, they die away a few seconds after the original note has faded. There are separate outputs for the main and sympathetic voices, which could be handy if recording, but you can also use a single cable to provide a mixed output. The Ravish pedal offers the user plenty of control over the sound. Of the five conventional control knobs on the device, three govern the levels of the original guitar sound, the sitar-processed lead sound and the sympathetic drone sound. The other two control the respective timbres of the lead and sympathetic drone sounds, making them progressively brighter and more harmonically complex as they’re turned

clockwise. Further timbral changes can be made using the Resonance control, of which more in a moment. Other important functions include using the rotary encoder to select the musical key and scale type, the decay of the lead sound, the amount of modulation on the sympathetic drone sound ( a slow and subtle chorus-like effect) and the resonance (when both the decay and mod LEDs are lit) of both the lead and drone sounds. This control is also used for saving user settings. The leftmost footswitch can be tapped briefly to scroll up through the presets (all of which


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can be overwritten), while holding it down during performance causes the sympathetic string sound to sustain indefinitely, even if you unplug and set fire to the guitar! In the key mode, you can also hold this pedal down to input new scale notes directly from your guitar. The rightmost switch bypasses the pedal, with the guitar signal being buffered before reaching the output. With a pitch pedal connected, the pitch-bend range can be set anywhere from one semitone to an octave upwards in pitch. When the left switch is pressed to freeze the drone strings, you can still adjust their timbre using the other controls. Furthermore, a pedal connected to the drone jack can both adjust the level of the drone strings and initiate the Freeze function. You can also control the volume of the sympathetic drone using a pair of optional expression pedals. In terms of sound and performance, the sitar emulation is best considered an

approximation — the hex-pickup systems from Roland and Line 6 probably do a better job in this respect. The lower strings sound more convincing than the higher ones, but all this said, the general impression of ‘sitarness’ is still pretty solid and it responds well to playing technique. The decay can be adjusted from a very short pluck to a much more extended note, while the timbre can go from warm and gentle to an almost synth-like resonant twang. Mix this with the main guitar sound, possibly processed, and the creative possibilities can seem endless. The real icing on the cake is the smooth way those sympathetic strings pick out the scale notes and drone away in the background. If you play a chromatic scale, you can definitely hear that only the scale notes excite the resonances. Used on their own, these can be used to create synth-like pad sounds and washes, so their application goes far beyond mere sitar emulation. There’s a somewhat

spring-reverb-like character to the drone strings — but because the Freeze pedal used a modified infinite reverb algorithm this is not unexpected, and the result sits nicely beneath the lead sound. There’s no ‘all notes’ chromatic drone scale as standard, but you can create and store one to allow the Ravish to be used more like the Freeze. I love the sheer creative audacity of the Ravish pedal, and I certainly wouldn’t use it only for making sitar noises. In fact, it can actually get pretty close to sounding like a genuine guitar synth, and it could easily form the basis of some rich pad sounds on stage or in the studio. Teamed with plug-ins, it could also provide the basis in DAW recordings for a much more warped and layered sound. I’ve been very impressed by both the Freeze and Ravish pedals — so let’s hope they’re just the thin end of Electro Harmonix’s creative wedge! Paul White $ $239.95 W

Zaor Stand Monitor 30.002/C Speakers Stands
Hopefully, we all know about the importance of supporting our monitoring loudspeakers on a firm and stable platform and, thankfully, there are countless monitor stands available. Some designs are beautiful to behold, while others allow the function to overpower the form! Few are height-adjustable, though, and most that are tend to be feeble and unstable. However, I recently stumbled across an Italian company called Zaor, who build a wide range of really interesting and attractive studio furniture, equipment racks, speaker stands, work consoles and hi-fi cabinets. The Italian styling is evident throughout, as is the attention to detail in the clever and often unusual designs. Of particular interest is the company’s Stand Monitor, which is height-adjustable from 80-129 cm using a simple but effective peg system, and is able to support speakers up to 65kg! The floor baseplate measures a modest 32cm square, and it has four adjustable flat feet, while the top plate is slightly smaller at 28cm square, and has an Aerstop expanded rubber insert to prevent speaker movement. At additional cost floor and speaker spikes are available, as is a special high-gloss black or white paint finish. The review model had a black-ash style finish, with a silver centre column. The construction employs a rectangular wooden tube rising from the base, inside of which an internal frame guides a separate rectangular shaft attached to the top plate. A spring-loaded metal peg installed in the outer tube locates in one of 12 metal sockets installed along one of the narrow sides of the top-plate shaft, with roughly 4cm spacing to determine the height of the stand. Despite the simplicity, this appears to be a very solid and stable arrangement. The stands are intended to be used with the wider aspect of the central column facing the listener and the locating peg on the outside edge, but I preferred to use them with the peg facing forwards, as in this arrangement the small amount of residual free-play between the inner and outer columns is on the side-to-side axis, rather than front-to-back. Just to be nerdy, I also packed some stiff furniture foam off-cuts into each side of the shaft to damp any remaining tendency to wobble or vibrate, as well as to minimise any acoustic resonances from the base tube itself. Being constructed from solid wood and plywood, the stand has virtually no self-resonances (unlike most metal stands), and I didn’t detect any audible ‘organ-pipe’ resonances from the open support column, even before installing the foam off-cuts, no doubt partly because the column has a large cut-out at the bottom enabling cables to be hidden by running them down the inside of the tube. The Zaor stands are very well built and sturdy, of a manageable weight if you need to move them, and quick and easy to adjust. The height range covers pretty much all bases from lounge seating to standing with most small and medium monitors. In fact, I was so impressed that I bought the review models. Hugh Robjohns $ $695 per pair. W

w w w . s o u n d o n s o u n d . c o m / January 2012



Turbosound Milan Mi0

he recently-introduced Milan range of portable self self-powered selfpowered speakers from Turbosound consists of three models, and comprises 10-inch and 15-inch two-way, way, full-range powered speakers and a powered 18-inch subwoofer. The new baby£ of the family is the 10-inch model, the Milan Mi0, which we’re looking at here. There’s a wide choice of self selfself-powered powered speakers on the market these days. Smaller units with eight- and 10-inch woofers can be found everywhere from garden fetes to upmarket bar installs, so any newcomer needs to have some sort of differential in order to attract attention, whether it be price, power or portability, for example. The Milan Mi0 has two advantages right out of the box, in that it says ‘Turbosound’ on the front, which is always a nice thing to have written on one’s live gear, and it’s a bit of a looker. Whatever it contains inside and whatever the performance figures say, the Mi0 is a stylish little item that’s very pleasing to the eye.


Active Loudspeaker

The Turbosound Milan Mi0 enters a competitive market. Can this little newcomer stand out from the crowd?

Construction Sight
I’m generally a fan of plain, square-ish black speaker cabs, as much of my work involves corporate clients who want to hear the sound but don’t wish to know where it comes from, but my first impression on unwrapping the Mi0 was that it would be

Turbosound Milan Mi0
$799 each
• Powerful, light and compact. • Excellent vocal clarity. • Easy to operate. • Attractively styled. • Technical support and spares available.

• Nothing significant.

The Turbosound Milan Mi0 is an extremely well constructed and impressive self-powered speaker. The cabinet design is practical, yet has a spark of sleek and individual style, and controls are simple to get to grips with. Sound quality and vocal projection are both top class, making this portable speaker a top performer all round.

at home at any posh product launch or business presentation. The attractive cabinets are made from the ubiquitous polypropylene using a gas-injection moulding process. As I understand it, this involves the use of nitrogen gas under pressure to counteract shrinkage during cooling of the moulded material, producing an accurate shape and a nice surface finish. Apart from the stylish dark-blue colour, which Turbosound call ‘black-blue’ and is specific to the Milan range, the moulding is clearly of high quality, including fine ribbing down both sides, as well as crisp, sharp detailing. Since everything is moulded into the enclosure itself, there are few bolt-on parts to rattle or work loose over time, and the review models certainly didn’t make any untoward noises when bumped around on a hard surface. On top of the cabinet, there is a single ‘scoop’ handle that offers a secure grip

and is quite adequate for lifting the Mi0, which weighs less than 12kg. The Mi0 is symmetrical, and you can treat either edge as the top or bottom when you’re using the speaker as a floor monitor. This allows a choice of where the high-frequency (HF) end is located, which is useful for very close or individual monitoring applications, such as an orchestra pit with very limited space. For ‘flying’ or permanent installations, M10 rigging points are provided on the top and bottom, so the cabinet can be positioned vertically or horizontally. Working with portable speakers at a range of venues, I value the quality of the metal front grille, as it tends to take a lot of punishment and is, of course, what the punters/clients will see most of. The Mi0 has a strong galvanised-steel grille with a matte silver, powder-coat finish. The grille sensibly covers the whole front of the cabinet and derives good rigidity from its folded edge shape where it fits tightly,


January 2012 / w w w . s o u n d o n s o u n d . c o m

securely and neatly into the bodywork. It serves as an example of just how well this cab is put together that there are actually no fixing screws holding the grille in place, it’s just a very accurate push fit and would have to be carefully prised out if necessary. There’s no foam liner inside the grille, and it’s possible to see the driver components inside, but the punched metalwork should offer reasonable protection against the odd accidental spill or splash. I like how the rear of the cabinet extends way back beyond the control panel, providing a level of protection for the controls and heat-sink fins that’s much better than usual. This is an excellent design point. On the bottom of the cabinet, there are three fairly soft, grippy rubber feet of good size, which are far better than the hard plastic scratch-n-slide things favoured by some manufacturers. The pair of pole-mount sockets provides a choice of mounting angle: the speaker can be mounted so that it faces directly forward at a right angle to the ground, or it can be angled downwards at the audience. In most smaller indoor venues, the latter would be the one to go for, as the sound is aimed at its intended target and the dispersion of the horn will still ensure that a decent balance is achieved for those at the back.

compression driver mounted on an elliptical waveguide, resulting in 90 x 60 degree high-frequency dispersion. The overall frequency response is quoted as 55Hz to 22kHz at -3dB. The rear panel hosts all the connectors, controls and indicators. Everything has been kept simple and straightforward, and the panel layout is easy to understand at a glance for users who may be unfamiliar with this type of gear. There are two XLR inputs, microphone and line, which have independent level controls and can be mixed. The line-input XLR is also able to accept balanced or unbalanced jack plugs, which is handy for solo performers or situations where an external mixer isn’t needed. I tried using the Mi0 as a keyboard amp, and it performed very well. The mix output that’s sent to the internal amplifier section is also duplicated by a balanced XLR line output, for onward transmission to another cabinet or audio destination. The Mi0 includes a simple two-band shelving equaliser with center-detented pots. As the manual suggests, they’re useful when applying a bit of bass (and maybe treble) cut in a floor monitor application.

HK Audio’s FAST active PA speaker provides a similar level of performance to the Mi0, as do QSC’s K8 and K10 K-Series active Speakers.

Power-up & Away
Switching the Mi0s on is an undramatic and quiet process, with a couple of very small clicks as the LF and then the HF sections come to life. Switching off and on again quickly didn’t produce any unwanted pops, and any idling noise comes mainly from the LF section, with very little hiss from the horn. Overall, the noise level while idling is low, and acceptable for performance applications, though it might be distracting in a very quiet environment. I compared it to two other active speakers from wellknown manufacturers, and found that the Mi0 produced much less noise than one and slightly less than the other. The SPL quoted in the published specification is 119dB continuous with 125dB peak output, and I can confirm that these little speakers definitely pack a lot of wallop. I hooked them up with and without a subwoofer; in full-range mode they had a surprisingly full sound with plenty of warmth and depth, and I didn’t find the sound fatiguing even at high levels. When pushed hard, they simply delivered more output, and remained well-balanced, like a pair of powerful monitors. At low volume levels, there was still plenty of warmth to the sound and, for me, their real forté was clarity and projection in the vocal range.

When I ran the pair of Mi0s with the Milan M18 subwoofer, I was able to produce even higher overall levels of volume, as the Mi0 speakers were concentrating all their power on the high-passed signal fed from the sub (everything above 80Hz or 125Hz, depending on the M18 setting), rather than wasting power trying to reproduce the lowest frequencies. Despite their small size, the Mi0s do make a respectable rig at very high levels when paired with a decent sub! I took a single Mi0 outside for an open-air speech test in a field, pointing away from any buildings, and it performed impressively, covering a much greater area than I had expected. I was also able to use the speakers in a rehearsal for amplifying male and female vocals over a 20-piece band. I’d definitely put them in the top flight of powered portables for this task. I also used them to reinforce the top trumpet line of the band, both close-miked and as a section, and was able to produce a natural bright sound that easily cut across without harshness, blending nicely with the natural, un-miked sound of the rest of the band.

The Mi0’s amplifier module is housed in the back of the cabinet, with a control panel below it. The power-amp section includes a 180W (continuous) class-D low-frequency amp and a 50W class A/B design for the high frequencies. Drivers are a specially-designed, reflex-loaded neodymium 10-inch woofer and a one-inch

Milan Magic?
One of the things I like about Turbosound products is the availability of spares and, should anything go wrong, all the parts (including spare front grilles) are readily available to get your Mi0 back into service quickly and easily. There’s a parts list on the web site, together with a technical support service. Also, if you call the company someone will speak to you. Apart from finding a home with musicians, DJs and independent live sound providers like myself, I should imagine the Mi0 will find its way into many upmarket bar installs. Its versatility and power-to-weight ratio, combined with the strong design and build, should also fit the rental criteria for speakers of its size. I really like the Milan Mi0 powered speaker. It’s an impressive performer, is very well-constructed, and combines versatile practicality with attractive and individual design and styling. $ T W W
$799 each. AMS +1 800 4312609

The rear panel of the Mi0 is quite simple, including high and low shelf EQ and a simple mixer for the line and mic inputs. It’s also well protected by the extended rear of the gas-moulded cabinet.

w w w . s o u n d o n s o u n d . c o m / January 2012



PreSonus StudioLive 16.0.2
The StudioLive range of mixer interfaces now includes a compact 16-channel model. We find out how well it performs on the road.

Digital Mixer & Firewire Interface
PreSonus StudioLive 16.0.2 $1299
• Easy to use. • Well thought-out layout and design. • Very small and light. • Remote control capability.

he choice between an analogue and a digital mixer for live sound use is an interesting one, especially where small-format desks are concerned, because they’ll tend to be used for a wider variety of perhaps more modest events. They’ll often be used by ‘volunteer’ operators too, who don’t spend all that much time behind the console, and will need something that’s easy to understand and operate. That’s where the Presonus StudioLive 16.0.2 comes in. A dual-purpose desk


that aims to offer ‘digital facility with analogue simplicity’, the 16.0.2 is a compact, portable 16:2 design with 12 mono microphone inputs plus additional stereo input options on the highest four channels. It has plenty of built-in DSP and effects, as well as four auxiliary (aux) sends. There’s also a two-way, 16-channel Firewire interface for recording and playback, a MIDI control interface, and a dedicated talkback facility. Although it’s designed to be suitable for studio recording too, I’m going to focus here on using the 16.0.2 as a stand-alone mixer for live use.

• Slightly sticky fader action. • Lack of dedicated output meters.

The Presonus StudioLive 16.0.2 looks and feels like a simple little mixer, but has capabilities and performance approaching that of something a lot bigger. It also acts as a great introduction to digital mixers, combining the simplicity of an analogue mixer with the versatility of a modern digital desk. A compact and versatile piece of equipment.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Each channel has a dedicated rotary input-trim control and everything else, including the meters, is shared with other mixer channels or functions. The stereo output has its own dedicated fader, as do the four analogue aux sends. After reading the bits at the front of the Quick Start Guide to make sure I didn’t miss anything really obvious, I unpacked the 16.0.2 and started playing about straight away, to see if I could find my way around it without any instruction. The first and most obvious thing I noticed about the desk in use was that it doesn’t have motorised faders or any fader bank or layer arrangement. The 16 channel faders control the corresponding channel output levels and that’s all they do: there’s no possibility of confusion or of adjusting the wrong ‘layer’ by mistake. A strong point of digital mixers in general is the ability to store, copy and paste data settings, so that all effects parameters and settings can be copied and applied to another channel with a couple of button presses. They can also be stored as favourite settings for future use. The StudioLive 16.0.2 provides 80 scene memories that will capture comprehensive mixer snapshots for later recall. Various settings can be included or excluded from this process too. There are also some good ‘first base’ presets included for the more common channel assignments, such as kick drum, jazz piano and so on, which might be helpful while you’re getting used to the StudioLive 16.0.2.

channels, taking the total count to 16 inputs. The channels all have solo-in-place buttons that perform a second duty as channel mute buttons, lighting yellow for solo or red for mute. After passing through the trim control, the signal is processed by what Presonus call the ‘Fat Channel’, which provides non-effects processing such as dynamics and EQ. This is controlled and monitored by all the knobs, buttons and indicators on the main panel area above the faders. This area also hosts the 12 LED strip meters that display most of the adjustable parameters. A bank of buttons over to the left of channel one determines the function of 12 rotary encoders. These can be used to set the channel-send levels to the four aux mixes and the two internal effects buses, and are also used for EQ adjustment. Whichever function is selected, the LED meters will follow the action, and auto-range accordingly. Using the ‘Fat Channel’ is very straightforward: you press the appropriate channel select button and the ‘Fat Channel’ will control the selected channel until you switch its attention elsewhere.

very simple. It’s a single-knob, one-meter affair, so you just dial in the threshold you want. Just like the filter, the LED meter is directly above the encoder, this one calibrated in dB from 0dB at the top to -84dB at the bottom, in 6dB steps. The StudioLive 16.0.2’s compressor section assigns four encoders that are used to set the threshold, ratio, response time and make-up gain. Each control’s LED meter has an appropriate scale printed alongside, except for ‘response’, which just says ‘smooth’ (for vocals) near one end and ‘tight’ (for snare) near the other. That’s probably as much as is needed here. The compression ratio can be adjusted from 1:1 to 14:1, which should be enough for any live situation. The response time can also be set to ‘auto’, disabling the response encoder and applying broadly useful fixed values of 10ms attack and 150ms release. A separate limiter can also be engaged if needed, recommended for ‘danger channels’ that might have to handle occasional very hot signals. This is something I regard as mandatory on any channel assigned to the man on the disco.

Channels & Dynamics
Sticking with the signal flow, every input channel includes phase reverse, followed by a high-pass filter with a fixed 6dB-per-octave slope and a range of 24Hz to 1kHz — a range that’s printed alongside the accompanying LED meter. I like this simple visual indication, as it makes setting this all-important function as easy as it can be. I’d really have liked to see an ‘all channels’ view, as high-pass filtering is an easy setting to overlook when you’re rigging up a live show in a hurry. Having said that, the 16.0.2 is a digital desk with comprehensive scene recall, so I expect that filter settings are something I’d save into a ‘blank’ memory and use as a starting point for every new setup. Next we reach the gate, which again is

The basic channel equaliser covers three swept bands with a useful amount of control. The low and high bands can be used in either band-pass or shelving mode, and each has an encoder and a display used for setting the centre or shelf frequency, as well as another for the gain setting: a cut or boost of up to 15dB. The mid-range band can be centred between 260Hz and 3500Hz, overlapping the low EQ by 100Hz and the high EQ by a useful 2kHz. This practical range means that a smooth response ought to be easily achievable. In addition, the mid-range EQ section has a ‘Hi Q’ option, which increases the mid-frequency Q curve from its default value of 0.55 to 2.0, providing a much sharper tool for finding and dealing with problem frequencies. The adjustable EQ is a useful feature:

A Guided Tour
The best way to describe a mixer’s basic functions is to follow the signal route roughly from input to output. Channels one to eight are mono strips that can take either mic or line signals via balanced XLR or separate TRS connectors. The remaining four channels have mono inputs but are also configured as stereo

Getting Graphic
In addition to the three-band EQ available within the Fat Channel, there is a built-in, 31-band ‘GEQ’, which can be applied to the main bus. This function makes use of all 12 of the meters, the encoders and the LCD display, and provides much more accurate control over the mix than is possible with the three-band system. With only 12 meters, you can’t see all of the 31 bands from 20Hz to 20kHz at once, so they’re divided into three groups that are displayed according to which individual band you select on the LCD screen. It sounds like a fiddly system, but in practice it’s quite easy to use, and the setting can be stored in a ‘scene’ memory along with everything else.

w w w . s o u n d o n s o u n d . c o m / January 2012



The rear panel of the StudioLive 16.0.2 squeezes in plenty of connections, with the XLR inputs conveniently positioned at the top.

it’s not quite a fully adjustable EQ circuit, but nice on a desk this small and simple, and I found the mid-range sweep to be nice and responsive. I spent some time with the EQ on individual sound sources and I liked working with it, never having any difficulty obtaining the sound I wanted. As with all the elements of the ‘Fat Channel’, I found the EQ easy to use and, once I got used to the linear indications rather than a graphic screen display, I didn’t find any difficulty in achieving basic working settings. The EQ isn’t aggressive and there’s no lag between adjusting the controls and the display responding, something I found tended to make me a little less heavy-handed and a little more precise when making gain changes. The pan function has an encoder of its own, along with a horizontal meter. Being a rotary encoder rather than an analogue pot, the control doesn’t have a centre notch, but the meter has a red segment in the centre, which stays lit and makes the middle easy to find. This control operates as a stereo pan on any paired channels, including paired aux buses.

Buses & Monitoring
As the StudioLive 16.0.2 is a straightforward 16:2 mixer with no in-line groups, routing options are confined to the aux and effects bus settings. Sends to the four aux buses are controlled by the rotary encoders, depending on which of the auxes is selected. The aux buses can also be linked to create one or two stereo outputs. The main stereo output is under the control of a single fader, and — like the aux buses — can be selected on its own, so that all the effects within the Fat Channel can be applied. Your main mix is always where the faders are, so it’s possible to look at a graphic representation of the full mix and one aux mix at the same time,

which is very useful indeed. The StudioLive 16.0.2’s master section includes good talkback and monitoring facilities for a mixer of its size. There is a dedicated talkback mic input, situated on the back of the mixer with its own trim control, and the signal from that can be assigned to the aux buses in pairs. Talkback can’t be routed to the main stereo mix, which is perhaps both a good and bad thing depending on how you work. The ‘talk’ button latches the talkback function, which is a pity, but presumably it would only take a bit of software tweaking in a future update to provide a ‘push and hold to talk’ option. As for monitoring, there’s a solo bus that can pick up the content from any channel, including the aux buses, and can be switched to be pre-fade or post-fade. The monitor bus is used for feeding a control-room output on the rear panel, and also for supplying the headphone mix, which has an independent level control. This bus takes its input from the solo bus, the main bus and the Firewire return, and these inputs are summed so that you can select all three at once, if required. The headphone output is sensibly positioned on the front edge. This is a pretty comprehensive monitoring section, and wouldn’t be out

of place on a larger and more expensive mixer. The control-room outputs could, in a live application, be used to make an analogue recording or as an extra output for feeding a backstage relay.

System Control
There are many areas of the StudioLive 16.0.2 that I haven’t space to describe in great detail here, notably the Firewire recording capability and MIDI control, but those fall outside my basic remit of reviewing this desk as a candidate for live sound mixing. Of relevance to all applications, however, is the ‘System’ menu, which gives access to various global and housekeeping settings. It’s here that ‘pre’ and ‘post’ aux-send choices are made, the LCD backlight level is set, and security lockout levels can be changed in order to prevent inquisitive fingers from ruining your painstakingly crafted mix. For the more adventurous, as well as where the application requires more than just the hands-on direct control available from the 16.0.2 as a stand-alone desk, the StudioLive can be controlled from a computer, using the Virtual StudioLive software. It can also be remotely controlled from a laptop, iPad or iPhone using StudioLive Remote. When the desk is hooked up to a PC

Extra Effects
The StudioLive 16.0.2 has two internal effects buses, which are accessed in exactly the same way as the aux buses. The amount of signal fed to the effects processors is controlled by appropriate encoders, and an overview of the send levels is shown on the meters when the EffectsA or EffectsB button is pressed. The processor itself contains 100 presets that can be recalled by either of the two processors. Presets 1-50 are factory patches, of which you to change the various setting and save in place or to a different location between 51 and 100, all of which are reserved for user-built presets. The effects comprise a range of reverbs and delays, and I liked the most natural-sounding of the room and hall reverbs, which were very usable in their factory default states. Effects, particularly ‘straight’ reverbs, are, of course, a matter of personal choice, but I liked what the StudioLive 16.0.2 has on offer, especially the ability to tune parameters. I was thinking that the local Gilbert & Sullivan Society’s concert might benefit from an application of ‘Ping-Pong Purple Rain’ delay but, sadly, the right moment never quite arrived.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Meter Maid
The StudioLive 16.0.2’s metering is comprehensive and clear, and the only part I found confusing at first was the main EQ section, mainly because I was looking at it as a graphic rather than a parametric layout. The twelve LED meters are at the heart of most operations on the 16.0.2, and are certainly large and bright enough to be read easily in brightly-lit venues and, I suspect, even in bright sunlight. To the left of the faders there’s a little panel with meter options, and the meters can display the channel input levels (post-trim, pre-fader), gain reduction currently being applied to all 16 inputs, or the output levels for the aux and main stereo buses. The one thing I really missed on the 16.0.2 was a pair of dedicated output meters, plus I couldn’t find a way of viewing the main mix level at the same time as the input levels. Because the StudioLive 16.0.2 doesn’t have motorised faders but does have scene memory and recall, the meters can be used to recall stored fader positions. With the meters in ‘locate’ mode, all of them illuminate their centre segment only and any fader movement will be tracked on the corresponding meter as it’s moved. When only one segment is lit, the fader is in its original or stored scene position. It sounds clunky when described, but it is actually very easy to use in practice. Although it requires manual intervention to recall fader
The StudioLive 16.0.2’s meters, like everything else on the console, are very bright and clear.

settings, it does get you to where you need to be. This obviously isn’t useful for automated plot changes during a theatre show, but for less demanding live concert work it’s just fine. I tended to use it as a safety net, letting me know without a doubt that ‘this was where we finished the sound check’ . The meter modes respond automatically to whichever control you’re currently adjusting. For example, in the ‘locate’ mode it will show the twelve input strip faders, but if you touch one of the aux or main faders they instantly flip to show those instead — using meters 7 to 12 —

and will flip back again the moment you move a channel fader: a pretty neat feature. One thing to note is this: if you’re monitoring input levels, for example, and you make an adjustment in the Fat Channel, the meters will quickly flip to display the things you’re tweaking, and then flip back about one second after you stop making changes. But what caught me out here a couple of times was the channel selection: the channel that’s actively selected will be the one that the Fat Channel controls apply to, not the channel whose fader you last moved.

or Mac, the communication is two-way, and all parameters can be controlled and monitored. In this way, the StudioLive 16.0.2 can be thought of as having ‘virtual’ automated faders, as scenes recalled using Virtual StudioLive with fader positions enabled will actually change the channel and aux or main fader settings within the desk, overriding the physical fader settings. There’s a lot that can be achieved with this system, and having now described all this functionality and additional capability it’s hard to remember just what a compact, easy-to-operate little mixer this is, especially at its price point.

Good impressions
The StudioLive 16.0.2 boots up quickly (around six seconds) and during the time I spent with it was always stable and well-behaved, with no crashes or unexpected behaviour. I also like the way Presonus have designed the main panel, aside from the lack of dedicated output meters, with everything clearly labelled, despite the number of multi-mode controls and meters. The rubber buttons are positive in

operation and are illuminated so well that I thought they might be a bit bright in a dark venue, but actually they’re just right for me. The rotary encoders, meanwhile, are smooth and have a consistent mechanical resistance through their travel. The faders aren’t the smoothest I’ve come across, though: I found them to be a bit sticky when pushed on either edge of the plastic cap, but they were fine when shunted along the centre line. I decided to let a couple of friends try the mixer: people who had recently expressed an interest in purchasing a new mixer for their respective organisations. I had to help out a bit at first, mainly with the EQ indications, but they both ‘got it’ fairly quickly without reading the manual, and one would have given me cash for the review model there and then! They were both very taken with the way that settings could be stored and recalled, and they were impressed with the size, look and feel of the StudioLive 16.0.2 too. They were also both convinced that it would be easy to use for others in their organisations, who would only need to operate the mix faders and a couple of aux sends at most. They saw it as a good

step up from analogue, maintaining simplicity but adding all the digital bells and whistles they desired. I, too, see the Presonus StudioLive 16.0.2 as an excellent first digital mixer, as it retains an analogue feel with its dedicated faders, simple routing and ease of use, but is packed with all the essential ‘outboard’ processing needed to do a great live-sound job. It also has significant extra capability on tap for direct recording, remote software control and MIDI interfacing in the studio. In summary, it’s a small and easy-to-operate mixer with plenty of features. To appreciate all that the StudioLive 16.0.2 can do, you really need to get your hands on one, but if you can’t do so right away, a good first step would be to find the full user manual online, and have a good read. I am also very tempted to buy one myself, as I can think of so many jobs for which this desk would be the perfect tool. $ $1299. T Presonus +1 225 216 7887 W

w w w . s o u n d o n s o u n d . c o m / January 2012



Boasting two custom drivers, are Phonak’s new universal IEMs the best monitoring solution for mixing on the move?

Audéo PFE232

Dual-driver In-ear Monitor

s the music production software we know and love finds its way onto pocket-sized mobile computers, we find it’s still impossible to take our studio monitors on the train with us too. As well as helping artists hear themselves on stage, in-ear monitors (IEMs) are a practical kind of mobile monitoring, and the latest Phonak model is specially designed to provide high-resolution sound suitable for mixing. Phonak’s first IEM was a single-driver model which, like the PFE232s, was sold with various filters that gave the monitors different frequency-response characteristics.


However, the PFE232s use a pair of tiny balanced-armature drivers, as opposed to the earlier single dynamic driver. Interestingly, balanced-armature technology is common in hearing aids, and Phonak themselves have quite an extended history in that particular field of research. Balanced armatures tend to be used in groups along with passive crossovers, an arrangement that provides the resolution and frequency extension required for monitoring purposes. These IEMs use a closed-back design, so isolation from external sound is very good, and there’s no leakage to speak of.

a bit too bright for my taste, and the green filters sounded a little dead in comparison.

Clean & Clear
The most prominent sonic characteristic of the Phonak PFE232s is high-frequency detail. Bass is presented in sufficiently extended form, though it’s very ‘lean’: don’t expect the subwoofer-style bass impact and extension of a high-end dynamic IEM and you won’t be disappointed. The voicing of the PFE232s reminded me somewhat of the smaller ADAM and Genelec monitors, so users of these systems would be quite satisfied with the balance and detail. I find differences in the tone of different IEMs to be more drastic at first than with studio monitors, but that the adjustment period required is actually shorter. After I got used to making judgements on the PFE232s, I found them quite up to the task, and some demos I mixed on the move in the Multitrack DAW and FL Studio Mobile apps translated nicely in the studio. As with most mixes carried out on headphones and IEMs, it was the bass frequencies that benefited the most from a remix on studio monitors. I found the PFE232s’ frequency response to have a slight ‘smile’ curve, though not so much so that it was a problem. In terms of IEMs, where design is something of a balance of compromises, the PFE232s give a very detailed performance. If you have the budget available and are looking for a clean and detailed presentation on the move, you won’t be disappointed. $ $599 W W

The Phonak Package
Two detachable cables with iPhone controls are included, and a zip pouch is provided for protecting the monitors. Since these monitors are worn with cable running over the ear, a pair of cable guides are in the box, along with plenty of silicon tips and foam tips in three sizes. The PFE232s are incredibly light and small for a multi-driver earphone, so they’re very comfortable to wear too. The most interesting pieces of additional hardware that are boxed with with the PFE232s are a small blue tool and box of coloured circles. These circles are the filters, which can be popped into the outlet tube of the monitors in order to tailor the sound. The filtering itself takes the form of a broad high-frequency cut: the grey filters in place by default cut very little high-frequency content, while the black filters cut a little more, resulting in the flattest response. The green filters cut the most, resulting in a perceived increase in bass response. I preferred the black filters, which sounded the most balanced, as the grey ones were

Phonak PFE232 $599
• Clean and detailed sound. • Extremely light and comfortable to wear. • Detachable and replaceable cables.

• Slightly recessed mid-range.

The Phonak Audéo PFE232s sit comfortably with other universal IEMs in the price range. They improve significantly on the single-driver Audéo models in terms of extension and clarity, but maintain Phonak’s signature sound, with lean yet extended bass and detailed, crisp treble.

Sony’s MDR 7550 dynamic IEM offers similar quality with more of a focus on mid-range detail, as do Shure’s SE425 and SE535 multi-driver balanced-armature models. JVC’s HA FX700 and Monster’s Turbine Pro Copper dynamic IEMs have more bass extension and impact, with less treble detail.


January 2012 / w w w . s o u n d o n s o u n d . c o m


from Lynx.

Reengineering the Two Channel Converter.
It’s hard to know where to start with Hilo, so let’s begin with the most important – audio quality. Hilo is the best sounding converter ever made by Lynx. The pristine, open, transparent audio quality of Aurora has been kicked up a notch or two, for mastering quality AD and DA conversion using Lynx’s BiLynear conversion technology.

This high resolution screen allows Lynx to provide the analog style meter seen here, as well as several bar style meters. Hilo is open-ended as no other converter has ever been. The updatable FPGA-based design and versatility of the LCD screen allow Hilo to accommodate enhancements to features, screens, functions, and utilities. The LSlot expansion port provides an upgrade path to future interface protocols as they become available. So the Hilo you buy today will be continually improved and morph into a device that will also meet your needs in the future.
Hilo can be powered by AC or optional DC battery pack

Hilo completely redefines the two channel converter genre. In addition to the primary analog Line Out, Hilo provides Monitor and Headphone outputs. Digital outputs include USB, AES/EBU, S/PDIF coax, S/PDIF optical, and ADAT. To top it off, each output has its own unique mix of all input sources courtesy of the 32-channel internal mixer. Hilo’s headphone technology adds a world-class headphone amplifier to the mix. It is capable of driving today’s low-impedance headphones with extremely low distortion while maintaining accurate inter-channel gain matching. With all of the options available, it was obvious that the standard push-button/LED meter front panel would not be sufficient to control and monitor all that Hilo can do. As we called each other on our touch screen cell phones, the answer became obvious. Hilo’s innovative LCD touch screen monitors and controls all routing, metering, and settings.
©2011 Lynx Studio Technology. Hilo is a trademark of Lynx Studio Technology.

