Resolution V5.8 Nov/Dec 2006 by Resolution

AUDIO FOR POST, BROADCAST, RECORDING AND MULTIMEDIA PRODUCTION

V5.8 NOVEMBER/DECEMBER 2006

John Cornfield — Cornish landscapes and Soundscapes André Jacquemin: Monty Python and 50,000 commercials later Comparing stereo pair and multimic mixes Surround recording and playback with discrete layered sound Ten musicians who pushed recording boundaries Radio compaction and compression: why does it sound so bad? REVIEWS: Groove Tubes The Glory Comp • Sony Vegas 7.0/DVD Architect 4.0 CEDAR Cambridge V3 • Earthworks TC25 • Digidesign Mbox 2 Pro • SE Titan

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November/December 2006 V5.8

ISSN 1477-4216 THE PRO END-USER AUDIO PRODUCTION MAGAZINE

News & Analysis 4

Leader

News

Craft 16

Products

Sales, contracts, appointments, biz bites and the bigger picture.

Headroom

Télévision Suisse

Broadcast

John Cornfield

André Jacquemin

A broadcaster’s move and move to a tapeless production environment puts its audio considerations to the fore. Top indie/alternative rock producer talks about Cornish bands, headphone mixes and Soundscapes. Gear decisions and keeping it small have benefited Monty Python and more than 50,000 commercials.

Sweet Spot

Discrete layered sound — an interesting new approach to surround sound recording and playback.

New introductions and announcements. Not alone and what’s wrong with Watts.

SoundField’s B-Format mic system provides a simple and integrated means of creating surround for broadcast.

Broadcast

Compaction and compression — why does compressed audio sound so annoying and what can be done to make it sound better?

Stereo pair and multimic mix comparison A rare opportunity to compare stereo pair and multimic mix versions of the same performance directly.

Ten

Katz’s column

Continuing his mission to bring power to the people, Bob indulges in acoustic empowerment.

Musicians who pushed recording boundaries.

Business 50

Brand, image and value

The traditional model for breaking a band is now largely outdated yet there are aspects of the old approach that are as crucial now as they were then.

Your business

Fame is a fleeting thing, especially if you’re a record producer. Daley asks if its pursuit is a viable career strategy.

Technology 60

The Crystal Core Engine

Fairlight’s new CC-1 Crystal Core Technology explained.

Reviews 22 25 26 28

CEDAR Cambridge V3 Earthworks TC25 Sony Vegas 7.0 + DVD Architect 4.0 SE Electronics Titan

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, Dan Daley, John Watkinson

Slaying Dragons

Watkinson suggest that shipyards may know more about sampling than the audio industry.

Groove Tubes The Glory Comp

Digidesign Mbox 2 Pro

Lexicon MX400

Cycling 74 Upmix

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

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

news Appointments

DPA directors: (l-r) Dan Ingemann Jensen; Ole Brosted Sorensen; Lars Friis Ostergaard; Jens-Jorn Stokholm; Morten Stove. DPA MICROPHONES has appointed Lars Friis Ostergaard as managing director. He joins from BB Electronics (Suzhou) Co Ltd, in China, where he was general manager and has also served as CEO for Glunz & Jensen A/S and as development manager at Bang & Olufsen. DPA has experienced annual growth of around 25% for the last three years and Ostergaard’s experience in steering national and international businesses during rapid periods of growth will be called into play. ‘DPA has grown to the size where we need someone with a fresh perspective combined with a lot of experience to come in at the top,’ said Morten Stove, who has assumed the role of marketing director. HARMAN PRO North America has appointed Nick Owen as vice president of sales for AKG Acoustics, Soundcraft and Studer in the US. Nick joined the Harman Pro Group in 2002 as director of sales Europe, Africa and the Middle East for Crown International and was formerly the founder of Soundscape Digital Technology.

Leader

So the world wide web is 15 years old and its inventor Tim Berners-Lee had no idea how it would develop and admits to being constantly astonished by the creative things that people do with it and the potential for misinformation. At the same time he believes that the study of the web should be formalised as a type of science so we are better able to understand its dynamic and regular changes of direction. It’s all sensible stuff as the web continually demonstrates an ability to take something very small and make it very big, very quickly when it is harnessed and exploited. In this issue (p50), Nigel Jopson looks at the role the web plays in establishing bands and highlights just how integrated the web has become in our lives. Christmas shoppers are likely to go online in a big way compared to just a year ago where most remained uncommitted because of security hang-ups. It’s about achieving the critical mass that paddles the rock along. Nigel also mentions that with more information than anyone can handle available the need for community becomes stronger. People like to clump around their tribal fires and certain types certainly like to warm themselves on the blaze from some of the pro-audio chat groups. However, tribal alliance does not exclude from the web’s laws of science and we’ll find as much misinformation in here as we will most anywhere else. The thing about the web is that anyone can build a website and put up any old load of nonsense in the knowledge that the ill-informed may read it and believe it. The printed word remains sacred to many because the ideas communicated are legitimised by the presentation. I’ll say straight away that there are some chat groups that are extremely focused and useful and there are others that I’ve heard good things about but have not been allowed to join — they must be superb. The ones I have a problem with are those in which the qualification to play is undefined, you can remain largely anonymous, and that are populated by a lot of very active individuals. The same group is likely to have some really unsophisticated manufacturer ‘plants’ and be brimming with opinionated high-flyer music production types who I suspect have to break from pronouncing on the relative merits of the ‘early years’ M 49 as opposed to the ‘newer’ ones to pop downstairs when their mum calls them for their scrambled eggs on toast. It’s also around ten replies to the thread before the abuse and finger pointing breaks out. I can go to a pub if I want to see people I don’t know arguing about something they don’t really understand and I don’t care about. There is a serious side to this because in an age where a lot of audio production is carried out solo without interaction with peers and other influencers, these chat groups represent community and that’s unrepresentative and a little sad. I don’t like the idea of anyone coming up thinking that’s all there is to it. Zenon Schoepe

Cine-Impuls buys first Fairlight CC-1

Fletcher and Unity’s Kevin Walker. TFPRO HAS appointed Unity Audio as exclusive international distributor for its range of professional audio products. Ted Fletcher, the founder and designer of all TFPro products, has some 50 years of audio design experience that includes Alice Mixers and the JoeMeek.

©2006 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

Berlin television production company Cine-Impuls has bought the world’s first CC-1-powered Fairlight system – a 5-bay Constellation-XT console with built-in DAW. The news coincided with the launch of Fairlight’s Field Programmable Gate Array Crystal Core (CC-1) digital media engine (see p60) at the AES in San Francisco. Cine-Impuls started using Fairlight eight years ago when as a film postproduction facility it purchased a Prodigy. ‘Our clients are always surprised about the good

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.

quality and sound of our productions and the smooth and easy workflow in our studios. Now it is time to invest in a new Fairlight,’ said Clemens Grulich, technical director of Cine-Impuls. ‘We are moving to multichannel surround production and want to fascinate our clients once again with Fairlight’s cutting edge technology, integration and sound quality.’ • Soundfirm, Australia’s largest postproduction company with facilities in Sydney and Melbourne, has added a fifth Pyxis NLV.

S2 Publications Ltd. Registered in England and Wales. Company number: 4375084. Registered office: Equity House, 128-136 High Street, Edgware, Middlesex HA8 7TT.

resolution

AES SF shows 8% growth

According to the organisers, October’s AES in San Francisco attracted 17,445 attendees and 420 exhibitors; equating to 8% growth over the last time the event was held in the city. ‘Exhibitors seemed very happy with the floor traffic, and many attribute the high number of end-user attendees to the success of our second consecutive Retail Promotion Co-Sponsorship which doubled its number of participants over last year,’ said executive director Roger Furness. ‘San Francisco is clearly a favoured destination city. We booked a record 2000 hotel rooms for this show. ‘By concentrating Exhibition Floor activity to three days, exhibitors were able to take full advantage of Convention educational and technical sessions,’ he added. ‘The final results of this experiment are still being assessed, but initial response has been extremely favourable.’ The next US AES will be held at New York’s Jacob Javitts Center 5-8 October 2007. The European AES with be held in Vienna 5-8 May. Bob Katz has been awarded a patent for his circuit design ‘Process for Enhancing the Existing Ambience, Imaging, Depth, Clarity and Spaciousness of Sound Recordings’. Better known as KStereo and K-Surround, these processes recover lost or amplify hidden ambience, space, and imaging, and generate stereo from mono signals without adding artificial reverberation. The K-Stereo and K-Surround have been licensed to Weiss Engineering and Z Systems for use in the Weiss DNA-1 and the Z Systems z-K6 Stereo-to-Surround Processor.

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November/December 2006

news Tacet rental takes LRX2

Jayaraj takes MC for scores

MI7 HAS been appointed distributor for Audient products in Sweden, Norway and Denmark.

Digital s o u n d equipment h i r e company Tacet has purchased two SADiE LRX2 LAWs for its expanding rental business. The company specialises in renting equipment for feature film and TV drama and documentaries. ‘It is a really exciting tool, a great and compact multitrack recorder that is the size of a telephone directory opened up with a laptop on top, which is used as a screen and to fire controls at the LRX2,’ said Tacet’s Simon Clark. ‘The SADiE unit does all of the DSP and our experience so far is that it has been rock solid. It is also surprisingly cheap to rent — but then we would say that, wouldn’t we!’

CHARTEROAK ACOUSTIC Devices has added Continental Far East Inc (Japan) and Full Compass Systems (USA) to its distribution network. CharterOak microphones are distributed in 11 countries and include eight independent audio dealers in the US.

Observe’s SoundField for HD OBs

Ireland’s leading independent outside broadcast company, Observe, has bought a SoundField SPS422B mic for use on OBs. Observe handles OB events for UK and Irish broadcast clients and launched its first HD OB unit in the Spring. ‘I like the SoundField, its audio quality and how easy it is to set up,’ said Observe’s head of sound Colm Flynn (pictured). ‘We have the SPS422B at present, but I’m looking at upgrading to the DSF-2 digital mic [see p52]. We’re covering some of the Rugby Heineken Cup for Sky Sports in Ireland, and they recommended the DSF-2, as they’ve been using it for UK Premiership football coverage. The 422B’s ability to output in stereo and 5.1 is of great interest to us, but the DSF-2 can do that and also operate at distances of over a kilometre from its OB truck, which could be really useful for us. We often can’t get our trucks very close to events.’

Appointments

Leading Indian film composer Harris Jayaraj has taken delivery of a Euphonix MC media application controller to interface with his Pro Tools and Logic Pro. ‘Logic Pro is my central workstation application on which I write, record, and program MIDI keyboard tracks for my film scores, and I use Pro Tools for mixing and mastering my soundtracks,’ said Jayaraj. ‘Hence, the ability to conveniently move back and forth between these two applications is critical to the way I work. With the MC’s ability to sense these applications and reset itself to match, the process of alternating back and forth is greatly simplified and enables me to focus on the creative task at hand — not the mechanical process of switching between applications. ‘The interaction of EuCon makes controlling Logic with the MC faster and easier than any other controller I looked at,’ he added. ‘The learning curve is almost non-existent and I had no need whatsoever for the manuals. Since the MC can interface with virtually any application, I’ve also been able to control Pro Tools, browse the Internet, check email, and edit video.’

MTV Networks is US mc266 first MTV Networks has installed a 56-fader Lawo mc²66 in its 48-foot Mobile Unit 8 truck, which is used for live broadcasts, taped performances, and DVD and CD productions. As one of the leading trucks in the US for award shows and multiband performances, it is also the first mc²66 in the country. ‘Mobile Unit 8 has mixed over 150 major award shows and thousands of concert performances to date,’ said Marc Repp, MTVN staff recording engineer. ‘We decided on Lawo after looking at every other multitrack console in the market. The Lawo sounded as good or

better than their competitors, it felt solid, demonstrated impressive engineering, and as a company appeared to have the most stable and forward thinking infrastructure.’ The MTV Networks desk is equipped with an HD core that supplies 256 DSP channels and a routing matrix with a capacity of 4,096 x 4,096 crosspoints, all at 96kHz. It includes stage boxes of 112 mic preamps. The supplied console includes redundancy of the DSP, router, the port-redundant MADI connections to all Dallis frames and the PSU and includes a 24-fader backup control surface.

Chandler Collison, Aviom; Kevin Madden, Innovason; Xavier Pion, Innovason. INNOVASON HAS entered into a partnership with Aviom in which the Innovason Dio module, DioAV, which is installed on the DSP board of the Sy48 and Sy80 consoles offers direct connectivity via two Cat5 connectors to two 16-channel networks and Aviom’s Pro16 personal mixers via Aviom’s ANet proprietary protocol. C R O W N I N T E R N AT I O N A L has promoted Scott Potosky to vice president of engineering and Marc Kellom to vice president of marketing. Potosky has been with Crown for 18 years, most recently serving as a product development manager. Kellom has been with C ro w n s i n c e 1 9 9 4 , originally starting in the IQ Systems Group and most recently as product development manager. MARK HUMRICHOUSER has been hired as director of US sales at Shure. Most recently he was team manager for Professional Systems and the Music Industry at Sennheiser, following a number of years as a regional sales manager and regional market development manager.

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November/December 2006

resolution

news Appointments

AWS 900 for Chemical Brothers

The new DTS theatre at it UK headquarters in Twyford is equipped with Procella Audio speakers powered by Yamaha XP Series amplifiers and two DME64 digital mix engines connected via CobraNet for processing. The DME64s take the 7.1 outputs from DTS decoders and splits and processes these into 20 discrete speaker channels, applying crossover, equalisation and delays. The theatre can run at 96kHz and can store and recall scenes in the DME64 for switching from 35mm to D-cinema or home theatre sources such as DVD, HD-DVD or Bluray.

ANTARES AUTO-TUNE is now being distributed in the UK by Sonic 8 Distribution.

ALLEN & HEATH has appointed Shidco to manage the sales, distribution and service of its products in Iran. Asimetrik in Turkey re c e i v e d A l l e n & Heath’s award for Most Improved Territory in Europe (pictured), while EAD from China was awarded Most Improved Territory in Asia. Most Improved Territory in Latin America went to Masteraudio from Argentina, and Canada’s Erikson Pro received Most Improved Territory in North America. American Music & Sound was presented with Distributor of the Year.

Rowlands and Clarke. UK production duo The Chemical Brothers have installed an SSL AWS 900 in their private studio in Sussex. Designed by Recording Architecture, the studio was originally equipped with a Mackie console and contains vintage synthesisers and outboard. Nick Clarke, of Tech Nick Clarke Ltd, installed the console for Tom Rowlands and Ed Simons, who are The Chemical Brothers, and engineer Steve Dubbs. ‘We are really enjoying using the AWS 900. It’s a great desk because it gives us all the features of a high-end console at a reasonable price. Also, importantly for us, it has a small footprint,’ said Rowland. ‘It’s perfect for musicians and producers who want to get a high-end sound in a private studio. We love the EQ, mic preamps and dynamics, and think the whole feel of the faders and pots are great.’ • Mix engineer/producer Mark ‘Spike’ Stent acquired six XLogic E Signature units for a Maroon 5 project at Conway Studios in Los Angeles with co-producer Mike Elizondo. ‘I’m used to tracking with SSL consoles,’ he explained, ‘and we were using another desk. I took these E Signature channels and the results were outstanding. ‘The E Signature is amazing on the kick and snare, ambient mics and things like that,’ he said. Stent, who mixes on a G Series in the Mix Suite, his private room at Olympic Studios, is also using the XLogic Multichannel Compressor, which he has been running as two independent stereo devices.

Genex DSD mix engine

(l-r) Klaus Seitz (director sales export, EVI Audio); Gilbert Pernin (Pilote Films); Nico Lewis (sales manager, EVI Audio); Bettina Hülsmeyer (Pilote Films). FRENCH DISTRIBUTOR Pilote Films has won Telex/RTS’s Distributor of the Year award. ON-AIR SYSTEMS has appointed Eugene Cooke as sales and marketing d i re c t o r a n d T i m Rawlings as regional sales manager. Cooke previously held sales positions at Snell and Wilcox, Maddox and Megahertz. Rawlings previously worked for JVC, For-A and Metro.

