Resolution V4.7 October 2005

OCTOBER 2005 V4.7

THE NEXT GENERATION AUDIO PRODUCTION MAGAZINE

Lars Nilsson preamp techniques

Ken Thomas on staying current Why the SAN is shining in post Getting Dolby approval for your room The E-trap — a tuneable electronic bass trap Meet your maker: Dirk Brauner — Brauner Microphones Ten things to do with an old analogue recorder

REVIEWS • Solid State Logic C300 • Buzz Audio ARC 1.1 • Universal Audio LA-3A • • • Digigram VX882HR • and UAX220 • Focusrite OctoPre LE

AEQ DR-100 DPA 4006 TL Empirical Labs Lil FrEQ

• SE Electronics Icis

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October 2005 V4.7

ISSN 1477-4216 THE NEXT GENERATION AUDIO PRODUCTION MAGAZINE

News & Analysis 4

Leader

4

News Sales, contracts, appointments and the bigger picture

Craft 38

Lars Nilsson

Ken Thomas Sound reinforcement, desk manufacturing, and studio builds as an engineer, producer and mixer with prog rock, New Wave and the current crop.

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66

48 Capturing sound with a primary emphasis on using different mic preamps and changing the acoustics.

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14

52

54 59

Sweet Spot You’ll need a licence for that. After the design and build of a post suite we follow one man’s preparation and the process involved when dealing with the Dolby police.

60

Products

New introductions and announcements plus Digidesign and Steinberg platform news.

Headroom

Blumlein technique, peaked quality again and audio lookilikies.

In the picture

It’s Clash of the Titans yet again — Blu-ray vs HD DVD — another format war.

How bloody tedious.

Meet your maker

Dirk Brauner — Brauner Microphones is celebrating a decade. Its founder talks shop.

Steinberg Top Tips

Nuendo3: Warp To Picture for post.

We show you how.

Katz’s column

With Becky and Fred on honeymoon, Bob questions the logic of peaking to full scale to fill up all those bits.

Ten

Things to do with an old analogue recorder.

Business 50

Why the SAN is shining in post

64

New developments in network storage are putting the pedal to the metal for production workflow and changing people’s jobs in the process.

Your business

Daley wants to see the producer become the object of more commercial desires. He wants to see them selling T-shirts.

Technology 56

The E-trap

62

A tuneable electronic bass trap that offers new ways to think about low frequency room acoustics and acoustic design.

Slaying Dragons Good grounding is vital for safety and for sound quality, but is often neglected because it’s not exciting and/or because it’s not understood.

Reviews 20

Solid State Logic C300

22

Universal Audio LA-3A

24

Buzz Audio ARC 1.1

26

DPA 4006 TL

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, Keith Spencer-Allen, Terry Nelson, Jon Thornton, Neil Hillman, Nigel Jopson, Andy Day, Jim Evans, Kevin Hilton, Dan Daley, John Watkinson

28

AEQ DR-100

30

Focusrite OctoPre LE

32

Empirical Labs Lil FrEQ

34

SE Electronics Icis

36

Digigram VX882HR and UAX220

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 236681 Email: dean@resolutionmag.com

news APPOINTMENTS AUDIO TECHNICA Ltd has appointed Adrian Rooke as MD following the retirement of Paul Maher. Adrian joined the company in 2003, having previously held the position of group financial director at a leading UK building hardware distribution company. He was promoted to the position of financial director last year.

TUBE-TECH HAS appointed Audio Exchange International as its US pro audio distributor.

FAIRLIGHT HAS appointed Andy Duffield as regional manger for Europe, the Middle East and Africa. Broadcast and postproduction engineer Lukas Bower has joined Fairlight’s research and development team as project manager, Internet applications. Duffield joins after 12 years as sales manager at SSL. Bower joins after several years as chief engineer and CTO at LA post house World Wide Wadio.

Fairlight has appointed Ton und Studiotechnik as its Business Partner in Germany. Ton und Studiotechnik will oversee all sales and service of Fairlight products in the German market.

Leader

It’s a fact that every technology, craft or art has a ‘classic’ era that gets certain types dewey-eyed or visibly animated at its very mention. You can also slice a subject into a number of finer compartments to make the allocation of even more specific classic eras easier. Thus we can, with out too much trouble, identify the significant output of guitar plants in Fullerton and Kalamazoo to fairly eagerly agreed date spans (I can feel hackles rising). In more general terms, we can identify the time from the late 1950s to the late 1960s as a period of divinely inspired motor car designs. What captivates the connoisseur and the hobbyist in both cases is the freedom of thought and form combined with early ‘modern’ takes on issues that have never really gone away but have perhaps never been bettered for their purity of approach. There’s also a style and build quality thing going on because for things to last long enough for them to become classics they have to be built well or to be so beautiful that they encourage humans to preserve them.

When we look at our audio gear, each category type has a classic era and every long-enough-lived manufacturer also has a classic period. That’s great and I know what you’d say if I was to ask you to name classic mics, tape machines, compressors, EQs, and desks, among other things.

However, I can’t think of any other technologically-oriented industry that is as into the concept of ‘classic’ as the pro audio industry. The word is used for purposes of comparison, description and it’s named as a feature on new gear releases. We are totally obsessed by the classic. Is this because there is no, and doesn’t seem like there ever will be, a classic digital era? I can tell some classic stories about digital gear but that’s about as far as it goes. We yearn instead for classic qualities in our digital. Unfortunately this will never truly happen until we build longevity into it and make it a thing of beauty.

Zenon Schoepe

Midas adopts SuperMAC and HyperMAC

S E R G E A U C K L A N D, Preco’s sales director and 34-year veteran of the broadcast industry, has retired. During his career, Serge has worked at Michael Cox, Ampex, Pye TVT, Philips, Marconi, and Harris. We wish him well.

MILLENNIA MEDIA, the manufacturer of high-end analogue signal processing, has appointed HHB as its exclusive distributor in the UK. DK-TECHNOLOGIES has appointed William Boxill as vice president of its US subsidiary DKTechnologies America Inc. He began his career at Videotek and has worked for Tektronix and Leitch.

S A L Z B R E N N E R S TA G E T E C Mediagroup has appointed Syncrotech Systems as its distributor in Australia, New Zealand and the bordering Asian states.

©2005 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

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Midas and KT MD John Oakley; John Richards, general manager, Sony Pro-Audio Labs.

Midas has licensed AES50 (SuperMAC) and HyperMAC technologies from Sony ProAudio Labs, Oxford for implementation in Midas and Klark Teknik equipment.

‘In live sound applications, we are confident that the open standard AES50 approach has significant advantages over other technologies — in particular the exceptionally low latency and error correction,’ said Midas R&D director Simon Harrison.

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.

Midas says it prefers open standards wherever possible and views the publication of the AES50 standard as an important step towards the easier interconnection of large-scale digital audio systems. AES50, implemented by Sony as SuperMAC, can carry 48 bi-directional audio channels plus 5Mbps of generic Ethernet control data on a Cat5 cable. HyperMAC can carry up to 384 bi-directional audio channels plus 100Mbps Ethernet on Cat6 cable or fibre.

S2 Publications Ltd. Registered in England and Wales.

Company number: 4375084.

Registered office: Equity House, 128-136 High Street, Edgware, Middlesex HA8 7TT.

resolution

European HD firsts are Dolby The first three broadcasters to launch HD television in Europe will be featuring Dolby Digital surround sound on their channels as an integral part of the service. French pay-TV operator Canal+ Group, German pay-TV operator Premiere, and BSkyB in the UK will have Dolby technology in every receiving set-top box for their highdefinition services.

In addition to Dolby Digital 5.1, m a n y b ro a d c a s t e r s a re i n c l u d i n g Dolby Digital Plus in their set-top box specifications. Dolby Digital Plus is included in the Digital Video Broadcasting Project’s audio-video specifications, and has been selected as a mandatory standard for the new disc formats for HD DVD and as an optional standard for Blu-ray discs.

Dolby Digital Plus, an extension of Dolby Digital, is a versatile audio codec designed to adapt to the changing demands of future audio and video delivery and audio storage systems.

96-channel Dream for Grand Central’s HD Fairlight has installed a Dream Satellite at Grand Central post facility in London.

The 96-track DAW will provide the basis for the increasing number of Grand Central’s 5.1 and HD audio productions.

‘With the norm becoming 5.1 sound for cinema commercials, the advent of HD as the production format and the roll out of E Cinema in the UK, Grand Central required major increases in speed and track count in a single platform for the edit workstation,’ said Raja Sehgal, director of sound engineering at Grand Central. ‘The Dream Station offered the best current solution.’

‘The Dream offering allows for a highly “edit-centric” approach to sound editing and mixing,’ added technical director Ivor Taylor. ‘This coupled with a very high track density and depth of its editing tools allows us to significantly expand our creative abilities.’

• Human rights nonprofit organisation Witness will auction a one-of-a-kind, vintage Fairlight CMI Series III keyboard donated by Fairlight that has each key signed by musicians, composers and producers who used the Fairlight CMI from its early years. Personalities that have already agreed to take part include Peter Gabriel, Hans Zimmer, Thomas Dolby, Todd Rundgren, Mike Oldfield and Jan Hammer.

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October 2005

news Digigram changes A holding company created by Digigram’s senior management under the leadership of Digigram CEO Philippe Delacroix has acquired 10% of Digigram’s shares, as the company’s co-founders and majority shareholders have retired from operational tasks and have sold the major part of their shares. Co-founders Philippe Girard-Buttoz and Marian Marinescu have sold 19.52% of Digigram shares to the investment company Alto Invest that will have no influence on Digigram’s operations.

Digigram has also sold 100% of its subsidiary Innovason to the Innovason management team. As Innovason is an EtherSound licensee it will serve as a commercial and technological link between the two companies.

Korean success for Studer

F&P chooses Pyramix for post

F&P senior sound engineer Poul Sven de Haan.

Studer consoles have been supplied to Korean national broadcasters KBS, YTN and the MBC national radio network. KBS, the public service Korean Broadcasting System, and YTN, the only full-time news network in Korea, have both acquired Studer A928 analogue consoles.

Munhwha Broadcasting Corporation, one of Korea’s largest public TV broadcasters, is also the proprietor of a nationwide radio and television network of 19 regional stations and several of these have bought Studer.

Kwang Ju MBC is installing an OnAir 500 Modulo, ChungJu MBC has ordered two OnAir1000 consoles to use in new studios, Ulsan MBC has installed an OnAir 2000, and Chuncheon MBC is taking delivery of an OnAir 3000.

Russo works his Halo

Engineer and mixer Thom Russo has used his Metric Halo ChannelStrip on every project in the last two or three years.

‘Especially for important elements,’ he said. ‘I would say 95% of the time it’s on my lead vocal, and a lot of times on the drums sub-mix and on bass. Its transient response is leaps and bounds ahead of everything else that’s out there.

His client list includes Eric Clapton, Macy Gray, Audioslave, System of a Down, Enrique Iglesias, Juanes, Jay-Z, Babyface, and Kinky.

Leading Dutch postproduction studio Fröhlich & Plaisier in Hilversum, has chosen Pyramix as its new edit system. ‘Customers have become used to our speed and quality and Pyramix gives us both,’ said F&P CEO Sjors Fröhlich. ‘With the new Isis-remote control coming up we expect to satisfy our clients’ every possible audio requirement.’ Pyramix’s ability to import AKAI DD1500 files meant that F&P’s whole archive could be restored easily on separate tracks.

Czech National TV has bought 17 Vcubes for its postproduction department.

The machines are connected using a 1Gb Ethernet network and the audio postproduction facility employs 12 Fairlights and has a Dolby dubbing theatre.

Télévision Suisse Romande, the Geneva-based Swiss French television station of the Swiss Broadcasting Corporation, has ordered 11 Pyramix systems and ten VCubes as part of a major refurbishment.

Phoenix and Gateway merge at Pinewood Two of London’s recording facilities have merged to create a new orchestral recording facility at Pinewood Studios. The facility is called Phoenix Two and represents the re-emergence of Phoenix Sound from the ashes of the Wembley development and an alliance with Gateway studio at Kingston, which closed after ending a 21-year relationship with Kingston University. It will see Gateway founder and owner David Ward working in partnership with Pete Fielder of Phoenix Sound.

‘Combining our resources and client bases makes a lot of sense,’ said Ward. ‘Pete and I have talked about this for many years and have also been working closely on industry training projects.’

‘Gateway’s accomplished history will contribute to the strong industry brand

established by Phoenix, enabling this new partnership, combined with the comprehensive support of Pinewood’s spectacular facilities, to create a unique service to the film, television and record industries,’ added Fielder.

Gateway’s education and training facilities will also move to Pinewood where it will create a centre of excellence for professional development training.

The new studio is based around the 72-channel Neve VR formerly in Studio 1 at Phoenix Wembley with 5.1 monitoring and video projection. It will accommodate 30 musicians and will have tie lines to the enormous, and adjacent, F Stage at Pinewood, which will make full orchestral recording available in the future. A location recording facility will also soon be available and engineering staff from both studios will be at the new facility.

APPOINTMENTS AUDIO PRODUCTS International Corp has entered the professional and commercial sound markets with a range of studio monitors under the Energy Pro brand. The operation will be managed by Bill Calma, well known as the man who helped establish Tannoy in North America, who will launch the Energy Pro brand worldwide.

UK CABLING specialist V D C Tr a d i n g h a s appointed Nigel May as commercial director. He has worked at MI retailer Macari’s, Turnkey and Mediatools.

DARRYL ROSE has joined the Wohler Technologies sales team as director, special opportunities in Europe. He will be focusing on strengthening and expanding the Wohler UK sales base, especially in the OB arena.

ASPEN MEDIA has appointed James Bradley to its sales team from AMS Neve where he managed all the sales activity in the South of the UK. SONIC8 HAS been appointed UK distributor for Acousti Pro’s range of acoustic treatment products.

SYMETRIX HAS appointed Dan Gallagher to the newly created position of executive vice president — global sales and marketing.

He has previously worked with Ensoniq, Alesis, Korg, Marshall and Behringer USA.

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October 2005

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news APPOINTMENTS SRS LABS has appointed Michael Franzi as vice president of sales for its licensing business. He was previously chief marketing officer at THX.

C200s to Paris and Istanbul

DPA sponsors jazzfest

NAKAMICHI HAS appointed Arie van den Broek as general manager international.

He was Digidesign’s first overseas employee in 1993 and followed this as international sales director at Behringer, MD of Behringer Holdings in Singapore, and CEO of the Behringer Group of Companies.

T U R B O S O U N D H A S appointed Paul McMullan as UK sales manager. Previous UK Sales Manager Tim McCall becomes Middle East and Asia sales manager.

GERRY TSCHETTER has joined QSC Audio as director of product management. He previously worked at JBL Professional and Yamaha.

DINO VIRELLA has started VirellaPro Sales and Marketing, which combines a rep firm with a consultancy for small to medium-sized manufacturers.

Blue microphones has appointed VirellaPro Sales and Marketing to manage its sales and reseller channel development in the US. MI7 WILL distribute, support and market the entire range of DTS surround encoding products in Norway, Denmark, Finland, Belgium, Luxembourg, Portugal and Spain.

JAIME ALBORS has been hired by Crown International as the manufacturer’s intercontinental sales director. He joins from wholesale and retail operation Villa Music.

Blue Sound studios in Paris has invested in two SSL C200s for its Studios A and B. Best known for its sound to picture work for TV commercials, Blue Sound also offers music recording and mixing.

‘I was not happy with the sound quality in Studio B and having tried several “small” digital desks, plus the remote control solution, I eventually came to the conclusion that maybe a small C200 could solve the problem by providing the quality we were missing,’ said studio owner Carlos Duarte.

‘When we played our Pro Tools sessions through the C200 there was simply no more discussion: we all agreed that the sound quality was remarkable. Despite being a digital console, the C200 gave us the same warmth and good feelings that we got from an analogue desk — and without any of the drawbacks of analogue technology.’

Istanbul video and film postproduction facility Safak Studios is the first facility in Turkey with an SSL C200. The move is part of the facility’s intention to attract work from other countries and the desk is housed in a new Dolby Digital 5.1 room with facilities for 35mm and HD film postproduction.

‘The C200’s scalability and architecture will allow us to upgrade the console in the future to encompass technological changes and developments,’ said Safak Ustun with engineer CEO Idris Ustun.

Onan Karagozoglu.

ITALIAN PRO audio manufacturer Outline has appointed Discovery Lights & Sound Pte Ltd as its sole distributor for Singapore and Malaysia.

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DPA Microphones continued its sponsorship of the annual Copenhagen Jazz festival this summer, working as acoustic consultants and providing technical support throughout the festival, particularly at multipurpose venues not primarily designed for music.

DPA also provided IMK 4061 instrument miking kits, 3521 compact stereo kits, 4066 omni headband mics, 4011 cardioids, 4022 and 4023 compact cardioids, 4006 omnis and a selection of mounting accessories.

‘The Copenhagen Jazz Festival is very pleased with the support from DPA Microphones regarding acoustical improvements in our very different venues,’ said Festival manager Signe Lopdrup. ‘Each year we select new and often not very well tested venues for our concerts, and it’s particularly important to have the kind of professional support we’ve received from DPA over the last two years.’

Mediapro’s third Euphonix

CHYRON CORPORATION has opened its fourth European office. Chyron Italy is headed by Carlo Struzzi and joins offices in the UK (EAME Head Office), Germany and France.

STUDIO HAMBURG Media Consult International has been appointed distributor for PESA products in Germany, Austria, and Switzerland. The distribution remit of Netherlands-based distributor Burst Video has been expanded to include Belgium and Luxembourg.

DPA CEO Morten Stove at Tivoli Gardens.

Mediapro recently bought a 72-fader Euphonix System 5-BP for a newly constructed audio truck to be operated and managed by CLS Audio in Lisbon, Portugal. Mediapro now has three Euphonix consoles.

CLS Audio’s Andre Senos and Nuno Duarte serve as the architects of Mediapro’s

digital console upgrade programme.

‘The installation of a large format console represents a major investment,’ said Senos.

‘As a result, we took a close look at all the main players.Ultimately, we decided upon Euphonix, as our experience with the System 5 in the HD trucks has been faultless.’

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World-renowned pianist Niels Lan Doky, a long-time user of DPA mics, employed an armoury of models to record a jazz band for his new film, Between a Smile and a Tear: A Night at the Montmartre Club in Copenhagen.

DPA 4011 cardioid and 4006s omnis were used for acoustic bass, with 4011s as drum overheads, on the violin amps and as speech mics for the interviews.

‘In general I always use the 4011 cardioid or 4021 compact cardioid, plus the 4006 omni and 4040 special edition gold omni on recording projects, with the 4040s generally used on the piano,’ he said. ‘They are extremely precise and detailed over a very wide frequency range. I personally don’t know any other mics that I fully trust to deliver the same transparency in the sound reproduction.’

October 2005

news

news APPOINTMENTS DAYANG HAS appointed Steven Farmer as European technical operations director. He has worked for Drake Electronics and ClearCom in the past.

Dayang International has opened an office in Thailand and appointed Supat Sirithamrat as MD of the operation. He joins after 20 years with BEC Co Ltd where he was most recently deputy MD. Dayang International has appointed Maurice de Jonghe as director of sales and marketing, Europe.

JOHN COSENZA has joined Nvision as product development manager, Modular Products and Ray Bryan has been hired as customer service/support manager. John joins from Sierra Video Systems/ Kramer Electronics while Ray has held positions at Hewlett Packard, Sony, Tektronix/Grass Valley Group and Jones Futurex.

B R O A D C A S T TECHNOLOGY has appointed Stephen Harding as customer service manager. He previously worked as a teacher and has held positions at Sony Broadcast & Communications and Tandberg.

Martin Altham has joined Broadcast Te c h n o l o g y as international sales manager. He previously worked for Racal, NDS, Mindport Solutions, and Irdeto Access.

Broadcast Technology has appointed Bryan Fernandez as Middle East and Africa area sales manager. He started his broadcast career as a systems engineer at Racal Micro-Electronic and has worked for Quantel, Oxtel, Snell & Wilcox, BarcoNet, Teleste and JVC Professional.

ALLEN & HEATH Chinese distribution partner Eastern Acoustics Development director Y K Cheung (centre) being presented with the Distributor of the Year 2004/5 award by A&H sales director Bob Goleniowski (left) and MD Glenn Rogers.

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Ten System 6000s for Revolver

For the mixing of Guy Ritchie’s latest film Revolver at Luc Besson’s Digital Factory facility in France, film mixing engineer Vincent Arnadi employed a multitude of TC System 6000s.

Working with recording engineer Fabrice Saure, Arnadi used his own six System 6000 racks with Digital Factory’s four 6000s (plus a rented one for the final master). Including the double AES card, Saure and Arnadi. the setup represented 160 digital I-Os. A total of 60 effects channels, 30 channels for music, 30 ambience channels, and 40 channels for dialogue were used. Arnadi uses two screens displaying four TC Icon software editors to control the eight first units, plus two TC Icon hardware remotes for the two other units.

SR renovates its radio house

The renovation of the Saarländischer Rundfunk (SR) radio building has included all the studios of programmes SR1 – SR3 and youth radio Unser Ding.

All facilities have been designed identically with each consisting of two radio studios, one programme matrix plus two central news studios.

The master control room has two mutually redundant Lawo Nova73HD routers with a capacity of 3000 x 3000 mono channels.

They are controlled by two redundant line

scheduler servers with automatic replication of connects.

Studio Control Rooms are equipped with Lawo Diamond consoles with a Diamond matrix managing the lines and is also used as the programme switch. The mixing console surfaces integrate Delec T/B system operating panels. The Unser Ding studio uses Lawo Zirkons. The news area hosts two studios with Zirkon technology for central presentation of the news for every programme.

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news

news THE BIG PICTURE • ICEMOBILE’S VIDEOCALL2TV technology and formats enables viewers to make a video call from their 3G mobile phones to a TV programme and participate in the show with live video images. The VideoCall2TV technology works worldwide from any 3G device and on any 3G network and can be billed on a per-minute basis.

The technology includes the content management, monitoring and on-screen display solutions to integrate user-generated, live video content directly into the show.

‘IceMobile’s clients are media and telecom companies, such as MTV Networks, Endemol, Vodafone and T-Mobile,’ explained Ralph Cohen, CEO of IceMobile. ‘For years, these companies generated a lot of revenue with SMS to TV services, but are now looking for the next big innovation to entertain their audiences, and generate new revenue streams.’

• CINGULAR WIRELESS is adding Fujifilm’s Get The Picture Mobile service to its product set. The application enables Cingular customers to order prints directly from their handsets and pick them up at a local retailer in as little as an hour.

• DTS HAD a live IP multicast transmission presenting Swedish Radio multichannel programming at the IBC exhibition.

The programmes were fed live over the Internet from Stockholm, and replayed using a DTS developed multicast client running on a standard PC. Swedish Radio has also announced that the number of downloads from its website, www.sr.se/multikanal, has surpassed the 10 million mark, which makes it one of the world’s leading online multichannel audio destinations.

Biz bites

The UK government’s culture secretary Tessa Jowell has unveiled official plans for a complete transfer to digital TV broadcasting in the country, writes Nigel Jopson. Analogue switch-off will be phased in by ITV region between 2008-2012, with assistance for elderly and disabled Telly-watchers to acquire new sets. ‘When a new technology comes along, governments have two choices,’ announced Jowell. ‘They can follow it, trying to make retrospective sense of how society is changing as a result. Or they can be ahead of the curve, shaping the future and ensuring that the fruits of technology are evenly spread.’

Spreading the fruits a little thinner was the US government, whose Senate Commerce Committee decided to shelve legislation for a digital TV transition.

Meanwhile a NY Post report that Microsoft was about to purchase AOL from parent Time-Warner was received in a remarkably sanguine manner by US commentators.

AOL may have lost a few accounts recently, but 22 million subscribers still count for something. Bill Gates already owns My Computer, last month’s Resolution business column demonstrated he was taking over Your TV, now he’s making a bid for Our Network.

Once they fought to keep them out, now they invite ’em in: Napster announced the addition of 53 colleges to its subscription service initiative, including UCLA, Berkeley, Cal State and Brown. The campus service includes unlimited streaming and downloading from Napster’s 1.5-million song library, permanent song downloads for US$0.99 and albums from US$6.95.

An American popular music course at Penn State University has incorporated the service into curriculum offerings, and students get the service for free!

Brits be proud: researchers from Cambridge say their new computer chip design enables large amounts of data to be stored by using a complex interconnected network of nanowires, with functions performed where they meet using a similar approach to brain neurons and axons. Memory capacities could increase 200 times. Professor Cowburn and colleagues discovered it’s possible to reproduce key functions of semiconductor electronics in nanotech microchips using only the spin of electrons, responsible for magnetism, rather than the more conventional charge that traditional processors use.

Three years ago my discovery that illegal P2P music-sharing network Kazaa was holed up in Australia, rather than the Pacific island of Vanuatu they pretended to occupy, met deafening apathy from newsstand editors. Now an Australian Federal Court judge has ruled in favour of record labels, stating Kazaa managers authorised users to infringe copyright. The victory is not in the costs Kazaa must pay, or the software modifications it’s been instructed to make, but in the public demonstration that those behind P2P were not some 21st century versions of Marat, champion of the Third Estate — they were merely money-grubbing opportunists, out to turn a fast buck from music sharing by infecting PCs with prodigious quantities of spyware and advertising.

• BBC BROADCAST transmitted a specially crafted HD test channel simultaneously in Mpeg 2 and Mpeg 4 from its Broadcast Centre in West London to the RAI Exhibition Centre in Amsterdam throughout IBC. The test channel was broadcast using the Eutelsat Eurobird 1 satellite and included a selection of sport, drama, arts, nature and current affairs programmes.

• A SURVEY, published by independent think tank Pew Internet & American Life Project, found that high-speed Internet adoption in the US, after growing quickly in the past several years, has been losing ground and is poised to slow even further. It attributed the slowdown in broadband penetration to a maturing of the market.