Hilo – reengineered to offer you audio quality, control and versatility never before available in a two channel converter.

w w w. l y n x s t u d i o . c o m

Zoom H2N

Marantz PMD661

Hand-held Recorders

If you’re in the market for a portable recorder, there are plenty to choose from, with a variety of different features and at a wide range of prices. Before you buy, consider what you want to do. Some models allow you to use your own microphones, while some restrict you to their built-in ones. A few are even capable of surround recording, and can record multiple discrete channels, like a compact multitracker. These are just a few of the models that are currently available, with links to the SOS review where available.

Korg MR2

Zoom H4N

Roland R26 Tascam DR2D

Zoom H2N $349
The H2N has no fewer than five mic capsules built in, allowing it to record in X-Y stereo, Mid/Side, Blumlein and surround modes. It offers USB file transfer and headphone monitoring, and has an input-gain knob on the side to let you adjust the record level without having to navigate the menu. Samson Technologies +1 6317 842 200

Olympus LS3 $299
The latest in Olympus’ LS range, the LS3 offers 24-bit/96kHz PCM recording and has three mics built in, adding an omnidirectional capsule to the usual X-Y pair. It also has software features including a pre-record buffer and automatic file splitting during pauses. Olympus USA +1 800 201 7766

Korg Sound On Sound $299
The Sound On Sound recorder has just one omni mic built in, but it can accept stereo micor line-level source and also has a quarter-inch instrument jack input. It allows you to layer an unlimited number of recordings on top of each other, hence the name! Review: www.soundonsound. com/sos/sep10/articles/ korg-soundonsound.htm Korg +1 (631) 390 8737

Roland R26 $599
A high-quality recorder that, unlike most similar products, has the ability to record six discrete channels of audio. It has a built-in X-Y pair of mics, plus two preamps with phantom power, and a mini-jack input for compact stereo mics. Roland US +1 323 890 3740


January 2012 / w w w . s o u n d o n s o u n d . c o m

Olympus LS3 Korg Sound On Sound

Sony PCM M10B $299
One of this model’s selling points is that it has 4GB of recording memory built in — and that can be supplemented with a MicroSD card of up to 16GB in capacity, for a total of 20GB! It has a built-in stereo mic array (as well as inputs for line or mic sources) and a large LCD screen, and comes with a remote control. Review: www.soundonsound. com/sos/mar10/articles/ sonypcmm10b.htm Sony Professional +44 (0)208 412 9705

Sony PCM M10B

Korg MR2 $699
Korg’s MR2 is unusual for a handheld recorder, in that it offers 1-bit DSD recording: a high-resolution format that can be converted to any of the ‘normal’ sample rates and bit depths, making it useful for archive recording. It has an X-Y pair of mics built in and records to SD and SDHD cards. Korg +1 (631) 390 8737 Alesis Palmtrack

Zoom H4N $609
Winner of last year’s Best Hardware Recorder SOS Award, the H4N has two mics built in, plus two preamps, which can be used in addition to the onboard mics for four-channel recording. You can even use it with a computer as an audio interface. Review: www.soundonsound. com/sos/jun09/articles/ zoomh4n.htm Samson Technologies +1 6317 842 200

Tascam DR2D $269
The DR2D has two omnidirectional mics built in, and can accommodate external mics and line-level signals via mini-jack sockets. It allows you to perform overdubs and has a ‘dual-record’ function, which makes a duplicate of whatever you’re recording, but at a lower gain setting, in case the original file clips. Review: www.soundonsound. com/sos/dec10/articles/ tascam-dr2d.htm Tascam US +1 323 726 0303

Alesis Palmtrack $149
The Alesis Palmtrack is both compact and affordable, yet it has a stereo pair of mics built in, as well as being able to capture external mic or line sources (via 3.5mm jack). The internal and external mic signals can be mixed prior to recording, and it offers uncompressed 24-bit quality, up to 48kHz. Review: www.soundonsound. com/sos/jul10/articles/ alesispalmtrack.htm Alesis +1 401 658 5760

Marantz PMD661 $599
Marantz’s PMD661 is chunkier than many of its rivals, but it has two mic preamps built in, so you can use your own mics as an alternative to the built-in stereo pair. It records to SDHC cards (up to 32GB in capacity), at up to 24-bit/96kHz. D&M Professional +1 630 741 0330

w w w . s o u n d o n s o u n d . c o m / January 2012



Optimising vocal recordings and tackling some troublesome mixes are on the agenda this month, as the SOS team travel to Paul Shepherd’s studio in Nottingham.


aul Shepherd is one of those musicians who has enjoyed a busy and interesting part-time musical career, but has only just returned to recording after a long break. His previous experience was back in the days of C-Lab’s

Creator (the predecessor to Apple’s Logic DAW software), but today he has Pro Tools 8 LE running on a Mac Pro, which is teamed with a Mackie Control surface, a Digi 003 interface and a pair of Mission 762 passive hi-fi speakers powered from a Denon amplifier. His studio is set up in a small basement beneath his Nottingham home, and measures approximately two

metres by a little over three metres. The ceiling height is low enough to require taller people to duck. He records mainly original material along with his daughter, Madeleine, who, despite being only 13, has an enviably strong and well-pitched voice. She also has some specific ideas on how contemporary music should sound. Paul contacted us because his mixes didn’t pack the punch he felt they should, and they also sounded different when played outside the studio — a common problem and one that can have multiple causes. Over coffee and some very welcome bacon sandwiches, we played some commercial material from Hugh’s test CD.

Paul Shepherd’s studio, with Mission hi-fi speakers as the main monitors. Some thin foam had been put up, and this took care of the early reflections, but the room’s dimensions (including the low ceiling) gave rise to some low-frequency issues.


January 2012 / w w w . s o u n d o n s o u n d . c o m

In the corner where most of the vocal recording takes place, Paul White glued some foam to the wall and ceiling, adding to what was already stuck to the door, to kill off as much room sound as possible. Paul Shepherd’s vocal mic was initially set too far back into the Studiospares mic filter, so Hugh moved it level with the outer edges, to help ensure a less coloured sound.

Given the size and layout of the room, the results were fairly predictable. Paul had set up his studio facing across the width of the room rather than along its length (lengthways is generally safer in smaller rooms), but given that he’d also managed to squeeze a Roland V-Drum kit and a selection of keyboards into the space, it would really not have been practical to set it up in any other way.

possible to arrive at mixes that translate well to other systems. Paul had already put up a small amount of fairly thin ‘camera case’ foam at various points around the room, which, along with

all the equipment and shelving, neatly eliminated any flutter-echo issues. The Mission monitors were mounted quite high on the wall, but the unusual design placed the tweeter in the middle of the

Not-so-sweet Spot
The main problem was that Paul’s mixing position was almost exactly halfway between the wall behind him and the wall behind the monitors, and in small rooms with solid walls, experience has shown that this produces a dead zone where much of the bass end is lost. We demonstrated this to Paul by asking him to listen to a track from his normal position and then to move closer to his desk to take him out of the central area. He immediately heard that the bass end was better balanced when he moved forward from the dead zone. This physical acoustic problem can really only be solved with very extensive bass trapping, which would reduce the usable space in the room dramatically and unworkably. The best we could realistically achieve was to make sure Paul understood the issue and leaned forward when trying to balance the bass end of his mixes. In this kind of circumstance, it becomes really important to check mix balances in other rooms and on other systems. Although this slows the process down, it does make it

w w w . s o u n d o n s o u n d . c o m / January 2012



front baffle (with the bass driver at the top and reflex port at the bottom), and this arrangement conveniently put the tweeter at around head height. By chance, this produced surprisingly good results, given the room’s size, low ceiling and thick, solid walls, though both Hugh and I suggested that Paul check out the Focusrite VRM Box, which emulates various speaker types over headphones. While I wouldn’t advocate using a VRM Box for making all mix decisions, it works exceptionally well as a secondary reference for checking that your mixes are likely to translate well to other systems.

Vocal Point
For recording vocals, Paul uses the corner nearest to the door, with the door itself covered in more of the thin, camera-case-style foam. His Rode NT1A mic is teamed with a Studiospares reflection screen, plus a mesh pop filter, and he was getting reasonably dry results. We felt we could improve the situation, however, by adding some acoustic foam to the side wall, so that both sides of the corner behind the singer would be less reflective at vocal frequencies. The low ceiling was also a worry, so we decided to fix a two-foot-square tile there too, and as we had a few two-inch-thick Vicoustic pieces left over from a previous Studio SOS, we decided to use those. Alhough they lowered the ceiling headroom by a further two inches, the foam tiles reduced ceiling reflections, and provided a more comfortable experience for any taller people entering the room! Paul was happy for the tiles to be glued directly to the wall, so out came the spray glue and, 10 minutes later, the job was done. Hugh also observed that Paul had positioned the mic rather far back inside the curved mic screen, and in our experience the sound can become coloured unless the mic is brought forward to be roughly level with the front edges of the screen. Again, a couple of quick adjustments by Hugh remedied this, and the result was a noticeably more neutral-sounding vocal area. However, the screen and mic combination was mounted on a mic stand with a relatively small round base and, as a result, the whole thing was decidedly unstable, so Hugh recommended purchasing a stand with a tripod base, which would be a lot more steady.

Paul’s mixes to see what other issues he was experiencing. He’d identified the bass sound and vocal treatments as particular areas of concern, but had only a limited number of plug-ins at his disposal: the ones that come with Pro Tools LE, plus a JoeMeek compressor and Izotope’s Ozone Lite, which allows the user to access a range of presets, but not to edit them. Paul also confided that he’d tried to use level automation to keep the vocal level even, but hadn’t used any compressor plug-ins. The only vocal treatment was from the Air Reverb plug-in that comes with Pro Tools. Playing the track revealed a very capable arrangement with a good choice of sounds and a well-performed vocal, though the vocal sat rather too far back in the track and the relatively unsophisticated Air Reverb didn’t really do it any favours. There was also a lack of focus to the low end. Paul admitted to mixing with his monitoring cranked up to very high sound levels — possibly as a natural response to the inherent lack of low end due to the room’s acoustics — and this would certainly account for the relatively submerged vocals. Paul had used Pro Tools LE’s Strike drum instrument for all the drum parts, and a sampled bass guitar, again from one of the included plug-ins. He’d added a little electric guitar of his own, via a Line 6 Pod XT, and was using one of the Izotope Ozone presets to process the final mix. He felt that the overall sound was slightly abrasive and lacking in punch, but didn’t know how to remedy it.

Paul Shepherd, this month’s Studio SOS candidate.

Kick Out The Jams
As has been explained in many of our Mix Rescue articles, not all bass instruments need to produce large amounts of deep bass, as the human ear finds it quite difficult to separate sounds at the lower end of the frequency spectrum. In this mix, the kick drum was providing plenty of weight, but it was battling for space with the sampled bass guitar. Our solution was to roll off some low end from the bass guitar, and also to boost it at around 230Hz to emphasise that area of the bass sound that would be most audible on a limited-range system, such as a portable radio. We also rolled off some of the extremely low frequencies from the kick drum, as they were simply eating up headroom for no good reason. In most cases, there’s doesn’t need to be much going on below 40Hz, even on kick drums, so the trick is to use a low-cut filter, and then move the frequency up until you can just hear the sound being affected, then back it off again slightly.

Strike allows various aspects of the drum-kit sound to be balanced, so to clean up the drum mix we reduced the room mic’s contribution significantly and pulled the overhead mic level down slightly. Strike also has a fader that emulates the infamous SSL talkback mic compressor used on so many drum tracks in the ‘80s. This is a very ‘pumpy’ compressor that, when added to the dry drum sound, gives it a lot more attack and attitude. This is essentially parallel compression built into the Strike plug-in, and it can be used as subtly or unsubtly as you like. We found that a fairly subtle application gave the drums the right amount of edge. For the bass part, we attenuated the lows below 120Hz, using a shelving filter. It’s also a good idea to high-pass-filter vocal and electric guitar tracks, as these can include low-frequency elements that are not part of the wanted sound and just muddy things up. For example, a vocal mic can pick up low-frequency air blasts, even with a pop screen in place, while the electric guitar generates low frequencies when the strings are touched while palm-damping, and also sometimes during the picking phase of the lower strings. The lower-mid range of a pop mix can get very congested, so any unnecessary lows that can be cut will help keep this vulnerable area clear. After tidying up the low and mid-range frequencies in this way, we turned our attention to the vocal sound.

Compress Or Automate?
Aside from being mixed a little too low, the vocal part wasn’t as even-sounding as it could be. Given plenty of time, I’d generally use automation to level out as many level fluctuations as possible, then feed the vocal

Fixing The Mix
Having addressed the more mechanical problems, it was time to look at one of


January 2012 / w w w . s o u n d o n s o u n d . c o m


to a bus with a compressor inserted, so that the compressor comes after the automation. In my own studio, I often use Wave’s Vocal Rider instead of automation, as this does a great job and attends to the fine detail of level changes within individual words very efficiently, although I find that it sometimes helps to precede this with a low-cut filter, to prevent ‘near-popping experiences’ from triggering gain reduction. In this case, though, we bypassed the automation that Paul had already created, so that we could demonstrate what was possible using a compressor. As is the case for many readers, he was a little unsure as to how compression should be applied, so on the occasions he had used, it he’d dropped in a preset without realising that changes to the threshold or input level still need to be made to achieve the desired amount of gain reduction. For our vocal treatment, we used the JoeMeek plug-in to give us four or five decibels of ‘squash’, and then followed this with the included Maxim plug-in, which is essentially a type of limiter, to take care of any excessively loud peaks. That gave us the necessary ‘up-front’ and dense sound, so then it was down to looking at reverb and delay treatments. The original choice of Air Reverb produced a rather dated sound, with the reverb being a little too obvious, so we switched to D-Verb and used the Plate setting with around 30ms of pre-delay. The reverb time was reduced to around one second, and a second aux send was set up to feed an instance of the Stereo Delay plug-in. This was sync’ed to the song tempo, with one repeat at around 300ms and the other at around 600ms. Feedback was set to about

Having done what was possible with the recording and monitoring environment, Paul turned his attention to some troublesome mixes.

30 percent. By combining the reverb and delay at a level low enough that it was not too obvious, but was still ‘missed’ when bypassed, we made the vocal sound much more focused and contemporary. When solo’ing the vocal, the delays became very obvious, but in the context of the mix they were almost subliminal. This mix of delay and more reserved reverb treatments is very popular at the moment and helps keep the vocal at the front of the mix. In fact, Paul had already employed some more obvious delay treatments for selected phrases in his mixes, but rather than automate the send level to a delay plug-in, he had adopted the technique of copying the final syllables of some vocal lines to a new track, and sending that to his delay plug-in, to add delay precisely where he wanted it. This approach is just as valid as automating the sends, of course.

which we could adjust, so that Paul could follow the process and also save the result for later experimentation. For EQ, we chose Pro Tools LE’s included four-band equaliser, and created a very subtle ‘smile’ curve using 2dB of boost at 88Hz, 1.5dB of cut at 240Hz and a generous 4dB of shelving boost at 8.5kHz. This was followed by low-ratio, soft-knee compression of 1.4:1, with the threshold adjusted to give around 4dB of gain reduction. Then we used the Maxim limiter to shave another 2dB off the peaks. This combination produced a punchy sound that didn’t suffer from the harshness of Paul’s earlier mix.

Song Two
The second song required a different vocal treatment that would add life without sounding obviously ‘reverby’. For this, we chose D-Verb’s Medium Ambience program, opened up the high-cut filter to keep it bright, and then added around 30ms of pre-delay. In conjunction with similar compression to that used in the earlier song, this added a subtle density to the sound and eliminated the dryness. We also took the opportunity to show Paul how a higher reverb level and longer pre-delay time could create a vintage, slap-back doubling effect. For the drums on this song, Paul wanted a gated-ambience drum sound but didn’t know how to go about achieving it. We used PT LE’s Non-Lin reverb for this, and though it doesn’t deliver the best gated drum sound in the world, it does a perfectly good job. Essentially, it produces a dense burst of early reflections that remain at

EQ Tips
After we’d balanced the vocal a little higher in the mix, our final tweak was to use some high-shelving EQ to add a couple of dB of boost above 6kHz, to lend a sense of ‘air’ to the sound. We demonstrated how, by applying maximum boost while adjusting, it was possible to move the frequency control and hear at what point the EQ changed from adding high airiness to upper-mid harshness. Once you find the best corner frequency, you can adjust the boost level. For mix processing, we found some smoother-sounding Ozone presets than the one Paul had chosen, but we felt it would be more useful to use separate plug-ins,


January 2012 / w w w . s o u n d o n s o u n d . c o m

a fairly constant level, before cutting off abruptly. Getting the balance right is the key to the success of this sound, and adding in some of that talkback-mic compressor really helped achieve the desired aggression.

In Pod We Trust
That left only the electric guitar sound. Paul’s daughter, Madeleine, likes bands that use a fairly dirty rhythm guitar sound, so I suggested we listen to some of her commercial choices, as well as some Green Day songs, to see how the guitar should sound. I usually opt for miking a guitar amp, but in Paul’s small room a DI approach seemed more practical. I set up a couple of Pod sounds that we could combine to get the desired result. Paul wanted what he described as a ‘chuggy’ sound, but I know from experience that overdriven parts can get very messy unless you take care with both the sounds and the arrangement. I set up a Soldano amp model to give a dense overdrive with plenty of low end and a bit of mid-range scoop, for almost a classic metal tone, but with a little less presence. I then found a Vox AC30TB model that had enough drive to produce ringing chords with a nice dirty edge, but without losing definition. The idea was then to double the part, using the AC30 sounds for the full chords and the more dense Soldano sound for playing just the bottom couple of strings (in this case the root and fifth) of the chord. I tried this using Paul’s rather nice Rickenbacker and it sounded pretty good. However, you do have to play

‘chuggy’ parts appropriately, with plenty of attack, followed by releasing the lefthand finger pressure slightly between strokes to afford a little damping. Without the damping, you tend to get a more continuous sound that can be messy, whereas appropriate damping helps reinforce the chugging rhythms. We discussed a few other guitar strategies, such as doubling the part while playing only the higher strings of the guitar on the first pass and the lower strings on the second, and also layering two parts playing different inversions of the same chord. If in doubt, recording an unprocessed dry sound alongside the processed sound is always safest, as it allows you more flexibility when mixing. Often layering a dirty and a cleaner version of the same part will save the day if things are getting too messy, as mixing in the cleaner sound has the effect of adding some definition back into the dirty version. These and other guitar-recording tricks will be explored in greater depth in a future article dedicated to the subject. Our final piece of advice to Paul was that he consider upgrading to Pro Tools 9 or 10, as these versions offer several improvements, including plug-in delay compensation. This could be an important consideration, as Paul was feeling tempted by a UAD2 DSP card, and the effects that run on these incur greater latency than non-DSP plug-ins. We left Paul with plenty to think about, but he seemed to have found the exercise useful, and we both look forward to hearing his future mixes.

‘’Immediately fell in love with the Vintage series’’

‘’V67 midrange repeatedly blew me away’’

“Very warm sounding microphone, a great addition to collection”

Rafa Sardina

Paul White programming some guitar patches on a Line 6 Pod processor.

Lady Gaga Beyoncé Shakira


We aim for more punch, depth and clarity in a quirky urban track.

he mix files for the song ‘Daydreamin’ came to me as a massive project, with no fewer than 39 separate vocal parts (some with both ‘A’ and ‘B’ versions), all of which were to be used in my final mix. Along with the instruments and layered percussion, the original project The plug-ins Paul used to process the came to around 90 main vocal. Rather than undertaking tracks, but I managed detailed level automation, he used to whittle that down to Waves Vocal Rider to smooth out the a more manageable 45 levels, before applying a limiter to or so before I started catch the most wayward peaks. work on the mix in earnest — and even then the Logic mixer took up two whole There’s also an acoustic guitar rhythm 20-inch monitor screen-widths! part, an acoustic guitar lead part, organ I like this song — it’s slightly quirky, and piano segments, a string-like synth it feels contemporary, and it has a very pad and some nice synth arpeggios. Add well-conceived structure. Placing it to that a multi-layered percussion part, in a single category is quite difficult, a section of steel drums, a kick drum and though, as it features urban-style a fat but somewhat ‘lacking in top’ bass vocals underpinned with ad-libs and part, and you start to get the picture. multi-layered harmonies. All this sits With the exception of some over a strong backbeat, with the most slight tuning issues on the acoustic prominent instrument being a ukelele! rhythm-guitar part, it was all well played,


well recorded and well sung. Equally importantly, the arrangement worked pretty much as it was.

Mix Prep
I didn’t want to try to mix all 90-something tracks in one go: ‘multing’ is all very well where it’s required to do a job, but mixing is not about keeping all

Rescued This Month: Ivy League Sound ound & Nesia Beatz
Ivy League Sound are a production team based in Pittsburgh and comprising Mike Burger and Joseph Faulk. Mike studied piano performance at the Eastman School Of Music in New York. Joseph began playing guitar at a young age and was part of various bands writing and recording music throughout high school and college. The two began writing for artists in the Pittsburgh area in 2010, which led them to the artist Nesia Beatz. At that time, Nesia had been successfully producing songs in Atlanta and Pittsburgh for artists such as Wiz Khalifa, Yung Joc, Bonecrusher and Waka Flocka, but was interested in extending his talents as a vocal artist. Nesia came to Ivy League Sound in search of a more organic song with live instrumentation and an island/hip-hop vibe. Mike and Joseph created the entire instrumental track for ‘Daydreamin’ and handed the song over to Nesia to write the lyrics and melody. They recorded the vocals, ukulele and acoustic guitar using a Peluso 2247 SE mic, Vintech x73i preamp and EQ, Empirical Labs Distressor dynamics processor, and Lavry AD10 A-D converter. The rest of the song was constructed using a combination of Logic’s soft synths and samplers, Lennar Digital Sylenth and Spectrasonics Trillian. They also used Yamaha HS80M studio monitors.
Nesia Beatz.


January 2012 / w w w . s o u n d o n s o u n d . c o m

options permanently open. It never hurts to commit to some decisions, so after determining that the vocal pitching was perfectly adequate without resorting to artificial pitch correction (or perhaps it had already been subtly tweaked?), I bounced down all the double-tracked and triple-tracked parts to fresh stereo tracks after panning the individual lines to give a bit of stereo spread. There were also vocal segments recorded separately, but that could be moved onto the same track, which also helped to simplify the mix. When it came to instruments, the ukelele was split into verse tracks and chorus tracks, and I simply combined these. All the audio tracks placed the segments at their correct time positions, other than the ukelele chorus part, which I had to copy and move so that it could be used in both choruses. To aid navigation, I coloured the various parts according to whether they were vocals, rhythm section or whatever and routed the various vocal parts to their own stereo subgroup. A second subgroup was set up for the main drum and percussion parts. I’d also been supplied with dry and processed versions of the instrument tracks, so the processed versions were all muted and then hidden to free up more arrange-page real estate. To my ears, the original mix had the vocals pushed a little too far back, while the somewhat boomy kick-drum tended to dominate proceedings. There was no real definition to the bass part, and I felt that some of the effects were clouding the mix and robbing it of clarity, but even so,

the overall mix wasn’t a bad start. However, before I started to rebuild the mix, a little housekeeping was in order. Having bounced the vocal tracks that I could, I set about dividing the vocals into separate phrases, so that any spaces between phrases could be silenced. There were quite a few lip smacks and anticipatory breathing noises between phrases, so these were all removed in the process. While you can use a ‘Strip Silence’ function to do this (most DAWs now include one), I feel much more in control of proceedings when doing this sort of thing manually.

Rather than relying solely on EQ, the ukelele was brightened using an instance of Noveltech’s Character enhancer plug-in.

In my experience, vocal tracks often include a fair bit of low end, due to breath getting through the pop filter, so although

there may be no audible popping, putting a spectrum analyser on the track will often show up a significant amount of 20-40 Hz frequencies flapping about at the low end. To combat this, I routinely put a 12dB/octave, 80Hz filter on each track and then fine-tune the cutoff frequency by ear, sometimes taking it up to as high as 200Hz if it doesn’t affect the vocal tone. This goes before any compression, as the unwanted low end can otherwise trigger unnecessary gain reduction. I used Logic’s own EQ for this purpose, and also took the opportunity to add a little shelving air EQ above 8kHz to open out the vocals slightly. In addition to low-cut EQ and ‘air’ lift,

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compressors were put on all the vocal tracks to even things up a touch. Logic’s own compressors were used for most of the supporting tracks, and a UAD 1176 was used in Nuke mode (all buttons in) to firm up the urban vocal part. Most of the other compressors were set to work

this one.) More processing and tweaking would come later, but this gave me a good starting point. In fact, the only vocal part that was given more blatant processing turned out to be the one that carried the ‘Dreaming’ line on the choruses, which was treated to a generous helping of

To bring the kick more into focus, Paul used an instance of SPL’s Transient Designer to emphasise the attack, while applying a little low-end EQ boost and a touch of limiting courtesy of Waves L1 Ultramaximizer.

quite gently, levelling the peaks by no more than 4dB or so. The UAD 1176 was preceded by a Waves Vocal Rider to help keep the level of the urban vocal part even and followed by a limiter just to catch any errant peaks, though with the final mix settings the limiter proved unnecessary. Similarly, a safety limiter placed over the drum subgroup turned out not to be needed, as the peaks never triggered it. At this early stage in building the mix, I set up an initial vocal reverb using the Universal Audio EMT 250 plug-in (for their UAD DSP platform) to create a short, fairly bright reverb with 60ms of pre-delay, adding a sense of spatial presence to the vocals without actually making them sound processed. (The same principles would apply if using another reverb plug-in, but I’m particularly fond of the sound of

625ms delay with 30 percent feedback, via a Logic Tape Delay plug-in that I’d set up on aux send 2.

Drums & Ukelele

After the vocals, the percussion and ukulele formed what I felt to be one of the most important parts of the track. The ukulele was brightened up slightly using the Noveltech Character plug-in, running on my TC PowerCore system, and again I rolled out any unnecessary low end, this time below 270Hz, but otherwise left it dry so that it would cut through cleanly. (There are plenty of enhancer plug-ins that run natively and can add some brightness, but none that seem to do quite the A touch of distortion and EQ was all that was needed to enhance the same thing as Character. bass and get it to cut through the mix.

The nearest equivalents are probably Flux’s Bittersweet and the BBE Sonic Maximizer in the Sonic Sweet bundle.) Most of the rhythm-section parts worked well with little or no added treatment, although the kick needed a bit of work to bring it into focus without losing weight. This involved using the SPL Transient designer to shorten its release time, after applying EQ to tame the sub50Hz frequencies and to plump up the 76Hz region. A little 200Hz cut prevented the bass boost from making the midrange flabby, while a touch of boost at 3.5kHz added some clarity. I also placed a limiter at the end of the chain, just to catch the occasional high peaks and maintain some headroom. Noveltech’s Character was again put to work, this time on a multi-layered drum fill close to the end of the track, to give it a little more definition. The snare was livened up with a coarse early-reflections treatment from Logic’s Platinumverb with 16ms of pre-delay, preceded by Logic’s Overdrive plug-in set to full brightness, to introduce some snap. A mild dose of distortion was also added to the handclap sound, and this was followed by a couple of short delay repeats from Logic’s basic stereo delay plug-in at 70ms and 90ms, with very little


January 2012 / w w w . s o u n d o n s o u n d . c o m

Listen Up!
This article won’t make much sense if you don’t listen to the original and remixed versions of the track! We’ve placed both of those, along with some other before-and-after examples, on the SOS web site. W

feedback. By now, the rhythm section was starting to come together, but I still craved more punch, so I set up a parallel compressor fed from aux 3 and inserted the UAD plug-in version of the Empirical Labs EL7 Fatso Jnr, set to compress really hard using its ‘Spank’ setting, and driving it to achieve up to 25dB of gain reduction. When mixed back in with the main drum mix, this made the kick sound noticeably more solid and punchy.

Here you can see the full mixer setup for the final remix. Note that even on a heavily processed-sounding mix like this, Paul usually only used a couple of plug-ins on each source.

I was a little worried that the bass sound was going to get lost, as it had very little definition, so I added a modest amount of overdrive (this was getting to be a habit!) to stimulate a little harmonic growth, and then used EQ to try to push the sound a little in the 170Hz region while trimming away excessive low end using a 24dB/octave low-cut filter set just below 80Hz. A 12dB/octave high cut applied above 5kHz took care of any HF mess. This improved the sound a little, and now it actually sat quite well in the rhythm section without muddying the kick sound. Feeding some of the bass part into the parallel compressor along with the drums also added some welcome weight.

Now it was time to bring in the other instruments, which I treated far less than on their original mix. The piano was given just the barest hint of reverb and some presence boost, while the UAD’s Roland Dimension D plug-in lent it just a hint of movement. The Dimension D provides a strange and rather unique chorus effect, that’s much more subtle-sounding than

most. (If you don’t have a UAD card, you could also try equivalent plug-ins by ERS and WOK.) The string-like pad was left completely unprocessed, while only a tempo-locked 312ms tape delay was added to the arpeggio synth. A further chopped synth part was left as it was, other than rolling off the lows, while the organ was treated to EQ to add 3.5kHz boost and to roll out those unnecessary lows. As the acoustic rhythm guitar was not entirely in tune, and because it largely underpinned what the ukelele was already playing, I trimmed off some low end and then used it quite low in the mix, where it added a bit of body to the sound but wasn’t loud enough to make the tuning


Berklee is accepting now accepting Berklee is now applications for the position of Full-Time Faculty, Music Production and Engineering applications for the position of Full-Time Faculty, This is a 9-month full-time facultyMusic position with a renewable contract and an initial appointment to begin September 1, 2012. Academic rank and Production and Engineering.   compensation commensurate with experience. This faculty is a 9-month full-time faculty position with a renewable contract and an initial This member will be responsible for teaching courses with a particular emphasis on modern production appointment to begin September 1, 2012. Academic rank and compensation paradigms including DAW and related software, MIDI sequencing, beat making, and re-mixing, in addition to commensurate with experience. multi-track recording and mixing techniques.
This faculty member will be responsible for teaching courses with a particular emphasis Qualifications: on modern production paradigms including DAW and related software, MIDI sequencing, beat and re-mixing, in addition to multi-track recording and techniques. • 10making, years minimum professional experience, including national ormixing international recognition as an active producer, recording engineer, and/or mixer in a variety of contemporary music styles Qualifications: • Expertise with a wide range of modern music production tools
 • Degree in music or equivalent professional experience • 10 years minimum professional experience, including national or international • recognition Excellent interpersonal and communication skills as an active producer, recording engineer, and/or mixer in a variety of • contemporary Ability and willingness to work within a diverse faculty and student body music styles • Classroom teaching preferred. • Expertise with a wideexperience range of modern music production tools

Balancing Act
Playing the vocals, the ukelele and the rhythm section showed that the track was starting to take shape, although much balancing and some automation would ultimately be necessary to keep the vocal balance correct. The band had very strong ideas as to how the vocal balance should sound, so after improving the clarity of the vocals and bringing them up in level, I sent the band several mixes so that we could fine-tune the vocal balance to their liking.

• Degree in music or equivalent professional experience Interested candidates will and find communication application guidelines • Excellent interpersonal skills online at: • Ability and willingness to work within a diverse faculty and student body All application materials must be received no later than February 17, 2012. Incomplete applications will not be • Classroom teaching experience preferred. considered. Interested candidates will find application guidelines online at:

Berklee College of Music is committed to increasing the diversity of the college community and the curriculum. Candidates who can contribute to that goal are encouraged to apply and to identify their strengths in this area.

All application materials must be received no later than February 17, 2012. Incomplete applications will not be considered. Berklee College of Music is committed to increasing the diversity of the college community and the curriculum. Candidates who can contribute to that goal are encouraged to apply and to identify their strengths in this area.

w w w . s o u n d o n s o u n d . c o m / January 2012



problem obvious. In a perfect world, I’d have asked for this to be played again, but in this instance it wasn’t a major part of the arrangement, so it could be mixed quite far back. The acoustic guitar solo, which is in tune, plays towards the end of the song, accompanied by more vocals, so its level was automated to keep it audible while avoiding conflict with the vocal parts. Some ‘broad strokes’ vocal level automation was required to maintain the balance between the various sections, but nothing really forensic was needed. The balace was fine-tuned after I had sent rough mixes to the band to check, which is just as well, as they managed to identify a couple of short vocal snippets they wanted to retain that I’d faded down in my initial mixes. Panning was used to spread out the vocal layers and, to a lesser extent, the instruments, trying to maintain a nominally even left/right balance, as always, but as many of the multi-tracked parts had already been submixed in stereo, (and many of the effects were already stereo) it wasn’t necessary to use extensive panning to paint a wide picture.

Bus Processing
Overall mix treatment was provided by an SSL bus compressor emulation, and a very mild application of passive EQ from PSP’s NobleQ, just to lift the 80Hz region a hint and the highs by a couple of decibels. Then I put the mix through the a tape emulator (UAD Ampex), prior to limiting, as this seemed to knit the sound together nicely, and it also made the highs sound less aggressive without detracting from the overall clarity. Limiting was restricted to around 3dB of gain reduction on the loudest peaks, while the bus compressor added no more than a couple of decibels or so of overall ‘squash’, so the final mix processing was actually quite gentle.

seem, I thought it was exactly the right choice of instrument for this track! What can you learn from this mix? Well, if I had to summarise the main elements of my approach to this type of mix, it would be to first simplify the track layout as much as possible and to use colours to identify the various types of track, just to aid navigation. It makes it so much quicker and easier to get on with the mix. For similar reasons, groups were used, where possible, to provide overall control over the main parts of the mix, such as the vocals and the rhythm section. Then I’d go on to trim away unwanted lows from any tracks that weren’t a legitimate part of the rhythm section’s bass end, to edit out any sections that might include low-level noise, such as breathing or ambient background sounds, and to use any effects sparingly, especially reverb. Parallel compression is a valuable tool for adding weight without smearing the transient detail of a mix and is invaluable in urban, dance and rock production. It certainly suited this mix. Reverbs often work best when understated — I used just enough here to stop the performances feeling unnaturally dry: a fairly long pre-delay helped to create a confident vocal sound without smearing it all

over the place. Where something more noticeable is required, as with some of the chorus vocals in this track, a basic repeating delay often does the trick. It’s also interesting to note that overdrive distortion isn’t just for ‘dirtying’ things: it can occasionally be useful to make things actually sound cleaner, which is the opposite of its usual effect on electric guitar. Sources like snare drums and handclaps can often be sharpened up nicely using a simple overdrive plug-in and, as in this case, a harmonically-challenged bass can be given a bit more mid-range grunt by adding a touch of dirt. Then there are the old fallbacks of using coarse, early-reflection reverbs to liven up percussive sounds without filling up all the valuable spaces and of adding closely spaced, discrete repeats to claps to make them sound bigger. Ultimately, then, there were no unusual techniques applied to this mix. It was really just a matter of deciding what belonged at the front, what should stay at the back and how the lows should sound. As I said at the outset, having a well-thought-out musical arrangement really does help make life easier for the guy doing the mix!