Yamaha powers DTS preview theatre

Genex Audio has us to provide this launched the first technology to fully featured DSD market.’ Mix Engine, the ‘Oxford Digital Mix+. The Mix Engine Limited is a centre of exploits technology excellence for various exclusively licensed audio areas including from Oxford Digital, DSD processing,’ a spin off of Sony Proadded Ayataka Audio Lab, Oxford Nishio, senior systems and allows the user e n g i n e e r, S o n y to mix DSD projects Corporation. ‘It’s nice with no conversion. to see the end result of The Smart AV console (l-r): Kevin Brown, Genex; Nishio; this work in commercial communicates with Richards. products.’ the Mix Engine via Ethernet using the OSC The DSD mix engine is a 2U rackmount protocol and all Mix+ functions can be equipped with 56 channels into 16 accessed by the controller. configurable buses and features Oxford ‘This is a truly exciting development, the Digital EQ and Dynamics on all channels. realisation of the final part in the end-to-end A Waves effects processor is included with native DSD production chain to provide the system, an d routing capabilities at the the ultimate in quality,’ said John Richards, input and output offer flexibility. The system CEO of Oxford Digital. ‘It’s the product of can also be controlled via TCP/IP allowing a lot of work that we did prior to spinning the system to be operated from a remote out of Sony and Sony have authorised location with monitoring returns.

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(l-r):Andy Reardon and Dennis Georgiou, First Network; Steve King, LMC Audio; Karl Christmas, Yamaha. Middlesex based audio-visual and corporate event specialist First Direct has taken delivery of the 150th Yamaha M7CL console to be sold in the UK.

Producer and engineer Jim Lowe, who has worked with the Charlatans and Stereophonics, relies on a pair of SE Gemini microphones as an essential for his trade. ‘I bought them because I just heard on the grapevine there were these valve mics called Gemini that were great value and sounded great,’ he explained. ‘I’ve been using them as overhead mics and as room mics on drums, which I did on the last Charlatans and Stereophonics albums. I have also used them on vocals. They have a reasonably bright edge, but a great clarity. They are a terrific value valve mic. They have added another sonic dimension to the sound already in my head.’

November/December 2006

calrec.com

MIX 1 PART ACTION WITH 5.1 PARTS PASSION

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

Putting Sound in the Picture

news Appointments

Soundworks takes largest Icon

SHANE FELIX has j o i n e d Wo h l e r a s marketing manager from Tut Systems, a leader in IPTV headends and digital video networks. Prior to that she held positions in marketing management with ASP Computers, Everex Systems and as an independent marketing consultant. John Sheridan has joined Wohler as senior applications engineer from ClearCom where he was customer service manager and applications engineer. Prior to that he spent ten years as a television production audio mixer in live remote broadcasts. RUSS BERGER Design Group (RBDG), the designers o f re c o rd i n g a n d broadcast studios, has added Chris Long and Stephen Wettermark to its staff. Long works in project development and worked previously in the live audio and recording industries. Wettermark is a new architectural designer with the firm and will work closely with Long and the rest of the RBDG staff to design facilities. RBDG has added architectural/ acoustical designer Zühre Sü to its project team. With experience in commercial and residential design, Zühre has a master’s degree in architecture and acoustics.

Soundworks in Cardiff has installed the UK’s largest Digidesign Icon — a 48-fader console with a Pro Tools HD5 Accel — in Studio 1 to partner the 16-fader D-Control already in Studio 2. Both were supplied by Scrub. ‘We’re committed to offering our film clients the best possible facilities here in Cardiff,’ said director and rerecording mixer Simon Jones. ‘Features and flexibility made Icon the console we wanted at any price but the relatively low cost meant that we could spec a big, 48-fader console which is important to give film sound mixers the visual feedback they need on larger projects.’ Fitzrovia Post in London has installed its first Pro Tools system through Scrub with one of its three dubbing studios now home to an HD3 Accel with an 8-fader D-Command.

Sinatra sessions at Angel

THE AMPCO/ FLASHLIGHT Group has appointed Dick van Berkum as MD. Van Berkum, a chartered accountant, began working for AFG in February 2006 as financial director with a growing involvement in the group’s management. FOCUSRITE HAS appointed Damian Hawley as UK manager for the Focusrite, Novation and KRK brands. He joined Focusrite as sales executive for UK South in 2004 and previously worked at JP Morgan and Synthax UK. S W I T C H C R A F T H A S appointed Sofia-based AT Company to stock and supply its connectors, patchbays and cable assemblies to the broadcast and professional audio markets in Bulgaria.

Angel Studios’ Steve Price was booked by Frank Sinatra Enterprises to rerecord some of Sinatra’s songs with new big band arrangements in Angel Studio 3. In 1957, Sinatra had the idea of doing a TV show in which he would sing all his favourite songs to camera with just piano accompaniment and the Nelson Riddle orchestra would be added afterwards ensuring perfect audio separation between

voice and orchestra. The show never materialised but the vocal takes were stored in the archives of The Sinatra Company. Charles Pignone, VP of Sinatra Enterprises, has now taken the performances and added the original Nelson Riddle arrangements in digital in London with producer Ken Barnes. The five sessions, which took place over two days, resulted in 28 ‘new’ Frank Sinatra recordings.

resolution

Holophone on the 7.1 effects trail

Wave Recording Studios’ Project Harvest, which aims to amass a comprehensive 7.1 sound effects library, is employing a Holophone H2-Pro surround microphone in acquisition. The project will take UK engineers Joss Gardner and Craig Loftus around the world, recording sound effects in real-time and real-life situations. ‘We selected the H2-Pro because it provides superior 7.1 recordings without the need for complicated set-up,’ explained Gardner. ‘You can just plug it in and know that it will work when you need it to -– in whatever situation you throw at it.’ The H2-Pro is being used in conjunction with a Zaxcom Deva V hard disk recorder at 24/96. ‘The sound produced by the H2-Pro is clear and the surround image is excellent,’ continued Gardner. ‘It has also survived being lugged up mountains, punched by soccer fans and travelling in the van with us.’

China dub

The Dubbing Centre in the Xinjiang Uyghur Autonomous Province in China, which handles the dubbing of programmes from Mandarin Chinese into the language of the local Kazakh and Uyghur population, has completed a refit and a move into new premises on five floors of a 21-floor skyscraper in Ürümqi, the capital of the province. As part of a ‘state-driven’ move to phase out tape-based production broadcast facilities, the Dubbing Centre has elected to employ a total of 28 networked Pyramix systems with Mykerinos boards for recording, editing and dubbing together with 10 native systems. The new Centre estimates that the system will increase its output of dubbed programmes to 2500 annually, increasing to 3000 next year.

November/December 2006

news The Big Picture BIZ BITES — During the first six months of 2006, writes Nigel Jopson, global shipments of all physical sound carriers shrank by 10% world-wide to a trade value of $8.4bn, down 4% from the corresponding period in 2005. The value of the digital music sector in the first half rose to $945m, up 106% compared to the same period in 2005. Senior label executives finally accept that physical distribution has a limited future. EMI Music chairman and chief executive Alain Levy (pictured) admitted the CD in its current form is finished, while highlighting multimedia physical releases. ‘The CD as it is now is dead, but a new version with added value will live on,’ Levy said during a recent discussion at the London Business School. ‘By the beginning of next year, none of our CDs will come without added value of some sort.’ EMI has been having a torrid time of late. The corporate’s latest trading update shows the company expects to report a decline in total revenues of approximately 3% for the six months to 30 September 2006. EMI attributes the year-on-year decline to a release schedule weighted towards the second half, and expects to deliver results in line with expectations for the full year. EMI’s trading numbers will be the first since the collapse of take-over talks with Warner Music in July. In South America EMI has suspended several key executives after uncovering a significant accounting fraud in its Brazilian business. The company’s present assessment is that this fraud has resulted in the overstatement of EMI’s revenues by approximately £12m and its operating profits by approximately £9m. ‘We’d be buying into this cock up,’ said Simon Wallis of broker Collins Stewart. Google snapped up YouTube earlier this month for $1.65bn (£884m) and says it will stamp out any copyright violations when it takes over the videosharing network. YouTube deleted more than 30,000 illegal video clips from its site after several Japanese media

New studio shows The Way

Based near London Fields in East London, The Way provides media services, such as website design, production plus on-line marketing, for clients in the music industry but also includes a commercial recording studio as part of its aim to provide a ‘one-stop-shop’ to help music clients across all stages of the music production and marketing process. The four partners in the studio are chief engineer and producer Luke Buttery, Steve Abu Nab (studio management and media design), Lee Ellis (media design, 3D animation), and Robert Offen providing business management. ‘We aim to deliver a big studio sound without the need for big studio budgets,’ said Offen. ‘Our ability to offer an integrated and cost-effective service across every stage of the music production process will help our clients connect their music directly to their audience.’ Centrepiece of the studio is a Neve 8078 custom-made in 1978 for CBS Sony studios in Tokyo. The 40-input, 48 monitor, 24-bus board features 31105 mic pres and 32264a compressor/limiters and predates the standard production 8078s. It includes such things as gold switches, which were specified by Sony as upgrades. They record to Pro Tools HD2 Accel, a 3M M79 16-track and Radar 2 with monitoring through ATC SCM100s and ATC and Genelec nearfields. The live room is augmented by three large isolation booths and the design is by White Mark.

Music 4, a leading UK creator of custom and production music for clients such as ITV, BBC Radio 1 and Virgin Radio, relocated its studios and offices from Maidstone to central London and needed to create a studio facility that would work as a postproduction facility during the day and double up as a music mixing and master suite in the evenings. AKA Design worked with acoustic design consultancy Harris Grant Associates and provided custom furniture for the facility. ‘We worked closely with AKA’s Guy Wilson who helped us create stunning furniture for our reception area, studio control room and voice booth,’ said Music 4’s MD Sandy Beech.

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Opry upgrades to Neumann and Sennheiser

The Grand Ole Opry turned to Neumann and Sennheiser mics to round out an audio improvement initiative that began when Steve Gibson, music director/manager of Creative Services, joined the operation five years ago. Gibson was asked by Pete Fisher, Opry vice president and general manger, to use his experience in the recording industry to improve overall audio procedures. ‘There were practices in place that we felt could be improved upon with regard to broadcast audio, so we took a look at all aspects to improve the sound,’ said Gibson. ‘We literally looked at every element in the signal path from microphone, all the way out the door.’ ‘This front-end upgrade with Neumann and Sennheiser was the last piece of the puzzle,’ said King Williams, broadcast mix engineer for the Opry. ‘The Opry upgraded two audio consoles in 2003. We were looking for a different sound off our deck.’ The Opry stage consists of the upstage area reserved for the Grand Ole Opry band, the mid-stage line that is occupied by the piano, various guitars and their respective amplifiers, and finally the downstage area that is closest to the audience. The new microphones are featured at mid-stage and downstage with a Neumann BCM 104 broadcast microphone on the upright bass and a total of eight Neumann, (four BCM 104s and four KMS 104s) for main vocals and instruments. One Neumann BCM 104 called ‘Big Rhythm’ is used as a wild microphone that travels around the stage as needed. Sennheiser Evolution 609 silver microphones are used on guitar amplifiers downstage. A Sennheiser MD421 is also on the steel guitar and on guest guitar amplifiers. ‘As professionals, we have always associated the Neumann and Sennheiser nameplates with quality and durability,’ added Gibson. ‘We were looking for something that set us apart from other broadcast production environments in our city and other cities. The new microphones have just an overall broad, nice sound. The front-end is all important, so the more natural and less manipulation, the better.’

November/December 2006

news The Big Picture

DR Byen powers up

companies accused the site of copyright infringement. This issue goes right to the heart of such social networking communities. Universal Music has this month launched legal action against two video-sharing websites — Bolt. com and Grouper.com. More than 250,000 tracks of previously out-of-print recordings by European artists have been sold since the launch of Universal Music’s digital catalogue reissue programme earlier this year, validating those of us who predicted a rosy future for the under-used archives of majors. If social networking sites become the go-to destination for online searches of deep catalogue material, majors will lose a unique opportunity to monetise such content.

Being on the other side of the glass has never been easy: UK star Crazy Titch shot and killed his producer Richard Holmes because the knob-twiddler made the mistake of working with the upcoming garage star Shab Shah, who insulted Titch’s half brother Durrty Goodz with the lyric: ‘Over the years things change in the ’hood, I used to have a lot of respect for Durrty Goodz — either singer or song. Not no more.’ But fame has a pay back: the star was convicted because a teenage rap fan who witnessed the killing recognised him from a TV appearance.

SHOWTIME SBES, Birmingham ......15-16 November Interbee, Tokyo...........15-17 November Tonmeistertagung, Leipzig ........................16-19 November CES, Las Vegas ........ 8-11 January 2007 NAMM, Anaheim............ 18-21 January Integrated Systems Europe, Amsterdam ..................... 23-25 January CabSat, Dubai .......................6-8 March Musikmesse/Pro Light & Sound, Frankfurt ............................28-31 March NAB, Las Vegas ................... 14-19 April AES, Vienna ..............................5-8 May

Described as the biggest ground-up broadcast build in the world and serving as something of a pointer to where broadcast technology can go if it is given the luxury of a clean slate, Danish Radio’s DR Byen site in Copenhagen has gone live in part with the remaining buildings expected to be completed and commissioned by the middle of next year. The site houses four identifiable blocks dedicated to sections of DR’s output with shared facilities and elaborate walkways and canals planned and being built together with the unprecedented inclusion of large areas for the public to relax and be entertained in. It is a complex of many firsts not least the widescale adoption of Studer OnAir 3000 desks dotted throughout the Radio operation. The 92 consoles vary in size between six and 39 faders and are linked in parts by IP control and realise widescale distributed control and interconnection. Integral to operation is the Studer Call Management System, which offers a comprehensive telephone interfacing communication system and permits control via a single GUI. It replaces telephone or ISDN-hybrid solutions Concert Hall. and offers Voice over IP and can also offer enhanced features, such as real-time caller statistics. ‘The design of the OnAir 3000 is very user-friendly and offers a lot of functionality in a compact design,’ explained Lars Lomborg, DR’s general project manager for audio facilities. ‘We are able to use the same modules to build up different consoles, from small six-fader units to big 39-fader production consoles. Each console can be scaled up or down if we have a requirement for another size of desk in the facility.’ The jewel in the crown of the site will undoubtedly be the massive and incredibly ambitious Concert Hall that is still being built and will eventually be encased on all sides by huge projection screens that will serve as a stunning visual focus for the whole area. It will house a large main hall, rehearsal areas and production rooms equipped with a selection of SSL consoles.

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Bay Area studio complex opens

The 5,000 square-foot Red House Studios complex in the Bay Area, US is home to a recording facility, several well-equipped rehearsal suites, private lesson rooms, and a 100-plus capacity performance hall. ‘There have always been places where people can rent a warehouse space for a few hours,’ explained founder Raja Singh, ‘but I felt there was an opportunity to create more of a community -– a place to not only improve your chops, but play live, record a demo, and all while meeting and interacting with others.’ Singh brought together a team of engineers, producers and instructors, creating a supportive environment where, for a monthly membership fee, musicians can take private lessons, join group seminars, rehearse their band, jam with friends, or record a demo and play a live gig. ‘A significant portion of our membership are people in their 30s, 40s and 50s,’ said Singh. ‘They have families and busy careers, but still want to play their instruments and need to do it on their schedule, and in a classy environment.’ The Music Lounges are pre-equipped rehearsal and recording suites with a Mackie Onyx 1640 mixer and FireWire interface connected to a PowerMac G5 with a Mackie Universal Control Surface and SRM350 monitors. ‘The mixer is simple and intuitive, so our customers and staff can get up and running quickly, but it’s also a professional piece of gear that people are excited to use.’ explains Singh. ‘Plus, the breadth of recording options make it a perfect songwriting and preproduction board.’ Those wanting more serious studio time can book the recording studio equipped with Pro Tools|HD2 and a Mackie 24fader control surface, Big Knob, Onyx 800R preamp and a pair of Mackie HR824 monitors. The Performance Hall is equipped with an Onyx 3280 32channel mixer and S400 Series passive loudspeakers.

November/December 2006

facility

Télévision Suisse A broadcaster’s part relocation and part temporary rehousing has given it the clear opportunity, that all would want, to take the leap to tapeless. ZENON SCHOEPE visits Geneva to see how the audio contingent has managed.

