Resolution V6.2 March 2007 by Resolution

AUDIO FOR POST, BROADCAST, RECORDING AND MULTIMEDIA PRODUCTION

V6.2 MARCH 2007

Mike Oldfield — On changing his working methods Can the Dolby LM100 do for TV what Leq(M) did for cinema? Latest generation Internet apps are transforming music discovery Acoustic Energy’s successor to the NS10 Meet your maker: John La Grou — Millennia Media Ten techniques to kill spill REVIEWS: Beyerdynamic Headzone • RSS Digital Snake • Apogee Ensemble Elysia Alpha • Waves V-Series and MaxxVolume • Radial JDV • TFPro P9

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March 2007 V6.2 ISSN 1477-4216 AUDIO FOR POST, BROADCAST, RECORDING AND MULTIMEDIA PRODUCTION

News & Analysis 4

Leader

News Sales, contracts, appointments and biz bites.

Craft 14

New introductions and announcements.

Ten

Broadcast

Katz’s column Bob talks us through the equipment and connections that allow seamless mastering of different-sounding sources.

Mike Oldfield

Sweet spot The NS10 has left a towering legacy and Acoustic Energy believes it has developed its natural successor.

Red Pipe Sound Design

On changing his working methods, classical music, and playing the console and the Mac.

Headroom Media Producer and Visconti’s autobiography.

50 New postproduction outfit serves Sweden’s sophisticated commercials market with image and design.

Products

Orchestral recording Jon Lord’s new orchestral scores offered an opportunity for a recording engineer to reappraise his choices.

Techniques to kill spill.

Meet your maker John La Grou — The man behind the Millennia Media brand talks anti-corporate philosophy and 15 years and nine products.

We consider whether the Dolby LM100 broadcast loudness meter can do for television what Leq (M) did for cinema.

Business 46

Beyond iTunes

Your business Financial house in order but who’s minding your time? Time bandits abound, says Daley.

The latest generation of Internet apps are transforming music discovery.

Technology 60

Spacial sound to air

Slaying Dragons John’s not finding fault this issue, he’s fault finding because failures are all too common.

Achieving ‘stereo’ reproduction from a compact loudspeaker source has baffled boffins for years.

Reviews 22

RSS Digital Snake

Apogee Ensemble

Beyerdynamic Headzone

Elysia Alpha

EDITORIAL Editorial Director: Zenon Schoepe Tel: +44 1444 410675 Email: zen@resolutionmag.com Editorial office: PO Box 531, Haywards Heath RH16 4WD, UK Contributors: Rob James, George Shilling, Jon Thornton, Keith Spencer-Allen, Neil Hillman, Nigel Jopson, Andy Day, Philip Newell, Bob Katz, Dan Daley, John Watkinson

30 32

Waves V-Series and MaxxVolume Radial JDV

TFPro P9

SE Electronics R1

Phoenix Audio Nicerizer 16 ADVERTISEMENT SALES European Sales, Clare Sturzaker, Tel: +44 1342 717459 Email: clare@resolutionmag.com US Sales, Jeff Turner, Tel: +1 415 455 8301 Email: jeff@resolutionmag.com

PRODUCTION AND LAYOUT Dean Cook, Dean Cook Productions, Tel: +44 1273 467579 Email: dean@resolutionmag.com

news Appointments Q U E S T E D MONITORING Systems has appointed Studio Logic Sound as its distributor for the US. CEO of Studio Logic Sound, George Hajioannou, is a 25-year industry veteran with experience in designing, consulting and equipping major facilities on the East Coast. BRENT ELDER has joined UA as VP of engineering as part of the expansion and restructuring of UA’s engineering group. Brent takes over the engineering helm from Bill Putnam Jr and Joe Bryan who will now take on new strategic roles within UA, focusing on business development, new technologies and platforms.

DIGIDESIGN HAS appointed Kyle Ritland as worldwide public relations manager. He was previously director of public relations at Loud Technologies, and has also held marketing positions at Carver, Digital Harmony, and MCA. Digidesign has appointed Chant Peck as Venue product specialist. He has experience as a live sound and studio engineer. G LY P H T E C H N O L O G I E S h a s established a division headquarters in Germany. The new operation will be responsible for sales, production and service for all European dealers and distributors, including Eastern Europe. A new European sales and service team has been established in Nettetal with Martin Richert handling sales and Stefan Leihsa overseeing production and technical support in Europe. Phil French has been appointed president and CEO of Glyph Technologies, assuming all operational activities of Glyph’s US and European entities. He was most recently VP of sales and president of European operations.

©2007 S2 Publications Ltd. All rights reserved. No part of this publication may be reproduced or transmitted in any form or by any means without the prior written consent of the Publishers. Great care is taken to ensure accuracy in the preparation of this publication, but neither

Leader

The incessant clamour for our attention by the various entertainment media continues unabated. Delivery methods multiply in their varieties and the technology that would receive them revises itself with unnerving regularity. We are already at the stage where we can receive the same content through completely different carriers on completely different devices. While I know that the ‘broadcasting’ industry has been touting convergence as a convenient long-term technological carrot, it is the convergence of content that is with us already. And that’s the major flaw. History has shown us time and again that technological take-up can be driven by unique content. It’s not enough that you can pay to watch a football match on your mobile phone any more than watching a scratchy video on youtube, with severely delayed and poor quality audio, constitutes an evening’s entertainment. Somewhere down the line all these disparate technical threads may be joined up and combine into the mother of all entertainment delivery systems but I can’t begin to even guess at when that is likely to happen. The main reason is my concern for the content. I appreciate independent news and documentaries and quality drama but I also like the idea of video on demand; the ability to request a humourous evening’s worth of Department S or Black Adder, for example, would be great. These two broad categories involve completely different business models. The first is current and expensive to produce; the second involves digitisation of back catalogue and is similarly not a trivial exercise. I also want new programming to be suggested to me so I’m not making all the decisions myself — I want to be entertained and informed — and I also like BBC Radio 4 and CDs. We’ve got this far without wondering whether I will prefer it to arrive via satellite, cable, broadband, or any other means because that is irrelevant providing the quality is there to make it a satisfying experience. As is so often the case, too much energy is being expended on the establishment and definition of the delivery medium with not nearly enough attention being paid to what will actually be delivered — football and Big Brother is not for everyone. Do we need it? There’s not an awful lot that I want to watch now but I muddle along. Will it be ‘better’ than what we already have? Unlikely. What’s certain is that none of this will give me any more than 24 hours in a day. Zenon Schoepe

Quantegy discontinues tape

‘New look’ PLASA07

The PLASA07 Show (9-12 September) at Earls Court, London will include a new layout that extends the show over one floor into Earls Court 1 and Earls Court 2. New facilities for visitors and exhibitors include business meeting, presentation and general seating areas, a wireless Internet zone, VIP hospitality suite and a dedicated media centre with extensive facilities. Earls Court 2 will also give the show a fast-track entrance on Brompton Road, dedicated to preregistered visitors, PLASA Platinum Club members, media and VIPs. On-the-day registration desks remain at the Earls Court 1 (Warwick Road) end, but will be accommodated in a more streamlined area inside the hall. Other features for Earls Court 2 include an expanded ESTA Pavilion, hands-on workshop areas, and a PLASA Theatre that will host a series of Master classes, presented by industry leading names. The Central Bar and Meeting Lounge will once again double-up as the After Show Bar, remaining open until 8pm on the first three evenings. ‘We’ve spent a lot of time talking to exhibitors and visitors in planning the first show under PLASA Events’ management,’ said director of events Nicola Rowland. ‘We’re seizing this unique opportunity to move the show forward and confirm its status as the ultimate entertainment technology business environment.’

Consumer electronics shows 7% growth

Quantegy Recording Solutions has discontinued various products from its magnetic tape product lines including GP9, 400 Series, 600 series, ADAT, and R-DAT plus video products that include BETA, Umatic, D1/D2, 196 and VHS and SVHS. Orders for these items accepted through to 28 February are intended to be fulfilling by the end of April 2007. ‘Obviously, as a company, we have very mixed emotions concerning this decision,’ said Quantegy Recording Solutions

S2 Publications Ltd or the editor can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the Publishers.

CEO Frank Foster. ‘On one hand, the company has played a defining role in the development and growth of this industry which has such a loyal following, and we have made many great friends along the way. On the other hand, we have to face the current role that tape plays in the industry.’ The development follows the closure of the doors and filing for bankruptcy at Quantegy Inc at beginning of 2005 and the rebirth of the firm under new owners in April 2005.

S2 Publications Ltd. Registered in England and Wales. Company number: 4375084. Registered office: Equity House, 128-136 High Street, Edgware, Middlesex HA8 7TT. Printed by The Grange Press, Butts Rd,

resolution

Factory-to-dealer sales of consumer electronics are projected to exceed US$155 billion in 2007, or 7% growth, according to the half-annual industry forecast released by the Consumer Electronics Association (CEA). CEA projects that display technologies will continue to be the star category in the industry and account for $22 billion in revenues for 2007. Flat panel displays are expected to ship a combined 19 million units. MP3 players will continue to drive the audio market and CEA projects that MP3 players will account for 90% of the $6 billion in revenues for the portable entertainment market. Thirty-four million MP3 players shipped in 2006 and an additional 41 million are expected to ship in 2007. In 2006, shipment volumes of laptops eclipsed their desktop counterparts.

Southwick, West Sussex, BN42 4EJ.

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March 2007

news Vienna AES

Andersson’s Duality in Mono

Vienna’s Austria Centre will host the 2007 European AES Pro Audio Expo & Convention on 5-8 May. ‘Almost 40% of confirmed exhibitors have decided to expand their exhibits or have returned as AES exhibitors after a short break from the Convention,’ said Thierry Bergmans, European AES exhibition manager. In the UK, the 22nd AES Conference to be held 11-12 April at New Hall College, Cambridge will cover Illusions in Sound the applications of psychoacoustics. In addition to the papers there will be two demonstration rooms in which current ‘withheight’ and planar spatial reproduction systems can be experienced. The AES welcomes everyone to a talk by Rupert Neve at the Royal Academy of Engineering, 29 Great Peter Street, London on 13 March. www.aes.org

Proel buys Turbosound Italian manufacturer Proel SpA has acquired the assets of UK Turbosound Ltd in a deal that is hoped will transform the profiles of both companies. The structure of the companies will remain unaltered and each will continue to operate independently on a day-to-day basis. Turbosound will benefit from additional financial backing, enabling it to increase its research and development activities, speed up time to market and introduce more new products. ‘Over the last 15 years, Proel has focused all its efforts on anticipating the future needs of its customers,’ said Fabrizio Sorbi, founder of Proel. ‘Its rapid growth has dictated a dynamic development model, but its core values have always been firmly grounded on maintaining a strong identity in the market and producing high quality products. ‘Turbosound’s team is a highly skilled and specialised resource which will benefit the whole Proel group. Through this acquisition, we will be able to meet all the needs of the sound reinforcement sector as well as our more traditional markets,’ he said.

Mono Music Studios, owned by composer/producer Benny Andersson, has installed a 96-channel SSL Duality. ‘I wanted to see a new generation of analogue mixers designed to work with a DAW rather than tape recorders,’ said Berhard Lohr (pictured), Mono Music’s studio manager (Resolution V4.1). ‘After testing several large-format desks, we were contacted by SSL, and they showed us the specs of the new Duality. I really liked the concept and felt straightaway that this was the next-generation analogue desk. It feels like you are working more within the world of the DAW you are using.’ • Chalice Recording Studios, one of Southern California’s busiest and fastest-growing facilities, has installed an AWS 900+ for its new overdub/mix Studio F making four SSLs at the five-room complex — two 9000 J Series and an 80-channel G+. ‘We have seen a movement toward all in-the-box mixing and wanted to catch a part of that market,’ said studio manager Stacey Dodds. ‘But we didn’t want to do it with just a standalone control surface and limit ourselves to only being able to use that for what it is. We wanted to have a console in the room that was usable for more than just controlling Pro Tools.’

YLE adopts Lawo for Eurovision

F o u r L a w o consoles from three generations and a Lawo matrix will provide the worldwide sound feeds for the Eurovision Song Contest from Helsinki in May. Host Finnish broadcaster YLE will broadcast the event live from the Helsinki Arena and Audio Broadcast Services (ABS) have been appointed by YLE to equip the main music control room and the master control room on a rental basis. Signal distribution will be handled by a Nova73 HD routing system, which will supply the master control room and integrate four Lawo consoles. The main music control room will be equipped with an mc²90 and will supply a 5.1 mix to the HD-1 OB truck rented from Prisma, using a Lawo mc²66 that will add the atmos and presenter feeds. The production in HDTV

and 5.1 surround will also be uplinked in these formats. Two further YLE-owned OB trucks will be in place for radio broadcast and backup; one is fitted with an mc²80 and another with an mc80. ABS will not only provide Lawo technology but will also equip the master control room with extra control panels and metering systems. ‘We have had extremely positive experiences using Lawo equipment in our studios and OB trucks, as well as at live events when we rented equipment from Lawo,’ said YLE’s technical audio director Matti Helkamaa. ‘For example, when onsite at the Olympic Winter Games in Torino, we used a Lawo mixing console and Lawo matrix on a rental basis. We are sure that the Eurovision Song Contest will add to this chain of positive experiences.’

Appointments HMG HAS begun large-scale expansion of operations at its Sandy, Utah-based headquarters. The announcement comes on the heels of Harman Music Group (HMG) President Rob Urry being promoted to the position of Harman Pro Group CTO and is part of HMG’s five-year strategic growth plan. The company completed a 26,000square-foot addition in June 2006. This enhancement to HMG’s 100,000square-foot complex houses expanded warehouse and distribution operations, as well as office space for 100 additional employees in the company’s R&D department. To date, more than £4m has been invested in the upgrade, which includes high-speed surface-mount and automated manufacturing equipment that has increased production capability by more than 300%. The Har man Music Group has appointed Reed Grothe as executive VP of sales. Most recently, he held the position of chief global business development officer for the Gibson Guitar Corporation. APT HAS opened two new sales offices in the US and made two sales appointments for its North American operations. Kevin Campbell, for merly European sales manager with APT, has taken up the post of VP North American operations and Ted Nahil, formerly of Harris Corporation, has been appointed as National accounts manager. APT has appointed Intelcom Services as its distributor in New Zealand. SYNTHAX AUDIO UK has relocated to new office units at its North Street premises in Melton Mowbray. In addition to more warehousing and increased staff space, it has a dedicated area for product demonstration and training. Synthax Audio UK, Units 2 & 3, The Old Dairy, 47 North St, Melton Mowbray, LE13 1NL, UK. Tel: +44 1664 410600. After three years at the helm of Synthax Inc in the US, Tom Sailor is moving on to start his own company. Hermann Vega Savon, the founder of Synthax Inc, will take over all the responsibilities as president. Brian McCall has been appointed VP of Operations.

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March 2007

resolution

news Appointments ROLAND SYSTEMS Group and Stage Electrics have forged a distribution deal in the UK. As well as being stockists, Stage Electrics will offer demonstration and installation.

Astley closer to AML1s

L o n d o n sound facility S o u n d e l u x ’s creative director Eddy Joseph and sound designer Martin Cantwell were part of the winning team that won the 2007 Sound BAFTA for Casino Royale. ‘Director Martin Campbell’s brief to us was to make the sound a raw, transitional experience for the audience,’ explained Joseph. ‘So instead of creating a dense, multilayered sound design for the film, the soundscape is made up of many carefully chosen individual elements that are based on the scene and the setting. You get a cleaner sounding film that really tells the story.’

TA S C A M H A S p r o m o t e d R i c k McClendon to VP, Tascam Division. Kenji Matsu is now international business manager, Tascam and Teac Electronics, and has been the link between Teac’s Japanese offices and the US for more than 14 years. FOLLOWING THE acquisition of Telex Communications by Bosch Security Systems in late 2006, the title and structure of the Bosch organisation that contains professional audio product brands Telex, RTS, Electro-Voice, Dynacord, Midas, and Klark Teknik has been named Bosch Communications Systems — a business unit of Bosch Security Systems. Bosch says it is now poised to become one of the world’s leading manufacturers and suppliers of professional audio, wireless, life safety and communication equipment. WEST LA Music/Westlake Audio and Minnetonka-based American Pro Audio have been appointed as US dealers for SSL’s AWS 900+, XLogic and Duende.

Long-time PMC user Jon Astley, mastering engineer at Close to the Edge Studios, recently took delivery of AML1 active 2-way monitors for installation in his Twickenham studio. These will supplement his current PMC 5.1 configuration. The AML1’s were acquired for mastering of the recent release from Jools Holland — Moving out to the Country — and he is currently mastering the latest album from John Mayall. Due to the AML1’s diminutive proportions and ease of integration Jon will be taking them to Martian Sound in the West Country for use on a project with Tori Amos.

Linett’s X Bus on location

PRISM SOUND has appointed ANK as its distributor in South Korea. Seoulbased ANK already distributes AMS Neve and Digidesign in the region and is headed by former recording and mastering engineer Tom Kim. Recent sales have included two MEA2 EQs to mastering facility Sonic Korea (pictured). A L L U S orders for True Systems products are now placed directly with the manufacturer’s Tucson headquarters. Sennheiser Electronic Corporation is no longer the exclusive US distributor for True.‘We intend to build a closer connection to our dealers in addition to offering direct in-house technical support and service to our end users,’ said Tim Spencer, president and chief engineer of True (pictured). ‘With fewer steps in the distribution chain, True is capable of improving core product features in a timely manner, adding value to the line by lowering prices across the board and improving our connection to end users.’

Sound BAFTA for Soundelux

Engineer/Mixer Mark Linett recently added a Mackie Digital X Bus to his Your Place Or Mine recording studio in Southern California. The 48-track system is geared toward location recording and postproduction for live concerts and broadcast projects. On location, the X Bus receives its signal via the optional FireWire card from one of two Genex 9048 48-track hard disk recorders. The second 9048 is typically used for backup, or on shows where more than 48 tracks are required, like the recent Santana tour. Additional tape backup is via six Tascam DA-78s. ‘Working live means making quick decisions, and I don’t have time to page through menus and mess with layers of parameters,’ said Linett. ‘The X Bus’ interface is really so well designed. The

dual displays make it easy to display a wide range of information, and since they’re touchscreens, I can access almost any parameter I need within one or two moves. It really does put the entire mix at my fingertips.’ Although most often associated with Brian Wilson, due to his long-term working relationship with the Beach Boys’ founder including the Smile LP (for which he received a Grammy nomination for Best Engineering), Linett’s has also collaborated with Randy Newman, Jane’s Addiction, Paul Simon, Santana, Eric Clapton, the Blasters, Love, Dixie Dregs and the Red Hot Chili Peppers. ‘The X-Bus was obviously designed by engineers who really mix,’ continued Linett. ‘It’s a console that works with me and adapts to my mixing style, rather than making me adapt my style to the mixer.’

resolution

Codec manufacturers group for compatibility A E TA A u d i o S y s t e m s a n d M AYA H Communications founded the international Audio-via-IP Experts Group at the end of last year to coordinate joint technical efforts to achieve true interoperability of audio codec products from different manufacturers. Orban/CRL has now also joined the group and the three manufacturers have successfully performed transmission tests of their codecs applying the SIP/RTP protocols according to the draft standards addressed by the EBU. Such tests covered bi-directional transmission using G.711, G.722 as well as MPEG Layer 2 and Layer 3 audio coding. Continuous extension of the featured compatibility stands on the road-map of the Group. ‘This illustrates the importance of the efforts being put into the Experts Group,’ explained MAYAH Communications CEO Detlef Wiese. ‘Codec recognition with SIP/ RTP is becoming mature, and automatic connection set-up has now become a reality for IP-based transmissions. In combination with predefined compliance to the draft EBU recommendations, there is a solid base for further developments which will be to the benefit of end-users.’

Famous Japanese film production outfit TOEI Company Ltd has acquired a Sonosax SX-ST8D. The new mixer has already been used at Kyoto Movie Studios. Pictured (l-r) Sonosax’s Jacques Sax, sound engineer Nobuhiko Matsukage, and Yoshio Horiike deputy manager, postproduction section.

March 2007

Master DVD today – HD media tomorrow.

Dolby Media Producer. ®

Introducing Dolby Media Producer. Dolby reference encoding and decoding in a family of convenient software applications. Supports today’s DVD formats and future disc formats as well. Comprises four applications that can be used individually or combined : Dolby Media Encoder is a professional, non-real-time encoder that offers network-based encoding of Dolby Digital, Dolby Digital Plus, Dolby True HD and MLP Lossless TM.

NEW

Dolby Media Encoder SE

has all the same features as Dolby Media Encoder, but is optimised to run on a single computer, making it ideal for smaller-scale operations.

Dolby Media Decoder offers reference decoding of Dolby disc formats, complete with transport control for audio-tovideo synchronisation.

Dolby Media Tools is a powerful utility program that saves time on audio file preparation by enabling the repair and update of previously encoded disc files without re-encoding.

To find out additional information, visit www.dolby.com or contact your local Dolby distributor.

news Appointments

TV2 refits with two Vista 8s

SE in Real World

LOUD TECHNOLOGIES has appointed MS Max as its distributor in Russia. D & M P R O F E S S I O N A L , p a re n t company of Denon Professional and Marantz Professional, has established the D&M Professional Education Solutions Division (ESD). www.education.d-mpro.com SYMETRIX HAS appointed Michael Worona as director of business development. He has been with Symetrix for four years as product specialist and previously worked at Loud Technologies in the digital development department. Symetrix has appointed Audio Audio Agent’s Dave Christenson. Agent as its sales representative in Latin America.

Norwegian commercial television station, TV2, is engaged in a multi-million euro investment programme to refurbish its central Oslo studio premises and has specified two 42-fader Studer Vista 8s for its new audio control rooms. Head of studio productions Stig Gøran Nordahl has created a versatile network, enabling two control rooms to link to three studios with full interconnectivity. The design of the Vista 8 allows TV2 to place three or four wallboxes in every studio, supplemented by a 24-in/8-out remote stagebox that can go anywhere. Tie lines to all the green rooms and VIP areas extend the network still further. With ambitious plans being announced by TV2 management for new programming and new channels, Nordahl has specified ‘a lot more inputs and outputs for the consoles than we need at this minute. We had to prepare ourselves for the possibility of a large live entertainment/Saturday night show, with a lot of guests and live bands. We’ve bought extra cards, so we should be ready for most things. Currently we can handle 72 mics plus 16 channels of wireless. If that’s not enough, we can buy another card.’ ‘There is a small but closely-knit community of sound engineers in Norway, and they’re all using Vista,’ added Nordahl. TV2’s sister company, OB Team, which operates the station’s outside broadcast fleet, has installed a Vista 8 in one of its flagship vehicles and Vista 7 and 8s are widely used by NRK, Norway’s national broadcaster.

INDEPENDENT SYSTEM integrator dB Broadcast has expanded its engineering team by appointing Phil Keeler as a project manager. Phil will report to engineering manager Gareth Evans. He has worked for The Mill, TVI Ltd and MTV Networks Europe.

First JAMES to the Leeds Met

THE INSTITUTE of Broadcast Sound has appointed Simon Jones as its first training coordinator. The post was a key recommendation of the IBS’s Training Seminar in April. His initial brief includes seeking funding to enable the IBS to expand its training efforts. STUART WOOD has joined Lab X Te c h n o l o g i e s ’ engineering staff after ten years with Bosch Security Systems. He will design embedded hardware, gateware, and firmware for Lab X clients, who include leading pro audio manufacturers. M I D I L A N D I N Seoul has been appointed distributor of CharterOak products in Korea.

Real World Studios has recently taken delivery of a range of SE microphones for its new Mill Side studio. ‘Real World own a Gemini, an SE2200a and a pair of SE3s — I also own a z5600a,’ explained house engineer Dom Monks. ‘Bruno Ellingham [another producer and engineer at Real World] first put me on to SE mics. He had a few on demo and was getting really great results from them. ‘The great thing about SE mics is their value for money. Because any producer or artist can afford their own SE mic they’re not restricted to where they record, so it’s that much easier to maintain continuity. And compared to other “low cost” mics on the market they just sound more expensive.’ ‘The package was just right: good quality mics, solid build, a selection of useful and lasting accessories and a decent aluminium case for each mic or pair,’ added senior sound engineer Marco Migliari (pictured). ‘Add this to a competitive price, and you’ve got an easy decision on your hands. I particularly like the tone of the SE2200a — it’s hard to find another mic in the same price range that could deliver a better performance.’

(l-r): Julian Slater, Hot Fuzz co-writer Edgar Wright, Nigel Heath. London Soho-based postproduction facility Hackenbacker carried out the sound editing and mixing for the Life On Mars UK TV Series. Production company Kudos Film and Television chose Hackenbacker for the audio because of its work on the first series. British comedy film Hot Fuzz also employed the facility with director Julian Slater given the task of supervising sound editor while business partner Nigel Heath mixed the final product.

resolution

There’s a humourous caption for this somewhere. Leeds Met has become the first university to receive a top music industry accreditation for its Music Technology courses from the newly formed JAMES organisation. JAMES (Joint Audio Media Education Services) was launched by the APRS and the MPG — the two main industry bodies responsible for assessing the quality and suitability of recording, music technology and music production courses in the UK. George Martin, who was awarded an honorary doctorate in music from Leeds Met in November, is the patron of this new organisation, which was founded to give industry support to educationalists and institutions working in the recording industry.

March 2007

enter the HeaDzone...

5.1 monitoring system.... Headzone has been designed to provide the most accurate headphone-based 5.1 surround-sound reproduction possible from today’s DSP technology, in the world’s most compact portable package. Headzone offers a unique patented ultrasonic Head-Tracking system which locates the orientation of the listener’s head with respect to the source material and adjusts the audio accordingly. This allows complete freedom of movement, while the source material remains fixed in position, just as with a ‘real’ 5.1 loudspeaker arrangement. For a demonstration contact matt.nettlefold@beyerdynamic.co.uk

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news Biz Bites J A N U A R Y USHERED in a night of the long knives for UK music executives, writes Nigel Jopson, as EMI chief executive Alain Levy and deputy David Munns were fired. Chairman Eric Nicoli took direct responsibility for management of the recorded music division, and shows himself (again) to be an astute corporate survivor. Underneath, the EMI story would be familiar to a time traveller from the 1980s: good in the

UK, crap in America. Steve Knott left his job as MD of HMV (the UK’s largest specialist music retailer) and Peter Jamieson was out at the BPI in favour of ex-IFPI man Geoff Taylor. Music Zone went into administration, and HMV and Woolworths produced profit warnings.

HD expansion at Levels Post

Billed as the ‘first postproduction HD super-boutique’, Levels Audio Post in Hollywood specialises in high-definition, 5.1 audio postproduction, music production, and sound design services and has installed five Digidesign Icons in its new 13,000sqft complex. Studio owner and mixer Brian Riordan worked with architect and studio designer Peter Grueneisen to create five Icon-equipped 5.1 audio dub stages along with an ADR/Foley stage, three audio editorial suites, five HD offline/online bays, two visual effects suites plus a central machine room. Also in Hollywood, Record Plant recording studios recently installed an Icon in its DigiPlant room. The studio installed a 16-channel Icon D-Control with Surround Panner option and an integrated Pro Tools|HD 3 Accel as the mix engine. Audio interfaces and peripherals include the 192 I/O, Sync I/O, MIDI I/O, and PRE preamp. ‘Serving our clients is one of our hallmarks,’ said Record Plant CEO Rick Stevens. ‘We tried a number of different systems in our DigiPlant room, but it was clear that the Icon system was the most well accepted among our top-tier clients and would best serve their needs. ‘All of our rooms offer Digidesign Pro Tools systems. And in fact, more than 70% of all mixes done at the Record Plant are done “in the box” these days,’ he added. ‘The ICON system provides the most efficient way for our clients to record overdubs, comp vocals, make edits, or anything else they need to do on a Pro Tools system before —- or during -— the mixing sessions in the larger mixing rooms. Then they can move files and sessions around the facility easily using our high-speed network.’