Remix Reactions
Joseph Faulk: “Mike and I have been reading SOS ever since we decided to start producing and recording artists, and it has been extremely helpful for us in so many ways. We are always devouring the Inside Track, gear review and Mix Rescue articles. After hearing a lot of the mix rescue ‘before and after’ MP3s, we were confident that our song would be in great hands. We had a few mix engineers eager to get their hands on this song, but with all that Sound On Sound has done for us, it seemed fitting to send it there.” “Paul’s mix added a lot of clarity and presence to the numerous elements of this song, helping it realise its true potential. He was extremely easy to work with and really understood our vision for this song.” Mike Burger: “We were most concerned with the sound of the drums and the ukulele, since those elements formed the backbone of the track. Paul’s adjustments, especially to the kick drum, took those elements above and beyond. We also had a lot of ideas both instrumentally and vocally as to how we were going to make each section of the song new and interesting. While good from an arrangement point of view, all of the ideas created a bit of a cloudy rough mix. Paul was able to clean up the tracks in such a way that all the parts can be heard clearly in their own space. Paul’s mix is everything we hoped it would be!” Nesia Beatz: “The mix is very organic and rich. There was so much to play around with but Paul put everything in a certain place. Sounds like candy: I love it!” Ivy League Sound: W W W!/IvyLeagueSound Nesia Beatz: E W!/NesiaBeatz W!/nesiabeatz

Looking Back
Comparing my version of the mix with the band’s original rough mix, the vocals were standing out more clearly and the bass end sounded tighter, but I’d still managed to convey plenty of punch and depth. The separation between the various instruments in the mix had also been retained, which was as much down to the good musical arrangement as it was to my mixing. I was very pleased with the ukelele sound and, unlikely as it may

This month’s Mix-rescuees, Ivy League Sound.


January 2012 / w w w . s o u n d o n s o u n d . c o m

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Clean Up Your Act
Noise-reduction Tools & Techniques
If hisses, clicks, thumps or hums wreck a great take, don’t panic: do something about it!

odern recording systems are clean and quiet, so electronic noise in the recording chain is rarely a problem. However, what was once an acceptable amount of noise in a mix (because it could be masked by tape hiss, for example) will now stick out like a sore thumb. Many of us also face challenges from noises such as the hum and hiss of guitar amps, mysterious digital clicks, camera whine, air-conditioning noise and so on. Audio-restoration software is quite affordable now, but you still need to know how to get the best from it, so in this article I’ll discuss the pros, cons and applications of various noise-removal tools.


Gating & Expanding
The humble noise gate (or simply ‘gate’), relies on modest levels of noise being masked when the wanted audio is present, and being audible only during pauses. The gate opens as the signal rises above a threshold, and closes when it falls below it. If the threshold is set just above the noise floor, the gate will close during pauses and open as soon as the wanted audio is detected. To avoid clicks due to the gate opening very quickly, and to prevent slow fades being cut off abruptly, most gates have adjustable attack and release times. Some also offer a ‘hold’ control, to keep the gate open for a short time after the audio falls back below the threshold and prevent the gate from ‘chattering’ when the audio features rapid level fluctuations. Masking requires the wanted sound to be louder than the noise in the same

frequency range, especially in the 500Hz to 5kHz region to which human hearing is most sensitive. So if the wanted audio is a low-frequency sound and the noise is a high-pitched whine, the noise will remain audible. Although you might improve the situation using a low-pass filter, this usually also dulls high-end details you want to keep. This all means that gates are only really effective when (a) the level of unwanted noise is fairly low, and (b) the wanted audio masks the noise when the gate is open. The greater the level of background noise or spill, the less natural the result will sound as the gate opens and closes. You can generally get away with higher levels of noise when gating short, percussive sounds than when working on delicate sources, but ultimately you must let your ears decide. In most applications, the complete removal of noise is counter-productive, as it draws attention to the processing and becomes a distraction. In practice, it’s usually far more effective and less obvious to use more subtle processing, even if the degree of noise reduction is more modest. Reducing the attenuation range of a gate to, say, 12dB (so it doesn’t completely close, but just attenuates by 12dB) usually sounds subjectively better than a hard gate action. A refinement of the gate (pioneered in Drawmer’s DS201) employs variable high- and low-pass side-chain filters. This helps prevent accidental triggering from pitched components in the noise that are significantly different in pitch from the wanted signal. The DS201 was popular for drum recording, as the filters could be set to reduce the risk of hi-hat spill triggering gates used on other drums.

Expanders work in a similar way to gates: when the signal falls below the threshold, it’s subjected to progressive gain reduction, much like a compressor in reverse. Some have a variable ratio, so that the higher the ratio, the more the signal is attenuated. Ratio figures are the opposite way around to those for a compressor: a compressor ratio of 2:1 means that the signal has to rise 2dB above the threshold to cause a 1dB rise in output; whereas an expander ratio of 1:2 means the signal has to fall 1dB below the threshold to be attenuated by 2dB. General operation, once you’ve selected a suitable ratio, is much the same as for a noise gate — but so are the limitations. The expander can be a little less abrupt than a gate, but once you’ve set the attack and release there’s often little subjective difference unless you use a very low ratio setting. Moving up a stage in complexity, the dynamic filter’s side-chain controls the frequency of a sliding low-pass filter. It’s most effectively used to tackle high-pitched noises such as hiss, as it works by pulling down the filter frequency as the wanted sound decays in amplitude. Like a gate, it relies on the wanted signal masking the noise, but it’s better at disguising noise during decaying sounds because most acoustic instruments decay faster at high frequencies than low ones. The result can sound quite natural if not overdone. Outboard dynamic filters include the Symmetrix 511 and 511A, and Drawmer’s


January 2012 / w w w . s o u n d o n s o u n d . c o m

DF320, but I don’t know of a plug-in equivalent. The best way to recreate this process in your DAW is to use automation to control a low-pass filter’s turnover frequency (make sure there are no resonant peaks in the filter), but it’s laborious by comparison. The hardware processors also included expanders that came in at very low levels to mop up any remaining noise during pauses. This approach is particularly useful in cleaning up tape hiss and electric guitar recordings, but it can’t tackle low-pitched noise, and, unless the process is set up carefully, note tails can seem a little dulled. A more sophisticated approach is to split the audio into multiple frequency bands, to create a multi-band expander or gate. Dolby did this with their CAT43, a four-band device based on the Dolby-A noise-reduction system operating in a modified replay mode. It’s since been recreated as a plug-in by Waves. Four sliders set the expander thresholds for each band, and a fifth acts as a master. This doesn’t work well for broadband noise such as hiss, but it can reduce the level of consistent mechanical sounds such as camera noise and lighting whistles. Heavy processing will affect the wanted audio and create unnatural tone-shifting artifacts, but where noise contamination is low or moderate it can work well, and it has proven very popular in cleaning location dialogue. Being optimised for dialogue, this approach may improve a dirty vocal but is unlikely to be the best tool for musical instruments.

Gates and expanders can help you remove or conceal many unwanted noises. Some gates, like this one by Melda, offer multiple dynamics processors in a single plug-in, which can be useful.

Digital Solutions
Many of the processes I’ve described were originally implemented using analogue circuitry, but high-power power digital processing has made possible some serious advances in de-noising noising technology. The earliest digital methods were similar to the CAT43 process, but they split the audio into a far greater number of bands, sometimes several hundred. Many more affordable plug-ins, ins, including the broadband section of BIAS Sound Soap Pro, work in this way. CEDAR’s DNS series of dialogue noise-reduction reduction systems, which are very popular in broadcast and film production, also use multi-band dynamic noise-removal removal techniques. It would be impossible to set hundreds of expander thresholds by hand, so most systems allow you to take a ‘fingerprint’ of the offending noise by playing a short ‘noise-only’ only’ section. This is analysed and used to set the expander thresholds automatically. This fingerprint sample only

needs to be a second or so long. Tape noise can usually be taken from just before or just after the recorded performance, as the noise is continuous, but for noisy DAW tracks where you don’t have this luxury, you’ll need to find an exposed section of noise. Where no solo noise sample is available, some systems offer a selection of typical noise profiles (tape, circuit hiss and so forth), which can be adjusted up or down using a single threshold fader. Plug-ins Plug using this system usually have a graphic display to

CEDAR Audio’s ground-breaking noise-reduction software has proved so useful that they’ve been able to release dedicated hardware noisereduction processors like the DNS1000 pictured here, to speed up ‘cleaning’ tasks in professional broadcast, restoration and mastering work.

show both the noise-curve threshold and the spectrum of the wanted audio. There may also be tools to adjust the shape of the noise profile curve, to help optimise the settings for minimum side-effects. The above approach only really works well when the noise level and noise spectrum are reasonably constant. Otherwise, the fingerprint quickly becomes invalid, and either some noise will rise above the threshold and become audible, or, if the noise level falls, some of the wanted audio may be attenuated. Nevertheless, when used appropriately, such processors can bring about a significant reduction in noise, and because of their multi multi-band nature they can work across all frequencies: if no audio is present in a specific part of the frequency band, the corresponding filter bands will close down. The masking process is thus very effective, as when audio is present it only needs to mask noise residing in the same part of the audio spectrum. Common side-effects when tackling higher levels of noise include a dulling of top-end detail (again!) and a watery ‘tinkling’ or ‘burbling’ sound, as the various bands being muted affect the wanted audio in an unnatural way. The latter effect can be minimised by careful algorithm

w w w . s o u n d o n s o u n d . c o m / January 2012



design, using intelligent linking between frequency bands to smooth out their effect, but it’s also related to the amount of noise reduction being applied. In most cases, adjusting for a noise attenuation of around 10dB or less produces a good compromise between unwanted side effects and a useful level of noise reduction. I also find that processing in a couple of gentle passes often produces better results than a single, heavier application.

Variable Noise Floor
A changing noise floor poses a greater challenge, and may be an issue when treating a complete mix in which noisy sources, such as guitar amps or old synths, are turned up and down. Companies at the cutting edge of audio restoration, such as CEDAR Audio and Sonic Solutions, have devised algorithms that can recognise noise in the presence of a wanted signal and create a dynamic noise profile to follow them. Exactly how they do this remains a secret (but it stands to reason that if the human hearing system can detect noise, an algorithmic solution must ultimately be possible). One method would be to use a process known as auto-correlation, which looks for constants present in a varying signal. Using that method, the longer the analysis time, the more accurate the process of separating signal and noise can be. Similarly, the faster the noise spectrum and noise level changes, the less well software will be able to track the noise. This is certainly true for many of the more affordable packages that offer an ‘Auto’ mode: if there’s an abrupt change in noise character, the noise may become audible for a few seconds while the system responds to it, and changes its own noise-profile settings. Most noise-reduction software aimed at the home user tends to dull the wanted audio more in its automatic mode than if a fingerprint is taken, but further manual adjustments can often be made to find the best compromise settings. Popular software capable of this type of processing includes products from Izotope, TC Electronic, Waves and BIAS. To avoid problems at the start of a piece of audio, where the system is still trying to figure out the noise profile, try temporarily pasting a piece of the audio before the start, just to give the algorithm something to work from. Once the audio has been denoised, this section can be discarded.

Some early analogue noise-reduction systems were based on multi-band expanders, and a multi-band dynamics processor, such as the Universal Audio Precision Multiband shown here, can be pressed into service for nosie-reduction duties. In the picture, the top band has been set as an expander that attenuates that frequency band when audio falls below the threshold, and applies gain when the wanted part of the signal peeps above the threshold in that band.

The same strategy can be used if the noise floor changes abruptly and significantly: break the audio into sections, process as just described, then rejoin them.

Hum, Buzz, Clicks & Crackle
Some noises are, by their nature, more problematic. For example: hum, buzz, clicks, vinyl crackle, and single outside noises such as a slamming door or a car horn. None of these respond well to the processing methods I’ve described, so specific processes have been devised for each.

“In most applications, the complete removal of noise is counter-productive, because it draws attention to the processing and becomes a distraction.”
Removing hum is fairly straightforward, as it generally relates to the mains supply frequency and its immediate harmonics. For example, in Europe, where the mains voltage comes in at 50Hz, there may also be harmonics at 100Hz, 150Hz and sometimes even higher. (In the US, the mains frequency is 60Hz, so the harmonics will occur at 60Hz intervals.) Not all hum-removal processors can switch between 50 and 60 Hz, so ensure you choose a product that covers your needs. Narrow-band notch filters tuned to the fundamental and its harmonics will attenuate the hum while

having minimal effect on the wanted audio. The narrower the filter bands, the less the wanted audio will be affected, but if they’re too narrow, they may not be wide enough to accommodate small fluctuations in mains frequency (or recording speed, or sample-rate variations). It’s also possible for very narrow-cut filters to introduce resonance-like artifacts. Buzz can be more serious. The fundamental is still often related to the mains power supply, but another factor introduces waveform distortion and a more extended series of harmonics. Lighting dimmers are a common cause of buzz, as they reshape the mains waveform, leaving square edges with a lot of harmonics. Buzz can be addressed by a more extended series of filters at multiples of the mains frequency, but in my experience this is usually only partially successful, as really bad buzz has enharmonic components that escape the filters. Digital clicks caused by clock-sync problems may be less common than they were in the early days of digital recording, but they still happen: a mastering engineer working for a top mastering house recently told me that he spends a lot of time correcting digital clicks on otherwise high-end studio recordings! An individual click can often be edited out in your DAW’s


January 2012 / w w w . s o u n d o n s o u n d . c o m

waveform editor by ‘drawing in’ a new point, of course. However, some software can automatically differentiate between the fast rise time of a digital click and an intentional percussive sound. It can then replace the sample(s) causing the click with values interpolated from the samples either side of the click. Vinyl crackles and clicks are a little harder to deal with, as they can extend over many samples and occur far more frequently. Algorithms have been created, though, which can usually discriminate between these artifacts and intentional percussive transients. As the events are longer than a single sample, the process of filling the gaps in such a way as to hide the edit becomes more complicated, but when done well it leaves a pretty seamless join. CEDAR pioneered this field with their Declick and Decrackle products, but many other companies provide similar tools. Sonnox have come up with a refinement that prevents their restoration software misinterpreting the sound of some harsh brass instruments as crackles: a display shows the detected crackles on a level/frequency graph so when a brass instrument plays, any apparent crackles appear as momentary bubbles in a specific area of the screen. It’s then possible to draw a box around this area to exclude it from processing. In the case of de-crackling software, the user has to make adjustments to find the best compromise between side effects and noise reduction. The main side effect is that where the processing is intensive, so much reconstruction takes place that the cleaned

Spectral editors like CEDAR’s Retouch and the one in Steinberg’s Wavelab (shown here) allow you to perform incredibly precise edits,.

audio suffers noticeable distortion. To aid setting up, this type of software often has an audition mode that lets you hear only the audio that will be removed by the processing. If you can hear music as well as crackles, you’re probably over-processing! Clicks are dealt with using a different algorithm, as they tend to be less frequent than crackles and are usually much higher in amplitude, but, again, samples are generated to bridge the gap as seamlessly as possible. When cleaning up vinyl, it’s common to treat major clicks and pops first, then address surface crackle. Sometimes a third stage of broadband noise reduction can be beneficial to reduce more general surface noise. There are many programs of varying sophistication aimed at those transferring their vinyl collections to a digital format, but you really do get what you pay for. The expensive, high-end systems are usually capable of better results, as their reconstruction algorithms produce less distortion.

Things That Go Bump...
Until recently, the one type of noise that couldn’t be dealt with by signal processing was the cough or door slam in the middle of, for example, a string-quartet recording. Once again, CEDAR tackled the impossible with their Retouch software, which displays the audio as spectral content against time: the different frequencies are shown vertically, and the signal amplitude as different colours. Often, the offending noise can be seen on the display as a noticeable discontinuity in part of the spectrum, and, once it’s identified, the operator can draw a box around it, and either attenuate it to lie below the noise floor or replace it with material interpolated from the region around it. The
De-clicking and decrackling plug-ins, such as the Sonnox De-clicker shown here, have become quite sophisticated.

idea is that only the frequencies present in the intruding sound are processed, and then replaced by what I can best describe as ‘spectral fog’ to match what’s going on either side of the problem section. Retouch has been used to restore and improve countless recordings (including much of the Beatles catalogue), as well as to eliminate individual instruments from mixes to facilitate a Rock Band game! I’ve used the system several times to erase squeaky chairs and floorboards from classical concert recordings, to remove squealing car brakes from an ambient soundtrack, and to get rid of organ ciphers. It can also be used to remove just enough speech elements to render a profanity unintelligible in a song for radio play, without destroying the flow of the song, and to remove... well, almost anything that shouldn’t be there, without damaging the audio in any audible way. Sounds covering only a limited part of the spectrum are easier to remove without audible side effects than ones with a wide spectral spread, and short-duration sounds are easier to fix than long ones. From an operational viewpoint, every rogue sound usually has to be located and treated manually, so it can be labour intensive, but if the sound repeats regularly, you can select the offending region, and then select repeat regions at regular intervals. Even if you end up having to do your selections manually, though, you’re achieving the impossible, so who’s complaining? Several companies now offer spectral editing software clearly ‘inspired by’ the CEDAR model (Steinberg’s Wavelab includes similar tools, and the latest version of Magix Samplitude offers spectral editing on every track), but they all seem to have their own methods of synthesizing the signal

w w w . s o u n d o n s o u n d . c o m / January 2012



Noise Reduction In Practice: Combining Tools
Here’s an example of noise reduction in practice. I’ve placed some illustrative audio clips on the SOS web site at www. articles/noisereductionmedia.htm. The example is a bass-guitar recording, recorded by an artist for a demo, with a lovely lazy character and a nice laid-back, lilting style. However, the recording clearly suffers badly from buzz, hum and other noise, and becomes worse when adding typical mix processing such as compression. The file had already been edited, so I didn’t have access to any suitable sections of audio to provide a noise ‘fingerprint’. The first step was to tackle the all-too-audible 50Hz mains hum, and for this I turned to Tone Boosters’ excellent TB HumRemover, which allows you to apply a notch filter at a user-defined frequency, and creates notches to remove the harmonics. Achieving the best result is a matter of balancing what you remove with what you wish to retain, by altering the number of notches and their bandwidth (Q) settings, and setting the overall amount of attenuation. I was pleased with the result, as the bass retained its character but was much more audible, but there was still audible noise at the top end. My first instinct was to see if I could

A series of separate processors was deployed on a bass track to clean it up for use in a mix. Note that the compressor is not part of the noise-reduction process. It can be worth auditioning typical processing after the noise-reduction chain, just to see how usable the resulting file is going to be!

use EQ to roll off the top end. This removed some annoyances, but unless I brought the filter down far enough that it interfered with the bass sound, I could still hear gremlins. To tackle the residual problems, then, I inserted an instance of TC Electronic’s De-noise plug-in. This works most efficiently with a noise fingerprint, but as none was available, I bypassed the low-pass filter (so I could hear what I was doing) and waded through the presets. The ‘Strong Noise’ preset got me closest, and juggling the threshold and reduction level controls brought further improvement. I then switched on both the ‘draw’

and ‘audition’ modes, which allowed me to hear only the noise that I wanted to remove and to customise the curve so that the processor wasn’t removing any of the wanted signal. This got me some way towards the result I wanted, and reinstating the low-pass filter after the De-noise plug-in yielded a reasonable result. Finally, to check that the result would be usable, I jammed a compressor-limiter on the end and gave the signal a good squeeze. Although there were some audible issues, they were not show-stoppers, and I was happy that they’d be masked in a mix. Matt Houghton

that’s used to fill the part of the spectrum where the rogue sound used to be.

Pops & Shocks
In the battle against plosive pops and mechanical shocks, CEDAR were once again in the front line, with their DeThump plug-in. I’ve not had a great deal of personal of experience with that software, so I asked one of the CEDAR boffins to explain how it works. “The first problem with thumps,” he told me “is not removing them, it’s detecting them... because low-frequency events of that sort are — for any existing detection algorithm — often indistinguishable from wanted signal in the same range of frequencies. The second problem is that once the offending events are detected, their durations are long, and a conventional declicking approach (the best of which is generally superb for a few tens of samples, very effective for a few hundred, but begins to become less appropriate for a few thousand) is almost useless for events lasting up to 100,000 samples (which is a 1s thump at 96kHz). This is because it would create a partial drop-out — a direct consequence of the mathematics — at the centre of the interpolated region.

“Currently, the best way to overcome these problems is to combine a semi-manual detection method with an algorithm capable of sustaining the energy in the interpolation over an extended region. CEDAR’s Dethump combines these approaches. Firstly, it allows the user to select the duration of the thump visually within a File Processor screen or, in the case of a plug-in, the host editing screen. The user is then prompted to enter the number of visible cycles of the thump within the selected region. This is a simple way of entering a ball-park figure for the range of frequencies to be addressed, thus helping the algorithm to detect offending components. Once those components have been identified, a different type of interpolator can rebuild long durations of signal with the nasty bits suppressed. This allows genuine low-frequency content to survive and eliminates the risk of the ringing that might occur if deep, precise filters were to be used.”

Clean Up Your Act!
That pretty much completes this whistle-stop tour of noise-reduction technology. I’ll leave you with a few guiding principles about how best to put the tools to use. First, remember

that although there are ways to tackle most forms of unwanted noise, most of the tools aren’t perfect, and some require a bit of skill to get the best from them — so try to get things right when recording! Second, noise is only a problem if you notice it. A little continuous tape hiss, for example, might not be a problem, and its presence might mask a few other low-level ‘evils’. So use your ears to decide what’s needed, rather than trying to clean every track. If you have identified problem noises, though, all of the aforementioned processes can be used more effectively when you understand how they work and appreciate their limitations. Often multiple passes with a process, removing less noise with each pass, will help you keep side effects to a minimum. Similarly, you’ll often find that some noise problems, such as guitar-amp noise, respond best to a combination of different treatments (probably de-hum, de-buzz and broad-band de-noising, in this case). There’s some pretty powerful software now, ranging from free plug-ins to pricey professional tools, so if your recordings suffer from noise, there’s no reason not to give at least some of them a try!


January 2012 / w w w . s o u n d o n s o u n d . c o m


Hands Off
Save time and protect yourself from RSI by learning to edit audio in Pro Tools from the QWERTY keyboard.

f you use a computer all day, every day, you are at real risk of developing repetitive strain injury — especially if you work intensively with the mouse. One of the great things about Pro Tools is that it’s possible to edit fast and accurately just from the keyboard, and this month and next we’ll be looking at the techniques that power users employ to get work done faster and save their wrists!


is in the top right-hand corner of the Edit window and is one of several ‘a/z’ buttons around the Pro Tools interface.

There’s no point in getting really quick at editing if you take an age to find the right section of your session and zoom in on it. When I watch people working in Pro Tools, I’m often aware that they use a lot of keystrokes and mouse clicks to get to where they want to be, so let’s see if we can make this process easier and much more economical on mousing.

Hit the ‘a-z’ button to activate the Keyboard Command Focus.

Linking the track and edit selections makes it easier to navigate your session from the keyboard.

Before we do anything, we need to make sure that the Link Track & Edit Selection button on the toolbar is active. We also need to activate the Keyboard Command Focus option, which

This enables a whole raft of additional keyboard shortcuts that are going to prove very useful. To adjust the track height without using the mouse, simply hold down the Ctrl (Windows: Start) key and use the up and down arrow keys to change the track height of whichever track your cursor is on. If you have a set of tracks grouped together, Pro Tools will highlight all the track names and the keyboard shortcut will change the track height of all the grouped tracks. To zoom in and out horizontally with the Keyboard Command Focus enabled, you can use the ‘R’ and ‘T’ keys respectively, remembering that, as you zoom in, Pro Tools will place the cursor at the centre of the screen. This is very useful, but you do end up hitting the ‘R’ and ‘T’ keys a lot, so don’t forget that Pro Tools also has five zoom presets, which

I rarely see folk using. With Keyboard Command Focus enabled, the ‘1’ to ‘5’ keys above the QWERTY section relate to the zoom presets: hit the ‘5’ key and Pro Tools will jump to zoom preset five and so on. You’ll probably find the factory presets aren’t quite right for you, but these are easily changed. Use the ‘R’ and ‘T’ keys to set a zoom level you’re happy with, then hold down the Command (Windows: Ctrl) key and click on one of the five number keys. The relevant button on the Edit window will flash to acknowledge that the setting has been saved. You can repeat this for all five zoom settings, and it’s probably a good idea to follow the presets in having numbers one to five correspond to increasing levels of zoom. Remember that zoom presets are saved in your session, not as part of Pro Tools Preferences, so you will need to set them for each session, unless you use session templates. If you do, consider editing your templates with your preferred settings so that any session created from your template will acquire your zoom settings too. Zoom Toggle is another under-used but very handy feature. It’s activated using a button immediately below the Zoomer tool or, more easily, by hitting the ‘E’ key with Keyboard Command Focus
With the default Zoom Toggle setting, Pro Tools will zoom horizontally to fit your selection to the window, but keeps the vertical track height the same.


January 2012 / w w w . s o u n d o n s o u n d . c o m

With ‘Fit To Window’ active, your Zoom Toggle selection is scaled both vertically and horizontally.

enabled. Make a selection on a track in your Edit window; then, when you hit the ‘E’ key, Pro Tools will zoom in, so that your selection will fill the Edit window. The default setting uses the last track height that was set, and if that isn’t to your liking you can use the Track Height shortcut — Ctrl/Start and the up or down arrows — to set the Zoom Toggled track height to your preferred height. Then, when you hit the ‘E’ key again, it will take you back out to your previous zoom setting, but when you reactivate Zoom Toggle, by hitting the ‘E’ key for a third time, it will zoom in to your preferred track height. You can also make a system-wide setting for the Zoom Toggle function in the Editing tab of Pro Tools Preferences. In the lower right-hand corner, there’s a section that configures how the Zoom Toggle will function. My suggestions are to set the Track Height to ‘fit to window’ and the Track View to ‘no change’. These both default to ‘last used’, which is why the Zoom Toggle can sometimes perform a little oddly. You can also tick the ‘Separate Grid Settings When Zoomed In’ box to have a different grid setting when you activate Zoom Toggle, and I prefer to

have Zoom Toggle Follow Edit Selection as well. The benefit of using ‘fit to window’ is that when you make a selection across multiple tracks and then hit the ‘E’ key, all the selected tracks will fit the screen. You can also use the track navigation buttons ‘P’ and ‘;’ to move the selection up and down tracks. When you’re done, just hit the ‘E’ key and you’re back to your normal view. Another way to economise on mouse or keyboard actions is to use the left and right arrow keys. Say I’ve used the Zoom Toggle to zoom in and perhaps zoom in some more; all I can see is my clip start. In this situation, I see many people zoom out, scroll to the end of the selection, and zoom back in, which takes loads of keyboard and mouse operations. But using the left and right arrow keys to the right of the QWERTY keyboard, you can jump from the start of the selection to the end without zooming out — the left arrow takes you to the selection start and the right arrow to the selection end.

Tabbed Browsing
Another keyboard-based navigation tool is the Tab key. When I’m training people, I always describe the function of this key as ‘doing what it says on the tin’: it moves the cursor to the right until it hits the next vertical line, which in Pro Tools means a region (or clip) boundary. The Tab key thus makes it easy to move the cursor around and have it precisely land on the start or finish of a clip, or at an edit point. You can also make it go backwards: Alt-Tab will make the cursor go left to the previous clip boundary.

With Tab To Transient mode active, Pro Tools will find the next transient event in an audio clip when you hit Tab; you can activate Tab To Transient using the button in the toolbar under the Trim Tool button, or by the shortcut Command-Alt-Tab (Windows: Ctrl-Alt-Tab). This means you can easily trim a region without leaving the keyboard: use Tab To Transient to find the start of some audio, then hit the ‘A’ key. With Keyboard Command Focus active, this will trim off the start of a clip to the cursor position — neat! If you add the Shift key to these combinations, the cursor will highlight and make a selection as it moves; this also works when you use the Alt-Tab ‘reverse gear’ as well. This makes it very easy to highlight from your cursor position back to a previous edit and delete it again, all without leaving the keyboard. (Not only does this save time and mouse use, it reduces the risk of you dragging to make a selection and leaving the smallest clip in the world, then wondering at a later stage why a crossfade won’t work.) If you add Ctrl (Windows: Start) and Tab, as you tab each clip (we used to call them regions but with Pro Tools 10, Avid have standardised on a nomenclature that is the same for their video and audio products) Pro Tools will highlight the next one. This is a fast way of navigating around your session using the keyboard. Now that you can navigate your session from the keyboard, in next month’s issue we’ll be looking at the variety of editing actions that can be carried out without the use of the mouse. Until then, happy zooming!

w w w . s o u n d o n s o u n d . c o m / January 2012



Against The Grain
We demonstrate that there’s more to Live’s Grain Delay than bizarre effects.

f you’ve experimented with Live’s Grain Delay Audio Effects plug-in or used other granular synths and effects, you may be left with the impression that granular processing is chaotic, hard to control and mainly useful for creating sound effects. However, with a little care in choosing your settings, you can use Grain Delay to achieve subtle and very musical results that are beyond the reach of other types of effect — and that’s what we’ll be looking at in this month’s article.


What’s In A Grain?
Grain Delay performs a fairly simple process: it samples incoming audio in very small chunks, called grains, and emits each grain after a delay whose time you can set in milliseconds or sync to tempo. You control grain size, pitch-shift amount, pitch and time jitter (randomisation) and output settings. The grain size is set in Hertz (Hz) using the Frequency control, which you can think of as grains per second — higher frequencies mean more and smaller grains. You can calculate the grain size in fractions of a beat at your song’s tempo

by dividing the tempo by 60 times the frequency. For example, at a tempo of 120 bpm, a frequency of 4Hz (four grains per second) captures a half-beat (an eighth-note) per grain. But you don’t need to be overly concerned with grain size; its practical significance is that larger grains (lower frequencies) are more stable when you use feedback, pitch-shifting or jittering with the Spray and Random Pitch controls. A good way to experiment with Grain Delay is to place it after an electric-piano virtual instrument. Start with the settings in screen 1 and play a short note. The only effect you’ll hear is the note repeated an eighth-note later. Increase the Pitch knob

1: A Grain Delay embedded in an Audio Effects Rack with its controls mapped to the rack’s Macro knobs. I’ve colour coded the controls by function: orange for grain settings, green for pitch, blue for delay and yellow for output. Spray (top-left) is actually a delay setting, but its effect is to add jitter to the grain spacing, so I think of it as a grain setting. The numerical box next to Grain Delay’s Sync/Time button has different functions and separate mapping assignments in Sync and Time modes, and I’ve dedicated a knob to each.

2: A simple eighth-note swing drum loop (middle) is processed with an eighth-note Grain Delay (top). The Spray and Random Pitch settings produce small timing and pitch variations that are heard as slight alterations in the drum sounds. The Beat Swing setting adds a 32nd-note triplet to the eighth-note delay, which emphasises the swing feel. The bottom graphic shows the hits added by the Grain Delay.

setting to 2, and the repeated note will be shifted up a whole tone. Add 50 percent feedback and you’ll hear eighth-notes cascading up the whole-tone scale. For a modified version of these settings that produces a familiar and playable electric-piano effect, change the pitch shift to 5 (up a Perfect Fourth) or -7 (down a Perfect Fifth) and set the Feedback to 30 percent. Use a 16th-note delay (the button labelled ‘1’) for arpeggios, or use Time mode with a delay of a few milliseconds for chords and slightly longer for rolled chords. Frequencies above 4Hz start to get blurry and produce artifacts. You can make this effect even more playable by mapping the mod wheel to the Dry/Wet knob. Try it with sustained notes, intervals and chords, as well as with pitch-bend.


January 2012 / w w w . s o u n d o n s o u n d . c o m

3: Here I’ve automated Grain Delay’s Feedback, Spray and Dry/Wet controls to turn a four-bar rhythm-section pattern (guitar, bass and hi-hat) into a more interesting 16-bar pattern. Alternatives to automation include MIDI controllers such as mod wheel and expression pedal or manipulating Grain Delay’s on-screen XY controller. (You cannot use the XY controller for parameters that are mapped to Macro knobs.)

You can create more interesting note patterns still by using two Grain Delays in series. Set the second Grain Delay’s delay to twice that of the first, use feedback on only the second Grain Delay and experiment with different pitch-shifts. Make the first pitch-shift larger and set the second to +/– 1 or 2. The series can, of course, be extended with even more Grain Delays.

can take a lot of the repetitiveness out of a groove built from short loops. In the example shown in screen 3, I’ve started with settings derived from those previously described and then automated two of the knobs. The groove in question starts with four-bar guitar, bass and hi-hat loops. For the guitar, which is a monophonic line, I’ve used a five-semitone pitch-shift with a very short delay time. Turning up the Wet/ Dry knob doubles the line a perfect fourth

example, you can turn the highest quality sampled grand piano into an unruly bar-room upright with just a few Grain Delay tweaks. Start with a 50 percent Dry/Wet mix and a low Frequency setting. Experiment with delay times in the one to five millisecond range for changing the piano’s timbre. You’re not limited to integer values: you can type in values to two decimal places, and a few tenths of a percent can make a huge difference. Alternatively, map Time to a Macro knob and set its range maximum to 5ms, then use the knob or automation to adjust it. Set Spray in the 1ms range to vary the timbre of each note. Although you can use Grain Delay’s Pitch parameter to detune the piano by typing in values of a few hundredths of a semitone, I prefer to use Random Pitch. Values between 10 and 30 sound most authentic, but you can push it up to 100 for a really out-of-tune beast. For a phaser/flanger-like effect that works well with strings and pads, start with the piano settings, but change Random Pitch to 0.00. Increase Grain Delay’s Frequency and Feedback to around 15Hz and 60 percent, respectively. You can use either a clip envelope or track automation to vary the Time Delay knob in a triangle-wave pattern

Re-purposing A Drum Loop
You can add a lot of mileage to even the simplest drum loop with a Grain Delay. The process repeats the drum hits in the loop with slight alterations to their sound, so you want a fairly sparse loop to start with. You’ll want large grains (low Frequency settings) in order to capture whole drum hits for repeating. Use the Spray and Random Pitch settings to introduce small variations in the drum sounds. The settings shown in screen 2 are a good starting point — you’ll easily hear that the sounds become unnatural when you push those settings too far. Depending on the original loop, you’ll generally want to use delays of an eighth or 16th note. Positive Beat Swing settings will add a swing feel. Depending on the original material, try settings of 8.33, 16.7 or 33.3 percent.