HE URGE TO GO tapeless is a strong one among broadcasters but like getting married or having children there is rarely a best time to do it. The upheaval and expense not to mention the mountains of forward planning for what are living 24-hour operations is enough to demoralise even the most vocal endorsee of the cause and encourage them to postpone the inevitable switchover. In an ideal world most broadcasters would want to have a completely new set of facilities built right alongside their old existing ones allowing for staff to flit between the two to follow progress and train and then finally to turn up for work at the new building when all is complete and working beautifully. Failing that you could just have the whole decision forced upon you so you just have to do it. Télévision Suisse’s headquarters in Geneva features a large tower that housed a good proportion of its facilities and offices. Trouble was it had been built with a lot of asbestos in it. Fortunately there are other adjoining buildings and enough other real estate in the city for things, departments and people to be moved around while the tower is being cleaned out, floor by floor, but the upheaval has been immense. The work that has started will last possibly four years and will eventually see all the office staff able to return to the tower. At the same time it was the opportunity to rethink the whole broadcast infrastructure because all departments were effected by the disruption in one way or another. Télévision Suisse is going tapeless. 14

The broadcaster had its audio postproduction department in the tower in among the admin and programme production departments that were housed there. Former office space in a newer building alongside was allocated to the new sound editing and mixing facilities which naturally enough were built to be integrated with other departments by sitting on a

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large cross-disciplinary network. The postproduction department now houses the largest Merging Technologies installation of its type in broadcast and rings all the bells that have been hung on the Pyramix DAW for the last few years in preparation for just such an application. Two Luxor severs, ten Pyramix DAWs and ten VCubes nonlinear video systems link through their own network and employ Merging’s Virtual Transport control when appropriate. Designed by WSDG with PMC IB2 monitoring, the department has six sound design rooms and two multipurpose rooms that can be used for mixing or sound editing. There are also two large Studer Vista 7-equipped mixing rooms. The update was needed. The broadcaster’s live production areas with their four studios and SSL Aysis consoles were recently refurbished and a plan of replacement was underway in video editing and graphics. The old audio post rooms had run for some 15 years and had become something of a Dyaxis power-user, with six systems for sound editing plus Pro Tools stronghold, mixing. It was not the most efficient of arrangements as the Dyaxis was strapped for tracks and project transfers could involve a lot of Beta. The DAW choice was a crucial one, according to Thierry Bonvin, head of audio postproduction. ‘When we started the project we looked at the market and selected three editors and saw them all. The Pyramix was very well rated and accepted by the sound designers, a little less so by the mixers. There was also the consideration of connection as we could use MADI on Pyramix — not on Pro Tools — and that saves money and we were on a tight budget,’ he says. ‘There were also considerations about openness to different formats. When we started looking, MXF was not proposed by Digidesign. Avid was being looked at for the video editing so we were looking at the solution that Avid and Digidesign proposed but we were not keen on that. To go to MXF was part of the whole project so that’s another reason why we chose Pyramix.’ While individual departments will have their own local networks for their own workflows they will all sit on a larger network and the objective is to be able to place an MXF file on the system that can be used for playout. While the audio department is up and running with its network and server, the first phase of the installation of the main server should happen by the end of this year and permit communication

November/December 2006

facility

Patrick and Thierry.

between audio post, video editing and graphics. The final integration of the studios and the heavy postproduction should be completed by the end of next year. Télévision Suisse’s audio engineers move around according to the job and could be mixing in a multipurpose room one day, sound designing the next and mixing on a Vista the day after. You get the impression that variety is encouraged and it’s one of the reasons why the standardisation on Pyramix makes sense. The sound design rooms can perform the digitisation of the video into VCube and this can then travel with the project around the network. Similarly, audio editing can be continued into the mix -- Pyramix is not just being used as a playback device here. There’s also a 40,000 CD library of sound effects with dedicated staff to manage and distribute it. While there is a search engine of the effects titles and their location, the discs still have to be retrieved manually. There are plans to place a select sound library on the Luxor and make it available for everyone across the department. It’s a massive investment for a broadcaster that serves a small percentage of a small country. Technical manager Patrick Boehm believes the approach is

cautious but sensible. ‘A point that’s worth mentioning is that we didn’t really invest in video tape machines and it’s why we are still using Beta SPs that are more than ten years old,’ he says. ‘We didn’t make the transition to Beta D. This is finally the opportunity for us to throw away those machines!’ It’s a modification of the approach that many have adhered to in the past — stick with a technology that is adequate for the job for as long as possible, work it to death and then throw it away when you jump a generation. Patrick adds that the lack of a common file format has consistently hampered progress in audio and that MXF promises a degree of compatibility across disciplines that will make a real difference to a broadcaster’s operation. The full integration of all the constituent parts is still some way off and is dependent, to an extent, upon the building work that is going on. Local network sections are happening but switching over and phasing in the integration looks daunting. What is clear is that the openness of digital systems is now a crucial factor in progress. It’s interesting to remember that in the early days of DAWs, being proprietary was a real advantage because it allowed you to control your users and your market. With

the DAW becoming central to what goes on in any studio, its degree of openness helps it to maintain and protect that position. Closed audio systems tend to look unappealing in the context of very much bigger broadcast infrastructures. The last words go to Thierry who says that other than the arrival of bigger drives and faster processors it is hard to imagine what the DAW could do next. ‘What I do know is that it would have been much harder to make these sorts of decisions ten years ago because anything we would have chosen would have been an intermediate step,’ he says. ■

Contact TÉLÉVISION SUISSE, SWITZERLAND: Website: www.tsr.ch

November/December 2006

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review gear

Products Equipment introductions and announcements.

Rupert Neve Designs desk

RND’s 5088 is a 16-channel, fully discrete, eight bus, expandable console with delivery expected early in 2007. It incorporates custom-designed input and output transformers and is expandable in 16-channel sections for 64 or more channels. All input channels offer bus/line/tape input selection, transformer-coupled direct outputs, eight mix buses, eight auxes and 100mm faders. The monitor/ master section includes four effects returns, switchable selection of three control room monitor outputs, six monitor sources and AFL/PFL solo modes, as well as talkback, oscillator and 100mm master and stereo master faders. Metering is via two VUs and an optional meterbridge. The Portico 5014 Stereo Field Editor is designed for mixing, mastering and live sound engineers and allows users to control stereo ambience and spatial positioning while manipulating source material with its difference channel equaliser and insert. A width control enables the user to increase or reduce the width of a stereo image and adjust the level of ambience inherent in the recording. The depth control adjusts the spatial positioning of elements in the sound stage. The difference channel equaliser and insert create new opportunities to manipulate stereo signals. www.rupertneve.com

Audient goes Black

Audient’s Black Series is a modular rack system combining the uncoloured audio path normally associated with Audient with a palette of tonal colours provided by discrete and transformercoupled circuitry. The Black Pre has discrete Class A circuitry, transformer balanced mic and line inputs and unbalanced instrument/DI with a HMX Harmonic Sculpting feature plus variable highpass filter, polarity invert and 12segment metering. The Black EQ is 4-band with ‘Enhanced Slope’ HF/LF shelving, high-mid Presence/Absence, and a sweep low

Platform news: Digidesign Digidesign X-Form is described as ‘the pinnacle of time compression/expansion and pitch-shifting technology’. Available for Pro Tools and Avid DNA systems, this AudioSuite plug-in is said to deliver time stretching and formant-correct pitch shifting with professional results even at extreme settings. X-Form provides a choice of two time-stretching/pitch-shifting algorithms to give best results with a variety of source material. The Polyphonic algorithm is for use on complex audio files, such as chords or stereo mixes. When used with the Transient Sensitivity control, it allows you to compress or stretch drum loops and other percussive material without losing attack transients. The Monophonic algorithm is best suited to solo instruments and vocals. Developed by Digidesign’s Advanced Instrument Research group, Velvet is an RTAS instrument plug-in that delivers emulations of electric pianos and is based on the Fender Rhodes Suitcase, Fender Rhodes Mk I and Mk II Stage Pianos, and Wurlitzer A200. Velvet also includes emulations of tube overdrive, compression, EQ, wah/filter, chorus, tremolo, tape delay, and other effects. Its controls work like the originals, according to Peter Gorges, director of the AIR group. ‘The sound of a vintage piano isn’t just what you record at its audio output,’ he said. ‘It’s also shaped by the amplifier and speakers — or a preamp — and most sounds rely on effects such as chorus, phaser, wah, and tape delay.’ www.digidesign.com

Platform news: Steinberg Tailored to the needs of postproduction and its Nuendo system and developed in cooperation with CB Electronics, Steinberg has shown a new synchroniser. ‘We’re very excited to be developing a new HD video-compatible synchroniser for Nuendo together with Colin Broad of CB Electronics,’ said Lars Baumann, Steinberg’s senior product manager for Nuendo. ‘Despite the current trend towards accomplishing ever more tasks using video within DAWs, many post people still have a requirement to integrate their audio workstation with other devices. The new hardware will couple the open, innovative and scalable Nuendo DAW technology with Colin’s 25 years of experience and his outstanding technical know-how in the field of machine control.’ The new unit will be fitted with a range of physical interfaces for machine control and synchronisation, including support for LTC, MTC/MMC, Sony 9-Pin machine control, Sony 9-pin emulation, Word clock and GPIO. The synchroniser will be tightly integrated with Nuendo with a sample-accurate VST System Link connection. It will offer lock to AES-EBU and video references including trilevel sync from HD video devices. Two RS-422 ports will enable Nuendo to act as a 9-Pin controller or device. As a 9-pin controller, Nuendo will slave to LTC or VITC with a single serial connection. The hardware will interface with Nuendo via USB. www.steinberg.net

mid. Overtone gives LF band specific harmonic enhancement, Glow offers LF-band specific dynamic equalisation and Tilt shifts the overall tonal balance around 1kHz. The Black Comp Class A opto compressor has a VU meter switchable between gain reduction and output level, switched Ratio, Attack and Release controls, and an Auto Release function. Overcomp is a FETbased compression circuit for overcompression, Smooth switches to a dual-time constant mode for overall mix bus compression and modules can be linked for multichannel use. The Black ADC uses an AKM chipset offering up to 24-bit, 192kHz resolution, AES, SPDIF and Toslink outputs and 12-segment metering with OdBFS and Over LEDs. Discrete inputs allow the module to be used as a standalone convertor.

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The BR-10 Black Rack can accommodate ten Black Series modules and the last four slots can hold analogue or ADC modules. www.audient.co.uk

UA Desktop and 2-LA-2 Universal Audio’s Desktop Console System (DCS) initially includes the DCS remote preamp and DCS monitor master. The DCS modular family of products each combine a base-unit with a DCS-Link connected remote controller and can each be used individually or interconnected to form a multiunit console. Features of the remote preamp include dual mono or stereo transimpedance mic pre/DI, M&S recording with

November/December 2006

review gear decoded monitoring, headphone amp with 3 x cue mix, reverb and EQ, a digitally controlled analogue signal path via DCS-Link, and up to 300-foot remote location from the base station via Cat5 cable. The DCS monitor master (speaker and cue controller) has a mixing/monitoring/recording ‘console master section’, 100% digitally controlled analogue signal path via DCSLink, and five stereo inputs (three analogue, two digital) and five stereo outputs. It also has mono sum, dim, mute plus LR solo, phase and swap, talkback and a 24-bit, 192kHz D-AC.

The 2-LA-2 Twin T4 Levelling Amplifier is based on the LA-2A design and offers two linked or independent channels and two T4 optical compression cells per channel, greatly expanding the uses of the original LA-2A. Features include the sound and circuit of the original Teletronix LA-2A, matched slow and fast optical T4 modules, linked stereo or dual mono operation, custom UA transformers and ten valves. www.uaudio.com

V4 for DS-00 and D4 Following up on the Version 4 combined software and h a r d w a r e upgrade for the D5 Live, DiGiCo now has Version 4 for the Soundtracs DS00 and D4, which incorporates new MIDI control. DUI allows the user to assign a console’s banks to control external MIDI devices or DAWs through worksurface faders, pans or mutes together with automation. Blocks of eight channels can be called up on the console

touchscreen and allows the use of record buttons to arm tracks on the DS-00. The window section provides access to edit and mix windows to bring up transport controls on the DAW. V4 for the D5 Live includes extra configuration f l e x i b i l i t y, expanded busing and routing options, partial session loading, enhanced onboard effects and graphic equalisers, new snapshot features, additional monitoring functions, and augmented aux and group bus control. www.digiconsoles.com

Waves Mercury The Mercury Collection from Waves features 91 processors with more than 200 component plug-ins. It includes the entire Diamond bundle, the L-Series Ultramaximizers and Multimaximizers, the GTR Guitar Tool Rack, the complete 360 Surround Tools collection of 5.1 processors, Waves Tune pitch correction and DeBreath breath eliminator, IR1 and IR360 parametric convolution reverbs, Z-Noise dynamic noise reduction, the complete Renaissance series, Q-Clone, the X-Series of archive restoration tools, and the Transform Series. It also introduces the V-Series plug-ins developed by modelling classic British analogue audio components. The V-Comp’s compressor has fixed ratio settings, fixed attack time, adjustable recovery time and automatic gain make-up and a limiter also with adjustable recovery time including an auto setting and variable threshold. There’s also the V-EQ4 4-band EQ and V-EQ3 3-band EQ. The MaxxVolume plug-in combines technologies from the company’s L2 Ultramaximizer, C1 Parametric Compander, Renaissance Vox and Renaissance Compressor for music production and broadcasting. The Live Bundle for Digidesign’s D-Show and D-Show Profile consoles includes 27 processors including the L2, C4 MultiBand and MaxxBass, C1 Parametric Compander, S1 Stereo Imager and the entire Renaissance Series. www.waves.com

HDOE and HROE output expansion Millennia Music and M e d i a ’s H D O E a n d HROE output expansion options for the HV-3D and HV-3R 8-channel mic preamps provide an additional two buffered outputs per mic channel for a total of three outputs for each microphone. The expanded outputs are on a pair of DB-25 connectors wired to the Tascam format with eight channels on each connector. The extra outputs can be used to provide splits for live, broadcast and recording feeds without resorting to transformers. The output stages are identical to the HV-3’s. The HROE option is field-installable in the HV-3R remotecontrollable mic preamp. HV-3Ds can be ordered with the HDOE as a factory-installed option. Existing HV-3Ds can be returned to Millennia to have the option installed. www.mil-media.com

November/December 2006

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review gear More XLogics

SSL’s XLogic Alpha Channel combines a mic preamp, filter, EQ and limiter as a front-end to a workstation. Key features include a ‘Variable Harmonic Drive’ circuit, High Pass Filter with selectable 40Hz, 80Hz or 120Hz cut-off frequency, analogue to digital conversion, auto sample rate sensing and self adjustment, 3-band SSL parametric EQ with mid Q control and dual LF curves, plus auto limiting.

The XLogic Alpha-Link Audio I-O Product Range is a They´ll Grow 05|05|2006 15:03There Side collection of multichannel audio convertors. are 1 initially three Alpha Link models all with 24-channel SSL

enhanced A-D/D-A convertors and offering a choice of digital formats. The Alpha Link MADI SX is a MADI and AESEBU-based convertor, while the Alpha Link AX and Alpha Link MADI AX are ADAT Lightpipe-based convertors. All Alpha-Link units can be used as standalone format convertors, but used in combination with a Soundscape Mixpander PCI card they provide a flexible I-O solution for native PC-based audio workstations. When the unit is connected to a Mixpander the I-Os can be routed individually from the PC using Soundscape Mixer software. With the XLogic Delta Link MADI HD, routers, digital consoles and convertors can now be directly interfaced with Pro Tools|HD. One Delta Link MADI HD replaces up to eight Digidesign 192 digital I-

sound engineering

Os, delivering 256 channels of digital audio interconnection from a single hardware interface. MADI HD reduces cost by using a single cable interconnect and occupies less physical rack space. Following its acquisition of Sydec, the makers of Soundscape, SSL has released its Console EQ and Filters plug-in for the platform. The 4-band parametric EQ includes two distinct SSL EQ flavours, one based on the G series and the other based on the latest version of the E series EQ. The Soundscape EACM EQ|1 plug-in is the first member of a new plugin family only available for Soundscape DSP-based workstations. The EACM engine was created by Algorithmix and the Soundscape DSP platform delivers the processing required to run multiple instances of the algorithms. SSL has announced its ongoing support and direct participation in the development of Ardour, the open source DAW. www.solid-state-logic.com

Control 1 Pro JBL’s Control 1 Pro 2-way loudspeaker has been re-engineered to provide smoother frequency response and more even coverage. It will ship with a mounting bracket included. An updated crossover network provides steeper crossover slopes and SonicGuard overload protection has been enhanced to provide better protection of the transducers. Previous versions of the Control 1 sold more than one million units worldwide. www.jblpro.com

Snake enhancements

THEY WILL GROW ON YOU - BOOTH 1305 -

AES-EBU I-O boards for the RSS Digital Snake allow you to mix and match analogue and digital inputs and outputs in the same system, the output board having sample rate conversion and Word clock per module. Each module has four inputs. Version 1.5 software enables remote control of preamp gain, pad, phantom power and memories from a Yamaha M7 or PM series desk. A convertor from RS-232 to RS-422 is required. The AR series includes the AR-3000R, AR-200R and AR-200S CompactFlash recorders. Editing of waveforms and message assembly is integrated and control is via RS-232, direct connection, binary or MIDI and is available on all units. The AR-3000R can have its playlists updated remotely using TCP/IP from an optional AR-NT1 expansion board. www.rolandsystemsgroup.co.uk

[GREAT SOUNDING TUBES SINCE 1985] Ultrasone Edition 9

LYDKRAFT 18

www.tube-tech.com resolution

The US$1,500 Edition 9 headphones use Ultrasone’s finest titanium-plated drivers and use black chrome in the ear cups and nameplates. Ethiopian sheep leather is employed for the ear pads and headband pad and the cans come in a metal attaché case.