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Riedel supplies Spain’s largest intercom

Artist S digital intercom matrices are at the heart of communications infrastructures of Swiss installations at Hallenstadion and Theater11 in Zurich, and the Paul Klee Centre (pictured) in Bern.

Spanish national broadcaster TVE has chosen Riedel’s Artist platform for the country’s largest broadcast intercom project.

The project consists of 29 Artist frames, 900 control panels, 150 Connect Duo ISDN codecs and 75 digital beltpacks.

TVE configured an integrated system that will link communications of 16 locations.

Three main centres, located in Torrespaña and Prado del Rey, Madrid, and Sant Cugat del Vallés, Barcelona, will receive Artist M frames linked by a redundant optical fibre ring. The installations feature Artist 1000 series LED control panels and Artist 2000 series LCD control panels, 4-wire interfaces, integrated ISDN codecs, AES signals, and Performer digital belt-packs.

Riedel’s trunking software, Trunk Navigator, will handle the communications between the centres.

The 13 regional centres will use Artist S frames, 2000 series LCD panels, and Connect Duo ISDN codecs to communicate with each other and the main centres. In addition, numerous remote panels with Connect Duo ISDN codecs will be added to the regional centres as needed.

Soundfirm orders three Smarts

French TV production company Visual Technology has equipped its HD1 production truck with systems from RTS Telex. The centrepiece is an ADAM with 80 x 80 ports and a total of 20 RTS Telex keypanels (4 x KP32-GR4F, 4 x KP32-16GRC4, 2 x KP632-24GR, 6 x KP12LK W/RC, 3 x MKP-4LK, 1 x UIO-256). The HD1 is a simulcast-capable, mobile control room that is equipped for the use of up to 24 HD cameras.

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Soundfirm, Australia’s largest audio postproduction company, has ordered three Smart Consoles for its facilities in Australia and Beijing. All consoles will be powered by Pyramix digital engines.

‘The addition of the Smart console, with its incredible flexibility and speed, will increase our competitiveness in all of our three facilities,’ said Soundfirm owner and CEO Roger Savage.

‘I‘ve been keeping a close watch on the progress of ARC Technology since the first concepts were shown to me in the mid-90s,’ he added. ‘The ergonomic strength of this approach really comes into focus when you’re trying to manage the multitude of tracks that can be thrown into the mix. This technology will revolutionise the future of mixing.’

October 2005

news

news THE BIG PICTURE • TDK HAS begun distributing its P ro f e s s i o n a l D i s c (PD-RE23CN) media, rewritable optical discs that provide compatibility with Sony’s Professional Disc System (XDCAM). With XDCAM, camera and media can obtain footage in challenging environments where traditional tape-based recording systems would fall short in performance and reliability, according to TDK. • SONY IS to cut 10,000 jobs, reduce costs by US$1.8bn and close 11 manufacturing sites as part of a midterm plan to bring the group back to profitability. The restructuring plan comes as the group revised its July profit forecast and said it would suffer an operating loss this year for the first time in a decade.

• SONY’S NEW handheld games and entertainment console, the PlayStation Portable, is now shipping and is a games console that can also play films and music, and can be used to browse the web wirelessly.

AC/DC rocks with Digital X Bus

Angus and Malcolm Young of AC/DC each have a Mackie Digital X Bus X.200 recording console in their personal studios in the Netherlands and London respectively. Both also have HDR24/96 hard disk recorders and HR824 monitors.

Acting on a strong recommendation from Paul ‘Pab’ Boothroyd, who is AC/ DC’s FOH engineer and audio consultant for their recording studios, Angus read every review and piece of information he could gather on the Digital X Bus. This was followed by a call to Mackie and an on-site demo.

In their previous studio configuration, they fed the output from their recordings into a Neve mixer, and then used separate Fairchild compressors to create the sound they describe as ‘that classic valve, phat thing’. After years of service their trusted compressors became unreliable, and the brothers opted to specify a Universal Audio UAD-1 DSP card for their new Mackie consoles to give them emulations of favourites like the 1176LN and their Fairchild.

• FREEVIEW NOW reaches more than 5 million UK households and 30% of all TV viewing is to multichannel services, according to Ofcom’s latest quarterly report.

explained. ‘On the next job, over a series of five separate performances, I decided to audition NT5s on all sections of the orchestra. They performed flawlessly, giving me plenty of the low end that I was after and an incredible off-axis response that I’ve previously only heard from microphones costing up to five times the price.’

• PACKET VISION claims its PV1000 could be the saviour of TV advertising because it screens different adverts to different TV viewers during the same ad break and shows commercials relevant to viewers.

It works on IPTV. TV companies can programme the PV1000 with demographic information about their subscribers.

‘Because IP is addressable, targeting can be much more granular than is possible with terrestrial, satellite or cable television — down to individual households or even specific TV sets,’ said Packet Vision MD Patrick Christian. ‘One ad slot can be used to deliver hundreds of ads with each targeting the relevant audience.’

• SAMSUNG WILL bring out a DVD machine next year capable of playing Blu-ray and HD DVD if backers of the rival standards fail to agree on a unified format, according to reports.

Samsung’s head of consumer electronics, Choi Gee-sung, told the Financial Times: ‘We would welcome a unified standard but if this doesn’t come, which looks likely, we’ll bring a unified solution to market. It won’t be simple but you’ll see our solution in the coming year. Consumers will be too confused otherwise.’

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ITV Anglia doubles up on Calrec ITV Anglia is the latest ITV News Group station to switch to digital audio with Calrec following the purchase of two 32-fader Zeta consoles. The installation brings the total number of Zeta consoles in ING (ITV News Group) facilities to eight: two Zetas have been installed at Tyne Tees with another at ITV Central and three consoles at ITV Meridian.

‘The installation at Meridian was a deciding factor on the purchase of the desks at Central, Tyne Tees and Anglia,’ said Paul Stevenson, controller of news systems projects, ITV News Group. ‘Meridian was developed as a template for the other ING new build installations. The Calrec consoles were chosen after extensive research by ITV and the consoles have proved to be an effective choice both technically and operationally.’

South Africa’s SABC has installed three 24-fader Zetas consoles at its Henley Television Facility in Johannesburg.

Supplying production, postproduction and broadcasting facilities to external clients in addition to it’s commitment to the SABC, all Henley studios are used principally for soaps and are capable of live transmissions.

UK OB specialist NEP Visions (pictured) has purchased three Calrec Alpha consoles for one new HD truck and substantial refurbishments to two existing units. The desks will be installed into Visions’ new HD3, HD4 and HD5 units and will cover live Premiership football for Sky. UK-based Telegenic has installed an Alpha console in its T8 digital OB unit as the company gears up for 5.1 mixing. T8 made headlines at its launch in 2002 as the first HD unit in the UK — the Alpha replaces the truck’s original S2 analogue desk and is Telegenic’s first digital console.

It runs two S2 series and a C2 series in three other trucks.

Metro FM relocates with Audionics routers

Sound mixer Jerry Eade has bought 22 pairs of Røde NT5 cardioid condenser mics from HHB to use on Placido Domingo concerts.

‘I was invited to try out the NT5s on a show featuring the up and coming opera performer, Anna Netrebko. Frankly, I was amazed at how good they sounded,’ he

HHB post division Scrub has installed the UK’s first Digidesign Icon D-Command in London post facility Silverglade. The facility has six offline suites and two nonlinear Avid online studios. ‘We’d run Pro Tools and a Pro Control surface in our smaller dubbing suite for two years and decided it was time to upgrade the larger one,’ said Silverglade’s head of sound Doug Robinson. ‘Looking for a product with longevity and which would integrate easily with our current studio equipment, we went for D-Command, and we’ve been delighted with its performance.’

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Audionics has continued its long association with EMAP Radio Group by supplying key equipment for the relocation of Metro Radio to a new studio complex in Newcastle from where it will broadcast FM, AM, DAB and Internet services.

Central to the operation are outside source and transmission routers supplied by Audionics. The routers have been customised to meet operational requirements and the outside source router has an embedded web server to enable control from standard web browsers — this reduces the number of hard control panels required and enables any desktop PC to control the router.

October 2005

news

gear review

Products Equipment introductions and announcements plus news on Digidesign and Steinberg.

JBL ‘INTELLIGENT’ MONITORS JBL Professional’s LSR4300 Series studio monitors have ‘network intelligence’ and a new automated version of JBL’s Room Mode Correction system.

The LSR4326P powered 6inch 2-way system and the LSR4328P powered 8-inch 2-way system are the first studio monitors to include Harman HiQnet network protocol, which allows all speakers to be centrally controlled from the mix position.

With Control Centre software the user can address system settings, create custom EQ presets, and store and recall system configurations from the computer desktop. The LSR4300 ships with a wireless remote control that allows control of all LSR4300 features.

There’s dedicated system volume control, mute and solo functions and input switching that allows the user to monitor a range of analogue and digital playback sources.

Each model includes 24-bit, 96kHz AES-EBU and SPDIF. A tri-colour meter display indicates system output and a compliment of illuminated controls provides access to all user functions including EQ, Solo, Input selection volume and RMC calibration. When networked, all speakers can be controlled from the front panel of any speaker.

LSR4328P and LSR4326P monitors are available in pair packs that include an accessory kit containing a calibration microphone, remote control, LSR4300 Control Centre software, USB and network cables. www.jblpro.com

YAMAHA M7CL DIGITAL DESK Yamaha’s M7CL digital desk sits between DM and O series consoles and the PM1D and PM5D and is targeted at the live production market.

It is available in 32 or 48-channel formats with four stereo inputs and three mini-YGDAI card slots. M7CL features include 16 mix buses, 8 Matrix buses, LCR bus, 8 DCAs and 16 omni outputs. On-board effects and graphic EQ are included and eight signal processors can be used simultaneously. Single function physical controls and all digital controls are accessed from two main display screens.

Long-throw 100mm motor faders are provided for all input channels plus stereo/mono masters, with a further eight ‘Centralogic’ faders of identical type for the central control panel. Each input fader has its own On, Cue, and Select key. Aside from input gain, pan, high-pass filter, and 4-band EQ, each channel also has two dynamics.

HS-Series reference monitors feature

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Platform news: Digidesign Mbox 2 is a ‘next-generation’ USBpowered audio/MIDI production system that builds on the performance and simplicity of the original Mbox.

Mbox 2 includes Pro Tools LE software and the Pro Tools Ignition Pack collection of plug-ins, creative applications, and additional tools (it includes more than 50 plug-ins and instruments).

Features include four simultaneous inputs with analogue and digital I-O, two analogue inputs (XLR and 1/4-inch) and two analogue 1/4-inch jack outputs, SPDIF, 48V phantom, 24-bit/48kHz audio and zero-latency monitoring.

Mbox 2 Factory offers more than £650 in plug-ins for only £60 more than Mbox 2.

Digidesign is now bundling Melodyne Essential pitch correction, vocal production and melody editing software free of charge with Pro Tools software. Melodyne Essential is a cut-down version of Melodyne Uno and works like a sample editor. www.digidesign.com

white-cone drivers and cover three models. There’s the 2-way bassreflex bi-amplified HS 50M with a 5-inch cone, the 8-inch coned 2-way bass-reflex bi-amplified HS 80M, and the 8-inch bass-reflex powered HS 10W subwoofer. www.yamahacommercialaudio.com

HHB FLASHMIC Described as the world’s first digital recording mic, the FlashMic DRM85 is intended for voice recording applications in broadcast and press and combines a Sennheiser omnidirectional condenser capsule with 1Gb of Flash recording memory.

Recording is instigated with a single button push to Wav linear or MPEG 2 files that can be transferred via USB to a computer for editing and onward transmission. A Date/Time stamp is stored along with the file, with the internal real-time clock set/synchronised automatically by the host computer.

The FlashMic is powered for more than six hours by a pair of standard AA batteries, with remaining battery power displayed along with time, level and status information in a backlit LCD on the mic body. There’s also a visual low battery warning alert.

Users can operate the FlashMic using default settings or create and store nine custom configurations using the software supplied. Parameters including Audio Mode, AGC On/Off, Record Level, Pre Record Buffer (0 - 10 seconds) and HPF On/Off can be individually adjusted. An Expert Mode allows all parameters to be accessed directly from the FlashMic body.

The Portadrive has received an upgrade with a new, higherp o w e r e d rechargeable battery and the availability of an 80Gb removable drive.

The new 71 Watt/hour NPL7 Lithium Ion rechargeable battery replaces the original 52 Watt/hour NPL50, boosting continuous operating time from two hours to around three. www.hhb.co.uk

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CEDAR FOR TOOLS The CEDAR To o l s r a n g e of restoration m o d u l e s , available as AudioSuite plug-ins on the Pro Tools (PC) platform, i n c l u d e s R e t o u c h , Declip, and Auto Dehiss.

Retouch is able to remove individual sounds and notes from complex audio and can eliminate unwanted noises such as car horns, banging doors and coughs. www.cedaraudio.com

OXFORD LIMITER

Sony Oxford’s Oxford Limiter is described as a high quality programme limiter that uses logarithmic sidechain processing with adaptive timing functionality and lookahead techniques.

An Enhance function gives volume and punch to material beyond that available from conventional limiting functions without overloading. Comprehensive metering is provided to display peak sample values and inter-sample overloads.

Sophisticated dithering functionality with selectable and variable depth noise shaping permits mastering output quality in 24-bit or 16-bit modes. www.sonyplugins.com

October 2005

gear review FAIRLIGHT I-O SX48 is a 2U rackmount that provides 48 channels of analogue or digital I-O with setup and other parameters established and controlled via Ethernet. The host connection is supported by MADI, sample rates of 44.1, 48 and 96kHz are supported and a 4-channel mic amp card can be included.

Fairlight’s MediaHub collects its studio asset management tools into one comprehensive package, allowing users to implement an effective workflow process throughout a studio. Connecting all facets of the studio environment, the MediaLink server networks multiple Fairlight workstations, recorders, editors and mixers, as well as file servers and Windows or Mac-based systems. Also included is MediaLink AV, Fairlight’s audio and video solution for networking Pyxis NLV and QDC engines using sustained bandwidth streaming technology.

AudioBase3xtreme is Fairlight’s audio library management system, Virtual Studio Runner automates file delivery to and from Dream systems, and AVTransfer is a server-based file conversion application.

Fairlight has joined the Autodesk Infrastructure Sparks programme and will develop audio media and metadata integration between Autodesk’s editing and visual effects systems and Fairlight’s nonlinear audio recording, editing and mixing systems. www.fairlightau.com

360° FOR NUENDO The Waves 360° Surround To o l s b u n d l e i s n o w available for Cubase and Nuendo native systems.

It gives nine processors designed from the ground up for surround production: Surround Panner; Surround Imager; Surround Reverb; Surround Compressor; Surround Limiter; Surround Manager; Surround Mixdown; Low-Pass Filter; and the Bit Re-quantizer. www.waves.com

RTW VECTORSCOPE RTW’s DigitalMonitor 10500 is an audio vectorscope, peak programme meter and status monitor that interfaces to professional digital production environments. The instrument is connected via XLR connectors and accepts signals up to 24-bit, 96kHz. All essential parameters and displays are permanently visible. Supplied as a tabletop unit with an adjustable stand, the 10500 can also be converted for front-panel mounting. www.rtw.de

A&H GOES DIGITAL ILIVE

Allen & Heath has entered the digital mixer domain with the iLive.

The heart of the system is the iDR-64 rackmounting modular mix engine providing 64 channels into 32 mixes, which can be assigned as auxes, groups, matrix and main outputs. The iDR-64 is a standalone mixer comprising the 8-channel audio interface cards, DSP processing, and communication ports. The unit is controlled via an Ethernet network and can be connected to a variety of controllers, including iLive control surfaces, Allen & Heath’s PL range of remote units, third party devices, or a laptop/PC. All channels and mix masters have EQ, dynamics and delay processing with on-board FX, and an option for a built-in 32-channel speaker processor. Branded plug-ins are being developed for the future.

There are three iLive control surfaces — iLive-112, iLive144, and iLive-176. The faders are motorised and grouped in three sections, each with four banks, giving up to 176 control strips on the largest surface. There is a multicolour backlit LCD display above each fader for labelling and colour-coding channel information. The channel controls for preamp, filter, gate, EQ, compressor and limiter/de-esser are laid out across the top of the surface on rotary controls with LED indicators, while a colour LCD touchscreen presents a graphical view of the processing and access to the automation and set up screens. Audio at the surface is handled by 8-channel analogue or digital I-O cards for up to 16 I-Os. www.allen-heath.com

October 2005 15

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gear review EUPHONIX UPGRADES Software and h a r d w a r e upgrades for the Euphonix Max Air (V1.4) and System 5-B (V2.8.1) on-air digital a u d i o mixing systems include live control surface module resynchronisation, configurable aux sends, Image Video TSI-1000 integration, ergonomic improvements to the surface, and faster embedded processors. Optional hardware upgrades enhance performance but are not mandatory to run the newest software.

There is now a choice on Max Air as to how many aux sends can be configured with a new option for 24 aux send buses instead of the original 16. Each of these sends can now be globally set to unity, pre/post, on/off, stereo/mono for all strips. Fader positions can be copied to an aux send and aux send knob positions may be copied to another aux send. It is now possible to control a selected aux send from the faders and the aux sends now have up to +12dB of gain to match the fader gain structure. www.euphonix.com

COLEMAN CONTROLLERS Coleman offers a range of passive analogue DAW monitoring controllers, passive switchers and analogue VU meters.

These include the TB4mkII stereo DAW monitoring controller with talkback, cue, speaker selection, engineer’s headphone out and master volume and the M3PHmkII

stereo DAW monitoring controller with 4-way balanced stereo source selection, 3-way speaker selection, engineer’s headphone out and master volume control.

All Coleman master volume controls track to within 0.05dB. www.kmraudio.com

ROLAND SNAKE The first product in new Roland division Roland System Solutions (RSS), the Digital Snake is a 40channel digital audio transmission system with remote controllable mic preamps, configurable inputs and outputs, and immunity to RF and electrical interference. The system is configurable in 8-channel modules for 32 x 8, 24 x 16 and other configurations.

Remote gain is adjustable f ro m - 6 5 t o +10dBu and includes scene recall on the 24-bit, 96kHz system. Interconnection is via Cat5 and system inputs can be split using Ethernet hubs. The Roland Ethernet Audio Communications protocol introduces only 375microseconds of latency. www.rssamerica.com

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Platform news: Steinberg The update to V3.1 of Nuendo is now available and includes features targeted at professionals in post and audio production.

The update is available for Mac and PC from www.steinberg.net ‘While Nuendo 3.1 introduces significant new features for the fastgrowing Nuendo postproduction community, it also includes an impressive list of improvements that speed up the workflow of any audio production,’ said Lars Baumann, Steinberg’s senior product manager for Nuendo. ‘The additional option to integrate Nuendo 3.1 with the new Euphonix hardware and the fact that Steinberg releases such a substantial update at no charge is a clear reflection of our strong and ongoing commitment to our customers.’

Nuendo 3.1 includes several advanced new features for postproduction in film and television.

Among them is Audio Pull up/down functionality for whenever a Telecine process or transfer from NTSC to PAL has been involved during the production. Frame rate support has been extended to include those required by HDTV. Network collaboration features have been extended by including support for Marker Tracks as well as the addition of a Transfer Status window for accurate monitoring of data transfer around the network. An OnlineMerge function enables users to merge selected tracks between local projects and available network projects without having to fully join a network session.

Nuendo 3.1 also provides the technological basis for its synergy with the Euphonix MC and System 5-MC DAW controllers via EuCon. Other new features include improved handling of video thumbnails, a safeguard against lost recordings due to power outages, and a handy ‘Enlarge Selected Track function. www.steinberg.net

NEW ATC COMPACT The ATC SMC 110A is a compact, active 3-way monitor. It has twin 9-inch ATC SL bass drivers, an ATC 3-inch soft dome mid and a soft dome 1-inch tweeter.

The active design matches six Mosfet amplifier blocks with the drivers to deliver a maximum SPL of 115dB. www.atc.gb.net

SAN UPGRADE An upgrade to the globalSAN X-24 system n o w o ff e r s 6 0 % m o re storage capacity and two additional software licenses.

The X-24 is now a 9.6Tb, eight-user SAN with dual RAID controllers and redundant power supplies. www.studionetworksolutions.com

CHARTEROAK FET CONDENSER The CharterOak E700 is a dual diaphragm FET condenser mic that employs two 1.22-inch gold sputtered Mylar diaphragms and pure Class A head amp electronics. The mic offers three selectable polar patterns and a two-step bass roll off. www.charteroakacoustics.com

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LIPSYNC LipSync1 is a stereo audio delay synchroniser with balanced analogue audio interfaces that is designed to resynchronise audio with video that has been delayed by processes such as format conversion or satellite transmission. Audio delay processing is carried out in the digital domain.

Delay settings are adjustable in 1, 10, 100 and 1000mS increments.

Input and output levels are adjustable and multiple units can be rackmounted and powered as a group. www.michael-stevens.com

DYNAUDIO COMPACT ACTIVE Dynaudio’s BM 5A Compact is an active 2-way nearfield monitor that measures 170mm x 260mm x 235mm and costs EUR 1050 (+ VAT) per pair.

It is equipped with woofers and tweeters of the same quality as in the BM 5A active monitor, which is nearly two times the size. Powered by two 50-Watt amps and claiming a frequency response of 55Hz-21kHz, the BM 5A Compact can be combined with the BM 9S sub for 5.1 monitoring. www.dynaudioacoustics.com

October 2005

gear review TWO IN ONE COMPRESSOR

Buzz Audio’s Dual Compressor System Model DCS-2.2 features two different compressor types in the one unit. It includes one channel of hard knee FET-based compression and one channel of softer opto compression. Routing switches allow the user to use the two compressors types individually, in series or in parallel. A blend control allows mixing between un-compressed and compressed signals, or blending between the two compressor types in parallel mode. Other features include a front panel switchable insert point, transformer saturation, LED metering and a sidechain frequency response tilt control. Intelligent linking to a second DCS allows for stereo operation. www.buzzaudio.com

SOUNDELUX E47C AND E251C Soundelux Microphones’ E47C tube condenser mic is said to faithfully recreate the cherished ‘cardioid only’ characteristics of the original 1950s-era German tube 47, which was typically used almost exclusively in that mode.

Like the original, the E47C displays a signature proximity effect that offers 12dB of boost at 100Hz when used at 1-inch. The E47C features a NOS Telefunken Large Plate EF814k valve, a P99E power supply, 20-foot Soundelux cable, and a 47 suspension clamp-type shockmount with a wooden box.

The E251C is a cardioid only, lower-cost version of the ELUX 251. The E251C uses the same capsule as the ELUX 251 but costs significantly less due to savings made in the design and manufacturing processes — a lower cost body assembly, no pattern selector switch, and an imported shockmount. www.transaudiogroup.com acoustic measurement software

ADAM IMPROVED On the surface little may have changed, but on a technical level the new ADAM matrices from RTS Telex have been improved and offer new features but remain compatible with the company’s existing systems.

The new ADAM employs the AIO-16 16-channel analogue I-O card, two new master controllers, two clock masters and (in the interests of maximum reliability) two specially designed power supplies.

ACCURACY THROUGH PRECISION

The ISDN 2005 Interface has two codecs that can be individually configured and used independently. The interface layout is modelled on a mobile phone for editing, storing and dialling of phone numbers. It can be configured using the supplied PC software or by using menu functions. www.eviaudio.de

} Tannoy Dual Concentric™ point source drive unit } WideBand™ SuperTweeter™ extends frequency response to over 50kHz } Digital amplification with analogue and digital inputs } Automated Activ-Assist™ software driven digital calibration

MULTIFACE IMPROVED The RME Multiface II is 100% compatible with the current model’s firmware and driver and offers the same set of features and settings and the same lowlatency, zero-CPU-load performance.

Some significant changes to the circuitry have been made: a high power headphone output with a volume knob; analogue input stages with improved CMRR, lower THD and an increase of the S/N ratio by almost 10dB; and analogue outputs with significantly lower THD. The RME Octamic’s synchronised and more effective internal switching power supply has also been added. www.rme-audio.com

October 2005 17

ACTIVE STUDIO MONITORS

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Tannoy United Kingdom

T: +44 (0) 1236 420199

F: +44 (0) 1236 428230

E: enquiries@tannoy.com

Tannoy North America

T: (519) 745 1158

F: (519) 745 2364

E: inquiries@tannoyna.com

Tannoy Deutschland

T: 0180 1111 88 1

F: 0180 1111 88 2

E: info@tannoy.com

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gear review NEVE 88D LAUNCHED AMS Neve has launched the 88D music production console. It combines a DSP engine providing 1000 tracks at 96kHz with classic Neve mic preamps, dedicated 8.1 surround monitoring, and Encore Plus automation. Encore integrates Pro Tools and Nuendo via HUI and Pyramix via Oasis protocols.

The 8816 Summing Mixer offers 16 input channels that include level, pan, cut and cue controls. A +15dB gain range accommodates -10dBv and +4dbm input levels. Multiple 8816 units can be cascaded together via a fixed level 2-track bus.

The mix outputs appear simultaneously as balanced +4dBm and unbalanced -10dBv. Mix Inserts are pre-fade and can be switched between normal insert29/6/05 mode and mastering style Res_Smart AV_07.05 10:40 am

sum and difference mode with a stereo width control. A post insert mix function can be used for mixing in a separate stem or blending the mix with a special effect.

An optional digital output converts the analogue mix to any PCM rate up to 192kHz or DSD. Output PPM meters are complemented by overload LEDs. A monitor section selects between mix out, 2-track return, channels 1 and 2 (DAW mix replay) and ‘i-mon’ front panel input jack — handy for quick mix comparisons. The monitor level is independent of the mix level and the monitor outputs can be switched between Main and Alt speakers.

A separate Cue Output can be fed from three sources and the built-in talkback mic. A USB connector allows snapshots of the 8816 setting to be saved to a PC and recalled. www.ams-neve.com Page 1

”The most exciting new product in years.”