4: These three Effects Rack panels reflect the settings for a bar-room upright piano, a phaser/flanger-like effect for strings and pads and extreme processing for a lead synth. The orange and black regions of the phaser/flanger Time Delay knob reflect clip-envelope modulation emulating a triangle-wave LFO. MIDI mappings for mod wheel (MW ) and sustain pedal (SUS) are also indicated.

Groove Enhancement
When you run up against the limits of your source material — construction kits with short loops and few variations, for example — Grain Delay can save the day. A little automation, either recorded or drawn in,

higher, and increasing the feedback doubles it again. The automation turns that into a 16-bar pattern. The 25ms delay adds some flam to simulate strumming. The bass part is doubled an octave higher with minimal delay, and because of the large pitch-shift, increasing the feedback sounds a bit like scat singing over the bass line. With the hi-hat, I’ve used a short delay and moderate pitch-shift, along with delay-time (Spray) and pitch jitter to change the colour without making the part busier.

Instrument Maintenance
Grain Delay is a great tool for processing acoustic and electronic instruments. For

and map the mod wheel to the Dry/Wet mix to fade the effect in and out. For lead-synth patches, concentrate on Time Delay, Frequency, Feedback and Pitch. I find the full range of those parameters useful, and I generally avoid Spray and Random Pitch. In particular, try maximum feedback and the upper-middle part of the frequency range. A pitch shift of 1 or -1 with a delay of around 30ms adds a granular up or down sweep to each note, and mapping the Pitch knob to a footswitch or sustain pedal with a range of -1 to 1 lets you change sweep direction on the fly. Mapping the mod wheel to the Dry/Wet knob is essential here — these are synth patches.

w w w . s o u n d o n s o u n d . c o m / January 2012




espite the well-designed user interfaces of modern DAWs, the repetitive moving, scrolling and clicking with a mouse can often feel clunky and sluggish, and in the long run can also cause problems with fingers, wrists, elbows and so on. While I’m the first to admit that defining and learning key commands doesn’t sound as exciting as, for example, getting to grips with a new soft synth (or even staying in on a Saturday night to wash your hair!), it’s something that will both make Cubase easier and quicker to use and help you to avoid suffering from mouse-related physical problems.


Key Facts
Mastering keyboard shortcuts makes Cubase easier to use and is good for your health!
defined, this is shown in the panel on the left side. If you select an unassigned command, you can then enter a key combination in the ‘Type in Key’ box. Usefully, if the key combination you try is already assigned to another command, Cubase politely tells you, and you can choose whether to overwrite the existing assignment or not. Otherwise, pressing the Assign button links the key combination with the command and, once the Key Command window is closed, the key combination becomes available for use. Key commands defined here are ‘global’, which means that they will be available in any Cubase project on the host system. Usefully, all the key command assignments that you make on your system can be saved as a preset. This is great if multiple users share the same host system and have different key command preferences. Providing you can find the file (search for it based on the preset name you used), it is also possible to move presets between computers, so you can take your key commands with you if you need to work away from your own system.
Key commands can be saved as presets to accommodate different needs or facilitate [moving between systems.

Right On Key
Frequently performed tasks such as project navigation — moving between windows or events, moving the playhead along the timeline, zooming in and out, or something more complex — can be made more efficient using Cubase’s key commands, providing you’re prepared to invest a little time in setting them up and practising using them. As an example of what key commands can offer, then, let’s learn how to ‘navigate’ using just the keys, and get there quicker. While there are a lot of useful default key commands in the standard Cubase installation, there are also plenty of additional functions that can be assigned a shortcut via the Key Commands window, which is accessible via the File/Key Commands menu option. By default, access to this window does not have a keyboard shortcut assigned (existing key commands are shown to the right of an item within a menu), so I’ll use this as a basic illustration of how to define your own key commands. At the top of the window, the search box allows you to find specific Cubase commands in the (very!) extensive list of possibilities. In the first screenshot example, searching for ‘key commands’ found an entry under the ‘File’ section. Where an item has a key command already
The Key Commands window allows you to define key combinations for your favourite Cubase commands.

Behind The Square Window...
Unless you’re fortunate enough to use a multi-screen system (and often even then), you’ll probably find that you’re always

switching between a number of the main Cubase windows. The Cubase ‘workspace’ system — with which you can define a window layout containing any combination of windows and then save them as a workspace — streamlines this process and is accessed via the Window menu. Once your workspaces are configured, the first nine can be recalled by pressing the Option (Mac) or Alt (PC) key and the assigned number key (1-9, but not zero, which is reserved for locking and unlocking the active workspace). The two obvious workspaces you might set up are a full-screen Project window and a full-screen Mixer window, although there are plenty of other possibilities. It’s also worth noting that individual workspaces can be ‘locked’, so even if you then make some temporary changes to it (perhaps opening a further window or resizing a window) while working, the original configuration of the workspace is recalled the next time you open it. There are other ways to open individual windows, of course. Default key commands will open the Mixer window (F3), the MediaBay (F5), the Video window (F8) and the VST Instruments list (F11), for example. (Mac users might want to check how they have the function-key behaviour configured in the System Preferences, but can still access these function key-based key commands in Cubase if they also hold down the ‘fn’ function key on the Mac keyboard.) Other heavily used windows also have pre-defined key commands, and a few of these are well worth memorising such as the Pool (Command-P for Mac, Ctrl-P for PC) and Marker windows


January 2012 / w w w . s o u n d o n s o u n d . c o m

Stickers & Keypads
If you’re one of those people who finds it difficult to remember which keys control which commands, you should consider investing in a set of keyboard stickers from Editors Keys, which aren’t expensive. These have both the standard QWERTY keys and shortcut labels on them. Bear in mind, though, that these will reflect the default shortcuts. The minute you start to reassign or overwrite shortcuts, the stickers will be incorrect! You could also keep your QWERTY keyboard clear of clutter by using a second, dedicated keyboard for the task. For example, I use a USB numeric keypad, to which I’ve assigned different shortcuts. By default, a lot of transport controls are assigned to the numeric keypad, but there can be more than 20 keys and two modes on such devices, giving a possible 40 separate commands — and that’s before adding Ctrl, Alt, Shift or other modifiers into the equation. This keypad even includes a couple of USB ports that enable me to keep my Cubase dongle and a USB pen drive with my Cubase preferences on it, so it’s easy to plug into another machine and be up and running quickly! Matt Houghton

Defining key commands for your most used ‘zoom’ commands can speed up project navigation considerably.

(Command/Ctrl-M), as well as the Project Setup window (Shift-S).

Project Prowess
For many Cubase users, the project window is where they spend the bulk of their time, and two sorts of keyboard-based ‘navigation’ tools can make life here a great deal easier, particularly on large projects: those for moving along the timeline and those for zooming in and out. There are a number of ways of moving along the timeline, each with their own particular uses. If you just want to get the cursor to step forwards or backwards, the Command (Mac) or Ctrl (PC) key and plus (‘+’) key combination, and Command/Cntrl key and minus (‘-’) key combinations move the cursor forwards and backwards along the timeline in increments controlled by the Snap setting. So if Snap is set to a bar or beat, each press of the key command moves the cursor by that
Workspaces can be defined via the Windows menu and then provide instant recall of screen configurations via key commands.

amount. Unless you have a penchant for working on 30-minute prog-rock epics, this is a very easy way to move around a project. An alternative approach uses the ‘N’ or ‘B’ keys to move the cursor to the next or previous event on the currently selected track. For example, if you have an audio track containing a series of audio events, each press of the ‘N’ key will take you to the start of the next event, the end of the same event, the start of the following event and so on — again, an easy and useful way to step through the timeline of your project. Incidentally, the up/down cursor keys can be used to step through the used tracks in the Project window Track List, so a combination of these two key commands allows you to move the cursor to anywhere in the project. A third approach is available to those who use the Marker Track and markers to highlight the locations of key points in the project. Shift-N and Shift-B move the cursor to the next or previous marker. So, for example, if you place markers at the start of each song section in your project, this provides a very rapid way to move along the timeline to the section you want to work on.

Shift-F sets the zoom level to fit it into the current Project window — magic! If you want to zoom in/out vertically on the currently selected track, the preset key commands Option-up arrow and Option-down arrow (use Alt on the PC) will get the job done. However, if you want to simultaneously zoom all tracks vertically in or out, you’ll need to set up some appropriate key combinations in the Key Commands window, as these are not defined by default. I’ve set these as Option-G and Option-H on my Mac, so that the ‘G’ and ‘H’ keys control both horizontal zoom (without the Option key) and vertical zoom (with the Option key). Of course, you could similarly use shortcuts to increase or reduce the height of the waveform within an event.

Endless Potential...
I’ve focused on the key commands that help you with the most common Cubase navigation tasks, because these offer the most immediate potential for streamlining your way of working with Cubase, but what’s covered here is merely the tip of the iceberg. Almost any operation can be executed via a key command, so if you find yourself regularly repeating a particular operation, learn or set up a key command for it. You don’t have to stop at the preset commands, either — you could create Macros (combinations of commands and processes) and assign those to a single key. It should be a simple matter, for example, to use a single key to jump 10 events forward or back on the arrange page. Learning key commands won’t make you ‘sound’ better, but it will let you capture and manipulate your sound more efficiently, more quickly, and with less chance of physical damage to your all-important hands and arms. John Walden.

Zooming About
Aside from moving around the timeline, the other operation repeated many times over in the average project session is changing the zoom level. Key commands provide access to a number of ‘zooming’ tools. At the most simple level, ‘H’ will zoom in and ‘G’ will zoom out horizontally. And if you want to see all of your project at once,

w w w . s o u n d o n s o u n d . c o m / January 2012



Winding Down
Take it slow with Logic Pro, as you get to grips with this vinyl effect...


hough over-used in some musical circles, the effect of a vinyl record or analogue tape being slowed to a stop and then starting up again can still be a useful musical device to add interest to a track. Logic includes tools for emulating this effect very easily, but before applying them, a little bit of physics helps us to understand what kind of curve is required for the slow-down part of the effect. The energy contained in a rotating flywheel, which is what a record deck is, when it comes down to it, is proportional to the square of its rotational speed. In practical terms, this means that if you put your hand on it to apply friction, the rate at which the turntable slows increases as time passes. In other words, the slower it gets, the less effort it takes to slow it even more, so the slow-down curve gets


steeper towards the end, rather than being a straight ramp. In Logic, however, you can change the shape of the curve according to what sounds good to you: you don’t have to stick to the laws of physics if you don’t want to.

Hidden behind Logic’s Fade facilities are all the tools you need to create a convincing vinyl slow-down effect in the middle of your track.

The Vinyl Countdown
So how do you apply this knowledge in a DAW? In Logic Pro 9, the speed-change tools are hidden away behind the fade-in and fade-out facilities: • The first thing to do is split the audio file where you want the speed change to occur and move the two sections apart. • Use the Fade tool from the toolbar menu to draw a fade-out and fade-in at the approximate positions you’d like the speed changes to occur. Don’t worry about accuracy at this stage, as everything is adjustable.

• If you play the file now, you’ll just hear fades, which is not what you want. To fix this, go to the Inspector at the left of the page, click in the fade-out area with the Fade tool active and then click on the word ‘Fade’ in the Inspector. This will open up a box with two choices: Fade or Slow Down. • Select Slow Down. Your fade curve will change from grey to orange. For the fade-in part of the following section: • Again, select the fade area. • Go to the Inspector, and this time click on Fade In, which opens up a box that gives Speed Up as an option. • Select Speed Up and, again, the fade curve changes to orange.


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To give each end of your split audio regions slow-down and speed-up effects, you need to change the nature of the fades in the Inspector.

• Play the file and you should now hear the first audio section slow to a halt, then pick up again after the break. It’s unlikely that the effect will be quite as you want it just yet, so use the Fade tool to drag the middle of the curve upwards, producing an exponential curve that gets steeper towards the end. You can then adjust the length of the fade ramp, using the same tool, to get just the effect you want. For the speed-up at the start of the next section, adjust the curve shape so that the initial speed-up is fast, levelling out towards the top of the curve. Again, this mimics the way a real turntable takes more energy to speed it up as the rotational speed increases. Tweak the curve length until the effect sounds natural and you’re done. Just bounce the regions ‘in place’ to capture your effects as finished audio once you’re happy with the settings.

the slow-down in two stages, bouncing the file between each stage, which is what I did for the audio example that accompanies this article. The first stage was to slow down only the last beat or two of the section, using the opposite curve shape so that it flattens

of the slow-down stretched out longer than would occur in real life — call it poetic licence. There’s another alternative, and that’s to use Logic’s Region Stretch facility to extend the length of the last beat or two before applying the speed

Although the above settings get close to the sound of the real thing, the tail-end of the slow-down can sometimes sound a little too abrupt. Having the last sound grind slowly into silence can be more dramatic, and there are a couple of ways to achieve this. The first is to do

out towards the end. After bouncing this to a new file, a longer slow-down was applied using the original ‘steeper towards the end’ curve shape. By taking a two-stage approach like this, you can depart from the true ‘laws of physics’ scenario to have the tail-end

You can now make your slowed-down and sped-up audio convincing by altering the speed at which the effect works. Grabbing the middle of the curve and dragging it upwards starts to create the effect you need.

Audio Example
To listen to this technique in action, go to articles/logicmedia.htm.

change. This entails first separating the beat to be stretched. It then has to be glued back onto the earlier part of the region after being extended. In my experience, the dual slow-down sounds more natural, but natural isn’t always what you’re after! And that’s it: an easy effect to create, but one that should be used wisely.

w w w . s o u n d o n s o u n d . c o m / January 2012



Flip For It
Have some phase-switching fun in Sonar X1!


effects, but let’s do a few things to make the sound more extreme: • Insert the Classic Phaser on an audio track and edit it for your phasing sound of choice. • Copy the audio from your primary track to create a second, parallel track of audio. You don’t want the Classic Phaser in this track, so if you clone the track, delete the effect, or at least bypass it. • In the Console view, click on the phase button in the audio-only track. • You can user the fader of the

ong-time readers probably know that, whenever possible, I try to make this column applicable to programs other than Sonar. Granted, many of these techniques are easier or more convenient to implement in Sonar, and reference Sonar features and effects, but if you use another DAW, don’t turn the page yet. As long as you have a way to flip a track’s phase, you’ll probably be able to translate these techniques to whatever DAW you use. Phase switches are like the Rodney Dangerfield of consoles; they don’t get no respect. A lot of people seem to think of a console’s phase switch solely as “that thing to flip when some idiot wired the XLR cable out of phase”. But being able to flip phase can give some really intriguing special effects, and Sonar makes the process easy: not only does every console channel have a phase-flip switch, but so does the VC64 Channel Strip (which is particularly relevant, as it allows for parallel processing). The general way to set up a track for phase-switching fun is to copy the audio from one track to another track to create a parallel signal flow, then change the phase on one track. It’s a little more complicated with buses, as buses don’t have a phase switch, but there’s a simple workaround we’ll get into later. Meanwhile, let’s look at some applications. First, something I like to call ‘Extreme Classic Phaser’. Cakewalk’s Classic Phaser plug-in does indeed give classic phaser

imaging gets wider as well. Use the audio-only track’s fader to set the desired amount of cancellation.

Dynamic Saturation
Here’s a really unusual phase-flipping application. I like using saturation from time to time, but find that with percussive audio, like drums, the ‘flattening’ of the waveform and general muddiness can reduce dynamics dramatically. You can always try the ‘mix in some straight signal’ technique, as you would with parallel compression, but then you start to compromise the saturated character. The following technique uses phase cancellation to help retain dynamics. This works because saturation affects the highest-level signals the most, which, of course, are the percussive peaks. The lower-level saturated signals are much more like the dry sound, so combining the saturated signal with the phase-flipped dry signal tends to cancel the lower-level signal while leaving the percussive peaks intact. Here’s the procedure. To get a feel for how it works, first load a drum loop or other drum part: • As with the Extreme Phaser application, clone the audio from your primary track to create a secondary, identical track. • Insert the saturation effect in your primary track. I generally use the Pro Channel’s Tube Distortion for this application because (depending on the source audio) being able to choose between the Type I and Type II saturation options can make a big difference to overall effectiveness. However, I’ve also tried this technique with the Softube Saturation knob that’s part of Sonar X1e, and it works too. • Start playback and adjust the Tube saturation controls for the desired saturation character. Don’t be concerned if you want to pile on the distortion. We’ll tighten it up. • Now turn up the secondary channel’s fader. As the level gets closer to matching the first audio channel’s level,

Here we’re looking at phase-switching applications using the Extreme Phaser. Note that the Classic Phaser is inserted on Track 1, while Track 2 has been flipped out of phase. The Classic Phaser and phase switch are circled. The Track 2 level fader sets the precise amount of phase cancellation.

audio-only track to change the level of the dry signal. As you change the dry level, you’ll find a point where the dry signal cancels with any ‘remaining’ dry signal in the Classic Phaser effect, leaving only those parts of the effect that are different from the dry signal. Not only does the phasing effect become more intense, the stereo


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We’ll use send buses for the reverb, so we don’t need to copy the audio to another track, as with the other techniques that flip channel phase. But we do need two buses. Enter Sonar’s Send Assistant: Right-click on the track where you want to add reverb, and select Insert Send / Insert Send Assistant. • Specify a New Bus, with Master as the Bus Output. Name the Bus ‘Reverb 1’. • For Choose Effect, select the Sonitus:fx Reverb and also click on Show Effects Property Page. • Click on OK to create the bus. • Now create a second bus in a similar way to how you created the previous bus, but don’t choose an effect, and give the bus a different name (for example, ‘Reverb 2’). • Start playback and adjust the reverb for a sound you like. For voice, a good preset is Aux: Vocal Hall 1. If you create your own preset, remember to turn Dry down all the way, as this is a send effect. • Make two edits before proceeding: choose Stereo instead of Mono (if needed) for the output, to create a bigger sound, and change the Decay Time to one second. • Once you have the right reverb sound set up, control-drag the reverb into the other send bus’s FX bin so that the
The vocal track is being sent to two reverb buses. Note that the Reverb 1 bus has Channel Tools inserted in the FX Bin to provide phase changes; unlike audio channels, buses don’t have a dedicated phase-flip button.

same reverb is in both tracks. • Now we need to flip the phase on one of the buses. Although buses don’t include phase-flip switches, there’s an easy solution: insert the Channel Tools plug-in into one of the buses. • With Channel Tools, turn Input Mode and Delay to Off, then enable the Left and Right phase-flip buttons. The reverb effect should now cancel. • On one of the reverbs, increase the decay time; try 1.8 to 2.0 seconds. This cancels the reverb’s initial decay, leaving only the ethereal tail at the end. For comparison, turn off Channel Tools; you’ll hear a huge difference. But note that, as with other effects that are based on phase-flipping, the level of the combined sound might be a little low. Grouping the bus output controls lets you raise or lower them together if needed.

Why Stop There?
We’ve covered three phase-flipping applications in previous columns: using phase flips to create vintage wah-wah sounds, telephone/megaphone voices and the kind of over-compressed, ‘sucking’ drum sounds used as a special effect in a lot of music from the psychedelic ‘60s. There are probably more cool sounds just waiting to be discovered, so give the phase switch some respect and you just might come up with something no one else has heard before.

The Main Audio channel incorporates the Tube Saturator, set for Type 1, in the Pro Channel; a slight bass boost compensates for any thinness that can result from the cancellation process. Channel 2 has its phase flipped.

the individual drums will become more distinct. Note how the channel meter indicates a more dynamic signal. The greater the cancellation, the more the level will tend to drop. As it takes some tweaking to get just the right balance of phase-flipped to processed audio, it’s helpful to group the level controls for the two audio tracks, so that they track each other if you want to change the level. Right-click on the level control for each track, assign each one to the same group, and now you can adjust the level without changing the distortion effect. You might want to increase the bass a bit to compensate for any thinness that results from the partial cancellation; distortion affects the high-frequency content more, so low frequencies will have more tendency to cancel.

In The Ether
And now the pièce de resistance: a reverb sound you haven’t heard before, except possibly in your imagination. This phase-flipping effect works best with algorithmic reverbs, like the Sonitus:fx Reverb. The result is an ethereal reverb sound that’s particularly useful with vocals, but works with other instruments as well.

w w w . s o u n d o n s o u n d . c o m / January 2012




here’s still plenty to get our heads round in Reason 6, so this month we’re going to have a look at two completely different aspects of the new program. Firstly, we’ll examine the new Pulveriser effect and then we’ll go on to explore what version 6 can do with time-stretched audio.


New Tricks
the Squash and Dirt amounts, and also Dirt’s accompanying Tone control. On more dynamic loops, the Release control will also come into its own — experiment with it to really get things pumping — but it’ll make little difference if your loop maintains a uniform level throughout. The next crucial feature is the Blend control. This is a wet/dry mix. We’re not used to seeing those on compressors and distortion units, but it’s a brilliant feature. When using Pulveriser to treat drum loops or patterns from Kong or Redrum, this lets you blend the crisp original audio with the squashed and dirtied contribution from Pulveriser. Simply adjust to taste. It’s effectively a simple version of parallel compression. This is a real power-user’s technique, often mentioned in SOS producer/engineer interviews, which is frequently used to fatten up drums without destroying their original character. If you only every used Pulveriser’s Squash, Dirt and Blend, you’d already be getting your money’s worth. But there’s more to it, and the remaining sections are a tweaker’s delight. The filter section is like a synth’s multi-mode filter. Select the Low-pass 24 or LP12 + Notch modes, turn down the Frequency knob a bit, keep Peak turned down, and the result is even more lo-fi, progressively numbing the

We continue our exploration of the new features and functions in Reason 6.
signal’s high-frequency content. The Comb mode adds flanger-like overtones to the signal, and turning up the Peak control adds weird robotic tones.

A Handful Of Dust
Pulveriser is one of Reason 6’s new effect devices, and with a name like that you might imagine some sort of brutal distortion unit, especially as the subtitle or tag-line is ‘high yield demolition’. But hang on a minute — hasn’t Reason already got one of those, in the shape of the Scream 4 Sound Destruction unit? Absolutely right, and in fact Pulveriser is actually a more subtle and flexible tool than you might at first think. What it offers is characterful single-band compression (Squash), old-school valve-type distortion (Dirt) and a multi-mode Filter with cutoff (Frequency) and resonance (Peak) controls. Used by themselves, these get you a long way. But there are modulation options too, as we’ll see. Try this for starters: load up a loop or two in a Dr OctoRex and hit Run. Right-click on Dr OctoRex and choose Effects > Pulveriser Demolition to patch in the new effect device. Immediately there should be a whole lot more attitude! The default patch combines some Squash compression with a bit of Dirt distortion, and nothing more. You can quickly tweak the effect by adjusting

Modulation Explanation
The remaining sections — Tremor and Follower — are modulation sources. They’re capable of producing more different effects than I can describe here, but here’s one example that demonstrates what’s possible. Try applying a Pulveriser to an electric piano or clavinet sound. For now, turn Squash and Dirt right down. Set the Filter to Low-pass 24, the Frequency knob to about the 12 o’clock position, and Peak to about 2 o’clock. The result so far is a more muted, slightly twangy version of the original sound, but it gets a lot more interesting when we introduce the modulators... Follower measures the level of the input to Pulveriser, and generates a corresponding modulation amount, which can be applied to the filter frequency parameter using the knob to its left. Try raising that knob to about the 1 o’clock position. Make sure the Threshold control is right down, and raise Release to about halfway. The effect is auto-wah — as you play, the filter now tracks the volume of your performance. Now, how about some authentic electric piano tremolo? For this we need the Tremor section, which is an LFO, pure and simple, with nine waveforms. Select the square wave using the up and down buttons, and set the rate control to about the 10 o’clock position. You’ll notice

One of Reason 6’s new effects devices, Pulveriser is more than just a character compressor — it can be used for a wide range of sonic treatments.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Tremor and Follower settings for a combined auto-wah and tremolo effect, as described in the main text.

sequence, and most likely it’ll be one, two, four or eight bars long. Listen to the results, experiment a bit and see what works best. 7. When you’ve nailed the length of your loop, you can easily repeat it to form the basis of a backing track. Just select it, and hit Command-C (Mac) or Ctrl-C (Windows) to copy it. The playback wiper helpfully jumps to the end of the loop. Then use Command-V (Mac) or Ctrl-V (Windows) to paste it as many times as you need. The reason this all works so well is because of Reason’s superb real-time time-stretching capabilities on playback. You can now vary your sequence tempo to whatever you need, and the loop will continue to ‘conform’ to your sequencer

from the knobs either side of the Tremor section that it can modulate the Filter or Volume control, or both. For tremolo, you need to control the volume, so raise the right-hand knob to about the 2 o’clock position. The result is tremolo, but it’s clicky. You need to soften the edges of the square wave, and that’s what the Lag parameter is for. As you raise it, the Tremor waveform becomes more and more ‘rounded’. There’s another option too: if you press the Spread button, the tremolo effect is split between Pulveriser’s two output channels, and the effect becomes auto-pan. The more you experiment with Pulveriser, the more you realise how versatile it is. Happy pulverising!

Reason will also confirm some technical details about the file. 3. Click Open. Reason will dismiss the file browser and place the loop into your sequence. If you’ve already selected an audio track (with the correct channel configuration), the loop is inserted in it at the playback wiper position. If not, Reason will create a new track for you. As the vast majority of WAV and AIFF loops don’t have tempo metadata embedded, the likelihood is that the imported loop won’t match your sequence tempo. So try this: 4. Make sure your loop begins exactly on beat 1 of a bar. If it doesn’t, the quickest way to remedy that is to first select it, then use the Position text field at the top of the sequencer to type in a location directly. For example, typing in 3. 1. 1. 0 moves the loop to Bar 3, Beat 1, 1/16th 1, subdivision 0. 5. In the sequencer, turn on Snap and set the resolution to Bar. 6. Select the audio clip, then hover your mouse pointer over the right-hand clip resize handle. Now hold down the Alt key (Mac) or Ctrl key (Windows) — the mouse pointer changes to a ‘scale tempo’ icon — and drag the clip. The length will snap accurately to bar divisions. You’re aiming to make the clip best match your

Audio clips can be time-stretched manually by dragging their resize handles while holding down a modifier key.

Audio Stretching
Reason 6 doesn’t have the Melodyne-like ‘elastic’ audio that some other DAWs offer, but it does offer more in this general area than previous versions did. Let’s look at one really common scenario where this can come into its own: importing audio loops. For an application that is so modern and progressive in other ways, Reason has surprisingly poor support for audio loops. You can, of course, load sliced REX-format loops into Dr OctoRex, either from the Factory Soundbank, commercial ReFills, or those that you’ve created yourself in ReCycle. But beyond that, the only audio formats Reason will deal with are plain old WAV and AIFF files. So let’s say you’ve got a loop in one of those formats that you want to time-stretch to use in a song. Here’s a good way forward: 1. From the File menu, choose Import Audio File, or hit Command-Alt-I (Mac) or Shift-Ctrl-I (Windows). 2. In the file browser that appears, navigate to where your loop is stored. You should be able to preview audio using the Audition section’s Play button, to make sure you’ve selected the right thing.

time ruler, as it remembers its length in bars and beats. There’s one thing to check, though, or to experiment with. The quality of time-stretching is controlled on a per-track basis and it’s set in an audio track’s corresponding Audio Track device in the Rack. That’s easy to find: just click the sequencer track in an empty part of its settings section at the left side of the sequencer, and the rack scrolls to show its rack device and flashes it for good measure. You need to be able to see the rack for that to work, of course. Then check out the Stretch Type pop-up menu. The Allround setting works well for most things, but Melody and Vocal can give a different effect, and might be a better choice for sustained textures.

The clip parameters at the top of the sequencer section are invaluable for setting clip length, position and transposition.
w w w . s o u n d o n s o u n d . c o m / January 2012


T E C H N I Q U E / D I G I TA L P E R F O R M E R

Mach Speed
Get up and running with MOTU’s MachFive 3, an ideal software sampler partner for Digital Performer.

achFive 3 is MOTU’s latest virtual instrument: a full-blown software sampler with capabilities to rival industry standards like Native Instruments’ Kontakt. As with all MOTU’s commercial virtual instruments (that’s to say the ones that aren’t bundled with DP), it’s developed in collaboration with French company Univers Sons, incorporates some of their UVI Soundsource libraries, and has a strong family resemblance to their UVI Workstation sample player software. Even though MachFive 3 is a genuinely cross-platform product, with Mac OS and WIndows versions, it’s still an ideal match for Digital Performer running on the Mac, so this month I’m looking at how DP users can get up to speed with this great new virtual instrument.


Installation Matters
Here’s a turn-up for the books: MachFive 3 doesn’t have a MAS version, only Audio Units, RTAS and VST. But since DP’s Audio Unit plug-in format hosting is so good, it’s really of no consequence. The AU version

will show up in your instrument plug-in lists along with your other AU plug-ins. Usefully, too, MachFive 3 won’t overwrite a previous MachFive 2 installation, but just co-exists with it. Then, if you open a DP project that used MachFive 2, it’ll

Multiple Outputs
MachFive 3 has a great on-board mixer for all its internal parts, but there are times when you might prefer to bring some or all parts individually into Digital Performer’s Mixing Board, to treat them with your favourite MAS or Audio Unit plug-ins. Here’s how you do it, working with a conventional stereo MachFive 3 instance: 1. In the MachFive 3 plug-in window, make sure you’re viewing the parts listing (the ‘Parts’ tab should be selected at the top left of the plug-in window). You’ll notice that, by default, all parts are set to use Main Out (visible under the volume slider), which means that they output through the Instrument track MachFive 3 is instantiated on. 2. To remove a part from that main output and pipe it directly into DP’s Mixing Board, click the output pop-up menu — directly on ‘Main Out’ works well — and choose one

of the other 16 stereo bundles. 3. In DP, create an Aux Track from the Project menu. Then in the Mixing Board or Tracks Window, for example, click on an audio input pop-up menu and choose the corresponding MachFive bundle.

still use MachFive 2 — and that’s great for consistency and stability. MachFive 2 and 3 will actually run simultaneously without any trouble. Oh, and if you’re wondering about how the new version gets along with MachFive 1, that’s actually a moot point. Version 1 only runs on PowerPC-based Macs, and version 3 only on Intel Macs, so never the twain shall meet… As far as backwards compatibility goes, MachFive 3 will open version 1 and 2 presets and multis with no problem, and will open their bundled sound libraries too. So if you’ve got a DP project that uses an instance of MachFive 2, for example, and you want to ‘upgrade’ it to use your new MachFive 3, do the following: 1. In MachFive 2, click the ‘SAVE P[erformance]’ button in the display at top left of the plug-in window, and save the file somewhere convenient, perhaps on your desktop, or in the DP project folder. 2. Now instantiate MachFive 3, and click on the button with the spanner icon near the top of the plug-in window. Choose Load Multi (notice the different terminology – it’s not ‘Performance’ any more) and use the browser to locate the MachFive 2

The naming scheme between MachFive 3 and DP won’t be quite the same. For example, the stereo out pair that a single MachFive calls ‘Out 2’ is called ‘MachFive3-1 3-4’ in DP. But if you remember MachFive has 16 stereo out pairs as well as its Main out, you can tally them up with the 16 DP MachFive inputs.


January 2012 / w w w . s o u n d o n s o u n d . c o m

performance you just saved. Double-click it, or select it and click the OK button. 3. Now you’ll have to work through all the MIDI tracks that were assigned to parts/channels in MachFive 2 and reassign them to the same ones in MachFive 3. You should now be up and running. The only thing I’m conveniently overlooking is any automation of MachFive 2 parameters. That’ll be coming from the MIDI tracks driving individual parts, in the form of MIDI continuous controller messages, and ought to map pretty well to the new MachFive instance. In the case of some very specific automation mapping, you may need to set this up again in MachFive 3. Once you’ve located the parameter, right-click it for a contextual menu from which you can assign it a specific MIDI continuous controller type (see screen showing MIDI CCs). Don’t worry that you’re suddenly configuring ‘modulation’ — that’s what MachFive 3 calls any control of its parameters, whether from its onboard modulators or an external MIDI message.

When It’s 64...
MachFive 3 runs as a 64-bit stand-alone application and plug-in if your OS and sequencer host support it. That said, with DP still very much a 32-bit application at the time of writing, DP users can’t yet take advantage of the memory-related advantages that a 64-bit plug-in architecture brings. However, there is one thing that a few users will want to watch out for, and that’s compatibility problems with the MOTU sound libraries Symphonic Instrument (often called MSI) and Ethno version 1. In short, MachFive 3 running as a stand-alone application refuses to open them. That’s because
Here I’m adding my main sample library location to the (favourite) Places section of MachFive’s browser.

it opens in 64-bit mode, even if you think your Mac and OS are only 32-bit. There is a fix. Find the MachFive 3 application in your [Hard Disk]/Applications folder, select it, hit Command-I to ‘Get Info’, and tick the ‘Open in 32-bit mode’ tick box. However, the fix is at the expense of that nice 64-bit memory addressing, which could hit some power users. And it’s not really clear whether the incompatibility persists when MachFive 3 is used as a 64-bit plug-in in a 64-bit host application. For their part, MOTU say they’re aware of the issue and are working on a solution. If and when we see a new 64-bit version of DP, I’ll revisit this thorny little issue.

Soundbanks & Places
MachFive 3 lets you store your samples and sample libraries where you like. There isn’t a ‘default folder’, and that makes it much easier to store your sample library where you need to, even spread across several locations. To deal with this, the file browser in MachFive 3 is slick and quick, but a couple of little tricks make getting at your sample libraries that little bit quicker still.

First, you’ll notice the top-level item called ‘Places’ in MachFive 3’s browser. By default, in my OS 10.6 installation at least, the locations included there are Desktop, Documents, Home folder and Music, none of which are where I store my sample libraries. So try this: open up the Devices item (by double-clicking or clicking its disclosure triangle), then drill down into your disks to locate a folder that contains your sample-library files. Then click and drag the name of the folder back onto the leftmost browser column. The column goes white to confirm your action, and if you drop the item there it’s permanently added into the Places list. Handy for next time! To zap those not-very-useful default Places while you’re at it, right-click them one at a time. There’s only one contextual menu command — Remove from Favourite Places — so if you choose it, the Place disappears. Second, there’s an even more convenient way to access the bundled MachFive 3 library, and other sample libraries by MOTU or Univers Sons: via the Soundbanks item. You might notice Soundbank names appearing there if you mount them manually (browse your way to them and then double-click them to ‘mount’ them). All very well, but these Soundbank shortcuts are not remembered by MachFive 3 next
Right-clicking on most knobs or sliders in MachFive 3 brings up a staggering array of modulation options, including MIDI CC messages from your controller keyboard or DP track.

time you load it, so to get all your MOTU/ UVI libraries easily accessible, do this: 1. Make sure the browser overlay is closed and you’re viewing MachFive 3’s main window. Then click the spanner icon at centre top. Choose Preferences. 2. In the Preferences window, open the Soundbanks tab by clicking its icon. 3. In the Search Patch column, click the browse button (three adjacent dots. 4. Navigate to the location of your sound libraries, select the folder, then click Open.