November/December 2006

review gear The Edition 9s use Ultrasone’s SLogic technology, which reduces sound pressure to the eardrums by 4 0 % a n d re d u c e s electromagnetic field radiation by up to 98%. www.ultrasoneusa.com

K + H O 300 monitor Klein + Hummel’s O 300 triamplified 3-way nearfield monitor delivers the same performance as the O 300 D without the digital input and control hardware. The waveguides are moulded from low-resonant LRIM while the elliptical shape of the high-frequency horn makes the dispersion angle in the vertical plane narrower reducing early reflections from the console surface. The unit has an 8-inch woofer with the midrange handled by a treated fabric dome with a three-inch voice coil. A oneinch high-frequency driver has a titanium/fabric dome. The O 300 is available in anthracite finish as standard with optional silver, white and custom colours also available. www.sennheiser.co.uk

Radar V3.38 V3.38 software for Radar adds 18 new features and more than 140 improvements. It includes streamlined FTP file transfers,

23.976 frame rate support for HD Video, 48Hz PAL Sync Mode, and one-step backup to DVD-R/CD-R. The 3.38 software upgrade is available for free download to current 3.3x users. Upgrade from 3.2x Radar software is US$149. www.izcorp.com

Dangerous D-Box Dangerous Music’s D-Box is a multipurpose hardware product for DAW users. The unit is designed for compact or mobile use and the 1U box combines eight channels of analogue summing and an integrated programmable monitor control section with analogue and digital inputs, on board 24 bit/96kHz D-AC, talkback, dual headphone amps, and a speaker switcher. The Dangerous Monitor ST and the Dangerous Monitor SR aim to solve monitoring problems associated with DAWbased recording and mixing in studio environments that don’t employ a traditional console. The Monitor ST is a remote controlbased input source and speaker switcher with integrated cue and talkback systems, including an onboard headphone power amplifier. The Monitor SR is a companion 1U expansion module providing 5.1 surround monitoring capability. Studios working in stereo can start with the ST and later expand to surround by adding the SR unit. With the Cat5 connected remote control unit, included with the Monitor ST, engineers can configure and control an entire system from the sweet spot. www.dangerousmusic.com

Camera slot-in receiver Sennheiser’s EK 3241 slot-in receiver for professional cameras has a switching bandwidth of 36MHz and frequencies that can be tuned in steps of 5kHz, making the true-diversity receiver the perfect partner for Sennheiser’s transmitter models SKM 5200 and SK 5212. With a choice of various adapter kits, the EK 3241 fits into the slots of Sony, Ikegami, Panasonic and Thomson cameras. The mini-receiver can be attached to other camera types in a special slot-in housing; the EK 3241 then gets its power from a battery adapter with battery packs. The receiver’s user interface is similar to that of the SK 5212 bodypack transmitter and displays battery status. It has a weatherproof and robust full-metal case. www.sennheiser.co.uk

Røde Podcaster mic

Based on Røde’s Broadcaster microphone, the Podcaster delivers cardioid response and connects to computer via USB. It comes with an onboard high-output headphone amp for direct monitoring purposes along with a dedicated volume control and a 3.5mm socket. A green LED indicates that the mic is active and ready for recording. Optional accessories include a suspension mount and an anglepoise table mounting arm. www.sourcedistribution.co.uk

the daddy. 2 channel JFET/tube stereo bus compressor “From vocals to bass guitar, the

“I cranked it to where the meters

“Many compressors struggle to preserve

1968 has a natural warmth with a

glowed with the attack of each note...

transients and overall tonality when asked to

classic American-style compression

it gave the bass a nice edgy growl

process a mix, but this one works beautifully...

character - think UA 1176 or even

that I haven’t been able to get with

in fact, this is probably the best-sounding

Fairchild 670.”

any other compressor.”

Drawmer compressor to date.”

George Shilling, Resolution

Mike Caffrey, Tape Op

Paul White, Sound On Sound

Tel: +44 (0) 1924 378669 • Email: info@drawmer.com • Web: www.drawmer.com

November/December 2006

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review gear A-T wireless focus

RF signals. The ATW-49CB Active Antenna Combiner Kit includes two active antenna combiners, each of which is a broadband device and combines two incoming RF signals into one outgoing RF signal. Following Audio-Technica’s enhancement of its 3000 Series UHF True Diversity Wireless Systems with the addition of automatic frequency scanning, the company is now offering the new ATW-T371 handheld cardioid condenser in its ATW-3171a system. Incorporating the Artist Series ATM710 element, the ATWT371 wireless microphone is designed to be handheld and has rugged metal construction, soft-touch controls, a multifunction LCD display and is powered by two AA batteries. www.audio-technica.co.uk

A-T’s wireless products range has been augmented with the launch of the affordable 700 Series. Comprising five different packages, the 700 Series includes the Vocal system with handheld mic/transmitter, the Presenter system with belt pack and lavaliere mic, the Guitar system including cable for guitar/belt pack hook-up, and the Active system that includes a headworn mic. For flexibility, the Options system comprises a basic belt pack and receiver with the customer able to select a microphone of choice from A-T’s Wireless Essential range. The 700 Series offers 8 selectable frequency-co-ordinated UHF channels, automatic frequency scanning with open channel selection at the touch of a button, and AudioCrowley and Tripp goes stereo Technica’s Tone Lock squelch system. Designed for use with Crowley and Tripp’s Recordist and the A-T 4000 and 5000 Recordist Ensemble Stereo Kit claim S e r i e s U H F w i re l e s s to ‘raise the bar for Blumlein ribbon receivers and ATW-A49 recording’. The company’s newest LPDA antennas, the ATWribbon microphone — The Recordist 49SP Active Antenna — is voiced for realistic reproduction Splitter Kit and ATWof classical and jazz performances 49CB Active Antenna and its high output, low noise design Combiner Kit may also be used with a variety of other doesn’t require special high gain antennas from A-T and other manufacturers. ribbon mic preamps. The ATW-49SP Active The Recordist comes with its own Antenna Splitter Kit low diffraction rotary mount for exact includes two active placement, a durable Pelican case antenna splitters, each and a three-year warranty. of which is a broadband The Recordist Ensemble Stereo Kit device that splits an consists of two matched Recordist mics, an extender bar, 1/2 page Horizontal H OLOPHONE A D incoming RF signal into and two low diffraction rotary mounts. 216mm wide x 125mm high R esolution two identical outgoing www.soundwaveresearch.com

Dugan Model E

Dan Dugan Sound Design’s Model E is an economy model that makes Dugan Speech System automatic mixing available to a wider range of applications. It is smaller than earlier Dugan products, fitting eight channels into a half rack width. The Model E connects to a mixing console’s insert jacks with standard TRS cables or through an 8-channel ADAT optical interface. When analogue I-O is used, the ADAT connectors can be used to link multiple units up to a maximum of 64 channels. The Model E can also be linked with the Models D-2 or D-3.

Mute and bypass buttons are provided for each channel. An internal web server downloads a Java applet that gives real-time control and monitoring of the controller from the user’s browser. The virtual control panel includes most of the Speech System controls that are on the Models D-2 and D-3. It can be operated locally or over a fast Internet connection. www.dandugan.com

Holophone Products Make Surround Sound Simple. Award Winning Products

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01.416.362.7790 November/December 2006

review gear Royer tube ribbon

Royer Labs’ R-122V is described as a compact, mono, bidirectional mic that the manufacturer claims is the first commercially produced vacuum tube ribbon mic. Available in a limited production run it uses the same patented large-ribbon transducer assembly found in the R-121 and R-122 active ribbons. The R-122V claims a flat frequency response and an output of -37dB, which is approximately 13dB hotter than an R-121. www.royerlabs.com

TFPro range

The TFPro range consists of the P38 VCA/optical stereo mastering compressor, P9 stereo switched 4-band mastering EQ, the M16 16-channel transformer based mic/line preamp, which can also be used as a 16:2 summing mixer with individual gain and pan controls, and the P10 Mighty Twin dual mono channel path mic/line/DI input stages with VCA/optical compressor and 4-band EQ. The P9 has high and low shelving bands, two parametric mids and high and low pass filters. All its controls are switched. www.unityaudio.co.uk

November/December 2006

AudioCube plugs for Tools

Cube-Tec has announced the availability of its AudioCube plug-ins for Pro Tools OSX (RTAS iLok). Previously available only f o r C u b e - Te c ’s AudioCube, Quadriga and Dobbin systems, the plug-ins will be released in stages starting with DeBuzz, Spectral DeHiss Expert, DeScratcher and DeCrackler and followed by Azimuth, RepairFilter, DeClipper, DePop, and a suite of mastering plug-ins. www.Cube-Tec.com

Streaming for workstations Source Elements’ Source-Connect plug-in is now available in VST. It offers the same high quality, low latency, real-time audio connections to a wide range of audio workstations as

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the original Pro Tools plug-in. Source-Connect eliminates the need for ISDN lines, hardware, subscription fees and minuteby-minute line charges, enabling affordable high-quality audio s t re a m i n g b e t w e e n s t u d i o s , voiceover talent and musicians all over the world. Incorporating a high-quality, low delay AAC codec, Source-Connect allows for communications at bit rates up to 192kbps stereo. Two types of streaming software cover Source-Connect for workstation-to-workstation collaboration and SourceLive for workstation to QuickTime. www.source-elements.com

VoicePro update

The V1.2 update to TC-Helicon’s VoicePro voice processor adds refined pitch detection and shifting. The pitch detection has been improved resulting in smoother pitch shifting in VirtuaLead and Harmony paths together with vibrato and inflection. Low frequency pitch detection is extended and the factory and preset banks have been extended to 275 presets each. TC’s UnWrap will soon be available for workstations. Originally developed for TC’s System 6000, UnWrap facilitates fast stereo to 5.1 up-conversion. Full versions of the plug-in will be released for PowerCore VST and Pro Tools|HD Accel. www.tcelectronic.com

review

CEDAR Cambridge V3 Restoration, cleaning up or the reconstitution of audio is a serious business be it in mastering, recording or forensics. User choice has certainly increased but if you’re really serious about it then this is the product you need to look at.

LACK MAGIC, WHITE MAGIC? Sometimes the dividing line is an indistinct and moving target. I’ve been rescuing dodgy audio for over 30 years from chopping clicks out with a razor blade and the Philips analogue declicker, all the way to today’s digital wonders. It’s an obsessive, esoteric arena. I wrote my first article about audio ‘restoration’ around 17 years ago and I still don’t have any definitive answers. What I do know is that the leading purveyor of digital tools to exponents of the art remains CEDAR Audio. As a variety of restoration algorithms have become generally available all manner of pretenders have arrived at the palace gates, promising alchemical miracles. Some are good, lots are not. There is a lot more to it than simply picking an algorithm out of the grimoire. Suitable cases for treatment can be divided into three broad categories; enhancing existing recordings for re-release, removing defects from original material for new productions, and forensic use. Priorities obviously differ across these areas. For remastering the most gentle touch must be applied with due regard for the original. Cleaning up material in the course of production can be more aggressive since other sounds, such as backgrounds and music, may well conceal artefacts. On the other hand, in forensic work, intelligibility is the principle criterion and the tools used to achieve this are often positively vicious. Here, provided that a transcriber can accurately interpret the speech, considerations of high fidelity are secondary. CEDAR offers a wide range of standalone, real-time processors addressing specific problems. Its flagship PC-based system, Cambridge, can be specified to suit any or all of these areas with a wide variety of processes supporting many strategies. The Version 3 demo system I had is a workstation class Cambridge ‘Q’ rackmount host PC equipped with two dual-core AMD Opteron processors. There is no proprietary DSP hardware. CEDAR optimises the Windows operating system at Registry level, therefore adding extra third-party software is definitely not encouraged. An RME card provides the audio I-O in this instance. All Cambridge systems include the TAC 22

ROB JAMES

(Timecode Automation Controller) and communicating with the PC via USB this 1U provides full automation of the CEDAR processes internally and when chase synchronised against external code. TAC is also a giant dongle, storing the license keys. In the Cambridge environment the Process Manager window is the command centre. From here you can select file or physical input and invoke the File Processor, the Set Up window and a variety of tools. This is also where processing modules are instantiated. This approach, with separate windows for logical components of the user interface, is well suited to the task. Windows can be arranged into a layout appropriate to the job in hand. The workspace thus created can be saved and any workspace can be set as the default for new sessions. Process Manager also relays information to the operator about CPU load, incoming sample rate, sample resolution and latency through the entire processing chain. The Channel Selector determines which channels or channel groups are selected for adjustment and also reflects values for each channel when the mouse pointer hovers over a control in a Process Module. Global Settings along with Channel groups are defined in the Set Up window. Automation is handled in the Event Manager window. An Event is a snapshot of all parameter values in every module loaded at the time the Event is created. Morphing between Events can be user determined from instantaneous up to the length of the entire Event. A timeline display in the File Processor window shows currently loaded audio files as waveforms and enables regions to be defined. Cambridge can handle up to eight simultaneous channels. Thus surround formats up to 7.1, or multiple files with any programme content and bit depth but the same sample rate, can be processed, for example, four stereo files at once. File, transport and render controls are also to be found here. Modules are added to the process chain by clicking on Insert Module and selecting a process. Clicking the blue block minimises/maximises the module’s control window while the On/Off ideogram duplicates the resolution

button function in the Module Control window. Signal flow is from left to right and clicking on a Module and dragging to a new position changes the process order. One thing Cambridge does not do is record. All processing is performed either on existing files or live on the soundcard inputs. CEDAR says that the majority of Cambridge systems are on networks and few users wish to record in the Cambridge environment. Properly set up, network file handling is completely transparent. Temporary copies of files are always created and these can be local or on a remote disk as can rendered output files. Rendering can be undertaken in real time for monitoring or in non realtime mode at up to 30 times faster. Cambridge is offered ‘a la carte’ and in four standard configurations aimed at Film, CD and DVD mastering and two levels of Forensic. Prices range from UK£6,250 to around £40,000 with a typical system coming in at around £20,000 (all plus VAT). In practice, the exact complement of modules supplied with even the ‘standard’ configurations varies according to the client’s requirements. Every Cambridge system has sample rate conversion and dither/noise shaping and ships with a number of core modules under the heading of Utilities. DC removal is normally the first step. Offering 12dB/ octave roll-off in the user defined range 1Hz - 20Hz, a display shows the extent of its action, if any, and which channels it is affecting. The exception to the rule is DeClipping. In this case removing DC offset first lessens the effectiveness of the process and DC removal, if required, should happen afterwards. The Gain module provides gain control, solo, solo in-place and muting. As with other controls this applies to all channels selected in the Channel Selector. Gain offsets can be achieved by simply deselecting channels you do not wish to affect. As with all other modules, multiple instances are allowed.

Metering offers four concurrent displays for each selected channel. Near instantaneous peak metering is shown by a horizontal bargraph with a fall timeconstant of around 1.4s. RMS is indicated by a cyan bar and numeric. Peak Hold (1) is a grey bar with a hold time of 1s. Peak Hold (2) is numeric and is retained until the window is closed or the value reset by clicking on it.

Spectrum analyser is a high precision (0.02Hz resolution) graphic display of the selected channel(s). In contrast, the FFTs often found in DAWs have a typical resolution of 50Hz. All graphs in Cambridge share a common control set for averaging modes, zooming and Markers. Once you’ve zoomed in far November/December 2006

review enough the display slows down to improve clarity. Markers are a very powerful Cambridge tool. A frequency identified and marked in the analyser can be used to set a parameter in a process module with a mouse click.

CEDAR processing continues to be the state of the art. 64-bit internal processing and meticulous software engineering endows freedom — freedom to use processes without fear of running out of headroom and tools to reduce the range for output. There is no real alternative. Lest this seem like too much praise, workstation users may find a few things about the user interface rather odd. Mostly these could be viewed as charming eccentricities. Within reason, whatever you do, the results can be heard, evaluated and modified without worrying about the limitations of the system. Given my film background, the favourite remains DNS, which still brings a smile at the miracles it can achieve. Linear EQ is highly desirable and, in the right circumstances, Debuzz-3, DeClickle-2 and NR-5 rewrite the definition of what is possible. If you need an audio sorcerer, CEDAR has the spells.