AudiopleXus goes 24-hour THE NORWEGIAN ombudsman declared iTunes illegal, while Steve Jobs adopted a holier-than-thou attitude as he dissed DRM in an open letter to major labels: ‘If the big four music companies would license Apple their music without the requirement that it be protected with a DRM, we would switch to selling only DRM-free music on our iTunes store.’ The IFPI’s John Kennedy said: ‘I think he’s expressing some frustration at being the bad guy ... and people like the Norwegian government beating him up, and he’s taking it out on us.’ EMI IS in talks to release DRM-free catalogue, and sent standard-bearer artists Norah Jones and Lily Allen out on plain old MP3. It’s suddenly fashionable for artistfacing execs at majors to spurn DRM, with many apeing Columbia MD Mike Smith’s example in predicting DRM would be abandoned within a year. Amidst all this sudden enthusiasm, the ‘Rights’ part

AudiopleXus Mastering in London is opening its facility for 24-hour operation in response to client demand last year, according to studio founder and chief engineer Chris Stilmant. Recent additions of Prism Sound A-D/DA convertors and new Weiss equipment have reinforced the operation’s stance as a predominantly hardware-based mastering studio with very little use of plugin software. ‘Now we have all the gear to master every genre of music and style and

also the facilities to sync to video and DVD for television and Internet streaming audio and video for MP3-MP4,’ said Stilmant. ‘By offering a free trial two years ago, musicians and artists where able to test the high grade quality of AudiopleXus mastering without any risk and found out for themselves that we are a very serious facility with the skill and talent to make it world class.’ Originally based just around St ilmant, a few freelance engineers are now also working in the studio on a regular basis to meet demand.

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Jackson employs Millennia on Streisand Bruce Jackson, FOH engineer for Barbra Streisand has once again chosen Millennia HV-3 mic preamps for her tour. ‘Whenever I need a standalone microphone preamp I always choose Millennia. They have proven themselves time and time again,’ said Jackson. ‘Studio products don’t always hold up on the road. The Millennia’s are built solid, with John La Grou’s personal attention to detail, and are the result of his personal quest for the best. ‘We have a bunch of consoles on the tour,’ he continued. ‘We put the Millennias across any microphones that are shared with multiple consoles so we don’t load the mics down and change the sound. We also use them to buffer signals to recording trucks. For example in Madison Square Garden, the truck is too big to go up the ramp so we park them down on the street. That results in an 800-foot run from the truck to the mics. Imagine the load on a mic once it’s split 3 or 4 times for the show and once again to the truck 800 feet away. The result is a dead sounding input overloaded with too low an impedance and a massive capacitance on the 800 foot run…enter the Millennias…clean, accurate amplification coupled with an output that has enough line level audio drive to light a light bulb!’

Thrall thrilled with Metric

Thrall (left), drummer Vince Wilburn (centre) and bass player Charley Drayton. Artist, engineer and producer Pat Thrall is redefining remixing through his work on recordings from the late 1960s and introducing a new generation to the music of Miles Davis and Sly and the Family Stone. Complementing the original 8-track masters with overdubs by a host of current music artists, Thrall relies on Metric Halo’s ChannelStrip plug-in. On the remix of It’s About That Time from Davis’ 1969 album In a Silent Way, Thrall engineered new contributions by drummer Vince Wilburn (Davis’ nephew and former band member) and bass player Charley Drayton. All three musicians co-produced the remix, which also includes overdubs by Carlos Santana. The remix is included on Miles Davis — Cool & Collected on Columbia/Legacy Jazz. ’ChannelStrip is basically my drum sound,’ said Thrall. ‘I use it regularly over all the drums, and have for years. I bought it when it first came out.’

March 2007

news Biz Bites of Digital Rights Management seems to have been forgotten. DRM was not invented for technology companies to build proprietary walled gardens around their music players, they just used it like that. The digital rental/subscription model can only deliver revenue to music creators if users’ listening preferences are somehow tracked. ‘This is 100% an issue for the music business,’ said Leanne Sharman, manager of Napster UK, ‘I don’t think that technology should be driving the music industry forward, but at the moment it is.’ LAST.FM, a music-focused social network that claims 12 million users (see this issue’s Business article p46), announced a deal to integrate Warner Music content. ‘We want to enable fans to experience exciting ways to uncover new Warner Music artists, and to enjoy innovative approaches to customising their digital music experience,’ said Alex Zubillaga, EVP of Digital Strategy and Business Development at WMG. I detect the start of a worrying trend, in which the owners of a particular right — the sound recording copyright — cut deals directly with corporates. LEGENDARY US act The Eagles announce their first new studio album since 1979 will be available only from Wal-Mart stores for the first year of release. Microsoft pays Universal a $ for every Zune player sold, Vodaphone offers MySpace on subscribers’ mobiles, and video-sharing site Bolt.com looks set to throw in the towel and pay millions to UMG. Bulk content licensing deals threaten to cut out producers and nonfeatured performers, as the collection societies who have for years ensured their stipend find themselves with no means of tracking usage and are bypassed in corporate-to-corporate agreements.

SHOWTIME Sounds Expo, London ......1-3 March CabSat, Dubai .................6-8 March Musikmesse/Pro Light & Sound, Frankfurt ......................28-31 March NAB, Las Vegas .............14-19 April AES Europe, Vienna ............5-8 May Musikmesse/Pro Light & Sound Russia, St Petersburg .....14-17 June BroadcastAsia, Singapore 19-22 June BIRTV, Beijing ............ 22-25 August IBC, Amsterdam ....7-11 September PLASA, London ......9-12 September AES US, New York ....... 5-8 October SATIS, Paris .............. 23-25 October SBES, Tokyo ......... 20-22 November

Berlin’s Mitte mixes with 5-MC

A 32-fader Euphonix System 5-MC has been installed by Studio Mitte in Berlin, which is using it with Nuendo, as part of a major upgrade and relocation for the film and TV postproduction facility. ‘The Euphonix setup was so smooth that our first project using the System 5MC, Julie Delpy’s film Two Days in Paris, was edited and mixed while the system was being installed,’ said Jörg Höhne, Studio Mitte’s founder, CEO and rerecording mixer. ‘With the 5-MC and the advantages afforded by EuCon, we found the perfect, most professional hardware to work with our favourite software.’

Fritz collaborates on musical experimentation

Netia at the heart of Radio France N e t i a ’s M a n re o r a n g e o f c o n t e n t management software is being used by Radio France to control all internal audio flows among its studios and to enable simultaneous mixing from workstations in different studios. The custom solution is a Manreo-derived, IP-based redundant control system for the Lawo Nova 73 audio routing matrix. The system was taken live in August as part of a master control room at La Maison de Radio France, the Radio France headquarters in Paris. ‘We rely on Netia’s Radio-Assist for all of our sound production tools, and for this master control project, the company demonstrated its ability to meet our needs for IP-based signal-routing software as well,’ said Jean-Marc de Felice, head of technical resources at Radio France. ‘The resulting system allows us to access audio more quickly and easily, with much greater flexibility in routing signals where they are needed. The Netia routing system gives us the ability to link equipment located all over the facility and has allowed us to free up space for other projects at La Maison de Radio France.’ The system controls 1120 x 1136 crosspoints and handles all signals that enter and leave the facility, including incoming signals from ENG and SNG, transmissions to the radio transmitter for FM and AM networks, and routing of feeds from one studio to another.

FogoLabs Holophones Junkies

Fritz, Neumann’s nickname for its KU 100 model human head, which features microphone elements located inside its replica ears, is being used by avant-garde composer Zeljko McMullen to create new binaural projects. McMullen, who also works as an installation sound artist and musician, says he has been experimenting with the Neumann KU 100 as a recording method and as a way to rerecord stereo music playback. Initially borrowing Fritz for three weeks from Mike Pappas, Neumann USA senior applications engineer, the KU100 sparked a slew of creative ideas. ‘I compose music for spaces, site-specific,’ said McMullen. ‘I go into a space and tune the music to the room, and I’ve always been interested in ways of capturing that. ‘I was working on some meditation music that I was trying to make into an enveloping

field. That was translated to CD and then I positioned different playback things around the room positioning Fritz in the room where there was the most resonance and where it sort of erased the room on the recording, so it didn’t sound like a distinct left-right. By putting a lot of different speakers around Fritz, I ended up having it function as an attendant in an installation.’ Having exhausted the possibilities of a stationary Fritz, McMullen really started experimenting. ‘What if Fritz were walking around in here? What if Fritz is in a gyroscope? I had a playback of very lowend sounds and some high white noise, and I cradled Fritz and flipped it upside down and spun it by hand.’ He explained. ‘On listening back to that I got dizzy, but it was pretty interesting. I didn’t know that the effect of what you heard on your balance could be so visceral.’

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Faced with the task of recreating the Cowboy Junkies’ performance at Toronto’s Church of the Holy Trinity for an upcoming HD concert DVD, production and postproduction facility FogoLabs selected Holophone’s H2-Pro to capture the audio. Originally recorded in 1987, most of the Cowboy Junkies Trinity Session was recorded in one day with one microphone. The new project, scheduled for release in the spring, features the Cowboy Junkies with guest performances by Natalie Merchant, Ryan Adams and Vic Chestnut and was filmed in HDCAM SR and recorded at 24-bit/96k. ‘To be faithful to fans of the album, I wanted to use one mic as the basis to record the band, as was done back in the day,’ explained FogoLabs producer/engineer Francois Lamoureux (Resolution V3.2). ‘We are in the surround generation now and the choice of surround microphone was easy. I selected Holophone’s H2-Pro because you are able to realistically capture 7.1 channels of audio with minimal setup.’ FogoLabs placed the surround mic a little off to the side near the drum section. Additional Shure mics, mostly from the KSM series, were placed close to each instrument to supplement the sound picked up by the H2-Pro.

March 2007

Photography by: Louis of Teddington

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Red Pipe Sound Design Sweden enjoys an enviable reputation for the quality and sophistication of its commercials market. ZENON SCHOEPE visits a new postproduction outfit in Stockholm that has taken the minimalist approach to an extreme and cultured a refreshingly different look.

NY NEWCOMER TO THE INDUSTRY who still labours under the misapprehension that it’s all about the gear and how item B when combined with item S will create an irresistible attraction to clients and fame, would do well to look at the postproduction business. While trailblazing in

their adoption of technology that makes them faster and more efficient, postproduction facilities have a habit of stabilising on equipment, once a formula and standard have been achieved, and then concentrating on earning a return. And, just to prove the point still further, post engineers have a habit of spinning off

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on their own or with other like minded practitioners and taking their clients with them. The clients are hiring them; not their box of spanners. They like their creature comforts and a nice environment but they trust the engineer to deliver the end result with whatever technology he deems fit. Red Pipe Sound Design was opened in Stockholm towards the end of last year by Tobias Norman, Erik Olsson and Mikael Frithiof together with music mastering company Cutting Room from nearby Solna. The three engineers previously worked at Riviera in Stockholm, formerly know as Mic Studio (Resolution V2.2), and opened the studios because they wanted to concentrate on sound rather than create a facility that catered for all disciplines under the one roof. Majoring on sound design and mixing for commercials, drama and multimedia productions it has two Genelec 8040-equipped 5.1 studios running Pro Tools HD3 with a shared voiceover booth between them. Built in what was previously office space the décor is high-style but not at the expense of performance. ‘When we built the studios we wanted to concentrate on the sound first and then the design and how they looked afterwards. We’re really happy with the results,’ says Tobias, adding that the rooms were designed for surround from the ground up. Being long-time Genelec users the monitor choice was a natural one. ‘Whenever you build a new studio, initially you’re not totally comfortable with it and you have to rely on the speakers — that’s something you can take across to a new room.’ Awareness of 5.1 is high in Sweden, according March 2007

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UR ORDER YO LOG FREE CATA to Mikael who says that even business-to-business presentations are more often than not delivered in 5.1. This multichannel connection is carried through with music mastering facility Cutting Room’s share in the studio. Cutting Room was interested in broadening its area of operation and the team saw logical expansion in its mastering for film DVD. ‘It’s quite common in Sweden to have the final feature mix, which has been done in a theatre, and to put that version on to the DVD release,’ explains Mikael. ‘That’s not good and we don’t believe that Swedish films sound as good as they could. ‘When we work with feature films we do the premix in our studios and then we take it up to Europa film studios and do the final balance there for cinema release. We do the final mix for the home theatre DVD here with mastering at the Cutting Room,’ he says. ‘If you look at a film like Borat, that’s not a film that needs to be shown on a big screen. It’s suited for the DVD market,’ adds Tobias. ‘We think in the future that feature films in cinemas are not going to be the main goal for the distributors and production companies because a feature film in cinema lives maybe two months — three months if it’s a really good movie. Then it is never shown again. The DVD release of a movie lives for ever — almost. Why not focus on the DVD release first? You can do that at the cutting edge of sound design and mix for DVD and then you can focus on the cinema release.’ This mastering connection not only benefits the quality of DVD releases but also has benefits in their other lines of work. ‘It means that the commercials, drama productions and whatever else that we sound design and mix here, we can also take to Cutting Room and make final mastering of it in exactly the same way that you would in the music business,’ says Mikael. Cutting Room has two studios for surround mastering and three for stereo and the working methods are complementary. ‘We work a lot inside the workstation with plug-ins but Cutting Room works with a lot of analogue and digital outboard – so we’re combing two different ways of working,’ says Mikael. ‘It’s also a way to give a different sound to our mixes and a sound that maybe no one else can get. It’s been interesting for us because we’ve never worked in this way before.’ There’s also a third company in the equation — DVD authoring company Born Studios, which is 50% owned by Cutting Room — so they also have

connections to another part of the process. They say that if they get involved early enough with a film project they’ll be able to offer all stages of the studio production. That said, it’s still early days and drama and feature films are only expected to account for some 20% of the work, with the remainder taken up by commercials in Sweden’s small but incredibly sophisticated market. ‘Swedish commercials have a really high-end focus on the sound and that’s fun to work on,’ says Mikael. ‘Commercials are very creative in Sweden; the directors are up for trying new things. When we work on feature films and drama we want to apply our way of working with sound design from the commercials market.’ Red Pipe is located in the centre of Stockholm within easy reach of all the agencies, and picture editing outfit Jolly Roger with an Avid Adrenaline editing suite also has a room at the facility. In response to the realities of production companies and how clients tend to budget based on an hourly charge, Red Pipe has simplified its rate card and has folded in many of the extras they used to charge separately for into the hourly rate. ‘It has been a real release for a lot of production companies because it’s easier for them now and they know that the price we quote them is the price they’ll pay,’ says Erik. Red Pipe aims to provide a ‘boutique’ postproduction experience because they believe that the service has to be more personalised on account of the commercials business being a relatively small community in Sweden. Clients, they say, like an element of exclusivity. ‘When you’re a director working on commercials you want to take your work to a guy that you’re used to working with,’ says Mikael. ‘It’s not the company; you go to someone that you like to work with and that you can rely on.’ ‘That’s a really good thing because eight years ago the main focus was on: are you working digitally, do you use Pro Tools, what do you use? The machine was in focus,’ states Erik. ‘Now they understand that you have the equipment, you have at least the basics, you know what you’re doing…so, who am I working with? The guy that is sitting there is important again. That’s good progress.’ ■

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March 2007

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Products Equipment introductions and announcements.

Tube-Tech modular system Tu b e - Te c h h a s announced a valve-based modular system built around three modules. The RM 8 tabletop enclosure contains the power supply for the modules with all the necessary voltages, separate in and output XLRs for each channel and DB25 input and output. This holds a maximum of eight modules in any configuration and can be added to at a later date. The CM 1A module is based on the CL 1B and its compressors can be linked together on two buses. The EM 1A equaliser is based on the PE 1C Pultec clone. The PM 1A has selectable input impedance, phase reverse, -20dB pad, +48V phantom power and a low cut filter. It has a coarse gain (10dB steps, 20-60dB) and a fine gain available in 2dB steps +/-10dB. All modules have fully floating input and output transformers and as with all Tube-Tech units the signal path is tube-based. www.tube-tech.com

Benchmark native 96kHz, 24-bit USB

Benchmark’s DAC1 USB is a 192kHz, 24-bit, D-AC with ‘Advanced USB Audio’, which provides high-resolution, bittransparent playback seconds after plugging into a computer’s USB port for the first time. There is no software to install, and there are no system settings that need to be changed. It is compatible with Vista/XP/2000 and Mac OSX. The DAC1 USB includes high-current output drivers that can be configured to mute upon headphone insertion. The original DAC1 does not include the USB option or special high-current drivers and both options are available in silver, black, and black rackmountable chassis. www.benchmarkmedia.com

Portico duo mic pre/variable phase DI

Rupert Neve Designs’ Portico 5016 is a 2-channel device combining the RND 5012 mic preamp with a DI, which may be used together or independently. A Variable Phase control allows the manual phase-alignment of the direct input source in relation to the microphone input signal. The 5016 features very short signal paths, minimal negative feedback and custom-designed transformers that, on the DI input, provide ground isolation and the

Platform news: Digidesign Structure is the latest RTAS virtual instrument plug-in from the AIR group. A sampler workstation for Pro Tools, Structure includes a sample library from EastWest and has support for importing SampleCell, EXS24, and Kontakt 2 sample libraries. Structure has a 128-level multitimbral universal sound engine and support for an unlimited number of nestable patches and up to 8-channel interleaved samples. It supports bit depths, sample rates, and surround formats up to 24-bit/ 192kHz/7.1 and with a database and integrated file browser, users can quickly find and load the sound they’re after or drag and drop Pro Tools Regions directly into Structure and tweak them using the plug-in’s integrated sample editor. Structure also has an integrated multi-effects processing engine. Reel Tape Suite is a set of analogue tape emulation plugins that recreate the warmth of analogue tape recordings and tape effects. Operating as AudioSuite, TDM, and RTAS, it features three plug-ins: Reel Tape Saturation, Reel Tape Delay, and Reel Tape Flanger. Reel Tape Saturation, which will also be offered apart from the suite, simulates analogue tape saturation effects by modelling analogue tape machine and tape formulation characteristics. Reel Tape Delay simulates classic tape echo effects while providing precise control over regenerating echoes. Reel Tape Flanger simulates tape machine flanging effects. You can select from the response characteristics of three different tape recorders and two different tape formulations, alter the tape speed or feedback level and the bass and treble feedback response of the Reel Tape Delay, and adjust the LFO rate and depth, wow and flutter, and delay time of the Reel Tape Flanger. You can also dial in the drive, noise, tape machine bias, and level calibration of the Reel Tape Saturation. www.digidesign.com

Platform news: Steinberg Steinberg and Euphonix have updated Nuendo EuCon to 3.2.2. The update brings full MIDI support, further extending the scope of the integration for media production. This includes control over MIDI tracks and channels including MIDI plug-in insert effects, as well as access to all elements of corresponding VST Instrument channels. Editing MIDI parts in the Project Window is now as seamless as editing audio parts. MIDI notes can be edited with the help of the soft keys, smart switches and two jog dials on the Euphonix MC. A set of additional preferences allows better organisation in projects containing many tracks and the transport behaviour can now be adapted to suit user preference. The display of key Nuendo software parameters on the Euphonix hardware has been enhanced to allow faster recording, editing and mixing with Nuendo on MC, System 5 MC or System 5 Hybrids. www.steinberg.net

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March 2007

gear virtual elimination of interference loops. The independent DI input can select between a high impedance instrument or line level source and provides up to 30dB of gain. A front panel Thru jack socket provides independent access to the unprocessed instrument DI signal. The microphone preamplifier section of the 5016 is identical to the Portico 5012 and is designed to provide very low source loading. Mic gain is from +72dB to -6dB via a stepped rotary switch and a continuously variable trim. A continuously swept 12dB/octave high pass filter covers 20Hz-250Hz and the ‘Silk’ is included along with switches for mute, phase reverse and 48V phantom power. www.rupertneve.com

Elysia Mpressor

The Mpressor dynamics processor from Elysia provides several functions that create ‘fat and freaky’ sounds by employing punchy control, characterful colourations and extreme settings. Each of the two linkable channels offers a switchable Auto-Fast function that allows fast and distortion-free compression without artefacts. Antilog circuitry generates compression effects such as pumping and breathing and variable area filters are included. Maximum Reduction allows the intensity of the compression to be accessed on one controller. Switchable external sidechain inputs are also included. The circuit design is based on discrete analogue technology and signal processing is realised by single transistors in permanent Class A mode and the sidechain and power supply have discrete circuits. Oversized power supplies, capsuled conductive plastic potentiometers, internal temperature stabilisers for critical components and special current feedback amplifiers are part of the technical spec. www.elysia.com

Apogee Symphony Mobile Apogee has introduced Symphony Mobile, a professional multichannel ExpressCard for Apple’s MacBook Pro. Symphony Mobile features all of the performance, quality and value of Apogee’s Symphony PCI-Express card with up to 32 channels of I-O and less than 1.6ms of latency at 96kHz in a portable form factor. Installed directly into Apple’s MacBook Pro and connected to Apogee’s X-Series or Rosetta Series convertors, Symphony Mobile delivers capability that rivals most desktop workstations. It is compatible with any CoreAudio software application and includes Apogee’s Maestro Software for control and routing, and Apogee’s VBus for virtual routing between CoreAudio applications. www.apogeedigital.com

Dolby Media Encoder SE The latest addition to the Dolby Media Producer product line, Dolby Media Encoder SE (Standalone Edition) is designed for smaller postproduction facilities and is a cost-effective single-computer software encoding solution that offers the same level of encoding capabilities as the original Dolby Media Encoder network version. Dolby Media Encoder SE encodes all Dolby audio technologies used in packaged media applications, including Dolby Digital, Dolby Digital Plus, Dolby TrueHD, and MLP Lossless. It brings file-based encoding to facilities producing audio files for DVD-Video, DVD-Audio, HD DVD, and Blu-ray Disc. www.dolby.com

When enthusiasm leads to passion ...

Waves L3-16 16-band peak limiter T h e L 3 - 1 6 Multimaximizer is the latest in Waves’ L-Series processes. It is the first to offer 16-band peak limiting ‘delivering unparalleled loudness with unprecedented control’. It features a built-in linear phase equaliser as well as linear phase filtering throughout its crossover network. The L3-16 is available exclusively in Waves Mercury. Powered by Waves PLMixer Peak Limiting Mixer technology, the L3-16 limits and maximizes the sum of all 16 bands to a set Threshold factor for maximum output while keeping the music intact and transparent. It has an EQ-style 6-band interface offering envelope control of Gain and Priority. The PLMixer uses psychoacoustic criteria to intelligently determine how much attenuation to apply to each of the 16 bands, so all available headroom is used. This is modified by a gain system that operates like a linearphase EQ section placed ahead of the limiter, plus a priority envelope across the entire audio spectrum to maximise loudness and optimise tonal character. Release Profiles control the behaviour of the L3-16’s built-in ARC (Automatic Release Control) engine across the band-split spectrum. The 16-band crossover network is linear-phase and thus delivers an output free of phase distortion and artefacts. www.waves.com

March 2007

SCHOEPS Mikrofone Spitalstr. 20 76227 Karlsruhe - Germany +49 721 943 200

Diana Mayer-Blaimschein and Martin Mayer are:

www.schoeps.de

www.mistermaster.at

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gear AKG new releases To mark its 60th anniversary AKG has released a limited edition C 414 LTD featuring handselected capsules and components for optimum sonic performance and a classic nickel finish and brass grille screen that evokes the first models of the microphone. It also has a transformerless output stage. The IVM 4 UHF wireless in-ear monitoring system incorporates Dbx digital signal processing and employs the same system expansion components as AKGâ&#x20AC;&#x2122;s WMS 400/WMS 4000 systems. The SST 4 includes12:55 an Projekt1 18/02/07 Side 1 integrated Dbx compressor, an

sound engineering

equaliser offering a large number of EQ presets, and inear-optimised binaural room simulations. The IVM 4 set includes the half-rack SST 4 stereo transmitter, a S P R 4 stereo bodypack receiver and pair of IP 2 earphones. The SST 4 features selectable RF power output up to 100mW, preprogrammed frequency sets and can be used with 14 channels in each of its 30 MHz bands. The SPR 4 bodypack has Auto Setup and Environmental Scan features and has an operating time of 7 hours. The newest generation of MicroMic series miniature microphones offers five products: C516 for accordions and for mounting on instrument amplifiers; C518 for drum and percussion; C519 for use on woodwind and brass; C520 condenser vocal mic; and the C555 head-worn mic. Each of the MicroMic

RM8 TUBE MODULES

models offers a flexible gooseneck and elastic suspension. The C516, C518 and C519 offer metal capsule housings and an adapter plate that allows them to be attached directly to AKG WMS 40 Pro and WMS 400 bodypack transmitters. The C5 condenser vocal microphone and D5 dynamic vocal microphone each feature a dentresistant, spring steel, wire mesh grille cap and a rugged zinc alloy die-cast housing. An integrated shock absorber system minimises handling noise. The C5 has a 24-carat gold-sputtered housing that protects the condenser capsule against moisture and humidity plus a removable Presence Boost adapter for increased intelligibility. The D5 includes new Laminate Varimotion diaphragm technology to deliver smooth frequency response and higher gain-before-feedback. The D5 is also available in a switched model, the D5 S. www.akg.com

UAD-Xpander UAD-Xpander is an ExpressCard DSP system for laptops that will be offered in three versions: the UADX p a n d e r X p re s s (US$999 including a UAD$500 voucher), the UAD-Xpander Xpert (US$1399 including a UAD$1000 voucher), and the UAD-Xpander Xtreme ($2199 including all UAD plug-ins to v4.5). The Xtreme will uniquely feature a highly polished chassis. Key features include 44.1-192k Mac/PC DSP audio acceleration for laptops, noiseless, fanless chassis design, ultra-fast (2.5Gbps) ExpressCard 34 interface, 14 powered plug-ins, a travel-case with compartments for all components and an optional UAD-Xtenda kit with PCIe desktop adapter. V4.5.1 software focuses on 64-bit driver support for XP X64 and Vista and is available as a free download for UAD customers. www.uaudio.com

Schoeps plug-in

RACK 'EM UP! - introducing 3 blue building blocks... Meet us at Frankfurt ProLight+Sound - Hall 5.1, B88

LYDKRAFT 18

www.tube-tech.com resolution

Schoeps has announced a free VST plug-in for double M/S surround recordings. Only three microphones are needed for double M/S surround: two cardioids with radial pickup, one of which points to the front, the other to the rear and one fig-8 pointing to the sides. They form two backto-back M/S recording systems with a common S-channel microphone. The microphones shown are one CCM 8 fig-8 and two CCM 4V cardioid compacts. More can be read about the topic by following the link below and those that register will receive an email with the plug-in attached. www.schoeps.de/dmsplugin.html

March 2007

gear RSS/Cakewalk REAC co-op

Cakewalk and Roland Systems Group have announced REAC (Roland Ethernet Audio Communication) Recording Technology for Sonar. This enables the recording of 40 channels of 24-bit/96kHz audio from a split of the RSS S4000 Digital Snake system by a direct connection to a PC’s Gigabit Ethernet port. REAC is the audio transport protocol developed by Roland for point-to-point 24bit/96kHz, low latency digital audio transport over Cat5e cable. www.rolandsystemsgroup.co.uk

Red switcher The RB-OA3 is a 1U rackmount, unity gain on-air switcher, capable of switching four stereo pairs between three studios. Each studio can control the transmission path, two peripheral paths for equipment, such as a codec or hybrid, and a ‘Last studio to offer’ bus, allowing for seamless and continuous broadcast from any multistudio radio network. A ‘sustain’ mode also allows for a sustaining system, such as a PC automation system, to control the broadcast. More switchers can be connected together to switch up to six stereo channels between five studios. The switching is achieved using relays, except the ‘last studio to offer’, which is switched by an analogue switch and the transmission path is switched using latching relays. This means that if there is a power failure to the unit, the transmission path will remain selected. The RB-SSML1 is a 1U rackmount contribution unit

comprising a source selector for compressing or limiting an incoming microphone or line signal; along with selectable level metering and headphone monitor outputs. It provides the facilities often needed in news-booths, voice-over booths and at remote studios for recording into a PC. www.sonifex.co.uk

Radial solutions The Radial Phazer is a 100% discreet Class-A analogue phase control that allows two sound sources, such as a microphone and direct feed, to be combined by time-correcting the two sources. Phazer can be inserted in series with any line level device and can also be connected via the insert bus using balanced XLR cables. The Radial JDX is a guitar amp direct box with reactive speaker emulation. The JDX connects between the guitar amplifier head and the speaker cabinet to pass along the direct sound of the amp to the mixing console. By combining the miked sound of the cabinet with the direct feed and using the Phazer to line the two sources up, a variety of tones are possible.