5. In the Preferences window, tick Indexed, Recursive and Automount. Respectively, these make MachFive add the programs within Soundbanks to its browser Search function, look in any folders inside the folder you just configured, and automatically add Soundbanks to the browser list. 6. Repeat steps 3 to 5 to add more locations if needed. Close the Preferences window. Now, when you bring up MachFive 3’s browser (by double-clicking the word ‘Empty’ in the part list, for example), all your Soundbanks will be immediately under the Soundbanks item.That holds true for all future uses of MachFive too.

w w w . s o u n d o n s o u n d . c o m / January 2012



Act On Impulse


ast month’s Reaper column looked at creating convincing convolution reverb, using the comprehensive processing capabilities offered by the ReaVerb plug-in. This month, we’ll investigate tools within ReaVerb that allow you to capture your own impulse responses of reverberant spaces on location, as well as some further creative possibilities offered by the Impulse Generation section of the plug-in.

Part Two: Record your own impulse responses with Reaper’s ReaVerb.

Get Ready
Prior to recording an impulse response on location, you’ll need to set up an appropriate project-file template in Reaper. This prepares the test (or ‘excitation’) signal for playback in the acoustic environment and the subsequent recording of the impulse response. • Open a new Reaper project and save it, naming it something like ‘IR template’. • Ensure that buffer settings on the audio driver are set as low as possible to minimise latency when recording. • Create three new audio tracks. The first
Creating a four-second sine-sweep test tone and saving it into the designated project folder. The Impulse Generation utilities are only visible when ‘File’ is selected in the Impulse Generation section of the plug-in.

audio track will be a utility track that is only used to generate and decode the test signal for the impulse response.

Name it ‘Utility’ and insert the ReaVerb plug-in into the signal chain using the FX button. • Press the Add button in the Impulse Generation section and select File. Since we don’t wish to load up a pre-existing impulse response, select the Cancel option when prompted. • Towards the bottom right-hand corner of the plug-in window are the ‘Impulse Generation Utilities’. Using the Generate Test Tone button, create a four-second test signal named ‘Sweep’ and save it in the project folder. This action produces a sine-wave sweep from 20Hz to 20kHz, used to excite the room acoustics when measuring the impulse response. The project file is then prepared for impulse response recording using the second and third audio track:

This is the impulse-response project template ready to go. Carefully placed locators surround the test tone, the auto-punch record mode is enabled and the third track is named to appropriately reflect the acoustic space being measured.

• Navigate to the ‘Sweep’ audio file on your computer and drag and drop it onto your second audio track in


January 2012 / w w w . s o u n d o n s o u n d . c o m

Bonus Content!
There’s more on the techniques discussed here in the expanded online article. Check it out at jan12/articles/reaper_notes_0112.htm.

• •

Reaper. The track name will change to reflect the name of the audio file. Set the locators so that they accurately encompass the Sweep file on screen. Right-click the record button on the transport bar to select ‘Record mode: time selection auto punch’. This ensures that the sweep and the recorded impulse-response audio files end up the same length as each other. The third track should be named according to the room or acoustic environment you will be measuring and is where the impulse response file will be recorded. Select the appropriate input(s) and arm the track for recording.

After the first impulse response is recorded, subsequent alternative responses can be recorded using the same test signal. The simplest way to do this is to create a new audio track for each response, name it accordingly and then record. Naturally, when using this method you need to take care to mute all other tracks containing recorded impulse responses.

Go Capture
Although a formalised series of ISO regulations exists for the measurement of impulse responses, it is most likely that recording engineers will employ an active studio monitor to reproduce the test signal, with omnidirectional condenser microphones situated at a sufficient distance to capture the acoustic response. A spaced (A-B) pair of omnis performs strongly in this application to achieve a sense of immersion and stereo-field spatialisation. If omnidirectional microphones are not available, a coincident pair of cardioids in X-Y or ORTF formation will suffice as an alternative, but might need to be placed further away to capture enough ambience. A laptop computer and a bus-powered audio interface providing phantom power are also essential parts of your armoury, minimising the requirement for power sources at opposite ends of a large space.

Experimentation, as ever, is the key to unlocking interesting reverb effects. You may consider capturing a number of responses in each acoustic setting with varying distances between microphone and speaker, making changes to their relative heights and tilting the speaker to emphasise specific reflections and resonances. Environments such as reflective stairwells provide interesting, resonant reverb effects for instruments such as snare and guitar, and the process represents a modern-day equivalent to genuine chamber reverb without resorting to any digital modelling. For further ideas on technique, refer back to the Creative Convolution article from September 2010, available online at www.soundonsound. com/sos/sep10/articles/convolution.htm. Maintaining a healthy signal-to-noise ratio throughout the process is crucial. The excitation signal should be played at a strong level through the speaker — use ear protection! — with care taken to eradicate rattles that might occur due to the speaker having to reproduce very low frequencies, or a lack of isolation between the speaker and its stand. Lightweight speaker stands may need to be damped by ad-hoc remedies such as draping with (excess) clothing, sandbags at the base and masking tape over the fixings to eradicate undesirable vibrations. Microphone preamps should be set to a reasonably high level of gain, with care taken to avoid the noise associated with maximum levels. Most important, though, is ensuring a quiet environment when recording responses. This often involves being patient and triggering sweeps between spells of external traffic and bird noise and getting those around

you to remain silent for the duration of the sweep, part of which will often not be immediately audible. For these reasons, many elect to go out and collect responses alone and under the relatively silent cover of nightfall...

Sine Off
The final stage of the impulse-response recording process is to deconvolve the sine wave from the recording made in the acoustic space. This clever mathematical process removes the sine wave from the impulse response and then ensures that all the reflections are compressed into an appropriate time frame, giving an accurate representation of the reverb time and tonal response of the space. Thankfully, the ReaVerb plug-in takes care of this involved number crunching on your behalf: • Call up the ReaVerb plug-in previously inserted on the ‘Utility’ track. • Select File in the Impulse Generation section. • Click the Deconvolve button and select the impulse response recording to be deconvolved. You will be prompted to specify a path and file-name for the finished impulse response: a good place would be in a new folder in your library of third-party impulse responses. After a few seconds, the deconvolved impulse response will be created and ready to use. The finished impulse-response file can be freely loaded into the ReaVerb plug-in to yield original and interesting reverb effects in productions, with further creativity afforded by the range of impulse-response manipulation processes explored in these articles.

Here we’re selecting the recorded impulse response to be deconvolved. Following this step, the user is asked to specify a path and file name for the finished impulse response.
w w w . s o u n d o n s o u n d . c o m / January 2012



Petula Clark ‘Downtown’
Photo: British Pathé

The single ‘Downtown’ gave Petula Clark a worldwide hit and rejuvenated her career. Presiding over the session was engineer Ray Prickett, who tells us how it happened...

Engineer Ray Prickett in the control room of Pye’s Studio 1. This and the following photos are taken from a 1963 Pathé newsreel which featured the Searchers recording their hit single ‘Sugar & Spice’. While the session was artificially recreated for the film, the pictures show a fascinating view of Pye Studios as it was at the time ‘Downtown’ was recorded in the mid-1960s.


owadays, orchestral pop hits are largely a thing of the past. It is no longer a common occurrence for contemporary singers to be tracked live in the studio alongside string, horn, and woodwind sections, as well as electric guitars, keyboards and drums. But it was the standard way of working back in October 1964, when Petula Clark recorded ‘Downtown’ at the Pye Studios on London’s Marble Arch. Released the following month, it became a number two UK hit that December and a US chart-topper in January 1965. This celebration of curing

life’s problems by revelling in the bright lights, neon signs and “music of the traffic in the city” was not only the perfect encapsulation of pre-psychedelic swinging ’60s optimism, but also a classic example of the brilliantly arranged, instantly infectious three-minute single that melded mod sensibilities with showbiz polish. And to think, the lyrics penned by its English composer/producer were not even inspired by the attractions of his nation’s suddenly in-vogue capital. “‘Downtown’ was written on the occasion of my first visit to New York,” Tony Hatch would later recall. “I was staying at a hotel on Central Park and I wandered down to Broadway and to Times Square and, naively, I thought I was

downtown — forgetting that, in New York especially, downtown is a lot further downtown, getting on towards Battery Park. I loved the whole atmosphere there and the song came to me very, very quickly.”

‘We Need A Hit...’
A child star of film, TV, radio and record, known in the 1940s as ‘Britain’s Shirley Temple’, Petula Clark had been signed to Pye since 1955 and enjoyed her first UK number one in 1961 with ‘Sailor’. Although Alan A Freeman had produced that track, as he had all of Clark’s recordings since 1949, his assistant on the ‘Sailor’ session was Tony Hatch, whose first song recorded by her was the 1963


January 2012 / w w w . s o u n d o n s o u n d . c o m

flop ‘Valentino’. Thereafter, Hatch took over as her producer and capitalised on Petula’s popularity in Europe via some French-language releases. At home, it was a different story. A string of nondescript singles meant that, by 1964, she was in dire need of an English-language hit, so Hatch visited her home in Paris to play her three or four songs he had acquired from music publishers on a trip to New York. Pet wasn’t impressed... until she heard a few bars of an incomplete soul number that Hatch intended offering to the Drifters. “We already knew that we had to make a record,” Hatch recalled in a 2009 interview with Gary James. “I had a studio booked with an orchestra, ready to do a new recording session with her. And she said, ‘Aren’t you working on anything yourself?’ Reluctantly, I played her the idea of ‘Downtown’, because I’m always reluctant to play half-finished songs. She immediately saw tremendous potential in it. She was the one who said, ‘Get that finished. Get a good lyric in it. Get a great arrangement and I think we’ll at least have a song we’re proud to record even if it isn’t a hit.’”

The Session
The ‘Downtown’ session took place about two weeks later, on 16th October, 1964. Thirty minutes before it commenced, Tony

Hatch was still fiddling with the lyrics in the studio’s lavatory. This was because when he took charge of a production — as he had already done with the Searchers (for whom he’d penned the hit release ‘Sugar & Spice’ under the pseudonym Fred Nightingale) — he insisted on everyone recording together at the same time, be they members of a four-piece group or, as in this case, a large ensemble. The musicians assembled included eight violinists, two viola players and two cellists, four trumpeters and four trombonists, five woodwind players with flutes and oboes, percussionists, a bass player and a pianist. Also present was famed session drummer Ronnie Verrell (not Bobby Graham, as has been erroneously reported elsewhere), female vocal trio the Breakaways, whose backing-singer credits would soon range from Dusty Springfield to the Jimi Hendrix Experience, and session guitarists Big Jim Sullivan, Vic Flick and Jimmy Page. “Quite often on the Hatch sessions, not only did we have Jim Sullivan and Jimmy Page, but also John McLaughlin sitting in as part of the rhythm section,” says engineer Ray Prickett, who recorded ‘Downtown’ in Pye’s Studio 1. “Can you imagine being around that collection of talent? For ‘Downtown’, looking down on the live area from the upstairs control room, the drummer was in the
Petula Clark with a gold disc for ‘Downtown’, 1965.
Photo: R McPhedran/Express/Getty Images

Artist: Petula Clark Track: ‘Downtown’ Label: Pye (UK)/Warner Bros (US) Released: 1964 Producer: Tony Hatch Engineer: Ray Prickett Studio: Pye
far right corner, the bass player was to his left (again, from our perspective), the guitarists were about halfway along the right-side wall, the percussionists were at the near end of that wall, the pianist was in the middle of the studio and Tony’s arranger Bob Leaper was conducting nearby. Opposite the control room, near the facing wall, was where Petula and the vocal group were set up, with a few small screens between and around them to provide a little bit of separation. The strings were to the left of them (as we were looking at the room) facing towards the conductor, and further left was the brass section, while the woodwind were in front of the string section. “When you listened to any of the mics, there wasn’t full 100 percent separation. Not by a long way, because that wasn’t what we were aiming for. The way I saw it, and Tony agreed with this, was that the sound wasn’t as good when we recorded different sections separately. When the whole orchestra plays together, something happens — all of the air is being moved by those instruments and that’s what gives you a big, ambient sound. This is why there was minimal screening even around the vocalists; maximum separation would have defeated the object of having all those people playing in that room. “Since their playing was well controlled, Pet Clark was somehow able to hear herself singing. The vocalists didn’t use headphones as often in those

w w w . s o u n d o n s o u n d . c o m / January 2012



A mono Ampex tape recorder from Pye’s studio 1. The control panel of Pye’s three-track Ampex (not the four-track used on ‘Downtown’).

days as they would later on, and there also wasn’t much [loudspeaker] foldback. Until we replaced it with a Neve in 1965, we had a four-track Neumann desk in Studio 1. Later desks would actually have more controls for foldback than for mixing, and the trouble was, when you had the ability to work more than two or three different mixes of foldback, it became a bit of a pain. At that point, everybody wanted to hear their own little mix, and this, in my opinion, detracted from the spontaneity of the overall sound.”

Career Engineer
Ray Prickett had no such concern when, as an 18-year-old in 1952, he gained his first studio experience. This was at a facility named Gui de Buive, where

name). This, in turn, paved the way for the fully fledged engineering work that Prickett did after moving to Pye in 1963, recording anyone from Lonnie Donegan and Kenny Ball to the Searchers and, of course, Petula Clark. “Whenever Tony Hatch was with me in the control room, we had such a good understanding that, by the time he’d ask me to add something, I had already done it,” remarks Prickett, concurring with what Hatch himself told an interviewer in 2005: “Bob Leaper was my musical associate. I would do a run through of each item with the orchestra myself, check notes, et cetera, get the right feel, then Bob would take over when I went to the control room. I worked with balance engineer Ray Prickett so many times... he knew exactly what I wanted. I’ve always

“Working four-track, we’d always do a straight stereo mix of the entire orchestra and all of the other musicians, there and then, completely live.”
he worked with tape and direct-to-disk recording before landing a job three years later as a cutting engineer at IBC. In 1958, IBC became the first UK facility to be equipped with a stereo lathe, and it was there that Prickett also tracked easy listening, orchestral-pop sessions, including several produced by the aforementioned Alan Freeman (not to be confused with the disc jockey of the same had a great rapport with musicians and studio personnel, so the job becomes an enjoyable team effort.”

Recording The Pye Way
Located inside ATV House, Pye had two studios: the 40 x 30 x 18-foot room where ‘Downtown’ was recorded and the 20 x 20 x 18-foot Studio 2. The former, in addition to its Neumann console, housed an

Ampex four-track tape machine, Tannoy dual-concentric speakers inside Lockwood cabinets, an echo chamber, a couple of rooms with EMT 140 echo plates, and a good selection of mics that included Neumann U47s, M49s and an SM2 (and, later, U67s, U87s and KM84s), AKG C12s and Sennheiser MD421s. “I’d use three mics to record the drums,” Prickett explains. “Usually a 421 on the bass drum and then two 47s, 67s or 87s as overheads — whatever was available at that time. We would train a lot of young engineers at Pye, and one of them once came to me and said, ‘Come and listen to this drum sound that I’ve got!’ This was when we’d already upgraded to 16- or 24-track. He said, ‘Listen to the snare. Doesn’t it sound great?’ and I said, ‘Yes, great, great,’ and then he asked me to listen to the bass drum. Eventually, I said, ‘Let’s listen to the whole lot together... Have you listened to the whole lot together?’ ‘Oh no,’ he said, ‘I haven’t done that yet,’ so I said, ‘Well, then do it.’ When he did, his face fell — he couldn’t believe the top-less sound that he’d got. ‘Why’s that?’ he asked me, to which I said, ‘You’ve got too many mics too close together. You got so much phase shift there, you’re losing all your top end. You’ve made it over-complicated.’ “That was basically my theory: if you’ve got too many mics on a drum kit, you get phase shift. Because a kit is a very tight little unit, you can’t afford to put separate mics on the cymbals, hi-hat and so forth. Whatever happens, you’re going to get spill, and the phase shift caused by the various spacings of the mics will detract from the overall sound. You should get


January 2012 / w w w . s o u n d o n s o u n d . c o m

Photos: British Pathé

a nice, clean, top-end sound on the kit, and that requires just two mics overhead and one on the bass drum.” Without a doubt, this worked on ‘Downtown’. Ronnie Verrell’s drumming, deft yet with a light touch, features prominently — courtesy of a solid bass drum, heavy snare backbeat on the choruses and crisp ride cymbal — while fitting in with the rest of the instruments. A case of perfect blending rather than perfect separation, this was balancing done the organic, old-fashioned way, and the results speak for themselves. “For the bass guitar, we didn’t DI much in those days, so we would have miked the amp,” Prickett continues. “We of course did the same with the electric guitars, there would have been a mic on the acoustic, and for the piano I used the stereo SM2 — the lid would be partly open with the mic inside, and then we’d put a cover over the top just to keep some separation. “For the strings, I’d use two mics as a stereo pair on the eight violins, and then one each on the pair of cellos and pair of violas. Then there would be no more than three mics on the woodwind section, two on the trumpets, another two on the trombones, and either U47s or U67s on the vocal and backing vocals. “In all, we probably did four takes — sometimes we got away with less than

that — and we also did some editing, maybe editing the brass middle-eight out of one take and inserting it into another take that we preferred because of the vocal. Working four-track, we’d always do a straight stereo mix of the entire orchestra and all of the other musicians, there and then, completely live. What we got on the session is what you hear on the record, and the other two tracks had Pet Clark and the backing singers.” Whereas most British studios of that era adhered to a strict timetable — with sessions running from 10:00am to 1:00pm, 2:00pm to 5:00pm and 7:00pm to 10:00pm — Pye’s engineers were used to often working beyond the regular recording sessions into the wee small hours so that they could finish a mix and meet the deadline for delivering the finished product to the pressing plant a few hours later. “I got blacklisted at a very early stage in the business,” Ray Prickett laughs when asked whether such work practices contravened union rules. “Pye was actually a non-union studio and it was meant to be that way.” For the mix of ‘Downtown’, since the stereo musical backing had already been mixed live, Prickett’s main task was to just add the lead and backing vocals at the desired levels. “We’d pan the mix as we went

along,” he says, “usually with the rhythm mainly in the centre, the strings spread left and right, the trombones and trumpets split from the point of view of the stereo mix, and the woodwind panned across the middle. So, it was a nice big spread of sound.”

Evidently, the record-buying public agreed. Following its release, ‘Downtown’ not only ended a two-year chart absence in the UK for Petula Clark, but during its three-week stay at number two in December 1964, the single was only kept from the top spot by the Beatles’ ‘I Feel Fine’. Certified silver for domestic sales of over half a million, it also hit the top five in countries as far apart as Ireland and India, while going all the way to number one in the English-speaking territories of Australia, New Zealand, South Africa and Rhodesia. Still, its greatest triumph was reserved for an America that was in the full throes of the so-called ‘British Invasion’. Given that Tony Hatch had first thought about writing ‘Downtown’ while standing in New York’s Times Square (which, he didn’t realise, is actually part of the city’s midtown), he was concerned that Warner Bros A&R exec Joe Smith would think Petula Clark’s unmistakeably English accent sounded ridiculous when she sang
Photos: British Pathé

Disc Cutting
Although Ray Prickett didn’t work as a disc cutter at Pye, his previous experience in this field nevertheless proved beneficial to the records that he engineered. “We had two cutting rooms with Scully lathes,” he says, “and I remember one of the engineers there asking me, ‘Why is it we never have to do anything to your stuff other than just cut it?’ I said, ‘Well, I used to be a cutting engineer.’ The guys there reckoned that all recording engineers ought to learn a little about cutting. “As we all know, in those days the LP was really a compromise in terms of capturing what was on the tape — particularly the stereo LP. There would be two drivers running at a 45-degree angle to the surface, and you could actually cut a groove shaped in such a way that it would be almost impossible to separate when they were pressing — it was like an under-cut. That’s why, if you listen to most records of that period, a heavy bass would usually be placed in the middle. Otherwise, if you put the bass on one side, it made it very difficult for cutting. These were the sorts of things one learned back then...”

Cutting Engineer Derek Moore in one of Pye’s two cutting rooms.

w w w . s o u n d o n s o u n d . c o m / January 2012



Ray Prickett looking down from the control room into the live room of Pye’s Studio 1.

Americanisms such as the song’s title, “sidewalk” instead of “pavement”, and “movie shows” instead of “cinemas”. He needn’t have worried. As soon as Smith heard the record in Hatch’s London office, he loved this “observation from outside of America” that, to his mind, was “just beautiful and just perfect.” In fact, such was Smith’s enthusiasm, it even inspired Pye executives to give the single a stronger sales push in the UK, where it would scoop an Ivor Novello Award as 1964’s ‘Outstanding Song Of The Year’. Released by Warner Bros that December, ‘Downtown’ repaid Joe Smith’s faith by making its way into the top 10 of the Billboard Hot 100 within five weeks. Then, on 25th January, 1965, it commenced a fortnight’s stay at number one, making Petula Clark Britain’s first female chart-topper in the US during the rock era (just over 12 years after Vera Lynn had achieved a similar feat). It also made her the first British girl singer to also

earn a gold record in America for sales of one million copies, and the recipient of a Grammy for 1965’s ‘Best Rock & Roll Song’. “She was a total professional and very easy to work with,” says Ray Prickett, who remained with Pye until 1978. His many other credits including all 14 of Petula Clark’s subsequent consecutive US top 40 hits — ‘Colour My World’, tracked at LA’s Western Recorders was credited by Phil Spector’s drummer of choice, Hal Blaine, as featuring his best drum sound there. Additionally, Prickett recorded artists such as Donovan, Dionne Warwick, Trini Lopez, Sounds Orchestral, Jay & the Americans, and Françoise Hardy before going freelance and focusing on location work, mainly with military bands. He ended his 56-year career by retiring in 2008. “I grew tired of working in the studio when there was no longer any longevity to what was being recorded and multitracking was making everything so time-consuming,” Prickett explains. “This was completely different to how

things were up until the early ’70s, when we worked quickly and compiled a great catalogue of material that would keep selling well into the future.” A case in point: the 1964 recording of ‘Downtown’ which not only surfaces regularly on radio, TV shows and film soundtracks, but has also been remixed and re-released three times; in 1988, 1999 and 2003. What’s more, in addition to the multitude of cover versions by anyone from Frank Sinatra to the B-52s, Petula Clark herself has re-recorded the song several times: in 1976 (with a then-trendy disco beat), 1984, 1988, 1996 and, in December 2011, as a duet with Irish rockers the Saw Doctors. Not bad for a chanteuse who has just turned 79. “We had no idea that we were recording a monster,” she said when discussing the original, classic version in a 1999 radio interview with Kool FM in Phoenix, Arizona. “You never do. You just go in and do a song you like and it’s the public who decides that it’s going to be a hit...”


January 2012 / w w w . s o u n d o n s o u n d . c o m

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Bio Engineering
Mandy Parnell: Mastering Björk’s Biophilia
The mastering engineer’s role is changing as artists explore new formats. And as Björk and Mandy Parnell discovered, what works for the iPad might not work on CD...

On its eventual release as an album, the music that makes up Biophilia would prove very different from its initial iOS incarnation.


astering is supposed to be the simple part of making records, right? You bring your stereo mixes along to someone with a fresh pair of ears, a nice monitoring system and a bunch of gold discs on the wall. He or she checks that there’s nothing wrong with your files, runs them through some specialist equipment you can’t afford in order to make them sound ‘finished’ and then generates production masters. Simple. Simple, that is, unless you’re Björk, and your latest album is Biophilia, the ‘world’s first app album’, released in conjunction with Apple. All of Biophilia’s 10 tracks are being issued as apps for iOS devices in collaboration with Scott Snibbe, an interactive artist who combines his visuals with images from National Geographic and narration by David Attenborough. They explore a variety of music- and science-based themes, forming a multimedia


collection “encompassing music, apps, Internet, installations and live shows”. Björk debuted songs from the album during a series of performances at the Manchester International Festival before it materialised in conventional music formats on CD, vinyl and MP3. By this time, a few months had elapsed since the music had originally been ‘finished’, and having performed the songs live, Björk decided there was more she could bring to the project. She asked long-term collaborator Leila Arab to help with some sound sculpting, and Leila suggested she should consult mastering engineer Mandy Parnell. “The idea of the original mixes for the apps,” explains Mandy, “is that when the song are played through the iPad speakers, you hear the mid and top end of the mixes, and when you plug your headphones in, this whole other dimension with all the sub-bass comes through. The app album was deliberately mixed with this in mind, so when you listened to the original master,

it was very prominent in the sub-bass area and the top end, but slightly lacking in the upper bass and lower-mid area. When Björk received her audio CD back from mastering without the visuals and the experience of hearing through headphones, she was not so convinced of the mastering for the CD. It really works amazingly for the apps, but she felt the album needed to be fuller. At the point where she had received the references for the CD, Björk had already performed a month’s residency in Manchester playing 10 incredible shows, so she had a completely different emotional and sonic connection with the album. The music had morphed into a different experience in the time that had passed, so after finishing the shows in Manchester she decided to go back into the studio with Leila Arab and rework the album.”

Mandy Parnell’s involvement in the project escalated rapidly. “It started off with me


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receiving a call from One Little Indian [Björk’s label] saying ‘She’s not happy with her vinyl, can you re-cut it?’ Then ‘She’s not happy with the album, can you re-master the album?’ Then ‘Actually… we’ve heard that you sometimes get involved with taking a mobile rig to another studio and mastering there, could you do this for us? Can you fly out to Iceland?’” This Mandy promptly did, to be greeted with yet another surprise. Curver Thoroddsen, one of Björk’s assistant engineers, opened up the files, and they were all mix sessions. There were no stereo bounces of the tracks. “I didn’t know that I was going to be presented with stems. I thought I was going to be working on the stereo masters!” Mandy was not aware of the changes Björk had made to the mixes at this point, and in case this wasn’t stressful enough, she was working against a looming and immovable deadline, with an unfamiliar setup, in a studio she’d never seen or heard before. “I went into Addi 800’s studio Ö&Ö. He’s worked with artists like Björk and Sigur Ros numerous times before, and he’s one of the top mixing guys in Iceland. We went into his studio to get set up, before I would meet Björk and go through the album with her. “I had taken my rig with me, including a Prism Sound Orpheus — I used this and SADiE as my mastering system.

w w w . s o u n d o n s o u n d . c o m / January 2012



Mandy Parnell’s own Black Saloon mastering suite in London is based around an EMI mastering desk and Tannoy dual-concentric monitors, along with a Neumann cutting lathe (not pictured).

Prism and SADiE is my normal system and it’s great for me now that SADiE is native, it makes my mobile rig more versatile as I can run it from my laptop. My signal flow was from Pro Tools into a Thermionic Culture Phoenix compressor, a rare Inward Connections EQ, a Crane Song STC8, then my Orpheus; I used the limiter of the TC System 6000. Addi’s studio has the Barefoot MM27 monitors with a Genelec sub and a Dangerous monitor controller. I had taken my Benchmark D-A as well, and I said I’d rather run the Barefoot MM27s from the Benchmark because I know the sound of it better than the Dangerous monitor controller. We set up the subwoofer with the Dangerous monitor controller to give us the option of punching it in and out during the mastering to check the really low subs. We realigned Addi’s Barefoots listening to lots of references while tweaking, so that they felt closer to my listening environment in my studio. The Barefoots were incredible to work with.” “I sat with Björk for many hours playing back the mixes, talking about the album, the concept, looking at the apps, hearing about the incredible instruments that had been built especially for the project. I made notes and markers for each song. We listened to each song, played with the apps and talked at length while Björk was giving me an overview of the whole concept of the album. Björk works on a very emotional level: she talks about what the intention of
Biophilia was conceived and developed as a suite of apps for Apple iOS devices, in conjunction with Scott Snibbe.

each song is, and what feelings it is going to conjure up in the listener. I said ‘Let me work on one track tonight, and I’ll send you a couple of different mastered versions to pick from. Give me some feedback and we’ll get going on the rest of the album.’ I decided to work on ‘Crystalline’ and spent a few hours trying different things, exploring the different textures and depths I could get from the mix. “Björk came back the next day with the version she had chosen, and then it was all systems go. All of the mixes had elements changed; for instance, on one song Björk had decided to change some of the choir arrangements and use some different takes of the recording. Björk and Curver were

editing that in another room while I was mastering in Addi’s room. She had also decided to use the live version of ‘Solstice’ rather than the recorded one, as she preferred the way the song had grown out from the live performances.”

Stemming The Flow
This would have been enough of a challenge had Mandy been asked to work with conventional stereo masters, but instead she was confronted with large Pro Tools sessions containing stems — and not just a few of them. “Most of the time when I am sent stems to master from, I am given the instrumental, main vocals and backing vocals. I would adjust the balance of the


January 2012 / w w w . s o u n d o n s o u n d . c o m

vocals with the mix during mastering. Working from stems is a whole other way of thinking for me, it opens doors to endless possibilities that I would not have in a stereo mix. Björk’s project was presented to me in a Pro Tools session with the mix ready to roll, or so I thought — as I started to go through each track I realised I could do fine adjustments to create the space in the mix before going into the mastering chain. ‘I would be given the choir vocals for instance, with many different edit sections that needed re-comping. I would have the drums, maybe on more than one stereo group, there might be a few different groups of Björk’s vocals, as well as other beats, various instruments on other groups and sound effects. Some of them you’d open and there’d be a page full of stereo stems, groups and tracks.” Despite being unfamiliar with Pro Tools as a mastering system, Mandy, with the help of Björk’s assistants, ended up using its automation and software EQ quite extensively to balance the mixes prior to sending the signal through her analogue mastering chain. “I had the chance to rebalance the mixes and process the separate tracks with EQ and compression, as well as the stereo mix bus. I really enjoyed working with some of the spatial effects on the separate stems inside Pro Tools, working to open up the choir in the stereo field, and being able to EQ them separately. This meant that the mastering side was just compression and a tiny bit of EQ: rather than trying to find that space in a stereo mix, I could create it before the mastering. It was great to have that flexibility. I have been working more like this recently — for instance, with Leila Arab’s album she brought her rig to my studio and did the final tweaks to the mixes there to get the sound and space she was looking for while monitoring through my mastering chain. “With something so complex, that has got so many layers of sounds, having access to the stems as a mastering engineer can allow you to get more space out of the mix. I can get it sounding wider and I can achieve more separation this way. It’s really about balance and pulling the vocal out. The amounts we’re talking about would not really be a lot in a mixing situation. The most might be 0.5 of a dB, while a lot of the time it could be 0.2 or 0.3. On one track, Curver, who sat behind me, said ‘I can’t believe that’s all you’ve done, and it’s made such a big difference.’ It’s just about approaching the mixes with a mastering ear. I’m not a mix engineer, I’m a mastering

Loudness Without Limiting
Mandy Parnell lays great stress on the use of techniques other than hard limiting or clipping to get greater perceived level from her sources. When asked to deliver a loud master, she prefers to use more subtle means. “It can sometimes be more about gain structure than just putting a limiter at the end of the path. Because I’m working in analogue, my signal will come out of SADiE into the [Prism Sound] ADA8, and if something’s given to me that is already very loud I am able to bring the output slightly down in SADiE, just so it’s not pushing into the converter too hard. Then I can change the gain inside the ADA8 if I want to, so I might do a mixture of the two. I then come out of the converter into the compressors, where I can also change the gain coming in or out. Additionally, I’ve got various gain options on my desk, so a lot of the process is gain staging along the way. It’s the same with compression: I might do a little bit on one compressor and a little bit on another compressor — if I need to do a lot of compression it won’t just be on one unit, I might layer it. You can also do a phenomenal amount with mid-range EQ to get level. “I was very fortunate to be trained by experienced mastering engineers who cut vinyl records. Back then it was about getting a good-sounding record. They had issues with levels, level was relative to time: if you had a long side how do you get it to sound loud when you can’t use gain? So you learn how to use limiters, compressors and explore EQ with mid-range to make something perceivably louder than it might actually be. “Luckily for a lot of us mastering engineers, now we are not only being asked for the loudest mastering — it is also about having dynamics for a lot of artists, producers and engineers. I think we are coming into a new era of mastering where the CD and vinyl are not the only formats we need to think about as mastering engineers. For example, record companies are asking for 88.2k, 96k, 192k and so on, 24-bit WAV/AIFF files for higher-resolution downloads. We need to encourage our recording studios, engineers, producers and artists to record and mix in these formats. “I was very honoured to be invited by Björk to work with her on this project. I feel it has opened a whole world of new possibilities for artists to present their music on different platforms and also change the way we will interact with music in the future.”

engineer, and everything I am looking for is about separation in the sound and the balance of the mix.”

Territorial Pressings
After much effort and fun, Mandy managed to deliver an album that Björk was completely happy with, before the deadline — and one that was radically different from its original incarnation in many places. Mandy was then able to return to her Black Saloon Studios in London, her work done… or so you’d think. In fact, the orchestration of delivering masters for all the different formats, record labels and territories involved kept her busy for another few days. “We spent 27 hours running production parts for all the different territories. I don’t think people realise how much work is involved for an artist of this calibre to deliver an album, let alone what Björk has done with this project. Not only is there the commercial CD, there are also deluxe versions for box sets and vinyl, as well as bonus tracks for different territories.” Alas, one thing that the mastering engineer has no control over at this moment is the conversion of music into lossy formats for online distribution. “It’s horrible,” says Mandy. “What happens a lot of the time is that there’s someone in a back room who does the conversions after it leaves the

mastering suite, and every platform wants a different codec, be it iTunes, Amazon or CD Baby. The record companies are being charged a nominal fee for this transfer by companies who are not employing trained audio engineers. There is no quality control for what the different codecs are doing to the audio; the record companies would not pay our fees as mastering engineers to check this, unfortunately. Years ago when we would cut vinyl records, the mastering engineer who cut the vinyl would always get a test pressing back to check the quality and the audio. We would be asked to approve the quality before it went into production. Now, once it leaves the mastering studio as files, DDP or CD, we do not know how it is transferred. We, as mastering engineers around the world, have to take back control of the final QC.” It’s a situation that neither Mandy Parnell nor her artists are happy with, but it doesn’t mean that quality mixing or mastering have become irrelevant. “With Leila [Arab]’s album, after we mastered everything, but before we finally signed off, we actually transferred everything onto a phone to check it. We were listening to see if the integrity and intention were still there, checking for if you’re losing the important elements of the song. The well-produced, well-mixed, well-mastered tracks will still cut through on a phone speaker.”

w w w . s o u n d o n s o u n d . c o m / January 2012



The A Team
Jake Gosling:
Producing Ed Sheeran’s +
Ed Sheeran’s phenomenal success depended on hard work, a few lucky breaks, and the talents of long-term co-writer and producer Jake Gosling.

ew artists in recent times have experienced the kind of vertigo-inducing rise that Ed Sheeran has enjoyed. When, in September 2011, his debut album, the symbolically titled +, entered the


album chart at number one, enjoying first week sales of over 100,000 before quickly going platinum, it seemed as if the 20-year-old singer-songwriter had appeared from nowhere. But Sheeran’s back story is actually one of an impressive level of dedication and dogged determination. As a teenager,

inspired by Irish singer Damien Rice, he picked up a guitar and began writing songs while still at school in Framlingham, in Suffolk. As soon as he left, he hit the road, appearing at open mic nights around the country and hawking CDs out of his rucksack. In 2009 alone, as the legend now goes, he performed a staggering 312 gigs. En route, he showed real bravery in stepping on a plane to Los Angeles at the age of 19, to perform solo in low-life bars, before by chance encountering his sometime mentor, actor/musician Jamie Foxx, who took him in and recorded him at his home studio, with Sheeran kipping on the Hollywood star’s sofa. However, none of the recordings Sheeran made with Foxx were to make

January 2012 / w w w . s o u n d o n s o u n d . c o m

Ed Sheeran’s success might have seemed an overnight phenomenon, but was built on much hard work.