THE PROCESSES The last Utilities module, 2-Channel Tools, provides a general toolkit for adjusting and monitoring 2-channel signals. A goniometer and +30dB balance bargraph give insight into the image. The balance control offers reversible gain shift between the channels and the Auto button keeps the image centred. Output controls enable the module to produce sum, difference or M&S outputs. Use of two modules enables M&S effects, such as width control, to be achieved. There is no ‘magic bullet’. If this is your Holy Grail, you will always be disappointed. All clean up and restoration tools are necessarily a compromise. Notwithstanding, the quest continues. But, make no mistake, although the price can be steep there is no substitute for the real thing.

EQ — CEDAR offers two 64-bit 96kHz EQ processes for Cambridge, Precision EQ and Linear Phase EQ. Both can be automated. Precision EQ is perhaps more applicable to forensic work, with seriously deep notches and brick-wall filtering, although in restoration it can help correct excesses perpetrated with analogue EQ. Linear Phase EQ is something of a find. I discovered that both subtle and very heavy equalisation can be applied with none of the usual artefacts. The prodigious number of filter bands, two LF and two HF shelving

with 8 variable Q parametrics positively encourage experimentation. Multiple deep, narrow notches, which would result in horrible phase effects and even pre-echo with conventional EQs, are meat and drink to this CEDAR algorithm. Constant power mode means you don’t have to worry about overcooking it. — A suite of a Compressor, Upward Expander and Downward Expander. Positive and negative ‘readahead’ enables the dynamics to be highly transparent or punchy when required. Each process has flexible linking options and side-chain EQ. DYNAMICS

Linking is available.

LIMITER — This Limiter is very different from a very high ratio compressor, employing a different signal path to minimise side effects. PHASE CORRECTOR — Sorts out static and varying phase discrepancies between channels with little effort.

CLEANS UP YOUR SOUND. Clarity and detail of sound. That’s what Ultrasone professional headphones with S-Logic ™ technology deliver. And because our headphones are ergonomically designed for professional use, they are also very comfortable to wear. Exactly what you want to hear when you spend hours in the studio or out on location. Visit www.ultrasone.com / UK distribution: www.ultrasoneuk.com

November/December 2006

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review

DEBUZZ-3 — Buzz is defined as unwanted audio with many high-level, closely spaced harmonics. Debuzz3 provides full-bandwidth processing, removing components across the entire audio spectrum, and is capable of removing all manner of buzzes and hums with fundamental frequencies as high as 500Hz. It can also track buzz varying by small percentages. Unlike alternative strategies it retains wanted signal even at the buzz harmonic frequencies by analysing the dynamics of the signal and buzz. In the past, I’ve not been overly impressed by products purporting to remove buzz. DeBuzz-3 has changed that perception and the results I heard were almost uncanny.

DECLICKLE-2 — Early digital declickers and decracklers could and did damage the sound of brass instruments, solo violins and sometimes singing. Declickle 2 has an improved noise detector and interpolator and offers improved performance by retaining useful information during clicks and crackles.

samples. The analogue audio waveform shows ‘flat tops’ where the real signal has been destroyed and replaced by false samples at maximum amplitude. Declip enables the user to identify and remove most clipping in one pass. If the analysis window shows ‘hard’ vertical edges, the signal has been clipped at that sample value. Analogue clipping is less well defined with a multitude of causes and exhibits a more rounded profile. Having spent many fruitless hours attempting to manually redraw audio waveforms damaged by clipping, the results obtainable here are quite remarkable. DEHISS 3 — Best suited for use when the unwanted broadband noise is consistent and uncoloured and when it is not possible or there is no time to use NR-4 or NR-5. AUTO DEHISS — Embodies a more advanced algorithm. It has a unique ‘Auto’ mode that enables the software to determine the broadband noise content and remove it without introducing artefacts. Manual mode offers control over all parameters to fine-tune the process. NR-4 AND NR-5 — NR-4 is based on developments of the original CEDAR dehiss algorithms while NR-5 uses the more recent Auto DeHiss algorithm. They share an almost identical interface. NR-5 lacks the modelling and brightness pane. These algorithms are radically different. In most cases NR-5 will remove more noise with fewer artefacts. The exception is when the noise is buried deep within the wanted signal. Experimentation is the order of the day to determine which process does the better job. As with all broadband denoise processes the controls are highly interactive and it takes time and experience to achieve the best results. An accurate fingerprint is useful. However, the interface enables the fingerprint to be adjusted or created manually using the drawing tools and the Noise Reduction EQ. This re-shapes the noise content contained in

VINTAGE DECRACKLE — Uses the same algorithm as the award-winning CR-1 Decrackler. It will also remove some forms of buzz and amplitude distortion from material spanning cylinder recordings to live broadcasts with lighting buzz.

DECLIP — Clipping results in harsh distortion, increasing in intensity as clipping becomes heavier. In digital clipping the signal contains many false 24

the input signal and therefore allows the operator to concentrate the noise reduction in the range where the hiss is most intrusive, while leaving other areas relatively untouched. The Noise Free EQ acts only resolution

upon that part of the signal identified as wanted. This can be used to introduce a mild high frequency boost to counteract loss of perceived brightness. NR-5 can produce the best results I’ve ever heard from this class of device in the shortest possible time. DNS — The modern world is a noisy place. Even at the poles there is no guarantee a jet plane won’t happen by. CEDAR’s Academy Awardwinning DNS process splits the signal into a large number of well-defined bands. Digital filters analyse each band, then independently suppress the noise in each. The really clever bit is presenting all this power to the user via relatively few controls, making it simple and quick to use. Especially applicable to dialogue clean-up, DNS remains one of the most powerful tools ever offered for sound for picture clean up. The DNS defaults will seem strange to those brought up using Dolby Cat43a and Cat 430 processors. However, a moment taken to read the manual will reveal that the approach is necessarily different. In forensic applications, DNS will remove motor noise, eliminate electrical interference and help clean up recordings made in unfavourable acoustics with poor microphone positioning.

ADAPTIVE FILTER PACK — The four Adaptive Filters are designed specifically for use in forensic audio. Adaptive filters are primarily intended to address constant unwanted sound. The unique CEDAR feature is the ability to retain out of band material free of processing, e.g. Sibilants and fricatives, improving intelligibility and listenability. Practically, this means more accurate transcription with less fatigue to the transcriber. The Attack time controls the rate at which the filter is allowed to react to changes. Speech can radically change inside a 100ms frame while the unwanted material may not. Filter length is the duration of the audio used to determine the filter coefficients. As filter length increases so does filter precision, however, long filters use a lot of CPU power. Where a reference recording of the unwanted signal is available, e.g. a broadcast, the results are truly astonishing. But don’t even think about using adaptives for music or film. They’re just not designed for it. ■

PROS

Unbeatable performance; easy for novices but with the depth for experts.

CONS

Price, which won’t matter if you really need it; cannot do the impossible; not much else.

Contact CEDAR AUDIO, UK: Website: www.cedaraudio.com

November/December 2006

review

Earthworks TCS Think of a high quality, small diaphragm omni for general-purpose studio and location recording and the chances are your thoughts head over to Germany or Denmark. Steer them over the Atlantic, though, and other options are there for the taking explains JON THORNTON

ASED IN NEW HAMPSHIRE, and formed by dbx founder Dave Blackmer, Earthworks designs and builds a range of microphones, preamplifiers and monitors for the professional market. In addition to microphones designed for measurement purposes, the company has three broad ranges of mics — the QTC range, TC range and SR range. All three ranges are small diaphragm capacitors with the QTC and TC ranges (Quiet Time Coherent and Time Coherent respectively) being omnidirectional, and the SR range features a cardioid pattern. The numeric designator that follows indicates the extent of the microphone’s HF response — a TC50, for example, exhibits a useful response up to 50kHz. This makes sense of what is quite an extensive series of microphones, and helps place the pair of TC25s reviewed here into context. It’s clear from the accompanying literature that Earthworks views this model primarily as a studio tool — suggested applications are ‘drums, percussion, amplified instruments and loud sound effects’. The microphone itself is what I’d term a ‘proboscis’ type — with a standard width body tapering down to a very narrow snout, at the end of which sits the capsule protected by a mesh grille. The prepolarised diaphragm, coupled to a transformerless, FET-based output stage gives a quoted frequency response of 9Hz to 25kHz +/-3dB. In reality, the response is pretty linear between 20Hz and 15kHz on-axis, rolling off gently to –3dB at about 25kHz and similarly below 15Hz. Response 90 degrees offaxis gives an earlier, but shallower rolloff in the high frequencies, starting at about 9kHz. Earthworks’ philosophy is built around ensuring that the frequency response is as linear as possible and also ensuring accurate phase and transient response. The TC25, it transpires, was originally engineered as part of the company’s drum miking kit (comprising two TC25s for overheads and a cardioid pattern SR25 for the kick drum). With all of this in mind, kit overheads seemed to be a good first test. Initially set up as a spaced pair directed at the snare/hats and rack tom — first impressions explain the suggested applications listed above. You’d expect a small diaphragm design like this to be relatively noisy, but with even fairly conservative amounts of gain these are noticeably so. The quoted 27dB SPL (A weighted) equivalent noise seems fair, but you aren’t going to want to record quiet sources with these particular November/December 2006

models. Fortunately the drummer in question wasn’t lacking in either enthusiasm or ability in the volume stakes, and the TC25s delivered a nicely balanced fairly neutral performance. They delivered plenty of attack to the snare, toms and cymbals, while maintaining good levels of tonal detail. Perhaps the cymbals sounded slightly splashy in comparison to a pair of DPA4006s — there is a slight presence lift around 10kHz on-axis, and their relatively low sensitivity forced me to drop their height a little more than I would usually — but a sound with plenty of depth and focus. There’s some experimentation to be had with placement to get the best results — despite being omnis, tilting the microphones in these positions so that they were about 40 degrees off-axis to the top skins of the drums softened that splashiness nicely. Acoustic guitar was next on the agenda and again the best description of the sound is balanced and neutral, both with a single TC25 and a spaced pair — there’s a superb transient response and HF clarity here that seems to place the sound right in front of you. It’s a shame that on quieter passages the TC25’s self noise was starting to intrude slightly. On an upright piano, the story was similar — nice tonal balance, particularly in the bottom octaves, but plenty of bite and attack to the sound — but fairly tight placement was needed to minimise noise on some passages, which was something of a compromise. With both the acoustic guitar and the piano, the vote remained clearly with the DPAs used for comparison but on the kit it was less clear-cut. I actually liked the slightly more present sound of the TC25s here, which helped pull the overheads into the mix slightly better than the DPAs. For the sake of completeness I tried a single TC25 on a 1 x 15 bass cab. Care has to be taken here, as they are quite sensitive to air movement, and in this case I had to point the ‘snout’ of the mic in the opposite direction to the cab to stop the resulting clipping sound, but was rewarded with a very clean, almost DI-like sound, but with the slight compression that the cab lends. In summary, the TC25s are good and versatile performers, which seem to be quite tolerant in terms of positioning — although in all applications tweaking a placement could improve the sound, first attempts never produced anything that was other than eminently useable. For those concerned about the noise issue, you could do worse than check out the resolution

QTC30, which offers much better performance in this respect. It’s possibly unfair to compare the TC25s with the 4006s — a more useful comparison would be DPA’s 4090 or 4091, but these weren’t to hand. If you were considering these, then the TC25s would definitely be worthwhile auditioning. ■

PROS

Extended frequency response; great transient response; very tolerant in terms of positioning.

CONS

A little too noisy for quiet sources.

EXTRAS

The Sigma 6.2 and 6.3 monitors take Earthworks principles of time coherent response and impulse response and apply them to the playback process. The 6.3 is essentially the same monitor as the 6.2 but features dual woofers for higher output handling.

Contact EARTHWORKS,US: Website: www.earthworksaudio.com UK, Unity Audio: +44 1440 785843

review

Sony Vegas 7 + DVD Architect 4 Sony’s Media Production Toolset has matured nicely and while the headline feature set hasn’t changed that, much the all important ‘oily bits’ have dramatically changed for the better. ROB JAMES rolls up his sleeves.

HE LAST TIME I took a look at Sony’s Vegas ‘Media Production Toolset’ it was at version 5.0 and now on a fast machine with plenty of memory Vegas is a lot slicker. The editing model has always been attractive to audio people but, by the same token, it has always seemed a little alien to anyone brought up on Premiere, Edius, Final Cut or Media Composer. More good news comes with the price, thanks to the exchange rate and fierce competition the Vegas 7/DVD Architect 4 bundle is now just UK£424 (+ VAT). Boris Graffiti LTD titling software, and Red Giant Magic Bullet Movie Looks 65HD video effects are included along with a sample CD of Sony sound effects. The last major upgrade to V6 brought a comprehensive re-write of the rendering engine and changed the feel of Vegas almost beyond recognition. Version 7 is not a flashy upgrade with a raft of new whizzy features. Rather it is a consolidation with further performance improvements, tweaks that make life easier and improve workflow, and support for new formats. Unsurprisingly, given that this is a Sony product, there is now extensive support for Sony HD and SD XDCAM MXF files in both native and proxy formats. All HD and SD XDCAM compression types (MPEG HD, IMX 30, 40, 50 and DVCAM) at all frame rates in all aspect ratios; multichannel audio and essence markers can be used. For input there are options of i.LINK (FireWire) or Ethernet. Extensive conform options are provided for proxy files and output can be mastered to disk. Native HDV performance has also been improved. You can edit native transport streams or transcode to a variety of intermediate codecs. Import from DVD camcorder has been added along with export to Sony PSP (PlayStation Portable). On the audio front, the multichannel BWF files generated by location recorders, such as Sound Devices and Nagra, are now supported. Vegas 7 recognises the time-stamps and automatically synchronises multichannel files across the timeline. Window docking is more flexible with less restriction on which windows can be placed where. Window layouts can be quickly saved and recalled, which will improve workflow since there are a large number of tabbed windows in the Vegas interface and it is all too easy to inadvertently hide things. Now the user can design screen layouts suitable for specific tasks and

recall them at the touch of a key. Timeline previews can now be simultaneously viewed in both preview window and external monitor. There are new scaling options including Auto-Fit, which makes the best use of the available preview window real-estate. Integration with Cinescore is a real sign of the times. Whatever we might think about semi-automatically generated music it is undeniably useful when you need a quick link and Sony’s new Cinescore appears to be one of the easiest music generators to use. If you have Cinescore, Vegas detects this and enables music to be generated to fit the picture without leaving the Vegas environment. CD Audio Extraction uses Gracenote Music ID to obtain information about CD tracks including title, artist, and song name. The ‘snap’ function has been enhanced with new object options making it easier to precisely place objects in complex projects. You can snap to objects on any track and colour-codes now denote snap object type, marker, region, event, cursor and grid. It is also worth noting that Vegas has a very neat method of indicating and repairing loss of sync between audio and video clips. Automation curves can now be drawn with a brush and there is an ‘autothinning’ function to reduce the number of points to a minimum. With the full package installed (absolutely no dramas), I imported the rushes from a project recently completed on another machine using a rival editor. Without recourse to the manual the first three minutes

or so were back together again in very short order. No small feat since the material was acquired using a different codec (also present on the Vegas PC) in another manufacturer’s editor — compatibility indeed. Previews were good and on this Intel Core2 Duo extreme-based machine the transport dynamics are as fluid and fast as you could wish. Hitherto I’ve not been using Vegas regularly but this might now change. I’m certainly going to use it on a project or two to get the ‘in anger’ feel of it. I’m especially impressed by its compositing abilities. DVD Architect is now at Version 4 and includes 5.1 Dolby AC-3 encoding. The first major new feature is the ability to create and run scripts integral to the DVD design. This will allow interactivity to be built into projects for games and kiosk automation with pass-code protection. The timeline can be panned and cropped. Objects, such as buttons and graphics can be animated with keyframe control in menus. Scene/Chapter buttons and button markers can be added to the timeline and button ‘hot-spots’ can be added over video with control over position, duration and link. This allows conditional branching, for example so that the viewer can access more detailed information about a subject of interest. DVD-R dual-layer discs are now supported for increased playing times. I have been using DVD Architect almost exclusively for at least three years. Once you get your head around it, it is perfectly possible to knock out a quick DVD, say for giving to a composer, in less time than the supposedly more user-friendly and simple authoring packages. On the other hand, highly sophisticated results can be obtained with a bit of patience and ingenuity. The themes and button designs supplied leave a lot to be desired but you can do much better with a bit of effort and without getting into the complexities of XML scripting. Vegas is the only application I can think of in this class that is easily accessible by newcomers while offering the scope required for professional use. From a relatively slow start back in the days when it was owned by Sonic Foundry, Vegas has steadily gained adherents across the spectrum of users and applications from casual amateur to broadcast professional. DVD Architect is simply a great program and Vegas is certainly close. At this price it is a bit of a no-brainer. ■

PROS

Easy, fluid editing; strong media management; compositing: if your needs are modest, save yourself a fortune; DVD Architect.