In an industry where new product launches come thicker and faster each year, reactions like the one above are extremely hard to come by, yet this is the most frequent comment we’ve heard from seasoned professionals and hardened critics alike on first seeing the revolutionary new Smart AV Console.

Designed for use with today's most popular DAWs including Apple Logic, Digidesign Pro Tools and Merging Technologies Pyramix, the Smart Console represents a quantum leap forward in ergonomics and is radically different from any other console on the market today. Operation via patented ARC technology is so intuitive that the learning curve is measured in minutes rather than days, and project completion times are slashed. A bold claim, but when you find out more about the design, it’s easy to see why: THE ARC

EQ FAN DISPLAY

The Console’s patented touch-sensitive ARC allows instant one-touch selection of any channel or group of channels from those currently in use on the DAW - custom sets of channels can be called up onto the faders in a second or two. What’s more, the ARC also allows instant muting, soloing (or any other custom function) of any channel, even if it’s not currently active on the console.

DISPLAYS

CHANNEL DISPLAY

EDIT PANEL

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In addition to the stunning metering on the ARC itself, channels marked on the ARC in your own handwriting appear in the super high-visibility electro-fluorescent display above each fader bank - this can also optionally display your DAW track names, or both. Meanwhile in the EQ department, dualconcentric touch-sensitive EQ pots show the current gain and frequency information on hi-res LED fan displays, whilst detailed plug-in control is available on the central LCD touchscreen with its own dual-concentric touch-sensitive pots.

The touch-sensitive aux send knobs are also motorized, as are the pan knobs with their handy LED surround pan displays and a central motorized touch-sensitive surround panner is available to all channels.

MODULARITY The entire Console is highly modular, and in additon to the components already described, also comes as standard with support for 3rd party surround monitoring controllers and remote mic pre-amps, a 48, 72 or 96 channel ARC, one or two optional ‘floating’ subsidiary ARCs, and an optional Edit Panel with high quality jog wheel, 40 custom function buttons and built-in 12" hi-definition LCD screen.

SOUND INVESTMENT

MOTORIZATION

Most of all though, the Smart Console makes good commercial sense - other than the undeniable client ‘wow’ factor, tests have shown that productivity is massively increased versus any other existing console design, and operator stress levels significantly reduced - how much could this be worth to your business?

The Console can be specified with either industry standard ALPS touch-sensitive motorized faders, or optionally deluxe ultra-fast Penny & Giles models when only the best will do.

Call today to find out more about the Console and flexible finance arrangements, or to book your personal demonstration.

+44 (0) 20 7692 6611 www.smartav.net smartav@mediatools.co.uk

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TRANTEC HTX HANDHELD

Trantec’s S6000HTX handheld transmitter is designed for use exclusively with its S6000 wireless radio microphone system and its sister products — the S6001 and S6002.

The HTX is a frequency-agile handheld transmitter that boasts a brand new ergonomic design and will offer a variety of inter-changeable capsules (dynamic, condenser, omni or cardioid).

It offers operating times in excess of 10 hour’s constant use with a single AA alkaline cell, a 70MHz frequency window, and infra-red frequency and operating options set up. www.trantec.co.uk

‘GREEN’ VOICEOVER BOOTH

IAC has developed what is believed to be the first ‘green’ voiceover booth. Developed at IAC’s R&D facility in New York, post-industrial organic fibre is used as an environmentally-friendly acoustic infill within the booth’s structural panels. The infill is sourced in the UK. The booths are constructed using the company’s patented Moduline system with acoustically-rated, modular acoustic panels forming the walls, floors, and ceilings. Each panel is 100mm thick and filled with the green infill that itself delivers a range of ‘green’ benefits: the manufacturing process uses far less energy than the fibreglass insulation alternative and it has mostly recycled content; it is non-toxic, and fire-, fungi-, and corrosion resistant; and, importantly, it provides equivalent acoustic and thermal performance. www.iacl.co.uk

LIGHT SPEAKER WHD’s WLS10/2 LED has the look of a stylish wall light but also functions as a 2-way full bandwidth speaker. Units can be combined with the Sub200 for a full system.

Each speaker includes 12 red, green and blue LEDs that result in long operation, low power consumption and symmetrical illumination. Integrated RGB control electronics make colour changes possible. www.beyerdynamic.co.uk

October 2005

gear PRISMX PRO AUDITIONING

PSP DELAY AND DYNAMICS The PSP 608 is an 8tap multimode delay processor plug-in for VST/DirectX/RTAS and AudioUnit/VST/ RTAS/HTDM. Each tap has selectable feedback, stereo image and position, delay time, multimode filter, modulation, drive/tape saturation, and reverb. The feedback and drive can be placed at the beginning or end of the signal chain and the modulation section consists of an LFO and Envelope follower. It’s 24-bit, 192kHz and fully automated via MIDI. The

PSP MasterComp stereo dynamics processor is double-precision (64bit floating point) and double-sampled. It features Over-threshold compression and expansion, advanced sidechain filtering, channel linking and compression tilting capabilities, blending of processed and clean signals and a high-quality brick wall output limiter with automatic release time. www.PSPaudioware.com

Sound Of Voices has a new addition to its production workflow-related software line up. PrismX Pro allows the auditioning of Broadcast Wave and AIFF polyphonic sound files up to 10 channels wide, and up to 32-bit resolution through any cheap stereo sound card. PrismX Pro streams audio directly from any internal, external or network media so only a minimum spec PC is required. The PrismX Pro mixer features are based on auditioning tools normally found on large-format audio consoles, and offer dB display, MS decoding, pre/post fader metering, phase inversion, panning, mute and solo on each channel, as well as a comprehensive master section. PrismX Pro also reads timecode and features a ClickPlay browser for the fastest possible auditioning of files. www.soundofvoices.ie

IMPROVED VOCALIGN

VocALign Pro automatically edits one audio signal to match the timing patterns in another and has been released first as a Pro Tools AudioSuite plug-in.

New features and functions over the original program include increased alignment accuracy, user settable sync points, protected signal blocks, comprehensive audio preview options, the ability to re-use an alignment path, and improved TimeMod (time compression/expansion). www.synchroarts.com

ADL 600 SHIPS

PreSonus is now shipping the ADL 600 high-voltage class A, 2-channel valve microphone preamp. It has six tubes running on 600-volt power rails for 75dB of gain and -100dB signal to noise ratio.

Features include variable microphone input impedance, instrument input, line input, 48V phantom power, -20dB pad, phase invert, variable high pass filter, as well as analogue VU and fast-acting ten-segment LEDs for metering. www.presonus.com

October 2005 19

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19

SSL C300 At a time when more and more attention is lavished on the workstation, a ‘proper’ desk has to have a sharp line in justification to warrant consideration. ROB JAMES looks at a digital postproduction console with plenty of features and front.

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IFE USED TO BE SIMPLE for console manufacturers. Each postproduction console had its own particular strengths but nobody questioned the necessity of having one. The playing field was reasonably level, although competition was often fierce, with the worst offenders making a shoal of barracuda look positively cuddly.

In recent years the whole mid to high-end market has been skewed, partly by some extremely curious commercial practices, but mostly by the rise and rise of the workstation. In the halcyon days the console ruled the roost. Even in the early digital age the console was the master. Both in the very real context of providing audio and timecode sync for the entire system but also in a more conceptual sense.

As the industry slowly moved from linear recording to random access, nonlinear hard disk, the console continued to be the tool of choice for making the major qualitative changes. The room with the big console was still the place where adjustments to equalisation and dynamic range were made, effects added and the final mix produced.

A few, more enterprising, practitioners had long been doing their own thing and producing semipremixed effects and even dialogue sequences, often to the considerable annoyance of the postproduction dubbing (re-recording) mixers. The DAW has its feet under the table and material arrives premixed, EQed and with effects in place. Notwithstanding, the final mix console remains the place where the pigeons come home to roost and all problems must be solved.

The workstation has long promised to replace not only linear recorders and sticky tape editing but also the console at what appears at first glance to be ludicrously low cost. In recent years the DAW has largely managed to achieve the first goal. Even the ‘dedicated’ recorders of today owe more to 20

workstation technology than to any linear recorder, despite appearances.

The battle lines are drawn and it is the traditional console that is under threat. It is perfectly fair to say that a modern digital post console has much in common with a workstation equipped with a hardware control surface. Whether the processing is contained within the surface or in a remote rack or PC/Mac the principle is the same. The user interface is merely a remote control for the processing and I-O. It is also fair to say that workstations are catching up fast as the sheer horsepower of host PCs and add-on DSP cards increases in accordance with ‘Moores law’. However, conceptually and ergonomically the DAW — even with a large scale hardware control surface — has a very long way to go before it can hope to truly supplant a state-of-the-art ‘big-gun’ console in efficiency.

Semi-pro workstation sticker prices have raised unrealistic expectations among the bean-counters. For some reason they seem to have lost sight of the fact that the person with the ears is the most expensive piece of equipment in the room. Over its working life a console doesn’t have to save much production time to more than justify a significant price premium.

Against this background the SSL C300 has arrived (starting at around UK£100,000 + VAT).

Building on decades of experience SSL has leveraged the technology developed for the other C series consoles and coupled it with heavy-duty ‘Timefreeze’ automation. A single Centuri core offers up to 512 mix inputs feeding up to 80 buses. While the rest of the C series are optimised for other purposes the C300 is quite definitely a post console.

Control surface paradigms can be loosely split into two groups. The first is so-called ‘knob per function’ with recognisable channel strips bristling with knobs and buttons as far as the eye can see. The assignable alternative usually comes with a decent number of resolution

fader strips but relatively few controls per strip and a central section with lots of knobs and buttons. C300 belongs to the latter category.

Layering and bank switching are used to control a large number of channels from a much smaller number of strips. Fixed Bays always show the same sets of channels — the relationship between strip and resource is fixed and Scrolling Bays provide the moving window across all the input channels.

Channels can be scrolled channel by channel, bay by bay or three bays at once. This follows common DAW practice so it doesn’t matter if the bay is controlling C300 channels or workstation channels. Apart from input channels there are Buses, Main Outputs, 16 assignable Direct Bus Access Channels and Control Faders (VCA-style grouping). User Layers can contain a user-defined mix of signals, input channels, buses, outputs and DAW tracks.

The C300 surface is constructed from just four main building blocks: Channel bays eight-fader strips wide, Channel Fader Strips, one or more Master Channels and one, two or three Centre Sections.

Fader Strips have a touch-sensitive moving fader, an input meter, two indicators showing linking and strip format, automation mode and write status plus a Select key for DAW tracks and Solo, PFL On and Off buttons, pan pot and linking buttons. The ‘Free Controls’ section has two buttons, two knobs with switches each with a four-character display and an 8-character name label. Multiple signal paths can be formatted as stereo, LCR or many other formats up to 7.1 to ensure the panning and processing is dealt with appropriately.

C300 offers three types of grouping — for formatting a channel or bus, channel linking and control grouping. All faders in a formatted link can be ‘hidden’ under a single master (greatly increasing the number of inputs and buses directly accessible from the surface) or ‘fanned out’ to show their constituent channels at the touch of a key.

C300 also has SSL’s Theta panning, developed for the K series, which allows you to deal with sound as a variable arc that can be rotated around the central position.

A Master Channel contains all the controls for a single strip — EQ, dynamics and so on — and far more controls than could ever be conveniently accommodated in a vertical channel strip. The optional motorised joystick X-Y controller can be mapped to control any two parameters and serves as an especially intuitive method of adjusting gain and frequency. Each operator position must have at least one Master Channel and a Master Channel can be fitted in each bay if desired.

The Centre section has the transport, automation and monitoring controls, all the global console set-up and switching and something SSL calls The POD, a dedicated area that controls what appears on the faders. The largest element is a huge touch-screen used for things you set-up and leave and also to display automation tracks, etc.

Buses and master faders have a killer feature, the Spill function. One key ‘spills’ all the source channels feeding a bus onto the surface, known as a Spill Layer.

In the meter-bridge TFT screens show meters appropriate to the format of the strip, a graphic representation of the EQ curve and various other displays depending on what you are doing at the time.

Monitoring is another area where workstations often struggle. Here a 128-channel (64 dual-input channels) into 8 buses monitor matrix offers flexibility with serial record control and up to 64 Pec/Direct switches.

October 2005

review

Downstream of the monitor matrix is the output matrix, an 8 x 8 array. You can have up to four sets of monitor outputs each anything from mono to 8-channels wide.

For each set there is a set of eight fold down stores.

This allows, for example, a 5.1 mix on output 1 to be monitored in stereo on a different set of speakers with a single button push. Calibration includes pink noise generator and time delays for each channel.

A monitoring insert point between the monitor and output matrices enables a codec to be used to check how the mix will sound in the final delivery format.

A 32 x 16 programme/re-record matrix allows for multiple final mixes in various formats to be made simultaneously and any bus can be fed back into an input channel (re-assigned).

The minimum configuration is 8 faders and the maximum is 128. There is room for 8 I-O cards in any mix of types from a choice of AES-EBU, MADI and analogue line and mic options. Similarly, the back-end processor crate can have a maximum of 8 DSP cards fitted. The DSP boards, each equipped with 12 Analogue Devices Sharc processors, provide a pool of dynamically allocated processing. One chip may be

running EQ processes, another dynamics. Each chip is fed from a dual-port RAM to its own serial routing bus on the card. The channels are constructed by signals running through the various DSPs with time alignment blocks. All processing is prefader.

There is no preset assignment of aux buses. You can configure each channel with a different set of aux buses with up to 18 sends, far more than anyone is ever likely to use. They go to formatted buses so if you want 18 x 5.1 sends you can have them.

Apart from the pool, one of three Mixer Models can be applied to each board independently. Short gives 32 channels feeding all 80 buses. Tall gives 48 channels feeding a continuous block of 56 of the 80 buses and Grande gives 80 channels feeding a continuous block of 40 of the 80 buses.

The Short model gives 4-band EQ, filters and dynamics on each channel — the others are less generous but still impressive. Cards can be reconfigured without a reboot and processing is allocated as it is required. For example, if an EQ is in bypass it isn’t consuming DSP and switching a filter from 12dB/ octave to 24dB/octave seamlessly adds the necessary extra processing. In addition to the total of 80 programme buses there are a further 8 as a permanent AFL/Listen in-place monitoring facility.

All this is calculated to give the maximum bang for the buck with a smaller, cheaper console fully capable of handling large productions. Multi-operator is part of the basic system — just add the control surface parts. This is not then a fixed configuration as a single operator can control the whole console when required.

Automation is key. The C300 automation comes out of a truly non-linear environment harking back to

the SSL Scenaria. Although it appears to be timecode based it is in reality based on a sample clock and is non-linear in the sense that it does not require playspeed timecode to operate. Automation can be read and written at any forward speed including stop.

The design eschews touch sense rotary controls in favour of good old buttons (in the knobs). This makes it much simpler to set up, say, an EQ and punch it in wherever you wish, even with the transport stationery if desired. You can pull up a frame in the project, adjust the controls to where you want them to be, hit Play and be recording automation from that point. Tri-coloured indicators show Read, Write and Preview. There are some unique automation modes. For example, if you rewind while in Write on a section that hasn’t been written before then when you arrive at the point where writing stopped, the control(s) automatically drop back into Write. The Join key (depending on the current mode) lets you adjust settings on the fly for controls in Write then roll back and drop the latest settings into the mix at the point where you hit the key.

The computers controlling all of this are proprietary SSL designs with their own operating system, so no overlay of PC or Mac baggage to get in the way.

Nowadays, if they want to be taken seriously, post consoles must support the enfant terrible DAW with more than just a nod. The closest thing we have to a lingua franca between disparate controllers and workstations is the Mackie HUI protocol, which uses MIDI. Unfortunately HUI is rather restricted. The C300 gets around this to some extent by providing multiple HUI ports to control up to three DAWs, each with 24 tracks, but the C300 Master Strip doesn’t control anything on the DAW. According to SSL design guru Chris Jenkins, the HUI protocol is best left alone. If you mess with it, you’ll find yourself redoing it when the DAW manufacturer comes out with a software revision.

The assignable paradigm poses many challenges for designers. Although the basic idea works well enough with simple set-ups and a small number of inputs and outputs, scaling to a few hundred inputs can make it difficult or impossible to keep track of what is going on. Automation can also be tricky unless the designer thinks laterally. Here in the C300 most of the obstacles have been elegantly overcome, although I have no doubt that, just as with any other console of this scale and power, there will be some initial head-scratching.

This is a ‘proper’ console. It hits all the right spots.

Ergonomics, impeccable sound and very powerful automation combine to make this a highly productive and cost effective tool. There remains a problem. No matter how worthy the product, the real issue facing manufacturers of proper consoles is how best to accommodate DAWs. The C300 is a postproduction console for the age.

It offers significant and solid horsepower with finely honed automation and an ingenious, intuitive and ergonomic user interface. ■

PROS

Excellent automation; sheer horsepower; scaleability.

CONS

Needs more comprehensive DAW control; not much.

Contact SOLID STATE LOGIC, UK: Website: www.solid-state-logic.com

October 2005

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21

review

Universal Audio LA-3A Is nothing sacred? Thankfully not, as UA continues to plunder its historical back catalogue of desirable and elusive outboard. Throw GEORGE SHILLING into a room with a classic audio leveler and we have the recipe for a little light entertainment. (I think I’m going to enjoy this. Ed)

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INCE THE RATHER SPLENDID 1176 and LA-2A reissues, UA received increasing requests for an LA-3A. This model is perhaps less celebrated than the former two, but mostly because of its diminutive half-rack 2U size, its unassuming appearance and its lack of glamorous valves. Designed in the late 1960s this was sold as a cheaper solid-state equivalent of the LA-2A but used the same T4 optical cell for its gain reduction circuitry. In the LA-3A, the T4 sits inside a smaller casing and the LA-3A uses different input and output transformers. It also borrows some components from Putnam’s 1176. Although there was a Teletronixbranded original, it is the more commonly found Urei version that this reissue is based upon.

The LA-3A was designed by Brad Plunkett, who employed reissue designer Dennis Fink as a design engineer in the 1970s. As we have come to expect, UA’s efforts to recreate the original were incredibly painstaking. The tracks on the bottom of the circuit board look like a surrealist join-the-dots line drawing, rather than the straight lines and angles that derive from modern CAD programs. But rather than tidying this up, the UA team, led by Dennis Fink, effectively traced the original pattern.

Some parts were successfully sourced, others like the transformers were commissioned using examples from original units as a guide. They even used the same internal cable and wire colours.

The LA-3A (UK£1049 + VAT) is shipped with a pair of rack ears included, one of which is effectively a half-U width blanking plate that can be mounted either side. For side-by-side mounting of a pair of units, you need to purchase a separately available dual mounting kit.

Build quality is the usual exemplary UA standard, with solid construction that looks as well assembled as a Victorian steam locomotive.

22

The front panel really could not be simpler. It has a Power toggle, a meter Gain Reduction/Output Level toggle, an excellent central meter that illuminates when power is on, and knobs for Peak Reduction and Gain. The rear panel is a little more involved.

All the original features are present, but this is where three important modern additions are implemented.

Like the original, there is a power transformer mounted outside the back of the box — modern units tend to put these inside, along an external cable, or in a wall-wart. But apart from looking a bit odd, I can’t see a problem with this. Next to this is a 115/230V switch, a fuseholder, and the first bow to modernity in the shape of an IEC mains socket — pretty much a legal requirement. At the other end is the old-fashioned tag strip connector, where bare wires can be attached for signal. This is accompanied by another upgrade — modern XLR input and output connectors. However, the terminal strip is still needed for stereo linking, and provides a means for attaching a second output in addition to the XLR. To this end, a 600ohm resistor is attached across the output terminals.

Also on the rear are a selection of adjustments.

Most important is the Limit/Compress toggle, and although most will leave this in Compress mode, I disapprove of useful controls being hidden on the rear of rackmounting units. Limit mode increases the compression ratio when pushed hard, which is sometimes better when taming wayward bass players, but the essential character of the LA-3A remains the same. However, I have to admit the other two toggles are definitely better here on the rear. That is because they have the potential to throw the output gain up by 20dB which is not something you want to inadvertently flip when the singer is wearing cans (…well… I suppose not. Ed). The Mod/Normal toggle controls a circuit that was not resolution

part of the original LA-3A spec, but which was a very commonly applied tweak. This sets the gain structure differently to achieve a significantly better signal to noise ratio, while still providing plenty of output gain, as the LA-3A has about 50dB gain to start with. The accompanying gain toggle is a 20dB input level pad, handy when not using the Mod. But the only situation I can imagine switching the Mod off in is when pairing with an original un-Modded unit. A screw-pot balances the gain reduction when pairing units and another screwpot sets high frequency sensitivity. This makes the compressor more sensitive to high and high-mid frequencies, a feature originally designed to prevent over-modulation when broadcasting. Both of these screw-pot adjusters are described in the manual as operating in the opposite direction to the panel legending, which appears to be correct.

In use, the sound is big and warm, without the valve richness that you benefit from with an LA-2A. Slightly cleaner and more natural than its sometimes larger-than-life big brother, there is nevertheless a subtle glow that helps most signals, and the twostage release works smoothly and invisibly. It is beautifully unobtrusive when taming vocals and controls bass guitar naturally. I used it for gently levelling female operatic vocals, where it was ideal as the artist and producer professed a dislike of ‘compressed’ sound.

I was intrigued to see if the HF Contour could be set to effect de-essing. With my opera singers, there was a little sibilance reduction, but more usefully, setting the pot fully counter-clockwise turned out to be an effective way of reducing inhaling breath noise and gasps between phrases.

No matter how hard you push it, the sound remains remarkably clean, and you feel the LA-3A is reassuringly in control of the situation at all times.

I was lucky enough to be able to compare the unit to Livingston Studios’ (London) original Urei model.

After I’d tweaked the old one’s HF Contour pot to flat they were virtually indistinguishable, although, when pushed hard, the new one retained slightly more transient clarity — probably more down to the age of the components than anything else. ■

PROS

A lovingly re-created classic; simple operation; high quality ‘invisible’ compression; excellent build.

CONS

Incorrect manual description of screwpot direction; hard to access Limit/ Compress toggle when rackmounted.

Contact UNIVERSAL AUDIO, US: Website: www.uaudio.com UK, SCV London: +44 208 418 1470

October 2005

The mc266 with 512 DSP channels The mc266‘s excellence is derived from its initial conceptual design: one of the largest audio matrices, the highest quality signal processing in every channel and sophisticated redundancy arrangements, from microphone input to program output. The compact DSP core with 512 DSP channels underlines the enormous capability of the mc266.

The control surface with its graphical support makes it the ideal tool for ambitious productions.

Fast access Flexibility of external control, sophisticated audio-follow-video and sequence automation guarantees the flexibility needed for live situations.

New: dynamic automation, networking of routing and mixing consoles, workstation remote control

BROADCAST PRODUCTION RADIO ROUTING OB VAN SERVICE

One generation ahead

AES New York, October 7th –10th, 2005, Stand 964

Headquarters: Lawo AG · Rastatt/Germany · +49 7222 1002-0 · www.lawo.de Lawo North America Corp. · Canada · +1 (416) 875 3355 · www.lawo.ca

review

Buzz Audio ARC 1.1 Just when you thought that all bases had been covered as far as the outboard channel strip was concerned, along comes Buzz and it’s built an ARC. GEORGE SHILLING goes down under with this analogue recording channel.

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’DAY. MY ONLY previous encounter with New Zealand’s Buzz Audio was an excellent sounding optical compressor with big knobs and minimal controls. So I was quite surprised when I first took a gawk — there are 18 rotary controls and 23 toggles crammed on the 2U front panel.

The rear is no simpler, with nine XLR connections and two jack sockets — and yet this is a mono unit. An all-encompassing voice channel, the ARC includes elements from previous Buzz units, along with features said to be inspired by suggestions from ‘audio professionals from around the globe’. This ambitious company has quietly developed a very fine reputation. The ARC, like other Buzz products, is well-built and assembled using quality components — there are no surface-mount chips in here, mate.

The unit comprises a mic/line preamp, EQ and compressor and each section’s input and output may be accessed separately from the rear panel. I therefore first tested the preamp in isolation. The mic amp claims exceptionally low noise and with no pad sounds consistent across the gain range. Instead there is a 15dB boost switch for use when recording distant conversations of neighbouring sheep farmers using a ribbon mic. Beaut. With a large condenser, the mic preamp’s character is neutral and sounded biggest with the Load knob fully clockwise, although the effect of this knob was subtle. Even at this setting, it lacked the warming enhancement I get from my own Neve-style preamp but clarity was supreme. The phantom power circuit features ‘soft stop/start’ — a nice touch. With a ribbon mic the Load knob dramatically increased the level output when turned fully to the right, the extra 15dB gain switch was useful, although there still seemed to be less gain than was available from my own preamp at full tilt. But the ARC’s mic preamp sounds great with exceptional detail.

So on to the rest of the ARC, and here some time is necessary for familiarisation due to the rather complex panel layout. Some study of the manual is required and thankfully this is comprehensive and good-humoured without being patronising or irritating (which is probably more than can be said for this review!)

Following the mic preamp section is a line input section with a separate Gain knob, a 10dB pad to prevent input overload, and selection between the balanced rear panel input and the front panel mono jack, which is high impedance for electric guitar and bass. There is plenty of gain here for the weediest of Stratocasters and my dodgy Squier sounded clear as a bell, particularly enhanced when flipping in the Tranny circuit in the following Output section, where a 24

deliberately crunchy transformer is available for subtle colouration of the low frequencies. This also added back the warmth I was missing from the mic preamp.

The output section includes an Output Gain and a Polarity switch, although I might have wanted that on the mic pre section if using it in isolation.

The EQ consists of a High Pass filter, High and Low Shelf sections and two Mid Bands. The panel layout is somewhat jumbled. Each band includes a threeposition toggle, where In enables the band on the main signal path, Ext switches it to the separate EQ In/Out on the rear panel, and SC sends it to the compressor as a sidechain shaper. There is no ‘Out’ setting, but by sending it to the sidechain and setting the Cut/Boost knob to the centre détente the band is effectively defeated without degrading any audio signal.