Sticky Studios is based around an old DDA console, and Jake Gosling’s preferred monitors are JBLs of even older vintage.

their way onto +. Instead, the album’s sessions were rooted in Sheeran’s relatively long-term relationship with producer and co-writer Jake Gosling, who runs his own Sticky Studios from a converted barn situated in an apple orchard in the small Surrey village of Windlesham. “It’s a great location,” Gosling enthuses. “It has a real country vibe. People love coming here, ‘cause it’s cut off and you’re not interrupted by anything, so it’s great for writing and all the rest of it.” Previously, Gosling was renowned for his work in UK urban music, producing rappers Wiley, Kano and Wretch 32, while remixing tracks for the likes of Lady Gaga and Timbaland. He first met Sheeran four years ago, when their shared publisher dispatched the prodigious 16-year-old to Sticky Studios for a writing session. “This little ginger kid turned up and he was really confident for his age,” the producer remembers. “He’d just moved

to London and he was living above a pub. We sat down, and we were talking about him moving to London and that became ‘The City’, which was the first thing we wrote together. I felt his lyrics were just insanely good.” One characteristic feature of Sheeran’s music is his blurring of acoustic balladry and hip-hop, which finds him switching between sensitive singer-songwriter and adept rapper. Gosling says that this cross-pollination of styles was something he encouraged from the moment the pair first started working together. “He loves urban music… he loved Wiley and all the rappers I’d worked with,” he says. “But I also love folk, so we connected on a musical level straight away. It was a perfect fusion of the two together, really. We were trying to create something new.”

Mate’s Rates
Before + came out, Ed Sheeran released a series of EPs between 2009 and 2011, all recorded with Jake Gosling at Sticky Studios. Soon after starting to work together, the pair settled on a loose arrangement where the singer would give the studio owner £1000 to cover an EP’s recording costs, which he would quickly recoup, before turning a modest profit selling them at gigs and through iTunes. “The only way he lived was through selling his EPs,” Gosling points out. “Because I believed in what he was doing, I was like, ‘Well, look, if you can cover my cost on the studio, I won’t recharge you, ‘cause I love what you’re doing.’ He’d get a couple of thousand printed up and just sell them at gigs everywhere he went. If he

made 30 quid or 40 quid, he was over the moon about it.” Style-wise, Sheeran’s formative EPs veered between full band arrangements, stripped-down intimacy and, in No.5 Collaborations Project, a full-strength blending of his songs with grime and dubstep influences and rap cameos from Devlin and Wiley. This followed Sheeran’s first appearance on SB.TV, an online urban music video channel where he quickly began to amass millions of views. “They asked him to do a few tracks down at the studio here,” says Gosling, “and suddenly the online thing went mad. All these urban kids that were loving Wiley were suddenly into Ed, because he’s rapping and he’s singing. Lyrically, it was really connecting.” Sheeran soon attracted the attention of Atlantic Records, and sessions for what would become + began at Sticky Studios in January 2011, with the singer determined not to move to a bigger or posher recording facility. In fact, as a highly amused Gosling points out, the majority of the album was recorded on an old PC running Cubase SX3. “I mixed ‘The A Team’ on that for the album and obviously we’ve gone platinum now,” he laughs. “Funnily enough, I got a message from Abbey Road through the record label when we were mastering, saying, ‘Who mixed this record? This is amazing!’ “I chuckled to myself, because there can be a real snobbery with musical equipment and what you have and what you don’t have. But I think half the battle is how you use it. I do it day in day out and really, I think, if you’ve got a good set of ears and you know what you want to hear and you

Photos: Carey Sheffield

w w w . s o u n d o n s o u n d . c o m / January 2012



really nicely and create your own loops out of loops. It’s really versatile.”

Nothing Fancy
Sticky Studios is built around a DDA AMR 24 console that previously belonged to Hans Zimmer. Gosling managed to get his hands on it when Chelsea recording facility Snake Ranch went out of business in 2004. His main monitors are a pair of early ’70s JBL 4311Bs that belonged to John Lennon during his Imagine period. “A mate of my dad’s used to work at his studio,” he explains. “So I recorded Ed’s album through John Lennon’s speakers, which is funny. I love mixing on them. I saw another pair on eBay for £100, which was insane. I’ve got NS10s as well, which I love, so I monitor between those two. And I’ve just got some Focal Twin 6s, which I’ve always loved but could never afford.” Mic-wise, Gosling tends to favour the SE Electronics range. “I did a lot of the guitars using a pair of SE 4400as. The main vocals pretty much were all done on the SE Z3300a, but I’ve just got a new Telefunken U47, which is lovely.” On +, along with his production and co-writing duties, Jake Gosling also served as the main keyboard player, utilising his array of vintage synths, which includes a Roland Jupiter 8, a Korg MS2000 and a Roland SH5. In addition, he has a Hammond, Fender Rhodes and upright Yamaha piano, miked with a pair of SE1as. For strings, he relies mostly on the EastWest plug-in range. “With ‘The A Team’, it was all EastWest violins,” he says. “But when we did the final track on the album, ‘Give Me Love’, we used the EastWest samples mixed with some live strings we recorded as well.” Gosling uses very few outboard effects apart from a Line 6 Pod XT for guitar sounds. “I use it for keyboards as well,” he points out. “But if I use the Rhodes, I’ll mush it up through a VST plug-in amp.” Otherwise, the producer insists, he stays in the box, effects-wise. “I love the Waves stuff. They’ve got such a massive
Where many producers favour exotic vintage microphones, Jake Gosling uses the SE Electronics range almost exclusively.

selection of reverbs and compressors and they’ve got the SSL plug-ins, which are fantastic… great on vocals, great on pretty much anything, really. I mix and match different reverbs with different instruments. I try different ones, high and low reverbs rather than one generic one.” Gosling is also a fan of cutting frequencies rather than boosting them, in attempting to get a sound to sit in a work-in-progress mix. “As we go along, I’ll make sure that everything’s got its place,” he says. “With guitars, I’ll literally EQ the bottom end out of it completely so you don’t get the full range, unless it’s a guitar on its own holding the track together.”

Writing & Recording
At the beginning of each session, if Sheeran and Gosling are starting work on a track from scratch, the latter will encourage the former to chat about his everyday life, hoping to mine his personal experiences and emotions for a song idea. “As a producer and songwriter, it’s my job to be almost like a psychiatrist,” he laughs. “Tell me your problems, Ed, what’s going on with your life? The beginning of that process will come from stories and things that are going on, mainly in his life. “Then we probably attack it in quite an old-fashioned way. We’re not really thinking about how we can sit down and write a hit song. We don’t necessarily go, ‘How long is it going to take for us to get to the chorus?’

For the most part, Jake Gosling uses only software effects and processors, but relies on his Line 6 Pod for many guitar sounds.

can bring out frequencies and work on stuff, you can get amazing results. I mean, don’t get me wrong, I’m having to probably work a bit harder than other people to get the right sort of sounds. But Ed’s got a very warm voice anyway, and I think with the acoustic guitars and blending in samples and all these other things, it created its own sound.” Even now, after a minor dalliance with Pro Tools, Gosling has stuck with Cubase, updating to version 5. “For me, it’s the editing,” he says. “I know it so well. I don’t really tend to use a lot of Auto-Tune, but I use Melodyne in it quite a bit, which can be handy for backing vocals or if you’re working with a singer who’s a bit dodgy here and there. It’s quite fluid and with the pitch-shifting, it’s really helpful.” Elsewhere, Gosling tends to turn to Ableton Live when it comes to constructing beats. “I’ve got a massive vinyl collection and loads of old sample DATs,” he says, “so over the years I’ve put them onto my hard drive. I’ve gone through records and taken little breaks and samples and built up loads of my own sounds, and I just swing them into Ableton. You can really play with them. You can take a snare and turn it into a synth. You can completely manipulate sounds really easily, like being a painter and just swishing around with your brush. Suddenly you’ve got some sound that started as something else entirely. Or you can time-stretch things


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I mean, I wrote something new with Ed the other day. He was writing this middle eight section and he began at the end and started writing backwards, lyric and melody. I was like, ‘Wow, I’ve never worked with anyone who’s written backwards like this.’ And it made complete sense. It was just incredible. “Usually, we just jam out stuff. Ed will grab a guitar and go, ‘Oh I’ve got a few riffs.’ So we’ll go through some riffs and decide upon ones we like. Then we’ll get a beat going and try to work on a feel thing and we’ll start recording straight away. Recording is part of the songwriting.” One unusual consideration, particularly in their early songwriting days, was how Sheeran would manage to perform the songs live, since it was such a key part of how he got his music across. On stage, he would make heavy use of live overlooping, forming a one-man band with his Boss RC30 Loop Station, creating beats by tapping his acoustic guitar body and human beatboxing, layering bass lines and

In contrast to his dependence on software effects, Jake Gosling owns and prefers to use real vintage keyboards, including his Roland SH5 synth, Hammond organ, MkI Rhodes piano and Yamaha upright.

riffs that he would then perform over, before kicking the pedal on and off at crucial points in the arrangement for dramatic effect. “Because that was his main performance thing, we’d often write in a loop way,” Gosling says. “We’d loop up a beat and a guitar part and build it from there, working through dropping the track out or cutting it back in, like you would use a loop pedal.” Nevertheless, the tracks on + are characteristically sparse, a feature which Gosling insists was intentional. “Yeah,

definitely. I think the power really comes from the words and the sonics that are there. I wanted to make sure that they were right, rather than filling up and overcomplicating the tracks. We wanted it to be quite stripped-back and to give it a lot of space. I think a lot of music tends to be overfilled these days. People shove everything in it. What we were aiming to do was let the songs breathe.

Perhaps I Do Need You After All
Jake Gosling insists that the process of writing and recording the album was relatively painless throughout, with two exceptions. The first complication came with the recording of the hidden track on +, an a cappella rendition of traditional Celtic drinking song ‘The Parting Glass’. “The reason why is because it’s all vocally done,” says the producer. “The backing vocals were all played on a keyboard and each note was sung four times, so you can imagine how long that took. We’d solo one note and Ed would sing that note four times, down and down the keyboard to the bass, as low as he could go basically. It hadn’t sounded right with piano and vocal, so it was like, ‘Well, let’s trying humming it.’” The real bugbear of the album’s sessions, however, was ‘You Need Me, I Don’t Need You’, Sheeran’s beat-driven one-fingered salute to the music business. “We did, like, 20 versions of that,” Gosling admits. “It was pencilled in for a single and there’s always that thing of trying to meet in the middle with the commercial aspect of what music’s about and the artistic aspect. The problem we had was that the first recording that was done was a live SB.TV thing and the energy and atmosphere of that track was what we wanted to capture. But, of course, with the magic of music and the way it goes, you can’t always capture that again. So we really struggled. “We did so many different versions: full-on drummy things and really stripped back stuff. Then we tried to do it how we recorded it originally, with the loop pedal. We even thought of actually using the original loop pedal track, but it was only recorded in stereo. But everyone loved the song and Ed really wanted it to be a single as well, so we eventually had enough of it. It was like, ‘We can’t take this any more, we’ve done 20 versions.’ So the label suggested that we get another guy [Charlie Hugall] in to try and do some drum programming on it and at that point, I went, ‘Yeah, fine.’ That became the final version.” It’s perhaps a touch ironic, of course, that Sheeran suffered record company pressure on a track that is basically telling the music industry where to go… “But I don’t think everyone knows the answers,” Gosling argues. “Even if you’ve got a really good clear idea, at the same time, sometimes it can help to give the record company’s ideas a go. If it doesn’t work, it doesn’t work. But with that song, I think we got to a place where we were all happy with it.”

w w w . s o u n d o n s o u n d . c o m / January 2012



“I mean, there are moments in the album, like ‘Give Me Love’, where we went all in and thought, ‘We need to create this moment of intense craziness,’ ‘cause that was the emotion we were trying to convey, with the strings and the hard drums and Ed just screaming, delayed and distorted. But as a whole, it was more about the groove, the beats, the sounds we used. We used a few live drum parts — we did some recording at SARM and some overdubs there as well. But I think that was more the record company wanting us to try and experiment, because it had all been done so simply.” Vocally, Gosling says Sheeran tends to work incredibly quickly. “He’s very fast. It’s more about the emotion than getting the perfect take. But his singing is amazing and his harmonies are great. He’s very clear. As an artist, he really knows what he wants. I think that’s half the battle and it makes my job a lot easier. We never did line-by-line stuff. Occasionally there would be one word or something that we’d have to get. But it was very raw recording. I treat it very much like an analogue process of recording.”

Mix Farming
As is increasingly the case these days, once tracking was completed on +, the mixing duties for the album were outsourced to a number of people, namely Guy Massey, Charlie Hugall and Ruadhri Cushnan. “All of those guys were really good, actually,” Gosling says. “Ruadhri was brilliant because I’d be talking to him on the phone all the time. The label were like, ‘We don’t really want to change what you’ve done, Jake, we just want to try and enhance what you’ve done on certain tracks.’ And I was like, ‘Fine, great, that’s what it’s all about.’ We worked with Guy at SARM and those tracks were a bit more straightforward. We just wanted to literally enhance what was

originally done in the room, You can’t keep rather than take it away and Ed Sheeran from change it that much. It was a good sofa... just fairy dust really, but then fairy dust can be the magic dust, of course.” Neither Gosling nor Sheeran attended the sessions, but the mix-in-progress tracks would be shuttled back and forth online for approval. “With technology now, it’s very easy. You literally get sent the track there and then, as they’re doing it. You can listen to it and come back with your feedback and things can be changed here and there. But I like people to bring their own things to the music too. It’s important to let people find their own feet in the track and see what they can bring to it. And luckily they didn’t really change things too much, so that was great. Nothing really surprised me. I was really happy with the remixes. Everything ran very smoothly.” Just one song on the album was done outside the team: ‘Kiss Me’, the only track to survive from exploratory writing and recording sessions in Los Angeles with No ID (Kanye West, Rihanna). “I just recorded the vocals and guitars on that,” Gosling says. “Ed did quite a few tracks out there, but that one really felt like it fitted with the rest of the album. It worked with the sound that I’d created. It tied in really nicely.” For his part, Gosling admits that, upon its release, he was somewhat taken aback by the instant success of +. “Yeah, it was insane,” he says. “I think we all felt confident about it. I mean, a similar thing happened with No. 5 Collaborations. That was done with just me and Ed and not even a record label. So for that to get to
The relatively small amount of outboard at Sticky Studios includes this Avalon voice channel, M-Audio Profire 2626 interface and Audient ASP008 eight-channel preamp.

number two in the iTunes chart without any promotion was just amazing. Then when we had the label we thought, well, yeah, 40,000 for the album in the first week would be great. But then when we hit number one, it was amazing. ‘Cause it’s been such a long journey as well. It hasn’t been overnight, it hasn’t been processed and put together.” Ultimately, of course, while + has proved the making of Ed Sheeran as an artist, it has also proved the making of Jake Gosling as a producer. “Well, yeah, absolutely. The weird thing is I haven’t really changed what I’ve done. I’ve done the same thing and it’s all been about timing. I think a lot more people are wanting deeper music and lyrics and stories, rather than another pop track singing about the club, y’know. In terms of album artists, people are wanting real songs, especially the younger generation. But, yeah, it’s been a lot of hard work and sweat and tears and all the rest of it that goes into it. And Ed sleeping on my sofa.” Which is perhaps the most surprising aspect of this particular success story. Even after going platinum, Sheeran has found it hard to give up his skint musician ways and is currently still sleeping on his manager’s couch. “He loves his sofas,” Gosling laughs. “I’m gonna buy him a sofa. Actually, I might just give him my sofa. Which he’s wrecked by the way. It doesn’t fold up properly any more…”


January 2012 / w w w . s o u n d o n s o u n d . c o m


The BEAST system being set up before a concert.
Photo courtesy Jonty Harrison


e tend to think of loudspeakers as devices that just make music louder. But what if the loudspeakers themselves were part of a musical performance? That’s the idea behind non-standard multi-loudspeaker diffusion systems, where diverse loudspeakers are placed everywhere across the concert room and wired to the bus outputs of a control surface or a mixing desk. In these setups, a performer directs recorded sound around the speakers, treating the array as if it were a concert-hall-sized musical instrument. The result is usually quite


Non-Standard Multi-Loudspeaker Diffusion Systems
Many musicians spend their lives trying to cram hundreds of tracks into just two speakers — but performing with an NSML system presents exactly the opposite challenge!
fascinating: the music surrounds the audience, moves around and keeps changing colour and shape. These unorthodox methods have been gathering hardcore fans around the world for a good 50 years. In this article, we’ll meet some of the people using them, look at what’s involved from a technical point of view, and find out what sort of ‘performances’ they make possible.

Starting Out
The first non-standard multi-loudspeaker diffusion systems (or NSML systems,

w w w . s o u n d o n s o u n d . c o m / January 2012



for short) were developed in San Francisco and in Paris, France, in the ’60s. On the West Coast, composer Stan Shaff and equipment designer Doug McEachern introduced their Audium system at the University of California in 1960, settling into their own theatre in 1968. Meanwhile, at the Groupe de Recherche Musicale in Paris, engineer, inventor and composer Pierre Schaeffer
Jonty Harrison manages the BEAST multi-speaker system at the University of Birmingham.

developed research work he had begun before World War II. One result was the Acousmonium, a loudspeaker orchestra that was made fully functional by composer François Bayle in 1974. NSML systems made it possible to include space as a full dimension of musical composition. As Stan Shaff puts it, when you listen to a performance at the Audium Theatre, “sounds are ‘sculpted’ through their movement, direction, speed and intensity on multiple planes in space”. They also provided an answer to the age-old problem of how to create engaging live performances of electronic music. At the time, electronic music was still a minority interest and, as a consequence, NSML systems became associated with ‘avant garde’ music. Unfortunately, the growth in popularity of electronic music in the ’80s didn’t change this, and the vast majority of performances based on recorded music in techno, trance, and so on are played
Speaker setup for a BEAST concert in Elisabethkirche, Berlin: a complex instrument!

on stereo setups, while NSML systems are still used mainly to perform ‘avant garde’ music. This is a pity, because NSML systems have much to bring to any kind of music. NSML systems in Europe include the BEAST (Birmingham ElectroAcoustic Sound Theatre) system at the University of Birmingham, which is run by Jonty Harrison, while in Paris, Vincent Laubeuf is the head of MOTUS, a privately held company that specialises in concerts involving non-standard multi-loudspeaker setups.

Central to the idea of NSML systems is the notion of performance, albeit of a particular kind. During these concerts,
Photo courtesy Jonty Harrison

a large number of loudspeakers are generally set up. Jonty Harrison says he usually uses between 70 and 90 speakers during BEAST concerts, and the MOTUS equipment list gives the client the opportunity to use a 100-loudspeaker system. The art of performing on such a system lies in taking advantage of multiple factors such as speaker placement, speaker colour, and room acoustic properties. To fully take advantage of those factors, NSML systems include routing matrices that enable any channel from the playback medium to be directed towards any loudspeaker, as shown on the diagram. According to Jonty Harrison, “You want to enhance the musical shapes and gestures that are already contained in the music,

This photo shows the elaborate rigging necessary to position the speakers within the Elisabethkirche.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Courtesy of Jonathan Prager

Synopsis of the MOTUS routing system. Notice the frequency-selective amplification.

Part of a MOTUS performance. Note the ‘ball’ and ‘column’ loudspeakers with their highly coloured frequency responses.

with the concert-room acoustics. Such knowledge is of paramount importance. For instance, according to Vincent Laubeuf, it’s part of the MOTUS performance vocabulary to be able to play medium-high frequencies on dedicated speakers close to the audience. But this must be done properly: how does that sound from the audience’s point of view? Is there a part of the music piece that’s particularly well suited to this particular colour? Are those loudspeakers going to be used as support, or as main sources? What about the other frequencies that are heard at the same time? How does this work with the adjacent parts of the music? Many decisions have to be made, and this can prove disorienting at first, especially when one is dealing with large and complex setups such as the one shown in the diagram.

by articulating the space with the musical material. You may, for instance, translate dynamic shift or energy to spatial shift or energy.” Translated into practical terms, this means that the beginning of a musical crescendo might be played on a limited number of loudspeakers, and the louder the music gets, the more loudspeakers would be used. Similarly, a thin, mono musical part in a mix would be played on a given, narrow zone from the loudspeaker array,

whereas a large original stereo image would be dispatched on a wider choice of loudspeakers. Staying faithful to the music, while enhancing the way it’s written: this is indeed performance. In music, generally speaking, performance requires knowledge of both the piece to be played and the instrument. This is also the case with NSML systems: live shows have to be carefully rehearsed so that the performer gets to know the music to be played, the loudspeaker setup, and its interactions

Photo courtesy Vincent Laubeuf

Sound In Space
An important question of NSML system diffusion concerns how many tracks of recorded music to start with. While it might seem a bit strange to play stereo material on a 100-speaker system, it’s actually perfectly feasible. Vincent Laubeuf even prefers it over playback of multitrack material: in his opinion, having a very limited number of original sources being played simultaneously via a large number of loudspeakers

Where Can I Hear NSML?
There are a number of active multi-loudspeaker diffusion systems around the world, especially in Europe. The list below doesn’t pretend to be exhaustive, and only includes systems that appear to be in regular activity. Name Acousmonium du Groupe de Recherche Musicale AMEG Audium BEAST, University of Birmingham Casa del Suono Klangdom, Zentrum für Kunst und Medientechnologie Karlsruhe MOTUS Musique & Recherche Music Circus Sonic Lab, The Sonic Arts Research Centre, Queen’s University Location Paris, France Geneva, Switzerland San Francisco, CA Birmingham, UK Parma, Italy Karlsruhe, Germany Paris, France Brussels, Belgium Osaka, Japan Belfast, UK Fixed/mobile Mainly dedicated to Radio France’s studio 116 Mobile Dedicated to the Audium Theatre Mobile Dedicated to the former Santa Elisabetta church Dedicated to the ZKM Kubus space Mobile Mobile Mobile Dedicated to the Sonic Lab space Web site TheSARCBuildingandFacilities/TheSonicLab

w w w . s o u n d o n s o u n d . c o m / January 2012



often gives better results than having many speakers behaving independently. This is all the more true on the MOTUS system, which is based on extremely dissimilar loudspeakers. As for Jonty Harrison, though he understands Vincent’s point of view, he personally prefers to work with a higher number of tracks (his own recent work is typically eight-channel), for more flexibility. Whether one chooses to perform from traditional stereo content or from multitrack material, the issue of spatial perspective necessarily arises. For instance, in mixes destined for stereo playback, a sense of distance is typically obtained by adding reverb and/or filtering the high and low frequencies. This creates a virtual image of a ‘faraway’ musical object, which will seem to be emitted from way behind the speaker position. In NSML diffusion, ‘faraway’ will be more literal, since it can be obtained simply by playing that particular sound on distant loudspeakers. Also, in stereo, groups may be mixed on different ‘zones’ of the stereo image. In NSML diffusion, this will be translated to moments or tracks being directed towards different groups of speakers. Since there are no instrumental performers to look at, and no stage, should the traditional ‘front-back’ / ‘left-back’ concert space organisation be abandoned in favour of a more flexible, ‘isotropic’ setup? At the beginning of MOTUS, traditional concert directivity was altogether forsaken, but Vincent Laubeuf says that it was gradually reintroduced: listeners appear to pay more attention and feel better when most of the sonic information comes from the front. Jonty Harrison’s opinion is that isotropy may help one person who’s completely out of range, but generates a lower general audio quality. Also, he feels it may distort the original composer’s intentions, by restraining the possible range of musical expressions: for instance, there is no possibility of ‘intimacy’ if the distance between the listeners and each speaker is random. He also states that this debate is often a point of disagreement between him and his students. In my opinion, a major point in favour of isotropy would be that it makes the notion of sweet

spot disappear altogether. Listeners are ‘inside’ a sound scene, and can watch it from any point of view they want, even moving from one point to another during the performance. The matter seems open to debate...

The BEASTmulch software that controls the BEAST system


A central issue when performing with an NSML system is the coloration of the original musical content. Understandably, playback on such systems is not like playback on professional-range headphones: first, the sound will be coloured by the loudspeakers, and then by the room acoustics. NSML system owners can adopt different attitudes towards this phenomenon. According to Jonty Harrison, trying to remain true to the original colour of the recording Computer Support is most important. This leads BEAST to use as many full-range loudspeakers as Given the number of loudspeakers possible, so as to minimise the coloration. involved, NSML system designers As for reverberation, it is naturally resort to software considered a component controllers. In Karlsruhe, of space, thus being Germany, the Institute of integrated into the spatial Music and Acoustics (ZKM) image. At MOTUS, Vincent use the Zirkonium control Laubeuf handles things a bit software, mainly to deal differently. He also considers with ‘super-panning’ across room reverberation part of numerous loudspeakers. the spatial image during the In Birmingham, the BEAST performance, and goes a bit team use BEASTmulch, further, regarding spectral a program written by Scott Vincent Laubeuf of MOTUS. coloration as part of the Wilson that manages all interpretation. As a result, although both aspects of the diffusion system: playback, MOTUS and BEAST use, for instance, routing matrices, ‘super-pans’, and so speakers facing walls, in order to be able on, running in tandem with real-time to get ‘extra far-sounding’ parts, MOTUS human control. BEASTmulch appears also uses a variety of unorthodox custom to be especially practical in its ability to

speakers to add ‘spectral performance’ on top of ‘spatial performance’. NSML systems are not restricted to the playback of pre-recorded music: acoustic instruments can be integrated into the performance too. As usual, different approaches are possible. Karlheinz Stockhausen’s Kontakte has been performed using the BEAST system alongside piano and percussion, and MOTUS sometimes invite instrumental improvisers who react to the spatial imaging that’s derived in real time from their performance. At the Groupe de Recherche Musicale in Paris, pieces are also specifically written for the local NSML system, the venerable Acousmonium.


January 2012 / w w w . s o u n d o n s o u n d . c o m

Courtesy of Scott Wilson

recall pre-recorded presets, so that clear and spectacular transitions in the spatial imaging can be made. Use of control software is not universal, though, and the MOTUS team prefer to do without, so that the human performer always remains central to the concert. Computer support proves indispensable when using regularly spaced loudspeaker arrays or other special spatialisation techniques. For instance, the ZKM in Karlsruhe uses a ‘dome’ of loudspeakers that hangs from the ceiling, and BEAST also incorporates a dome-like array of speakers in the system. Such domes are meant to project the sound in a very specific way, not unlike IRCAM’s Spat system, which requires precise volume control that only computers can achieve. Other examples include the use of Wave Field Synthesis techniques, for instance in Parma, Italy,

for the Lampadario Sonoro in the Casa del Suono. This special sound-projection technique dedicated to the creation of virtual acoustic environments naturally finds its place inside NSML systems set up in relatively small spaces.

NSML For Pop Music
As we’ve seen, for historical reasons, NSML systems are most often used to perform Schaefferian-style ‘acousmatic’ music. However, both MOTUS and BEAST have worked with musicians from the ‘pop’ scene — respectively, Aphex Twin and Scanner — and both Vincent Laubeuf and Jonty Harrison admit that the outcome was not really satisfactory. Apparently, the musicians were primarily concerned about being heard, and there was little time for them to engage with other aspects of the performance, such as spatialisation. This seems surprising, as

MOTUS & BEAST Loudspeaker List
Both Vincent Laubeuf and Jonty Harrison were been kind enough to provide the list of loudspeakers they use, and this information reveals a lot about the character of their systems. Whereas the people at BEAST seem to be primarily concerned about transparency, the people at MOTUS have no complex about ‘re-colouring’ the original audio content. MOTUS System ‘A’ (55 loudspeakers) Full-band loudspeakers Brand Model Quantity Boost MT502 2 Cabasse Cyclone 2 Custom built ‘JP’ 8 Elipson ‘Stars’ E50 3 Elipson Aria 3 2 EV MS802 4 JBL 4408 2 JBL 4312 4 JBL 4411 2 JBL 4315 2 Narrow-band loudspeakers Bose 802 Bouyer ‘Columns’ Bouyer ‘Pavillons’ RB 540 Custom-built ‘Boomers’ Custom-built ‘Cube’ Custom-built ‘Large Black’ Custom-built ‘Sound Rod’ Custom-built ‘Tupperware’ Elipson ‘Ball’ Raveland ‘Subs’ X8828 MOTUS System ‘B’ (45 loudspeakers) Full-band loudspeakers APG MX Bose M25 Elac ELT MKII 2 2 2 2 2 2 2 2 2 2 Brand Elipson Elipson JBL JBL Model ‘Balls’ ‘Stars’ E40 4412 Control 25T Quantity 4 2 2 8 2 4 2 1 4 2

Narrow-band loudspeakers Bouyer ‘Balls’ RB 34 Elipson ‘Cyclopes’ EquipScène ‘Sound Projectors’ FBT ‘Sub’ MaxX 10Sa Monacor ‘PAD’ SP3051 Monacor ‘Pavillons’ UHC30

BEAST Full-band loudspeakers Brand Model Quantity APG MC2 12 ATC SCM50A 2 ATC SCM50 6 Genelec 8050A 8 Genelec 8040A 16 Genelec 8030A 24 Genelec 1029A 8 Genelec 1037B 2 Genelec 1037C 2 HHB Circle 5 10 HHB Circle 3 8 Tannoy Lynx 2 Volt Home Studio Monitor 8 Dedicated subs Genelec 7070A Genelec 1094A 8 2

8 4 2

Dedicated tweeters Motorola Array of six piezo-electric 10 tweeters Motorola Wide-dispersion 10 piezo-electric tweeters

there is no obvious contradiction between playing loud and taking advantage of more than 70 loudspeakers — quite the opposite, really. In my opinion, the actual issue here might have been a certain reluctance to lay aside the traditional codes of the stereo image, starting with kick and snare in the middle, along with any lead vocals. It seems likely that proper playback of music pieces including these parts would require them to be on separate tracks so that they can remain immobile, while the spatialisation work would be made on panned material. Obtaining suitable multitrack versions of the material could be a challenge from the technical and copyright points of view. But, that said, imagine a piece like ‘A Warm Place’ by Nine Inch Nails, played on 80 loudspeakers surrounding the audience. It would make perfect sense in regards to the original intention, especially as far as the 16th-note hard-panned guitars are concerned. Also, still considering Nine Inch Nails’ music, look at the shift in space and arrangement between 3’01 and 3’11 in ‘Right Where It Belongs’: it almost sounds as if it’s specifically written for multi-loudspeaker systems, reduced to a stereo image for lack of better playback solutions. To sum up, in my opinion, despite probable initial difficulties to be encountered and solved, intelligent collaboration between NSML system owners and ‘pop’ musicians would be mutually beneficial to an enormous extent. There is something extraordinarily refreshing about NSML concerts. Very different from your usual orchestra or band on a stage in front of you, utterly unlike the same two loudspeaker stacks next to the dance floor, and far richer than austere 5.1 or 7.1 surround, properly used NSML systems immerse the audience in sound in a way no other situation can. I sincerely advise every person interested in the sonic aspect of music to go to such a concert at least once in their life. It’s a shame that the use of such systems is, for the moment, restricted to a particular kind of music. Let’s just hope that NSML system developers will keep on building bridges with other styles of music, and that ‘pop’ musicians, labels and concert promoters will realise the added value that non-standard multi-speaker setups can bring into small and medium-scale live shows.

w w w . s o u n d o n s o u n d . c o m / January 2012



Secrets Of The Mix Engineers: Randy Staub



he way the music business works,” says Randy Staub, “is that you get pigeonholed as an artist, a writer, a producer, an engineer and as a mixer. If you’ve had some success with heavy rock, like I had, people will naturally think that that’s all you do. I do like rock music, but I don’t like music because of its genre. I like it because it’s good. It can be extremely heavy, or it can be Hank Williams, or 50 Cent. I love all kinds of music, but sometimes, when I’m 13 tracks into mixing a heavy rock album, I find myself wishing that I was mixing a girl singer with an acoustic guitar!” While Randy Staub’s sentiment is understandable, it’s also understandable that he’s regarded as a living legend in the world of rock, having worked with Mötley Crüe, Nickelback, Metallica, Bon Jovi, Iggy Pop, Alice In Chains, Bryan Adams, Hinder, Lostprophets, Evanescence and many more. Hailing from the town of Prince George in British Columbia, Staub always wanted to be an engineer. He recalls,

The multitracks for Evanescence’s third album were so big that they required two maxed-out Pro Tools rigs to play back!
“I always liked the sound of records and the technical aspect appealed to me, so to become an engineer was the obvious thing.” After leaving high school in the late ’70s, the Canadian attended a summer recording course in Rochester, New York, and spent a while doing live sound before being employed as an engineer at Phase One Studios in Toronto. It was there that he met fellow Canadian Bob Ezrin, who recommended him for a job at A&M Studios in Los Angeles. Staub spent three years at A&M, but was eventually persuaded by another top Canadian producer, Bob Rock, to return to Canada, to work in the latter’s Vancouver studio. The first record Rock and Staub did together was Mötley Crüe’s Dr Feelgood (1989). Two years later they worked on Metallica’s eponymously titled album, also known as the Black Album, which went 15 times platinum in the US. Unsurprisingly, Staub has, as he says, “worked non-stop since”. Since Rock moved to Hawaii in the mid-’90s, Staub has mostly worked out of Bryan Adams’ The Warehouse

‘What You Want’
Written by Amy Lee, Terry Balsamo, Troy McLawhorn, Tim McCord, Will Hunt Produced by Nick Raskulinecz


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studio in Vancouver. Until 2001, his credits were fairly evenly divided between engineering and mixing, but for the last decade, 95 percent of Staub’s work has been mixing. “It was a conscious decision to focus on mixing,” Staub explains. “It has always been the thing that I enjoy the most.”