The S3 possesses quite astonishing amounts of headroom, and will quite happily put out peaks approaching +30dBu all day long without sounding the least bit flustered. Jon Thornton, Resolution Magazine

I used the S3 on a drum subgroup and it allowed me to shape the drum space with an amazing amount of control in a way that I never would have been able to do with EQ or full bandwidth compression. Thom Monahan (Engineer), Tape Op Magazine

Forest Audio’s Q6 counter coil equaliser combines passive and active stages in each band. The front panel features six separate EQ bands, each of which is equipped with a 12-position boost or cut control. These coil-based passive circuits overlap and individual EQ bands have a bypass. A high-cut filter is also included. Drag Control load correction simulates the relationship between an electric guitar and a valve amplifier and a separate 5MOhm input buffer is also provided. www.radialeng.com

Duende Drumstrip SSL’s Drumstrip is a drum processing plugin for Duende. Features include Transient shaper with Transient Invert to bring the attack phase of drums to life, Transient Shaper Audition enables you to isolate and monitor the effect of the Transient Shaper, a dedicated drum Gate with range and independent open and close threshold controls and high and low frequency enhancers. There’s also the classic SSL Listen Mic Compressor with additional bypass function to alter the band-limited compression to full-range compression. The order of the processing can be changed and input and output level metering with additional RMS meters are included. www.solid-state-logic.com

March 2007

The new Signature Series S3 from

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gear Anamod tape sim Anamodâ&#x20AC;&#x2122;s ATS-1 is an analogue tape simulator that is entirely analogue and is the first to be designed by a new technique called the Anamod Process. Mathematical analysis similar to the type used to create DSP plug-ins is used to design analogue audio processing hardware, eliminating the need to digitise the audio. The ATS-1 Analog Tape Simulator has all the

important features of a 2-track analogue tape machine, including speed selection, bias, low frequency and high frequency EQ, and the option of adding tape hiss, if desired. It precisely models the behaviour of specific tape machines and tape formulations, and allows the user to select from up to four machine and tape types. Optional machine and tape types will be available on easy-to-install SIMM cards. Unlike digital plug-ins or processor-based outboard gear, there is no latency, and no A-D or D-A conversion. Anamod LLC is a new company founded by Dave Amels (Bomb Factory Digital and Voce) and Greg Gualtieri (Pendulum Audio). www.anamodaudio.com

New Lex

The Lexicon MX500 FireWire dual stereo/surround reverb processor is a 4-in, 4-out device with â&#x20AC;&#x2DC;hardware plug-inâ&#x20AC;&#x2122; technology via Lexistream â&#x20AC;&#x201D; an audio streaming FireWire technology and VST/AU interface for use with software recording platforms. This allows users to have complete plug-in functionality with up to 96kHz streaming audio from their MX500 within any VST or Audio Units software environment, enabling users to control an MX500 exactly as they would a software plug-in. The MX500 also offers 4-channel surround algorithms.

The MX300 is a stereo reverb effects processor with â&#x20AC;&#x2DC;hardware plug-inâ&#x20AC;&#x2122; technology via a USB connection and VST/AU interface for use with software recording platforms. www.lexiconpro.com

A-T camera-mount

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Think outside the box! WE DID and have delivered you a media platform for the 21st century. DREAM II is a family of integrated audio mixers, recorders, editors and HD video recorders. The groundbreaking Crystal Core engine used in DREAM II deploys cutting edge FPGA (Field-Programmable Gate Array) technology to redeďŹ ne the price/performance ratio of digital audio production. With up to 230 super hi-resolution audio channels each equipped with 8 bands of EQ, 3 Stages of Dynamics and a fully integrated 192 track disk recorder, DREAM II puts the power back in your hands. Call your Fairlight Business Partner today and ask for a demonstration. UK Tel: +44 01701 620 090 â&#x20AC;˘ Germany Tel: +49 30 25 92 4460 France Tel: +33 1 49 21 2000 â&#x20AC;˘ Head OfďŹ ce Tel: +61 2 9975 1777

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A u d i o - Te c h n i c a â&#x20AC;&#x2122;s 1800 Series dualchannel cameramount UHF wireless system is designed f o r b ro a d c a s t e r s and videographers r e q u i r i n g simultaneous operation of two microphones. It provides UHF reception with up to 996 frequencies user-selectable in 25kHz steps. The ATWR1820 battery powered diversity dual receiver features two completely independent receiver channels in a single unit that mounts on a camera. Two balanced outputs enable signals to be mixed, or independently assigned, while independent audio level controls provide mixing and monitoring options. It comes with a choice of body-pack and plug-on transmitters. The ATW-T1801 UniPak body-pack transmitter has a locking 4-pin connector while the ATW-T1802 plug-on transmitter has a locking XLR connector for adapting wired dynamic and condenser microphones to wireless operation. Both transmitters are powered by two AA batteries and offer selectable high (30mW) and low (10mW) transmission modes to conserve battery life or maximise RF power output. T h e AT H - M 5 0 i s the companyâ&#x20AC;&#x2122;s a new flagship pro monitor headphone. A closedback design employing A-Tâ&#x20AC;&#x2122;s proprietary 45mm large-aperture drivers with neodymium magnet systems and CCAW voice coils, it has a collapsible design for portability and storage. The circumaural earpieces swivel 180Âş for single-ear monitoring. The AT2010 handheld cardioid condenser is a stage companion to the AT2020 studio model. It shares the large, 16mm, low mass diaphragm of the AT2020 side-firing mic. www.audio-technica.co.uk

March 2007

gear Dynaudio expands BM family Dynaudio Acoustics’ next generation BM range of speakers are represented by the BM 6A MK II and BM 12A, which offer more power, more headroom and wider frequency response than the originals. The BM 6A Mk II is a 2-way active nearfield with a 6.9-inch woofer and 1.1-inch soft dome tweeter. Onboard LF/MF/ HF placement filters and subwoofer high-pass filters are included along with an input sensitivity switch (-10/+4dB), a clip indicator LED, slow attack optical HF protection and thermal amplifier protection. The BM 12A is a 2-way active nearfield monitor with an 8-inch neodymium magnet woofer and 1.1-inch soft dome neodymium tweeter. www.dynaudioacoustics.com

Cardioid FlashMic

Firefly FireWire

HHB has expanded its FlashMic range with the introduction of a cardioid version — the FlashMic DRM85C — to accompany the original, omnidirectional FlashMic DRM85. Both FlashMic models share the same 1Gb Flash memory, USB audio data transfer, a high quality preamplifier with manual or automatic gain control (AGC), an illuminated LCD display, and nine user templates that can be configured externally using FlashMic Manager software. www.hhb.co.uk

Phonic’s Firefly 302 FireWire interface is a half-rack unit with an XLR mic input, two ¼-inch inputs, two phono inputs along with two ¼-inch outputs, two phono outputs and a headphone output. It offers 24-bit/96kHz performance, dual FireWire ports, MIDI I-O, SPDIF, 48V phantom power and is bus or AC powered. It ships with Cubase LE software and is described as ‘plug and play’. www.phonic.com

True P-Solo Ribbon True Systems’ P-Solo Ribbon is a single-channel microphone preamplifier that uses the circuitry of its more expensive multichannel preamps with tweaks to maximise the performance of ribbon and dynamic microphones. The P-Solo Ribbon uses the same balanced, transformerless circuitry, is built with military-grade, handmatched components and delivers a response that dips down to 1.5Hz and reaches up to 500kHz. Features include high-pass filtering, dual analogue outputs, high-impedance instrument input, and four-level metering. It has no phantom power. www.true-systems.com

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Audix DP-7 drum mic pack The DP-7 from Audix is a seven-mic package for miking a 5-piece kit in live and studio applications. The package includes the D6 dynamic for kick drum, the i-5 snare drum, two D2s for rack toms, a D4 for floor tom and two ADX51 condensers with pad and roll-off for overheads. Additionally there are four D-Vice rim mounting clips, which eliminate the need for mic stands on snare and toms. The mic kit is packaged in an aluminium case. Accessories for the DP Series include right-angle XLR mic cables, the D-flex mounting clip, the D-Clamp clip for lug nut-mounting, and the Stand-KD pedestal mic stand for use with bass drum and guitar cabs. www.audixusa.com

ATR Magnetics tape ready ATR Magnetics says it is ready to supply analogue mastering tape and will meet the market demand following the discontinuation of tape manufacture by Quantegy (News p4). Its first formulation, ATR Master Tape, has completed beta testing and the plant is ramping up to full volume production and building inventory of 2-inch, 1-inch and 1/2-inch reels. ATR Magnetics says it has modified the classic manufacturing process of how magnetic oxide particles are disbursed into the mix and its new methods have directly lowered mid-band noise and improved magnetic field switching for an ‘enhanced depth of field listening experience’. The process has also resulted in higher operating levels, according to the manufacturer. www.atrservice.com

March 2007

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review

RSS Digital Snake The concept of a digital multicore or ‘snake’ is certainly not a new one and anybody who has ever had to lug and string a large analogue multicore around will generally agree that it’s one form of aerobic exercise that you could easily live without.

HE RAPID UPTAKE of digital consoles for live production has been accompanied by a variety of digital snake and remotely controlled mic preamplifier solutions. For those still firmly in the analogue domain as far as mixing is concerned, there has been little choice in terms of a straightforward, robust solution that just plugs and plays. Enter, from the slightly unexpected quarters of Roland, the RSS Digital Snake. In its simplest form, the system has three main components — the S-4000S 40-channel I-O modular rack, the S-4000H Front of House unit, and the S4000R remote controller. Starting with the bulkiest component, the S-4000S (which for the sake of clarity I’ll refer to as the stagebox from now on) is a 6U rackmount. Most of the front panel is taken up with a series of I-O modules. The standard system ships with eight SI-AD4 modules, each of which provides four balanced inputs on XLR connectors, giving 32 inputs in total. Each input features a wide range preamplifier, capable of dealing with signal input levels from -65 to +10dBu. Each input also features three LED indicators, showing the presence of phantom power, signal and clipping (after A-D conversion). Two SO-DA4 modules are also

JON THORNTON

provided as standard, giving a total of eight balanced outputs at line level — again on XLR. The rear panel is given over chiefly to three substantial cooling fans, mains input, and a DC power input connector for use with an optional external power supply. Effectively then, what you get is a 32-channel microphone preamplifier, coupled with 32 channels of A-D conversion. The heart of the Digital Snake system, though, lies in the transmission protocol developed by Roland. Based on Ethernet technology, and dubbed REAC (Roland Ethernet Audio Communication), this allows for the transfer of up to 40 channels of digital audio (24-bit/96kHz) down a single Cat5E cable. Transmission is bi-directional, and maximum cable length is 100m, although this can be easily extended by using standard Ethernet switches as ‘repeaters’ up to a maximum of 500m using four switches. Crucially in live applications, system latency is low with deviceto-device latency being around 0.4 milliseconds. Two REAC ports are provided on the front panel of the stagebox — one as the main connection, and the other as a redundant backup (of which more later). Connection between the stagebox and the Front of House (FOH) unit is via a heavy duty Cat5e crossover

resolution

cable. Roland supplies these as options on diminutive little cable drums, terminated in very rugged Neutrik EtherCon connectors. If ever you need confirmation of the advantages of such a system, you certainly get it as you saunter along carrying two of these drums under one arm — that’s 200 metre’s worth of multicore! The FOH unit is functionally the mirror image of the stagebox, giving 32 balanced analogue outputs and eight analogue inputs — so in essence the system is a 32 up/8 down multicore. Remembering that mic preamplification is performed at the stagebox, the inputs and outputs on the FOH unit appear at line level in blocks of eight on 25-pin DSub connectors, so break-out tails will usually be needed although this does make the unit less bulky than the stagebox. Two thoughtful touches are the provision of cable tie-bars in front of each of the D-Sub connectors, and the pin-outs for them are silk-screened to the front panel of the unit (after all, the chances of finding the manual in an emergency are probably going to be slim). Both the stagebox and the FOH unit have a Mute All button, which if pressed for more than a second or so will mute all inputs or all outputs respectively. The final piece of the jigsaw is the S-4000R remote control. This plugs into a 9-pin remote port on either the FOH unit or the stagebox, and it can quite happily be hot-swapped between the two. The remote unit allows gain levels, phantom power and pad switching to be set for each of the input channels, in addition to providing metering for each of the channels. Operation is straightforward enough — a channel is selected with a rotary knob, its number appearing on a numeric display. A rotary encoder then allows gain setting, and buttons select pad or phantom power for the channel. Signal level for the channel (in dBFS post A-D conversion) is monitored via an eight-segment LED bargraph, while an overview of signal present and clip is provided for all channels via an LED matrix at the top of the unit. It’s also possible to link channels together so that adjustments can be made in stereo pairs, and up to 10 complete snapshots of the settings can be stored and recalled. So far, so good and in the basic configuration described above, the Digital Snake is incredibly straightforward to set up and use. Audio quality is exceptionally good. I was particularly impressed by the quality and low noise of the preamplifiers and the total absence of any zipper noise when adjusting gain levels. But for serious live use there are still a number of issues that need reporting on. The first, as mentioned earlier, is latency and I can report that it is minimal. Compare input to output and sum them together and there are phase artefacts in evidence — but in general use this isn’t going to be an issue. The second is build quality, and with the possible exception of the rack ears used to mount the remote panel, which flexed quite alarmingly, it appears absolutely rock solid. The review units from Roland arrived tidily packaged in flightcases, and seem eminently roadworthy, although quite weighty when stacked together. The last, and by no means the least, is system reliability. If there is one advantage to a conventional analogue multicore, it’s that if a connector or cable fails, you only lose the one channel. With a digital system, if the Cat5 cable fails it’s something of a show-stopper. With this in mind, Roland has built a level of redundancy into the system, in that the FOH and stagebox can be connected together using two separate Cat5 cables. Should the primary cable fail, the system will switch noiselessly and automatically to the back-up. Sounds great in theory, and in practice it works just March 2007

review as advertised. Setting up the two links and then repeatedly disconnecting and reconnecting the primary link cause no audible glitches whatsoever. And even disconnecting and reconnecting both cables, while resulting in loss of audio, didn’t throw the system into meltdown. On reconnection, there was a second or so of silence, a tiny bit of digital noise, and then normal service was resumed. But perhaps the biggest advantage of the Digital Snake system is the ability to scale it up into something altogether more complex. Getting more channels up and down is simply a case of buying additional stageboxes and FOH units. More significantly, the use of a protocol built around Ethernet means that providing splits from the stage box — for a monitor desk or broadcast feed, for example — is as simple as connecting the REAC output from the stagebox to an Ethernet switch, and then taking multiple Cat5 cables from the switch to multiple FOH units. As there is the potential for conflicting data here — remembering that the protocol is bi-directional — each unit in a system has a switch that sets it to one of three REAC modes: Master, Slave and Split. A master device can send signals to a split or slave device, and receive signals from a slave device. A split device can only receive signals from a master device; it cannot send signals back to it. So, in the scenario above, the stagebox would be the master, the FOH unit at FOH would be a slave, and another FOH unit would be a split device for a broadcast feed. Although the REAC protocol is based around 100 BASE-T Ethernet, Roland recommends the use of 1000 BASE-T switches to minimise problems and Resolution Half Page 7/12/06 17:15 Page possible added latency from the switch itself. Any

gigabit Ethernet switch that supports 100 BASE-T will suffice, although Roland has recently produced its own. Packaged as a 1U rackmount, this features two separate five-port switches enabling the use of main and backup REAC links to all devices. It also features two separate power supplies (with two separate mains connectors!) again in an effort to build the maximum redundancy into the system. In summary, the system seems absolutely bulletproof in terms of build and performance. Operationally it’s a doddle to use, and even complex configurations don’t require you to be a networking guru. Admittedly, there are some little details that could be confusing, such as using crossover Cat5 cables to connect REAC devices to each other directly but straight through cables if going via an Ethernet switch, but even this isn’t rocket science. Quibbles are very few — the flexing of the remote rack ears mentioned earlier, and the fact that all of the units, but especially the stagebox, exhibit a fair amount of fan-noise. Admittedly, this is probably not going to be a problem in live production applications 3 but the flexibility and sonic quality of the system

MIX 1 PART ACTION WITH 5.1 PARTS PASSION

actually means that using the system as a front end for live recording would be absolutely viable — or indeed any application that requires audio distribution and routing over long distances. My single biggest initial quibble was the lack of any digital interfacing — for live and other applications this could be a deal breaker for some customers. But this has been solved by the recent release of 4channel digital input and output modules for the system (2 x AES3 connections per module). The RSS digital snake is certainly not the only system of its type out there and part of me sighed inwardly when I saw yet another transmission protocol being touted. In terms of absolute flexibility of configuration, there are Ethersound or CobraNet-based boxes from various vendors that I’m sure can be made to solve the most complex of install briefs. But for sheer ruggedness, quality, simplicity and reliability — with absolutely no IP addressing anywhere in sight — it takes some beating. ■

PROS

Rugged and easy to use and configure; can use standard Ethernet hardware to configure complex systems; reliability and system redundancy very impressive; very quiet and clean preamplifiers.

CONS

Fan noise could be an issue in some applications.

Contact RSS, JAPAN: Website: www.rolandsystemsgroup.co.uk

As surround sound becomes more widely used, especially in sports, viewers can enjoy all the excitement of being there. But additional audio signal paths demand more console capacity – and with the increased complexity of today’s productions, that could be a problem. Fortunately now there’s a solution. Our revolutionary Bluefin technology more than doubles the signal processing capacity of conventional systems – all on a single card, occupying just a fraction of the space. Even better – it cuts the cost per channel by half. It’s the sort of innovation you’d expect from a company exclusively dedicated to live production and on-air broadcast audio mixing. If you share our passion, find out more at calrec.com

calrec.com

March 2007

Putting Sound in the Picture

resolution

review

Apogee Ensemble The multichannel audio interface must balance the importance of convertor quality, preamp quality, I-O numbers and DAW integration against cost. JON THORNTON says this box boasts a collection of functions from a company that has been previously best known for predominantly one thing.

OR ANY COMPANY designing or manufacturing audio peripherals for DAWs, the current Holy Grail must surely be to make the ultimate multichannel audio interface. It’s a path that many have trodden, some more successfully than others, but along the way most efforts to date have been dogged by the ‘C’ word — compromise. It’s understandable really — there is a price point that the consumer will stand, so what is most important? The quality of the convertors? Should it have microphone preamplifiers or not, and if so how good can they be? How many inputs and outputs should it have? How much monitoring flexibility? And how about ease of use and integration with popular DAWs? Enter Apogee, who until this point has been largely comfortable staying in the domain of doing one of these things (convertors) very well. The Ensemble (UK£1195 + VAT) sums itself up by name alone — it’s Apogee’s attempt to try and do all of the above with as little compromise as possible. A multichannel audio interface using FireWire 400 to

communicate with the host computer, it is currently available for use with Macs only. This goes some way to explaining the styling, which takes some obvious cues from the current Mac G5 and Mac Pro ranges with its aluminium finish and use of pure white LEDs. OK — let me get this off my chest straight way — this unit is scarily bright when you turn it on and the plethora of blue and ultra-bright white LEDS go through their start-up routine. It might look cool at first, but it does get almost distracting after a while. That said, let’s look at what you get starting with the rear panel. Four microphone level inputs are provided, the first two with balanced analogue insert points on separate TRS jacks (post gain but pre AD conversion). Microphone inputs 3 and 4 have a TRS jack connector next to them for high impedance

sources — plugging anything in overrides the XLR connector. Two additional high impedance inputs on the front panel actually do the same thing to the first two microphone inputs — useful if the unit is racked and you need a quick DI. In fact referring to these first four inputs as microphone level is a little misleading, as they can easily be switched in software to operate at balanced line level as well. Another four balanced analogue inputs (line level only) appear on individual TRS jacks, as do the system’s eight analogue outputs. Digital I-O is catered for by a stereo SPDIF I-O and an optical I-O on Toslink connector, which can carry SPDIF, ADAT or SMUX format digital I-O — the last of these sacrificing channel count for higher sample rates. Word clock in and out on BNC with switchable

"Work shouldn’t be this much fun" Ronald Prent - Europe’s leading surround engineer Clients include: Rammstein, Manowar, Lacuna Coil, SimpleMinds, Herbert Grönemeyer, HooverPhonic, Kane,

www.pmc-speakers.com

Omar Hakim & many more The masters of surround monitoring

Image features IB2S Activated 5.1

resolution

March 2007

review termination completes the back panel line-up; not bad for a 1U and all inputs and outputs are simultaneously and separately available. The front panel is comparatively minimalist. The two additional high impedance inputs mentioned earlier are topped by a tiny status LED that shows valid clock source and FireWire communication. A large rotary encoder, surrounded by pin-point white LED tallies, is used to provide gain control for the eight analogue inputs. Pressing the encoder cycles through the inputs, indicated by a set of eight LEDs, while turning it sets the gain level. A set of small blue and red LED bargraphs show signal level on the analogue inputs or outputs, while two further bargraphs show signal present on any channel of the SPDIF or Toslink digital interfaces. Another rotary encoder sets output levels and can control the main outputs of the device or either of the two headphone outputs that appear on the front panel. If that rather extensive list of functions seems somewhat at odds with the number of controls available, it’s because much of the control and set-up of Ensemble is performed in software. Installation of this software (called Maestro, and also capable of controlling other Apogee hardware) is straightforward, and Ensemble will work happily with any audio application that supports Core Audio — no drivers required. In fact, it works rather better with one particular application — Logic Pro — as most of the control options needed for Ensemble are available as a window within Logic itself. This is neat and certainly saves on screen real estate, but the Maestro software is easier to use and navigate in my view, and adds some additional functionality. Two control panel tabs cover the main settings, which include clock source, digital I-O format for the Toslink connections, and whether the front panel meters are showing input or output. Radio buttons set nominal operating levels for the analogue inputs and outputs (+4 or -10 for the line level inputs and outputs, and an additional microphone level option for the first four analogue inputs). In addition, Apogee’s Soft-Limit process can be applied to any of the analogue inputs. The company’s patented UV22HR dithering process can also be applied to any stereo audio path (input or output) selected in a pull-down menu. The second tab here allows gain levels to be set for the four mic preamplifiers, together with phantom power selection and polarity reverse for each. In truth it’s easier to use the front panel encoder to set gain, although both work inter-dependently. A neat feature is the ability to group together any combination of the four controls so that gain levels can be ganged, even allowing for offsets between grouped controls to be set first. A mixing and routing page brings up a straightforward grid for inputs and outputs, which allows any physical input to be assigned to the appropriate input designation of the DAW being used, and vice versa for outputs. Input sources can also be directly routed to physical outputs, bypassing the DAW host completely. And a final panel allows

any stereo signal source to be assigned to each of the headphone outputs, and allows the format for the main system output to be defined. This can be set to ‘None’, which means that the eight analogue outputs are simply individual outputs unaffected by the master level control, or to stereo, 5.1 or 7.1, which uses the first two, six or all eight analogue outputs, and enables the overall level of whatever is routed to them to be raised or lowered with the encoder. This reference to system output also warrants an explanation of the Ensemble’s final trump card — the inclusion of an onboard, software-controlled digital mixer that is optimised for low latency. This enables all of the unit’s hardware inputs to be mixed together with a pair of outputs from the DAW, before heading to the system’s selected main outputs. This certainly helps in combating latency issues when setting up cue mixes for tracking. It’s clear that Apogee has come up with a very flexible and versatile unit — I actually found it hard to think of a feature or function that wasn’t available — and if I did think of something I inevitably found that on further investigation it actually had been implemented. But has this flexibility resulted in the dreaded ‘C’ word? No. Starting with the convertor quality, the Ensemble is up there with anything in its price range and some dedicated boxes that cost considerably more. Auditioned against Digidesign 888/24 convertors and those on a RADAR 24 and they sounded as detailed and solid as the RADAR, which is saying something. It was a similar story with the mic preamplifiers. Fully expecting it to bug out at this hurdle, recording tests with an acoustic guitar and various recording paths revealed a good amount of headroom and lots of transient detail and mid-range definition. It wasn’t quite as good as an external (9098) preamp feeding the Ensemble’s line inputs, but a world away from the microphone inputs on a Digi002, and in fact from the 9098 preamp feeding the 002’s line input. Some tiny issues remain. The mixer software is somewhat idiosyncratic in the way it works — it effectively inserts itself between the DAW and the system outputs, but presents itself in a slightly different manner. And a colleague detected a random clicking sound on one of the headphone outputs from time to time when using it but I was never able to replicate this — admittedly on a different CPU. You could also argue that maximum flexibility would be afforded by having all eight analogue inputs capable of being operated at line or mic level. But these are all relatively minor issues and the reality is that Apogee seems to have got closer than anyone to date to the Quest of a multichannel interface that really can be all things to all men — almost without compromise. ■

Contact

The

Digital Snake

■ Affordable digital multicore ■ Analogue or AES-EBU ■ Uses simple CAT5e cable ■ Controllable from Yamaha FOH desk control surfaces

be Totally prepared

APOGEE DIGITAL, US: Website: www.apogeedigital.com UK, Sonic Distribution: +44 1582 470260

PROS

Highly flexible and versatile routing; good range of I-O options; tight integration with Logic Pro; high quality convertors and mic preamplifiers.

CONS

Mac only; a little bright to look at after a long studio day….

EXTRAS

The Future of Digital Multicore

Apogee has announced a Stand-Alone Mode for Ensemble and with the release of new firmware, users can now operate Ensemble without a FireWire connection to a Mac allowing it to perform as a high-end convertor and mic pre, independent of the computer. Low latency and routing settings need only be configured in Maestro once and Ensemble remembers the settings and all front panel settings, such as mic preamp gain and output levels, are fully functional while in Stand-Alone Mode.

The UK’s Digital Snake Experts 01527 880051 | sales@totalaudio.co.uk

March 2007

resolution

www.totalaudio.co.uk

review

Beyerdynamic Headzone It’s one of the great remaining challenges — how do you reproduce multichannel playback with headphones? ZENON SCHOEPE says you now can.