The Filter is 12dB/octave continuously variable from 25 to 450Hz and the High and Low Shelf sections claim 17dB cut/boost. The High Shelf features a selector labelled Tight or Broad that affects the steepness of the curve, which is set around 12kHz. On the Broad setting the slope is very gentle and affects mid frequencies also while in Tight mode it could be used as a 12kHz low pass filter. The Low Shelf features a fairly shallow shelf with selectable frequencies of 60Hz or 120Hz with an inductor used for a tighter bass sound. The Mid Bands feature huge continuously variable frequency bands by virtue of x10 toggles and are fully parametric, with

bandwidth adjustable down to wah-wah pedal values.

Interestingly, these bands are designed based on Steve Dove’s CAPS network, proposed in Resolution’s spiritual ancestor Studio Sound back in 1981.

The Compressor uses an opto-sensor with a softknee circuit and switchable settings for Attack, Release (both including Auto settings) and Ratio, plus variable Drive, which acts like a Threshold control. This is a very versatile and musical sounding compressor that can be routed to pre or post the EQ or switched out of the main path onto its own dedicated rear panel connectors.

When set with a fast attack, the compressor can seem fairly invisible, when set slower it sounds appealingly punchy. The Auto release setting generally seems fairly fast, the whole character of the compressor is very fluid, with none of the syrupy pumping sometimes associated with optical or vintage units. There are plenty of bright LEDs on the gain reduction meter for a good indication of what is occurring, alongside a Level meter for input or output metering, or Off if the flashing lights are too much for you. Thankfully, the Over LED remains active even in this case.

Finally, a Limiter section with three different Release settings is available for taming peaks, this uses a FET and sounds great.

The ARC (UK£1595 + VAT) takes time to understand but it is undoubtedly worth persisting with as the rewards are a very powerful and comprehensive set of tools with excellent sound quality. You’ll be dead chuffed with one of these. Try it before you suck the kumura. ■

Contact BUZZ AUDIO, NEW ZEALAND: Website: www.buzzaudio.com UK, ASAP: +44 207 231 9661

PROS

High sound quality; clear, detailed mic preamp; lovely compressor; powerful EQ; clever and thoughtful circuitry design.

CONS

Front panel layout slightly jumbled.

EXTRAS

The new Buzz Audio Resonance Equaliser gets its name from the use of switched inductors and capacitors as the reactive elements in the EQ stages.

With true Class A discrete transistor amplifiers throughout and individual discrete power supply regulators for each stage, the unit boasts a variable saturation circuit that introduces varying amounts of transformer colour.

resolution

October 2005

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review

DPA 4006 TL The brand responsible for the widescale acceptance of omnis as workable studio mics has taken its original superperformer and hot-rodded it to an improved spec. There are also a variety of options available. JON THORNTON gets stuck in to rediscovering a favourite.

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PA’S 4006 MICROPHONE will be familiar to many and has attained a deservedly classic status in recording applications where honesty and transparency are the main priorities. Although the 4006 has been around in its classic form since 1982, DPA has recently released a revised version featuring an electronic output stage rather than the transformer previously employed. Claiming significant improvements in low frequency extension and sensitivity as a result, it’s a good enough excuse to put this versatile performer through its paces once more.

The kit supplied for review was a complete stereo kit, comprising a pair of matched microphones, microphone clips, a stereo bar and a complete set of grids, nosecones and ball-shaped Acoustic Pressure Equalisers that allow the response of the microphone to be tailored to fit different applications. All of this is supplied in a purposeful looking Samsonite briefcase with a secure place for everything in the custom foam inlay.

The capsule and acoustic design of the 4006TL remains the same as the original — a small diaphragm pressure transducer with a pre-polarised backplate. As standard, the microphone is fitted with a nearfield protection grid, which gives a neutral and linear response to on-axis sound sources, and in this configuration two microphones were initially mounted as a near-coincident pair over a drum kit.

I challenge anyone not to be impressed by this microphone’s initial performance, with a focus, accuracy and openness of sound that makes playback a truly three-dimensional experience. It’s not a microphone that will flatter or one that will cover up any shortcomings in the source — but in this configuration it’s as close to the truth as you are likely to ever want. This is helped somewhat by the natural tendency of a small diaphragm pressure transducer such as this to show significant off-axis attenuation at higher frequencies, which aids in focussing more on direct sound with the diffuse sound-field being partially rejected. In applications where ambient sound is more important, the standard grid can be unscrewed and replaced by a nose 26

cone that counteracts this directionality and gives a more-or-less perfect omnidirectional response at all frequencies.

In addition to this, two other grids, supplied with the stereo kit, can be fitted to the microphone. The first is a close-miking grid, which results in a much earlier rolloff of high frequencies around the 15kHz mark. This is a useful addition as moving to an acoustic guitar with the standard nearfield grid tends to give the impression that the microphone always wants a bit more distance from the source. It’s not that it doesn’t remain accurate but that it struggles to deliver a musically pleasing response. Switching the grids makes an immediate

and useful difference in this regard, rounding out the sound without making it overly dull. A final grid tailors the response of the microphone for working at longer distances by creating a linear diffuse field response up to 15kHz, but creating a 6dB boost at 15kHz in the on-axis response.

Not content with that, and realising that a perfectly flat response might not be the ideal for all applications, DPA’s acoustic pressure equalisers can also be attached and all 3 sizes (30mm, 40mm and 50mm) are supplied with the kit. These are ball-shaped devices that slip over the front of the microphone so that the tip of the grid is flush with the top of the sphere. They all, to different degrees, modify the on-axis response to include a presence peak, while simultaneously increasing the rejection of these frequencies off-axis, thus focussing the microphone more on the direct sound than reflected sound.

These work extremely well, and being purely acoustic devices, achieve an exceptionally smooth tonal shift. The smallest of the three, particularly, was of great benefit in pulling a little more clarity and definition out of a female vocal — and this is an application where I would have instinctively reached for a large diaphragm cardioid. I found myself pleasantly surprised with the results. In effect, the 4006 with all of the accessories gives you seven different microphones, which starts to make it look almost cost-effective.

But the real question is how the new transformerless version compares with the original and fortunately I was able to compare them directly. My original versions aren’t exactly shabby performers but the 4006-TL had a definite sense of being more open — particularly in the sense of solidity in the lowfrequency range. And while I might have been imagining it, they also seemed to demonstrate a touch more bite and attack on stringed instruments and percussion. Not radical differences, but ones that are positive enough for me to prefer the new version over the old. This surprised me, as I had always actually preferred the sound of the old 414B-ULS to the transformerless version, for example.

The good news if you are an existing 4006 owner is that DPA is offering a factory based retrofit to upgrade your microphone to TL spec. And if you aren’t an existing owner, you owe it yourself and your recordings to give this microphone a try. ■

Contact DPA MICROPHONES, DENMARK: Website: www.dpamicrophones.com

PROS

Detail, clarity and transparency of sound; flexibility in conjunction with APEs and alternate grids; stereo kit is superb.

CONS

Not cheap — especially in stereo kit form; some might find the truth a bit too revealing in some closemiked applications.

EXTRAS

4006 owners can upgrade their mics to the 4006-TL for Euro 415. This covers factory retrofitting of a transformerless preamplifier, close-miking grid, DPA 4006-TL type ring, calibration chart and new mic case.

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October 2005

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review

AEQ DR-100 Field recorders have got smaller and smarter just as mobile phones did before them. NEIL HILLMAN can’t believe that he can’t shout ‘I said I’m just getting on the train!’ into this new offering from Spanish manufacturer AEQ.

I

T’S BEEN REMARKABLE to watch the miniaturisation of solid-state field recorders. From the first big models such as the Sonifex Courier, Mandozzi DART or Nagra Ares-C, their progress towards our palm-top has mirrored the evolution of the mobile telephone. Once just a device enabling you to bellow your moment-by-moment homeward transport arrangements to your other half — and to the rest of the passengers — mobile phones are now so advanced that you can routinely listen to FM radio, take pictures, receive e-mails, txt msgs 2 yr m8s, and would you believe it, even converse with other humans in real-time using well-proven voice technology.

The DR-100 (UK£365+ VAT) is claimed by its manufacturers AEQ to be the world’s smallest professional digital recorder. The size of a cellular phone, it includes a built-in microphone, an audio editor and an FM radio tuner, and stores high quality audio either to its internal Flash memory or to a removable SmartMedia Flash card. The recorded audio files may be edited on-board, or easily transferred from its internal memory to a PC for editing within another audio program.

The device itself is pleasantly uncluttered to the eye. Taken out of the supplied protective leather cover, its plastic case is conventionally and conservatively finished in silver and gunmetal grey and appears suitably robust for the job in hand. The unit is light, too — just 98g all up, including the 3.6V, 700mA Li-Ion battery that provides up to 10 hours of duty between charges.

The top face of the DR-100 is the busiest, housing a monotone LCD used for navigating through the recorder, editor, player, FM tuner, timer record, USB transfer, file transfer and setup functions; their associated menu options accessed by a centrally located Menu button surrounded by four menu option buttons. These are used for parameter adjustments when the device is in any mode other than record.

Two buttons, OK and Cancel, set immediately below the display screen, are used for executing menu commands. The last controls on this top side are three narrow-ellipse buttons, semi-recessed on the bottom third of the top face and used for making A-B play point/Select, Pause/Cut, or Play/Stop choices.

The left hand side of this slim unit has a sliding Record Lock switch and a Record Ready pushbutton at the top; two buttons for Volume up and down in the centre and a master Off, On and Key Lock at the bottom. With all of the DR-100’s controls either thumb or index finger operated while the device is in hand-held mode, the texting generation of journalists

are ideally prepared for this recorder.

The right hand side accommodates a mini-FireWire USB socket and two 3.5mm sockets for Line In and Line Out. The top edge houses the tiny built-in microphone perforations, and two further 3.5mm sockets: one for an external mono microphone (stereo signals can only be recorded via the Line In socket) and one for the stereo headphones (the lead of which acts as the antenna for the programmable FM tuner). The bottom edge houses the Nokia battery charger socket and the Remote file transfer RS232 modem socket.

The review version of DR-100 recorded audio using MPEG 1 Layer II, MPEG 1 Layer III (MP3) or G 723 although the imminent production versions will also offer 48KHz uncompressed recording.

The internal Flash memory allows for 4.5 hours of material to be recorded using G 723, or 1.5 hours using MPEG layer II. Only Layer II files can be switched between mono and stereo recordings, or edited with the on-board editor; which in itself is no real hardship because the in-built, non-destructive editing capabilities are very basic: namely Cut, Delete or Merge. Even with the aid of some zooming of the pixels in the display screen, sensible editing would realistically only take place after transferring the recordings from the device to a host PC. This, as it happens was a reasonably straightforward experience using the supplied transfer software and the USB port. Even better news though is that the newest production version will sport another significant improvement — the DR-100 will be detected automatically by Windows XP as a removable drive, bringing into play the increased convenience of simply dropping-and-dragging files for editing.

Remote File Transfer is also possible through the RS 232 port of the DR-100 and an external modem; allowing a point-to-point connection on any standard POTS (Plain Old Telephone Service) line using the low-bandwidth, twisted-pair wiring and associated equipment of the local telephone exchange that provides voice calls and low-speed modem connections. Should a break in the data be detected,

the transfer will automatically resume from the point of interruption; a modern variance of the old radiohacks trick of croc-clips on the phone-line.

So is it a serious field recorder? The AEQ DR-100 certainly takes its job seriously enough — the integrity of its recorded data is beyond question — but can it be taken seriously by its intended market? Primarily aimed at radio journalists for location acquisition, its over-sensitivity to handling noise in mic mode means that it performs best when fed with an external line source; which therefore requires daisy-chaining in another piece of kit even if that is only an in-line preamp for a decent mic. That could be seen as undoing the convenience of it being a small selfcontained package.

We thought it was amazing when mobile phones shrank in size and started to take pictures; I’m just amazed that I can’t make a phone call on this one. ■

PROS

Its quirky way of working may well give rise to a new breed of operator — the texting technician.

CONS

Beware of finger and thumb repetitive strain injury, or eye strain from attempting to edit on a tiny screen.

Contact AEQ, SPAIN: Website: www.aeq.es

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resolution

October 2005

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review

Focusrite OctoPre LE Take a unit that helped to define a genre and then strip it back and make it cheaper — is that ‘Lite’ or a different box altogether?

JON THORNTON wrestles with another OctoPre.

I

’M NOT A GREAT FAN of the ‘LE’ tag, as it implies that somehow you’re being shortchanged, or at very least compromised on features and performance in favour of price. The original OctoPre (UK£552 +VAT) is a smart little unit, featuring 8 channels of mic preamp each with a clever and simple to use dynamics processor to aid the process of tracking to DAWs. The OctoPre LE (UK£297 +VAT) is also, fundamentally, an 8channel mic preamp — so what has changed to earn it that LE designation?

Starting with the front panel, the most noticeable difference is the absence of any form of dynamics processing — this is a preamp pure and simple. Each of the eight channels has a rotary gain control, a switchable high pass filter (12dB/octave at 120Hz) and a single LED indicating signal clipping. The first two channels have the ability to switch their input impedance to a lower level of 150ohms rather than the standard 2.5kohms, and the first input channel alone has the addition of a phase reverse switch.

Phantom power is available but with a single switch applying this globally to all channels rather than individually per channel.

This lightening of the feature set on most of the channels clears a lot of front panel real estate, and this has been used for 8 line level inputs on 1/4-inch TRS jack sockets. Plugging anything into these inputs automatically overrides the XLR mic inputs on the rear panel and alters the gain range of the amplifier accordingly. The first two channels are also capable of accepting a high impedance source from an electric instrument via these inputs, selected via a switch on the channel in question, which makes for quick and easy DIing.

The rear panel houses 8 XLR sockets for mic level inputs, and balanced line level outputs on 1/4-inch TRS Jacks. An external power supply feeds the unit via a 3-pin connector. In the current day and age a unit like this really can’t survive without some form of digital interfacing, and this is offered as an option.

Currently the only option is a card providing ADAT lightpipe interfacing at sample rates of 44.1kHz or 48kHz. Front panel controls allow the selection of sample rates, or the selection of an external TTL Word clock or 256x Superclock. Word clock output 30

is also available on the rear panel if the OctoPre LE needs to be configured as the master clock source.

Interestingly, the digital card has a lightpipe input as well as an output — in other words the card has A-DC and D-AC capability, which is a useful bonus. Not only does this allow the unit to synchronise to an embedded ADAT clock, but the analogue outputs on the back of the unit can be switched to output the 8 channels of ADAT format input, rather than the output of the preamps.

This means, for example, that the OctoPre LE can be providing eight channels of input to a DAW digitally, while simultaneously providing up to 7.1 monitoring from the DAW. It’s obvious so far that the OctoPre LE has undergone a significant feature cull when compared to the original. Fortunately, this doesn’t seem to have extended to the quality of the preamps themselves, which are based on the designs of Focusrite’s Green range. There are no surprises here, as on a variety of different microphones the unit proved capable of resolving good levels of detail, with the kind of transparency and openness that most people will associate with the Focusrite brand. Not a device that is gong to add a distinctive sonic character to a sound, it nevertheless extracts a useful amount of information from most sources with no fuss and little in the way of unwanted noise.

A subject I haven’t touched on yet is metering, as this is one aspect of the unit I found a little strange.

As mentioned earlier, each channel has a single LED indicating clipping, set at +22dBu, which corresponds to 0dBFS at the A-DC stage. The unit also has a circular mechanical meter, calibrated in dBFS, with

an associated switch that allows it to toggle through each of the eight channels to show peak signal level for that channel. The clip LED on the channel being metered illuminates to show which one is currently selected. Although the ballistics of the meter seem reasonably fast, and the meter undoubtedly shows the ballpark that a signal is in, I can’t help thinking that I would have been more comfortable with even a relatively coarsely stepped LED bargraph in this particular application.

Overall, the OctoPre LE is most definitely a ‘Lite’ version of the original — to the extent that it’s almost a different beast entirely. Yes, corners have been cut and features omitted, largely in the interest of saving costs. This has been done with obvious thought though, and I can see the target market that Focusrite has for this box — the front panel inputs, clever D-AC and A-DC capability, and the implication that some functions are only ever going to be used on one channel at a time — these all point to a home in any small project studio. The preamps themselves are certainly up to the job, and you get a lot of them for your money. But in other applications, for example live recording which the original OctoPre excels in, you’ll need to decide whether you can live with the limitations of global phantom power, only one phase reverse switch, and a 48kHz limit on the A-DC. ■

Contact FOCUSRITE, UK: Website: www.focusrite.com

PROS

Cheap; preamps faithful to Focusrite values; useful D-AC capability.

CONS

Metering; not all features available on every channel; 48kHz limit on A-DC option.

EXTRAS The original OctoPre provides 8 channels of Focusrite Class A mic pre and compression routing directly to a DAW via eight analogue outputs or a choice of optional convertors. Each channel’s compressor/ limiter circuit has a ‘warm-sounding’ compressor that ‘morphs’ into a brick wall limiter.

resolution

October 2005

review

Empirical Labs Lil FrEQ Dave Derr’s company is not exactly prolific, but with his previous two units, the Distressor and the Fatso, he has won many fans, including GEORGE SHILLING who bought a Fatso after reviewing it. Our expectations are naturally high for this 1U mono EQ unit.

O

NE OF MY FAVOURITE THINGS about the Fatso was its entertaining manual. Yet astonishingly EL has not yet finished one for this unit, although I was allowed a sneak preview and I can confirm that it will be full of useful information when complete. It’s good to see the familiar Input and Output gain knobs book-ending the front panel — these are very accurate and easily recallable with great accuracy. These pots were a bit cranky on my review unit but I was looking at a slightly battered example that I gather was the first one acquired by the UK distributor, so it’s forgivable.

The Input level is accompanied by an LED marked ‘Bad! Clip’ to indicate 1dB below where real nasties happen. In between are a number of smaller knobs and pushbuttons for the EQ sections. There are no less than eight EQ functions with seven separately bypassable sections here. Each of these sections features an In button with accompanying LED. At the far right is a Bypass All button, with a red LED that lights when Bypass operates, although slightly confusingly this happens when the button is out rather than in, so really it is another In button. The rear features XLR or jack input, two different output XLRs (more of which below) plus a jack output. On the front is a switchable Instrument input socket.

The first section is the Hi-Pass filter. This is a ‘coloured’ filter that seems to add a boost just above the corner frequency, from where it rolls off steeply.

The frequency is chosen with nudge buttons, with eight settings from 30Hz up to 330Hz. The LEDs glow dimly when the section is bypassed, or brightly if switched in. Even at the 30Hz setting there is some obvious enhancement of the bottom end in the mix when this is switched in (although I had to use just one side of the mix, having just one unit).

The second section features simple High and Low Shelf Cut/Boost knobs, simple hi-fi style tone controls, with a range of +/-10dB. Across a mix or most individual signals, these are sheer temptation as the added top sounds very sweet. They are very broad and the added bottom end is hugely satisfying.

They do turn anti-clockwise to cut as well, if you so desire!

The following four sections are parametric. They each feature +/-14dB cut/boost knobs at the top, a bandwidth control labelled 0.1 to 2 octaves, and frequency knobs. At halfway, the bandwidth setting of 0.7 octaves sounds as narrow as many other EQs are when set at their narrowest. There are no centre 32

détentes for the gain knobs, but with individual In buttons these are not necessary. There is plenty of overlap in the band frequency ranges: the Low band runs from 40Hz to 600Hz, the Low Mid from 100Hz to 2.7kHz, the High Mid from 400Hz to 10kHz and the High from 830Hz to 20kHz. Boost does not diminish as the bandwidth is broadened, so a huge amount of level boost is possible with wider settings but the Clip LED will warn when danger is present even if a section is not switched in. The parametric band characters are juicy, warm, open and musical.

The designer describes them as ‘surgical’, and while this is a good description, they are more fun to tinker with than that word suggests. They claim incredibly low distortion figures, and these are believable, with a sweetness to match the GML 8200 yet an even more straightforward and open character.

The final section is labelled DS, this is a dynamic EQ for de-essing or high frequency softening. A fourLED meter indicates the amount of reduction and two knobs set Threshold and Frequency respectively. A button selects HF or wideband mode. The latter is rather potent, and I sometimes found the release to be audible, but in HF mode things seem gentler with a softer knee. The LED meter remains active even when the section’s In button is out, which is either very useful or potentially embarrassing/annoying depending on your preference.

The manual suggests experimenting with this section to soften a track or emulate analogue tape, and it is certainly worth a try if your signal is thin or harsh. Amazingly this section seems to operate independently of the input level, so de-essing remains constant once set. Clever.

While the overall level knobs are easily recallable, the same cannot be said for all of the aluminium knobs, as the black banded ones — such as the Shelf and Bandwidth — do not feature any marking on the barrel.

The Class A transformer output can add some oldstyle warmth that is especially rich when pushed.

This can sound quite different from the DC-coupled ‘clean’ output so it is worth wiring both up. On the front, an Instrument input turns the unit into an exceptionally good DI box that claims incredible figures for frequency response and noise and sounds wonderfully present.

The still incomplete manual is excellent and full of titbits of wisdom, and even a terrific page-long bulleted list of why EL is such an environmentally friendly company. But like a proper bunch of hippies, after many months they still haven’t finished writing it! (Hey, relax man. Ed).

Like the Distressor and the Fatso before it, the Lil FrEQ works hard to be the best at what it does and to add features that are genuinely useful to the hard-working engineer. Again, Derr has designed something rather special, worth every penny of the rather steep asking price (UK £1385 + VAT). The only problem is that, as with the Distressor, you will want two of these. ■

Contact EMPIRICAL LABS, US: Website: www.empiricallabs.com UK, Unity Audio: +44 1440 785843

PROS

Cleanest EQ imaginable; surgical and powerful yet joyous to use; you won’t run out of bands or boost; great DI; set-and-forget de-esser.

CONS

Crammed panel makes legending squashed; manual unfinished!

EXTRAS

The EL7 FATSO is a digitally controlled analogue device that offers many of the ‘musical non-linearities’ exhibited by older valve, class A discrete, and magnetic tape mediums. The two-channel processor will increase the apparent volume without increasing peak levels. Two channels of Empirical Labs compression are provided with several compressor ‘types’ with fixed attacks and releases.

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October 2005

WARM - THE NEW COOL Ms Dynamite is one of the most promising artists to emerge from the UK urban scene in recent years. Her intelligent and innovative debut album, 'A Little Bit Deeper' has undoubtedly become a modern day classic and her second album ‘Judgment Days’ is sure to secure her a deserved place in the forefront of the Urban and Mainstream music scenes.

Winner of many awards, Ms Dynamite has been using a range of Ivory series products for her recent live performances – including the Live 8 concert in Hyde Park, London, the MOBO awards, and several other appearances including an exclusive set at Abbey Road studios - which will be televised at a later date.

Read exerts from our exclusive interview with Ms Dynamite at:

www.tlaudio.co.uk/msdynamite

Ms Dynamite and the TL Audio 5051 Tube Processor

TELEPHONE +44 (0)1462 492090 // EMAIL info@tlaudio.co.uk // All trademarks acknowledged and respected.

PASSIONATE ABOUT TUBES

review

SE Electronics Icis Proving that you can improve the breed with attention to form and feel, SE has a valve cardioid that looks the part. JON THORNTON puts it through its paces.

H

AVING SPENT THE LAST FEW years garnering something of a reputation for producing inexpensive capacitor microphones, which while delivering good sonic performance were always — how can I say it — not exactly the last word in innovative styling, some of SE Electronics’ latest offerings are showing that this Chinese manufacturer is paying as much attention to form as to function.

Icis is a case in point — a large diaphragm, fixed pattern valve microphone. Although it features the same 1.07-inch diaphragm as the variable pattern Z5600 — albeit in single sided form — and indeed the same tube (ECC83A), SE has elected to give it a very different look to its stablemate.

And very fetching it is too with a slender body finished in light grey and a curved head grille assembly.

The look extends to the supplied shockmount, which features a metal tube into which the microphone drops, held in place by a locking ring screwed to the microphone base. A thick metal ring surrounds the elastic suspension, lending a very modern look to the whole assembly. Modern, but heavy — all of that shockmount assembly with the microphone in place makes for a weight that will tax some mic stand clutches, although the swivel joint and locking nut assembly on the mount itself seems to deal with it well enough.

Like all SE mics, the Icis (UK£297 + VAT) can be taken apart easily, and a glance at the internals reveals a tidy looking circuit board, a socketed valve and the output transformer. Connection to the remote power supply is via a supplied cable with locking 8-pin connectors, and the power supply itself has nothing but an XLR output and a power switch with associated LED. Its fixed cardioid response would seem to position the Icis as a microphone intended primarily for vocal work, and so this was its first task in the studio. Given

34

that it shares the diaphragm of the Z5600a, I was expecting it to turn in a very similar performance, but that wasn’t the case. And this isn’t necessarily a bad thing as I had found the Z5600 to be a little too sharp sounding and unforgiving of sibilance on occasions — particularly on female vocals. The Icis still has a distinct presence on female vocals, and one that has a little more edge than, for example, a U87, but at the same time delivers the ‘body’ of the vocals in a very smooth and warm fashion particularly when used close up. It’s difficult to know whether this is down to deliberately different tuning, or simply the presence of a single diaphragm, but it certainly isn’t just a single pattern Z5600a.

On male vocals, the story was similar, with the microphone turning in a nicely textured sound with good clarity and definition overlaid on a nicely warm fundamental. Bass extension is good — almost too good in some applications, and the lack of any highpass filter on the microphone meant that I had to turn to the console’s filters quite regularly. The slight edge to the vocal sound disappears quite quickly with even small movements off-axis to the mic, leading to a sound that is very warm, verging on a little woolly to my ears, so this might not be a great choice for very sibilant singers unless you’re prepared to wheel in the de-essers.