Evans Above
One of Staub’s recent high-profile projects was his mix of the entire third album by the American goth metal group Evanescence, including lead single ‘What You Want’. The disc, simply titled Evanescence, ticks all the boxes that characterise much of Staub’s work, with heavily distorted rhythm guitars, monolithic bass and drums, and an in-your-face sound image that is very, very loud. Singer Amy Lee’s dramatic voice and crystal-clear acoustic piano (she’s classically trained) are an essential part of the band’s identity, and extensive string arrangements form the icing on the cake. Though it has yet to match sales of the band’s debut album, Fallen (2003, 17 million worldwide sales) or its follow-up The Open Door (2006, five million), it reached the top spot in the US, and has so far peaked at number four in the UK. The band began work on the album early in 2009 with producer Steve Lillywhite (U2, Peter Gabriel, Talking Heads), but eventually shelved these recordings, and a year later switched to producer Nick Raskulinecz (Foo Fighters, Alice In Chains, Rush). Raskulinecz and Staub had earlier both worked on Stone Sour’s Audio Secrecy (2010) and Alice In Chains’ Black Gives Way To Blue (2009), amongst other things, experiences that set the tone for a fruitful working relationship on Evanescence. “Nick and I have done a few records together, which have generally been big rock records, and that was definitely the approach here,” says Staub. “I always tell people that you have to create the sound of the record that you want right from the beginning, so that when the mixer first pushes up the fader, the record is there. Everything sounds the way it should, the arrangements are good, the emotion is there, the balance is there, and as a mixer

Randy Staub mixed the Evanescence album on the SSL 4072 in Warehouse Studio 1.

your job is to take it to another level. You pump it up and make it sound larger than life. This rather than trying to create something that wasn’t there to begin with. Luckily Nick works like that. His stuff has its character recorded into it, so I don’t need to mix it into it. The arrangements are great, and Amy’s vocals are powerful and recognisable and tell the whole story of each song right from the start. So it was a matter of making her vocals sound even more powerful and making the drums sound bigger and making sure that the strings could be heard. There’s lots of everything, and it all had to sound big! “I mixed 16 songs over five weeks during the Summer of 2011. So on average we mixed a song every two days. Generally, my process is to work on a song and get it to sound great during the first day, leave it overnight, listen with fresh ears the next morning, make some changes, and then spend the afternoon printing all the different versions. One thing that can take time is waiting for feedback from the artist and producer, if they aren’t there. I’ll post mixes for them online, and hopefully they’ll get back to me right away. But it can take an hour or a day before people get back to me with comments. I then make some changes, and post the updated mixes, and then wait again for comments. “I like to talk with the producer and artist before I begin mixing, to get as clear a picture as possible of what they want to achieve, and for the same reason I like to hear the rough mixes, because

people will have spent a lot of time on them, and they give me a basic idea of the kind of balance and perspective that the artist and producer have in mind. Zach Blackstone, my assistant, and I generally work by ourselves here in Vancouver, but for these sessions Nick and Amy came over for a couple of weeks, and I really enjoy that, especially when it’s at the beginning of an album mix. When you’re doing an album it can take a few songs to find where the album is going to sit, sonically, thematically, musically. Once the first two or three mixes are done, you know what ballpark you’re in. You hopefully have a roadmap and you know where you’re heading in terms of a vision for the album.”

‘What You Want’
Randy Staub: “We mixed this song first — the record companies always want to begin with the first single, even though, as I mentioned before, it can take one or two songs to know where you’re headed. This [Pro Tools] session was very well-organised, Nick is really good at that. In fact, we’ve recently been getting more and more sessions that are well-organised and clearly labelled. Enough engineers and mixers appear to have been impressing on people that it’s important to send clear, clean sessions. We don’t need Playlists, we don’t need stuff hidden. When I open a session, everything that the artist and producer want in there should come up immediately, so I can see it straight away. One issue with this

w w w . s o u n d o n s o u n d . c o m / January 2012



session was that we needed two Pro Tools rigs to run it! Apparently, unlimited tracks is not enough any more. The session was at 96/24, and at 96k you can max out a Pro Tools system. “Zach gets in the studio before I do, and he will organise a session in the way that I like — I’ve been doing sessions the same way since the beginning of time. He’ll colour-code everything, clean up tracks and adjust levels, making sure everything is within reason and nothing will be distorting the console when it comes up. Because of the amount of tracks on ‘What You Want’ [around 150] he combined some of the tracks, and I also had to premix things before they go into the desk. I have a 72-input console, but if there are nine guitars all playing the same part, they will usually come up on the same channel on the desk, and the strings will tend to come up on two faders, although if there’s a string part that needs its own treatment, it’ll come up on a separate set of faders. “It is a big session, with about 23

kick and the snare. The reason is that a lot of the sound of the drums is in the top mics. It’s fairly easy to get a good sound from the close mics on the kick and snare; it really is the sound of the whole drums that’s trickier. Is the hi-hat in phase or out of phase, is the second floor-tom out of phase, and if so, why? Are the room mics in phase with the overheads? On occasion I might nudge stuff in time. Once I’m happy with the drums, I’ll put in the bass, make them fit, make sure all bass tracks are in phase, maybe go back to the drums, and so on. Once I have the drums and bass the way I like, I go to the vocals, and after that I’ll blend in the other things. But the bass, drums and vocals are the foundation.” Drums: Waves SSL EQ, desk EQ and compression, AMS reverb, Pultec EQP1A, GML 8200, SSL G384. “Everything you see in the session came from them [ie. Raskulinecz and the band], apart from a couple of kick samples that I added, which I like to do, because you can EQ them very radically. They

Randy Staub: “I always tell people that you have to create the sound of the record that you want right from the beginning, so that when the mixer first pushes up the fader, the record is there.”
tracks for the drums alone, 20 vocal tracks, and 72 string tracks, but the arrangement is the key to it working. That’s often forgotten these days. Like if the bass is very low, as in this track, you can change the voicings of the rhythm guitars to make them sound lower as well. With strings, particularly in rock music, it is all about the parts that they are playing. It’s so easy to have strings just playing chord changes, kind of like a pad, and if it’s a pretty aggressive rock track, the strings will just blend in and you won’t hear them. The fact that you can hear the strings in this song is mostly down to the arrangement. They are well arranged and all the individual strings play good parts. “When I begin a mix, I will normally push all the faders up right at the start, and get a rough balance of everything. I’ll then concentrate on the drums. I won’t switch the instruments and the vocals out, I simply turn them down, so I can still hear them in the background and hear the drums in context. I normally begin with the top drum mics, like overheads, hi-hats, toms, rooms, and then I’ll blend in the had an NS10 [a Yamaha driver used as a microphone] going to record the bass drum, and also added some tom samples. ‘Kicksnare Compex’ at the bottom of the screenshot is something they did, probably a mic between the kick and snare, on which they added a Compex limiter. It sounds very compressed, very trashy. The volume rides on the toms were also done by them — I would have done them if they hadn’t. I put the SSL channel plug-in EQ on the tom tracks, and one of the snare mics, I think because I wanted these tracks to sound more similar. The five room tracks gave most of the space to the drums, and I had them coming up on a stereo pair on the desk — you can see the desk channel routing in the I/O column. I also had a little bit of AMS reverb on the drums, but not much. I used desk compression and EQ on the drum
This composite Pro Tools screenshot shows the session on the ‘A’ rig. From top, the tracks are colour-coded purple (drums), green (guitars), grey (bass), and red (vocals). The grey and blue tracks at the bottom are auxiliary tracks containing send effect plug-ins.


January 2012 / w w w . s o u n d o n s o u n d . c o m

channels, and also some outboard EQ like the Pultec EQP1A and GML 8200 on the kick and the snare channels. “On the console, I sent each small drum fader to what I call a drum compression group, on which I had an SSL G384 compressor and the GML 8200 EQ. Most of all, I’d send the kick and snare there, but also, in lesser amounts, the overhead and room mics. The amount of compression really varies per track — sometimes it’s quite extreme, sometimes it’s very little. On this song, I had a pretty solid amount of compression and EQ, and I’d blend the compression group in with the original channels. It’s something that Bob [Rock] and I started doing years ago, and that’s now pretty standard. I also often send some of the bass to the drum compression group. It is intended to make the drums and bass sound punchy, and larger than they really are. Part of mixing rock music is to get more excitement in a track than really is there, and compression seems to do that. Again, it makes it sound larger than life.” Guitars: Neve 1073, desk EQ and compression, SRS Wow Thing. “The guitars were underneath the drums in the session. There are two times four tracks of rhythm guitars, each representing one part that was played via two amps and each amp was recorded with two mics. One set of four, played by Troy [McLawhorn], came up on channels 9 and 10 and was panned left. The other, played by Terry [Balsamo] came up on channels 11 and 12 and was panned right. You’ll notice the volume automation on Terry’s guitars, which was to take out some of the noise in the quiet bits between the chords. There were no plug-ins on the guitars, just outboard. I had the Neve 1073 EQ and console EQ and compression. I don’t generally use a lot of compression on heavy guitars, because the big amplifiers generally have compression and distortion built in anyway. If the sound is a bit muddy and it’s a very choppy part, I may use compression to get some control over the lower notes. There’s also a single track of chorus guitar that came up on channel 13 and that I simply EQ’ed to make it fit. “One effect that I often use on guitars is a small effect box called the Wow Thing that I bought for $20 many years ago. I’m letting my secrets out here! At the time, I was listening to music on a computer at home, this was maybe in the mid-to-late

Secret weapon: the SRS Wow Thing.

’90s, and it sounded really good and super wide. I wondered how it was done, and when I dug into the computer I found this piece of software made by SRS Labs called the Wow Factor. The higher you pushed the slider, the wider the sound got. It really is just a phasing program that makes stuff sound wider, to the point that it may sound outside your speakers. So I went to their web site and bought this small hardware box that cost 20 bucks, and I found that it sounds great on guitars. I will send them to the box using the small faders and spread them out left and right. SRS also make professional spatial effects devices, and I do have one of them, but I find that this small box still works the best for me.” Bass: desk compression and EQ, Pultec EQ, Focusrite Red 3. “‘Taurus Moog’ immediately below the guitars is a low pad-type sound, and ‘Audio 1’ and ‘2’ are two brief vocal parts on which I put the [Waves] L1 and the [Sound Toys] Filter Freak. It’s an effect vocal that appears in the middle of the third verse. Below that are five bass tracks, one from an Ampeg amp, one DI, one from a sub cabinet, and the JMP1 and Sans tracks have a distorted bass sound, using the Sansamp. I had the Ampeg track on channel 20, and grouped the DI and sub on channel 21 and the distorted tracks on channel 22. I used desk compression and EQ on these. If I did use outboard on the bass, it would have been fairly minimal, perhaps some Pultec EQ on the DI and sub, and if I did use outboard compression it would have been from the Focusrite Red. But these were pretty solid bass tracks and I didn’t need to make them nice and solid.” Vocals: Waves SSL Channel and Renaissance Compressor, Avid Lo-Fi and Revibe, Line 6 Echo Farm, Retro 176, Dbx 902, GML 8200, Eventide DSP4000 and H3000, Yamaha D5000, Lexicon PCM42, Urei LA2A. “The first three vocal tracks — ‘Verse Vocal’, ‘Chorus Vocal’ and ‘Bridge Vocal’

w w w . s o u n d o n s o u n d . c o m / January 2012



Warehouse Party
In the context of modern budgets, the five weeks that Randy Staub was given to mix Evanescence sounds luxurious, not least because he was also working on a desk in a commercial studio. Staub insists that he’s not in the least compelled to go down the working in-the-box-in-one’s-own-studio route. Instead, he’s proud of his working methods, which continue to centre on The Warehouse Studio 1’s SSL 4072 GTR with black E-series EQ. “I learned my skills on an SSL, so I became comfortable and proficient on it. The SSL has a big, punchy sound that lends itself especially well to rock and hard rock. The SSL I work on dates from the mid-’90s, and it sounds really good. A guitar player will have a favourite guitar that feels and sounds good and that he knows how to play, and this desk is like that for me. It’s my instrument. I cannot ever see myself mixing in the box, because it just doesn’t feel right, and it sounds different. I’m not saying it sounds better or worse, it just sounds different from what I like. I also like to be able to reach for knobs, and often two knobs at the same time, which is difficult to do when you’re working in the box. “I like working in The Warehouse, because of the great-sounding room, the technical facilities and the staff, which all make my job much easier and allow me to focus on getting a good product. It’s at the same at A&M, where I usually mix when I’m in LA. But it’s amazing that studios can still stay in business. I don’t think rates have increased since the 1970s. One of the pitfalls of many people doing stuff at home now is that the craft of recording is at its lowest point ever. With the studio culture disappearing, nobody is learning any more how to record properly. People buy studio equipment, put it in their house and immediately think they’re a recording engineer. But a lot of the stuff that I get in to mix has been recorded horrendously. People have no clue what phase relationships are, they don’t look at meters, because a lot of recording gear has no meters anymore. I have sent tracks back, saying that it isn’t good enough yet, and that I don’t want to waste their money and my time. “Pro Tools is a fantastic program, but a lot of the time it doesn’t get used in a good way, with bands banging out a couple of takes and everything then being edited together and fixed in the DAW. Back in the day, a whole team would take three months to make a record, and over that time the musicians would have to keep on playing until they got it right, and they became better musicians in the process. But three guys today spending months chopping things up in a DAW probably adds up to the same amount of man hours! I don’t have a problem with Pro Tools as such. I mostly use it as a storage medium, even as it’s also very good for editing and has some pretty good-sounding plug-ins. A record is not going to sound good or bad because it was recorded on tape or Pro Tools. It will be good or bad because it’s good or bad, not because of the recording medium.”

— are the lead vocals. You’ll notice that the chorus and bridge vocals both come up on channel 24, while the verse vocal comes up on channel 23. I wanted to give the verse vocal a different treatment, something that appears to be fairly common nowadays. The verse vocals are often sung with different volume and energy and require different treatment. The plug-ins on all the vocals were already

there, and I left them. When a session has plug-ins, I tend to use them, because I assume that the sound they give is what’s intended. The verse vocal has the SSL Channel plug-in on the insert, which I rarely use because I have an SSL console. ‘R’ is the Renaissance compressor, and you’ll notice that it’s on all the vocals. ‘L’ was the Lo-Fi plug-in, to give the verse vocals a little bit of edge.

“The sends go to reverb and delay effect tracks that also were already on the session. I generally use my own reverbs and delays, but they were happy with the sound that they had going, so I used that as well. They’d spent a lot of time and effort to get something they liked, so I used it and hopefully improved on it. [Send] 2 has the Revibe reverb and 3 and 4 have Echo Farm delays. You can see these

Total Recall
As well as providing full screenshots from both Pro Tools rigs (see elsewhere in this article), Randy Staub kindly supplied recall sheets for the hardware used during his mix of ‘What You Want’. The recall sheets are much too large to reproduce here, but can be downloaded from the SOS web site at jan12/articles/insidetrackmedia.htm.


January 2012 / w w w . s o u n d o n s o u n d . c o m

effect tracks at the bottom of this Pro Tools session. They came up on channels 39-40 on the desk. Below these three effect tracks, you can see my two mono Echo Farm and two stereo Revibe reverb tracks, which were coming up on channels 47 and 48 and 49-50 respectively. I recall that after I mixed this song, ie. when mixing the other songs, rather than send the effect from Pro Tools to the desk, I sent the signal from the desk to Pro Tools, the reason being that if I made a volume adjustment on the desk, it wouldn’t change the level of the reverb. “I really like the sound of Revibe, and I use it like a piece of outboard gear. Whether something is in an outboard rack or in Pro Tools, it’s still a digital reverb. Other rack outboard gear that I used on vocals included the Retro 176 Limiting Amplifier as my main vocal compressor, the Dbx 902 de-esser and the GML 8200 EQ, and I may have used a GML compressor as well. As always, I would have used console EQ and maybe some console compression. The Retro is my own, and I really like using it on vocals. I also may have used the Eventide DSP4000 and H3000 for chorusing and delays and the Yamaha D5000 and PCM42 for delays on the vocals. I’ll treat the background vocals differently than the lead vocals, to try and separate them a little bit, and may have had a ‘Dual 910’ program on the H3000, or different chorusing and delays. All EQ and compression on the BVs would have been done on the console, with a little bit of LA2A as well.” Keyboards: desk EQ, Pye compressor. “Underneath the vocals is a ‘Piano Effects Print’ track, which was the only piano track that I had. It had the effects that they wanted, so I did nothing else to it, other than desk EQ it to make it sit in the track. The other keyboards are in the other Pro Tools session, at the top and bottom of that session. At the top are some pretty weird-sounding synth overdubs, so they have their tone and sound built in, and I didn’t do much to them, other than some desk EQ to make them fit. There’s also a loop, on which I had a Pye compressor. At the bottom are some more synth sounds, arpeggiated ones, and bells, and stuff, and again that was purely a question of EQ’ing them to make them fit in the track.”

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w w w . s o u n d o n s o u n d . c o m / January 2012



The ‘B’ rig Pro Tools session contained synth parts and loops (top and bottom) and four takes of a small string section.

Strings: Avid Revibe, desk EQ. “All the strings came up on 41-42. I would balance and pan the strings in Pro Tools, with the violins hard left, the violas soft left, the cellos on the right and the basses in the middle. There was a mono room sound on all string takes that I did not use. Instead I sent all strings to a Revibe effect track in Pro Tools, that also comes up on 41-42. The Revibe has a medium dense church setting. Takes one and two came up on subgroup 21-22 within Pro Tools, and take three is a different part that was routed to subgroup 25-26 while take four went to busses 29-30. You’ll notice that there’s a big volume ride in the middle of the bridge on Take 3. Take three is panned hard left and take four is panned hard right. Because take three and take four play different parts from takes one and two, I wanted them to stick out a bit more. In addition to this, I only used EQ from the console on the strings.” Master bus: SSL Quad Compressor, Sontec MES 432. “I mixed to another Pro Tools rig that was also at 96/24, via the Apogee PSX100 [A-D converter]. I put an SSL Quad Compressor over the stereo mix — it is built into the console — and after that the master fader went to the Sontec MES 432, which is one of the best EQs ever made. The amount of compression I use on the Quad can be from almost none to 8-10dB at 4:1. In this case it was 2:1, with a fairly slow attack and release. I then sent off the full-bandwith file to the mastering engineer. What happens in mastering is, generally speaking, out of my hands, it’s between the mastering engineer and the producer and artist. Records have become too loud, and I don’t like it too loud. You can only go so loud until you reach a point where records don’t sound very good any more. They become too linear, are fatiguing to listen to, and will distort on consumer electronics, which can’t take the level. I think Nick is on the same page here. Yeah, this album is loud, but I don’t think it’s the loudest thing out there. We definitely try to impress on the mastering engineer and artist that the track needs to have some dynamic range, so that people can turn it up and it will still sound good.”


January 2012 / w w w . s o u n d o n s o u n d . c o m

Notes From The Deadline
TV Music From The Inside
We all have musical skeletons in our closets, but at least media composers don’t have to wheel them out every night at Wembley Arena.

n June 1979, the American music industry magazine Billboard listed a song called ‘Ready ’n Steady’ by D.A. at number 106 in its Bubbling Under chart. The song rose to 102 a couple of weeks later, before disappearing altogether. And it didn’t just disappear from the charts: it disappeared entirely from human record. No copies are known to exist anywhere, and even the most dedicated record collectors and pop historians have concluded that while it was almost certainly released, sold and performed across America — physical sales and radio plays being virtually impossible to fake on that scale — the record, its sleeve and the band or artist that created it have simply fallen off the face of the earth. And who among us doesn’t have at


least a few pieces of work out there that we could wish a similar fate upon?

Out Of Control
One of the biggest myths about this job is that we somehow have any real control over our careers. Like so many in this industry, the only tangible power we have is the ability to say ‘No’ to a particular gig, and even then, what we bring to the party can often be so pushed and pulled out of shape by the committees of the production process that occasionally we can feel like mere bystanders in the creation of a mutant. A mutant that carries our name and could potentially be heard by millions. And in situations like this, I don’t know about you, but I always feel a bit like a male porn star who keeps his socks on. He turns up for work and delivers the goods, but you somehow get the feeling that his heart just isn’t in it.
“One down... nine hundred and ninety-nine to go...”

The chain of command in an expensive TV production (and they’re all expensive) means that creative freedoms tend to be seen as a bit of a luxury item. And, to be fair, there are usually an awful lot of careers riding on the success or otherwise of even the humblest broadcast endeavour, so radical creative lurches in any particular direction are seen as incredibly risky to all concerned. This is why the most common complaint levelled by viewers at TV is that it plays too safe and is frequently derivative. In fact, it’s a rare and wonderful skill to be able to take what often seems like insane, meaningless and contradictory criticism of your work, yet still pull something out of the hat to the satisfaction of the production team, the audience and, ultimately, yourself.

It Could Be Worse
But at least media composers get to move on. You want to know why rock stars are mostly insane? Because their audiences usually only want to hear what they created 30 years ago, and no-one buys the new stuff. At a Radiohead concert, the rule seems to be: anything from the first three albums — crowd goes bananas. Anything from the last few albums — crowd claps politely. So, while media composers all have things we aren’t especially proud of in our back catalogue, we should rejoice in them and see them as steps along a never-ending road of creative self-improvement. If hearing something you did 10 years ago makes your cheeks tingle with embarrassment compared to what you can do today, take comfort in the thought that at least it isn’t the other way round.

w w w . s o u n d o n s o u n d . c o m / January 2012




We examine the production of some recent hit records to help you brush up your listening skills.
Here’s a production that feels to me like it’s trying a bit too hard to sound big and clear on low-quality systems, giving owners of pimped VW Golf GTIs some cheap sub-100Hz thrills and ensuring crispy ringtone transmission at the top end, but at the expense of a deficit in the 300Hz ‘warmth’ zone. The HF transients and sibilance feel overcooked too, and if they had been limited, to take off some of their plasticky edge, I reckon the lyrics would have come through better and you’d have been able to turn things up more before your ears started bleeding. Despite my reservations about the overall mix tone, though, a couple of nice rhythmic features did definitely catch my ear. The first is at 0:38 where the lead rap’s lyric “on my brand new white trainers” is echoed by The second rhythmic trick is of a similar type, but is created during the song’s outro (from 2:47) using the main mariachi horns sample. If you compare this to the intro (0:00-0:16), you can hear that Rizzle Kicks have effectively moved the trumpet line’s characteristic single high-register note an eighth-note early. Admittedly, it’s always possible that it’s actually played like this during another part of the original recording they sampled, but given the careful manipulation and editing of the samples elsewhere in the track (especially in the choruses), I’m inclined to believe that the credit is due to them. Whatever, the message I want to get across here is that the general idea of shifting repeated musical material into a different pocket in the groove is seriously worth investigating for any chart material, because it lets you have your cake and eat it: you can repeat your hooks more often, so that they worm their way better into the listener’s consciousness, but still keep them sounding fresh, so that all that repetition doesn’t just bore people to death. Mike Senior to really hear the details and sustain of his playing, particularly on the snare. It worked for ‘Back In Black’ and it’s just as useful in the verses here. Speaking of the verses, notice how the guitar texture’s mid-range pretty much evaporates when the vocals arrive, allowing Grohl’s restrained lower-register delivery to carry through — although that spectral energy reappears briefly to emphasise the sustained-note hook riff that straddles each new phrase’s ‘missing’ downbeat. Then, where the guitars regroup at 0:56, the vocals mostly join the snare in the riff’s perforations. You can gauge how effective this is in retaining the audibility of the vocals by realising how much more difficult it becomes to make out the one lyric that does trespass over the guitars (“I thought I’d save my breath for you”). It’s only when Grohl reaches bona fide howling registers for the chorus at 1:10 that he can afford to take on the full might of the guitars, and even then with a fader hike to help him out. This is common sense, but it amazes me how often I encounter home-brew recordings that haven’t taken even these kinds of rudimentary concepts on board, and hence struggle unnecessarily to achieve a good balance at mixdown time. Finally, here’s another mastering-related poser: although this track is mostly flat-topped to 0dBFS throughout, the brickwall level reduces to -0.3dBFS just for the solo (3:08-3:28). Is this some kind of last-minute ‘fix it in the shrink-wrap’ post-mastering level tweak? Have they tried to splice together files from two different

“Here’s a production that feels to me like it’s trying a bit too hard to sound big and clear on low-quality systems.”
the doubles, but with the rhythm starting in a different part of the bar. The result is that all the vocal stresses in the repeated phrase occur in a different relationship with the main beat, making it sound fresh and interesting even though you’re hearing it for the second time. Simple, but very effective.

‘Walk’, the other big single off the Foo’s album Wasting Light, has one of my favourite arrangement drops at the start of its final choruses, but I’ve chosen to focus on ‘Rope’ here because it’s a great example of some stalwart rock arrangement fundamentals in action. An easy way to make your drummer sound powerful in the face of epic guitars is to punch a few holes in the main guitar/bass riffs, allowing you


January 2012 / w w w . s o u n d o n s o u n d . c o m

mastering engineers (Emily Lazar & Joe LaPorta are both credited)? Your guess is as good as mine... Mike Senior

A successful production isn’t all about mixing and mastering. In fact, the songwriting and vocal performance can often be sufficient to carry a track. Indeed, I suspect that the runaway success of Christina Perri’s low-budget debut single ‘Jar Of Hearts’, following its appearance on the TV show So You Think You Can Dance, will have taken the artist and the production team by surprise. As the songwriting is clearly what’s been selling this track, I’d like to point out a few interesting musical features that I think might have helped it stand out from the crowd. Firstly, note that Perri’s not afraid to deviate from simple four-/eight-bar phrase lengths, as with the nine-bar chorus, ‘middle 10’, and six-bar outro. The book-ending of the chorus lyric with “who do you think you are?” is canny too, both repeating one of the song’s more evocative turns of phrase and giving a sense of closure to the section. This is especially relevant at the end of the song, because not only is the repetition missed off the end of the first final chorus, underlining the fact that we’re not yet finished with the chorus material, but it’s repeated three times after the second final chorus, closing more strongly to offset the double-length chorus, as well as rounding off the song as a whole. The harmonies also reward examination. The rate at which the chords change is varied very effectively, clearly differentiating the pre-chorus and mid-section from the verses and choruses, but what characterises this song most strongly is its internal major/minor tug-of-war. You can see this in the contrast between the minor verses and mid-section, and the major choruses, but the same tension is apparent as you zoom in further. The most obvious example is the major-to-minor shift in the chorus’s IV-I cadence, but the repeated outro cadence carries a minor flavour too, because of the way its diminished-seventh chord has more in common with Eb minor than Eb major. A couple of little musical patterns help with the song’s longer-term flow, the first of which is the verse’s Eb-Bb-Fm-Cm cycle of fourths. Although cycles of fourths are

slightly less common harmonically than cycles of fifths, they’re no less effective at creating harmonic momentum under the right circumstances, and are particularly suited to a song which, like this one, features a IV-I cadence as its main chorus resolution. The second pattern is a melodic one: the descending chromatic bass line underpinning the middle section. Not only does this proceed through C-B-Bb-A four times, but continues the trajectory onwards to Ab and G to complete the section. Given the quality of the singing and songwriting, producer Barrett Yeretsian sensibly appears to have focused his time and effort on those, rather than getting too hung up with expanding the arrangement or improving the realism of the programmed string parts. That said, the piano pushes a little beyond the earnest quarter-note repetitions that so many budding singer-songwriters get bogged down in, gently supporting the vocal. I’ve lost count of the demos I’ve heard prefacing the singing with a few such barren bars, but here the piano is stripped right back to place the vulnerability of the lead vocal centre-stage. Numerous understated figurations also provide musical punctuation between phrases, such as the rhythmic motif heard before “I learned to live” at 0:25 and 1:32, or the answering melodic fragments in the second verse at 1:12, 1:18, and 1:25. Even when the figuration settles into a more regular rhythm during the choruses, the balance between quarter- and eighth-note divisions remains fluid, giving some leeway to increase the intensity of the performance towards important section boundaries, and through the song as a whole. My favourite bit, though, is the mid-section, where the piano’s change of register and emphasised, oscillating, eighth-note pattern really help clear the air before the final chorus. These good things notwithstanding, I’m a bit underwhelmed by the vocal mix, as the tonality feels over-strident and a bit abrasive over the song’s duration, and I’d have welcomed more energy in the octaves either side of 1.5kHz. The vocal balancing also seems a bit rough and ready, with many consonants feeling over-emphasised (presumably on account of a hard-driven compressor’s fast release), yet despite this processing I get a sense of the singer travelling backwards and forward

in the mix. Take the section at 1:14-1:25 (“... if I am anywhere to be found, but I have grown too strong...”). In the first phrase, the section “to be fou-” seems to leap forward for a moment, before taking a big step back for the second phrase, and then hopping back towards you temporarily for “too stro”. To be fair, Perri’s wide tonal and dynamic variations would probably have required more than simple fader automation, but I reckon vocal ballads like this justify going that extra mile. Mike Senior

There’s lots to learn from Nicki Minaj’s rapping on this record, and from the brilliant daftness of the chorus vocal hooks, but what intrigues me most is the difference between the original version on the Now! 79 UK singles compilation, and the remastered version on her follow-up album Pink Friday. A/Bing both tracks into my DAW revealed that the album is roughly 3dB louder subjectively speaking. Even when you compensate for that, there’s a considerable tonal disparity: it feels as if the album version is down about 3dB below 200Hz, and is also tilted upwards from 1kHz by a decibel or so. Furthermore, almost all the vocals seem to be higher in level on the album version. To summarise, then: on the album, the track is louder, brighter, and less subby than on the single, and it has louder vocals. That’s exactly the opposite of what I’d normally expect! Single versions are usually loud, brash and vocal-dominated, whereas album listeners are treated to more lows, a greater dynamic range, and a smoother high end that’s better suited to louder, wide-bandwidth playback. So what’s the thinking behind this? My best guess is that the success of the original version took Minaj’s production team by surprise, and that in the glare of publicity someone decided that it didn’t sound competitive sonically alongside other artists’ records, especially in the light of its unforeseen pop-crossover appeal. If that guess is on target, those two versions may provide some practical insight into what the higher echelons of the industry do and don’t consider to be ‘radio friendly’. Mike Senior Mike adds more to his analysis of these tracks on our Mix Review web site thread,, where you can also contribute your own comments.

w w w . s o u n d o n s o u n d . c o m / January 2012



Big Fish Audio
I’ve yet to clap eyes on the Kurtis Chance Big Band Orchestra, but on the evidence of this new collection of 15 construction kits (comprising around 7GB of audio), there’s no doubt in my mind that they’re an absolute riot! It’s rare to hear so much fun being had on a sample collection, and this big band ensemble are clearly breaking a serious sweat to entertain us, producing a series of fantastic, swaggering performances. I’ve nothing but praise for the stylish arrangements too, which incorporate lots of great dynamic changes and variations that drive the track forward and keep you coming back for more. The musical and intelligent drumming is also commendable, especially when twinned with such muscular, funky bass parts. Styles range all the way from classic ‘40s swing, through blues (John Belushi would be proud!) to funk, with the odd Latin and Dixieland diversion along the way. The tempos are medium to uptempo, nominally 91-193 bpm for the Acidized WAV and Apple Loops formats, but there is, of course, some flexibility, given the inclusion of REX2-format versions. Each kit provides around 10 interchangeable song sections, each of which is split into a series of loop layers isolating the drums, bass, guitar, piano, saxes, trumpets, trombones, and any other parts, so that you have separate control over them. The brass are presented dry in mixed sections, typically, although a strong selection of solos is provided too. Unusually

Big Bad Horns


for a construction-kit library, multitrack drum parts (kick, snare, hi-hat, overheads and toms) for all the loops are included, although only in WAV format, which means a minor headache if you need to beat-match them. Indeed, coupled with locked-in harmonies and arrangement details, this means that Big Bad Horns clearly isn’t the kind of library you want to use after you’ve already started a production; it’s much better to work with it right from the start of your creative process. Sonically, however, it’s a bit of a mixed bag. Although the majority of the mixed drum loops are pretty serviceable, some sound rather more lo-fi than I’d wish, and the toms feel rather over-balanced on a number of occasions. The multitracks fare similarly: there’s a good measure of decent capture, but I did find myself sucking my teeth at trashy hi-hats, sqwanky snares, and muffled overheads on more than one occasion. Likewise, although the electric bass parts have much to recommend them in terms of solid low end and effortless mid-range cut-through (a well-judged instrument tone there), they are not without blemishes, such as obtrusive background buzz or the odd unattractive, booming unevenness. Uncontrolled resonances are even more of a concern with the less frequent upright bass parts, and here the tone is also not very well suited to purpose, easily becoming submerged in the mix. Fortunately, the guitar and piano sounds are respectable and work well in the arrangement, while the brass — the stars of the show, after

all — are shown off to pretty good advantage, which gives you more chance of sweeping any other sonic shortcomings under the carpet. Although it’s a bit disappointing to encounter engineering issues, given the higher-than-normal pricing here, there’s so much energy and enthusiasm in these performances that I think they offer reasonable value for money, nonetheless, if you need raw materials for building this kind of music from scratch. Mike Senior $129.95


Indie Rock
As with other titles in Ueberschall’s growing catalogue of Elastik-based collections, this is essentially a loop-based library organised into a number of construction kits, all presented via the small (but very efficient) Elastik front-end. In terms of pure numbers, the library comes in at just under 4GB, with 1500 loops divided between 18 construction kits. The loops themselves are dominated by drums, guitars and bass, but the occasional keyboard loop is also thrown in. Two very welcome features of the construction kits are worth noting. First, while pre-mixed stereo drum loops are provided, each drum loop is also presented as a series of multitrack loops (snare, kick, hi-hats and overheads, for example). These mean that you can remix the kit to your own taste or drop some elements

out for added variety, although do note that the recordings are from a multi-miked drum kit, so there is spill between them. The second feature is the inclusion of an excellent range of song sub-sections in each construction kit — intro, verse, chorus, bridge (or breakdown section) and outro — with the last of these featuring a proper ending. Many construction-kit libraries don’t provide this kind of detail, so top marks to producer Kai Reuter! Indie rock is a pretty broad field but the material here got me thinking of a number of fairly mainstream bands. For example, the kits titled ‘Bittersweet’ and ‘Lost You’ showed a touch of Kaiser Chiefs-style humour, while ‘Backtrack’ could have come from a Feeder album. There were other nods towards My Chemical Romance (in their more pop-orientated moments), Fall Out Boy, Lostprophets, Mexicolas, Paramore and even a hint of rock-flavoured Blink 182 pop/punk (the ‘Run’ kit, for example). In terms of recording quality and sound, the library is also 5 spot-on. The drums and guitars are particularly impressive, the guitar sounds ranging from grungy to glassy, and the playing is really good. Most welcome are the choices available in each song sub-section. The same basic chord pattern or riff is often supplied with different guitar tones or performances, so you can build plenty of variety into your compositions. If anything, this collection makes things too easy! Building a very credible backing track from any one of the construction kits is