CONS

Strange UI if you are familiar with other video editors; more effects and nested effects would be nice; DVD Architect themes and buttons need a makeover; DVD Architect not available separately.

Contact SONY, JAPAN Website: www.sonymediasoftware.com UK, SCV London: +44 208 418 0778

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November/December 2006

review

SE Electronics Titan At first glance, this Titan seems something of a misnomer as it’s actually quite a diminutive, ever so slightly squat looking microphone and seems positively miniscule when compared its sibling, the Gemini. But size, as they say, isn’t everything, according to an unapologetic JON THORNTON

T IS ACTUALLY the internals of the Titan that account for this mic’s appellation — namely a diaphragm made of titanium rather than the more usual gold-sputtered Mylar. This isn’t a new idea, indeed there are several pieces of microphone exotica that feature titanium diaphragms, most notably Neumann’s M150 and TLM50. The attraction of using this notoriously difficult-to-work material lies in its low mass and high rigidity — meaning that lighter, thinner diaphragms with corresponding improvements in transient response are possible. At £850 (+VAT) the Titan is a cheaper proposition than either of the two Neumann’s — but still one of SE’s most expensive offerings. The question is, does it deliver the goods? Overall build is typical of what we’ve come to expect from SE — good and constantly improving, if somewhat utilitarian. The light-bulb shaped housing is finished in the same matt grey used throughout the high-end of the range, and unscrews easily to reveal a fairly densely packed circuit board that seems well constructed and laid out. Electronics are FET-based, married to a dual-sided, centre-terminated capsule with those famous diaphragms. Switches provide a choice of omni, cardioid and fig-8 patterns, a -10dB pad, and a high-pass filter. The whole kit comes complete with an elastic suspension (which rather grandly has a ‘Titan’ name plate fixed to its front) and a microphone cable, all neatly packed in a hard case. Sitting on the end of one of SE’s own mic stands (see sidebar), the whole ensemble has a slightly alien retro look about it — strange but quite pleasing. Setting a female vocalist in front of the microphone to start with, first impressions are of quite a hard sounding microphone. Not harsh, although there is a hint of this with sibilance, but with a definite edge to it. Coupled with a very taut sounding bottom two octaves, and there’s a sense that the sound is ever so slightly unnatural. What is surprising is the degree to which the tonality changes when switching

Stand and deliver

Partly, I suspect, to deal with the not inconsiderable weight of its Gemini microphone, SE has started manufacturing its own boom stands. I first encountered a preproduction version when reviewing the Reflexion Filter (Resolution V5.5) and was less than impressed at the time. Mostly this was to do with the base, which looked like drum hardware grafted onto some rather over-engineered pieces of steel tube. The end result was somewhat less than stable, shall we say. I’m happy to report that the stand that arrived with the Titan is an altogether different proposition. It still has that industrial, ‘made from girders’ look about it, but the folding base is much more substantial and, critically, creates a much broader tripod. Everything about the stand, from the clutch, to the knurled wheels that lock off height and reach, look like they will probably outlive you and mechanical adjustments are made easily and firmly. Some nice touches are springs located within the telescoping sections, which give a little shock absorption should you inadvertently loosen off a section without keeping hold of it. Minor niggles are the design of the clutch, which makes it impossible to point the boom straight up vertically and the sheer weight of the whole assembly. However, there’s no doubting its ability to hold the heaviest microphone in your collection, even at maximum reach, with no trace of instability or droop. [SE makes three models of mic stand: the Stand 1 being looked at here for £135 (+VAT); the larger Stand 2 which offers a maximum extension of around 2.5m for £185 (+VAT); and the orchestral boom Stand 3 for £467 (+VAT).]

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polar patterns. The fig-8 pattern, for example, almost sounds like a completely different microphone, losing a lot of its edginess in the upper-mids. Off-axis response on the cardioid setting drops off the HF response quite dramatically — so care needs to be taken on vocals if used close up with singers who move around a lot. I suspect that this is a microphone that needs careful matching to a singer’s tonality — certainly with the female vocalist in question it wasn’t the most flattering match. Moving on, and intrigued by that edginess and tight sounding low frequency response, the Titan was put on kick drum duty for a fairly minimally miked jazz kit. Positioned just slightly in front of the shell and about 45 degrees off-axis, the result was a really nicely defined, punchy sound with plenty of air and low frequency extension, coupled with a really good attack. Not what you’d expect from that particular placement, but very nice indeed. Encouraged by this, electric guitar via a venerable old Roland JC120 was next on the agenda. Close in, and things started to sound very brittle, but moving the Titan further away, eventually settling on about a meter away, opened the sound out considerably — again preserving the attack to the sound but giving lots of body. There was a pattern emerging here and in many ways it was almost the exact opposite of another of SE’s line-up, the Icis. With this, I found that it really liked to be worked close-in, whereas the Titan seemed to give its best results when given a little more distance from source (Sounds like an interesting dual-mic combo. Ed). All of which makes the Titan hard to categorise. There’s no doubting that it has a very distinctive sonic character but in some cases this might prove a little too distinctive for anybody wanting a microphone for all seasons. ■

PROS

Good transient response; taut bass response; distinctive character.

CONS

A little too ‘hard’ sounding for some sources; might not be flexible enough if this was your only large diaphragm capacitor microphone.

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

November/December 2006

Pro Audio

ESSENTIALS

AKG’s new Perception 100 and 200 Large Diaphragm True Condenser Studio Microphones combine classic warmth and transparency with exceptional value for money.

The new Soundcraft Vi6™ Digital Live Sound Console incorporates Vistonics® II and FaderGlow™ technologies to deliver sophisticated digital mixing with new levels of control & intuitive operation.

The Lexicon® MX400 and MX400XL Dual Stereo/Surround Reverb Processors offer 4-in/4-out operation for live sound and studio applications through their intuitive front panel or ‘hardware plug-in’ control.

JBL LSR4300 Studio Monitors featuring RMC™ Room Mode Correction, JBL’s exclusive technology that analyses and corrects the response of each speaker for absolute accuracy at the mix position.

The dbx 162 SL Stereo Compressor/Limiter with AutoVelocity™ matches a stunning front panel design with equally impressive dynamic range, sonic clarity and ultra-low distortion.

The BSS DPR-402 Dual Compressor/De-Esser/Limiter features a double side chain and subtractor architecture for unrivalled flexibility in conventional applications, and unlimited effects.

www.harmanprouk.com

review

Groove Tubes The Glory Comp Few boxes can match the big, heavy and beautiful retro chic of the GT generation. We’re talking gain reduction here so GEORGE SHILLING goes to Heaven and comes back to tell us all about it.

HE GLORY COMP (GC) is a sister product to the ViPre microphone preamp. A similarly overengineered all-valve design, this new product is equally robust — it weighs a whopping 37lbs in old money, so care (and strength!) is required getting it out of the box. Side braces are provided to take some of the rear weight — these attach to 1U of rackspace above the 3U GC. A 1U space is also required below for ventilation — the unit uses about 95 Watts and relies on a large rear heatsink and vents to dissipate this. Groove Tubes founder Aspen Pittman is an unashamed valve obsessive (GT has even acquired the GE valve production line to produce its own tubes) and the GC contains seven valves to handle the programme audio. So, preposterously, 5U of rackspace is required for one mono compressor! The GC’s appearance is striking, with military-style build using huge black knobs with clear pointers, stiff toggle switches, useful rack handles and a super VU meter. The meter is accompanied by a control with positions for displaying Output at various levels, Program Input, Gain Reduction with the option of a double scale for checking heavy compression amounts, Side Chain, and a setting that turns on the internal Oscillator for balancing the large Plate and Cathode trim pots on the front panel. This knob, along with many other rotary controls, is stiffly switched, while Input, Output and Glory knobs have a nice oily damped feel. Conventional controls are provided, along with one or two (such as Glory) that require explanation. But layout is logical and clear, with boldly etched labels and legending. First, there is the large Input pot, which is great for fine adjustment and ranges from -20 to +10dB. The Threshold knob is switched from 0dB downwards in 3dB steps. Fine tuning can be obtained by adjusting the Input to drive the compressor input, and I generally found I had to turn the input up past zero for a reasonable amount of compression. The unit has a huge amount of headroom, so this isn’t a problem. The GC uses a ‘variable transconductance tube gain stage’ for signal compression. This uses an own-brand valve (easily available at reasonable cost) that provides clean gain reduction up to 20dB in a single valve. And it does this in a smooth linear fashion, lending the GC an unusually suave character for a valve compressor. Compression is frequently quite subtle in character, such is the smoothness of the GC. The knee is such that the selected Ratio is reached 2dB above the 30

Threshold set. Attack is stepped from 1ms up to a quarter of a second, the fastest setting is remarkable for a valve unit. Release ranges from 50ms to 2s. However, there is an accompanying Release Mode toggle switch, and the labelled range refers to Log mode. In Linear mode, the range is 25ms to 1s and this provides a snappier release character, rather than the tapering off of Log mode. The wide range of settings means the GC works well with all kinds of programme material. Ratio too is stepped from 1:1 up to 6:1 with plenty of choice in the lower orders. On the far top right the large Output pot has a similar range to the Input. The output circuitry is incredibly powerful, allowing full use of all the dynamic range available at the input to any DAW. Bypass separately hard-wires the input and output XLRs plus the input and output TRS jacks. There are CV linking sockets for chaining multiple units, with a toggle on the front for selecting Master or Slave status, plus a useful Local position for removing the unit from the chain without having to unplug. Two GCs would undoubtedly make a superb stereo mix compressor, and a 5.1 arrangement would undoubtedly be something to behold, with 30U required rackspace (plus the reinforced floor and personal trussing. Ed). There is even a Cal screw on the front for calibrating the link circuit. Connections are also provided on XLR

and jack (outputs, thru/mult and inputs) for external sidechain control of the gain reduction, but internally a pair of EQ controls are available for contouring the side chain. 10dB of cut or boost (in 2dB steps) is available for the LF and HF bands, and this is most useful on programme material with dominant HF or LF content. These are set at 50Hz and 10kHz respectively and works extremely well, although the degree of control is perhaps over-generous. The continuous pot labelled Glory has Earth at one end of the scale and Heaven at the other. As the control is turned towards Heaven, low-order even harmonics are added in the 40-700Hz range. So there is distortion in the afterlife! The legending alone certainly impresses non-technical clients. While this is subtle on signals with predominantly high frequencies, deep basses tend to exhibit slight fuzz when turned to Heaven. The manual explains at length how this effect is uniquely implemented, but although this can enrich a lifeless signal, I felt there was something slightly contrived about it — mixing a bit of fuzz in with a clean signal is not the same thing as pushing a vintage valve circuit that is straining to cope. A touch of this hairy distortion can be useful, but it never sounds as rich or grainy as a Drawmer 1968 or ADL1500 (respectively). It is fast enough for crunchy drum ambiance, but not as convincing as the Drawmer. Where the GC works best is on acoustic guitars, bass and vocals. This is obviously a unit for fiddlers and tweakers, unlike, say, the two-knob LA-2A. There is far more to think about here, but the Glory Comp (UK£1749 +VAT) has a character and sound of its own, albeit more neutral than most vintage valve compressors. It sounds as big as it is and is as proper as the military look suggests; it’s incredibly smooth without becoming totally bland. ■

Contact GROOVE TUBES, US: Website: www.groovetubes.com UK, Stirling Trading: +44 20 8963 4790

PROS

The ultimate in smooth valve compression.

CONS

‘Heaven’ is slightly disappointing(!); 5U of strong rackspace required!

EXTRAS

The ViPre all-valve mic preamp offers eight tubes, a fully-differential (balanced) signal path with 75dB total gain via 1dB and 5dB stepped attenuators, four variable transformer impedances and two balanced-bridged inputs. It has five variable rise-time settings, separate +4dB and -10dB transformer outputs, and microphone and +4dB line level balanced inputs.

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November/December 2006

review

Digidesign Mbox 2 Pro The latest in the Mbox range of Pro Tools entry level products is a FireWire 6 in, 8 out audio interface designed to work with Pro Tools LE. ANDY DAY reckons it’s a well balanced package.

Y FIRST IMPRESSION of the Mbox 2 Pro was the large size of the unit for a relatively small amount of I-O, but it actually works quite well when using it with a laptop because you can stand the computer on top and have easy access to all the controls. Despite its size the Mbox Pro is very light and portable making it ideal for ‘on the road’ and location recording due to the power being supplied through the FireWire port. I used the Mbox Pro with a Mac-book 2GHz laptop and could get about 3 hours of recording with a full battery charge (without phantom power). You can also use the supplied AC adapter, which comes with plug adapters for every known power socket in the world. The interface consists of MIDI I-O, 6 analogue inputs, 2 front panel DI jacks (for guitars and basses), 2 rear panel XLR/jacks for mic inputs and an aux input, which can be line level or direct from a deck. This is quite a nice touch for DJs wanting to hook up a deck into Live or Reason. The mic inputs can be phantom powered, padded and trimmed from the front panel. The DI jacks are also handy for quickly plugging in a guitar and recording, plus with two headphone sockets (each with individual level controls) you can always monitor what’s going on. Another nice feature for DJs is the ability to switch one headphone socket to monitor outputs 3 and 4, allowing a cue mix to be routed directly to the headphones in a live DJing situation. One of the main differences between the Mbox Pro and the normal Mbox is the inclusion of Word clock IO and this makes integration into other digital studios easier. The lack of AES connectivity is a concern as most digital studios have AES connections rather than consumer SPDIF. Outputs consist of six analogue jacks (two of which are unbalanced) and one SPDIF. Generally the Mbox Pro is well speced and quite well laid out, albeit slightly lacking in digital I-O. As with the basic Mbox, the Pro comes with Pro Tools LE V7 and the fantastic ignition software pack. This is definitely the best thing about the package, you get Reason 3 adapted, Live 5 lite, Melodyne, Amplitude LE, Samplitude LE, some rex loops and BFD real drum simulator. This is great for musicians, as you

can compose, mix and produce tracks straight away, all with the Mbox package. I installed all the supplied software and after a few registration hiccups I was up and running. I already have Live 5 so I used that instead of the supplied lite version, but it worked with the Mbox first time around. This was my first look at Pro Tools 7, because I work mostly in postproduction and the new features in 7 are more about MIDI. I was very impressed with how Digidesign has improved the MIDI features of Pro Tools and the Rewire integration with Live and Reason is great. The ignition pack represents great value for money, as all the supplied software can be upgraded to the full versions at a preferential rate. But the versions supplied are not just glorified demos and represent great pieces of software in their own right. I recorded some MIDI tracks in Pro Tools and routed those to instruments in Reason and Live, and for geek value recorded them back into Pro Tools to check for Rewire delays. On my Mac-book everything was spot on for timing. Things have definitely changed for the better with MIDI in Pro Tools 7, however, I prefer to use Live as it’s my main composition tool nowadays. The cool thing about the Rewire integration is that you can flip between Pro Tools, Reason and Live seamlessly. It’s easy to record MIDI in Live, route the output to Pro Tools to add RTAS plug-ins or record for later mixing. And because Pro Tools LE is compatible with Pro Tools HD you can take your session to another studio to mix on a larger HD system. Another nice inclusion in the Mbox package is a training DVD on Pro Tools 7. This is useful for beginners and experienced users alike, I discovered a few tips myself from watching a few chapters. This is definitely the way forward for all DAWs and hopefully something other manufacturers will copy. Finally, regarding Pro Tools LE 7, I was surprised that it still only works in stereo, especially with the possibility of outputing 6 channels from the Mbox2 Pro. 5.1 capability would be the next obvious step, if for nothing else than to justify the 6 outputs on the interface. The sound quality of the Mbox 2 Pro is very good and it does have the capability to record at 24/96k,

although I don’t think many people are actually doing this yet. The mic amps seem reasonably quiet and there is plenty of gain range. I recorded some vocals using a Neumann U87 (which I powered from the Mbox) and the results were amazingly good, plus, with the included RTAS plug-ins, there’s a reasonable choice of EQ, dynamics and effects. There are some additional add-on packages for both Mbox products that give more functionality in music or postproduction applications. With the Music production toolkit you can add extra plug-ins and unlock some of the more professional elements of Pro Tools HD. The DV toolkit gives you timecode display within Pro Tools LE and the ability to import/export OMFI files, as well as a few post-orientated plug-ins. These toolkits close the gap even more between the Mbox family and Pro Tools HD, especially as host processors become more powerful, allowing more RTAS plug-ins to be used in LE. I would be very interested in trying the Mbox 2 Pro and DV toolkit combination within a postproduction workflow. My final conclusion is that the Mbox 2 Pro successfully bridges the gap between Mbox 2 and Digi002. But for the extra £180 over the Mbox 2, I would expect more pro features like AES I-O (or ADAT 8-channel I-O) and ideally 5.1 support within Pro Tools LE. Despite these points, it’s a good product with plenty of uses in music or postproduction and with compatibility with Pro Tools HD systems. A good platform to base your projects on. ■

PROS

Compatible with Pro Tools HD; excellent package of software with Pro Tools 7 LE and the ignition pack; useful DVD tutorial.