Turning to acoustic guitar, those same characteristics give a particular sound ‘out of the box’ as it were, with some nice high-end detail, good transient response and a fullness that flatters even slightly dull sounding instruments. Using it in this application isn’t without its problems — chiefly of positioning, as the weight of the assembly and the rather intrusive ring of the shockmount make this something of a challenge. This wouldn’t ordinarily be that much of an issue, except for one thing. You see, Icis seems to really be most comfortable when worked relatively close to a source, whether that be vocal or guitar. To confirm this, I tried it as a room microphone with a drum kit. Although it turned in a reasonable performance, and to be honest its cardioid pattern probably did it no favours here, compared with an Audix SCX25 in the same position it seemed to lose its sense of warmth, and that nice edge to the sound started to sound more brittle than defined.

Noise isn’t a problem — the specs quote 16dB equivalent noise (A-weighted) — and neither is sensitivity as a very healthy output appears from the supply unit. So given its liking for close sources, this makes the absence of any pad a little hard to understand — this wasn’t actually a problem in my testing, but then again I didn’t really use any really loud sources.

Nevertheless, viewed as a vocal microphone with some other applications, these shortcomings aren’t really that problematic, and its looks are certainly bound to appeal to a certain sort of client.

As a general-purpose microphone there are probably resolution

more flexible alternatives, but if you are in the market for something to add to your vocal mic palette, Icis is certainly worth a listen. ■

PROS

Nice balanced vocal sound; elegant and distinctive looks; works well close up.

CONS

No pad or filter; shockmount might be too intrusive in some applications; not as versatile as it might be.

Contact SE ELECTRONIS, CHINA: Sonic Distribution, UK: +44 1582 740260 Website: www.sonic-distribution.com

October 2005

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review

Digigram VX882HR and UAX220 If you run a computer then you’ll be needing a means of getting the sound in and out of it in a professional and robust manner.

Digigram has had plenty of experience in this business and builds them well. ROB JAMES investigates two.

T

HE PC CAN BE a truly professional audio tool.

The accent is on the ‘can’. As the Editor recently pointed out, computers are now a commodity item, built to a price, but this is not a great recipe for a machine you are going to earn your living on. As with the PC itself, so with the soundcard. The old mainframe computing adage, ‘garbage in, garbage out’ or GIGO Document1 AM and Page 1 was never truer.12/8/04 Domestic 10:18 soundcards built-in

sound chips are designed mainly for entertainment and reproduction and real-world performance figures can make shocking reading. The environment inside a PC is electrically noisy and distinctly not conducive to maintaining professional standards with analogue audio unless stringent measures are taken to guard against interference. To put it another way, the sound card must be properly engineered for the purpose and that is expensive. If done properly, the results can be impressive. At UK£1999 + VAT the VX882HR/BOB 8 combination certainly isn’t bargain basement but the price is reasonable for the standard of engineering.

Digigram does not have a terrifically high profile outside broadcasting circles but it has been making professional computer sound I-O solutions for many years. The VX882HR is the latest in a long line.

Perhaps the most useful formats for much of the work done today are two channels in and out and eight channels in and out. The former for obvious reasons and the latter both for surround work and to provide access to external processing. This unit has 8 channels of I-O in analogue and digital flavours. For now, this is only available to PC users, Mac drivers are currently unavailable.

VX882 HR headline features are four AES-EBU

stereo inputs with hardware sample rate convertors (up to 96kHz) and four AES-EBU stereo outputs (up to 192kHz). There are eight balanced analogue mono line inputs, eight servo balanced analogue outputs, one AES-EBU stereo sync input (up to 192kHz in play and record), one Word clock I-O (up to 96kHz), one video sync input, and one SMPTE/LTC sync input. The AES/ EBU sync connection can also be used for audio I-O. The analogue convertors all operate at up to 192kHz 24-bit. The ‘servo balanced’ analogue outputs automatically adjust the output level to accommodate balanced or unbalanced signals.

Obviously, the main context for this card will be the applications you intend to run. Drivers are provided for Digigram np, DirectSound and Wave, although I could find no trace of an Asio driver in the current release. I’m guessing Asio may require specific application support.

A software development kit is available.

If you use the DirectSound drivers the Windows DirectSound Control Panel gives access to volume, balance, muting, clock and input selection. The card operates at up to 192kHz. Although the digital inputs are limited to 96kHz this can be sample rate converted to 192kHz. The card itself is a hefty, short length, Universal PCI 62-bit/66MHz with backwards compatibility for 32-bit/33MHz 5V PCI and support for the PCI-X slots

PROfessionals need PROline MU Metal shielding 36

resolution

www.ultrasone.com October 2005

review

UAX220 USB audio interface

found mainly on Xeon workstation motherboards and not to be confused with the latest PCI-Express slots.

Construction is tidy and dense. There are three PCBs and evidence of considerable effort put into screening.

Two on-board connects cater for linking to a second, companion board and inter-board sync.

I happened to have a PCI-X slot free, so that is where it went. Installation is pain free once you’ve killed Windows’ attempts to do its own thing. Apart from the drivers, the installer also adds a couple of Applets. A very simple two channel recorder/player, Digigram Play/Rec, and the Digigram Control Centre. This provides diagnostics and some settings for the drivers. Apart from setting the buffers if your application doesn’t do this, the rest are probably safer left alone. The card can access the usual Windows sound sources, the DC/DVD-ROM drive, Media Player, etc.

Physical connections to the outside world go via a 62pin Sub-D socket — not a cable I would want to make.

Digigram offers two options for break out — a cable that divides into two branches, analogue and digital plus

October 2005

sync with XLRs for audio and AES-EBU sync, BNCs for Video and Word clock sync and a phono for timecode.

The alternative is Digigram’s BOB 8 breakout box. This connects to the card via a 62-pin Sub-D cable. With this number of possible connections the break-out box is a much neater and more robust solution.

With the BOB 8 the VX822 is a convenient, professional solution for PC audio I-O with high sample rate capabilities for those who need them. ■

PROS

A professional mix of features; good breakout box; well engineered.

CONS

Drivers could be better; no Mac support.

Contact DIGIGRAM, FRANCE: Website: www.digigram.com UK, SCV London: +44 208 418 1470

resolution

The UK£360.00 (+ VAT) UAX220 is a very different creature. A cigarette packet-sized blue plastic box with six flying leads, the unit is supplied with a neat handbag for protection when not in use.

It is literally plug-and-play on Windows XP, Linux, and Mac OSX. There are no drivers to install, it uses the USB Audio drivers installed with the operating system. The UAX220 provides two channels of balanced I-O on XLRs plus a 1/4-inch jack headphone monitor with volume control and Direct monitor push switch that simply connects the analogue inputs to the outputs. Power is supplied by the USB (1.1, 2 compatible) connection. It currently operates at 48kHz but a sample rate conversion is automatically applied by the operating system if required. Other sampling rates up to 48kHz are available for Windows XP users through a firmware update application available for download from the Digigram website. The UAX220 will be ideal for people on the move with a need to work with audio on several computers.

PROS

Ridiculously simple; monitoring; convenience.

CONS

Lack of mic pre limits utility for reporters; doesn’t feel very professional.

37

Lars Nilsson Not many engineers create their sounds through a primary emphasis on the use of different mic preamps to impart character and suitability for the application. Lars Nilsson has been doing it for ten years together with different mic choices and changes to his room acoustics. KEITH SPENCER-ALLEN meets him in Gothenburg.

E

NTER ‘LARS NILSSON’ with qualifying words into Google and up come references to a Nilsson who engineers and produces classical recordings that frequently find their way onto recommend lists in the hi-fi press. The search also brings up a Nilsson with a long and growing list of credits for jazz recordings. The clue that these are in fact the same L. Nilsson comes from the location — Gothenburg, Sweden, and the studio name, Nilento.

Having trained as a classical musician, Nilsson moved into basic mobile recording using early lowcost digital recording systems whose lack of facilities encouraged him, out of necessity, to develop a very particular approach to recording that was something of a throwback to earlier years. With little access to EQ, mic technique and acoustics were his main tools, together with detailed knowledge of how different mic preamps coloured sound. His interests naturally spanned classical music and jazz and he found that techniques developed for the former transferred easily to the latter.

38

His studio base at Nilento is a fascinating exercise in what happens when you prioritise the need to create optimum sounding local environments for acoustic instruments. Set deep in the countryside but still close to Gothenburg, the rural setting of an old farm is ideal. He has a close working relationship with local studio designer Claes Olssen of Audio Support who realised the current studio and is periodically called in to ‘tune’ areas in the studio to meet very specific recording requirements.

The mobile recording work is now much less as Nilsson has brought more work into Nilento. His jazz recording is drawing an increasingly international clientele with a reception area losing wall space to awards. An increasing number of projects are calling on his production expertise in both musical areas and with such a detailed knowledge of the recordings he makes, he’s also frequently requested to master the recordings, and even other producer’s work in a similar vein. resolution

You’d been a professional classically trained musician for ten years before moving into recording. What happened?

I was interested in sound and always had been although I didn’t realise it. When I was learning trumpet I wasn’t so keen to practice if I couldn’t use the room where I preferred my sound. If the room was busy I didn’t practice. It wasn’t exactly the way I thought then but I can see why now. Later, sound was what I got jobs on — I’m not a very virtuoso trumpet player but I am a good section player, playing every type of trumpet and I could change my sound to fit. My brother sold Atari computers in the 1980s and I remember him showing me how to edit with Sound Designer, which I thought was great, and I knew then that I wanted to work with computers.

That started it and then I saw the first Alesis ADAT digital recorders a few years later, bought a couple together with one of the first Pro Tools systems in Sweden, and started location recording. It was just a natural transition.

October 2005

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Was all your work mobile and did you have a studio at all?

I had a small control room here 15 years ago but I was mainly editing and monitoring two channel mixes I’d made on location. I’ve never enjoyed mixing like that.

I like thinking about a mix and completing a balance in my control room so I can listen properly.

I started very much in the classics — a lot of woodwind and brass ensembles, symphony orchestra, and chamber music. Here in Sweden orchestras have now got less money and it’s become tougher in the market, and I don’t do so much now. I still record and produce the Gothenburg Opera but I do more jazz now.

The jazz musician wants an acoustic sound. Many have been using pop recording techniques but I came from the acoustic side with my classical approach, applying that to jazz and it seems to work.

October 2005

How long has the studio been completed as it is now?

About two years. I’ve tried to make it a relaxed place with a lot of windows so you can watch the horses while playing jazz. There are six different rooms because I want different sounds ranging from the big room, which is fairly live, to a very dry space. I work a lot with the rooms and I always have, right from the start, because I had few other tools.

Do you keep modifying the recording spaces?

Yes, I always ‘EQ’ my rooms. For the last two years Claes Olsson [the designer] has been here every six months to fix and adjust specific acoustics, tuning the spaces. I want to get the right sound there

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Do you still do location recording?

Not so much now but I have done a lot. I do record some live performances — I was in London and Paris to record the Danish jazz singer Caecilie Norby. But now I just do mobile orchestral recording — maybe just 5-6 times a year — 90% of the time I work in the studio.

When you record on location now what kind of gear would you be using?

The main equipment would be a Pro Tools HD3 system with 40 inputs, a TC 6000 processor for reverb — I’ve two because they interface so well with the HD system — a Genelec monitoring system, a monitoring matrix so I can listen in surround, and up to 40 channels of mic preamps that include the Neve input modules I’ve already mentioned, API 312s, Focusrite Red 4s, Crookwood Paintpot, and some specialist hand made discrete Swedish designs from Mats Larsson of Audio Support and Bosse Hansens.

I’ve worked with mic pres like this for over ten years and the idea is to keep it simple, feeding the digital system with good mics.

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Pioneering that type of digital recording must have been difficult?

I’d started with a 4-channel Pro Tools system, moved to 16 and then I could take the Alesis ADAT recordings into Pro Tools and start editing. I don’t know that this was so good but it was what I had in mind from when I started. There was no EQ on the early system but I had learnt a lot about the characteristics of my mics, how to work with the musicians, where you place the microphone and so on. It was a good training because digital EQ didn’t sound too good even when it did arrive.

I’d bought an old Neve ‘Melbourne’ console and this gave me some 3115 input modules with simple EQ. I’d use any EQ needed before recording and if there was still a need for more it would have to be a couple of the Neve modules again.

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resolution

27/6/05 1:40:04 pm

39

craft in the room. I don’t want to touch my console EQ if I don’t have to, although sometimes you must. When I started recording in my new room it was a little ‘tough’ at 3kHz with strings and the like and also with drums.

I told Claes and we fixed those frequencies, and then I heard some reflections that I didn’t like and so we then fixed that, and so on, just refining the acoustics.

My control room has been rebuilt four times and now I’m satisfied!

I often have a dialogue with Claes when something doesn’t seem right to me — he listens to my recording and we discuss whether it is my room, my ears or my ideas that aren’t working.

What kind of mic techniques are you using — do you favour a lot or a few?

I can work with one microphone and sometimes it is 12.

It is so different. When I did the Lars Danielsson Libera Me 5.1 surround SACD, which got very good reviews in the press, the bass was featured as a solo instrument over an orchestra, 16 channels of percussion, piano, and contra bass. How do you do that bass in surround?

I experimented for 2-3 hours while he played and I stopped at seven mics, creating a surround signal on a bass solo instrument with one in the middle, two in stereo and two at a distance behind, a DI output, and I had a little extra mic. Seven mics on the bass and I thought I’m crazy but it sounds good in the surround.

Always lots? with a modern orchestra.

Not always, sometimes the opposite. I was in When recording a symphony orchestra you might Norway recently to record a woodwind orchestra. just want to use a 5-channel system but I always I went with 30 mics but when I mixed I just used have mics on each section even if I may not use them. two because they had a great internal balance. They I hate to sit in my control room afterwards and say were using historic instruments on trumpets and ‘Oh I missed that! I wish I had more of that melody  so   The   trombones I got the perfect balance between them line.’ conductor may tell me that he must have and the quieter clarinets and flutes which is difficult more clarinets on this phrase. He should have said

that on the recording session but I never say that — just fix it and raise the level.

Are there similarities between recording classical music and jazz?

Both can be very close to working with natural sounding acoustic instruments. But it can also be very different because you don’t need to modify the output

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40

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October 2005

craft of an orchestra. Jazz has a more erratic output so you need to add a little limiting and so on. It should sound acoustic but we also expect a little more of the sound and have to work a little differently.

So you work more on the colour in jazz?

I can’t do a recording of a jazz trio by the same rules as classical recording. My U48 or whatever I have isn’t half a metre away from the bass because I want a little more attack on it. But then sometimes I do older style recordings and then the mic might be about 1.5 metres from the bass. But it is quite near the technique for an orchestra, and you have to have good musicians in both cases. You cannot get great sound without that. Many times I’ve had good musicians for a recording and people ask me how I got that sound and the answer so often is that I did nothing — perhaps just a bit of limiting but that is enough with a good musician. Where so many go wrong is that they think the good sound comes from a lot or processing.

But here in the studio I assume you don’t take such precautions?

When I record here I know the equipment, how I place the mic, how I set it up, and the acoustics. When I do a jazz recording I can take half hour, maybe an hour to do the sound — never more.

I do a lot of pre work before the session — I never ask what sound they want before. I talk with them before and get a feel what they want, and what I want, and then let them comment on the sound I get.

What do you do for monitoring I use Genelecs in the studio. I have a 5.1 surround system with 1030s and a sub woofer and a 1031 stereo system. I take a Genelec system on location

but some of the monitoring situations are not very good and I have to use headphones which is OK because I just need to check the sound of the mic and phase because all mix decisions are left till I’m back in the studio.

You also master your recordings?

Yes, a lot of them. I’ll complete a mix and then wait for comments from the client or record company. If they ask for changes such as more or less bass, I won’t master off the mix but return to the Pro Tools elements, going through the individual tracks to see if there is a single track I can treat to remove or add bass, such as an overhead, rather than take bass down over a complete mix. It is a great way to do it. ■

Do you have a large collection of mics?

Yes but I always choose mics to fit with a digital system — it is different and people don’t talk about that. I use mics from Neumann, DPA, some Schoeps and some AKG. The mic pres that go into my digital equipment are my real reference.

So how do you use mic pres for sound?

When I want a guitar to sound a little lighter or you want a Pat Metheny-type (fast strum) sound, I use a Crookwood Paintpot with a Schoeps mic. And if I want it more light and open I take a DPA 4006 on the guitar. When I do a jazz recording I take a M149 or Schoeps with Neve mic pre because I want a smoother, less prominent treble. The same on piano — in a big band recording I never use Neve mic pres because I want it light and a little transient on the top, and I don’t have that other than on the cymbals and some high trumpets. So I’d use the Crookwood in that case. And on the bass drum I use API, and if I have some jazz with a little funk in it I always take the bass DI via the API but if it is a little smoother jazz then I use the Neve. That is my approach.

So you are painting colours with mic pres…

Yes, I do a lot with that. With an orchestra I take the tuba and bass drum through API which is very solid.

And then I take all the brass and trumpets through the Neve, and then I create a little more lightness on the woodwinds.

Does this have as much effect as the choice of mics?

It changes a lot of the character of the sound. Take the use of a KM84 as an overhead on drums with a Crookwood, it is quite different to using an API; I get more attack but not so much high treble. With the Crookwood you have more highs and more transients but with jazz generally I don’t want that. Mics sound very different according to what mic pre you use.

Would you say that you use different mic pres for different areas and depending on what you hear from the instrument you will decide which mic that you want to use?

Yes, you can say that but now if I do a recording and I don’t know the ensemble I don’t have the time to test that. I have learnt this technique over 10 years and I don’t have to experiment any more. Now most of the time I know the ensemble but if I don’t, I take a few more mics.

October 2005

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41

Ken Thomas His background includes time in sound reinforcement, desk manufacturing, and studio build ups. As an engineer, producer and mixer he’s spanned prog rock, New Wave and the current crop of interesting and innovative bands.

K

EN THOMAS STARTED as a PA engineer for RJ Jones, doing sound reinforcement for the BBC, working at the Eurovision Song Contest, Usher Hall, Edinburgh, and arriving at cricket matches in a van with valve Vortexion amps and a pair of horns on top. Gerry Kitchingham was the firmly established studio engineer at RG Jones, so the ambitious Thomas wrote to Trident and landed a job there. Having recently converted to 24-track they appreciated Ken’s technical experience. He had learnt electronics, assuming that the engineer should know how things worked, and this worked in tandem with his love of music and sound. At Trident he was simultaneously a tea-boy, tape-op, tape librarian and also helped out in the cutting room and spent time at the desk manufacturing facility. Thomas learnt engineering skills working with Ken Scott and Robin Geoffrey Cable.

Following this spell he moved to Advision, which had the reputation for the best engineers. He recorded string sections, jingles, and worked on War Of The Worlds. When punk came along he worked with Martin Rushent, who had started producing, on sessions with The Buzzcocks and The Stranglers, and he also engineered with Martin Hannett and with Steve Brown on a Rush album. When he left Advision 42

GEORGE SHILLING

he started producing a lot of New Wave bands and helped build up Andy Fernbach’s Jacobs Studios (roping in the gardener to help out, a youngster called Mark Stent) and later helped Tom Astor set up Orinoco Studios.

As an engineer, producer and mixer, Ken has an astonishing CV that includes a variety of fascinating and engaging recordings including three Sigur Ros albums, Psychic TV, Sugar Cubes, and recent projects that include Clinic, Dave Gahan and Hope Of The States, whose new album Ken was mixing when Resolution quizzed him at Olympic Studios. (Photos www.recordproduction.com)

What keeps you motivated?

I still find it incredibly interesting, I still get jobs, I still enjoy it, and I still feel there’s so much more, to break rules and so on, it’s like painting a picture. If I can help someone make a good record it’s fantastic.

How have things changed?

It’s totally different because you’ve got Pro Tools and loads of tracks, but I think it’s just as good, I think it’s even more interesting now. I’ve been enjoying the bands I’ve been working with, so I’m just gauging it on that. resolution

How do plug-ins compare with analogue effects?

I don’t know, I’m never too hung up if it’s a plugin or not a plug-in. It’s got a different character, but sometimes that digital character is great, and sometimes the analogue character is good, but until you plug it in you never know. I’ve got no set rules of how something should be.

What guides you in effects choice?

It’s all to do with feel, if it sounds good and you react to it, that’s fine. You’ve also got to talk about the whole picture. If you use a lot of valve gear it’s going to be quite warm and quite thick, so it’s going to be quite hard to gel together if you’ve got a lot of instruments, but if you haven’t, it’s easy. If you’ve got Elvis Presley it’s fine — or even The Beatles, it’s quite simple.

Where did you track the Hope Of The States album?

We started at Earth Terminal where the engineer there helped me track drums and bass, then various studios before going to Prague where we recorded strings, brass and vocals, and drank a lot. Then we went to finish bits and pieces at Strongroom, where the psychedelic colours inspired us, and the bar downstairs as well, that was very good. And then we came here to mix.

Do you usually work with an engineer these days?

On this project I’ve got James Loughrey, he’s been great, doing all the Pro Tools. Sometimes with Sigur Ros a guy called Biggie helped us on the new album, but on the first two I did everything myself, engineering, teas, cooking, built the studio, psychotherapist, the whole caboodle.

October 2005

craft Do you contribute arrangement ideas?

Sometimes you have to. If you’ve got a band who are maybe trying to be too ambitious, they just need to calm down, if the drummer’s doing fills all the way through and he can’t get them, you might say, halve them.

Is there an overriding philosophy?

The first album, I remember we decided we should have a lot of bass. I overdid it, but I’d heard some Enya song and it just had a good feeling of bass end.

Do you prefer that, or do you like someone telling you you have to have it done by Friday?

We won’t have it done by Friday! When it’s ready, it’s ready. Things sometimes take longer. I went through a stage of doing albums really quickly, ‘Dreams Less Sweet’ only took two or three weeks. Albums take a lot longer now.

Why is that?

It could be the bands I’m working with! I don’t know,

Pro Tools does take time, it gives you a lot of options.

If you hear something slightly out of time you could go along that route. With Sigur Ros I wasn’t really bothered about time, there’s lots of stuff that’s all over the place. I don’t know, it just seems to take longer.

It’s good for studios.

Are you concerned about the demise of so many good studios?

Not really, nothing really concerns me! There’ll always be good studios, something new turns up. Sometimes it’s good that everything gets cleared away and you start fresh. I think there’s a new concept of studios.

Some bands don’t want to be where Led Zeppelin recorded, it’s kind of boring.

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What about with an experimental band like Sigur Ros where the structures are less conventional?

I’ve been with them right from the beginning, we don’t really have to talk about it. We all go, yeah that’s right, or that’s wrong. We just do it, and sometimes, like on one of the songs, I just thought I’d put everything through distortion, (drums) and Jónsi came in and said it’s great, and I said alright we’ll just put it in. Then Jónsi will push it up more than I thought it would be, and I’d go, alright that’s fine, maybe I shouldn’t have done that. They’re just fun, people think they’re just sitting around with incense around them, but we’re having a laugh really.

Do you enjoy having a deadline?

Well we haven’t with Sigur Ros, we break every deadline possible.

Do you think records generally sound better these days?

When Pro Tools first came out a lot of records sounded thin, but it was just because it was the character of new technology, and ADATs too, but if the song’s good I think you can get away with it.

Do you enjoy the styles of music you are known for, or do you have a secret desire to do hip-hop?

Not really. One of the things I’ve always gone for is I do like working with nice people. That’s the crux of it. And I’m a bit arty-farty, so I like something that is breaking the rules a little bit. I know you’ve got to sell records, but it’s great, it doesn’t matter what age, but someone going, yeah, I wanna do this, and I don’t care a shit. Fantastic! I just want one note all the way through — fantastic! I just think that’s a breath of fresh air. I’ve always been interested in The Getty or reading about John Cage. I like everything, I’m not just into one thing. It all helps.

Does the band’s increasing technical knowledge threaten the role of the producer?

Not really, I think the producer’s there to help the band make the best record they can in the easiest way possible. You’re born, you get from A to B the easiest way possible, then you die!

Presumably in the days of punk the band would barely go into the control room…?

Punk was amazing because you’d just put a pair of headphones on a new band, and that was the first time they’d ever heard themselves, and they’d just be beaming, like, Bloody hell, this is amazing! I think something is going to happen with all this Pro Tools stuff, if someone could use it like Jimi Hendrix, then it would be amazing.

Are you conscious of the budget and the deadline?

That’s my job, I have to be. I did an album with Dave Gahan, and it was great working with Daniel Miller because he keeps it real, you’ve got to finish it by then, that’s it!

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dj and producer

personal creativity 13/09/2005 17:58:02

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craft Don’t you think a bit of history is good?

I love it, I did Dave Gahan at Electric Lady in New York, and I was just buzzing. It was just a small room upstairs, but it was fantastic, a great atmosphere, and he was just great to be with, we just had a good hoot.

Maybe with younger bands it’s different. But some studios are just not good, maybe the people who own it are pissed off or something. But Olympic’s on its own, Abbey Road too.

Do you think people get too hung up on perfection?

It’s so open, there’s such a new wave where everyone wants to make it rougher and 60s or singer-songwriter vibe, and then they shouldn’t, but they clean it up.

But then pop music to go on the radio has to be able to serve the standard. But I don’t know, I’m hearing new bands like Bloc Party and it’s not hi-fi to me. All these new bands sound like they’ve been processed a little bit, but maybe that’s the new sound. Some of the heavy metal stuff to me is just brilliant, the hip-hop stuff with all the low end, I think Missy Elliott and all that stuff is genius.

Do you rehearse with bands?

On the first Sigur Ros album I did, I hung out with them in the rehearsal room, and then on the last one we did it as we went along. Hope Of The States came round to my house and we went through drums and bass and ideas, and get really drunk at night.

I’d play them some Mussolini music or something!

And Sam would wake up in the morning and say, I’ve got a headache! I think you have to, before you start something you have to get your head around it, otherwise you waste a lot of time. Tempos are incredibly important.

What tricks and techniques are used on the new Sigur Ros album?