January 2012 / w w w . s o u n d o n s o u n d . c o m

a complete breeze, simply because everything is provided to make this as straightforward as possible. For instrumental composers requiring 60 seconds of ‘indie rock’ to meet a tight deadline, this is ideal. However, stick some suitable vocal talent over the top of one of these construction kits and you could easily create a Kerrang-friendly indie track. Within the obvious limitations of any construction-kit-style sample library, Indie Rock is top notch. John Walden $113.56


Broken Wurli
Despite never quite attaining the hip status of the Fender Rhodes, the Wurlitzer 200A electric piano has been a favourite with keyboard players since it first appeared on pop hits of the ’60s. So, when Massive Attack’s producer Neil Davidge told Soniccouture he had “an old Wurli with a dodgy distorting speaker that sounds great”, the UK samplists 5 were quick to negotiate a loan. A Massive sampling session then ensued… Calling the resulting 8.5GB library ‘Broken Wurli’ doesn’t really do this instrument justice, and we should be thankful that Soniccouture haven’t applied the same marketing terminology to a used car business. The piano’s left speaker is indeed, as we say in the trade, f***ed, but its other speaker is as good as new. The producers miked up both and also took a DI

feed, so you have a choice of super-clean, moderately lo-fi and distorted signals. With nine velocity layers, three round robins and an accurate emulation of the trademark Wurli tremolo, this is a forensic sampling job that captures the instrument’s tone all the way from demure, mellow, harp-like tinklings to strident, reedy stabs. I found the basic instrument’s key-off samples to be over-loud, but was able to quickly fix that by adjusting one of the nice, big ‘virtual shiny metal knobs’ that grace its elegant user interface. Other than that minor issue, the default patch (which features the DI and clean-speaker feeds panned respectively hard left and right) is enormously playable and touch-sensitive, with no noticeable latency. Unlike the Rhodes Suitcase model, the original Wurlitzer 200A had no stereo tremolo/vibrato option, but this sampled version does, courtesy of a splendid auto-pan effect. This can be combined with excellent built-in chorus, auto-wah, phaser, Leslie cabinet, stereo delay and convolution reverb effects, to produce glorious, lush electricpiano washes. The producers go to town with these effects in their sound-design section, providing heavenly, ambient, clean textures as well as spiky distorted timbres and funky ’70s soul patches. Funnily enough, I found the ‘broken speaker’ effect to be one of the less engaging aspects of the library; its fizzy, fuzzbox-style distortion certainly adds bite and drive to single-note lines, but on sustained chords it tends

to sound merely synthetic. Because each distorted note was sampled separately, you don’t get the cumulative effect of multiple notes simultaneously overdriving a speaker. However, that racket can be simulated fairly well by dialling up the onboard distortion and amp-simulator effects. 5 Thanks are due to contemporary sound companies whose diligent, highly detailed sampling jobs preserve iconic instruments of yesteryear for future generations. We still have an original Wurlitzer 200A in our house, but haven’t recorded it this century — and if you notice its distinctive tones on a future track bearing my name, it’ll probably be these samples you’re hearing. Dave Stewart DVD $99, Download $89


Number One Hits EZX
EZ Drummer/Superior Drummer 2 Expansion
As off-putting as I found the name of this drum expansion library (I’m pretty sure that a sample library alone isn’t going to bring me a number one hit!), I have to say that I was quite taken with it. In terms of the drum machines sampled here, there’s nothing like the comprehensive list that you’d get from, say, Best Service’s Drums Overkill, but the quality is universally good, the MIDI side of things is well thought out, everything’s well organised and easy to access and the processing twists (that

were reportedly done by Niklas Flyckt of Britney Spears’ ‘Toxic’ fame) really do breathe new life into the familiar old sounds. Flyckt has also apparently donated some of his own sample collection, so there are a few sounds here you won’t get anywhere else. Sonically, it’s a flexible library and it should lend itself to any genre using electronic drums, from pop and electronica, through house and techno to hip-hop and its myriad related genres. All samples have been processed to some extent, via an impressive recording chain (according to the published list, though who really cares what it was recorded through, as long as it sounds good?), but there are both dry and reverbed versions of most sounds. That’s a nice touch, because the reverbed versions are well-judged and make for some great instant satisfaction if you plan to build your beats first and then construct your track around them, while the dry ones either allow you to go for a drier sound or put more work in to get the ‘right’ ambience for your track. Given the price and quality on offer, I’ll give a five-star rating, although it would really be nice to see a little more content here, and it seems a shame that you have to have EZ Drummer to use this library; there are an awful lot of Kontakt and EXS24 users out there who I’m sure would like to be able to add these sounds. If you do have EZ Drummer or Superior Drummer, though, and you need a palette of electronic drums, I don’t think you’ll be disappointed with this. Matt Houghton €69

w w w . s o u n d o n s o u n d . c o m / January 2012



Readers’ Music Reviewed
and some of the guitar solos are a bit roomy on this EP, as if the recording space could have done with a little more damping. ‘Planet Of Love’ has the best mix of the lot, boasting the greatest degree of clarity. J G Harding W

Janne Lappalainen, or Lapplander to his fans, is a difficult musician to categorise. Perhaps the easiest way to go about it is to think of him as a one-man band who happens to own some expensive recording equipment, but isn’t always sure how to use it best. Musically, the tracks veer from eerie space music — which reminded me of the scene in The Man With Two Brains where Steve Martin does crazy brain operations — to the electronic beats you hear in the classic 1982 film Tron, during the ‘light bike’ chase scene. If you’d enjoy being locked in a room listening to the mind of a potential genius (who probably needs to learn a few more techniques for recording electronic music), you will probably love Lapplander. Personally, I think he needs to develop his production skills further before releasing another album to the world. That said, I fully expect to hear of him writing the soundtrack for Back To The Future 2015. Sarah Bowden W

Syd Arthur
Syd Arthur’s music brings to mind early ’70s progressive rock, fusing Led Zeppelin, funk, jazz and feel-good harmonies. The result is quite ‘summery’, and gives the impression that the musicians are really enjoying themselves at a festival, jamming to a field full of cider-addled punters. Syd’s vocal is quite individual, with a clear English accent in the lower registers and a kind of classic rock tone when he sings at a higher pitch. All the instruments are played to the fullest, with some blazing violin and mandolin from the wonderfully named Raven Bush, as well as a crazy flute solo on one of the tracks. The songs themselves aren’t that ‘hooky’ and sound like they’re the result of full-band jams rather than songs that have been brought to a band, then arranged. The recordings are perfectly listenable: they don’t set the world alight or have a particular production sound, but they document a band that likes playing together live. Still, a little more effort on the production side would bring the energy the band have to the fore. The drum sound

Lelio Padovani
Electronic EP
Lelio Padovani stands out among the sort of people who send in instrumental electronica for review, mainly because he seems to have a grasp of concepts like suspension and inversion, and as a result, his chord progressions actually progress. In a world where the thought processes of many wannabe producers run along the lines of “Maybe I’ll just loop that C major arpeggio for another 64 bars,” that sort of old-fashioned musicality makes a refreshing change. The flip side, though, is that this musicality is paired with an equally old-fashioned attitude to synth programming and sound design. Despite having produced his music entirely in the box, Padovani attempts little that couldn’t have been recorded in the mid-’80s, and rarely departs from a template which could be summed up as “Vangelis, but with more widdly guitar”. It would be nice to hear him attempt something that is sonically a bit fresher and more original. Sam Inglis W


Audnoyz Project Vol 2
The mysterious “master musician” behind Audnoyz is nothing if not ambitious. The aim behind his second full-length collection of material appears to be to include absolutely everything. Upright bass, techno beats, guitar feedback, Indian and African instruments: all are grist to his mill, and they are often overlaid with fragmentary, heavily processed vocal samples. In fact, little here is immune from heavy processing, and one of the most impressive aspects of the album is Audnoyz’s ability to retain a focus on the bigger musical picture while throwing himself into some very detailed sound design. There’s not much here you can dance to, but there are plenty of hidden depths in which you can immerse yourself. Sam Inglis W


January 2012 / w w w . s o u n d o n s o u n d . c o m

It should be a stress-free ascent, but our Apple guru finds himself confronted with all kinds of complications when moving to the Cloud...

confession: for some time now I have been spending my time between two operating systems. While waiting for Lion to mature from a newborn cub to the adolescence of 10.7.2 (brimming with the enthusiasm of youth, occasionally petulant but, contrary to type, happy to wake from sleep on demand!), I have been keeping a venerable install of 10.6.8 going for day-to-day administration tasks and using Pro Tools 9.


To The Cloud!
However, now that Pro Tools 10 is fully qualified with Lion, I’m moving on and, as part of that transition, I decided to embrace iCloud and all that it has to offer. For newcomers to Macs, the process of setting up hardware and an Apple ID for the first time is designed to be blissfully simple. But for the rest of us, moving a legacy of multiple Apple IDs, mail accounts and personalised workflows raises a few issues. I have (or had) a MobileMe account and a second Apple ID (a non-Apple email address) that I use for iTunes Store purchases. I could have associated either with iCloud (or even set up a new Apple ID), but I decided to go with the MobileMe account, as that would give me 25GB of cloud storage (until next June) instead of the usual 5GB. For MobileMe users, transitioning to iCloud needs to be activated at Plenty of warnings are offered during the process, and rightly so. The iDisk, iWeb and Galleries functions are not supported by iCloud, but will continue

to function until next June; plenty of time to forget to back up all that critical data you spent hours uploading to the world’s slowest online storage solution! For Calendars and Contacts, you are advised to back up before making the transition. Don’t ignore this advice: in my case, Calendars moved to iCloud without a hitch, but Contacts did not, so having the archive to hand was a lifesaver! You can back up both iCal and Address Book by going to File / Export and selecting the relevant option. For the Mail app, it’s always worth archiving mailboxes as part of a backup routine. Be aware, however, that MobileMe mailboxes set up in Snow Leopard’s Mail app will stop working after transition and, no matter what you read on discussion groups about resetting passwords and recreating mailboxes, you will not regain their functionality. Other mailboxes for POP3 and Exchange accounts remain unaffected.

that is independent of iCloud, and is activated on a Mac by signing into iTunes with the Apple ID that you use for iTunes purchases. You can use the same ID on iOS devices by going to Settings / Store. On the Mac, this allows you to download previous purchases (music, apps and books) by choosing ‘Purchased’ from the Quick Links on the right of the iTunes Store homepage. In order to download all previous purchases, select one of them from the list and the ‘Download All’ button will appear at the bottom of the page. To enable automatic downloads of purchases

Purchases you make can be automatically downloaded from iCloud to your other devices.

Getting In Sync
One reason you may feel the need to switch to iCal is to access iTunes In The Cloud and iTunes Match. As this is a common source of confusion, let me clarify that iTunes In The Cloud is a service

made on other devices, go to iTunes / Preferences / Store. On an iOS device associated with that Mac, automatic downloads are activated in Settings / Store. Non-iTunes purchases can be accessed by syn’cing — no change there, then — but this can now be achieved wirelessly by going to Settings / General / iTunes WiFi Sync.

No Match
The service I was hoping to access was iTunes Match, in order to legitimise my music collection by paying an annual subscription — sorry, what I meant was: obtain high-quality, non-DRM AAC versions of all the music I have transferred to my iPad from CDs I own, in order that I can then stream those tracks on demand to all my other devices — but I can’t tell you very much about this yet, as it is only available in the US at the time of writing, and this situation may persist well into the new year. The same goes for volume licensing when purchasing multiple copies of iOS apps!

You can sign in to iCloud from Lion using System Preferences — but are you prepared?


January 2012 / w w w . s o u n d o n s o u n d . c o m

Sound On Sound editor Paul White delivers the definitive guide to recording and mixing in the project studio.
Featuring 350+ full-colour pages packed with pro techniques, practical photos, detailed illustrations and hands-on walkthroughs, The Producer’s Manual brings together everything you need to take a mix from initial recording to final master, including: • All you need to get great recordings: from vocals and drums to guitars, bands and acoustic instruments. Choose the right mic, review classic recording techniques, learn how to tame spill and get the most from performers. • In-depth 101-style guides to dynamics and compression, reverb, pitch correction, studio acoustics, monitoring and more. • Taking your mix to the next level: explore the techniques and the pitfalls. Essential jargon-free theory backed by practical insights on everything from EQ through mixdown approaches to classic hardware profiles. • How to master your own material when the budget doesn’t stretch to professional mastering.

“Stacks of information, top tips, problems and solutions make this book a joy to read. 10/10.” Music Tech. “If you can’t make a decent sounding recording after reading this, the problem is with you, not the book. Highly recommended! 10/10.” Waveformless “An indispensible book that anyone involved in music production, recording or mixing should own. 5/5.” Sounds and Gear
















Pro Samples/Books/Apps/Courses/Events

With expected component shortages and the consequent rise in hard-drive prices, what can the PC musician expect from 2012?


n the current economic climate, punters are, perhaps understandably, more cautious about splashing out on a new PC so, in the UK at least, some distributors have ended up with high inventory levels (although perhaps not quite the ‘PC mountain’ that some industry pundits have suggested). Sadly, some of these distributors are probably now relieved that they do still have plenty of stock, since the dreadful flooding in Thailand has caused hard-drive prices to skyrocket (Western Digital and Toshiba have been hit particularly hard by the floods). There are warnings of possible PC and associated component shortages at Christmas, as well as going into 2012.

Bargain Beware
For those that do take the plunge with a new computer this Christmas, notebooks are now by far the most popular choice, accounting for over three times as many sales as desktop PCs. However, in an effort to boost Christmas computer sales, as I write this in mid-November some of the largest retailers are already offering sizeable discounts on notebooks. Unfortunately, as I’ve said many times in this column, buying a notebook PC for low-latency audio use is nowadays a real gamble unless you go to one of the specialist retailers, so tread very carefully if you find yourself tempted by an apparent bargain. The third most popular computer format choice at the moment is the touchscreen tablet. Apple’s iPad range is still the leader in the popularity stakes,

although the release of Windows 8 in 2012 is expected to result in a shift in emphasis in the tablet market. We’ll just have to wait and see whether it heralds a new era for the PC platform. Meanwhile, one casualty of the shortages and notebook discounts is the new slimline ‘ultrabook’ laptop format that I first discussed in PC Notes October 2011, which is currently deemed desirable, but too expensive. Some expect these new contenders to steal a sizeable chunk of the tablet market, but only if they drop significantly in price. Along with the recent 20 percent increase in the Chinese minimum wage (which, while laudable in itself, will undoubtedly mean further computer price increases across the world), it looks as if 2012 is going to be an interesting year for the PC platform, in more ways than one!

pocket-money product from newcomers Klanghelm ( the VUMT (VU Meter and Trim) plug-in. There are alternatives available with similar switchable VU/PPM meter ballistics, such as PSP’s freeware VintageMeter, but what I love about VUMT is its smooth and accurate needle animation, its additional ‘hold needle’ displaying the current maximum VU reading, and its amber/red

Are you lost without VU meters? Klanghelm’s VUMT is a versatile bargain with plenty of different applications.

A Room With A VU
Closely monitoring peak audio levels is critical to avoid nasty digital clipping, which is why most DAWs offer peak-reading meters. However, when setting up mix balances or using the latest ‘console colour’ and ‘analogue warmth’ plug-in simulations, you should ideally be watching a VU meter that displays more average levels instead. This will both help your mix balance and achieve suitable gain staging for your ‘analogue’ plug-ins, so they better emulate the hardware that inspired them. Most such plug-ins include some sort of metering, but they rarely offer ballistics (the rise and fall times of the level pointer) that match real-world VU meters. Enter the first

clip LED that illuminates at a definable user level and clip point, so you can simultaneously keep an eye on peak levels. The extremely handy +/-20dB gain trim is also perfect for your gain-staging experiments: just insert one instance of VUMT before and after your plug-in, with complementary trim settings (such as -10dB and +10dB, respectively) and you can quickly alter its ‘drive’ level for more or less warmth. VUMT has four switchable skins, is supplied in 32-bit and 64-bit stand-alone and plug-in versions, in both single- and double-meter formats (the latter offering stereo linking and M/S metering options) and, best of all, costs just €6.28, making it an impulse purchase. Highly recommended!

Cram It With RAM!
Back in PC Notes June 2011, I discussed the advantages of Intel’s then forthcoming Z68 chip set, which was more overclockable than the H67 and more versatile than the P67. However, six months is a long time in the PC world, and by the time you read this Intel’s even newer X79 chipset will have been available for a few weeks. Ironically, this component costs motherboard manufacturers 50 percent more than the Z68, yet offers almost no new features, apart from the almost inevitable new format of CPU socket (LGA 2011 to replace LGA 1155) to partner the latest Sandy Bridge-E CPUs. Nevertheless, this release offers positive news for the professional musician. LGA 2011-based motherboards will have either four or eight DIMM sockets, each of which can support up to 8GB of DDR3-1600 RAM in quad-channel format. So anyone with huge sample requirements (such as those running the Vienna Symphonic Library) will be able to install up to 64GB of memory without breaking the bank. Meanwhile, the Z68 chip set is expected to die out quietly. It’s a tough world.

What features a jaw-dropping 2011 protruding pins and supports Intel’s new Sandy Bridge-E processors? Yes, it’s the new LGA 2011 socket.


January 2012 / w w w . s o u n d o n s o u n d . c o m



Is s the quality of S/PDIF connections on soundcards variable?

Is there any difference between the quality of S/PDIF connections on low-end and high-end soundcards, or am I right in thinking that a low-end card with S/PDIF I/O (and the ability to clock from the A-D converter) should be adequate? Via SOS web site SOS Technical Editor Hugh Robjohns replies: In theory, S/PDIF is quality

Via SOS web site SOS contributor Martin Walker replies: I doubt that you’ll hear any difference in practice by increasing the bit depth from 16 to 24. As long as you leave a few dB of headroom to give your MP3 encoder some ‘space’ to perform a clean result, the main decision to be made with MP3s is the target bit-rate. MP3 files can be created at CBR (Constant Bit Rate) values from 8Kbps to 320Kbps. Spoken word is still perfectly intelligible down to about 24Kbps, which is usually perfectly sufficient for podcasts, talk radio, and so on. Solo acoustic music performances could be acceptable at 48Kbps, although 64Kbps is probably more in line with AM radio quality.

independent, assuming that the physical interface is engineered reasonably in the first place. It is purely about transferring the data — there’s no jitter to worry about — so, provided you have decent 75Ω cables of modest length, it should just work. I’ve had very few problems with S/PDIF interfaces, and the few issues I did find were actually caused by ground loops. Personally, I prefer AES3 interfaces, because they will cope with longer cables and are always ground-free, transformer-coupled connections (often S/PDIF is as well, but not always). And XLRs are so much more reliable than RCA phono plugs!

If a soundcard is well made, its S/PDIF interface should be quality independent, so the difference between low-end and high-end cards should be minimal in this respect.


Can you help me with MP3 file conversion?

Can you explain a few things about creating MP3s? I’m currently converting WAVs to 24-bit, 44.1kHz and then converting them to MP3, but I’m not entirely sure what effect this kind of conversion has on the sound. Will my method have a higher-quality outcome than 16-bit WAVs converted to MP3?

For reasonable-quality ensemble music, many people consider 128Kbps a good baseline, especially if the intended destination is computer speakers or in-car audio systems. However, when listening on a hi-fi or on studio playback gear, many musicians find 128Kbps difficult to listen to, especially since the frequency response falls off rapidly above 16kHz, high-frequency sounds such as cymbals sound distinctly harsh, and you can often hear a low-level background ‘warbling’ sound, which is the main reason that some people dislike this rate. If you’re looking for the best compromise for your MP3 files between compression ratio and audio quality, bit-rates of 160Kbps or 192Kbps are generally recommended, with 192Kbps, in particular — often being classed as ‘near CD’ quality — suitable for complex music or tracks with lots of bass content. Only on expensive playback systems can most people tell the difference between 192Kbps and CD quality.

Further up the scale, if you want some compression but minimal degradation in sound, 256Kbps is a good compromise compared with CD audio, since the frequency response is generally identical to the original up to about 18kHz, and the difference between the two is barely discernible by most people, even on high-end systems. For ultimate MP3 quality, you could choose 320Kbps, but so few people can hear the difference between this and 256Kbps (or real CDs, for that matter) that it’s generally a waste of disk space. Of course, the main point of all these conversions is to reduce file size, and most MP3 encoders also offer a choice of VBR (Variable Bit Rate), in which the bit rate is altered dynamically during your track. Because VBR can rise during complex passages and drop during simpler sections, with some material it can sound significantly better when compared to a similarly sized CBR file, and instead of numeric values you may be offered a quality setting anywhere from ‘highest’ to ‘lowest’. However, VBR is rarely used for online audio streaming because its constantly changing data stream encourages glitches and errors, as it does on some older MP3 players, particularly when fast forward or rewind controls are used. The above guidelines are fine for the average punter, but as musicians, what should we really be listening for when deciding on bit-rate? Well, because of the way MP3 encoding relies on one frequency ‘masking’ another nearby at a lower


MP3 encoding reduces file size partly by Frequency Masking (discarding information that is unlikely to be heard because of nearby louder tones), but it can be fooled by some types of music, such as gliding or pure tones.

level, any instrument that glides from one frequency to another (such as fretless or acoustic bass, guitar whammy-bar excursions, Theremin or trombone solos) may result in audible artifacts, so listen out for these and use a higher setting if required.

w w w . s o u n d o n s o u n d . c o m / January 2012

Another killer combination for the MP3 encoder is a pure solo tone, such as a long, high vocal or flute note, or guitar feedback tone, with complex but quiet instrumentation behind it. Listen out for distortion or general fuzziness where the encoder has decided that parts of the background instrumentation are redundant: they might be with a rock band and screaming guitar solo, but not with a quartet featuring a flute solo. Ultimately, though, all these choices pale into insignificance if your MP3 files are intended for online streaming. Many sites, such as Soundcloud, YouTube, and so on, convert incoming audio to their own chosen format so, unless you’re offering downloadable MP3 files, you’re often stuck with whatever quality choices these other delivery sites choose for you (typically around 128Kbps).

Top: The arrangement our reader is currently using to produce a wider stereo image, which results in reduced bass on the right-hand side. The fake stereo image arises because some frequencies are stronger in one side than the other, due to the offset comb-filtering resulting from combining the original and delayed signals with the same and opposite polarities. More ‘fake Side’ (‘S’) level results in a wider perceived image. In the lower diagram, the fake ‘S’ signal has been high-pass filtered, avoiding bass cancellation in the right-hand side.


How can I create a ‘fake’ M/S setup that is mono compatible?

I thought I had the whole M/S thing down until I listened to a commercial record and realised their stereo image was wider than mine and yet still perfectly mono compatible! In order to convert a mono source to an M/S pair, I bus the source audio to two separate tracks. On one track, the audio is unchanged and routed to the stereo bus centre (which I label ‘Mid’). The other I delay by around 10ms, then split it to the left and right stereo bus, with the right side inverted (I label this track ‘Side’). The ‘Side’ track cancels when I sum to mono. The problem I’m having is that the stereo image is not very wide. While it is clearly in stereo, it does not reach the extremes of the stereo field as it would by utilising the Haas effect. When I use a simple Haas trick, I achieve the width I desire (ie. a hole in the middle), but it is not acceptably mono compatible. Is there a trick I am missing to achieve the width that I desire in my pseudo-M/S setup, yet also maintain mono compatibility? Via SOS web site SOS Technical Editor Hugh Robjohns replies: When you use this method of creating a fake stereo signal from a mono source, the apparent width is determined entirely by the amount of Side signal relative to the amount of Mid signal. No Side means mono. Loads of Side means wide perceived stereo image. Too much

Side signal means a hole in the middle and quiet mono! So you should be able to make the track as wide as you want — even to the level of a hole in the middle — just by further pushing the level of the Side signal. Changing delay time also affects perceived width and size. Larger delays (30-70 ms) create more hall-like effects, while shorter delays (5-30 ms) are more subtle and less ‘roomy’. However, this kind of M/S-based fake stereo is never as convincing as real stereo. You inherently end up with a mush of frequencies spread across the sound stage; your original mono source is spread across the image like butter on bread. There is no discrete spatial positioning, and no coherent imaging. Basically, it isn’t real stereo, it never can be real stereo, and comparison with a real stereo recording is pretty pointless and always disappointing! Moreover, created in the way you describe, the stereo image will tend to be bass-heavy on the left-hand side, because the relatively short delay you are using will tend to allow low frequencies to sum in phase on the left and out of phase on the right. This can be cured by inserting a high-pass filter before (or after) the delay line to remove bass from the Side signal. Set it to about 100-150 Hz to ensure the bass content stays central. The Haas effect — which you can employ by panning the original source to one side and a short-delayed version of it to the other — can sound very wide indeed, but usually isn’t very mono compatible. The only way to make a single source fill a stereo sound-stage

in any kind of convincing way — in my humble opinion — is to record it in stereo in a decent-sounding acoustic space, or use a really good reverb processor to achieve a similar thing. There are other techniques that can be used to create pseudo-stereo effects using dedicated stereo-width enhancers and variations on the distortion and chorusing theme, but they all tend to change the tonality to some extent, which may not be what you’re after.


Do mixes benefit from low-pass filtering at mixdown?

I’ve heard a lot about high-pass filtering tracks to reduce clutter at mixdown, but not as much about low-pass filtering in this context. Would mixes suffer or benefit from doing the same at the opposite end? For example, would it be easier to bring out ‘air’ in a vocal if other parts were low-passed? Via SOS web site SOS contributor Mike Senior replies: Particularly in small-studio environments where the low-frequency monitoring fidelity is questionable, there’s a lot to


January 2012 / w w w . s o u n d o n s o u n d . c o m

Q&A is sponsored by SE Electronics. Each month, SE are kindly donating an SEX1 large-diaphragm condenser microphone, worth $249, for the best question.
Although fairly systematic high-pass filtering is very sensible in home-studio mixing, as you can see in this screenshot from a recent Mix Rescue project, it’s rarely beneficial to apply low-pass filtering in a similar way.

be said for high-pass filtering in a fairly systematic way to head off problems at mixdown. However, widespread low-pass filtering offers fewer benefits, simply because so many instruments in a mix will have harmonics and noise components that extend right up the spectrum. In practice, I find peaking/shelving cuts are, therefore, more appropriate for dealing with typical mixdown tasks, such as frequency-masking problems. Yes, in theory you could make your lead vocal sound airier by low-pass filtering the other parts, but you’d still have to consider how the mix as a whole will sound during moments when the vocal isn’t active, so achieving an airy vocal in practice isn’t usually as simple as this. Having said that, there’s nothing wrong with low-pass filtering if you really want to kill the high frequencies of an instrument for balancing reasons. I would most commonly do this with amped instruments, such as electric guitars, which are capable of contributing a lot of undesirable amplifier noise in the top two octaves of the audible spectrum. However, this has to be evaluated on a case-by-case basis, because it’s very easy to dull the overall mix if you’re not careful.

me to get for that? It would be great if you could give me some information on setting up and using System Link, as there is very little on the Internet to help! George Morton via email SOS Reviews Editor Matt Houghton replies: Personally, I wouldn’t recommend using System Link, as there are alternatives around that allow you to link two machines without requiring a second audio interface. The one I have most experience with is FX Teleport, by FX-Max (www. There’s a free demo that you can try using any type of network connection, including USB and Firewire, but if you decide to use it you’ll get better results from a faster network connection, such as Gigabit Ethernet. The one additional piece of hardware I’d recommend investing in is a KVM box, which allows you to use a screen, mouse and keyboard with multiple computers. That’s pretty much essential when working in this way. I’d thoroughly recommend FX Teleport if another computer is the answer to your problems. Before you invest, though, do make sure that it is more CPU power that you need. Availability of memory, or hard-drive loading, could also be a cause of problems. It could be that, for example, results are limited by your hard-disk performance, particularly if you are running an operating system, audio files and streaming sample instruments all from the same disk, in which case running those three things from separate drives might help. Memory is often not a huge problem area, although it can be an issue on some 32-bit systems, particularly where you


Could I use Cubase System Link to slave my VST plug-ins?

I’m looking to set up another computer alongside my main computer to slave all my CPU-draining VST instruments and effects. I’m currently using Cubase 5, but looking to upgrade to 6 soon, and would like to utilise the VST System Link option. After doing some Internet research, I know System Link will only work with certain audio interfaces. I have a budget of around $500 to buy the two interfaces, and I’m wondering what you could advise

have a lot of hardware installed. First, there’s a maximum of 4GB available in Windows XP 32-bit, of which only 2 or 3 GB is available to each application — and all of the plug-ins running within Cubase count as one programme! On my old XP system, I had 4GB of memory installed, but had only 2.3GB available to applications, due to the way in which Windows allocated memory address space to my various DSP cards. In 64-bit versions of Windows, this limitation is removed. You might run into problems with older 32-bit plug-ins if you try to run the 64-bit version of Cubase, but in my current system I’m running 32-bit Cubase on Windows 7 64-bit, with the JBridge utility allowing me to run 64-bit plug-ins (such as Kontakt) in their own address space. If you haven’t tried it yet, I’d also suggest experimenting with Cubase’s Freeze facility, which allows you to ‘freeze’ audio and instrument tracks. This essentially performs a temporary render of those tracks and unloads any plug-ins to free up precious computer resources. You can unfreeze at any time if you need to go back and tweak, and even when frozen you still have access to features such as level and pan automation. Finally, another option is to consider upgrading to a modern multi-core PC. That might not be quite within your budget, but if you’ve not upgraded for a few years, you’ll be amazed at how much more you can do in a single system. One advantage is that you’ll only have the noise of one machine to put up with, because remember that the more computers you have running, the greater the sound of whirring fans will be in your studio!


w w w . s o u n d o n s o u n d . c o m / January 2012

Is the customer really always right?

hen a band walks into the studio and utters the words “we want to sound like Radiohead”, I experience a shiver of discontent and an uncontrollable rolling of the eyes, and I take a deep breath. Then I offer the applicable, if not appropriate, retort of “Who wants a cup of tea?” This first meeting sets the tone for any recording session: it gives the producer an idea of what they’re working with and trying to achieve, as well as just how much extra work will need to be put into keeping the artist happy. I know that exposure to egocentric megalomaniacs is an occupational hazard in the music industry, but am I alone in thinking that encouraging their delusions is getting pretty old? I didn’t sign up for giving a daily ego massage to every skinny-jeaned, tightly coiffured and socially misunderstood musical aspirant, but when I do find myself in that position — which I invariably do — I would at least like it known that they’re not given out until the band put in a decent amount of effort. Within the business that is ‘music’, as I have observed it, there are two dominant behavioural patterns: arrogant and subservient. The dynamic between the two dispositions
Adam Audio AKG Acoustics Alan Parsons Alesis / Numark Industries Argosy Console Audio Engineering Associates Avid / M-Audio BeesNeez Benchmark Berklee College of Music Best Service Big Fish Audio CASIO America Clearsonic Earthworks Emotiva Eventide Fingerprint Audio Focal USA Focusrite Full Compass Grace Design Grimm Audio (Dist Grace Design) Great River Electronics Guitar Center Professional JBL Professional JZ Microphones Kurzweil Liverpool Performing Arts


is especially apparent between a recording artist and their producer. The balance can alter many times during a session, the changes often spurred on by the emotional frailties of each party, but whose responsibility is it to swallow their pride and back down for the sake of the session? Having spent an unfortunate portion of my life working in the hospitality trade, I’m well aware of the theory that the customer is always right, and I understand that it makes perfect business sense to create a comfortable and hospitable working environment for your clients. But does one party have to feel more important than the other? Isn’t mutually beneficial gain better than self satisfaction? It seems counter-intuitive to bring hostility into the creative process, but perhaps I’m in the minority for thinking so. I’d like to feel that I have my head screwed on pretty tightly when it comes to dealing with clients. People, with the odd exception, don’t often surprise me, and although choosing to live in London certainly compounds my cynicism, I think that for all the emotional walls I come across as a producer, I can generally work out what ‘The Talent’ need in order to perform at their best. When it comes to buying a round of beers for the band, to try and loosen the singer’s vocal
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delivery, I’m your man! But if Rebecca Black had strolled through my door with a lyric sheet of Gregorian calendar days and teenage car-pool seating arrangements, I think I would have been eating pot noodles that month, if you know what I mean. Personally, I don’t dislike working with people. In fact, I spend so much time staring at graphical representations of waveforms that the ‘human factor’ is what keeps me somewhat sane. But I have found that many of my engineering colleagues have struggled to create a rapport with their clients simply because they won’t assume a subservient role. They have an old-school methodology placing them at the top of the social ladder, the kings of their respective castles, live or in the studio. We all desire recognition and we look for it in different places: artists still clamber up onto a stage and vie for the adulation of the crowd, but gone are the days of the awe-inspiring and dominant producer. It seems to fall to those behind the scenes to be of a fickle and malleable disposition, to be not only a skilled engineer but also a fan, confidante and friend of the artist. To those ‘yes men’ that use agreement as a quick fix, I’d say that sometimes you have Lynx Studio Technology McNally Smith College MicW Native Instruments Nova Musik Novation Music Primacoustic Project Sam Radial Engineering Royer Labs SAE Sample Magic Samson Schoeps Mikrofone Shure Sonnox Ltd Solid State Logic Sound Pure Stedman Corporation Steven Slate Drums StudioPros Sweetwater Sound Synthogy (Dist Ilio) Telefunken Trident Audio Development Universal Audio Vintage King

About The Author
Drew Bang (www.drewbang. com) is a freelance producer, sound engineer and composer. He lives in East London and is intolerant of studio-based intolerance.

to be cruel to be kind. In fact, I can’t help but think that if Amy Winehouse, for example, had been surrounded by more people who challenged her, rather than tolerated or encouraged her destructive persona, a great talent would still be with us. The truth is that a producer or sound engineer at any level would probably benefit more from a crash-course in practical sociology than from attending a seminar on advanced Pro Tools key bindings.
If you would like to air your views in this column, please send your submissions to or to the postal address listed in the front of the magazine.

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