CONS

Lack of AES I-O; no multichannel support in Pro Tools LE; slightly higher priced than other audio interfaces.

EXTRAS

Mbox 2 Pro (UK£480) includes Pro Tools LE software and more than 50 effects and instrument plug-ins, including DigiRack, Bomb Factory, the Xpand! sample-playback/synthesis workstation, and the Pro Tools Ignition Pack. For UK£60 more, customers can purchase Mbox 2 Pro Factory, which includes an additional five Digidesign and Bomb Factory plug-ins (worth more than £650), the Mbox 2 Pro system, and an iLok Key.

Contact DIGIDESIGN, US: Website: www.digidesign.com

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November/December 2006

Magic happens...

...outside the box.

Duality - Master Production System

If the magic is missing from your mix, put some analogue soul back into your “workflow”. Duality, the new music production console from Solid State Logic, delivers timeless analogue audio processing alongside state-of-the-art workstation integration in a console packed with highly evolved technology. The VHD™ circuit can take the pristine mic pre’s from crystal clear to driven and edgy, adding warmth to your DAW signal path. Two classic SSL EQ selections are available at the touch of a button. Sharp TFT displays give unprecedented levels of visual feedback. Advanced Total Recall allows multiple operators to recall settings simultaneously. And a unique signal architecture can place the console’s analogue processing at the heart of your DAW. Of course, there is more, but you must experience the full power of Duality for yourself. Call us now to arrange a demonstration. Your world is about to change.

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

www.solid-state-logic.com

• Pristine SuperAnalogue™ mic pre's and signal path • VHD™ circuit : from 50’s tube warmth to 70’s transistor ‘edge’ - adding character to your DAW • Unique SPLIT channel signal architecture : allocate processing to your DAW record or monitor path • Total DAW & plug-in control direct from the console • Selectable legendary SSL E & G Series channel EQ • Low power consumption means no machine room and reduced overhead

review

Lexicon MX400 Would you believe it? A dual stereo/surround reverb effects processor that you bolt into a rack. ZENON SCHOEPE says it can now be regarded as retro and is therefore cool.

HAVE TO ADMIT that it has been a quite a while since I last looked at a new dedicated hardware effects unit. In fact I had a quick look around and was somewhat dismayed, but not altogether surprised, that the Lexicon MX400 has its price bracket and feature set largely to itself. There was a time when a box like this would have had some ferocious competition. It strengthened the breed, so much in fact that many simply went soft and retired from the hard game. Mind you, the MX400 has that angle covered too as the USB port on its rear let’s you connect to a computer and employ the supplied MX-Edit software as an editor/librarian (how quaint) while VST and AudioUnits software allows the box to appear within your workstation as a hardware plug-in and benefit from its automation. The point to remember is that the audio will be passing through the physical box, which is what I want to concentrate on here. Two versions of the MX400 exist: one looking at the outside world through balanced jacks, another with XLRs — called the MX400XL. I can’t pretend I’m not baffled by this as it seems an unremarkable distinction to make when all else remains the same. The MX400 costs £425 (+VAT) while the XLR version hits £468 (+VAT), which, to put it into some context, is within around 50 quid of the MPX1 and that offers both types of I-O. Admittedly what differentiates the MX400 is that it offers two fully independent stereo channels of processing and the rear panel has two pairs of inputs and two pairs of outputs to reflect this. The fact that these pairs are marked Front and Rear tells you that it can be configured as a surround processor in an LR LsRs sense. Phono SPDIF I-Os also reflect this convention and run at 44.1 or 48kHz. You can switch the MX400’s mode to stereo, dual stereo or surround and you’ll be presented with appropriate varieties of presets to experiment with. I wasn’t entirely happy with the monitoring setup I threw together to demo the surround capability and the fact that I didn’t have a broad enough selection of programme to pipe through it. But the algorithms are certainly linked for 4-channel reproduction and display some interesting movement between the front and rears while giving a convincing impression of being of one space. That said, for most people the surround processing is unlikely to be the main reason for buying the MX400, it’ll be an added bonus. The strength, to my mind, is in offering two really rather 34

good stereo reverbs. I say reverb but this unit does embellish this palette with more. It offers 17 reverb types though halls of various size and application, to plates, rooms and other spaces plus seven delay types, a compressor and de-esser, and eight essential mod effects. Operation is fairly straightforward if you understand Lexicon’s take on using assignable dials to alter parameters values. The screen starts to get a little crowded and compromised when running two stereo processors simultaneously as it becomes divided in half horizontally at the top menu level to display one preset identifier in the top half and the other in the bottom half. In order to get in and adjust one of the two stereo processors you have to first select it on the Page/Select dial and then activate it with a press. Doing this reliably was a source of continued frustration for me because I kept getting it wrong, or thought I had, requiring regular back stepping with the Exit key. Once you are into the edit, matters improve as the whole screen is dedicated to the stereo processor you’re working on. It certainly helps if you can get the box up to a position where you can see the screen straight on. The front panels, has individual input pots for the two stereo channels with rather meagre 4-LED metering that is largely redundant as the unit is pleasantly unfussy about keeping levels high to keep the processing sounding good. Four display-related mode LEDs indicate whether you’re in surround, stereo or working on stereo channel A or B in dual stereo mode. The Page/Select dial accesses menu levels and an Exit button takes you back a step and usefully has an integral LED to tell you there is something to go back to. A Tempo key also flashes to tell you to ‘tap tempo’ in a relevant preset. A ‘local’ Bypass allows you to mute a selected effect process for instances when you’re working on another constituent in a preset. Then there are the three edit knobs that adjust particular parameters in editing; not always logically but if you’ve used this type of Lex arrangement before you’ll get the drift. Store commits your edit to the user memories. Stereo mode has 99 factory and 99 user locations, dual-stereo has the same, while surround has 25 factory and 25 user programs. These are scrolled through on the Program dial with press to make. Finally there’s a global Bypass. All effects boxes are judged on their presets, it was always so and it’s why plug-ins, with their resolution

gargantuan libraries, eventually wear the user down into believing they’ve got some great presets in there somewhere if only they could find them when they really need them. To my ears the MX400 delivers a credible presentation of its abilities with a selection of immediately usable patches. The Lexicon sound means different things to people now than it used to but I like what I heard through this unit. It’s not overly bright and sparkly in an artificial and forced way but neither does it pretend to play the convoluted reverb card. It’s a synthetic reverb that draws from some excellent algorithms that have been perfected over years of experience because they have been found to work. That really is the key here. You can edit, but it’s structured in a way that allows you to adjust the things that make most difference most immediately. If it’s not what you want then there will be another algorithm that can be adjusted so it will be. It’s fast, you’re using your ears and not looking at a distracting screen, and you can get on with the production like you are meant to. There was time when I might even have considered joining the army to own a box like this. Today I’m just very impressed with the value and performance of the MX400 but I suspect that I may not be a typical user and there will be some for whom the prospect of a physical rackmounting processor will seem a little archaic even with its hardware plug-in capability. To them I’ll say that there is no substitute for the convenience and immediacy of this density of power, sound and practical control on a hardware unit that sounds this good for this sort of money. It’s a shame that people don’t get quite as excited about the retro concept of an outboard effects processor as they do about compressors and EQs. Maybe that time will come. Until then I still think this is a remarkably proficient and fine sounding unit and there is not much else to touch it at the price. ■

PROS

Fast; convenient; sounds great; high value.

CONS

Some won’t see the point; XLR or jack? Oooh, decisions, decisions!

Contact LEXICON, US: Website: www.lexiconpro.com UK, Harman Pro UK: +44 1707 668181

November/December 2006

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review

Cycling 74 Upmix In an increasingly multichannel world the need to repurpose stereo into a meaningful 5.1 representation is a real one. ROB JAMES fixes on his bicycle clips and heads for the open road.

.1 IS HERE TODAY and it isn’t going away. This poses a number of challenges to audio professionals, just as the advent of stereo and Dolby Stereo did a few short years ago. Re-purposing mono material for stereo is at best an interesting curiosity; at worst the results can be horrendous. Fortunately, with true stereo material there is much more to work with and most engineers dealing with surround mixes have a number of strategies for producing convincing surround from stereo sources. The one thing all of these have in common is the time they take to set up and fine tune. Combining the bits and pieces into a plug-in is an attractive notion even though set up and tuning will still be required. Cycling 74, the company responsible for Upmix, is a new name to me. Loosely based in San Francisco, Cycling 74 is one of the new breed of companies with a distributed workforce, only possible thanks to the net. Upmix is in reality a suite of plug-ins. The eponymous Upmix leads a pack consisting of Rotator, which enables a 5.1 mix to be moved in a circular fashion, and ReRoute, which enables output channel routing to be changed to match your system’s internal routing. The others are subsets of the full Upmix plugin. FoldDown is a compatibility aid for checking the effect of folding a mix down to stereo and ReBalance simply provides a set of faders for each of the surround channels. The last member of the gang is LFE-6chan, which is a low-frequency generator/adjuster. The Upmix bundle retails for US$495 and currently works with Nuendo, Cubase SX, Motu Digital Performer (AU) or Digidesign Pro Tools TDM (RTAS). In the fullness of time other manufacturers are likely to make their VST support compatible with surround plug-ins, for example Cakewalk’s Sonar and Apple’s Logic. Upmix uses the Pace iLok challenge and response security system, as used by Waves and others. The authorisation can be transferred to an iLok USB dongle on request. All the plug-ins have a channel format selector switch that offers a choice between SMPTE, DTS or Film. This affects the channel routing through the plug-in and determines the input and output formats. 36

A bypass or compare switch is another common factor, enabling the processed signal to be compared with the original. Upmix itself consists of five modules. Input selects the input type. Center is for the generation of a centre channel from a stereo (or mono) source and tuning the result. Surround Process generates and adjusts the surround channels. LFE generates and tunes the LFE channel and Output Levels provides faders, delays and monitoring control of the resultant surround output. Input offers a straight choice between stereo and mono (Left only), Center sums left and right with adjustable LF roll-off and a single band parametric EQ to help pull things into the centre. Surround offers a choice of processes to produce the surround channels with a depth control where appropriate. A variety of algorithms are provided to cater for a wide range of circumstances. Used in conjunction with the delays (judiciously if folddown compatibility is an issue) these should cover the majority of requirements. The LFE generator also offers a choice of input source. Stereo input can sometimes result in unwanted cancellation and consequent reduction in LFE level, hence the choice. A source filter sets the cut-off frequency for the source and gate trigger. The Gate governs the input to the Dry, Wet and Oscillator sections thus acting as an oscillator trigger. The oscillator itself allows a static tone to be added to the LFE signal with choice of sine or square wave, frequency and volume. Downshift adds a pitchshifted version of the input signal to the LFE output. Shift is in semitone increments for -1 to -24. The Dry fader passes the filtered LF signal regardless of the downshift setting and Wet controls the proportion of downshifted signal. An output filter limits the LFE output to the relevant specification (80Hz or 120Hz) with a 24dB per octave High-pass filter at 20Hz. Since the LFE generator is capable of producing extremely low frequencies the Low Cut button invokes a 48dB per octave filter at 20Hz. Rotator enables the five full-range channels of a 5.1 mix to be rotated around the centre point in five-degree increments. Panning is equal power with no divergence control. At 90 and 270 degrees this can result in resolution

overload, therefore an overall gain control is provided. ReRoute is a simple switcher that can be used to rearrange the channel order. Since there are a multiplicity of surround channel formats this can fix annoying routing problems quickly but presets are provided for all the common conversions. Momentary solo buttons offer a quick check on what is where. Upmix uses the ‘standard’ ITU/SMPTE format internally, i.e. L, R, C, LFE, Ls, Rs. ReBalance is a six-channel level control with the same extras as the other plug-ins. It’s useful for rebalancing an existing interleaved mix without the necessity to split it out into individual mono files. The stand alone FoldDown module offers a useful guide to downmixing results and can also be used to produce a stereo downmix for distribution. It offers user adjustable parameter presets that follow the Dolby metadata standards. Two further controls determine the level of the centre and rear channels with -3dB, -4.5dB and -6dB options for the centre channel and -3dB, -6dB and infinity for the rear. LFE-6chan is the LFE section of Upmix with some bells and whistles. For a start, LFE-6chan can accept a full six channel input meaning it can modify an existing LFE channel while passing the other channels intact. Input source choices are Stereo, Mono (L only or the LFE channel). A Merge button adds generated LFE to the existing LFE content. The headline plug-in — Upmix — is undoubtedly the star of the show. It produces a credible surround field from a variety of sources with minimum effort. The LFE generator is very helpful if used carefully. The rest of the bundle is a nice bonus, although I cannot envisage too many circumstances where I’d want to be using Rotator. There is plenty of scope for a lengthy philosophical debate on this subject, but not in this article. In short, if your work uses a lot of stereo material for 5.1 then Upmix is well worth a demo. ■

PROS

Relatively painless re-purposing of stereo into 5.1; easy to use; useful tools.

CONS

Only works with certain platforms; LFE generator could be more sophisticated with modulation, envelope, etc.

EXTRAS

Octirama is a dynamics plug-in for 5.1 with proprietary DSP technology that is said to preserve the surround image and is available exclusively for Pro Tools TDM.

Other plug-ins from Cycling 74 include the Pluggo collection of more than 100 real-time plug-in audio effects and virtual instruments, Mode is a set of plug-ins that combine classic synthesis and effects technique with ‘performance control’, and Hipno is a set of more than 40 effects and instrument plug-ins featuring the Hipnoscope user interface.

Contact CYCLING 74, US: Website: www.cycling74.com

November/December 2006

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John Cornfield Despite starting from what must be the most remote studio in the UK, he’s become a top indie/alternative rock producer with hip names such as Supergrass, Muse, and Razorlight, glued to the back of his own when mentioned in A&R meetings. John Cornfield talks to NIGEL JOPSON about Cornish bands, headphone mixes and Soundscapes.