I suppose I’m just looking for sounds to sound engaging, so whatever I do to them, sometimes I’ll have one sound that I EQ really hard, and compress.

If it’s all getting a bit bland, it’s like taking a picture it’s just focusing, it might be on something that isn’t the vocal, like the snare drum or something that needs to engage you, take you somewhere.

That’s about the only trick I have at the moment — over-EQ and compress something! We try and record everything through amps — a lot of the stuff on Sigur Ros I put back through amps, because it was sounding too bland. And Kjartan the keyboard player was great, we couldn’t find a good amp so we stuck it through Georg’s bass amp. So it would work like, Piano’s not very good, haven’t got much EQ, let’s put it through the bass amp. I did a vocal with

Jónsi, and he said, The sound is shit. I found this old ribbon mic, and said, just give me an hour. The mic had that old 50s feeling about it, so I put it up and just put a speaker and re-recorded it. I couldn’t do it on the EQ because it sounded cold, we’d used some Focusrite thing we tried out, so putting it back through something worked. You just try things, it’s like being a sculptor. It’s nice just to smell the room and smell amps going.

What’s next?

I’m talking with Andrew McKenzie of the Hafler Trio about doing a conceptual piece, and we were talking about using the Zuccarelli head. Those little projects are great. He’s been crunching up sounds for many years. He’s one of the pioneer noise-makers, he just sits with seven computers around him all day. It’s another field, which is nice to walk into. ■

Holophonics and Psychic TV On Psychic TV’s Dreams Less Sweet album, it says ‘No microphones at all were used in the recording’ presumably because it was done with Zuccarelli Holophonics, tell me about that…

‘I was doing a filmscore with Rick Wakeman, and Zuccarelli turned up and said “I’ve invented this three-dimensional microphone head.” And I said I’ve always been trying to get cymbals high using different midranges and sometimes have a bass drum sitting low beneath the cymbals, and I thought this guy is right on my wavelength. So he played a plane going over, and I rang up Gen [Genesis P. Orridge] and said, right, we’re going to do this album holophonically. And I was always into photo-montage, and because they were quite poetic I could do what I wanted with them. I sat down with Gen and Sleazy and we came up with this idea of the Temple Ov PsychickYouth, and I said, alright, we’ll make it, this whole world, by using this head. I never really understood it, we couldn’t use it for more than twelve hours a day because it had to sleep, and it was fun. We made a hole and had Gen drive over it in a car, made loops, and went to the Hellfire Caves and did drums. We went to Caxton Hall and did the choir. We did oboes in a church and I was swinging it around.

It was a skull that had chemicals in it, and it also gives out a frequency. He did a lot of research on bats and basically was always interested in intuition.

Psychic TV were too, so I thought this was a good thing. Part of your brain is always beaming without you knowing it. Pink Floyd used it, but I think the drummer found it too hard to work with, he’d be hearing himself, like looking at himself. It never really got anywhere.’

Listening to that album there seems to be a lot of similarity with Sigur Ros, is that a sound you’ve lent to both projects?

‘I just think you find your tribe. You just find people who are similar to yourself. I’ve never really thought of my influences on any of these people, I never look at it that way. You go in, you do it, but I’ve never thought, that’s me. I think the attachment to me is boring anyway.’

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sweet spot

Life in the round Following his design and build, which has been covered in previous issues, JIM BETTERIDGE is now after a license for his Stationhouse post studios and explains his preparation and the process involved when dealing with the Dolby police.

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NLY A FEW YEARS AGO the worlds of television audio post and feature film audio post were set wide apart. Some upmarket television shows were mixed for Dolby Pro Logic offering some possibility of surround sound in the home but TV was mainly L/R stereo at best and was watched on modestly proportioned CRT sets. In contrast, the cinema was about big budgets, big screens and big sound coming from all around you; all something of a mystery to the average TV dubbing mixer.

Today with DVDs, games consoles and the threat of the HDTV broadcast revolution, plus 5.1 sound systems fairly flying out of the shops for under £50 a pop, every postproduction sound person undoubtedly sees their future in the round. And if you’re purely interested in mixing music, DVD, television or radio sound there’s nothing much to stop you. The technology is widely available to encode and decode the signal in Dolby, DTS or even SRS Circle Sound, as are DAWs, consoles and sound cards with multichannel outputs. If your sights are set on cinema, however, you can’t be so cavalier. To mix for mainstream cinema you need the approval of the godfather of cinema surround, Ray Dolby; or at least one of Dolby Laboratories’ representatives.

Dolby has achieved an amazing dominance in the cinema sound market. Having first applied the technology for the original LCRS matrix surround format in the 1970s it went on to establish a set of minimum technical criteria as a guide for cinemas wanting to show Dolby encoded films. Similarly, studio facilities wishing to mix in a Dolby format for theatrical release are required to achieve a set of minimum technical standards before Dolby will sign the official agreement that allows them to do October 2005

so. The standard has been adopted internationally and is an extraordinary achievement. When cinema goers see the Dolby logo they can be fairly sure of technically good quality sound. I say fairly sure because although the requirements for a studio license are strictly maintained the guidelines for cinemas are only guidelines; there’s no enforcement.

Equally, when working on a Dolby mix you can move from one studio to another knowing that they’ll be compatible. In a world teeming with incompatible formats that dog audio professionals most days of their lives you’ve got to feel a bit grateful.

Of course, Dolby’s isn’t the only audio format in use in the cinema. Most major feature films these days additionally carry DTS (Digital Theatre Systems) and occasionally Sony’s SDDS format, both of which are excellent and have a lot to be said for them. This does incur costs, however, and only the Dolby mix is considered a virtual must for a movie of even modest budget destined for cinematic release. It’s also worth remembering that it’s in Dolby certified rooms, aligned to Dolby-approved curves, that the vast majority of DTS and SDDS theatrical mixes are made and that there are no DTS or SDDS equivalents to the Dolby certification process. For simplicity, then, I will largely be limiting this discussion to Dolby technology and standards. I’ll also leave aside Dolby Digital Surround EX which is identical to Dolby Digital with the addition of a centre surround channel.

If you’re thinking of building or adapting a room for cinema work, Dolby is happy to send out an application pack including a 3,000-word document detailing the minimum requirements for feature film mixing. If these are too steep you might be able to attain the only slightly less rigorous requirements resolution

for a ‘trailers and commercials’ license. As the name suggests this licenses you to mix cinema trailers and commercials in SR or 5.1. Basically, you have to have a well presented, reasonably large (5m from back edge of your console to the screen for feature work), quiet, acoustically controlled room with a 5.1 sound system capable of certain SPLs and frequency responses.

You also have to have a projected image although it no longer has to be actual celluloid. Another thing to consider is that you can’t use bass management; i.e. you can’t have a crossover that throws anything below a certain frequency in the five main channels into the LFE channel to provide the necessary bass extension. The LFE channel is not intended to be used as a sub bass for the whole system but only as a discrete low frequency effects channel. You also have to have a mixing console capable of mixing in 5.1 and interfacing with the Dolby equipment.

After reading the documents and talking it over with Dolby a payment of £1,200 starts the process rolling. If for any reason you fail to get certification Dolby will return £1,000 of the fee — could they be fairer? The fee normally includes an initial visit to discuss what you’re going to need to do plus a final day’s visit by a Dolby engineer to check and align your new system.

I quickly discovered that my control room was a metre or so too short for features and so I went for the trailers and commercials license. The room was designed and equipped by Roger Quested with surround in mind and so we were fairly confident it would measure-up acoustically and I’d employed an acoustically transparent screen to allow proper positioning of all three front speakers (Resolution V3.6, V3.7). Great care had been taken with the air con, the boxing-in of the projector and the banning of all noisy kit to the machine room so that the ambient noise was very low. The mixing console, a Yamaha DM2000, was also designed for 5.1 operation and I knew it would be fine. However, Dolby wisely requires a diagram of how things are going to interface before they’ll give you the go ahead to purchase and take delivery of the necessary encoding/decoding kit.

Working in 5.1 with its five discrete channels is in many ways easier than the matrix soup of LCRS where the positioning of a sound is far less precise 45

sweet spot or predictable. Hence, whereas with 5.1 there is no need to have an encode/decode system while you’re mixing, with LCRS you need to monitor through the entire chain to hear what you’re doing. To this end Dolby provides on loan, free of charge to every certified studio, a DS4 encoder; this also means that Dolby can take it back at any point should your standards drop too far.

Into the DS4 you feed the discrete LCRS analogue outputs of your desk and out of it comes the LtRt (Left Total, Right Total) equivalent. This is then encoded through a standard pair of Dolby SR (CAT280) noise reduction cards. It is this LtRt mix that is married to the final 35mm print to be brought back into LCRS through a Dolby decoder in the cinema. So in the studio you need to have a similar decoder through which you can monitor. For all new installations this will now tend to be the digital CP650S, although until quite recently second hand examples of the older analogue CP65S were still commonly being installed. The CP650S will cost somewhere between £6,500 and £8,500 depending on which options you choose.

I used a CP65S which, linked with a DS4, gave me an entirely analogue encoding, decoding and monitoring system. When installed in a cinema the CP65 tends to be in a rack next to the power amps and so the fact that its outputs are unbalanced isn’t an issue. For a professional studio environment where everything is balanced and the power amps may be at some distance, an optional balancing board is fitted, resulting in the addition of the suffix ‘S’ for studio.

In addition to various stereo analogue ins and outs the DM2000 console has eight assignable analogue outputs called omni outs. One way to approach the interconnection with the console would have been to have separate paths for the record signals and the monitor signals. For this I could have used the omni outs for monitoring but would have needed to give up one of the DM2000’s 16-way I-O card slots to provide more analogue outputs for the record path thereby losing I-O flexibility and potential monitor channels for my Pyramix and Akai DAWs. The other way, quite commonly implemented these days, is to use one set of analogue outputs to feed both the monitoring and the record path. To anyone used to having separate control over the record level going to the master recorder and the level for the monitors this will seem a bizarre suggestion. The thing is that when Dolby aligns your room it establishes a fixed relationship between the level ‘on the master tape’ and the acoustic level in the room. Thus, when you’re mixing you have an absolute reference for how loud it will be in the cinema.

With complaints about excessive loudness in cinemas growing in recent years a set of guidelines has been presented by Dolby for maximum SPLs for features and a set of limits are actually enforced for commercials and trailers. Just as in television, the

MIX FROM DM2000 5.1 Digital

LCRS

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sweet spot latter have always been the biggest offenders in terms of high sustained levels. So as part of the installation for a commercials and trailers agreement is a Dolby loudness equivalent meter known as an Leq meter.

This takes the six analogue outputs from the desk and calculates and displays an average level over the duration of presentation as a Loudness Equivalent (Leq(m)) number. The idea is to allow occasional loud peaks and preserve exciting dynamics while avoiding the constantly high volume that seems to upset and fatigue audiences. The magic number that mustn’t be exceeded for a 5.1 mix is 85dBLeq(m). So it’s clear that the relationships between your meters, your monitors and the Leq(m) meter must be fixed.

With this in mind the DM2000 has ‘Set SPL85’ and ‘Snap to SPL85’ buttons to assist in the initial line-up and in returning to reference level when doing a Dolby mix. The desk will work in stereo, LCRS, 5.1 and 6.1 and allows you to assign output buses to each speaker position and also to assign an omni out to each of the buses; all this can be stored in memory. The console is generally very flexible in terms of assignable inputs and outputs but I decided anyway to bring up everything of any interest on an analogue patchbay, half-normalled for 5.1 and patchable for LCRS or stereo; time spent on a patchbay is rarely wasted.

The console has a range of other useful facilities: displays of graphics of signal paths and how the desk manages fold downs; detailed adjustment of delay, level and EQ on all outputs; and 5.1 reverbs.

Bearing in mind the need to switch between stereo and surround operation there’s a facility whereby the main stereo monitor outputs can be switched to become the Left and Right of the 5.1 system, using omni outs 1-4 for the remaining channels. Although probably quite workable, in practice I found it easier and more accurate to use omni outs 1-6 for 5.1 and simply repatch for stereo work. Likewise, I patch the PPMs across whichever outputs are in use and use line amps to align them, bearing in mind too that Dolby aligns its meters for -20fsd = PPM4, as opposed to the -18fsd television standard.

Although I did use the DM2000’s level trims on a couple of outputs to very precisely match levels reaching the Dolby processors, the CP65S itself provides level and EQ for the monitoring chain.

James Seddon from Dolby spent a day checking everything out with an array of microphones and some proprietary Dolby test equipment to achieve what is known as the Dolby X Curve — similar to the original Academy curve — on all the main channels.

Happily, the installation as it stood was not far off ideal and took relatively little EQ.

October 2005

One last word of caution. I’m no stranger to the soldering iron but the backs of the DS4 and CP65S are a daunting mass of barrier strip and confusing legends. Not only that but interfacing Dolby kit of differing vintages requires a certain degree of improvisation. You’ll need help so budget a few hundred quid for either Dolby or a third-party specialist. Apart from Dolby I asked Norman Brown of Norbro Electronics to help straighten a few things out and also to do an inexpensive mod on the CP65S to allow its inbuilt stereo Dolby SR cards to be used for decoding, thereby requiring me to purchase only two channels for encoding.

The alignment produced a marked improvement in terms of consistency when panning a sound around

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the room and an overall more cinematic sound, also mixes taken from Stationhouse for completion in larger feature licensed rooms have transferred seamlessly. Perhaps more important though, the Saturday night DVD with a few beers and a few mates has never sounded better. ■

Contact USEFUL SOURCES Dolby: www.dolby.co.uk DTS: www.dtsonline.co.uk Sony SDDS: www.sdds.com Norbro Electronics: (Dolby installation specialists) +44 208 549 1681

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in the picture

Blu-ray and HD DVD As if to prove that when you get a good strong story it will stand up to any number of remakes, they’ve gone and reshot the epic format war feature Clash of the Titans yet again. Coming soon! To a screen near you — maybe.

W

ITH AN ALMOST UNPRECEDENTED level of hype and general pontification (this article included, I suppose) Blu-ray (Sony) and HD DVD (Toshiba) are about to be unleashed on the unfortunate consumer. Press releases are flying like confetti and the national press and broadcasters are lapping it up, salivating at the prospect of yet another format war.

Sky has announced its intention to begin High Definition transmissions in the UK next year and various providers in Europe and the US have already begun. High Definition TV will be (maybe already is) the next ‘big thing’ and, whether the broadcasters like it or not, the main driver will be recorded material and (take a deep breath) games. In the UK initially there was a great deal of consumer confusion about compatibility but, once this dissipated, buoyant sales of LCD and Plasma screens with an HDTV Ready sticker ensued.

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People are supposed to learn from history. The Greeks, the Romans and the Hundred Years War. More recently and certainly more relevant, one might suppose that the great Betamax against VHS war might have taught the manufacturers something. More recently still, there are lessons to be learnt from the SACD versus DVD-A debacle.

It would be a mistake to underestimate the influence of games on the likely outcome. There are now a couple of generations weaned on the video game, many of whom now have high disposable incomes and the ability to discriminate on grounds of picture and sound quality as well as game play.

When the Sony PS-2 was launched in 2000, just as DVD was arriving, some surveys had it that more movies were watched than games played. Another significant bellwether has yet to commit to either format. Unlikely as it may seem, the pornography industry was pivotal in the success of VHS. resolution

ROB JAMES

HD DVD can be considered as an evolutionary step from DVD. Blu-ray’s origins lie in a format developed to replace DV tape in camcorders and mass duplication was not a primary design goal.

Both formats use the same form factor discs and blue-violet laser diodes. The reduced wavelength and therefore spot size over the red lasers used in DVD are responsible for the greater capacity. A single layer Blu-ray disc (BD) can be up to 27Gb, October 2005

in the picture double-layer 54Gb as opposed to 15Gb and 30Gb for HD DVD. Both systems offer the promise of higher capacities and transfer rates in the future. Both will be backwards compatible with DVD and both use similar codecs.

The surface layer of a Blu-ray disc is a mere 0.1mm whereas in an HD DVD disc the layer is 0.6mm thick, as with DVD. Turning a potential snag into an advantage, BDs will have much greater scratch resistance than DVDs thanks to a coating technology developed by TDK under the name Durabis. But they also require a different manufacturing process. Existing machinery cannot be simply modified to produce Blu-ray discs. HD DVDs use similar duplication techniques to DVDs and little capital expenditure will be required. This gives HD DVD an initial price advantage.

Toshiba questions whether the extra capacity is worth the higher manufacturing costs, but they would say that. Looking at the historical Betamax/ VHS battle one might be tempted to agree. However, Betamax lost not because of higher tape and duplication costs but because of market penetration driven by licensing and, at street level, deals with rental companies. Today, given a level playing field, the larger capacity format will win. But in any case, this time the positions are reversed. Sony has the huge advantage of the PS-3 and, unless one of the other games console manufacturers decides to use HD DVD and comes up with the killer game of the decade, things look pretty bleak for HD DVD. After the DVD CCS farrago, both camps are paying considerable attention to copy protection — one of the more bizarre ideas is the ability to ‘switch off’ players if the security of a particular make and model is compromised!

Both camps may well use the Advanced Access Content System, ACSS. Unlike CCS in DVDs the protection scheme can be changed if it becomes compromised. I don’t pretend to understand this stuff and believe it is largely pointless when faced with a horde of hackers who won’t rest until they’ve cracked it but if it keeps the studios happy…

Blu-ray and HD DVD will arrive in market conditions similar to those encountered by DVD-A and SACD i.e. the benefits to the consumer are much less tangible.

DVD succeeded beyond the dreams of avarice because despite such stupidities as Region Coding it was the only game in town and the studios resisted the urge to profiteer. CD took longer to establish itself because of the price premium over vinyl.

The abject failure of DVD-A and SACD to make any significant impression on the market should serve as a timely warning of what can go wrong.

Despite initial glowing reviews of their advantages over CD, the consumer has not been convinced.

The absence of inspired marketing hasn’t helped.

Truth is, only the golden eared and serious anoraks really care about the (minor) increase in quality and very few people seem to be interested in the various clever tricks they offer. These formats could still succeed with the right marketing, but that just hasn’t happened.

Blu-ray and HD DVD demonstrate far more obvious benefits to the consumer than SACD or DVD-A. The side-by-side split screen comparisons used to kick-start DVD will help sell the idea to the non games-playing audience. Sony is already using these in-store in the USA. Just in case you thought this battle would be the end of the story, think again. A new pretender already has its name down for primary school.

October 2005

Holographic Versatile Disc (HVD), the brainchild of Japanese company Optware, combines red and blue lasers in a ‘co-linear’ single beam. In essence the two beams travel along the same axis through a single lens. HVD is big, more than 1Tb and fast, 1Gbps, 40 X DVD. Toshiba has just invested in the company…joining Fuji Photo Film among several others.

One thing is certain, there will be a lucrative stream of remastering and authoring work for the foreseeable future.195x142.qxd ■ Audient Sumo 22/9/05 1:30 pm

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business

Why the SAN is shining in post New developments in network storage are putting the pedal to the metal for the production workflow. They’re also piling in the data bytes and changing people’s jobs in the process.

P

OSTPRODUCTION STUDIOS HAVE ALWAYS been at the cutting edge of networking technology.

As distributed corporate data processing reduced the entry cost and upped the data capacity of networks, Ethernet kit was used to send short motion sequences between edit suites over computer networks. As bandwidth demand escalated planned wiring was installed, hubs were replaced with switches and routers were added to segment traffic. SNMP monitoring and diagnostics tools were used to tune systems. RAID (Redundant Array of Independent Disks) systems were introduced: data from a single file is split over several drives, normally in the same rack, with parity information recorded separately. A small number of huge virtual disks could be created from a large number of smaller physical devices, with hot-swapping of disk modules often possible. The terminology of pocketprotector-man was soon being bandied about by arty London Soho types…they learnt that a ‘Jaybod’ 50

NIGEL JOPSON (JBOD — Just a Bunch Of Disks) is merely a volume concatenation not a RAID…and that redundancy may mean a wait of over 4 hours while a RAID system rebuilds a failed SATA (Serial ATA drive).

The traditional way of deploying all this storage is to attach it directly to network servers on a SCSI bus.

This method is scalable to an extent, but has several drawbacks, especially where very large (in traditional IT terms) contiguous files are concerned. The server may be very heavily loaded if multiple clients on the network want to access data simultaneously. This will mean the LAN will have to support a large volume of storage traffic, which will slow other network traffic.

The server with direct-attached SCSI storage will constitute a single point of failure, and SCSI cabling is limited to 30m so all storage has to sit by the server.

The truth is, even if a Cisco 6500 series switch can support 5.456Gbps, the client-server network model developed for corporate database use is not well suited to the huge files associated with digitised motion pictures.

Fortunately for media boys and girls, some heavyweights with budgets the size of Scrooge McDuck’s money bin had already tackled a similar problem. The aerospace and petrochemical industries provided the impetus for new network topologies capable of handling huge contiguous files — seismic data collections run into petabytes (millions of gigabytes). A SAN (Storage Area Network) using Fibre Channel resolves the bottleneck issues of the LAN. The cable backbone is a high speed fibre-optic system intended to support transmission speeds of up to 10Gbits/s. The storage traffic is brought to a separate network, relieving traffic on the LAN, and the data can be shared on an any-server to any-storage basis.

The management of data can also be simplified with software tools that probe individual devices as well as giving an overall view of the SAN traffic. The backbone is scalable depending on requirements; ranging from small SANs consisting of a single switch or hub, to large thousand-port enterprise SANs. Special topologies can be set up that allow higher data rates (using multiple links between devices and switches) or native remote mirroring, since Fibre Channel cabling supports distances to 10km. Not only is the single point of failure, the storage server, removed, but remote mirroring and redundant storage can also be mapped very efficiently using the Fibre Channel protocol. Without SAN shared storage, each server has to be equipped with the maximum amount of storage anticipated plus a buffer for temporary files.

‘With disk drive prices being what they are, it’s no longer merely a question of capacity, but also reliability, availability and access,’ says Steve MacPherson, technical manager at The Moving Picture Company. MPC creates high-end digital visual effects and computer animation for the advertising, television and feature film industries. ‘For Harry Potter 3, we needed a high-speed data bridge between our CG and Compositing (2D) departments. During periods of peak activity, we needed a system that could catch a high volume of CG rendering while simultaneously releasing resolution

that material on approval to the next step in our pipeline. This involves a huge amount of writes (from CG) and reads (to 2D). The Sledgehammer is deployed in exactly this capacity and has performed flawlessly.’

Sledgehammer is a networked storage solution from US firm Maximum Throughput, optimised for moving digital content at speed onto or off an IP-based network.

Technically speaking, Sledgehammer is not a SAN on fibre, it is a super-NAS (Network Attached Storage). As production centres move to fibre, SANs are often being deployed alongside existing networks, with hardware moving between the two backbones.

‘The SAN was a more cost-effective way of doing what we needed to do,’ Darren Woolfson, technical director of London’s Molinare Film and TV facilities house tells me. ‘We moved to Fibre channel because we needed to do online work, as opposed to offline work that we use the LANshare for. We have 15 Avid Adrenalines here, nine on the LANshare and six on the TerraBlock SAN. The SAN we use for our DI work has about 12 terabytes, which will get much bigger as time goes on.’ As top houses like Molinare set the standard, SANs have become the latest must-have and are racing off the shelves at dealers.

‘SANS are currently big business,’ advises John Harris of root6, SAN resellers for TerraBlock from Facilis, Apple’s XSAN and DVS-SAN. ‘TerraBlock is a hugely successful product and we are currently installing it at the rate of one a week. Its key advantage is price/performance. While it certainly does not do everything that Unity can, for example, it’s a perfectly adequate solution in many applications,’ John informs me. Silverglade, a prominent supplier of postproduction services for the UK’s terrestrial and satellite broadcast market, has just taken delivery of a TerraBlock system.

Founder Charles Frater explains: ‘We currently operate four offline and two online suites and we were looking at a number of options to increase capacity. Ways of working have changed quite a bit over the years and there is now a much greater volume of material available in the offline — consequently these suites have a rapacious appetite for memory. We opted for a 12-bay 4.8Tb system which ensures that projects are never prejudiced through insufficient memory and, to date, the system has proved very reliable.’

In-house facilities are also treading the SAN route, with Parthenon Entertainment, a leading UK supplier of factual programming, recently spending £250,000 on HD production kit. Two new suites based on Sony’s XPRI HD finishing system have been added, together with an Avid Xpress Studio HD suite, all connected to a large TerraBlock SAN. Currently the company has almost 70 hours of HD programming in production in its key genres of Nature, Science, History and People.

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business ‘HD is obviously the way forward,’ asserts MD Carl Hall, ‘but to ensure quality we felt it was important to keep material in its native HD format. XPRI enables us to do just that, 6Tb of storage may seem excessive but when you consider a one-hour wildlife programme may have 90 hours of material for editing, you soon come to appreciate it.’

The new SAN networks are not just about storage, they are about increasing workflow. With NAS all work centres have access to the same file, if that file is updated, it is automatically updated for everyone accessing it. If the update changes the asset and the user wants to make all users aware of the change, he can give the file a new revision number. When an editor opens a project that uses that asset, it will be marked offline and the replacement will be identified.

Having an industrial-strength network is one way to enable collaborative workflow. Users working on a project can save their output in a universal format such as Targa or TIFF. The next person in line can import that image or sequence and take things from there. But there is a limit to how much this improves workflow.

Metadata about the project generally does not flow with the image file and if you have two people working in different areas of the same project at the same time, this model does not help.

‘The networks have got quicker, the drives have got cheaper and smaller, you can write and read to them quicker, and it’s more cost-effective. You’re using a bit of fibre instead of copper, it’s pretty dull stuff, really!’

summarises Molinare’s Woolfson. ‘It only becomes interesting when you look at what people can do with it all. We had an Apple Xsan system here, 20 terabytes across 4 RAIDS, we did the Tour de France off it for four and a half weeks. It allowed us to record our three line feeds live onto the SAN which we could then edit, play out and so forth — all while we were still recording.’