HILE THERE HAVE been attempts at addressing the issue before, Beyerdynamic’s Headzone is the most convincing commercially available product that delivers multichannel audio via headphones to date. It employs some pretty nifty digital trickery to achieve the impression of discrete out-of-head sources from a fairly standard pair of headphones and you can also move your head within that sound field just as you would in the real world with monitors. That’s what it does but let’s familiarise ourselves with the bits in the box or, rather, the case as the Headzone kit (UK£1500 + VAT) comes in a briefcase-sized bag. The key components are the base unit, a pair of DT880 Pro headphones with a tracking sensor on the top, and a tracking receiver that detects your head’s position. The main unit has a FireWire port, which allows you to program the unit and deliver digital audio to it, and six discrete phono inputs (with a +4/-10 switch) corresponding to the incoming analogue 5.1. At its simplest, you plug your 5.1 feed into the phonos, you plug the tracking receiver into the back of the unit and then position it above your principle video monitor or other central point of reference, you plug the headphones into the main unit and enjoy the experience, hoping that you don’t catch sight of your reflection because the tracking sensor sprouting from the top of your head will make you startle. Base unit front panel controls are Analogue/Digital Input select, a Bypass that gives you a downmixed stereo of the multichannel input, albeit in the same virtual listening environment you created for the multichannel, and a Headtracker Status LED that turns red when your headphones are out of range of the receiver. The DT880 headphones would not have been my first choice — I’d never heard them before — but you have to use them because the system is optimised to the flatness of their reproduction and the tracker unit is built into the headband. You can, however, forfeit the tracking info and plug in any other type of headphone and just get ‘stationary virtual’ multichannel. I tried all my favourite cans on this and had to concede that the DT880s were by far the best for this application. They’re really rather good although I’m not so keen on the velveteen ear muffs. When in full Headtracking mode you can move your head within the sound field although you are limited to around 180 degrees either side of dead 26

centre in front; so you can’t turn your back on the LCR and listen to the stereo rears, for example. There’s a useful button on the top of the headphone band that allows you to re-centre your middle point of reference for instances where a video monitor, for example, is slightly to one side. You program the box via the FireWire interface on a neat piece of software that allows you to alter various parameters. Permanent computer connection is not essential for ongoing use of Headzone as the base unit remembers the settings on disconnection. Within the software you can change the position of the virtual loudspeakers, adjust the room size, your distance from the monitors, and the ambience. There is danger here of flying up your own trouser leg because the degree of adjustability can create rooms that are of no practical use to anyone. There’s a page with metering –- you only get the Input LED on the front panel reddening to warn of approaching clipping –- and volume faders for the individual channels. You have a number of options for the LFE, which has to regarded as somewhat academic given that we are a talking about headphones, but it’s interesting to note that you could employ the LFE input as a comms input in a broadcast situation. The programming is worth experimenting with but the manual does encourage the concept of creating a ‘perfect’ personal virtual multichannel environment, which I think is a little ambitious. My experience suggests that different types of incoming 5.1 required different ‘rooms’ to best represent them in the Headzone environment. Some prerecorded mixes sounded for more ‘around the sides and back’ than others within the same Headzone ‘room’ because they were mixed differently. That’s a different scenario to one in which you are working on your own 5.1 mix because in this instance you have control of the positioning of the sources within that virtual environment. So there are two distinct interpretations — playback and mixing — but that’s not terribly different from using real loudspeakers. The point to remember is that all the issues resolution

pertaining to 5.1 monitoring in a real room with monitors do not go away once you put the Headzone cans on; they are still there and you still have to deal with them. Raising the level of the rear channels for reasons of listening clarity, for example, like you might do with the rear speakers, has exactly the same implications within Headzone. How your Headzone mixes transfer to real monitors is a difficult one to generalise on because it depends on what you are attempting. What I will say is that not-too-adventurous, multichannel sound field mixing transfers quite consistently and, crucially for me, the LR transfers solidly. I was worried that Headzone, with its objective to surround the listener, may have neglected the stereo but I don’t think it does. Of course, you can hear the process and it’s more apparent on certain types of material than others — a thick, no dynamic range, rock track shows it up, for example. It seems to me that the more space and dynamic there is, the more the process likes it. The artefacts are not unpleasant but if you get a track with a steady pan going across the front and you move your head, then it’s quite ‘interesting’; but it would be interesting with monitors too. I will add that your proximity to the process does accentuate your sensitivity to it and I reckon you listen more critically in Headzone than you might to lacklustre monitoring in a poor room. Given that you are so directly enveloped in the sound, it’s a marvel that it works as convincingly as it does. It’s a perfect solution for personal monitoring for a sound operator in a noisy OB truck, a good way to appraise multichannel privately, and its FireWire connection means you can integrate it elegantly with something like Logic by bringing its audio down the cable digitally. It’s an obvious choice for audio folk in the games market. There are connectivity issues; it could do with balanced analogue, independent digital inputs, and FireWire’s not the universal DAW panacea that it could be. Programming the base unit via USB might have been smarter. There’s no getting away from the fact that Headzone uses some very clever digital processing to fool you into thinking that you’re hearing sounds from all directions while wearing headphones. The only thing that’s missing is height information; something that you certainly employ and encounter with real monitoring. I would differentiate Headzone from every other ‘spatially widening’ headphones playback environment I’ve heard purely because it works for me; and that’s coming from someone who is chronically unable to appreciate most psychoacoustic spatial trickery and is suspicious of it. Set up correctly, you can walk a sound around the back of your head — no doubt about it. It’s very, very impressive. You have to hear this. ■

PROS

The best multichannel headphones representation; great flexibility of set up; sounds great.

CONS

Connectivity options; programming via FireWire only.

Contact BEYERDYNAMIC, GERMANY: Website: www.beyerdynamic.co.uk

March 2007

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review

Elysia Alpha It’s a mastering oriented compressor that doesn’t look like anything else and has its own sound signature too. NIGEL JOPSON hits the big Elysian lights as he finally gets his gnarly mitts on a bit of gear porn to review.

T HAS BECOME an article of faith in some mastering circles that a round trip to analogue and back is a necessity to extract the very last drop from a song. The tools for this are the last bastion of no-compromise analogue design, the audio equivalent of hand-beaten alloy panels on custom sports cars. The Alpha compressor, from German company Elysia, is aimed directly at this echelon of mastering. Designed by ex-SPL crew member Ruben Tilgner, the Alpha has attracted considerable attention for its dramatic appearance, with illuminated blue knobs and styling reminiscent of a software plug-in. Style can be as important as substance at the audiophile end of pro audio, just as it is in hi-fi, and — to this reviewer at least — clothing an innovative product in an innovative shell is more inspiring than the fauxvintage approach adopted by some designers. The Alpha (Euro 7990) is a dual-channel compressor with internal Mid/Side encoder-decoder, switchable sidechain EQ, integral signal path EQ, Soft Clip limiter, and additional colouration transformer. The channels run independently or linked, and innovative switching enables direct and compressed signals to be mixed — the so-called Parallel Compression technique — that normally requires a mixer and a deal of tiresome level matching. The compression circuitry uses genuinely new technology, a fully discrete Passive Current Attenuator or PCA: linear current attenuators with variable gain are not unknown in electronics, but I believe this is the first music application. Elysia employs power amp output stage transistors, the 16 discrete devices at the Alpha’s heart are kept at a special temperature by an ‘exclusive heating system.’ The aural result is unlike any type of analogue compressor we have become familiar with: adjusting attack and release times gave some unusual 28

impressions. This can be a very fast compressor, but without the bite normally associated with quick attack times. There are also some nice 100ms+ attack times marked on the control, but I didn’t hear the ‘peak followed by crush’ sound I’d normally associate with such settings. The Feed Forward button switches the sidechain junction ahead of the gain control section for harder compression, and the Autofast button results in noticeably more gain reduction on transients, but seems to preserve slower developing musical passages. I’m reluctant to use the ‘L1’ word because of the negative connotations this may conjure in some minds, but my most immediate thought when first hearing the Alpha was: Ultramaximizer in analogue! One of the Alpha’s strong points is its solid bass management. There’s an internal sidechain section with boost/cut and centre frequency controls, the extreme left and right settings of the SC Gain convert it to low or high pass filters respectively, with the Freq control then operating as turnover rather than boost frequency. With most mastering compressors the tendency is to use a sidechain to cut bass frequencies to avoid excessive pumping, the Alpha is perhaps one of the few where one might wish to experiment with a bit of LP or boost to tame a boomy track. The EQ to the left of the SC controls is similar in operation to the ‘Tilt’ equaliser found on Quad 33/34/66 hi-fi preamps. Rather than a bell boost/cut at the selected Hz/kHz, signals above and below the chosen frequency will be boosted by up to 3dB and synchronically cut by a maximum of 5dB. This is rather too coarse an adjustment when mastering for CD, but using the Alpha in M/S mode the EQ was useful in evaluating compression and taming muddy bass or excessively bright ambience for the Side component. In fact, because the Alpha excels at gentle compression resolution

settings, I found myself using it in unlinked M/S mode most of the time. As the Alpha has such a great built-in M/S matrix, insert points would be a useful added feature. Stereo analogue equalisers are always a pain to accurately adjust for mastering, which is one of the plus points of using M/S. The usefulness of the Alpha to a studio where it was a trophy purchase would be considerably enhanced by outside access to the Alpha’s Mid/Side signals. Most world-class mastering houses already have a matrix in their consoles, so the Alpha’s encoder will never be used by them. In fact, the Alpha is so nearly an all-in-one switching and balancing mastering solution, I wonder if Elysia ever considered building two companion products: a controller and a separate compressor. The Mix control, with it’s associated Direct and Compressed buttons, allows auditioning of the original, compressed or (with both buttons in) a mix of both signals. No unsettling level jumps or clicks intrude when these controls are used, I found it useful to leave the M/S configured Alpha in Active mode, and then seamlessly switch in mid or side, treated/untreated/mixed signals when setting up a track. It just shows that, although analogue is a sunset technology, good designers can still come up with clever ideas. The Alpha’s Soft Clip limiter is apparently designed to protect subsequent A-D convertors from clipping, but I found its curve too gentle to have much usefulness in this regard. The manual warns ‘noticeable distortion appears if you drive this circuit too hard,’ and this is most unfortunately the case. Some mastering engineers at the cutting edge of the loudness war are actually deliberately clipping occasional peaks into their Prism or Lavry convertors, as a means of employing slightly less (or softer) analogue limiting. Perhaps a circuit that emulated this technique would be more useful for owners of less forgiving A-DCs. The Transformer button adjacent to the Clip control engages a Haufe ‘to add that certain amount of iron to your sound’. It must be a different Haufe to those installed in my vintage Telefunken pres as the effect proved rather too subtle for my jaded ears to appreciate. The Alpha is full of relays and beautifully laid out PCBs. I can imagine it being the processor of choice for those in the jazz and classical fields who are now required to bow in the direction of 0dBFS. It will be a suitable tool for mastering engineers who wish to gently caress the songs they are charged with improving, rather than those who need to carry out a full-on audio face lift. ■

PROS

Exemplary build quality; noise-free makeup gain; transparent sound; gentle and useful increase in perceived level when used in moderation.

CONS

Tends to accentuate hi-hats and splattery HF with more extreme settings; no insert point for M/S matrix; impractical Soft Clip function.

Contact ELYSIA, GERMANY: Website: www.elysia.com

March 2007

Studio Legends. Refined over 35 years. AKG® C 414 B-XLII | C 414 B-XLS | Since 1971, the C 414 has been recognised as one of the premier studio condenser microphones. That’s why you’ll hear it on literally hundreds of classic recordings. These latest incarnations represent the pinnacle of this classic design. The stunning clarity and beautifully detailed sound are joined by enhanced features including a choice of five polar patterns and 3 switchable bass filters. The B-XLS delivers superb quality recordings across a wide range of acoustic sources, while for solo vocals and instruments the B-XLII has been specifically designed to capture every detail of the performance. When it’s time for a new microphone, choose a legend.

www.akg.com

Distributed in the UK and Eire by: Harman Pro UK T: 01707 668181 E: info@harmanprouk.com W: www.harmanprouk.com

review

Waves V-Series and MaxxVolume Waves have gone soft — instead of Diamond, the new number one collection is the Mercury Bundle. This contains 91 processors and everything in their range except SSL emulations. Included are the new V-Series and MaxxVolume plug-ins — both presently exclusive to the bundle.

GEORGE SHILLING

OLLOWING ON FROM its modelling of SSL hardware, Waves has turned its attention to the next obvious candidate and its everpopular classic processing represented by the V (Vintage? NeVe?) plug-ins. However, the N-word is not mentioned and the GUI is oddly (but attractively) rusty brown rather than battleship blue. But the manual freely mentions Neve model numbers and the suite comprises two EQs and a compressor. The V-EQ3 Equalizer is modelled on the 1073 and 1066 modules, and like those, the plug-in offers selected fixed high and low cut-off points and a bellshaped curve, with +/-18dB gain. High-pass filtering switches from 45HHz to 360Hz. A useful meter (dBFS) is provided, along with an Output level knob. As with the SSL plug-ins there is an Analog button that turns off the distortion and noise (yes, hiss is added!) characteristics of the modelling, and there is a separate EQ bypass so you can employ the analogue character without EQ. Switching both buttons off is similar to overall plug-in bypass. On an HD Accel each instance uses just 11% of a chip. EQ frequency selectors add points in-between the 1073 values, and the High shelf is selectable between 10 and 12kHz. This top-boost seems slightly more aggressive than the Universal Audio UAD 1073 emulation, but it’s pretty close. The analogue modelling adds a nice subtle syrup to the sound, it’s slightly subtler than the UAD one, it can be indiscernible depending on the source, but there is some sweetness added here. Low end boost seems slightly more solid and slightly less squidgy than the UAD, but it’s big and very warm, and you get the 1073’s 110Hz and the 1066’s 100Hz along with the other frequencies. The bell-curve also seems a little more pokey than the UAD version. Overall this is a fantastic sounding EQ, it’s a little sharper and less furry than the UAD 1073, (although this depends somewhat on drive level). The V-EQ4 models the 1081 module. Like the V-EQ3 the panel layout is reversed compared to the sideways hardware reissues. With the 1081’s far more comprehensive specification, there are no extra frequencies here, this is a straightforward copy. Adding HF shelf boost definitely makes things thinner and more edgy than similar settings on the UAD 1081 which is again furrier, even in SE guise, but of course comparisons should be taken with a pinch of salt, as furriness is level-dependent (Ladies and gentlemen — George Shilling! Ed.) Bottom end is very fat and Neve-like, but overall the UAD seems to add a little 30

more of this, plus some honk in the midrange. I marginally preferred the UAD, but the V is not far off, differences are subtle. The V-Comp Master Bus Compressor models the 2254, and the controls are broadly similar although the layout is different. The Analog control here varies from Off to 25, 50 and 100%. Even at 100% this can be subtle depending on programme type and levels, but these settings can be useful as they introduce subtle hiss and noise. Noise is at about -80dBFS, but chaining additional instances adds 10dB more noise per instance! On vocals and acoustic guitar the analogue-style warmth is immediately evident, with a clean input signal things can be warmed nicely, it never gets horrible, and is pretty convincing. Compression-wise, this sounds as fat and warm as one might hope. I missed the 33609’s fastest compressor release setting which this lacks, and this plug-in is based on the earlier 2254 without variable Limiter attack. But otherwise, all expected controls are present, including the useful De-Ess setting that works great — an excellent subtle de-esser for those who hate de-essers! This is a great sounding plug-in, if you like compression you should enjoy this. MaxxVolume combines four different level-related functions into one easy to use plug-in, combining technologies from Waves’ L2, C1, Renaissance Vox and Renaissance Compressor plug-ins. Central to the plug-in is a large bargraph ‘Energy’ meter. The Low Level Compression brings up the level of quiet signals, much like an upward expander, while the High Level Compressor works more conventionally. The Leveler is a slow acting compressor that is reminiscent of cassette recorders with auto-gain recording level. There are no controls for attack or release timing on any section, and the High Level compressor does have a tendency to pump if pushed. The Gate is actually a mild-mannered downwards expander, so it is more forgiving than might be assumed. This is probably more of a tool for broadcast and post than for music. However, as a quick fix for reducing dynamic range on any signal, it works pretty resolution

well, and set accordingly can give a desirable boost to individual sounds in a mix situation, or keep a handle on dynamic signals when recording. The graphics of all these plug-ins are excellent and clear, even better than other Waves plug-ins, although the V-Comp and MaxxVolume take up an unnecessarily large amount of screen space — Waves seems to have adapted its designs to match trends towards larger screens. All of Waves’ software is rock solid, and if you can put up with slightly tricky authorisation processes and the much-maligned WUP, these are super tools. ■

PROS

Great sounding emulations; comprehensive multiple formats supported and good compatibility with different DAW hosts; large easy-read graphics.

CONS

WUP expensive; Mercury Bundle enormously expensive; these plug-ins only available with Mercury — real 1073 is cheaper and probably lasts longer…

EXTRAS

The Mercury Collection from Waves features 91 processors with more than 200 component plug-ins. It includes the entire Diamond bundle, the L-Series Ultramaximizers and Multimaximizers, the GTR Guitar Tool Rack, the complete 360 Surround Tools collection of 5.1 processors, Waves Tune pitch correction and DeBreath breath eliminator, IR1 and IR360 parametric convolution reverbs, Z-Noise dynamic noise reduction, the complete Renaissance series, Q-Clone, the X-Series of archive restoration tools, and the Transform Series. The MaxxVolume plug-in combines technologies from the company’s L2 Ultramaximizer, C1 Parametric Compander, Renaissance Vox and Renaissance Compressor for music production and broadcasting.

Contact WAVES, ISRAEL: Website: www.waves.com UK, Sonic Distribution: +44 1582 470260

March 2007

review

Radial JDV Built like a very small brick outhouse and sporting features that you normally wouldn’t expect to find on a DI, the JDV isn’t just another DI, it is a DI that you can compare others to.

ZENON SCHOEPE

NE OF THE DOWNSIDES of the pricepoint-hitting malaise and far eastern ‘cost saving’ manufacturing is that some of the ‘feel good’ factor, which has always been part of the gear experience for me, has been eroded. I encounter boxes that work fine and sound fine but are just not special enough to get me excited about them. The humble DI has suffered more than most and become a rather unremarkable and utilitarian sort of affair that your sales man will quickly justify by asking what you would expect for the price. For what is in essence a pretty straightforward electrical procedure, it is surprising that DIs are so hit-and-miss. While many of us can probably name a DI that we prefer, many more will be able to mention instances where they’ve encountered ones that have underachieved. It’s all the more annoying when so small a component can give rise to issues that really shouldn’t be issues at all because and a bad DI can introduce noise, change the fundamental character of the connected source and also do strange and unpredictable things to the dynamic range and attack. It’s actually the last of these I find most irritating because I don’t want to hear a DI and I don’t want it to impart an imprint — there are too many other boxes vying for that particular duty. It’s perhaps significant that a number of mic preamp-type outboard processors are blessed with critically acclaimed DIs. That‘s good, but for me there are times when a small box on the floor is all it should take and all it should need. I feel better just by looking at a Radial product because it reminds me that I want to be surrounded by exceptional gear; proper stuff that does the job and will continue to so. If you’re aware of Radial from Canada then you’ll know it builds a very wide range of versatile interface-type products. It’s quite a traditional outfit in this respect because very few other manufacturers can be bothered to concentrate quite so much effort on such a specialist and professional portfolio. Build quality is quite remarkable as the devices employ a type of ‘book-cover’ construction that has a super-tough metal deep ‘U’ serving as the wrap around cover to the innards, leaving the controls and connectors to be arranged on the three remaining sides; additionally protected by a generous overlap of the ‘cover’. They’re weighty and have soft rubberised bases so they don’t slide around. They look like they will last a lifetime. One of the benefits of the construction method is that certain units in the range can be stripped of their ‘book-covers’ and slotted into a rackmount, complete with power, for instances where multiple units are required. The JDV (US$449) is now in Mk3 guise and is a staple of the range. Radial chooses to split the controls into sections for input, the musician and the engineer in its description of the box and this happily coincides with three panels. The input side has LED indicators for power, signal 32

a n d overload, followed by switchable 80Hz high-pass and 7kHz lowpass filters and a switch that activates an exceptional 3.9MOhm input impedance or fully variable ‘Drag’, or loading, control on a pot. Because there are two jack instrument inputs, you get to select between them on a switch. There’s a direct Thru passive output and two ClassA active Aux outputs for effects or other amps, plus a tuner output. This is already far in advance of most DIs and gives you many more possibilities, whether that’s for visiting musicians or your own recording guitar set up. A speaker input pad switch means you can also take in a post-amp signal. Finally the side that the engineer will be most interested in has an XLR balanced mic level output and switches for polarity, a -15dB pad (you may well need this as the box is capable of remarkable output), and a ground-lift. Power comes up through an external 42V DC supply. You won’t have to worry about the JDV once it is connected up because it just gets on with it. The filters are handy and the Drag control offers useful optimisation for different instruments. Best of all, it is transparent and just gives you a good wide and strong representation of what you’re plugging in; then it’s all in the

fingers. What I like about the Radial range is that many of the boxes are complementary and you can apply them together to sort a problem or to enable a creative solution. Radial is big on DIs, splitters, and re-amping and the recent introduction of the JDX Reactor guitar amp and cabinet direct box and Phazer variable phase control (see Products p19) are particularly interesting. If you haven’t thought about your DI for a few years then it is time to see how things have moved on. No one is putting as much attention and thought into developing the concept as Radial is. Frankly, if you’re up for a DI then this is where you start and end. ■

PROS

Transparent; great I-O capability; very flexible.

CONS

Nothing.

Contact RADIAL ENGINEERING, CANADA: Website: www.radialeng.com

X-Amp

The concept of re-amping, whereby a recorded signal is sent via an interface to an amp and speaker combination for rerecording, is not a new one. However, it needs to be done properly if you want the full benefit of delaying your creative decisions to a later time. What the X-Amp does is give you two outputs on 1/4-inch jacks — one is a direct feed the other is isolated — so you can double your possibilities. Power comes from an external 15V DC supply and the input comes in on balanced XLR and has a ground lift switch. The isolated output gets a polarity switch, which is a nice touch, and a ground lift operates on both. Essential to the whole process is the presence of an output level pot with a clip LED. It works exactly as expected and it doesn’t corrupt your signal while it’s doing it.

resolution

March 2007

AWS 900+ - GearBox, UK

Duality - Studios Piccolo, Canada

redefining the music console. again.

Real magic happens outside the box.

X-Rack - SSL SuperAnalogue magic in a rack

Over three decades, SSL E, G & J Series analogue consoles have been the first choice of the professional recording industry. With our latest console innovations, Duality and AWS 900+, we deliver SSL analogue magic to your studio DAW system. Record with SuperAnalogue™ mic pre’s (and the new VHD pre), sum your mix to an SSL SuperAnalogue™ mix bus, use legendary E, G & K series EQ & Dynamics and recall your mix settings effortlessly with Total Recall™. Installed in 300 locations worldwide, the compact AWS 900+ offers 24 SuperAnalogue™ channels and established new standards in console and DAW integration. The large format Duality, available in 48, 72 & 96 channel frame sizes, adds even deeper DAW control and a highly-evolved feature set including a dual signal path & routing system enabling you to patch SSL analogue channel processing directly into your DAW. To find out more about the new standards in music consoles, visit www.solid-state-logic.com

Oxford +44 (0)1865 842300 New York +1 (1)212 315 1111 Los Angeles +1 (1)323 549 9090 Paris +33 (0)1 48 67 84 85 Milan +39 039 2328 094 Tokyo +81 (0)3 5474 1144

www.solid-state-logic.com

For a more modest budget, X-Rack is a modular system featuring SuperAnalogue™ processing circuitry and Total Recall™. Six modules are now available making it possible to configure X-Rack as a rackmounted mixer & analogue summing system, or as a standalone processor for studio & live applications. • Eight Input Summing Module • Four Channel Input • Master Bus • Mic Amp • Channel EQ • Dynamics Module

review

TFPro P9 TFPro is offering a new range of outboard equipment options. GEORGE SHILLING discovers rich analogue warmth for tracking and mastering.

LTHOUGH THERE ARE more expensive rivals, this new dual channel mastering EQ is certainly priced in the pro bracket when compared to other outboard studio EQs. However, you pay extra for mastering versions with switched knobs, and the P9 (UK£1750 + VAT) employs these throughout. The house style is two-tone blue, using similar knobs to those found on Manley gear. Although the knobs are obviously quality components, having 24 of them packed tightly into 2U seems slightly disproportionate and makes the unit appear cluttered. That said, the labelling and legending is clear and, impressively, its designer Ted Fletcher also squeezes in 14 toggle switches and a pair of 8-segment LED meters, plus — generously — two power On LEDs! The rear panel includes inputs and outputs on XLRs, plus a bonus pair of TRS jack outputs 6dB down on the main outs, perhaps useful for supplying external zero-latency monitoring when recording direct to DAW through the unit. The two channels each feature four EQ bands, high and low pass filters, and input and output Level controls. Each EQ band provides four fixed frequencies, and each associated gain knob uses a toggle switch to select Lift or Cut. Unfortunately there is no bypass for each band — perhaps a middle position on the toggle switch would have usefully provided this. I did discover a workaround though: when setting the frequency of a particular band, it is possible to lodge the knob between frequencies, at which point the band is bypassed. However, I also found a problem with the lowest frequency band on each channel. The in-between positions on this band caused a full-band boost of around 8dB — a potential irritation if you are monitoring loud. This was pointed out to the designer during the course of the review, and was acknowledged as a minor problem that will be remedied on future production models. For the EQ bands, Ted has based his design on the late Barry Porter’s Trident A-Range EQ. These were simple designs, all-discrete Class A, but with only just enough open-loop gain to work sensibly, according to Ted. Thus, if you push an A-Range gain stage, they soon go into 2nd order distortion — perhaps the reason these old consoles are so loved by some. The P9 employs more elegant ‘constant current’ transformer balanced loads for the front ends improving linearity at high levels and avoiding extremes of distortion, but retaining the tendency towards 2nd order distortion. To this end, separate input and output Gain knobs allow variable drive through the unit. These gain stages remain in circuit whether or not the EQ is in. 34

Also featured are high and low pass filters that stay in circuit when the EQ is bypassed — this is sometimes slightly inconvenient. The filters are set at 25, 50 and 100Hz, and 15k, 12k and 8kHz respectively; the first two positions of each are usefully subtle. This is desirable, as aggressive filters are rarely helpful in mastering. Each band’s boost/cut knob uses an extended scale at the lower end, so half-dB steps take you to 2dB, then 1dB steps go up to 8dB, and the final position is 12dB boost or cut. This low-level precision is most welcome as the Q seems fairly broad. Small amounts of boost (or cut) are more obvious than one might expect. Midrange boosting has a warm, sweet and rich effect. The two mid bands boast a toggle to select wide or narrow curves. Wide seems, well, fairly wide. And even when switched to narrow, the perceived effect is not surgical, rather there is still a broad change going on, albeit with a pointy narrow peak on the top at the chosen frequency. The Low Mid band’s four frequencies range from 450Hz to 2kHz and I thought the low mid should extend a little lower. With some juggling using the low band and the high pass filter, the desired effect can usually be achieved, but not always. Across one mix I found reducing 450Hz affected the vocals too much, when what I wanted was to take some of the boom and ‘clogginess’ out of the bass and instruments while enhancing the very low end. The Hi Mid’s lowest frequency is 2.8kHz, so the Lo Mid band is in demand also for its 1.2kHz boost, a small amount of which adds a nice touch of ‘analogue’. The High shelving has such power available to it that you are in danger of emphasising harshness with the lower pair of frequencies (8k and 10kHz), but slightly boosting either of the upper pair (12k and 15kHz) can sound like a blanket being removed from the tweeters. The low end shelving band is a simple Baxandall filter with a corner frequency from 40Hz up to 250Hz, with 75Hz and 125Hz in between. This can add enormous warmth and weight, with the low filter keeping things in check if necessary –- even at 25Hz this tightens the bottom end nicely. When looking for outboard EQ, the choice is resolution

smaller than with compressors, so a unit like this is most welcome. However, the broad swathes and perhaps excessive power of tone change available seem slightly at odds with the apparent precision of the controls. On the plus side, the P9 can sound rich and warm, with gentle boosts and cuts. Crank any band past 2dB and some frequencies can sound somewhat aggressive and this makes it powerful for tracking applications, such as guitar EQ, although it lacks the incisive precision for rescue jobs and surgery. Nevertheless, it sounds big and, used subtly, the P9 adds a rich analogue warmth and tone for tracking and mastering. ■

PROS

Characterful EQ; easy to make subtle and/or broad changes; recallable settings.

CONS

No separate band bypasses; no output level fine trims; level controls and filters active in EQ bypass; a few more frequencies would be welcome; all knobs the same colour and size; ‘Mastering’ price!

EXTRAS

The TFPro range also includes the P38 VCA/optical stereo mastering

compressor, the M16 16-channel transformer based mic/line preamp, which can also be used as a 16:2 summing mixer with individual gain and pan controls, and the P10 Mighty Twin dual mono channel path mic/line/DI input stage with VCA/optical compressor and 4-band EQ.

Contact TF PRO, UK: Website: www.tfpro.com Unity Audio: +44 1440 785843

March 2007

review

SE Electronics R1 Adding to your ribbon mic choices, SE has weighed in with a typically competitively priced quality performer. JON THORNTON takes the ribbon from his lair, shakes it loose and lets it fall…fortunately it will be repaired.