Molinare’s crew have become Apple poster-boys as they pushed the boundaries of production. Editor Peter Wiggins adds: ‘Fast turnaround sports is the last bastion of tape editing -— other non-linear solutions have not been able to stand up to the pressures of such an operation. Traditionally, the Tour de France has been edited with two tape suites, but this year it was decided to use a combination of Apple’s FCP and Xsan storage to improve the coverage. Visitors who came to look at our system couldn’t believe I could play a clip on the timeline to within five seconds of it still recording — I had to show them to prove it! With over 450 hours of storage on Xsan, we didn’t need to delete anything, ever. Every frame of every part of the race could be accessed instantly by anyone. We also had seven years’ worth of pre-digitised archive for all those Lance Armstrong packages we knew we were going to edit.’

As audio facilities find their storage requirements expanding, and work closer with their vision counterparts, similar solutions are being deployed. Canal Plus in Paris has installed three Pro Tools D-Control systems on two Unity systems (one for adverts, one for the sports October 2005

vintage design

channel). Avid sequences are imported to Pro Tools, extra recording and editing is done, the mix is rendered and taken back to the Avid, which in turn sends everything to the playout servers. Swedish TV in Gothenburg also has a large installation of three Pro Control-based B-10 systems. Audio and picture on Unity still need to be segregated to different drives, so the network is not quite as transparent as a SAN theoretically should be.

Most Pro Tools on Unity installations have been for broadcast, as there is a 48-track limit when PT uses Unity rather than local storage. Third-party storage network solutions for audio are also becoming available. Studio Network Solutions (SNS) unveiled the newest product in its globalSAN family, the X-4, at the IBC show. ‘We’ve built the X-4 to be within reach of every multiroom audio studio. Price, performance, capacity — all factors were taken into consideration to make this the best choice for Pro Tools facilities,’ says Gary Holladay, president of SNS. It uses iSCSI over gigabit Ethernet (not Fibre) to keep the price down to just US$6,999, and includes 1.6 Terabytes of SATA RAID storage in a 1U enclosure plus two client licenses of SAN software. Of course, any network arrangement for Pro Tools requires careful integration with the Digidesign Workspace window, which controls whether drives are available for recording, playback or transfer: network drives generally register for transfer only.

New technologies like SANs are flavour of the month because they help top creative people in the production industry work the way they want to.

George Lucas re-equipped his Skywalker facility for Star Wars Episode III to use Pro Tools throughout the production, erasing the distinction between editing and mixing for picture. Sound designer Ben Burtt was quoted as saying he and Lucas ‘have always wanted to have one integrated system for all editing and mixing, including picture…have it all behave as if it were Pro Tools…do a picture change at any moment and ripple that change through all the soundtracks.’ Lucas just wants to sit on the couch, watch and listen, and put it all together simultaneously. To a music production person, accustomed to interrupting a mix to replace a vocal or solo, this approach makes perfect sense. In the more hierarchical world of film and TV production it’s really quite revolutionary. But, provided the results are up to standard, postproduction professionals are increasingly finding it makes better use of resources to blur some of the boundaries in what was previously a very sequential process.

MPC colourist Max Horton was not apprehensive about editing on a Quantel iQ system: ‘It was a great experience to do a bit of editing on the iQ. You don’t want to book an online suite just to drop in a couple of shots, so it makes great sense to me.’ Most technicians have a keen respect for the skills of their colleagues in other disciplines, but also appreciate that it can often be workflow time, as well as skill, that influences the overall quality of productions. As Darren Woolfson summarises: ‘My life is about making the work, from a technical point of view, move through our facility as smoothly as possible. In a perfect world we’d never buy anything, our expertise would be in the people we employ, we’d use cheap equipment and just charge a premium for our services because of the skills of our people!’ SANs are part of a new wave of technology ripping up boundaries and extending those skills beyond their original disciplines. It’s a data technology that will eventually extend the production process right out of the studio, to production people working at home or in remote locations and finally, whether we like it or not, to the client’s desk. ■ resolution

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meet your maker

Dirk Brauner Brauner Microphones is celebrating a decade this year and its company founder still can’t believe the success he’s had. He talks about the technology, the secrets, the passion and the art of microphones.

ZENON SCHOEPE

D

IRK BRAUNER GREW UP with a strong interest in electronics, music and natural science and designed his first electronic circuit when he was 14. He was in a professional recording studio for the first time at the age of 18 and was amazed and fascinated by the sound of old vintage valve mics.

He wanted one but could not afford the rare types that attracted him and wasn’t satisfied with newer alternative designs. ‘There was an element missing I could not describe at that time and that some of these old microphones had and then I discovered that tube gear in general was different. The sound was so much more natural and attracting,’ he says.

This fascination with valve gear led him to create his own and he studied tube circuits and built preamps, power amps and guitar amplifiers and discovered that tube electronics allowed him to ‘sculpture’ the sound in fine detail and offered an enormous variety of colours and possibilities. He had his own 8-track recording studio and with his microphone experiences still strong in his mind, in 1993 he designed his first large diaphragm tube microphone — the first prototype of the VM1. ‘It took me about one and a half years to learn about all the different aspects of tube microphones and I finally ended up with a microphone that was the essence of what I admired: a microphone that sounded truly natural, had an extremely low noise floor, fully variable characteristics and that was built like a tank,’ he says.

The final VM1 of 1994 was built entirely for his own purposes and Dirk claims he never planned to make a business out of it. But then people heard it and borrowed the mic and liked it so much that they urged him to build one for them too. He founded Brauner microphones in 1995 and the VM1 played a big part in the renaissance of the valve mic in the mid 1990s. In ten years Brauner has enjoyed enormous success and is now regarded as one of the premiere mic companies in the world.

What is special about Brauner products?

We follow a different approach from anyone else in the market. Many microphones today are built to a budget, which does not necessarily mean they are bad! Some Chinese microphones are really good, even superb when you look at the price performance ratio.

The technical specs are good and they sound good too. They are still light years away from the technical standard we can do in Europe, but that is a matter of time and most people don’t even need such high standards since the market has changed so much and we have so many semi professional people in our industry today — they don’t care about all these wonderful technological details. I don’t say this with disrespect, whoever comes up with a great result ‘did it’, no matter how he got there! But high standards make a huge difference — at least to the people who know the difference and are able to perceive it.

This is one of the reasons why the high-end microphone market has changed into a niche market, but this might pass when the Chinese catch up and 52

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they’ll do so soon enough, I think. But I don’t fear Chinese microphones or any other competition. If you are unique you are unique — anything else is a copy.

Look: there are such an awful lot of Neumann copies on the market — do any of those reach Neumann standards? Not even close! And they never will. You cannot wipe away those strong roots and all the decades of experience.

I want to talk about what is special about Brauner microphones but it is also a question of what is special about German microphones, since I’m part of this culture and I would be lying if I said I was not influenced by it. There can be no tribute strong enough for what Georg Neumann did — he was a genius — but I built my reputation not just by following because I have my own thoughts and took my own steps in my own direction, which is totally different and will become even more different in the future. Different from all other concepts.

I am in a niche market that will always remain a niche market because there are so many things that industrial mass production cannot do. It cannot care about the individual product outside of given margins.

As high as these margins might be and as good as the products can be that are made in industrial mass production, they have to compromise at a certain point. They will never be able to reach the quality and originality of a manufacturing process that allows no tolerances and that is not compromised with regard to every single product that is being built.

There is a huge world that you cannot squeeze into fixed margins and that cannot be automated and this different process of manufacturing does allow you to be free of all these limitations. Have you ever seen a robot listening to a microphone for hours and tweaking it until it is perfect? And also there are so many secrets, so many little details that you cannot see and these secrets I’ll never tell. They make our products unique.

Which microphone designs have you admired as being pivotal to the evolution of the species?

Without a doubt, the condenser microphone in general. This concept still offers a lot of interesting variety in so many aspects and the condenser microphone principle will surely always remain.

But there might be things coming that will be new.

I’ve been ‘brainstorming’ with Mr Heisenberg and Mr Plank and we’ve came up with some interesting ideas. Question is if these ideas work in practice and how long it will take to make them real.

How has the manufacture of Brauner mics changed in the last ten years?

It has mostly changed in details. The basic design is still the same. All the small things you improve when you learn how to do it better over time and when you are in love with a subject as much as we are there is barely a single day without an ‘aha!’

You don’t do a small diaphragm mic, why?

Schoeps does small diaphragm mics. The best in the world. No one can do this any better. Why should I do small diaphragm mics? A great small diaphragm tube mic could be interesting though.

What is the critical element in a Brauner mic?

The whole mic is the critical element and this is also why we are different. We always look at microphones October 2005

meet your maker from the holistic standpoint. Everything counts and the sum of all defines the final result. Most people focus on the capsule or a special tube or a special design concept. It is far more than that. It is an organic process. You cannot just build it around a single piece. There is such a strong interaction and such a huge variety of possibilities within microphone design and even within a fixed and a single concept of realisation there lies such a great potential of stray and uncertainty that you can only focus on each final product and all of its different aspects to make it constant. That involves a very clear and detailed understanding of what you do. In every single step of manufacturing you have to exactly know what you have to do to achieve that one result you are looking for and you have to focus on this entirety rather than getting lost in a single detail only.

What differentiates your designs from cheaper models made in China?

Its soul and its nature. A Brauner is more than a technical thing; it is a piece of authentic art. Each is individual and unique but clearly part of just that one family. We build to our understanding of the microphone as a piece of art rather than a science. Not just to meet fixed standards, like being technically, price-wise or marketing-wise in a certain range; it’s where we are unique. I see some competitors trying to

October 2005

follow our approach but I see very few really getting close and they are not necessarily always the ones that claim to do so.

How can you remain competitive from a price standpoint?

Through being noticeably unique, noticeably different and always worth the money! You’ll find nothing like a Brauner microphone. I’m not saying you will find nothing that you’ll like better — just nothing the same and nothing like it. It has a spirit of its own and our huge success speaks for itself. We don’t care about being competitive. We just care about our passion, regardless of price. Brauner microphones are entirely built in Germany, which is much more expensive, but it is worth it, since the value is higher and we also want to secure our workers’ and suppliers’ jobs. It is also a responsibility for this German part of culture to remain and to keep it vital. This culture cannot just be established in a different part of the world. Yes, it comes at a price but it’s more than worth it since you own a piece of authenticity that will remain. I also think that we are still offering an excellent price-performance ratio. A Brauner microphone is like a Rolex watch — it will never lose value and it is built to last longer than a lifetime.

If there is still technical progress to be made in traditional mic designs, where can it be made?

At Brauner! As I say, there is more to it than just bits (and bytes) and pieces, more than specs and dBs, marketing and money. There’s the dream of the perfect microphone. Dreamed by people who love microphones and who know what a microphone really is — a way into a different world, a world that has its own rules. The world of the electro acoustic form of art. Understanding how we can be the most natural part of this world, that’s where real progress is made and that’s were we will even further differentiate ourselves from the rest of the market — through our future development. There’s a lot we have learned and a lot that we are working on that is another huge step forward from where we are today and we are at a very high level already. We don’t want to decrease from that and the progress will be made in the understanding of the nature of the human perception of sound and in the understanding of the nature of the subject we are dealing with and how to expand our view and possibilities. We are experiencing some very fascinating aspects that will result in future products.

Where do you stand on the digital microphone issue?

On the side of sound, dynamics and versatility!

Do you regard your mics as instruments or tools?

Both. It is just the perspective from which you approach and use microphones — the perspective of an artist or the perspective of a technician. Again, how you get there doesn’t matter; what counts is the result. When you create an image, and this is what we do when we record, we need to focus on how we make this image transport the information we want it to contain in the way we think is best for maximum effect.

Keeping the impression of this image as a whole and perceiving things in their entirety is what matters and not to get lost in details.

That to me is the art of recording and also the basis of my understanding of how to master my field in this art — the art of microphones. ■ resolution

53

steinberg top tips

Nuendo3: Warp drive for post editing Editing audio events of a fixed size to video often requires time-consuming workarounds. This is especially true when a certain portion within an audio file is the wrong length. The section has to be cut out, altered and then crossfaded, often a time-consuming affair. Nuendo 3 offers an elegant solution for this task: Warp To Picture. This technique allows the fluid stretching and compression of any portion of an audio file directly to video. Not sure how this can save you valuable time in, say, SFX editing? ANGUS BAIGENT says read on.

I

’VE USED A DEMO PROJECT from the Nuendo 3 installation DVD to demonstrate Warp To Picture on an FX file. In the scene pictured in the first screenshot (Figure 1), the Tyrannosaurus Rex growl starts a bit too late. The growl begins after the T. Rex’s mouth is already open, rather than beginning as the T-Rex starts to open its mouth, which would sound more ‘natural’. Rather than move the clip forward, which would mean that the audio ends too early, Warp To Picture allows me to stretch portions of the audio file very quickly, with frame-accurate precision to the video material.

First of all, I activate Edit mode for the project. This is a very useful Nuendo facility for editing audio to video because the video playback in Nuendo follows the Transport, so if you use fast forward or rewind the video will follow the project cursor. The video player will also follow editing tasks like resizing, moving and, importantly for Warp To Picture, stretching using the Nuendo Audio Warp system.

Now I open the audio file that needs correction in the Sample Editor. The Sample Editor window should have its timeline set to Timecode, which allows me to see exactly at which frame the cursor is located just by moving the mouse around over the audio. In the example, I scrub through the waveform in the Sample Editor, and it’s easy to pick up the second part of the effect that comes too late, a low growling sound, which is louder and clearly visible in the Sample Editor window. To readjust it to fit properly, I activate Warp mode by clicking the button in the Sample Editor depicting the cursor next to a clock.

When I move a Warp marker, only the audio between the one I am moving and the next marker

on either side will be stretched or compressed. If no other markers are set, I will be Warping the audio of the whole event. This means that I need to isolate the portion of the audio event that I want to Warp by placing the Warp markers at the beginning and end of the required area. I add a Warp marker at the start of the ‘low growl’ phase that I picked up by scrubbing through it. As can be seen in the screenshot, the Warp markers are denoted by yellow lines with a small yellow triangle at the top. You can also see that I’ve placed Warp markers at a point well before the low growl and at its end. This means that when I use the Warp marker at the beginning of the offending portion of the audio, only the audio between that Marker and the one before and after it is stretched or compressed; the rest of the clip is left untouched. Of course, if I place any of the markers at the wrong point, I can remove the Warp Tab with a single click: I hold down shift to turn the Cursor into an eraser, and click on Warp Marker in the Timeline.

I repeat adding a Warp Marker until I’m satisfied it’s in exactly the right place.

Now it’s time to get stretching. By dragging the Warp Marker and watching the Video Player, I can drag the start of the ‘low roar’ to the point where the T. Rex opens its mouth, from 3:51:06 to around 3:50:22. This Warps the audio in real-time, and the beginning and end points of the audio remain where they are: only the start point of the ‘low roar’ has moved (Figure 2). And because I’m still in Edit Mode, Nuendo resets the Project Cursor to the beginning of the currently selected Event, namely the roar event, when I press Stop.

Conveniently, this means that just tapping the space

Fig. 1. The audio file before being stretched. On this screenshot, the Video Player window is a lot smaller than it normally would be, to fit all the windows needed into the same screenshot. The Video Window would normally, of course, be placed on a separate monitor or screen in full picture view.

54

bar again replays the event, now with the Warped audio, so I can check if the stretch has been done right without having to move the project cursor, which would mean changing to the Project Window. And because Nuendo 3 has, of course, unlimited undo, I can undo and redo the steps as often as I like.

When I’m actually working on the audio, the stretch is calculated in real-time using the aptly named Nuendo Realtime stretch algorithm, which is a highly CPU-efficient stretching algorithm used whenever a real-time Audio Warp task is performed on audio, like the one illustrated above. When I’m finished working on the clip, I render the changes I’ve made using another non-real-time but higherquality algorithm included in Nuendo. First, I close the Sample Editor and right click on the audio event (or Ctrl-Click on Mac) to open the contextual menu.

There I select Realtime Processing, and then Freeze Timestretch. This opens a dialog box where I can specify which algorithm I want to use to render the changes I’ve made. I use the MPEX 2 algorithm on the highest quality setting, which produces first class results.

Warp To Picture really is a remarkably flexible function: you can Warp any portion of an audio file, regardless of the size of the area you need stretched or compressed, and you can Warp as many points in an audio file as you need. And the most important reference for audio post, the video material, is dynamically displayed in real-time as a reference for any stretching or compressing tasks you carry out.

It’s no surprise to me that Nuendo users in audio post save substantial amounts of time over an entire project using this elegant little function. ■

Fig. 2. The correct portion of the audio file has been stretched in only a couple of seconds to match the video file with single-frame accuracy.

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55

technology The E-Trap. A concrete room known as the boathouse was chosen as a test location because of the extreme intensity of its low frequency standing waves. The construction is solid concrete on five sides. Figure 1 and 2 show two different measurements within the boathouse.

A

The E-Trap electronic bass trap The E-trap is a tuneable electronic bass trap that offers a new way to think about low frequency room acoustics and provides a precise tool for acoustical design. Bag End’s JIM WISCHMEYER takes us the through the thought process behind its creation.

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BASS TRAP IS A DEVICE that absorbs low frequency sound. It converts sound energy into another form of energy, typically a negligible amount of heat, in order to improve the uniformity of the acoustical characteristics in a room. Bass traps of various types have been in existence for decades and are employed in many critical listening and recording rooms. To better understand the bass trap it may be useful to review the basic problem and the solution the bass trap provides.

When sound is produced inside an enclosed space, like a room, the acoustical properties of the room are dominant in the distribution and characteristics of sound within. Sound waves reflect off room boundaries and interact with their own various reflections. The interactions cause higher or lower intensities of sound that vary depending on the location and frequency.

Typically the goal of a listening or recording room is to minimise the magnitude of these variations so that their effect upon the sound is minimal. Typical approaches to achieve this are to absorb and/or to diffuse the sound waves.

The effectiveness of passive acoustical absorption material is directly related to its physical size and placement. In general, the thickness of the absorption material and its position in the room determines the amount of absorption available at a particular frequency. This is directly related to the wavelength of the frequency. High absorption is achieved by an acoustical panel whose thickness is 1/4 of the wavelength. A quick review of wavelength vs frequency will show that a 1Hz wavelength is 1130 feet long, 1130 is also the typical speed of sound in ft/sec. A 10Hz wavelength is 113ft, 100Hz is 11.3ft, 1000Hz is 1.13ft. The relationship here becomes clear and it can be seen that lower frequencies have physically longer wavelengths and therefore require physically larger passive absorbers.

October 2005

technology

Practically speaking, the midrange and upper frequencies can be effectively absorbed using fibreglass wall panels and other room treatments.

Lower frequencies require larger absorbing treatments.

Professional studios and control rooms may employ 5 and 10- foot chambers with absorbing material to absorb low frequencies. Other techniques employ mechanically resonant membrane absorbers or acoustically resonant Helmholtz absorbers.

Previously we mentioned that the goal of the room designer is to minimise the magnitude of the acoustical variations within the room. The design of a good sounding room is a combination of art and science, where the room dimensions, construction and the implementation of absorption and diffusion will vary with the designer’s preferences.

It is uniformly agreed, however, that a dominant reflection that causes the room to ring or have a large increase in magnitude at a single frequency is detrimental to the sound. The prominent presence of a reflection in the bass range is due to one of the lowfrequency standing waves of the room and causes the room to sound boomy.

What is actually happening is a single frequency is reflecting back and forth between two walls causing an increase in magnitude and a ringing, or persistence in time, even after the original sound is no longer present. It is this increase in magnitude and ringing in time that causes the blurring and loss of definition to the sound.

What a bass trap does is actually dampen the ringing. To better understand this, think of a tuning fork that, once excited by an impact, will ring for a long time at a single frequency. In the case of a room resonance, the excitement comes from the sound source, such as a sound system, or is acoustically generated by a piano, drum or other instrument.

Just as a tuning fork ringing along with a music

Fig. 1. The magnitude vs frequency in the boathouse shows very strong resonant peaks. The red trace is with the E-trap tuned to two frequencies and turned on.

passage would be distracting, the room resonance in much the same way is booming along with the bass, masking and blurring the low frequency sounds. Bass traps add damping, much like putting your finger on the tuning fork. This reduces the ringing and allows all the sound to be heard more clearly.

This problem has been well known by designers for decades and is well controlled in the best room designs. These designs require extensive engineering and detailed construction and passive bass traps are routinely employed to dampen resonance modes.

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Typically passive bass traps, since they are large, can also affect the midrange and upper frequencies and require careful integration into a room.

Reactive absorbers such as Helmholtz resonators (HR) or quarter wave tubes are commonly used as the most effective solution for treating the low frequency standing waves. The problem with these low frequency absorbers is their large size.

Moreover, they normally split the mode targeted for treatment into two adjacent modes. In addition, reactive absorbers including HR can only be tuned

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technology

Fig. 2. A test signal consisting of sine wave tones — on for 0.5 second, then off for 0.5 second — starting at 27Hz and rising 1Hz each time it turns on to 40Hz, was played in the room using a flat response Infrasub system. The blue trace is the response within the room. The red trace is the response with the E-Trap on.

to a single frequency. When absorption at multiple frequencies is required a number of these absorbers tuned to different frequencies should be used. In practical applications, where space and cost are considerations, passive absorbers have limitations.

And unfortunately the re-tuning of a passive bass absorber, once deployed, is impractical.

Equalising the bass portion of the sound system

is sometimes suggested as a method of reducing the energy at the resonance peaks, thereby improving the flatness of the room response. Even those promoting this solution would agree that it would be far preferable to design the room so that it has little or no resonance modes and requires little or no equalisation. It is far better to fix the room acoustically by adding damping than to equalise the

signal. Equalising the sound affects the quality of the original sound and is not effective in reducing the ringing in the time domain when compared to adding acoustical damping to the room mode.

And, obviously it is impossible to ‘equalise’ a room where there is no sound system present, such as a recording studio where acoustic instruments such as drums or piano are recorded.

The electronic bass trap, while not likely to replace all passive bass trap implementations, offers a precise tool to attack the very worst problems with a high degree of effectiveness in a fraction of the space. It also offers a practical low-cost solution to existing rooms where problems are present and room redesign or large passive absorbers are not an option.

The E-trap can be viewed as an electronic acoustic absorber. It incorporates a feedback control scheme into a loudspeaker making the speaker exhibit the same dynamics as that of a reactive absorber. Because it is active it is capable of adding considerable damping to a room and still be very small in size (18-inches x 13-inches x 10-inches).

The small size allows the designer to place it in acoustically strategic locations without affecting the room’s upper frequency characteristics or impacting floor space and cosmetics.

The E-trap offers precise tuneability of two target frequencies simultaneously. The frequency and amount of damping is adjustable via controls. PC measurement software (for Windows) is included with the E-Trap to allow the user to pinpoint the frequency that requires damping. Once the E-trap is placed and tuned, it requires no additional attention.

The precision, ease of tuning and small size offered by the E-trap provide an additional tool to absorb low frequencies and can provide a dramatic improvement in the sound of a room. ■

Contact BAG END LOUDSPEAKER SYSTEMS, US: Website: www. bagend.com

katz’s column

That pesky peak meter Having sent Becky and Fred away on honeymoon bliss, BOB KATZ has got the place to himself and takes the time to ask if we should peak to full scale to fill up all those bits.

R

EADERS TELL ME they’ve received advice to peak levels to full scale so they can ‘take advantage of all those bits’. This is totally unnecessary with 24bit recording; you would have to reduce the level of a 24-bit recording by 48dB to arrive at ‘16-bit resolution’. So there’s plenty of room at the bottom. A 24-bit recording peaking to -10dBFS still has 20 to 30dB better signal-to-noise ratio than the analogue recordings of the past.

A good reason not to adjust gains for peak is that mic preamps and line amps that are maxed out tend to be noisier; total system gain structure will more likely be optimum if you adjust gains closer to unity. The Mastering CDs in 1980 next reason is that components with filters, such as equalisers, convertors, codecs (e.g.

MP3, AC3, DTS) and broadcast processors introduce intersample peaks that are above the level shown on a standard peak meter. I recommend that in all stages up to mastering, you should allow at least 3dB headroom for intersample peaks so the recording will not distort down the line. Because of the current state of level ‘competition’, in mastering I do peak to full scale but I use an oversampled peak limiter that prevents intersample peaks.

Please reserve bus-peak limiters for the mastering stage; they almost always make the sound worse and distortion accumulates.

When mixing, use a VU meter calibrated to -20dBFS, leave yourself 20dB of headroom and don’t worry about the peaks, which will never overload. Working to -20dBFS average, you will discover a new freedom in mixing. adjusted to 0 VU and then passed to the PCM-1610’s Liberated from the meters you can concentrate on the 16-bit A-D convertor as seen in the diagram. mix clarity and fullness. You can mix with your eyes Audio level was referenced, or ‘normalised’ to 0 VU closed with no fear of overloading. Many newcomers = -20dBFS (calibrated with a 1kHz sine wave). Note may think this a crazy idea — how can you mix rock ‘n’ that in the good ol’ days we often let the peak level of roll if you can’t compress? But I’m not suggesting that the analogue tape fall where it wanted, which in this you turn off your compressors -— by all means keep the picture is ‘only’ -6dBFS, due to the tape’s lower crest compressors that help get your ‘sound’. I simply suggest factor. Despite that, our transfers from analogue tape that you can remove all sound-degrading ‘protection’ sounded just as loud as digital sources peaking to full devices. scale, because the VU meters read the same.

If you must deliver this mix to clients who do not So, even though we all know that the VU meter lies understand what a raw mix sounds like, first try to and cheats and steals, there’s one thing it does much educate them that a mix is not a master, teach them better than the peak meter and that is regulate the how to listen to raw mixes. In the old days, clients average level. In the halcyon days of LPs, average levels knew what raw mixes sounded like but often today were very standardised; I could put an audiophile LP on they have unrealistic expectations. If you are concerned the turntable and then switch to Simon and Garfunkel, that inexperienced clients will panic just because they and probably not have to move the volume control. have to turn up their volume control, then try to make a Try that with two arbitrary CDs made today and you’ll ‘fake mastered’ reference for them. Don’t make the fake probably jump for your volume control. master any hotter than necessary or you will complicate One of the earliest PCM adapters made by Sony, the mastering engineer’s life by adding false expectations. model 1610, had digital peak meters with a scale marked Then send the unprocessed mix for mastering. +20 at the top. 0dB was supposed to be aligned to 0 VU Back in 1980 we did not have digital compressors, (+4dBu), that was the original design of the system. equalisers, or limiters. There was no competition for level; By this simple artifice, Sony made it clear to us digital no one was pressuring us to get the loudest CD. We were pioneers that full scale (+20) was supposed to be the lucky that we could reduce level in the DAE-1100 digital ‘unattainable’ level that you hit only on rare occasions. editor. If we needed to raise level we had to pass through As the first CD jukeboxes appeared, with no user an analogue gain stage, which we avoided if possible. volume control, the clients started asking ‘can you Analogue tape sources went through a transfer console, make my master louder?’ For better or worse we October 2005

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obliged. The first thing we discovered was that there was some peak wiggle room for the analogue tapes so we boosted their level to reach full scale.

As soon as we began to normalise all masters to the peak, this created an automatic loudness discrepancy of about 6dB depending on whether the source was analogue or digital tape. The VU fell by the wayside. Now the clients wouldn’t tolerate having their digital sources lower in level than their analogue tapes and so we had to invent digital compressors and limiters, and then we were off to the downhill races.

Welcome to lawless Digital Dodge City, 2005. Between 1980 and 2005, average CD levels of pop works rose by more than 16dB! Reissues are frequently remastered with excess compression so they fit on the CD changer along with current releases, and thus they usually sound much worse than the original versions. We have absolutely no standard for level (recordings’ average levels now vary by as much as 20dB!) and mastering engineers are resorting to using severe clipping to push the average level nearly to full scale! Say, why don’t we record square waves instead of music — a square wave has RMS value above full scale! It’s getting so bad that even classical clients are wanting their recordings to be louder. But loudness comes at a price; extra loud masters are distorted, irritating, lack impact, clarity and depth.

Don’t you wish those wise Sony/Philips inventors had standardised on an average level for the CD? Please, Sony — you got us into this mess, now help us get out of it. Every new release medium you invent will be ruined in the same way as the compact disc unless we find a way to enforce an average level standard.

Here’s my simplest suggestion: the new dual-disc, a hybrid DVD-CD, could become our saviour. The patentholders could make a rule that the average levels of comparable material on each half of the dual-disc must match within a couple of dB or the master would be rejected at the plant. The ostensible reason is for sonic consistency, so that listeners will not be jolted by level differences between the DVD and the CD halves, but my ulterior motive is that it will encourage CD producers to lower their own levels in deference to the superior sound quality of the (hopefully) less-compressed DVD half.

Mastering engineers doing dual-layer SACDs tended to follow this practice. Pass this idea on to the corporate parties in charge! ■

Information Resolution recommends Bob Katz’s book Mastering Audio — The Art and the Science as an essential source of information for every pro audio enthusiast who cares about sound. You can buy it on line at www.digido.com

59

ten

Things to do with an old analogue recorder You have an old analogue tape recorder sitting in the corner, unused and gathering dust. No one wants it but you keep it out of loyalty and sentimentality. KEITH SPENCER-ALLEN offers up ten ways that might make it useful to you again — from the very simple to the unnecessary (not all will work with all machines).

THE SHORT DELAY — This wouldn’t be your delay method of choice but it was once the only way.

Simply lace up a reel of tape, feed a signal to the tape machine input, switch the tape machine output to off-tape, and press record. Make sure that the input level to the tape machine is reasonably high so that the tape is well modulated and tape noise isn’t an issue. The delay time is determined by the distance between the record and replay heads, and the tape speed, but with average head spacing and common tape speeds it will be between a fifth and a twentieth of a second. If there is a varispeed facility this delay can be fine tuned to match a tempo. The sound of the delay will vary with the tape speed, and at the long delays (slower speeds) it will be less a discrete repeat than a ‘delayed fullness’. We’ve assumed use of a single track but even a multitrack could be pressed into service with the subsequent multiple delays presenting some unique opportunities.

same as the short delay but when the off-tape signal is returned to the console mix, it is also mixed back into the signal feeding the tape machine input. This regenerates the delay and adds delays on the delays.

By varying the level of ‘regen’ the delays can be adjusted to decay after a set time, or increased for permanent repeat. You need to exercise care because each subsequent regen is a ‘copy of a copy’ and in the analogue domain the quality suffers noticeably after half a dozen or so cycles,

THE STEREO REPEAT — A slightly more complex version of the multiple repeat and requires at least a two track tape machine. When the original off-tape signal is returned to the console mix, it is sent to a different track on the tape machine, also in record and set to off-tape replay. When this signal returns to the console mix, it is mixed back into the original tape machine feed. If the two returns are panned left and right, a ping-pong repeat effect results. Set the Res_horizontal_07.05 27/6/05 5:39is pm THE MULTIPLE REPEAT — The concept the Page tape 1speed and the amount of regen correctly and

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the effect can be a wall of sound from a quite simple original signal. And there’s no need to stop at just two tracks if more are available.

BACKWARDS — Making a recording on the tape machine, removing the tape off and rethreading it in reverse is still one of the simplest ways of creating a backwards musical track for spinning back into a mix.

FLUTTER EFFECT — On most tape machines, tape speed and consistency is transferred to the tape via the capstan. This is one of the precision parts of the deck mechanism and it doesn’t take too much to upset it (Many neglected machines will offer this ‘effect’ as standard. Ed). Adding any form of inconsistency to the capstan shaft will have an instant audible effect, and because of the speed of capstan rotation, will be a frequency modulation that has elements of the ‘underwater’. Preferred means of doctoring the capstan shaft used to be editing tape but any smooth low tack tape would probably do. Try using this as

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ten a delay to a reverb send. This technique probably does damage to the tape machine and the tape so beware! As the tape path is vulnerable to mechanical upset you can always try disturbing other parts of the machine to create effects — tapping the capstan or the tape feed spool has possibilities.

DOUBLE TRACKING — This really requires a tape machine with 30ips speed (or the imprudent use of lots of sticky tape round the capstan shaft) and varispeed to approach the delays that suggest a double track.

In all other ways this is just a simple off-tape delay, although the varispeed is needed to introduce a little variation in delay for a degree of authenticity to the effect. With some practice, the way the delay can be ‘played’ may add artistically to the end result.

TAPE SATURATION/COLOURATION — While it isn’t uncommon for basic tracks to be initially recorded on analogue multitrack and then transferred to a hard disk while retaining the analogue sound, handling overdubs like this isn’t so easy. Again it is the basic off-tape scenario but we are looking at a different aspect. The incoming overdubbed signal goes direct to the analogue tape and the off-tape signal goes directly into the hard disk system simultaneously, and it is possible to drop-in. The tape machine becomes almost an equaliser with the tape speed determining the end result. The harder the tape is driven, the more that the tape machine gains ‘compressor abilities’ as the tape heads into saturation. All that remains to be done is to dial-in an offset on this overdub on the hard disk system to bring it into alignment with the original track.

FLANGING — There are some who still maintain that tape-based flanging is deeper and more effective than any electronic means. However, it is a tricky operation and really requires two machines although with the will it should be possible to use our one machine, twice. Make a tape copy of the signal to be flanged. Play it simultaneously with the same signal originating from the hard disk system with the two signals panned to opposite sides. Using the varispeed or fingers on the tape flanges, speed or slow the tape replay against the hard disk signal. The output of the tape machine should be recorded back onto the hard disk system. This procedure should then be repeated using the just recorded track from the hard disk rather than the hard disk original. With the signals panned as before you don’t hear a ‘flange’ but a sense that the

image wanders to one side or the other and the most intense effect will be when this happens. Any offset required to realign the new tracks should be applied equally to both. The depth of the flange effect will be determined by the balance between them. I’ve never tried this but I guess it’ll work.

LOOPS — I hesitate to suggest this because there are far better ways of doing it but if you want analogue lo-fi characteristics then a heavily modulated 7.5ips tape signal on loop might be ideal. Make your tape copy of the signal and edit it up as a loop. This can then be laced up so that it runs past the tape heads, around any rollers and then over the front of the tape machine so that it runs freely — if the loop is long

enough to reach the floor that should provide enough ‘pull’ to maintain tension over the heads. If it is too short for the floor, place a small plastic spool in the loop to add some weight. There will also be the need to fool the tape machine into thinking that the reels are in use so trip arms, etc. may need a little tape to secure them in an ‘on’ position. And then lo-fi will just keep on getting lower and lower as that loop runs and runs!

AS AN ANALOGISER — Pull the analogue machine out, clean it, check the heads, line it up accurately, and you may even find that there is still some life in it doing what it was designed for! (I guess it’ll work but it’s far too radical. Ed)

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slaying dragons

Grounding Good grounding is vital for safety and for sound quality, but is often neglected because it’s not exciting and/or because it’s not understood. JOHN WATKINSON argues that it deserves more attention.

john watkinson ‘It is impossible to build a highresolution audio system using unbalanced connections. The fact that high-end hi-fi systems frequently use phono connections says it all.’

A

LTHOUGH THERE’S ONLY one type of electron, there are two categories of electricity as far as audio is concerned. There’s the type that works the lighting, the air conditioner and the coffee machine, and the type that carries the sound waveforms from one place to another. Unfortunately electrical signals have no idea of their purpose and simply obey the laws of physics.

As a result we can inadvertently mix the two categories. I can’t recall a case where the audio signals interfered with the power. One milliWatt of audio 62

isn’t going to have a big impact on a ten kiloWatt air conditioner. It’s always the other way round.

Thus achieving good sound quality in an electrically powered environment is like achieving harmony in a relationship where one partner trains circus elephants and the other collects porcelain.

Electrical power uses voltages high enough to kill, whereas with audio signals the biggest health risk is nausea. From an electrical power standpoint, safety has to be paramount and that is where grounding comes in. If the metal cases of all electrical devices are connected to ground, then no-one standing on the ground can receive a shock from them. If a live wire comes off inside the device and touches the case, then it will fry the fuse instead of the operator.

Devices like this need three-core power cords. There is another category of device that is constructed in such a way that it is physically impossible for the body to become live even in the worst case. These devices are referred to as double-insulated and can use a two-core lead. For example, an electric drill with a plastic body simply needs a plastic gear in the transmission so that the chuck can’t become live if the motor insulation breaks down.

In all electric power systems there will be currents in the grounding system. All devices have finite leakage resistance, so some leakage current flows to ground.

All devices have some capacitance and the power is alternating. In many devices, there is an interference filter at the power inlet. This will often have capacitors between the power leads and ground, guaranteeing ground currents. Even a double insulated device can have ground currents due to capacitance.

Thus if the grounds of two devices are connected, the chances are that some ground current will flow.

Every time we connect up a lead having phono plugs we take that risk. The screen of the phono cable is the common terminal of the audio signal, yet we are allowing ground currents to flow through it. Unless the screen has zero impedance, the ground currents create a potential that is added to the audio waveform: the common-impedance effect.

Consequently it is impossible to build a highresolution audio system using unbalanced connections.

The fact that high-end hi-fi systems frequently use phono connections says it all. The use of audiophile phono cables with massively over engineered gold plated plugs and exotic wire makes about as much sense as gold-plating a cow pat.

From a linguistic standpoint, the term audiophile is derived from the Greek philos, meaning love.

However, in the conventional English usage, it can mean love of an unwholesome or perverted nature. I think there should be a register of audiophiles so we can keep our children away from them.

Fortunately there is a solution that works perfectly and it is, of course, balanced signalling using twin wires, twisted and screened. In such a system, the ground conductor acts only as a screen and if ground currents flow in it they have no effect because the signal is carried differentially in the twisted pair. resolution

Consequently the ground connection of twisted screened cable should always be made at both ends, otherwise the screening ability will be compromised.

In a well engineered system, each device is enclosed in a metal case that connects via pin 1 of the XLRs to the screens of all the cables so that circuitry and signal paths are surrounded by continuous metal screening. Inside a device, the common terminal of the circuitry is connected to metal in one place only.

Such a single point connection prevents the formation of ground loops.

I often hear cases where the screen was disconnected at one end and the problem went away, thus ‘proving’ my theory wrong. However, if disconnecting the screen removes a problem, there is a design fault. Pin 1 of an XLR connector should go directly to the metal enclosure and nowhere else, unless that one XLR has been designated as the single ground point for the internal circuitry. Modern XLRs have an insert that connects pin 1 to the mounting screw. Often XLRs are grounded to the printed circuit and then earth currents can flow in the circuit board where common impedances with the audio signals exist. Thus in an adverse ground current environment lifting the screen at one end would reduce the hum. However, in an adverse RF environment, lifting the screen would make it worse.

It is easy to check XLRs for proper bonding to chassis by injecting a high AC current between pin 1 and chassis with the connector active. If the system hums when the ground current is injected, there is a design fault. A solder gun of the type where the heating element is a copper rod running from a single turn transformer secondary makes an excellent current injector if the element is removed and short leads with clips are fitted.

Large installations may have a separate grounding system, called technical ground or technical earth, which is used as the safety ground for all of the technical equipment. A separate grounding system is used for all other electrical devices. In principle the idea is a good one because the magnitude of ground currents can be reduced in the technical ground.

However, it requires constant vigilance to ensure the technical ground is not compromised by connection of inappropriate devices. Also there will be a ground potential between technical ground and nontechnical ground.

I remember well the case of a large digital mixing console in Paris whose sound quality varied throughout the day. It transpired that the building had a superb technical ground installation. After two days of tearing the place apart, I found that the main ground of the mixing console went to non-technical ground. A major load on the non-technical ground was the restaurant. When they started cooking, the difference in ground potential between technical and non-technical grounds grew and with it the apparent jitter on the AES-EBU signals received from the console.

Thus although digital systems have more immunity to the environment than analogue systems, it is as well to remember that the immunity isn’t necessarily total and good grounding practice is still needed in digital systems.

Grounding and screening are related subjects.

Unwanted signals can be picked up from alternating electric and magnetic fields. In the case of electric fields, typically radiated from power wiring, the coupling mechanism is capacitive and an electrically conductive screen that is grounded will block the pickup. This is the purpose of the braided wire October 2005

slaying dragons screening around audio cables. However, conductive screening will not stop magnetic fields induced by currents in power wiring or leaking from transformers.

True magnetic screening requires ferrous metal so that the flux prefers to pass through the metal rather than the device being screened. However, for a magnetic field to induce a voltage, there has to be a conducting loop with a finite area. Balanced wiring is twisted so that the loop area is effectively zero. Thus differential mode pickup is near zero. Common mode pickup is rejected by the balanced receiver.

On the subject of balanced receivers, they will only reject interference if they are operating as true subtractors. The problem is that many balanced inputs only have this characteristic over part of their frequency band. At higher frequencies true subtraction fails and the common mode signals get added to the signal path.

A correctly engineered differential receiver has identical impedance on both inputs at all frequencies.

The CMRR must be tested to find the highest frequency at which it is maintained. Low pass filters must be placed in the inputs to remove signals for which true subtraction is not possible.

Naturally audiophiles would want those filters removed because they reduce the bandwidth of the audio system from absurdly ultrasonic to mildly ultrasonic. Go ahead, remove them and then wonder why the local taxis break through on the audio.

I suppose that brings us to further meanings of grounding. Marketing people, who frequently don’t have a grounding in audio, can come out with the most preposterous statements that are, intellectually speaking, a form of noise. People who are grounded are better equipped to identify and reject that noise. ■

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Use balanced signalling wherever possible.

Ensure screen is connected to pin 1 of XLRs at both ends.

Check pin 1 of XLRs goes direct to enclosure metal and not via the PCB. Keep technical and non-technical grounds just that.

DON’T

Use unbalanced signals.

Disconnect grounds randomly, you may fix one fault and introduce another.

Give credence to any audiophile marketing nonsense.

October 2005

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63

your business

Pilot Fish DAN DALEY doesn’t want much but he would like to see the producer become the object of more commercial desires.

He wants to see them selling T-shirts.

dan daley

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RECORD PRODUCER HAS come to be viewed a lot like James Dean in that wonderful black and white photo in which he is walking through Times Square on a blustery night with shoulders hunched against the wind and overcoat clenched together in front: the ultimate lone wolf. With actors, of course, as soon as ‘Cut’ is yelled out, the camera pulls back to reveal a flotilla of support personnel. The record producer, on the other hand, will more likely remain the only person on the set. Production can sometimes be a collaborative effort, but it’s even less often an ensemble one. Few end up like Sir George Martin, as the Fifth Beatle. The producer’s role is by nature often adversarial — your craft is, ultimately, telling someone what to do with their art. They, in turn, will often tell you what to do with various parts of your anatomy.

Well, you’re not so alone anymore. The Internet is bursting with sites that will tell artists how to play, perform, compose and dress, and there’s no shortage of audio engineering sites and threads and forums. The producer, amorphous and often anonymous by nature, has also been getting more notice (that may or may not have been helped by Phil Spector’s proclivity to use handguns on house guests) and, thus, more support.

In addition to a growing plethora of business managers angling for producer business, the solitary producer now has more resources to turn to on the Web. Two that come to mind immediately are RecordProduction.com and StudioExpresso. com, based in the UK and the US respectively.

‘We’re helping them do something they don’t often do for themselves, which is promote themselves to the pool of artists out there.’

RecordProduction was founded by Mike Banks, who has a day job at Solid State Logic (Resolution enjoys a tight and mutually agreeable association with Recordproduction.com. Ed).

Studio Expresso was created out of Los Angeles by Claris Sayadian-Dodge, a multifaceted entrepreneur who also manages four producer/arrangers of her own. The sites differ in tone and look, but they share certain technologies and philosophies. Both feature interviews on Real Player videos, shot on mini-DV cams, of producer and producer/engineers, and both act as virtual coffee shops — Studio Expresso was named so for exactly that reason, says Sayadian-Dodge — creating a network linking producers and artists and others.

The decline of the conventional studio and record company infrastructure left producers in a bit of a business limbo. An engineer could always take a job at the BBC or NBC, if necessary; producers, though, tend to survive on wits and wisdom, less immediately saleable commodities in the everyday world. These sites fill a networking gap left that once was handled by recording studios and A&R expense accounts.

‘We’re filling a void that was created when the record labels began to disappear or get smaller,’ says Sayadian-Dodge. ‘We’re creating a place where producers can be introduced to new artists and record labels. But it’s not the same business as it was even ten years ago: with so many people able to make their own records at home, we’re also trying to make the process of record-making more transparent and understandable to more people, particularly in terms of explaining the role of the producer to more people.’ (This is reminiscent of exactly what happened a little over a decade ago when home-made recordings began to get released commercially on a larger scale: the equipment

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your business instruction manuals may have explained things like gain structure and equalisation, but it neglected to mention that it’s a good idea to have your recordings mastered before releasing them. Few of the new generation of recordists had any notion of what mastering was. It’s not surprising that the producer also is a void in many people’s experience.)

The interviews and biographies and discographies available on the sites also help address a seemingly innate lack of awareness about marketing and selfpromotion among many producers. ‘People don’t talk about the business end of producing records enough,’ comments Johnny Jaskot, who arranges and shoots many of the North American video interviews for RecordProduction.com. ‘The people we do record interviews with seem very grateful about it. We’re helping them do something they don’t often do for themselves, which is promote themselves to the pool of artists out there.’

These pilot fish swimming alongside the producers have something in common with their heroes in that regard: they often let their passion overtake their revenue stream. And thank goodness for it, since without personal ardour it might not get done at all. Jaskot will leave his home in the town of Hampstead, New Hampshire, 40 miles from Boston in the hills of New England, and travel at his own expense to New York or Los Angeles to video interviews. ‘We accept donations,’ he says, innocently. He makes his living as a programmer and as the operator of Babblefish.com, a website that does Internet-based language translations.

RecordProduction.com does not charge to make the video interview or store a downloadable version on its site.

Both sites have some advertising but neither supports itself from that. At Studio Expresso, the more capitalistically assertive of the two (and just barely so), successful referrals made through the site’s service between a producer featured on it and Res_MTvertical_06-04 pm an artist have a very informal20/4/04 nature. ‘The1:37 producer takes care of us,’ says Sayadian-Dodge, who relies

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October 2005

more on Karma than written agreements. Says Jaskot of his main compensation, ‘I’ve made quite a few friends.’

Regardless of the economics, I think he’s on to something there. Still, history has shown that content — any content — has enormous potential value. Something as simple as a photograph is often still a rare commodity in this business — seen any recent pictures of Mutt Lange lately? And despite pleading in this column several months ago for producers to take charge of their own promotional materials, I still have three stories hung up in production for lack of pictures. Having extensive full-motion video of a producer at work strikes me as almost luxurious. As luxurious as it is valuable.

Who would have thought Bob Dylan’s autograph would have been worth much in 1961?

The nature of the producer’s way of work has left the rubric with little to make investing in support services seem worthwhile. I mean, there are tons of musical instrument retail websites that include editorial content and how-to departments, all geared towards wooing the MasterCard out of the wallets of musicians. A proportionally smaller number of pro audio websites operate similarly, catering to sound engineers. But rarely is the record producer singled out by name as the core audience for this type of service. And perhaps that’s not so surprising. After all, what do record producers buy?

You need not be a musician or an engineer to be a producer. That’s great, but it gives those who want to sell something little to peg their hat onto. And that’s where services that support careers spring from the ability to monetise providing that support.

Not that these labours of love are not a great thing for record producers. They are. There just should be more of them.

There are two, somewhat more formal, support organisations for record producers — the Music Producers Guild (MPG) in London and the Recording Page 1 Academy’s Producers & Engineers Wing (P&EW) based in LA. They’ve been the core organisations for producers for years and remain quite valuable in those roles. But I want to see them start to sell Tshirts. I’d like to see the producer become the object of more commercial desires. The closest I’ve seen producers as a group come to leveraging their own collective worth was META, the group formed late last year comprised of Phil Ramone, Elliot Scheiner, Frank Filipetti, Al Schmitt, George Massenburg, Ed Cherney, Chuck Ainlay, and Rory Kaplan. I’m not sure where that venture is at this point, and perhaps neither are some of the principals, who anyway had never seemed completely on the same page about what they were doing or how much of it was for commercial gain and how much was pro bono. Again, getting producers to organise themselves in a focused manner is a lot like trying to herd cats.

Perhaps the best approach is what’s evolved here now: a handful of organisations, such as MPG and P&EW, that provide some kind of formal infrastructure that producers can enter and leave at will and whim, and the passively commercial infrastructure provided by RecordProduction.com and Studio Expresso — a bit of health care insurance from one and a bit of cocktail-hour networking from another, perhaps. ‘Every producer is very different in the way they work and look at what they do,’ says Sayadian-Dodge. ‘They’re a very different breed than the rest of the music business.’ Hear, hear. ■ resolution

headroom BLUMLEIN TECHNIQUE Tim Summerhayes in his Craft interview (V4.6, p40) made a ridiculous statement about Blumlein mic technique. I quote ‘Those people who can go out and even record an orchestra with a stereo mic just sound crap to me.’

Coincident arrays have provided fantastic results in all sorts of venues and acoustics. A self balancing ensemble will always produce a musically satisfying result. The recordist’s role is to capture this moment not flood it with technology. This is better done on the venue floor than later. Film music is not classical music. A million pound mobile is not required for acoustic self balancing ensembles that are directed or conducted.

Alan Dower Blumlein’s M+S arrays still engender beauty and simplicity of design, 70 years after their inception, all that plus high def TV and centimetric radar, for which he died. What a man!

Roger Long, Stroud, UK Just a brief response to the many comments I have read about the interview I gave last month. I mean no disrespect to anyone but I do stand by what I said.

There was much comment on my remarks about the stereo mics. The Blumlein innovation has been the source of many fine recordings over the decades and I have certainly used it as a starting point for almost all classical work, but I do believe that by supplementing this with modern recording techniques a far superior result is achieved.

Many have used the BBC as a yardstick; therefore I would suggest we all go and see how the Proms are recorded. I was a paying member of the audience at three of the concerts this year and you wouldn’t believe how many mics I saw!

Tim Summerhayes, Sanctuary Mobiles, UK

PEAKED QUALITY AGAIN In response to your leader and subsequent letter from Richard Poole of the BBC World Service (V4.6) it would seem that many of us are broadly in agreement that in the ‘fast-food’ world of entertainment audio quality has been pushed into the sidelines. And much as we may talk about the need for education or the ‘talking up’ of the importance of audio, who we target and how we go about this remains an important question.

And as Richard Poole implies and Zenon Schoepe states the consumer doesn’t seem to care anyway even though we take pride in our craft and feel the need to maintain our professional standards. Having said that, just because we live in a disposable, quick-fit age it doesn’t mean that we don’t want things to be good and to work properly. As consumers we may take what is given but part of that ‘given’ is that what we get works well. So maybe in some ways the bar has not been lowered but rather raised. Maybe there are new higher levels of excellence to be aimed for.

In decades to come will anyone ever say I wish we’d paid more attention to the audio when we still had the chance?

M. Jones, Manchester, UK I think your last outro point is the crux of it all. We may still be buried in the ‘soft knee’ section of the compression curve — if I can use a crass analogy — but we won’t really know if we under or over did it until it all shakes out on playback in the years to come. It is the retrospective element that puts things in context and it would be truly dreadful to realise, in years to come, that we lived in times when things still could have been changed just by being a bit more precious about the audio.

ZS

AUDIOlookilikies

The first in a series (We got these out of the way first so no one can complain).

Send your audio lookilikies to lookilikies@resolutionmag.com

Zenon Schoepe, Resolution

Swiss Toni, Used Car Salesman

Clare Sturzaker, Resolution

Sue Barker, Tennis Star

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