O WHAT’S GOING ON? After trawling through a whole bundle of new ribbon mics for the last issue, it seems that the ribbon renaissance is in full flow, as SE Electronics’ contender landed on my desk in a sea of bubble wrap. Taking this as an indication that I should exercise additional care in opening the package, the bubble wrap gave way to a compact hard-shell case containing the R1 microphone, an elastic suspension mount and that’s about all. [R1s now come in a black aluminium flightcase with a shockmount and 2m cable. Ed]. The lack of any accompanying documentation had me scratching my head for a while, as I didn’t know what might be lurking in the R1’s matt grey body in terms of fancy electronics or output buffers. Deciding that prudence was the order of the day, phantom power was switched off from a channel of the console and the R1 was plumbed in for a guitar tracking session that had started to go just a little pear-shaped… It turns out that the prudent approach was the correct one, as the R1 (UK£595 +VAT) is a classic ribbon design in every sense of the word. At its heart is a 1.8 micron aluminium ribbon, matched to a simple passive output stage. Externally it’s finished in the same battleship grey colour scheme as SE’s large diaphragm condensers. A slatted grille lends a distinctly retro look to the microphone, and internally it seems that there’s been some attention given to providing additional blast protection for the delicate ribbon assembly. So, back to that session. A ribbon microphone wouldn’t usually be my first choice for recording acoustic guitar — the low sensitivity and early HF roll-off often don’t quite capture the

attack of the sound. However, the guitar in question was being played in Nashville Tuning, where the lower four strings are replaced with strings of a narrower gauge, and tuned an octave higher than usual. This is a great technique for getting rhythm sounds to really cut through a mix, particularly when doubled with a guitar in standard tuning. Unfortunately, ‘cutting’ was the operative word in this case and not in a good way. Despite throwing all of my usual suspects at the problem, there was no getting away from an irritating, grating edge to the sound. Out of desperation more than anything, the R1 was brought in. Initially using the desk mic preamps (Audient), the results were a little underwhelming — not helped by the gain cranking needed as a result of the microphone’s low-ish sensitivity and low output impedance (<300ohms). Swapping the desk pres for an Amek 9098 preamp made a big difference — similar amounts of gain needed, but with improved noise and a much more open sound. But the biggest surprise was in the tonality — I was expecting something softer, but probably at the expense of transient detail, in fact the R1 softened the sound considerably while still managing to preserve a level of edge that was exactly what was needed. Part of this is because the R1’s response seems to extend that bit higher than some ribbon microphones before gradually rolling off, and partly the fact that any proximity effect seems minimal. Of course, there is always the temptation when a microphone solves such a pressing problem to brand it as a clear winner, but more investigation was needed. As such, the next port of call was on electric guitar via an Orange guitar cab — again quite a bright, ever so slightly harsh source. Positioned slightly off from the centre of the speaker, and about 15cm away, the R1 did its magic once more — a nicely gritty sound without any hint of harshness. It seemed to pull up the mid-range (about 2kHz) nicely, bringing out some presence in the guitar sound. There was plenty of authority to the lows and low mids as well, but without ever sounding overly boomy. For comparison, a Royer 122 was placed in the same position, and this sounded just that little bit harder, and in all honesty a touch fuller in the low mids than the R1. It’s a subjective call, but in this particular case the R1 won to my ears. Putting a little distance between the guitar cab and the microphones told a slightly different story though — at about 4 feet away both microphones were getting significant ambient pick-up in their rear lobes, but the R1 seemed to lack the focus on the direct sound that the Royer was still delivering, and started to thin out the lower octaves somewhat. With pop-shield firmly in place, male vocals were next on the agenda — both spoken and sung. I’ve recently started to use Royer ribbons on occasion for recording voice-overs for certain male actors, and I was interested to hear the R1’s performance in this application. The lack of a noticeable proximity bump means that, with care and a good popshield, you can resolution

get the talent right onto the R1, giving an intimacy to the sound that counters the slightly attenuated HF response. The result is a voice that oozes depth and warmth. It’s not a transparent sound, you always know that there’s a microphone in the way (Qué? Ed), and the voice has all of the rough edges removed, but it’s a very useable sound. The final outing for the microphone was purely out of interest but as there was a drum kit set up and a willing drummer to bash it I tried the R1 as a single overhead — roughly three feet above the cymbals and slightly forward of the kick drum. In truth, I wasn’t expecting much, as I’ve found other ribbons a little too dark to give the required definition in this type of position, but I was surprised. A nice, balanced sound, not overly bright but with a good sense of definition and attack to the cymbals and toms. Again, it was a touch thin in the 100–400Hz area, but that’s not necessarily bad in this application. The ribbon element in the R1 is that little bit thinner than most other ribbons out there, which probably accounts for this but this might also put off prospective purchasers, particularly if they are new to ribbons and have heard all the horror stories about their fragility. In an attempt to persuade people to make the leap, SE is offering to replace up to three ribbon elements damaged or broken free of charge. I should point out that this isn’t a statement about the microphone’s fragility — indeed UK distributor Sonic Distribution told me they haven’t yet had to replace one — but it should serve to put a few minds at rest. I really was pleasantly surprised by the R1’s performance. I was expecting something much darker sounding and more akin to the multitude of low-cost ribbons currently appearing. It’s priced closer to the Royers and Coles of this world than anything else, and it really does justify it. ■

PROS

Smooth sounding microphone; good HF extension and detail; compares very well with ribbon models from a higher price bracket.

CONS

Slightly less weighty sounding than some other options; can lack a little focus when at a distance from source.

EXTRAS

The USB2200a from SE is a USB2 microphone based on the studio sE2200a capsule. It records via USB (record path 16-bit/48kHz, output path 24-bit/48kHz) directly to DAW. The mic includes low latency headphone monitoring (less than 1ms) with mix control to allow the user to set playback versus record path levels to monitor live takes, 10dB pad, bass cut and an ‘analogue switch’ that enables the user to use the mic via an XLR connector with 48V phantom power. The USB2200a employs proprietary chips. One transforms the 5V power supplied via the USB cable to power the capsule after first removing noise and spikes from the current. A second chip includes the appropriate software drivers for plug-and-play use with any DAW.

Contact SE ELECTRONICS, CHINA: Website: www.seelectronics.com Sonic Distribution: +44 1582 470260

March 2007

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review

Phoenix Audio Nicerizer 16 Regardless of your beliefs about the relative merits of ITB and OTB mixing, the fact remains that enough manufacturers believe there is a market for the summing box. GEORGE SHILLING listens up and nods sagely…nice.

HE NICERIZER SERIES comprises two distinctly different models and the 8 has a slightly different concept from the 16 which we have here. Both are summing devices for out-ofthe-box DAW mixing. The 8 features eight channels with individual level controls, and individual inputs and outputs, with two separate mono buses. The 16 gains a further eight channels, but loses the individual outputs, and the knobs on the front panel for each channel are pan pots, feeding a stereo bus. It should be pointed out that this model is not brand new — it even predates the Neve 8816 — but some improvements have been made since its initial launch, with extra monitoring features and redesigned internals for easier servicing. Phoenix was originally known mainly for retrofitting Neve consoles, but although its panel designs encourage comparison, its circuitry design is considerably evolved from the 1970s ethos. The all-discrete Nicerizer 16 (UK£1945 +VAT) is contained within a 2U case with metallic finish vintage Neve-style knobs and a fairly straightforward layout. The pan pot arrangement mirrors the XLR input sockets on the rear. It would seem probable that most users will arrange the channels as multiple stereo stems, panning alternately hard left and right. The pan pots are loosely damped and lack a centre détente or bypass, so there is a slight concern that the knobs might become inadvertently knocked off their intended positions. Each channel also features a +8dB pushbutton to drive the Class A circuitry a little harder; again there is a slight possibility of error as there are no indicators for these, and their travel is relatively shallow. On the output section, along with a master level fader knob is a stereo width control. This can be bypassed with a toggle but when in-circuit allows continuous variation from Mono via normal Stereo to 38

+25% wider. This is fun, and a touch of extra width can certainly add some life to flat sounding mixes with no perceived unpleasantness with such things as phase. Even in the stereo position, some enhancement is perceived due to the extra Class A stage. The rear panel features main outputs on a pair of XLRs, these are always fed the main programme and are suitable for sending to a 2-track recorder or back into the DAW for printing the mix. Additionally, there are jack sockets for Aux Output, 2-Track Input, and Insert. The Aux outputs take the same feed as the main outs, but with a separate dedicated mini level control on the front panel. The front panel Monitor section is flexible, providing a path with a headphone output with selectable sources from each pair of inputs, LR out (post main output level knob), Prefade signal, 2-track input, and aux. A simple toggle allows switching from normal stereo monitoring into mono, selecting either Left or Right signal to feed both sides. The monitor output, while set up as a headphone connection, will happily drive an external monitoring setup or a talkback system with the correct adapter leads. The bonus of 2-track monitoring is welcome but there seemed to be a slight LR wiring fault on the review model, as although the channel pairs monitored correctly in stereo, the meters were reversed L-R, and when toggled to mono L-R was round the wrong way on the monitor output. The manufacturer has put this down to a one-off wiring mistake. Making scientifically valid comparisons is particularly difficult when the differences are subtle, but I bravely set up a few scenarios to compare inthe-box mixes with Nicerized ones. Simply running audio through the unit in stereo changes the tone subtly, generally bringing a slightly more musical richness to the programme. However, splitting a mix into three stereo buses of Drums/Bass, Vocals, and resolution

everything else, there was a noticeable difference from the in-the-box version. Oddly, the Nicerized version seemed more rhythmic and punchy, taming some low-mid mush and bringing clarity to transients. However, this was an unscientific test so on another day, with another track results may vary significantly. Pushing the +8dB buttons in (and matching levels again) seems to exaggerate the differences ever so slightly, but also the output level is a factor (not to mention how hard you drive your monitoring inputs!) The Nicerizer has enormous headroom, the monitor meters go up to +16 but it will happily go to +26, and the tone of the unit suggests plenty of power in reserve. Perhaps some cost saving could be made by dispensing with some or all of the pan pots. In normal use, most signals will need to be either stereo or mono in the exact middle. Four stereo bus inputs, four mono centred, and four panning inputs would surely suffice most of the time. However, the present arrangement is simple and easy to use, with little possibility for confusion. Using external preamps, the Nicerizer also makes a useful combiner for multiple mic setups in a recording situation. In this instance, the 8 model might be a better solution, with its separate level adjustments, but nonetheless the 16 performs this well, adding some desirable subtle colouration. The subject of ITB or OTB mixing is something of a can of worms, but there can be little doubt that the Nicerizer 16 imparts a desirable sonic character upon all that passes through it. Rivals may offer fancier features, but I think you’d struggle to find a better sounding summing box than this. ■

PROS

Sounds terrific; adds life to your mix; Enhanced Stereo Width option surprisingly pleasant; also a useful recording ‘mixer’ or combiner.

CONS

Monitor section L-R wiring confusion on review model.

EXTRAS

The Nicerizer 8 has 8 channels of Class A, discrete, truly balanced transformerless input and output with individual level control for each channel.

It has 8 balanced TRS inputs, 8 balanced XLR outputs, two mix buses (linkable to each other and other units), two individual mix bus output level controls and LED level metering switchable for each channel.

Contact PHOENIX AUDIO, UK: Website: www.asapeurope.com ASAP: +44 207 231 9661

March 2007

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craft

Mike Oldfield Still producing creatively challenging output many years after his initial album set a benchmark for what a musician could do when left to his own devices in a studio, Mike Oldfield has built a new facility for himself that boasts a Euphonix System 5MC. He talks to GEORGE SHILLING about his changing work methods, classical music, and playing the console and the Mac.

INCE PROVIDING BRANSON’S fledgling Virgin Records with an iconic catalyst in the shape of the first one-man concept album, Tubular Bells, Mike Oldfield has been renowned for creating luxurious soundscapes, famously undertaking most of the hard work alone — composing, playing, engineering and producing. Tubular Bells has been revisited several times over the years, with an orchestral version, live recordings, two sequels and an entire rerecording of the original in 2003, complete with 5.1 mix. The Macintosh computer has gradually come to dominate Oldfield’s working methods, and having recently relocated to the southwest of England, he now uses two adjacent rooms for working, with both rooms connected to the same central computer 40

system. His recently acquired Euphonix System 5MC is in the ‘little monitors, big mixer’ room in a converted garage, while the conservatory is the ‘big monitors, little mixer’ space with a Tascam fader controller. He has co-written a soon-to-be-released autobiography Changeling (Virgin Books), which goes into great detail about his early years, Tubular Bells and The Manor, and his astonishing career path, including childhood difficulties and struggles with depression. He describes doing the book as being like a ‘marathon psychotherapist session’. Your intrepid reporter had had a professional encounter with the man some 10½ years previously, engineering a series of sessions at his Neve Capricorn-based studio in Buckinghamshire, so it was interesting to see how things had changed. (Photos www.recordproduction.com) resolution

How do you work now compared to the old days? All the old equipment got carted away. Gradually over a period of two or three years, more and more I moved over to the digital audio workstation, and one day I realised I wasn’t even going through a mixer at all. I thought, What shall I do with it? It was a beautiful machine, the Neve Capricorn, but it was just not part of the signal path any more. I found I could do everything native in the Mac, even though it was very slow at the time, and used to crash. So I did a deal with a company to come and take the lot off me. Since then I’ve been battling with the Mac, trying to make it be a proper studio. Which it kind of did, but it was too much workload for it. Eventually, the Intel Mac has come out, which is the first proper machine to do the job. It’s a question of arranging windows so that they look like tape machines, and the Arrange Window in Logic looks like my track sheet. So I’ve got a track sheet, kind of a tape machine, plus I’ve got a mixer. That’s why I’ve got two big screens. It’s very difficult to do something with one square screen, so I’ve worked out how to do that. The last album I made completely on the Mac, no faders, nothing. You can do it, drawing points on a curve is different from moving a bunch of faders. I looked into all kinds of possibilities of control surface. One of them is the Tascam which has 24 March 2007

craft faders, and that’s pretty good. And then I looked into the Euphonix System 5MC, which really looked like a real mixer. But then when I first got that I was working on a previous generation of Macs, and that slowed it down. System 5 works beautifully with Nuendo on a PC, and I’ve worked out how to install Windows on a Mac using Boot Camp, so I’ve got one machine which has Nuendo working on Intel Mac running Windows XP, and it works perfectly. And it’s pretty good with Logic, that works on a second Intel Mac, but there are certain things that slow it down like adding channels, deleting channels, packing and unpacking folders. When I’m doing that kind of work, I disconnect the Eucon system from the Mac and it works fine, it never crashes, or hardly ever. Plus I’ve got two places to work now, I can work outside in the conservatory, which is lovely, or if I want to be more concentrated and don’t want to look out of the window at the view, I can work in the next room which has got the mixer in it.

or read about it or listen to anything. I think in the music business these days, the best job would be choreographer or make-up artist, they seem to be doing most of the creative work. It’s all about the image and the dancing. Or a programmer — you’ve got so many plug-ins. People work very hard to make them do special things, and from time to time I catch by accident something that I really like. The musicianship, and I would imagine the engineering side of things has fallen away and become less important, because anybody can do so much. Even without being a musician, with a reasonable piece of software, in a few minutes you can have something that sounds very professional. It’s for that reason I’m trying to make a transition to the classical world, because there’s no two ways

about that, you’ve got something written out, and you’ve got acoustic instruments actually playing it, so it’s not false, it can’t be faked. I’ll try that. If that doesn’t work, I don’t know. It’s actually fun working on computer music, I really like it. On the last album, it was great. You’d dial up a plug-in, find a nice preset, off it goes… Dooga-dagga-dooga-daggadooga-dagga, add a bit of guitar, a pad, it’s great, oh, this is easy now. Then you realise that everybody’s making that kind of music. Go into my local Sound Control in Bristol and over in the corner somebody’s doing exactly that, because it is so easy. I’m in an unusual situation, in that I’ve done pretty much everything I wanted to do. This is the only thing I haven’t done. I can’t do it on my own because I don’t have the training, I don’t want to sit down and wrestle with crotchets and quavers…

What other gear do you use? I’ve got the SSL Duende. It sounds and feels like a real mixer. I’ve got the plug-ins for PowerCore, but that doesn’t work on Intel Macs yet, I’ve only got that working on the card in the Nuendo system. And I’ve got the UAD card for Intel Mac, but the SSL just feels nice. Do you ever play the guitar these days? Oh yes, I do, I’m writing a piece of orchestral music. I’m making a demo of the whole thing, keeping all MIDI parts, so that the MIDI parts can be transcribed into Sibelius, and a proper orchestral score can be worked out. It’ll be all orchestral apart from classical guitar, and grand piano, I’ll be playing those two instruments. For an album? Yes, a classical album, it’s going to be based around the festival of Hallowe’en, rather than the Hollywood horror film. The ancient festival goes back to BC. I’m talking to the composer Karl Jenkins about collaborating on it, hoping to record it somewhere special like Abbey Road No.1, and we’ll do a live concert. The advantage of that is we’ll have an orchestral score, which anyone in the world will be able to pick that up and play it. How do you use the Euphonix? I remember being inspired by seeing you play the Neve like a musical instrument… I can do that now with the Euphonix, but I’ve got used to working just with the mouse really, drawing curves, editing points on a graph. The way I work at the moment is just with the mouse, but then I’ve got the automation. I’ve got to almost re-learn how to play the mixer, but it’s capable of doing it now. It wasn’t until this generation of Mac that it was able to do that. You’d put too much automation, and — Crash! Fingers crossed, I think the machine is fast enough now. Do you always work here or do you have other studios? No, I work in here or next door with the mixer. It’s nicer in the evening in there, it’s lovely. I’ve spent most of my life in dark, dungeon-like studios, I can’t bear to be locked up any more. I want light, I want to see the clouds. How do you feel about the music business these days, compared to the 1970s? I have hardly any contact with the music business March 2007

resolution

craft What motivates you to work? Motivation is a problem, sometimes I wonder whether I’ll continue making music in the future, I’ve slowed up incredibly. I’m able to go in the studio and force an idea out of almost nothing, and make it sound respectable, it doesn’t mean it’s that good. But there are little ideas that pop up, with the frequency of comets or shooting stars. I have to wait until — call it being visited by the muse — these moments arrive, they don’t come very often. So it takes me a lot longer to do something, I don’t want to force and push it, squeeze it out when it doesn’t really want to come.

And it can be a bit frustrating sometimes. So that’s why I’ve arranged my room, I’ve got my nice sofa, I’ve got a few books, if I don’t feel like working I can lie down and read a book. I’ve got other hobbies now, a new child Jake, he comes and plays with his mixer, I read my books, we fly model aeroplanes, I go out on my motorcycles.

Just for the record, Jake uses an 01V96… [Raucous Laughter] Well I started off trying to use that, I made the last album with that, but I don’t even need that with this project.

Are you a musician, engineer, producer, composer, what’s your main role? Since I very first started I just had a picture of what I wanted to hear, and I’d go to any lengths to make that come alive, and in doing that I had to use technology so I learnt how to be an engineer. I naturally just ended up producing, simply by sitting down at a mixing desk and wanting to do the best mix possible, and staying up all night, and the next night if necessary. I know how much can be done in the mix, and that’s why playing the mixer is so important. It’s something I can do again now, rather than trying to mix with the mouse. So I didn’t start out trying to be any particular thing, I just had a sound picture of what I wanted to hear. If I’m anything I’m a guitar player, because I love playing guitar, I started off very, very young and I still love playing guitar. And do you have any collaborators in those other roles? No, I do all the plugging in, installing and authorising software, I do everything and don’t have anybody helping me. In fact, I find I enjoy that almost more — I’m quite happy when I’m trying out that card with that interface, that monitor with that USB down there, thinking if I can just get that keyboard down there, or if I can send that bit of ADAT optical down there and a real buzz when it pops up somewhere else, I love that! It’s just tinkering and I really love doing that. I spent ages plugging it all together. Things like some Mac monitors, they are so fussy, especially if you want over the 1.5 metres you get. I ended up going to PC World and buying these little monitors which you plug in anywhere and they work. I’ve just come off a big tour in Germany with a symphony orchestra. That was quite exciting to be in front of a big audience again, we’ve got two more concerts in Spain in March with the same orchestra. I suppose I’m like the leader of the orchestra. The conductor Robert Groslot, and the leader of the rock band John Miles, they’re very good people, but apart from that I don’t collaborate. With Trevor Horn on Tubular Bells II we tried working together, but it just didn’t work. He ended up taking bits off to work on them in his studio. When he brought them back, wow! But being in the same room, grrrrr! The lovely thing about doing the remake of Tubular Bells in 2003, first of all I had the first one to copy. And my engineer at the time, Ben Darlow –- I’d go in there and play six takes of acoustic guitar, and he’d compile them while I’d be out the back flying my model helicopters. You did Tubular Bells in Quadraphonic, what do you think of 5.1 and current marketing of surround? Actually, it wasn’t Quadraphonic, it was this strange fake out-of-phase system. We tried to remix it, but the actual mix of Tubular Bells was so difficult without automation, that we gave up after one day. We were doing it at Abbey Road, and I just left it to the engineer to put some bits and pieces of delays at the back. But with the remake of Tubular Bells we mixed it properly in 5.1 at my last studio; that was great. Obviously the mixer was fully automated. At the beginning we made a carousel so that everything was slowly spinning round, a very slow, dreamy merry-go-round. The chap from CD Magazine complained bitterly, ‘Why does he keep moving everything around?!’ I got a telling off for that, but I loved it. I don’t know if 5.1 will ever take off. Most people get these extra speakers with their home TV system and end up slinging them somewhere, or the dog bites the cable, kids knock it over, they end up in the

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craft cupboard anyway. It’s not ever going to be anything apart from the serious audiophile. And even now, although I can do 5.1, it’s a novelty, and it wears off. I’m used to hearing stereo with a virtual centre, I found the real centre speaker off-putting, just because I’m not used to it I suppose.

Did you have formal training? In the very early days I didn’t really learn, but I got inspired to learn to teach myself, I suppose hearing the guitar playing of people like John Renbourn, Bert Jansch, classical players like Julian Bream, flamenco players like Paco Peña inspired me to want to be like that. I worked very hard to try and emulate them, and in so doing developed a technique that allowed me to write my own music. And then in recording studios I was fascinated by the mixer and the electronics, and wanted to know how it worked. It was only when I got to The Manor, I watched the engineers, when they left I could get in there and fiddle. I remember when I was mixing a bit of Tubular Bells Part Two, the ‘Caveman’ section, I just got so into it, I spent all night doing a beautiful mix. That was the moment I felt I had become an engineer. Now I’ve got to rediscover that. With this new project, there’s no automation at all at the moment. It’s confusing to have that running when you’re in the working stage because suddenly a fader shoots up and you can’t remember when you did it. But once I’m in mix mode I’ll go to the Euphonix and try to mix again. Do you think the inaccuracies and mistakes of classic records are what lend them their charm? I’ve been unhappy with Tubular Bells since I made it; it was done in such a rush. There are clicks and pops all over it, there’s a mains hum over the whole thing,

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there’s bits where we chopped out clicks, it’s out of tune, it’s out of time, there’s distortion on the tubular bell, and I remade it — perfectly [laughs]. And in my local record shop they’ve had the original Tubular Bells there since I’d lived there, and they put the new one up, and after a month they took that down and put the old one up! So you’re absolutely right. The

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production, and even the tuning doesn’t matter, it’s the force and energy, idiosyncrasies that give it its character. If you think of the great eccentrics of this world –- Patrick Moore, for example, we love him for his monocle and way he talks, we love him for that, if he didn’t have those he wouldn’t be Patrick Moore. Funny old world… ■

ten

Techniques to kill spill Microphone spill, bleed, leakage — whatever you call it — is like a weed in the garden; it can get everywhere and it’s hard to control. If you can’t learn to embrace spill and value the benefits of musicians playing together, there are many ways to reduce the ‘sonic crossfertilisation’ of microphones pointing at different instruments. With Spring clearly in the air, farmer KEITH SPENCER-ALLEN looks up from his hardy perennials to offer ten techniques to help reduce those unwanted sonic elements found on your multichannel mics.

TOTAL ISOLATION — Ranging from a different room separated by solid walls to a glass fronted iso-booth, this is the simplest way to prevent spill. However, results vary considerably. First thought is to place the loudest item in there but often the iso space is small and loud sources are frequently the ones that sound best in larger spaces. So argue that placing the Tibetan finger cymbals in the iso makes more sense than the brass section. The downside of full isolation is that musical communication is reduced and everyone’s performance can suffer. Lastly so many iso booths don’t sound that good inside (Or smell that good. Ed). A first choice, maybe, but sometimes a last resort. LOOK AT THE ROOM — Spill is less in large rooms and decreases with the ‘deadness’ of the space. While there are tools for reducing spill from direct sound, look at the principal reflective paths by wall and ceilings. Drums direct much of their output in a vertical plane so the ceiling and floor is important because that energy has to go somewhere after reflection. That rug under the kit or an absorbent panel near the ceiling over it can help, but taking it as far as the open drum areas popular in 1970s and 80s studio designs with their low screens and low dead ceilings is too far for modern tastes. Untreated recording spaces can have resonant points where certain frequencies can envelop the room — if low frequency sound appears

to be ‘everywhere’ then you’ve hit one so move the instrument. There’s no simple solution. DISTANT SOLUTIONS — Distance between instruments generally reduces spill. However, the nature of the spill changes with distance as the ratio of direct path spill to reflected falls giving a distant roomy characteristic. This may be more unpleasant than a higher level of spill from a closer position. It also depends on how the two sound sources will be positioned in the mix — greater distance in the left/right pan position emphasises roomy spill. In fact a good test to see how problematic a certain level of spill may be is to pan the two signals close together, set an approximate balance as they will be used, and then judge the effect of the spill. With a close position on the mix (almost mono) the spill will often give a perceived depth, back into the mix rather than an unwanted ambience across it all. SCREENS — There’s an art to positioning screens that comes through experience in a particular room. Unless you are trying to increase the separation between two moderate level sound sources, most screens are of limited use. They’re fine at HF but lower frequencies pass straight through, around or over them. Sometimes this is OK provided there is enough separation at the frequencies that give the definition as

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EQ or low level gating can clean up the rest within a mix. The best use of screens comes with knowing the room. It can enable you to separate instruments over a greater distance and then use the screens to reduce the reflected room spill which is easier. REFLECTIVE SCREENING — The effectiveness of screens can be increased by angling them to act as reflectors of unwanted sound rather than barriers. This can lead to degrees of isolation that wouldn’t seem possible in an open room. With a moderate ceiling height, a pair of full height screens positioned in a Vshape with the point of the V aiming directly at the louder sound source, a far quieter instrument can be positioned inside the V facing the louder source. With a tight cardioid pattern mic, in some circumstances it is possible to virtually eliminate troublesome spill on an acoustic guitar just 10 feet away from a full drum kit. Using the screens as mirrors to reflect the sound works well close as most of the spill is direct (unreflected) and its path can be predicted. CONTAINING THE LOUD — If there are no options other than having a loud instrument in the same room, try to contain the sound at source rather than after it’s in the room and fighting to get into all those delicate low level signals. An isolated riser, sitting on a bed of rubber or mineral

March 2007

ten fibre, can reduce the energy entering the floor in the case of drums or bass amps. Pointing an amp towards an absorbent trap in a room corner with right-angled absorbent screens behind can help. CHOOSE THE MIC — Spill can be reduced by the use and selection of mics. Where possible, placing the mic as close as possible to the instrument increases the ratio of wanted sound to spill. However, this doesn’t always deliver what is wanted sonically. Instinctively we might start off by looking at a cardioid pattern mic response and preferable a tight one where the response falls steeply off-axis. However, you need to look at the polar pattern for the low frequencies as some cardioid mics are really only cardioid at MF and HF with an unpleasant sounding rear response — the most likely direction that spill will originate from. Cardioids that have a pronounced proximity effect are useful because rolling that off reduces low end spill. Although it may sound odd, it is possible to achieve good results with omni patterned mics because they pick-up a more uniform sound all round. If you can lower the level arriving at the rear of the omni mic through distance, screens, etc. the spill that results will not have the unpleasant colouration that can result from a cardioid pattern and is therefore less obtrusive.

experimenting over their positioning to achieve a balanced response from the instrument this can be mixed with a little signal from a mic positioned in the normal way (with the same precautions over level and polarity as with DI and amp signals). The results can be stunning with very little trade-off in sonic quality, the mic adding the air around the signal, although sustain and attack will be enhanced so may need adjustment.

to replace it, you can double or triple track it, thereby reducing the audibility of the spill. • Deal with spill on basic tracks ASAP. The colouration that it adds changes how you look at the next round of tracking, or you become used to its fuller sound. • Spill from click tracks, handclaps, tambourines is pernicious — it gets everywhere, usually through low level leakage from headphones during recording. Listen carefully as it’s a pain to remove. • EQ, gating and compression can be used to remove the effects of spill provided the instruments are not over exposed in the mix. • For LF spill on recorded tracks that cannot be EQed, try splitting the signal into bands and using the original signal causing the spill to key a gate on the LF component of the signal suffering the spill. It will be audible but may be masked within the mix. • Muting spill on a mic when its own source isn’t playing can emphasise the spill effect when it is. Gain riding is less likely to be noticed. • You can experiment with loose hanging drapes and blankets for isolation — there are also proprietary designs of quilts to fit bass drums and pianos. • A little spill is beneficial because it ties the instruments together and gives the sense of performance. • If you want to hear what performance with minimal spill is like, try recording outside. All that sound energy normally reflected and contained in the room to act as potential spill is, one second later, 331 metres away from any mics! • If all else fails, overdub, shamelessly. (i.e. give it a generous top dressing of well-rotted manure. Ed). ■

PLACING THE AMPS — Very load guitar amps work best as overdubs. In small and medium-sized rooms the available air within the room can be driven ‘into compression’ — where the amp output is equally loud everywhere — and once that happens nothing will prevent massive spill. An alternative is to split the guitar signal to a smaller amp in the playing area and a louder amp in an iso space. TEN EXTRAS WITHIN THE TEN — If an instrument track has more spill than you want but you don’t wish

MIXED MIKING — Quiet acoustic instruments, such as guitar, violin and double bass, and acoustic spill-vulnerable instruments, such as the piano, can be problematic. The easy solution is the use of contact mics particularly the tape types. With some

Studer – the evolution goes on Studer Vista 5™ – Ultimate Control Gets More Compact The Vista 5 is a highly flexible compact digital mixer, designed for broadcast production, live broadcasts and performance venues. The 32-fader desk uses the same Vistonics™ screens as its larger brothers, the Vista 6, 7 and 8, allowing the user to work faster and focus on creativity. Facilities include N-1 outputs, off-air conferencing, GPIO, and extensive monitoring with 5.1-to-stereo downmix functions. The console’s internal matrix may be controlled from a variety of third-party controllers and video routers, so can replace a router in many installations.

OnAir 3000Net – from Studer Studer continues the evolution of the OnAir 3000 with the OnAir 3000Net, migrating the desk from a stand-alone unit to an open and networked part of the overall infrastructure of your broadcast center. Interconnecting several OnAir 3000Net results in a transparent and networked broadcast audio infrastructure with complete ergonomic control.

OnAir 500

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OnAir 3000

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March 2007

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business

Beyond iTunes It’s like radio without the talking and randomly dipping into your record collection without the predictability — music recommendation services are buzzing. The latest generation of Internet apps are transforming music discovery, says NIGEL JOPSON.

Pandora.com’s Tim Westergren.

’M SOAKING UP the winter rays in Florida and listening to my music. I didn’t bring a player, so I’m listening to Internet radio on a stereo. But it’s a radio that only ever plays music I like, I didn’t have to choose playlists, I just steered the selection by giving the names of a few artists and tracks. Then I trained my radio. I gave the thumbs up or thumbs down when tunes I did or didn’t like played, and it learnt my taste. It learnt well, now it only plays a blooper occasionally — I just flick the remote from the couch to skip the song with a thumbs down. Perhaps 30% of the tracks are by artists or from albums I’ve never heard before (or have forgotten). The technology behind all this is from Pandora.com, and there’s a host of other innovative start-up companies jostling for position in the race to provide clever music recommendations tailor-made for listeners. Visit any online music store and you’ll see why these Web 2.0 technology mavens have recognised an opportunity. Take iTunes, the industry leader: beyond the front page and 20 genre pages, the browser is left with a pretty uninspiring text-based discovery experience. Suppose I’m looking for music from Oran-born singer Khaled. He’s not on the World genre page and he’s not in World iTunes essentials. In text-based browse, Khaled isn’t under World/Africa or Afro-beat or Afro-pop or Middle East or World Beat. Searching for the misspelling Kaled I find a US rapper, but no further suggestions. Only if I know to search for Rai will I find one of his albums. Khaled is the King of Rai Music — winner of countless awards — from César best movie soundtrack, through BBC World Music Award, to Montreal International Jazz Festival. He’s signed to Universal and given prominence on its website, he’s hugely popular in France, is mentioned in every book on world music and has recorded over 25 albums, many selling triple-platinum numbers. He has 15 albums or partial albums/compilations on the iTunes store. Not exactly unknown, then ... just hard to find, and without any of the signposts and resolution

information the World Music neophyte might have expected for such a prominent artist. The first generation of Internet-based music applications, spawn of the Shawn Fanning-era Napster, aimed mainly at music acquisition: searching for and downloading a track the user already knew they wanted. The new breed of music applications focus on discovery, with actual ownership becoming increasingly less important. To discover music for users you first have to define their taste, and build a framework for analysing music — and there’s the rub — it’s harder than expected to fathom listeners’ preferences, and to define music. Any Amazon.com shopper will know that the ‘People who bought x also bought y’ system doesn’t work: I may have bought Nelson Algren’s cult classic The Man With The Golden Arm, but I also bought Philip Newell’s book Recording Spaces and a Harry Potter ... it’s a wonder the word ‘recommendation’ isn’t already mud. But everyone can name at least one artist or song they really like, and the idea of seamlessly displaying a menu of synergistic music choices to such a large audience of untapped consumers has got a lot of techies and venture capitalists going. It’s rapidly turning into a frenzy of ‘my scientist’s algorithm can beat up your scientist’s algorithm’ as rival services battle for eyeballs. Scratch the surface of music recommendation engines and you’ll find foundations based on a phenomenal amount of work by music-enthusiast techies. Many had no background in large-scale data processing and discovered some clever ideas did not scale well to millions of songs and searches. Just identifying the current song in a user’s player can be a headache, with metadata often partial or inaccurate. Robert Kaye, who slaved for years on MusicBrainz, an online music database, turned the project over to a non-profit group. Moodlogic’s similar project ran out of money in 2003, and was bought by AMG. Relatable’s acoustic recognition failed to scale, and has largely been supplanted by MusicIP’s proprietary acoustic fingerprint engine, which powers offerings from AOL, Disney, Fiql.com, and the new and groovy Soundflavor DJ for iTunes. Some truly unusual and innovative ideas have recently come to fruition: try visiting the liveplasma. com beta site. Type the name of an obscure act and watch as the most amazing graphic map of their influences, progeny and peers unfolds. Wacky ideas are becoming reality, and real money is being invested in them. Ticketmaster recently bought a 25% stake in iLike.com for US$13.3 million. Arena tours for the dinosaurs of rock is a mature market, but maybe there’s an opportunity to promote lesser-known artists playing in smaller venues — iLike can introduce you to your next favourite band and Ticketmaster can sell you tickets to their show. iLike is a community site/iTunes sidebar, grown from early dotcom-survivor Garageband.com. The nascent recommendation services broadly divide into two categories: those that generate recommendations from a database of professionally compiled genres, artists and musical characteristics, and those that monitor registered users’ song plays to compile databases and automatic recommendations. Two market leaders have emerged, Pandora.com and Last.fm, and they’re respective representatives of these disparate approaches. Both have far more builtin intelligence than older recommendation engines (such as Rhapsody), which always had trouble differentiating old-school country and new Nashville stars, or between IDM and downtempo electronica. Pandora is easy to start: just type a favourite artist or track into the Macromedia Flash-based interface March 2007

business and Pandora generates an instant personalised radio station. A small picture of album art displays as songs play, you can click to read more about the artist, or train your station by giving the track a thumbs-up or thumbs-down. Start with a couple of artists you know well and you’ll soon be thinking: how do they do that? Over to founder Tim Westergren, who explains: ‘We have just shy of 40 musicians, they typically have at least a four-year degree in music theory, and they are all practising musicians, they do their music analysis one song at a time. I was composing music for film, in that capacity I had to figure out the taste of film directors ... I got to thinking about music taste and musical attributes, and in 1999 when online music started to really explode, I realised that if we could find some way to automate that it would be really powerful.’ Each song takes 20-30 minutes to analyse, before the musos start working for Pandora they get 40 hour’s training on the so called ‘genome’ — Tim’s work started as The Music Genome Project, which tabulates over 400 musical attributes. The analysts are given benchmarks to judge what might constitute tags like ‘a lot of vocal vibrato’, and Westergren concedes at least 10% of songs in the database have been analysed twice. Two employees have full time roles scouring every source to uncover new music, and songs are added at the rate of 7-8000 per month. ‘We are not only doing new releases, we are also going back, we will have early jazz music as well as the latest hip-hop tracks. We have hundreds of thousands of songs in the genome, we have music from around 15,000 artists.’ I found it quite revealing when Westergren explained: ‘... any given musician may vary, they may sing some song in a lower register and fairly straight ahead, then sing another tune with a lot more vocal acrobatics and maybe in a higher range: when they do that, as far as we are concerned [in categorising the song] they are a different vocalist.’ This explains the success of Pandora in quickly assembling sympathetic playlists. The UK-based Last.fm is an Internet radio and music recommendation system that merged with Audioscrobbler in 2005. Last.fm builds a detailed profile of each user’s musical taste, recommending artists similar to their favourites, showing their top artists and songs on a customisable profile web page. Audioscrobbler began as a computer science project by Richard ‘RJ’ Jones while at Southampton University. Jones developed the first plug-ins, and then opened the API, after which many music players on different platforms were supported. Audioscrobbler monitored music that users played on a registered computer, allowing for charting and collaborative filtering. Last.fm users build a musical profile either by listening to personal music on a software player with the Audioscrobbler plug-in, or by listening to Last.fm Internet radio. Songs played are automatically added to a log from which personal top artist/song bar charts and musical recommendations are calculated. Cofounder Martin Stiksel explains: ‘I think the way music automatically finds you on Last.fm is of massive benefit to music producers, music promotion happens automatically. Instead of something like MySpace, where you need to upload your songs and send your friends to the page, you have to put promotion in yourself. Last.fm takes out the hassle, if your music is good it automatically travels from listener to listener, a lot of people get to hear it — the right people, as well — not just on a broadcast level but actually the people who are interested in it.’ Last.fm now has 12 million users a month, 65 million tracks in the database and 7 million artists. The advantage for community sites like Last.fm is March 2007

that users are doing all the donkeywork tagging and playlisting, so they grow like topsy. The downside is there’s less control over terminology. ‘Some people might consider Napalm Death to be Operatic: because of the tagging system being open to all users, there’s also a bit of leverage in there for slight abuse,’ Martin admits, ‘but we are working on a means to marginalise this. Essentially we really welcome the input of users on this mood level, and it made great radio stations.’ Pandora’s Westergren gives the opposing view: ‘Because it relies on usage patterns it’s a popularity contest, and that means that it’s not going to be very good at surfacing, in our case, music or artists that aren’t already fairly well known. It’s an age-old problem in the music industry that a tiny fraction, less than 1% of music, every year accounts for 80% of what is sold. There’s so much great music around that never gets heard.’ This is the well-known ‘cold start’ problem that afflicts social recommenders: before a new item can become recommendable, it needs time to accumulate enough popularity to rise above the system’s noise. Pandora is comparing songs’ inherent qualities, not their popularity, so it can recommend an artist the first day they are in the system. There’s probably room for both approaches. Social sites are going to work for individuals with the spare time to pose online and groom choices, professionally assembled recommendations will appeal to the cashrich-time-poor who want quick answers and readymade solutions. There’s a huge potential in the latter category for returning lapsed music purchasers. Some individuals I’ve introduced to Pandora would think nothing of spending several hundred dollars to get portable players filled with ‘their’ Pandora stations. These people don’t hang out in music shops or malls. Pandora’s CTO Tom Conrad admits: ‘Practically speaking there’s just not enough margin in CD/ download sales to even begin to cover our licensing and bandwidth costs’. Direct licensing deals like Last. fm cut with Warner Music on February 6, content bundling for mobile players and mobile phones are all possible solutions/untapped revenue streams for recommendation engines. At the end of 2006, Pandora signed a deal with Microsoft to power MSN radio. MSN Radio’s monthly visitors had fallen 9% to 2.5 million from October 2005 to 2006, Pandora attracts more than 1.1 million monthly visitors. Both Last.fm and Pandora are viewed as potential take-over targets for larger players such as Google or Clear Channel. Analyst opinion currently favours Last.fm, probably because community sites like Youtube and MySpace are flavour-of-the-moment. Community sites rely heavily on users clicking around in a computer-based web browser, so it seems to me that professionally constructed databases like Pandora’s will scale better to hardware such as the Slim Devices’ Squeezebox it works with, and to the small screens of mobile phones. Global mobile phone connections topped 2.5bn in 2006, with WiFi phone sales up 327% from 2005 and predicted to increase by 1,300% over the next four years. Mobile WiFi connections currently rely on islands of hotspots, but it’s likely technologies like Wimax will broaden coverage and enable music streaming to cell phones. I’ve discovered more new music since I’ve been a $36 Pandora subscriber than I thought possible. Right now I’m listening to a band I’d never heard of before, the aptly named Audio Learning Centre, who’s Cope Park album I’m surprised to learn (from Pandora’s Backstage pages) was produced by Joe Chiccarelli (Resolution V4.2). Perhaps someone, somewhere, is listening to some small forgotten nugget from my musical career. The potential for music discovery resolution

Last.fm’s Martin Stiksel.

seems almost as great as when the wireless radio was introduced. There’s just one, tiny little nagging doubt in my mind as I groove to my trained digital music player. If someone is grooving in this manner to my music instead of buying a CD, will I get paid? ■

LISTENERS WHO LIKE LAST.FM AND PANDORA.COM ALSO TRIED: soundflavor.com/; ilike.com/; mercora.com/; music.yahoo.com/launchcast/; musicmatch.com/ download/music_discovery_intro.htm; playground. musicip.com/; liveplasma.com/; goombah. com/; musicstrands.com/; upto11.net/; indy.tv/; musicmobs.com/; stage.fm/

broadcast

Pain through gain The subject of inconsistent broadcast television loudness and how it’s become an increasing irritation to viewers is now well established in our collective consciousness; and none more so than in the minds of station duty officers and dubbing mixers. NEIL HILLMAN considers whether the Dolby LM100 broadcast loudness meter can do for television what Leq (M) did for cinema.

LTHOUGH MIXERS ARE obliged to meet exactly the delivery standards set by individual broadcasters — or face a costly Technical Review failure and painful questions from the producer — a broadcaster’s off-air offering often results in carefully crafted and balanced mixes being drowned by highly processed commercials, promos and trails. This leads to frustration for the mixer and disappointment for the client, not to mention the threat of repetitive strain injury for the poor viewer who is left clicking constantly on the volume control of the remote. In fact, a whole new range of medical conditions attributed to domestic television have already been reported iincluding ‘4, five, Finger and Fumb’, ‘ITVitis’- strains 1, 2 and 3 and ‘Digital Digit’. Almost universally based on peak levels, and measured on peak programme meters (PPM), the broadcasters’ long-standing delivery specifications require conformity to an absolute measurement to prevent clipping or distortion, and to ensure signal matching throughout the broadcast chain. These delivery standards were established when analogue ruled the waves; and when the television sound supervisor, or dubbing mixer, would reverently check levels through the desk: speech peaking between PPM 3 and 5½, music at PPM 5, and heavy music and effects (M&E) somewhere between PPM 5 and 6. All fine up to a point: except as every mixing engineer instinctively knows, the human ear doesn’t work on absolute, peak levels; it works on the average level of a signal. This is the reason why viewers adjust the volume of their television sets to hear dialogue comfortably –- as opposed to explosions — in a programme; and why our present delivery specifications may protect the transmitter but fail to take into account the viewer’s listening experience. 48

Add to this the fact that over the last decade or so compression equipment has become widespread in use as dynamic-control devices have become more sophisticated, and the result is that commercials and promotions are louder; their signals continue to meet line-up and peak level criteria, but the average loudness of these 30-second bright bursts of energy has crept up to within 1dB of peak value. Listening to this way of mixing is unpleasant — unbearable in fact — on longer-form, full programmes: compressor artefacts quickly become apparent and a mix requiring artistic interpretation, subtlety, or any light or shade, is almost impossible to achieve. So a dubbing mixer’s only real solution is to drive peak levels close to PPM 6 wherever possible, limiting the headroom to a minimum above the dialogue peaks. It’s making the best of a bad situation, but by reducing dynamics in this way, and aligning peaks, a rough control of a programme’s loudness can be

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achieved; although huge discrepancies can still be heard between programmes and between stations. Digital broadcasting hasn’t eased the problem either — in fact it’s even worse, mainly due to the larger dynamic range available: measured differences of up to 17dB are not uncommon across the major European satellite stations. Cinema has already been here: some time ago it recognised that loudness needed to be controlled. Historically, unduly loud commercials upset audiences and led theatre owners to turn replay levels down, which resulted in louder commercials being mixed, which again were turned down; a spiralling process that eventually began to harm the reproduction of the expensive main feature. This cycle was broken by the introduction and adoption of Dolby’s Cinema equivalent-loudness measurement, Leq (M), based on a long-term measurement that is close to subjective levels, taking into account frequencies that are a source of irritation to theatre audiences. As the Leq (M) standard is specific for theatrical playback, Dolby turned its experience towards television through the LM100 broadcast loudness meter, which claims to offer sound supervisors, postproduction dubbing mixers and station quality control operators a ready way of measuring loudness in soundtracks for television. The Dolby LM100 uses the Leq (A) scale, where a measurement is defined as the level of a constant sound in a given time period that has the same energy as a timevarying sound. Relating directly back to Fletcher and Munson’s equivalent-loudness contours of the 1930s, it measures on a scale weighted closely to the frequency response of the human ear at normal conversation levels: which as we know, is exactly what the home viewer is trying to achieve when he adjusts his television’s volume between ad breaks and channel hops. So can Leq (A) in the form of the Dolby LM100 do for television what the implementation of Leq (M) did for cinema; and if it can, isn’t it about time a new QC measurement standard was introduced? Should there be a standard that can address the irritating issue of varying loudness and adapts its yardstick to recognise that the needs of mixers supplying terrestrial stations broadcasting PCM stereo and Dolby Surround LtRt Pro-Logic mixes, are significantly different to those

March 2007

broadcast operating in the digital domain with AC3 Dolby Digital? Because for some time to come, the majority of UK dubbing mixers will continue to supply their output in an analogue-esque fashion — albeit via a digital tape or as a broadcast .WAV — until the uptake of 5.1 gains greater momentum in broadcasting, and plentiful commissions start originating from the digital stations, high definition (HDTV) broadcasters and video-on-demand (VoD) services. Dolby claims several advantages of using the LM100; it can operate with an analogue or digital input; with PCM stereo, Dolby Surround or Dolby Digital signals; it can provide a single loudness figure that is repeatable, easily understood by operators, without interpretation and in real time; it also measures true peak and it’s relatively inexpensive. So is the Dolby LM100 the solution we’re looking for? During the mixing stage of the latest series of The Gadget Show, consisting of 13 45-minute episodes for UK terrestrial broadcaster five, and the International reversioning of ten 30-minute episodes of the motoring series 5th Gear, I added a Dolby LM100 to The Audio Suite’s standard metering array of a twin-digital PPM and a DK Audio MSD600M jellyfish display with PPM-IIA (BBC) ballistics. This gave me a good opportunity to see if achieving Dolby’s range of suggested loudness figures for the programmes impacted on the way I had to mix, while still mindful that the broadcaster’s standard delivery requirements still needed to be met. Both series were supplied for their QC Technical Review, as usual, in Dolby Surround and mastered onto DigiBeta tapes, and by the way, they all passed. I mixed the first couple of programmes without paying too much attention to the LM100, other than to note what arbitrary figures I was seeing for loudness, and to compare peak levels to my ‘standard’ meters. Not surprisingly, both PPM meters showed under on peaks by about 2dB, compared to the true peak meter of the LM100. I’d configured the Dolby unit to look at ‘all’ of the PCM stereo input, and selected ‘dialogue intelligence’ from the easilynavigated set-up menu. Selecting this parameter allows the LM100 to automatically base its loudness measurements on the portions of the input signal that it considers contain the characteristics of dialogue: that which the viewer prizes most highly. The portions that don’t contain these dialogue elements are not included in the loudness value and therefore, while it’s a really useful feature, it isn’t foolproof. This might yet be an interesting Technical Review/QC point: if dialogue is present, but coincident with other types of signals, such as heavy music or effects — pretty much the definition of a modern, fast-paced motoring programme — the algorithm can get caught out and fail to account for the dialogue that is present. The way in which the loudness value is calculated by the LM100 is chosen from two menu choices: short-term or infinite mode. Infinite is the best option for QC and ingest stages, where a programme can be measured over its full duration, and an accurate average loudness figure arrived at. Up to 480 deviations can be logged as events against timecode, and recorded via a remote interface cable to any PC able to accept ASCII text strings. Alarms can also be user-set for input-clip detection, modulation overload, loudness above threshold, silence below threshold, dialnorm threshold and AES input loss. For postproduction mixing, I favoured the shortterm mode where the Leq (A) measurement is shown for the previous 10-second period, updated each second, as the programme progresses. This felt a much more natural and dynamic way to operate, as it allows the mixer to see short-term variations in March 2007

the loudness. But how did the mixing process itself compare? Unspectacularly, as it happens. Like most dubbing mixers, I’ve got used to mixing pacey, midevening, magazine programmes reasonably hard; and so the suggested loudness figure from Dolby of 20 for this type of programme pretty well matched what I was already achieving, so no real change at all was required for mixing in Dolby Surround, for a terrestrial broadcaster, with an LM100 in tow. The suggested levels for a drama are in the range of 24 to 20, but the most telling figure is this: presently, we can expect to see commercials registering at the appreciably louder

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figure of 17. This is where we came in. So what are we waiting for? Logically, we’re going to need LM100’s for future digital delivery: Leq (A) is the ATSC (advanced television systems committee) A54 standard in the US, it’s been adopted by broadcasters using Dolby Digital, Discovery US has made it a stipulation that their supplied programmes must now meet stated LM100 values, and we have an existing loudness problem here in the UK that we might solve quite easily. I’ve got the telephone numbers for Channel 4 and five; any volunteer’s to call ITV? ■

sweet spot

Designing the AE22 nearfield The NS10 has left a towering legacy and its suitability and success as a studio monitor is now understood. Loudspeaker engineer PHIL WARD says that his work with Acoustic Energy has resulted in a natural successor.

Figure 1. Acoustic Energy AE22 nearfield monitor.

OU’RE PROBABLY asking, just as I might on seeing this article, ‘Does the world really need another nearfield monitor?’ And just like me, your first instinct might be a decidedly negative one. But let me run a question past you. How is it, in a world that we’ve already established has too many nearfield monitor choices, that the monitor that we come across most often, and the one that the majority of recording engineers would run to save if the studio caught fire, was designed in the 1970s and discontinued in 2001? I’m talking about the Yamaha NS10 (in its various revisions) of course. Even stranger, despite the fact that the recording world has an incurable case of Gear Acquisition Syndrome, I’d wager that the majority of Resolution readers, despite endless opportunities to do so, would be reluctant Figure 2. AE22 bass/mid driver. to give-up their NS10s Note the unusually for any of today’s all thick magnet singing, all dancing top plate. ‘equivalents’ — never mind how many adaptive learning network bells and whistles they boast. Few it seems have a problem spending, say, £500 on a couple (more) compressor plug-ins but spending £500 on a new pair of monitors to replace the 50

NS10s? Now you’re asking! There has to be something going on here. I suspect what’s going on is simple — the current crop of nearfield monitors, despite offering some impressive technology, just don’t cut the mustard. They simply don’t do the basic job as effectively as the old NS10s. If you’re still with me you probably have an inkling where this is heading. Yes, I’m here to write about a nearfield monitor I’ve designed in cooperation with Acoustic Energy. But first, before I get on to the AE22, a little NS10 history and a little discourse on loudspeaker design. On the surface, there was nothing special about the NS10 other than its unusual commercial life cycle. It was introduced in the late 70s as a domestic hifi speaker and was unexceptional in all respects: a 10.5-litre black-ash box carrying a 180mm paper cone bass/mid driver and a 35mm soft dome tweeter. It was relatively well engineered but introduced no new technology and was met by indifferent reviews in the hifi press and achieved unexceptional sales in the domestic sector. The NS10, it is safe to say, has an uneven frequency response with a mid-prominent balance and even now you’ll find hifi enthusiasts and audiophiles that, in moments of ignorant criticism, blame its alleged shortcomings for the ‘sad state of recorded music’. But, while on the surface the NS10 was unexceptional, it perhaps by chance combined technical features and performance characteristics that uniquely suited it to the nearfield monitor application and some unknown recording engineer first discovered it worked well in the role (Who was the first? Does anybody know?). The rest, as they say, is cliché. The happy accident of the NS10’s characteristics wasn’t widely appreciated at all at the time, and perhaps still isn’t now, but was demonstrated very clearly in a paper presented to the Institute of Acoustics by Philip Newell, Keith Holland and Julius Newell in 2001 and published in the last (and hen’s teeth rare) issue of Studio Sound magazine [Actually rarer than hen’s teeth. Ed]. How ironic that the first serious attempt at working out why the NS10 worked so well as a monitor happened in the same year that it was discontinued. The Newell/Holland paper — The Yamaha NS10 — Twenty Years A Reference Monitor. Why? — sets the NS10’s in the context of 35 other nearfield monitor speakers by measuring each one’s amplitude response, harmonic distortion, step response, and amplitude response waterfall (a measure of how well a speaker performs in the time domain) [Much of this was drawn from the monitor tests we ran in Studio Sound. Ed]. Newell(s) and Holland argued that a good result in the last three of these characteristics in particular is vital if a speaker is to perform well resolution

as a monitor because they describe how accurately it is able to communicate what’s actually inherent in the signal (as opposed to speaker artefact) in what’s finally perceived by the listener. Of the 36 speakers tested, the NS10 effectively came top. Its distortion levels were among the lowest, its step response was the quickest and its waterfall was the cleanest. The validity of Newell and Holland’s opinion that step response, distortion and waterfall are vital undoubtedly has currency. There’s much anecdotal evidence from professional and respectable hifi opinion that these parameters do reveal a great deal about a speaker’s musical accuracy — by which I mean dynamic, harmonic and temporal accuracy. And musical accuracy, I would suggest, is more important than tonal accuracy (by which I mean a subjectively neutral and wide tonal balance) in nearfield monitoring applications. Often, however, musical accuracy comes second to the market-led (or should that be ‘misled’) demands of a subjectively neutral response and a wide bandwidth. It gets worse too because an emphasis on such criteria can conspire against musical accuracy, encouraging as it does, high-order reflex bass loading to extend the LF bandwidth, lossy thermo-plastic cones to ‘control’ the upper mid-range response, and high crossover frequencies to allow the use of small tweeters and extend the HF bandwidth. The results tend to be a poorly controlled time domain response and high levels of distortion and compression. Speakers designed thus can work at first appearances because it’s actually not rocket science to ‘voice’ the tonal-balance of a speaker to sound neutral — even if its time domain response is poor, it generates high-levels of harmonic distortion and it suffers from both thermal and electro-acoustic compression. Such ‘engineering’ happens all the time. As expected, the NS10 was by no means the ‘flattest’ speaker measured by Newell(s) and Holland in their 2001 paper. Its amplitude response is characterised by a mid-range emphasis with a slight down-tilt above 10kHz, and while it might be assumed that a flat anechoic amplitude response is desirable for studio monitoring, that is not necessarily the case. First, if a lumpy amplitude response is not the result of serious underlying driver resonance problems and is still within reasonable limits, it is a relatively benign fault that listeners quickly learn to adapt to. There’s no better example of this ‘tonal adaption’ phenomenon than the NS10. You know its tonal balance isn’t neutral, but you can still intuitively balance a neutral mix with it. Second, because nearfield monitors are generally heard close-up and from a variety of laterally displaced positions (as the listener works along the desk), the most important aspect of the amplitude response is not its absolute flatness on-axis but its horizontal off-axis consistency — especially in the mid-range. In choosing a low crossover frequency (2kHz, made possible then by the NS10’s larger than usual tweeter dome), Yamaha helped make that NS10 more consistent off-axis. The low crossover frequency also helped the waterfall and distortion performance of the speaker by filtering the bass/mid driver at a relatively low frequency. March 2007

sweet spot

Figure 3. An under-hung voice coil improves linearity and reduces thermal compression but is expensive to implement.

Figure 4. Two sets of curves illustrating the low frequency amplitude response (black curves) and delay (blue curves) characteristics of reflex and closed box systems based on a similar driver in the same cabinet volume. The reflex system has an extended bandwidth, but far greater delay. The closed box system represents the AE22.

Third, the manner in which nearfield monitors are often used (standing on the meterbridge of a big desk or up against the wall in smaller rooms) and the demands of long listening sessions mean that a response specifically voiced to take these factors into account (mid-prominent with a gently falling top-end. Ring any bells?) is more appropriate than one that is ruler flat on one measurement axis in an anechoic chamber The NS10 had one other feature that I believe is vital for a musically accurate monitor — it was a closed-box design (or rather, it wasn’t a reflex design). Recent work, again by Newell and Holland and published in this very magazine (Measuring low frequency response accuracy, V4.3) has shown how reflex loaded speakers, while undoubtedly offering extended low frequency bandwidth, are inherently flawed when assessed in terms of signal intelligibility criteria. The flaw comes not only from the inherently poor time-domain response of reflex loading (a typical reflex loaded speaker will introduce a delay to low frequencies of around 15ms or more — see Figure 4), but from the reflex loading mechanism itself introducing distortion, compression and noise. An example of this occurred on a session I played bass on in 2005. A pair of nearfield monitors was loaned by its UK distributor and installed on the meterbridge just as a session percussionist was setting up his numerous instruments. The percussionist started by playing a small(ish) hand-drum, and even at low levels the monitors utterly failed to reproduce the signal. The reflex port noise — best described by the term ‘farting’ — rendered the speaker unusable. Switching back to the NS10s solved the problem. Now, maybe the speaker got unlucky and the peculiarities of that hand-drum excited a reflex port turbulence that would never otherwise be heard, but the fact that it was heard, revealed the fundamentally primitive, approximate and suboptimal nature of reflex loading. Granted, not all reflex loaded speakers are as bad — the one in question critically incorporates no ‘aerodynamic’ port flaring for a start — but its example demonstrates that the extension of LF bandwidth through reflex loading can extract a heavy price. The price is sometimes obvious, as in this case, and sometimes subtle, but it’s always there and always colours what you think you hear. Ever wondered why, down the chain, the mastering engineer struggles to get the bass/mid balance right?

It’s quite possibly because your reflex loaded nearfield monitors were indulging in a little low frequency improvisation all along. What you heard, wasn’t what you recorded. Having read my potted analysis of the NS10 and why it worked, you can probably guess what we’ve tried to do with the Acoustic Energy AE22 — everything that the NS10 was good at, and more. The AE22 is a low distortion, closed box speaker with an exceptional time domain performance (better than 40dB down within 5ms — see Figure 5) and good horizontal off-axis consistency. It doesn’t have the flattest anechoic amplitude response you’ll ever see, nor the widest bandwidth, but from 60Hz to 25kHz it is about as musically accurate as moving-coil driver technology can achieve at the price of a couple of compressor plug-ins. More specifically, the AE22 (available in passive and active versions), features a 200mm anodised aluminium cone bass/mid unit and a 25mm fabric ring-radiator tweeter. The low frequency alignment is a critically damped closed box and the crossover is a symmetrical 3rd order Bessel function at 2kHz. The low crossover frequency with relatively steep filter curves ensures good off-axis consistency in the near field. The AE22 bass/mid driver is unusual in a number of respects. Its use of a metal cone ensures a linear amplitude response and minimal time-domain errors up to frequencies well above the crossover point. The metal cone also acts as a heat-sink that increases thermal power handling and reduces thermal compression. The driver also has an ‘underhung’ magnet/voice-coil system. An under-hung system — a short voice-coil in a long gap — is inherently more stable and linear than the more usual, and far cheaper, over-hung arrangement (Figure 3). The voice-coil is 50mm in diameter (25-38mm is typical at this price point) and has a black anodised aluminium former — again to increase thermal power handling and reduce thermal compression. The AE22 tweeter is a proprietary unit chosen for its good linearity, low compression, high power handling and ability to operate with a relatively low crossover frequency. It is often seen in high-end hifi speakers and is characterised by a clean and detailed sound with better than average high frequency dispersion. Finally, there’s a further element of performance that I’ve touched on and that we’ve worked hard

March 2007

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Figure 5. AE22 Anechoic Amplitude Waterfall curve 200Hz to 20kHz. The output is better than 40dB down within 5ms.

on with the AE22 -– one at which the NS10 didn’t particularly excel; thermal compression. A typical moving-coil loudspeaker dissipates around 95% of its input power as heat, and the AE22 is no different in this respect. Unless a speaker is engineered effectively to dissipate heat, as it is driven harder, the drive-unit voice-coils will increase in temperature and their electrical resistance will rise (a doubling of voice-coil resistance is not unheard of). A significant increase in resistance will have three consequences. First, the broadband sensitivity of the driver will fall. Second, the low frequency alignment will change (it will become less well damped). Third, on passive speakers, the amplitude response around the crossover frequency will vary as the increase in voice-coil resistance causes the characteristics of the crossover filters to change. These three phenomena, all aspects of thermal compression, will result in a speaker that slowly varies its audio ‘footprint’ in response to changes in level. While the effect is subtle, as by definition there are no sudden changes, it can be a significant factor in the subjective response to monitors that should above all else be consistent over time and level. The AE22 is probably the only cost-effective nearfield monitor designed from the outset to have minimal thermal compression. The benefits are obvious as soon as you begin to apply EQ — you get consistent results over time and at different volume levels. ■

katzâ&#x20AC;&#x2122;s column

Routing it all Having explained his mastering choice options for that Latin-Jazz three-CD compilation in the last issue, BOB KATZ now talks us through the equipment and connections that allow seamless mastering of these different-sounding sources.

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Figure 1. Routezilla matrix. resolution

SET UP MY MAIN digital processing route using a Z-Systems 32x32 AES-EBU router and control software that Iâ&#x20AC;&#x2122;ve developed, called Routezilla. Figure 1 shows Routezillaâ&#x20AC;&#x2122;s control matrix, and Figure 2 the series chain of processors: the knee bone is connected to theâ&#x20AC;Ś oops, I mean that signal starts with SADiE outputs 3/4, which is connected to the TC System 6000 inputs 1/2. Inside Engine 1 of the TC, I may apply some parallel compression, sometimes for tonality, and sometimes to â&#x20AC;&#x2DC;fill in the holesâ&#x20AC;&#x2122; (if any need to be filled in). TC outputs 1/2 go into Weiss Compressor A, which is standing by to possibly add some high or upper mid frequency compression or to control some unruly peaks. If not needed, Iâ&#x20AC;&#x2122;ll set the threshold to 0dBFS, which in the purist Weiss is bittransparent. Weiss Comp A feeds the K-stereo processor, which I may use to enhance the depth or spatiality of sources that might be missing some, especially since the stereo separation in the 16-bit transfers was not always that great. Next is the Weiss EQ, and Weiss Comp B, which may be called upon to adjust level or compress or expand a bit, depending on the need. Comp B goes into the second engine of the TC, which I have assigned to its superb Brickwall Peak limiter (comparable to the well-known Waves L2 but with more flexible auto release time). TC output 3/4 then feeds into SADiE, where I can capture a safety 96kHz/24-bit master. SADiEâ&#x20AC;&#x2122;s output feeds Weiss SRC/B, which takes the signal down to 44.1kHz â&#x20AC;&#x2DC;liveâ&#x20AC;&#x2122;, in real time. Since the Z-Sys is an asynchronous router it can handle multiple sample rates at one time, and so it can patch the 44.1kHz signal into the inputs of DAW #2, which is running Steinberg Wavelabâ&#x20AC;&#x2122;s excellent software, though I am simply using Wavelab like a tape recorder, to capture a 44/24 file which will be later dithered to 16 bits. And Wavelabâ&#x20AC;&#x2122;s output feeds the Cranesong Avocet monitor controller, where I can listen to the 44.1kHz result through the entire chain. Letâ&#x20AC;&#x2122;s look a little deeper inside the TC System 6000. It contains four independent DSP engines, each of which is a powerful 48-bit processor with up to 8 inputs and outputs. In Figure 3, we see the normal route inside the TC. Signal from SADiE comes into inputs 1/2 and feeds engine 1, which contains the first processor in March 2007

katz’s column

the signal chain, the parallel compressor, and then seamlessly meet up with the big effects scene. Film goes out of outputs 1/2 to the four Weiss Processors mix engineers might commit to the dialogue scene, you saw in Figure 2. At the output of Weiss Comp B but roll back and punch into the effects scene to make signal is brought to inputs 3/4 of the TC, and in Figure sure the transition is good. With full automation there 3 you can see that feeds Engine #2, which is doing is no need to punch into the master, you automate, the brickwall limiting. preview, then go back to the beginning and roll it. A This long chain might do well for mastering combination of automation and punching is probably track 2 of our EDL, but what about the first track, the most efficient way to get it done. And the end which is already mastered? Even though the Weiss listener will have a better listening experience. Unless and TC processors are extremely transparent, if the he’s using a pair of cheap, noisy computer speakers… level and sound of the already mastered track is but don’t get me started! ■ just right, I would want to bypass the extraneous processors. And for this I use the versatility of the TC System 6000, which has an automated synchronous router. Figure 4 shows an alternate route. Notice how inputs 1/2 of the TC are now patched to Engine #2 thus bypassing the entire processing chain except for the peak limiter. And I’ve set the limiter to Off; like the Weiss, the TC Brickwall is designed to be bit-transparent when it is off. So I have routed a clone of the original 96kHz mastered material to the sample rate convertor. A synchronous router can change its routing without glitching or losing clock, so I can use a sequencer (like Digital Performer) to send MIDI commands that change routes in the TC and send program changes to all the processors, while I’m running the CD Figure 3. TC normal route. master from beginning to end. I know this sounds like a complex crossword puzzle (and it is), but the advantage of doing this automated is that you can have your cake and eat it too! You can listen to the entire CD master in the making, not overprocess, and every piece of processing gear automatically changes as we move from track to track. The CD master is fluid until I commit to it. I’ve even found ways to (semi) automate some of the analogue gear. In the old days of mastering, we had to capture each track on its own, hopefully get it right, and then work on the next track and sometimes redo the first if it didn’t fit into the whole. It wasn’t easy to build this automated system, but it makes the process of creating musical, holistic works easy and enjoyable. This approach could Figure 4. TC Special route. be accomplished entirely ‘in the box’ Information using automated VST plug-ins, but the ergonomic weakness of using a mouse to control everything, Resolution recommends Bob Katz’s book Mastering Audio — The Art and and the lower sound quality of some plug-ins are the Science as an essential source of obstacles waiting to be overcome. information for every pro audio enthusiast Motion picture mix engineers would not be strangers who cares about sound. You can buy it on to this way of working. Action films are mixed in line at www.digido.com context, because the quiet dialogue scene has to March 2007

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Boom for strings Jon Lord’s new orchestral scores offered an opportunity for a recording engineer to reappraise his choices. GEOFF MILES and his magic Pelusos capture a fairytale session at Odense’s Hans Christian Andersen Hall.

Photo: EMI Classics/Paul Mitchell (paul@mitchellstudio.co.uk)

Conductor Paul Mann below centre Peluso 22 251.

AVING RECENTLY RELOCATED to Oslo, and with a two month old son occupying most of the brain space I’d otherwise have been devoting to work, I was very happy to receive an enquiry from EMI Classics about a potential orchestral recording project in Odense, Denmark, to take place in December 2006. They planned to record some new orchestral music written by Jon Lord, and since one of his closest collaborators in the classical arena, Paul Mann, is chief conductor of the Odense Symphony Orchestra, this had become the obvious orchestra and venue for the project. Two pieces were to be recorded: Disguises, a suite for strings, and Boom of the Tingling Strings a piano concerto scored for large orchestra. Neither producer Jørn Pedersen nor I had worked with the Odense Symphony before, so we visited the orchestra while they were rehearsing for a concert in 54

September. The concert hall was an immediate and very pleasant surprise. I’d heard some good reports about the acoustics from orchestral musicians, but the fact that the hall dated from the 1980s, and was part of a hotel and conference complex, hadn’t given me great expectations. In fact the Hans Christian Andersen hall in Odense has to rate as one of the best orchestral halls in Scandinavia (and I’ve worked in a fair number). It’s a classic shoebox shaped concert hall, with a stage area placed forward of the wall to facilitate audience seating on an upper level behind the orchestra. Although the basic building material is a utilitarian concrete, much of the wall is covered with wood panelling, and attention has been paid to getting the right amount of absorption from the seating area. The result is a clear and very even acoustic, with a resolution

reverb time of around 3 seconds. Early reflections from the walls were distinct enough to give a feeling of space, but without confusing the direct sound from the stage. Even at the very back of the hall the orchestra sounded well-balanced and clear. We were much encouraged by this and by the helpful and friendly attitude of the orchestral stage management, who were happy to provide us with a decent control room area (complete with a 40m built-in cable run to the hall) and to build a stage extension at the front of the orchestra for the piano, so we would not have to recess the orchestra on the stage while recording the concerto. Having returned from Denmark happy, we needed to finalise some details regarding the equipment that we would require. I always use my own microphones and preamps for the main pickup since these determine the basic sonic identity of the recording. We would need the option of using quite a large number of additional spot mics for the piano concerto, since there was a great deal of detail in the score that might not necessarily come through on a minimal setup. To this end we hired a recording rig from Oslo-based Pro Musica, providing us with 30 extra Neumann, AKG and Schoeps microphones plus stands, cabling and control room equipment. B&W Scandinavia supplied us with a pair of Nautilus 800 monitors and Classé monoblock amps. We were ready for action. We allowed an extra day before the recording to set up the control room and get used to the monitoring environment. Multitrack recording on location without a proper inline console always produces a rather convoluted control room setup, and this would be no exception. The danger is that you end up with a nice array of different mic pres and input consoles, but with inadequate fader control of your monitor mix, and a less than intuitive gain structure. I planned therefore to use a Yamaha 02R coupled to the outputs of a 24-track Pyramix system purely as a monitor mixer, and to set the control room up so that I had a very clear view of the input level meters. Tape backup would be to three Tascam DA-98HRs. The main microphones were fed directly to a Studer D19 A-D convertor from Tube-Tech MPA1 mic pres, and the spots were fed from Amek and DDA mic pres, with some submixing of brass and percussion mics. The Nautilus 800s initially sounded rather disappointing — dull and congested, and there was some noise in the right hand channel, which pointed to a problem with the accompanying Classé amplifiers. We switched to Pro Musica’s Norwegian built Dynamic Precision amplifier and the speakers came alive. For these sessions I would use my standard main microphone setup of a tree of very widely spaced omnis, modelled on the classic RCA setup. I have always used omnis for orchestral recordings, but have experimented with spacing and configuration. I’ve moved from the German four-microphone (gross AB, klein AB) setup, to the Decca five-microphone (tree LCR plus outrigger LR) technique, and ended up with a three-microphone LCR configuration. Why? Well, it’s partly nostalgic — I regard RCA’s early Living Stereo recordings as some of the greatest sounding orchestral recordings ever made. I have always wanted to emulate that mixture of contact and space. I found that there were more inherent phase cancellation problems with the four- or five-microphone techniques and that even in a mix that uses a lot of spot mics for a full orchestra the RCA technique gives a cleaner more spacious sound, although the width of the image might disturb engineers who are used to a narrower soundstage. The ideal playback setup for such a technique would use a centre speaker, so for two-speaker stereo March 2007

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flexible_res_amend.qxd

Photo: EMI Classics/Paul Mitchell (paul@mitchellstudio.co.uk)

March 2007

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The Flexible Recording Route

MMA-4XR 4 Channel Analogue Mic Preamp

Control room playback during the session. Seated (front) l-r: producer Jørn Pedersen, pianist Nelson Goerner and conductor Paul Mann. Composer Jon Lord is seated at the back of the room.

I find I need to drop the level of the centre microphone by 4 or 5dB. I find also that the technique really does work best with vintage style mics –- it’s difficult to get close enough in over the strings with most modern solid state condensers without the result sounding hard. I’ve tried Schoeps CMC5s (mk2, 2S, 2H and 5 omni capsules), Sennheiser MKH20s, Neumann KM130s and TLM-50s, and some of the modern valve designs such as the Neumann M150s and the Brauner VM1 without being completely satisfied with the results. Then, some years ago, rummaging through the microphone cupboard at Funky Junk in London, I discovered Neumann KM253s. They have much of the character of the more famous M50 and are difficult to beat on a string section, although arguably they do lack something in their bass response, and of course they have a vintage colouration that is not to everyone’s taste. For this recording, however, I had another option. I’d had a very interesting telephone conversation with John Peluso, of Peluso Microphone Labs, Virginia, who agreed to ship his Danish distributor, Morten Jacobsen, three Peluso 22 251 microphones for demo. These are modelled on the classic ELAM Telefunken 251s, and come with nine-pattern variable directivity, controllable from the power supply units. I had been looking for a set of mics that had the same warmth and vintage quality as the KM253s, but could give a better bass response and more impression of acoustic space. The Peluso mics seemed to promise this on paper, and at an extremely low price, but were they too good to be true? There’s always a buzz about orchestral sessions. They represent a challenge to an engineer, not so much because it is more difficult to record an orchestra than any other group of musicians, but more because you are aware of time pressure, and the lack of room for mistakes. In this instance there was excitement within the orchestra also — this was the first recording they have made for EMI, and Jon Lord’s presence helped make this feel like an important occasion. Boom of the Tingling Strings had been performed to great acclaim the preceding week, and so was fully prepared and ready to record. I was aware, therefore, of having limited time for balancing and experimentation. I had rigged both Neumanns and Pelusos, and as the orchestra started to play, we had to make a quick decision about which way to go. My first impression was positive — the orchestra and the hall sounded great (on both sets of microphones) and with the help of a pair of Royer R121s the piano integrated well into the picture without sounding muddy or congested. Actually both Neumanns and Pelusos had their strong and weak points. The Neumanns sounded warm and detailed on the strings, but lacked space and that feeling of bigness. The Pelusos had a better bass response, and made the orchestra sound huge and three dimensional, but gave a slight thinness to the upper strings and lacked some of the detail of the Neumanns. Taking the Pelusos one click off omni to a wide cardioid setting improved their reach and seemed to provide the best balance between direct and reverberant sound in this acoustic. We decided to go with the Pelusos, and everyone, including composer and musicians, seemed happy with that decision. The spot microphones added a little extra sparkle and precision, but the large orchestra and piano balanced extremely well due largely to Jon Lord’s orchestration and the hard work of the conductor, orchestra and soloist Nelson Goerner. The sessions ran very smoothly, and we completed Boom on schedule. I changed the main mic setup for Disguises, using a combination of Pelusos and KM253s to get a little more warmth and detail without losing space. This balanced well enough in a good acoustic to allow us to dispense completely with the spot mics — always a good sign. I left Denmark happy that our initial feeling about the hall had been right, and pleased to have discovered in the Pelusos a powerful new addition to my microphone cupboard. ■

16/2/07

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The MMA-4XR and ADA-8XR provide the perfect hardware compliment to your choice of computer software, no matter what platform you choose. That’s why Sting, Mark Knopfler (Dire Straits), Casino Royale, George Massenburg, Bob Rock and The Flaming Lips are some of the customers that have chosen Prism Sound.

www.prismsound.com Email: sales@prismsound.com +44 (0)1223 424988 +1-973-983-9577 NOTE: Digidesign, Steinberg Media Technologies GmbH, MOTU Inc and Apple Computer Inc do not endorse or support the Prism Sound ADA-8XR unit. Not all ProTools, Nuendo, Cubase, Logic or Digital Performer controls or configurations are supported by the Prism Sound ADA-8XR. Prism Sound reserve the right to amend their own product specifications without notice. Digidesign, Pro Tools I HD are trademarks of Digidesign a division of Avid Technology Inc. Nuendo and Cubase are trademarks of Steinberg Media technologies GmbH a division of Yamaha Corporation, Logic is a trademark of Apple Computer Inc, Digital Performer is a trademark of MOTU Inc

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meet your maker

John La Grou The man behind the Millennia Media brand talks anti-corporate philosophy, the importance of the human ear, how the most important aspects of recording still occur in analogue, and why, after 15 years and nine products, cost remains irrelevant. POW-R Software is licensed by a broad range of technology companies including Apple Computer, Avid/ Digidesign, Merging Technologies, SADiE, and many others. An avid acoustic music recording engineer, La Grou has engineered and/or produced hundreds of orchestral, chamber, and world music recordings. John is a member of the AES, IEEE and NARAS, and has written numerous technical articles and papers for magazines and journals. His hobbies and avocations include tending the family vineyard, winemaking, energy and sustainability research, post-foundational religious studies, composing music, HD filmmaking, and Sierra trekking. La Grou co-holds US Design Patent No. 6049143, which is the heart of a new generation of home and commercial electrical safety standards called SafePlug. In 2001, John and Cynthia established the Millennia Foundation, which supports non-profit activities in areas of social justice, humanitarian assistance, resource sustainability, and post-foundational media/ education. The annual Millennia Scholarship was established in 2006. The La Grou’s reside in the Sierra mountains of California, near Lake Tahoe.

OHN AND CYNTHIA LA GROU formed Millennia Media in 1989 to pursue their passion in fine and applied arts. John lectures in audio engineering at colleges and industry conferences and also chairs the POW-R Consortium, a software development corporation focused on audio applications.

What’s special about Millennia Media products? A product is only as good as the people behind it. Millennia is fortunate to have attracted exceedingly bright, dedicated individuals who share a passion for great sound. Moreover, every person at Millennia has been chartered as a quality supervisor. In every area of business, from product development to procurement to assembly to final inspection, each person has authority to stop production if a quality issue, no

ZENON SCHOEPE

matter how small, is found. Most gear makers are market-driven and profitfocused. That is, they identify a market niche and define a product to meet that niche, making necessary compromises in sound quality, component quality, and build quality to meet their cost targets. From day-one, Millennia has done things quite differently. Instead of targeting a price point or market niche, we strive instead to maximise audio transparency and invisibility. Cost is irrelevant. Marketing is not considered. A product can undergo years of development. Nothing is shipped until it is sonically correct and ready to offer publicly. When we’re satisfied that a design is as good as we can make it, we calculate the cost of components and multiply that by a fair mark-up. That’s our business plan. I’m an active recording engineer in the classical music world. I’m also a gear-a-holic. Millennia was born out of my need for more and better recording equipment. This has been our business model for 15 years. It’s never changed. Nearly all Millennia products are conceived to fulfil a requirement for our real-world production work. The HV-3 mic amp was born of my need for a pristine acoustic music preamplifier. The TD-1 was born of our need for a compact recording channel. The new HV-3R remote mic amp allows orchestral gain adjustments from the recording booth. In the 1980s I was one of the first US managers in what today is the world’s #2 notebook computer maker [Acer]. My division was very successful. We built notebook PCs for IBM and many others. This was a pure business model where we targeted markets and strove to maximise revenue and profits. We were wildly successful. But big industry has no heart, no soul. Millennia reflects this desire to do business with heart and soul — with a passion for things that really matter and have lasting value. We have proven that a business can be sustained on a higher level of premises and expectations.

Why don’t more manufacturers build gear like you do? It gets back to corporate mission. Most companies are focused on maximising revenue and profits. Larger public companies are, by charter, focused on maximising shareholder wealth. Millennia’s mission is to design the finest audio paths it can envision, regardless of cost or time-to-market. We’ve spent over four years developing A-D conversion circuits. They’re still not perfect. We’ll continue to experiment with A-DC designs until we’ve made it the best we can possibly achieve. Until we’ve achieved our optimal design, the product will not go to market. Has this anti-corporate philosophy restricted our growth and profitability? Sure. Could we be making more money by focusing on optimal market strategies and Chinese manufacturing? Of course. What we’ve achieved instead is something far more personally valuable and satisfying to us — a well-deserved reputation for pristine audio performance, products that last forever and maintain exceptional resale value, and relationships with our customers that are second to none. 56

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March 2007

meet your maker

What have people forgotten about analogue? I’m 50 years old. I started building analogue recording gear when I was 15, before anyone knew about digital recording. Our technical staff at Millennia are in their 40s, 50s, and even 60s. We’re analogue guys who haven’t forgotten! We extol the tremendous virtues of analogue. Mics, mic amps, mixers, outboard, amplifiers, monitors — all part of our essential analogue recording and postproduction chain. We do not see this changing in our lifetimes. Have younger engineers forgotten about analogue? We see a new generation of recordists who are

March 2007

learning anew that it’s an analogue world, and that the most important aspects of recording occur in the analogue domain. As we say, ‘there’s no undo on your microphone preamplifier’.

Technically, what’s at the core of Millennia products? The human ear is the technical core of Millennia products. We use our ears to make design decisions on every step of product development. The human ear is the most advanced design we can imagine in the world of sound — it has dynamic range far surpassing

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the best recording and playback chain, it can detect a single molecule of air movement, it can repair itself. Most importantly, it makes us aware of design issues that our sophisticated audio analysis equipment cannot begin to resolve. As for the products themselves, we look for unique design approaches that set precedent, rather than re-do what’s already been done. Proprietary alldiscrete high voltage audio design techniques, handcrafted electro-mechanics, and components sourced for longevity and audio purity — this is the technical core of every Millennia product.

meet your maker

Are modern components and modern manufacturing legislation making the creation of high quality gear easier or harder? It’s my understanding that a respected professor of environmental sciences has said that the EU ROHS Directive is actually going to cause greater environmental damage than non-ROHS manufacturing. It’s a ‘net energy’ equation, and it

Shouldn’t you be offering at least a few digital products in your range by now? We’ve been working on a top flight analogue to digital convertor, and it sounds awesome. But it’s still not at the ultimate level we think we can achieve. Our new remote mic amp (HV-3R) will employ Ethernet, MIDI ports, and a beautiful software control environment running under XP, Mac OS, and Vista. Someone

appears that the EU got it backwards. That said, Millennia is compliant with WEEE and ROHS, but we find the whole affair a political mess of the highest order, with many unanswered questions. As for maintaining quality, we’ve not found any ROHS component that negatively impacted the sonic quality of our gear. Will ROHS manufacturing techniques last as long as non-ROHS techniques? Time will tell.

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March 2007

09.02.2007 08:18:07

meet your maker once said ‘digital has no soul’. There are a pile of digital products on the market claiming ‘analogue transparency’ but most of the software and digital hardware we’ve auditioned is dreadful. Maintaining real-world analogue life in a digital signal path is not trivial. We continue to learn.

What’s toughest to do really well — preamps, EQs or gain reduction — and how does a good one distinguish itself? Good question! I think every audio designer will give you a different answer. When I was developing the HV-3 mic amp it was the hardest circuit design I had ever done. It took years to get right. How does a good design distinguish itself? It’s all about sonic quality. The HV-3 was designed to record classical music, so it required enormous dynamic range, ultra-low noise, fast slewing, and most importantly it had to sound like the performance. It had to deliver lifelike timbre and spatial realism. Today, as we approach 30,000 HV-3 channels shipped into the most demanding applications on the planet, being used on well over half of all major Hollywood film scores, I think the HV-3 has proven its worth many times over. Look also at our client list on the website. It’s no accident that we have attracted a large number of the world’s most critical recording artists and engineers. I’m proud of our reputation, and I think our extensive key-customer list gives a more convincing argument for our products than anything I could say.

US Library of Congress has asked us to redesign our LPE-2 archiving system with some interesting new features. We will release this updated product in 2007. We’re experimenting with some very exciting new technology for dynamics control. It’s not FET, not VCA, not opto-resistive. It is something very unique that I can’t talk about yet. Quite a few engineers and musicians have been asking us to offer a sub-set of the TD-1 half-rack recording channel — something more along the lines of a top-end DI/Reamp/Speaker Soak/routing box, without the HV-3 mic amp or parametric NSEQ circuits. We’re considering that. I’m exploring a new digitally-controlled analogueixing environment, employing core elements of our Ethernet environment developed for the HV-3R remote mic amp system. We actually have way too many ideas, and not enough human resources. By the time a product does get to market, it’s often gone through years of

circuit revisions, listening trials, and experimentation. Most audio companies can define a product and have it to market in six months. We’re not going to sacrifice our basic principles just to have more products in the market. It’s taken 15 years to get essentially nine products to market. Maybe we’ll get faster in the future, but I doubt it. ■

Contact MILLENNIA MEDIA, US: Website: www.mi-media.com

How do you reconcile wide bandwidth, headroom and dynamic range with the MP3s most consumers end up listening to? We try not to worry about things that are out of our control. We focus on maintaining our products as industry references; products that engineers can trust to deliver an abundance of margin; gear that injects no ‘weak link’ into the audio path but remains absolutely faithful to programme source. You know, even MP3 audio can be improved when we manage original programme at a high level of sonic integrity. Remember, a codec works on the entire signal, so a cleaner, less noisy one produces better results. Why does analogue mean sonic character and signature to some and sonic neutrality and transparency to others? Analogue offers the entire sonic spectrum, from pure invisibility to thick colouration, and everything in between. Digital plug-ins attempt to emulate some of the great analogue colouration gear, but digital can’t emulate invisibility. You can’t go backwards and extract neutrality from a coloured signal. This is why we think it’s important for DAW users to capture their analogue programme in a pure and unadulterated manner. Colour can be added in mixing and postproduction, via analogue and digital signal processing, but colour added at the mic/pre cannot be reversed. What’s left to design? Millennia is currently looking at ways to expand our remote mic amp hardware and software into digital control of every analogue product we make. We want to complete our A-DC in 2007, but will only ship it if it performs more transparently than everything else we’ve tested. We’re close, but not quite there. I’m personally working on some custom mastering equipment for Georgetown Masters, which will be on the website later in 2007 in our custom shop area. We’re always experimenting with new discrete amplifier designs and will announce at least one new high voltage Class-A FET device in 2007. The March 2007

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technology

Spacial sound to air Achieving a ‘stereo’ reproduction from a compact loudspeaker source has baffled boffins for years. Airsound’s Ted Fletcher explains the thinking behind a process that he believes is the solution.

HE PATENT APPLIED for and achieved by Alan Blumlein (UK patent No 394325 dated December 14 1931) was the basis for stereo recording and reproduction for the 20th century, and it’s illuminating to try to get inside the thinking of the man and to see how history has been shaped by his ideas. The original problem that Blumlein was trying to solve was in two distinct parts. 1. When watching motion pictures on a large screen, the sound of the dialogue often did not appear to come from the characters on the screen; the characters moved around and the sound stayed fixed. 2. When listening to reproduced sound in a domestic environment, the sound was often ‘muddled’ and distorted by the intrusion of reflections and reverberation in the recording environment; an effect that is overcome in real life by the rejection of such interference due to binaural hearing. Blumlein tried what we now think of as being obvious and normal types of experiment, using two microphones to simulate the ears and reproducing the sounds via headphones. And he got the expected results, which are that it is possible to get reasonably good stereophonic effect; but when he tried to reproduce those signals from the two microphones through loudspeakers he writes ‘The effect is almost entirely lost and such systems have therefore not come into common use as they are entirely unsuitable for purpose.’ (!!) On page one of the patent application he writes: ‘… with two channels entirely separate, it is known that this (stereophonic) directional effect can be obtained, for example, in a studio. But if the channels are not kept separate (for example, by replacing headphones with loudspeakers,) the effect is largely lost.’ We have to bear in mind here that the year is 1931 and there is no concept of ‘stereo’. Previous medical and experimental engineers had done work on directional hearing back in the mid 1920s, this

The little cube (110mm wide) is in production and with a small sub will be used for an iPod dock or a TV loudspeaker. 60

was work associated with defence, to do with locating submarines and aircraft. They had said that directional hearing was determined by a combination of phase and amplitude; that is, our ears can determine the direction of a sound source by the phase relationships of the signals and by the differences in volume level as the sound hits our ears. They stated that at low frequencies, up to around 700Hz, the phase is all important, and above that frequency, the amplitude becomes dominant. It would seem that Blumlein accepted this as fact as he mentions it in his specification; he proposed a way of emulating the ‘direction mechanisms’ of the ears in electronic and acoustic terms, and he came up with a simple method of recording that we now know as ‘Blumlein pair’, or with slight modification, M/S. The simple Blumlein pair is a pair of ribbon microphones set as close together as possible and with their axes at 90 degrees. The modified version, and to me the more interesting is the M/S variant (the M/S stands for Middle and Side) H i s argument was that by using a direct source microphone and a fig8 response microphone at 90 degrees, one could The classic M/S microphone set-up. record all spatial information and reproduce it effectively and controllably via loudspeakers by manipulating the directional information. His arguments in the patent are that by filtering of the ‘S’ signal, you can achieve the dominance of phase difference at low frequencies and amplitude differences at high frequencies. For most of the last 75 years we have become accustomed to fine realistic ‘stereophonic’ recordings that can be fully appreciated sometimes with headphones, but more often by nailing ourselves to a sweet spot at the apex of the triangle made by us and the spaced loudspeakers. I have pondered many times about the inadequacies of stereo, and the idiocy of manufacturers straining to produce any sort of spatial sound out of pairs of loudspeakers close together in the same box. One night in August 2005, after preparing notes for a lecture on psychoacoustics and the realities of what we perceive rather than what we ‘hear’, it struck me that while debate raged about numerous systems’ attempts to reproduce spatial sound in true three dimensions it was all based around sound being generated from A commercial Pure Tempus spaced loudspeakers. modified with Airsound. resolution

No-one seemed to have tried to develop the leads hinted at in the Blumlein patent; hints about sum and difference recording and reproduction. The basic theory of ‘sum and difference’ or ‘mid and side’ is well known and well used; after all, it’s how VHF radio is transmitted. But what is less well appreciated is the flaw in that basic argument when it comes to recording. The system works because a cardioid microphone, set up facing a performer is the ‘sum’ signal; that is, left + right. A fig-8 microphone set up at 90 degrees to the performer is the ‘difference’ microphone, that is, left – right. When the outputs of the two microphones are added, then the answer is 2x left (the right signals cancel out). When the output of the difference microphone is subtracted from the output of the ‘sum’ microphone, the answer is 2x right (the lefts cancel). So we have created a ‘stereo’ signal, left and right from these two closely spaced microphones….. haven’t we? Sorry, the answer is no. What we have done is to manipulate the outputs of two microphones, not much more than that. The ‘left’ information collected by the ‘sum’ microphone is subtly different from the ‘left’ information hitting the ‘fig-8’ microphone; its phase might well be coherent but the pick-up is very different, particularly in the indirect sound content; and this is the most important area involved with appreciation of depth. However, there is no argument that recording using this set-up can produce wonderful results, and completely overcomes one of the banes of ‘stereo’ production, the ‘hole in the middle’. So how could this thinking be applied to reproduction? If you can get a beautiful stereo image from a co-incident pair of microphones, why can’t the system be reversed? In 1968, while I was running a small demo studio in Denmark Street, London, a man called James Holm was writing a patent application in the USA based on this very principle; it was a patent for musical instrument amplifiers, to get them to sound ‘bigger’. The system was not a commercial success, but it was modified and developed to become a product marketed by Fender, and more recently by Groove Tubes*. It is a single loudspeaker enclosure with drivers mimicking the ‘sum and difference’, and with electronics producing spatial information that gives an impression of width. I found this original work by Holm, and decided to do some experiments to see why this seemingly obvious improvement in reproduction had not been adopted everywhere. I tried simply processing conventional left/right into ‘sum and difference’ in the same classic manner and reproducing them through a mono loudspeaker (sum) and a dipole (an open backed loudspeaker set at 90 degrees)**. The results were impressive, but very much lacking in detail, and certainly ‘mushy’. So I came to the conclusion that the errors implicit in the arithmetic of sum and difference — the fact that left is not necessarily left — were affecting the results, and that to achieve ‘good stereo’ like this, then probably

A selection of DAB digital radios. March 2007

technology there would have to be some modification of recording technique. By thinking that way, I was prising open a can of worms! As a recording engineer, to make a good recording you have to have an appreciation of the performance, and a very clear understanding of how the listener will interpret it from the recording. To do that, we need to know more about what our hearing considers good and acceptable and if there are any aspects of ‘not so good’ that have to be watched out for. Consider a voice in front of a microphone. Our brains imagine a sort of idealised sound of a voice; the extraneous noises are ignored and the voice is clean and pure. But this is not the truth, of course, in the real world there are all sorts of confusing noises and reflections, but to make our recording acceptable, we need to idealise it, so we place the microphone close to the voice. This has the effect of minimising interfering noises, and it also ‘softens’ the sound because of the proximity effect. If the sound source is very close to a microphone, there is a noticeable lifting of the bass response. Because of the proximity, there’s also a possible problem with wind blast and overloads, so some compression may be necessary. So already we are starting to depart from the attitude that seemed so right to start with. Listen to a sound, put up a microphone and record it, and we will get a true recording of the sound. Not true. I suppose this is the biggest lie of the recording business. The sound we think we are hearing in real life is an idealised version of the real world. What we want to reproduce in a recording is that idealised version. And what we want to hear is just as idealised, and so we come back to my loudspeakers. With that simple and ‘ideal’ sum and difference setup what we want to happen is that the main recorded information comes from the central mono loudspeaker as truly and accurately as possible. As this sound goes to the sides it ‘mixes’ in the air with the difference signal, is heard directly, but also by way of reflections from other random surfaces in the room. Our ears register and recognise the direction of these areas of mixing by virtue of direction, and by content; they are the sorts of sounds that our brains expect to come from those directions, so the spatial image is confirmed. The mono signal reproduces the main information while the difference signal reproduces the spatial information. In practice this partially happens, but in the real world, to get sufficient spatial clues to the ears with the simple dipole loudspeaker, the difference signal needs to be at a higher level than realistic, introducing serious errors in both spatial perception and even in the dynamics of the material. The answer lies in some careful design, the use of a pair of difference loudspeakers (in the same enclosure) and another quirk of physics — the ‘surface effect’. Walk around a lake late at night in the quiet and an extraordinary effect takes place; it’s possible to hear conversations from the opposite side of the lake an impossible distance away. This is the surface effect where sound runs along a surface ignoring our A small ‘Marshall’ guitar amp famous inverse with very wide sound. March 2007

square law. Applying this to the loudspeaker, using just a small amount of surface effect on the spatial signal makes it effective and realistic. Introduce some response filtering into the signals and some careful design of the loudspeaker enclosure and we have a practical system that reproduces remarkable width and depth and is completely free of that sweet-spot effect! Considering the conventional way of listening to music, ideally, the pair of loudspeakers should be set at 1.6 metres apart and the listener should be placed about 2 metres from each loudspeaker and exactly equidistant from each. Any deviation from this position will cause errors in the perception of the stereo image generated by the loudspeakers. Any movement to the side alters the path length ratio to the speakers, so there will be phase and timing errors in the sound reaching the ears. The effect is that the brain loses the ‘picture’ and the sound becomes even more muddled than the image was in the first place, that original ‘muddling’ being created by reverberant paths and reflections from the two loudspeakers and the walls of the listening room. But, if the higher frequencies are reproduced by directional drivers, and provided the listener is accurately at the sweet spot then good stereo is certainly attainable; better than Alan Blumlein could have imagined in 1931. The ‘cons’ of conventional stereo are many; the largest is certainly the necessity of maintaining the exact sweet spot for the listener but in the real world of mass communications the most difficult problem is the provision of spaced loudspeakers. Manufacturers cram horrid little drivers into small boxes and laughingly call their systems ‘stereo’, domestic HiFi so often has to bow to the interior design requirements of a small lounge so that any thought of a sweet spot is lost forever. One of the biggest travesties of all is the pub or club where any appreciation of a record is difficult when all you can hear is one side! For Airsound to give reasonable results, the single source spatial system needs a little space, and some flat surface. It does not work well cramped into a bookshelf but what does? Sitting on a tabletop, or on a shelf with its back to a wall, the spatial information will jump out, and if conditions are ideal, then as well as the enhanced depth in the sound, a good ‘stereo’ panorama will appear. With a difference, and that difference is that the image will be the same from any listening position in the room. ■ * There have been other systems of reproduction designed to enhance spatial effects and remove the reliance on exactly spaced loudspeakers, specifically from ‘Embracing Sound’, ‘KEF’, and ‘Bose’, all use manipulations of ‘difference’ information combined with controlled reflection. ** The actual principle of reproducing sound by way of ‘sum and difference’ was proposed by Blumlein and others in the 1930s and so represents ‘prior art’ in patent terms. The details of how to achieve the results are what make the patent, both of Holm, and my own.

Contact AIRSOUND: Website: www.airsound.net Airsound technology is the subject of three patent applications to date and is available to manufacturers under non-exclusive licence. Airsound LLP., of which Ted Fletcher is a partner, is a company set up specifically to develop and administer the licensing of Airsound technology. The first areas of application for the technology is likely to be where spatial sound is most difficult to achieve, such as small radio receivers and iPod docks.

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slaying dragons

Fault finding: what to do when it doesn’t work Failures are all too common, quite often accompanied by time pressure, especially in broadcast or live sound applications. Here JOHN WATKINSON argues that dealing with failures and getting the show on the road is an art form.

HAVE BEEN FIXING things for as long as I can remember. As a student the motivation was impoverishment. Subsequently I fixed computers for a living, moving on to teach others how to do it. Today I get to investigate technical problems where the severity has led to litigation. Fixing things can be very rewarding. We get accustomed to things and it can be quite distressing to be told our favourite gadget is beyond repair, which today is a euphemism for we don’t know how to fix it, we want to sell you a new one, or we don’t know where to get the parts. Actually fixing things is also a very good way of learning to design. It teaches you about what does and doesn’t work. Over the years I have come to realise that if the right approach is used, faults can be located in anything, whatever the technology and whether one is familiar with it or not. If the wrong approach is used, faults may never be fixed and time will be wasted. Thus an efficient and rational approach to fault finding may make the difference between the show going on and giving the audience their money back. Fixing things can also be extremely entertaining. I recall being observed fixing a carburettor by a couple of Jehovah’s Witnesses who finally enquired if I thought there was anything in the Bible that was relevant to modern living. I replied that there was not much in it about automatic chokes. The greater the urgency, the better each faultfinding decision has to be and good decisions cannot be made when surrounded by headless chickens. People sometimes think a fault will be located sooner if the person fixing it is told in minute detail how big the disaster will be if it’s not fixed, what they think of 62

the manufacturer and speculation over the illegitimate birth of the salesman. The reverse is the case. Another interesting phenomenon is that the vehemence with which the cause of the problem is alleged is often inversely proportional to the likelihood of it being correct.

‘People sometimes think a fault will be located sooner if the person fixing it is told in minute detail how big the disaster will be if it’s not fixed, what they think of the manufacturer and speculation over the illegitimate birth of the salesman.’ I remember one large systems problem I was sent to sort out where my major contribution was keeping the customer away from the engineers. Whatever the urgency, the initial phase must always be to learn as much about the symptoms as possible and this must be done calmly. Where there is a lot wrong it may be necessary to adopt a form of triage so that manpower can be put on the worst problems first. It amazes me that, before anything has been learned about the nature of the problem, people expect resolution

me to know exactly what sophisticated test gear I will need. In this early stage, my sophisticated equipment consists of a mug of coffee (white-without) and a blueberry muffin. Faults come in categories and a speedy solution relies on an early identification of the category. A spontaneous failure of something that has already given a period of satisfactory service requires a different approach to something that has never worked. If something used to work but no longer does, possible reasons include component failure, sustaining damage, incorrect maintenance or a change in the environment. I wish I had a pound for every time I have stressed the need to look for all the stupid things before assuming a complex problem. Power supplies and fuses should be checked early on. Make sure the fans rotate and the filters aren’t blocked. If something has never worked, the possibilities are unfortunately greater. Perhaps it isn’t meant to do what is asked of it, or what was delivered isn’t what was ordered. Perhaps it hasn’t been installed properly. Consider cables swapped over, configuration links or DIP switches wrongly set, shipping brackets still in place or even parts missing before assuming a difficult fault. Again if the unit has never worked, try to find out if there is another one somewhere that does work. It may be possible to inspect that unit to look for differences in configuration or settings. I had promised myself not to air any prejudices about software, but I have seen too many problems to keep quiet. All I will say is that the standard of software seems to have declined dramatically of late. In a complex device such as a mixing console, the chances are much higher that, rather than not working at all, it will suffer from problems where certain features or channels don’t work. In order to locate faults of this kind it is extremely useful if a skilled operator is available who can place the unit into different configurations to see what does and does not work. In conjunction with the block diagram, different signal paths can be tested that incorporate or by pass different elements and, by considering which paths pass a signal and which don’t, the location of the fault can be deduced. The same approach can be taken with equipment such as routers. In some cases a system can be broken into sections that can individually be tested. The quickest approach here may be to use the so-called binary split technique. The system is cut in half arbitrarily and each half is tested. The half that doesn’t work is then cut in half again and so on. One of the most difficult faults to deal with is the intermittent. The more intermittent, the harder it can be to locate. With a steady fault, it is possible to test for the right signal at various places and thereby isolate the cause. Intermittent faults do not respond to this method, as in between failures all of the signals will be correct. Procedures for finding intermittent faults can be divided into passive, active and intellectual. Passive procedures include connecting logging equipment to capture the fault conditions, and designing test procedures to find out the configurations in which failures occur. Active procedures include deliberate attempts to provoke the fault, and trial replacement of parts of the system. In the purely intellectual approach, everything that is known about the fault is gathered together and, from an understanding of how the system is built, hypotheses are put forward about what the fault might be. This approach often works well if it takes the form of a debate between two people. The debate can and perhaps should take place at a different location to the fault. I can remember many occasions where an intractable fault that had resisted all morning was identified while discussing it March 2007

slaying dragons over lunch. Often the proximity of the problem makes it hard to see everything clearly and there will be distractions. It is vital to keep an eye on the big picture when looking for really intractable problems. Intermittent faults are often due to the combined effect of several influences and can be seen as a form of tolerance build-up. Tolerance build-up is where individual components are not faulty as such, but just a little out-of-spec. In combination with other good parts they may work fine, but in combination with other just-out-of-spec parts there can be a problem. This type of fault will often respond to deliberate provocation, know in the trade as margining. Imagine a computer bus or a network cable that has a weak line driver. The transmitted signal is a little down in

level, but the receivers can still pick it up. However, if into this case we add some electromagnetic noise, and an unstable AC supply, it can be seen that there might be the odd error. It might require low power supply voltage in conjunction with high interference to cause the errors. In margin testing, the system is deliberately degraded to make intermittent faults more common. There are various ways of doing this. In hardware, the power supply voltage can temporarily be adjusted down. In network and interface cables, an extra length of cable or a device called a ‘cable clone’, that simulates a length of cable, can be inserted. This technique works extremely well with AES-EBU and SDI interfaces.

Finally there is a political aspect to all of this. It is in the manufacturers’ interests if products cannot be repaired because then a new one is required. I for one do not subscribe to this throwaway society and often repair things on principle. However, it’s getting harder to do because doors are being closed. We now find that charity shops no longer sell used electrical goods. We are told that it is for liability reasons and because of new safety regulations. I wonder if there is any evidence of wholesale slaughter by second hand appliance, or whether this effect was the true agenda and safety was the excuse. So what happens is that the dumps fill up with discarded appliances that have minor faults and we have to buy new ones. ■

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March 2007

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your business

Tempus Fugit You may have gotten your financial house in order but who’s minding your time? Time bandits abound for producers with heavier-than-ever workloads. Time management is the key, says DAN DALEY, who also adds some key info on producer royalties in the download age.

‘The 1970s was an era of leisurely-paced productions, of going to the studio not just to record an album but to write it as well. All at card rate. No wonder everyone was so happy back then. And you thought it was the drugs.’

OT LONG AGO, I was standing on the threshold of Studio C at Hit Factory/Criteria Studios in Miami. It’s not a huge room, but it is an historic one. The record that was made there that stands out though, was the Eagles Hotel California. The making of that record stretched over six months. The 1970s was an era of leisurely-paced productions, of going to the studio not just to record an album but to write it as well. All at card rate. No wonder everyone was so happy back then. And you thought it was the drugs. Producers today will work on one or two tracks at a time but on multiple projects simultaneously, often in conjunction with other producers, sometimes on the

other side of town or the other side of the planet, in a variety of studios and in a cornucopia of formats. A lot has changed since the Eagles were making unhurried records. Unfortunately, the earth’s revolution on its

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axis hasn’t, meaning there are still only 24 hours in a day. Learning to manage that time has never been more crucial to your career. Stuff is coming at you left and right. Anyone with even a modest amount of success (and today that could include a ringtone or 30 seconds of an indie band on an Orange commercial) can be quickly and constantly besieged by incoming demos from aspiring artists. How do you manage them all? You could take a page from music publishers. Years ago, in response to a slew of lawsuits alleging that they stole a song from some unknown that had sent it in, publishers created the ‘envelope code’, a word or phrase or just some alphanumeric gibberish that they would give out very selectively. If an incoming envelope didn’t have the code on it, it went directly to the round file. Saved a lot of time and legal fees. Why that helps a producer is illustrated by David Z’s experience. Z, who has produced and engineered Prince and Johnny Lang, tended to listen to every artist demo that came his way, an old habit that was hard to break. However, he gave up on song demos after one came in through the younger brother of a childhood friend. Years later, when a song with the same name (but no other similarity) showed up on a Jody Watley record that Z hadn’t directly produced but on which his name appeared, the songwriter sued Z, MCA Records and Warner Music Publishing. The label and publisher paid $9,000 to make the ‘nuisance’ suit go away, and Z was implicitly responsible for a third of that. ‘Sometimes, if you make time for something like that, it backfires,’ he says. But most of the time, making time for things works to your benefit. Unless you’re doing commercials or film, it’s likely that your actual work -– music sessions -– start later in the day or in the evenings. Putting time aside one or two days a week a couple of hours before your commute to the studio begins can be dedicated to catching up on accounting and other mundane tasks. (Assuming the studio is not in the spare bedroom, in which case you have no excuse.) On a weekly basis, time needs to be allocated to conferencing with a manager, if there is one (managers have multiple clients and squeaky wheels get oiled); to reviewing any ongoing logistics for current projects (note to self: ship rack to Manchester!); and touching base with clients and prospective clients. Email is great March 2007

your business for that sort of thing but in a digital age an analogue conversation can produce wonders. Organising time into modules like this is probably one of the things that you’ve laboured so hard to get where you are in music to avoid having to do. Besides, the best marriages I’ve encountered in this business are ones where time has been budgeted for family. Doesn’t sound very romantic but it works. Some producers successfully apply technology to time management and it pays off for them. John Boylan, who produced Boston, Linda Ronstadt and Charlie Daniels, uses (in order of importance to him) Micrologic Info Select 8, Eudora (for email), the Lotus Organiser (for scheduling), Microsoft Excel (customised for managing budgets and for tracking sessions), and Contract Creator, which has templates for key agreements including the American Federation of Musicians’ session contracts. ‘I suppose the time management software I use rivals the amount of software I use as a producer, which includes Digital Performer 5.1, Pro Tools 7.1 and Peak 5.2,’ he says. ‘Some of it I’ve come to rely on very heavily; for instance, I couldn’t exist without Info Select.’ The production assistant, or PA, is something we’ve discussed in detail on this page a couple of years ago, but let’s quickly review the topic. PAs aren’t for everyone. First, you need to be billing enough to warrant a salary or a retainer, even on a per-project basis. I’ve seen some producers, especially in Nashville, who will share the services of a PA. There is a lot of redundant work involved -– more often than not you can book two sessions at the same studio in the time it takes to book one. However, with the changed nature of music production -– more small bits to work on rather than all large projects soup-tonuts –- the level of detail can bog down even the most industrious PA. Furthermore, I’ve had some producers tell me that they’d rather not take the chance of having too much information about their business available to other people. I’m not saying that PAs are gossips but people do love to talk. Old dogs can teach young ones some new tricks. Ron and Howie Albert, who recorded about half the soundtrack to the Boomer generation, including Layla, James Brown, Jimi Hendrix and Aretha Franklin, are old hands at this matter. ‘Digital definitely gives you more to do but I personally think it does in a good way,’ says Ron. ‘While in the production mode, our experience allows us to think ahead and know what trickery we can do digitally later, allowing us to keep the session moving. If we were still in the analogue world, we would have to do a certain amount of that trickery while we still had the artists and players in the studio. We also do a lot of automation within Pro Tools. In fact, most of the trickery we do these days is in the box. It’s not that digital is so much better sonically; it really is all about saving time so we can do more productions, increase cash flow, and not compromise the quality. I think we started figuring this out when automation first came along years ago.’ Time is money and this is a good time to mention some significant pricing changes in music, changes that will ultimately affect what flows into producers’ pockets from royalties. Apple’s iTunes had set the bar for digital download pricing at 99 cents (about UK£0.55). While iTunes putatively gave the record business the digital business model they so desperately needed, major labels quickly cooled to the idea of selling everything for the same price. Earlier this year, with the licensing contract between Apple and the majors up for renewal, the labels argued strongly for tiered pricing: a popular track should cost more, while less popular tracks could sell for less in hopes they March 2007

would build demand. Apple’s existing pricing model prevailed, but there are plenty of other schemes out there that are offering music at a wide range of prices. In October, Aimie Street went online with prices as low 23 cents, mostly for indie artists. MySpace, the wildly popular Internet social networking site, is planning to sell song downloads and allow the artists and labels to set their own pricing. All this is going to make hash out of the accounting for producers whose revenues include production points (not to mention create a new dimension to the piracy problem, but that’s for another time). In fact, the producers I contacted about it hadn’t given it much thought yet; apparently, neither have their managers. As has been the case for much of the digital era,

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legislation that would cover this sort of thing has yet to be drafted, much less enacted or enforced. The short-term solutions will come from a combination of individual negotiations between producers and artists/labels, and non-legislative blanket agreements between distribution outlets and copyright holders, such as the deal announced in 2006 between Warner Music and YouTube to assure that the online video portal is using only licensed material and that those entitled to participate in its royalty payments do so. Nonetheless, you cannot count on record labels, management or anyone else to do the right thing by you purely on principle. If you can’t find a way to front-load your payments (my mantra), you’ll have to scrutinise your royalty statements that much more intensively in the future. ■

headroom MEDIA PRODUCER Thanks for the review of Dolby Media Producer in Resolution (V6.1). I was interested in the comments about the Dolby Media Encoder element seeming expensive compared with the good value offered by Dolby Media Tools and Dolby Media Decoder. As you pointed out, Dolby Media Encoder is designed specifically to meet the needs of high turnover DVD authoring facilities, and includes support for three networked encoding workstations using a central encoding server, rather than just a single standalone workstation. So this is something that big facilities will perhaps only need to buy one of — not copies for each studio — and I think will prove to be good value for these users. But we do agree that single studio facilities will also want access to the features of Dolby Media Encoder too. I wanted to flag that we will very shortly be making an announcement about a new, lower cost version of the software designed specifically for these users. To nitpick on a couple of minor technical points in the original article for your future reference. It is stated that in Dolby Media Tools, the Dialnorm metadata parameter cannot be edited. This is not the case — it is possible to adjust Dialnorm and we expect this to be a very useful feature. It is noted that Dolby TrueHD requires a decoder in the player. Of course, this is mandatory in the HDDVD standard, so every HD-DVD player should have this capability. Jason Power, Market Development Manager, Dolby UK Product information on the new Dolby Media Producer SE (Stand-alone Edition) is included in the Products section p17. ZS

Book review Tony Visconti – The Autobiography A look at the life of one of the top-ten control room maestros is a rare treat and underlines just how few books there are on the sound profession that do not concern themselves completely with the technology and science of the subject. Visconti charts his life from childhood music lessons through to the present day taking in his golden years with Bolan, Bowie and a host of other acts I didn’t know he’d worked with along the way. I found the tempo of the book a little inconsistent. He starts off with enormous and entertaining detail on his childhood and adolescence in New York and the pace increases inline with the speed of his life as his career takes off. The account of his arrival in England and his formative work is well observed and reveals a work ethic and enthusiasm that is staggering and undoubtedly contributed to his success. His persistence and resolve is an example. You get a very candid and honest view of the music studio scene in London in the late 1960s — he namedrops studios that disappeared decades ago — its habits, attitudes and the record making process. It was a different world, but it was a business. At its heart, it was driven by people who understood that artist creative output was what made the business turn profit and that this model required facilitators around them to make it happen and to keep it happening. Such a different world; and I’d imagine that younger readers may be shocked at just how raw and energetic it was. Visconti’s story is fascinating because his involvement as a main player spans the industry’s opening up, maturation and gradual eventual decline.

This was also accompanied by rapid technological development, which he consistently adapted to and employed to good effect. You get the impression that he enjoys talking about his earlier years more than the later ones — we don’t get to Berlin until well over halfway through by which time the recounting has become brisker and more shortform. However, all the key events and sessions are here along with some interesting cameo appearances and exchanges; he doesn’t always like everyone. We also get an insight into his personal life, which inevitably seemed to have been at odds with his professional life. We all know Visconti for Trident and Hansa Ton, his impeccable balancing on the T.Rex singles, putting drums through an early Eventide, and opening room mics on a vocal. What is not so well known is the fact that he came into the industry from the music side and got his first breaks as an arranger, something he continued to do while crossing over into a pivotal role in the control room. Never afraid to travel for work, he’s played on sessions, he’s engineered, produced, mixed, he had a proper early home studio, a record label, a commercial studio. He’s done it all and he avoided draft to Vietnam because he’d recently come off heroin. It’s a recommended read purely for the insight into what he was thinking and how he managed some pretty remarkable bits of recorded history under often very difficult circumstances. He emerges as the consummate professional. ZS www.amazon.co.uk

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March 2007

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