OHN CORNFIELD BEGAN his recording career in classic fashion, bodging together mixers from veroboard and mic stands from tent poles. Visiting his first real studio with a band, he mended the main monitor amp when it blew up and scored a job at one of the most secluded and characterful residential studios in England: Sawmills, located in the remote South West county of Cornwall, accessed by boat (but only at high tide). He learnt his trade looking over the shoulder of producers like John Leckie recording the Stone Roses, and then engineered for a string of top acts including XTC, The Verve, Robert Plant and Catatonia. John’s break as a producer came with Supergrass (for whom he had engineered the debut I Should Coco album) with the ambitious sounding In It For The Money, which the NME famously described as 38

‘more fun than watching a wombat in a washing machine.’ He mixed the epic Origin Of Symmetry for local-gone-global alt-rockers Muse, recorded some of Definitely Maybe for Oasis, and produced Oasis-on-speed rockers Dogs album Turn Against This Land, with a raw sound that turned Paul Weller into a fan. John hit the charts big time when he jumped into the producer’s seat for Razorlight, and scored a number 1 with their first album Up All Night. Now he’s on the producer ‘A list’ for major label’s priority band signings like Boy Kill Boy, The Marshals and Morning Runner. But Cornfield shows he hasn’t pocketed the corporate shilling by keeping eclectic acts like The Swans, Kitchens Of Distinction and the jazzilicious Afro Cortex on his discography. We talked to him at Sawmills Studio (Photos www. recordproduction.com). resolution

Did you learn a lot from the producers and acts who came to Sawmills to escape big-city scrutiny? John Leckie came to Sawmills with The Dukes of Stratosphear [XTC] in 1987 to record Psonic Psunspot, which was an extremely interesting project to get thrown into that early in my career. The whole world of John Leckie and Andy Partridge was fascinating ... to say the least! I learnt a lot from John, and went on to do quite a few projects with him. But New Model Army was what really kicked it off for me, from just being some Joe Bloggs sitting in a little studio down here in Cornwall ... it was all pretty laid back. Justin and Rob from NMA came down to do an album for Justin’s punk-poetess girlfriend, we hit it off, and they were quite impressed with the sound we were kicking out here from rather basic equipment. They came back to record their own album [Thunder & Consolation], and that was my ‘in’ to getting records out into the mainstream. Working at Sawmills I saw a whole lot of engineers and producers doing things different ways, sometimes I thought ‘I wouldn’t have done it that way, that’s upsetting the band and pissing them off, surely that’s not getting the best out of things.’ You get to pick out the good and bad points from other people, and then kind of decide: this is the way I’m going to do it because this is the way it works — for me. November/December 2006

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Every song on Origin of Symmetry is so full of arpeggio sequencers, octave-divider basses and mad piano riffs â&#x20AC;&#x201D; how did you approach the mix? Thereâ&#x20AC;&#x2122;s actually even more on the multitrack than we used. Originally, I wasnâ&#x20AC;&#x2122;t totally convinced of Dennisâ&#x20AC;&#x2122;s decision to go with Muse, but then when I went to see them live supporting Bush â&#x20AC;&#x201D; I was just literally pinned against the wall â&#x20AC;&#x201D; I understood what it was all about. I felt this really hadnâ&#x20AC;&#x2122;t been captured yet, so when I mixed the album I thought Iâ&#x20AC;&#x2122;d approach it as though they were on stage â&#x20AC;&#x201D; Mattâ&#x20AC;&#x2122;s guitar off to the left, bass distortion effects over to the right hand side â&#x20AC;&#x201D; we kind of structured it from the bottom up, rather than top down. We decided that all these layers of overdubs should be kept to a minimum, apart from where they were essential to the melody. Thereâ&#x20AC;&#x2122;s also an incredible live vibe on the Turn Against This Land album you produced for Dogs. Itâ&#x20AC;&#x2122;s a cracking representation of a band â&#x20AC;&#x201D; it sounds like it was recorded in 5 minutes. If only, eh? When the takes came ... they might have been done in 5 minutes, getting at those takes was a lot of it ... getting through the blur of the hangovers! We did have a laugh making it. I always insist on going to see a band live before we take it anywhere. If you see the live performance and think: â&#x20AC;&#x2DC;bloody hell, this needs a lot of looking at,â&#x20AC;&#x2122; then weâ&#x20AC;&#x2122;ll go in and sort it out in preproduction. If the demos have a vibe about them and the live performance is cool, then I pretty much just go into the studio and do it.

Do you compress or EQ your mixes before you send them off for mastering? This has been a bit of a bugbear of mine for a while now, the current trend in mastering is that your CD has got to be louder than everybody elseâ&#x20AC;&#x2122;s to the detriment of the dynamics in the music. Years ago radio stations used to compress the shit out of it, which made music work on little speakers, but when you got the record you could still turn it up loud to get the full dynamic range. In the past I used to leave my mixes quite open when sending them off to labels for approval. Recently I worked on a project where I did a load of rough mixes, then I banged them through

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Youâ&#x20AC;&#x2122;re a prominent user of the Soundscape DAW, how much of your production is done in-the-box? It varies. I first bought a Soundscape system in 1996 and at that stage I was really just using it as a digital tape recorder, a means to edit without having to sit there for hours with a razor blade. As the system has developed over the years, Iâ&#x20AC;&#x2122;ve used more of its internal processing, but the Wire Daisies is the first time I have mixed a rock album within a computer. November/December 2006

better than the Powercore 24-7, but I use the TC MD3 mastering plug-in, itâ&#x20AC;&#x2122;s the same algorithms from the System 6000 ported across.

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What have you been working on recently? Iâ&#x20AC;&#x2122;ve just finished producing the second album for the Wire Daisies. They have some beautifully done acoustic stuff, just off the cuff playing with live vocal and acoustic guitar. They built a little rehearsal facility, and got themselves a few bits and pieces like a Nuendo system and a 32-channel TL Audio M4 desk. I took all my gear and basically recorded and mixed it in their rehearsal room. I couldnâ&#x20AC;&#x2122;t get into Sawmills because it was booked, the Wire Daisies are a local Cornish band so this solution avoided â&#x20AC;&#x2DC;going up to Englandâ&#x20AC;&#x2122;! I was doing the monitor mixes within Soundscape as we recorded because there wasnâ&#x20AC;&#x2122;t an analogue monitoring board, and by the time we were ready to go into a studio to mix I was thinking â&#x20AC;&#x201D; hang on a minute, weâ&#x20AC;&#x2122;re nearly there! We just spent a couple of weeks doing some rides and zinging it up a bit, so I ended up finishing the whole album in their rehearsal space.

Thereâ&#x20AC;&#x2122;s obviously quite a lot of analogue processing done when I record, Iâ&#x20AC;&#x2122;m using all the nice valve stuff and compression plus EQ when Iâ&#x20AC;&#x2122;m going in, so a lot of the time Iâ&#x20AC;&#x2122;m not actually doing a lot to the sound with the computer as I record. When recording youâ&#x20AC;&#x2122;ve got to work in a zero-latency environment, for overdubbing you canâ&#x20AC;&#x2122;t use any VST plug-ins because of the delay. I have been using some of the internal Soundscape DSP plug-ins recently, and Iâ&#x20AC;&#x2122;ve been pleasantly surprised by how well they work: thereâ&#x20AC;&#x2122;s the Drawmer SDX100 compressor which is really good, their own basic EQ and also a new EQ which Iâ&#x20AC;&#x2122;ve been beta testing. I use those while recording, then when Iâ&#x20AC;&#x2122;m mixing I fire up all the UADs, the TC Powercore and all the VST plugs. I tend to use some of the whackier stuff from the Powercore, sometimes I use the Intonator and Filtroid, I like the UAD 1176

Was mixing Museâ&#x20AC;&#x2122;s Origin of Symmetry album a landmark for you? It was a landmark to get to the end of it! Iâ&#x20AC;&#x2122;d worked with them before, because Muse sort of started their career off at Sawmills, Dennis [the owner] had pulled them through and got them up there. I had been in and out with them and done some mixes for them here and there. Paul Reeve and John Leckie had recorded most of their first album while I was away making Supergrass 3.

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a limiter to peak them up to CD level. Michael Brauer was doing the final mix on this project. Much later he phoned me up saying ‘What the hell did you do to the rough mixes?’ — the US label had rejected his mixes saying they preferred my monitor mixes. I explained what I’d done, so he hired an L2 [Waves] hardware box, ran his mixes through it and sent it back to the record label. They said ‘Great’, thinking that he had remixed it! ’Kin’ hell! Why don’t they just turn the volume on the stereo up if they want it loud? Michael couldn’t believe they weren’t hearing it, I was as dismayed as he was about what was going on. We were both batting from the same side, but you get caught in a dilemma where if you don’t do that to your mix people think there’s something wrong with it. It’s purely a level thing, how do we reset it? I don’t see how we can put it right. I now

spend quite a lot of time trying to find ways of getting the mix to be loud without making it sound like it’s been clipped to bollocks. It’s not purely down to which peak limiter you are using, it depends on the programme material. I’ve certainly found the TC Electronic MD3 multiband does do it better than the rest, at the moment.

Have you got the MD3 set in M/S mode? Do you use mainly the peak limiter or are you applying soft clipping? I’ve been using both, with the MD3 as a kind of mix compressor, seeing how far I can push it without making it sound like shit. I was surprised at how well the M/S mode works, but M/S can sometimes produce rather odd artefacts and change the mix balance. I’m still getting my head around it.

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So when you worked at Sphere Studios with Razorlight, did you take the Soundscape with you? It goes everywhere with me! I’m just working on making it a little smaller and easier to move around at the moment. If I’m working in a studio with a big enough analogue board with sufficient monitoring then I use the Soundscape system more like a tape recorder — 1 in, 1 out. How do you arrange headphone monitoring when recording with a DAW? If you’re working within a native DAW there are latency issues to deal with, but it’s only microsecond convertor delay from the Soundscape channels. At Sawmills we’ve got the Furman HRM-16 headphone distribution system which is brilliant — it just saves

November/December 2006

craft so much time and grief. All the musicians have their own little 8 mono plus 4 stereo input mixers. If musicians have to share a limited set of headphone mixes then in the end someone always has to say: ‘sod it — I’ll put up with it,’ because they can’t resolve between themselves how the mix should be balanced. This way, musicians can do what they want, and I’ve noticed quite a bit of difference in performance levels.

Do you take the feed for the Furman from the console monitor bus inputs? What I do is feed the drum sound as I have it grouped on the monitor board as a stereo pair, but I also send a separate feed of the kick drum and the snare, everything else just comes straight off the multitrack out. When I was producing the Wire Daisies I had a similar thing set up, all done in Soundscape. I sent direct outs into a little Allen & Heath stage monitoring board, so they all had their own mixes, albeit with the controls in one location. Did you record the Wire Daisies straight into Soundscape? I was using the Daisies’ TL Audio M4 mixer basically as a bunch of mic amps plugged into the Soundscape inputs, direct out from each M4 channel. You can drive the M4 channels quite hard and get a sort of tape saturation effect, I thought it sounded quite good, I like that board. I was surprised, having spent most of my life working in studios, how good the results were. The Wire Daisies album hasn’t been in to a conventional studio at any point. Is there a future for the fixed-location, nonorchestral recording studio? Well, what I had set up with the Wire Daisies was pretty much a recording studio! I’ve got quite a lot of experience putting studios together, we spent the first 10 days of the session doing a bit of acoustic treatment in the room and wiring things up. They had already done the construction work — so it pretty much was a studio — albeit without a huge amount of outboard and flexible wiring: it was just wired for one mode of operation. I know when we describe these recording projects it seems like they were just done in a front room, but that’s not the case, it was a studio, really. I think there is still a role for proper live-band recording studios. Have you got a trusted set of reference monitors? At home I’ve got a pair of big Mordaunt Shorts — I’ve actually got about four front stereo pairs and five or six surround systems — all in the same room! In the studio, to be honest, unless I have a pair of NS10s I’m lost. But I use NS10s with a subwoofer, I can’t bear listening to them without one. I just use whichever sub is available, obviously you’ve got to spend a little time fiddling around, I’ve got three subs of my own, one of the best I have is a Paradigm. It just fills out the bottom end and lets you know what is happening right down there, I always set them up quite aggressively because I’m a bit of a bass junkie, if I haven’t got that bass, I’ll wind too much into the recording and end up over blowing it. What are your overall guiding principles for making great records? In the end it’s the music that counts, and no amount of sonic trickery really changes that. We all love messing about with all the gear, it’s good fun. You can embellish stuff but if you haven’t got a song and a performance — you haven’t got it! ■ November/December 2006

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craft How did you work? We would book some studio time, they’d turn up, do all their sketches and you had to be really on the ball because invariably we’d do three or four sketches, have a break, have a cup of tea, come back and think, ‘OK lets just go back and pick up on that.’ So your documentation had to be spot-on. Then I’d put a rough assembly together, throw some sound effects in and then whoever was assigned to oversee it would come in and we would work out any changes. Python was involved in theatre and film and, obviously, we did our own commercials when we had product out. Then there’s DVD. For example in 2001 we remastered Holy Grail which had only ever been out in mono. We managed to get hold of a triple track split with dialogue, effects and music and just rebuilt the whole thing over a period of months, then did a final mix with Robin O’Donoghue at Shepperton. Holy Grail was the biggest of films they made, especially in America. It’s rumoured that even Elvis had it on his shelf. Were they well behaved? They were very good. In the ‘Embarrassment’ sketch for instance, a lot of the effects were done live as we recorded the voices. They would hold things to drop, paper rustles or whatever. Anything that was physically possible we did at the time. Sometimes things just develop, like the ‘audition’ sketch, put together and made up on the spot. We used bits of that in various other sketches. It will be one 15-minute sketch on the out-takes album. There were moments of crack up time. Once Terry said, ‘Do you know, it’s just not very funny this. Maybe if I took my trousers down it would be funnier.’ So he undid his trousers and did it again and of course it was hilariously funny.

André Jacquemin Swept along by the success and historical significance of the work he did with Monty Python, André has collected an amazing credits list and reckons he’s worked on more than 50,000 commercials. He talks to ROB JAMES about keeping it small, gear decisions and why he and Michael Palin could have been Starbucks.

NDRÉ JACQUEMIN HAD his baptism of fire aged just 17 when he agreed to do the second Monty Python album. His career never looked back and he has amassed an almost unbelievable body of work composing, recording, editing and mixing everything from commercials to music and feature films. At a time when VCRs had yet to appear, the only way Monty Python fans could get their fix was to wait for the repeats or buy the records. The first album, mostly a rehash of the first TV series, was recorded live at Camden Town Hall and didn’t work very well. In 1971 André was working in a little commercials studio in Wardour Street, Studio G, under the tutelage of Radio Luxembourg recording legend Alan Bailey. Michael Palin wandered in one day, looking for a studio to record a voice-over demo tape. Bailey was already booked so André got the gig and more than 30 years later he is still inextricably linked with the Python franchise. Since 1975 Jacquemin has had his own studio, Redwood. Over the past 30 years Redwood has been through many incarnations and changes of address. Currently located in Great Chapel Street, at the heart of London’s Soho creative hothouse, Redwood is once again ahead of the curve. The latest project, just completed, involved remastering new versions of eight Monty Python albums with a load of new material, sketches and 42

interviews. With Spamalot playing to packed houses in London after its phenomenal success in New York the rerelease seems ideally timed to catch the Christmas rush.

How did you get on with Michael Palin? I thought the sketches were really funny but I didn’t know who Mike was because I was always working, not watching TV. Mike would pop in between filming to record bits. Meanwhile I’d edit the voices, put sound effects and music behind it and every time he was terribly impressed because outside the BBC nobody was really doing this kind of material. Eventually, when we finished, Mike said, ‘Look, I’m making an album, would you like to help us put it together.’ I said, ‘yeah’, and off he toodles. About six months later Mike phoned, ‘Could you come down for a meeting?’ They were all there, messing about. The doorbell rang and in walks John Cleese and that’s when the penny dropped. I was terribly young and I thought, I hope I can handle this. There was this twofoot high pile of scripts and I worked out a schedule for them. I thought my best plot would be to book Alan Bailey, who by this time had returned to Radio Luxembourg. It was a kind of security blanket for me. I had a fantastic teacher in Alan. He’s one of those people with the knack of putting a mic up in a room and it just sounds wonderful. resolution

How did you approach the remastering? We have this thing called the Python Bible. Around 15 years ago we archived onto CDs and DA-88s and we took the opportunity to document and catalogue everything. I’ve got the short version here. There is a bigger version... Are the DA-88s OK after all that time? Well, so far they’ve pulled back fine. I’ve had to dip into them for some of the song remixes and I’ve had no problem with them or any of the CDs although I have heard that other people are having trouble. Python paid for the archiving and, as you can imagine, there is a lot of material. We had 4-track, 8-track, 24-track and tons and tons of ¼-inch. We had it all baked and John Du Prez has them all stored somewhere, just in case. Not so much for us really, but for future generations. How did Redwood begin? I was talking to a friend while we were cutting the first album at Abbey Road and Mike overheard me suggesting building a studio. He said, ‘I couldn’t help overhearing but it would be really nice if you came over for dinner and a chat. I wouldn’t mind getting involved with you in the studio idea.’ I went over for dinner and Mike made a proposition about lending me some money to set up a studio and that’s how the Redwood came about. I did try to talk him into a themed caff instead but Mike wasn’t having any of it. He said, ‘I know what you’re good at and the studio is a good investment.’ Looking back, we could have been the new Starbucks. Did you build it? I converted my dad’s greenhouse. He’d only finished it a month before but gracefully said we could have it. November/December 2006

craft changed everything. So much work is now done before you hit the dubbing theatre and this is why they are getting worried and so many have come and gone. Dubbing time gets cut to a bare minimum. That’s a shame because it means you’re rushing that element of the process. We also do a lot of cartoon work. Dave and I did a series last year and the year before, called Boo, for the BBC.

Because it had loads of glass we had to pad it and do all the work ourselves. I had enough money saved to buy some equipment. We had two EMI TR52s, I bought a brand new Chilton broadcast desk, some AKG D202s and one Neumann U87. We also had a Grampian spring reverb and later on we invested in a 4-track Teac. Out in the garden, effects weren’t too much of a problem. You want outdoor atmos? Just stick a mic out of the window. We stayed there a few years then we had an offer from a record company to move into the West End. We were in Wardour Street, then later Neals Yard. We moved from Covent Garden to Camden Town then back into the West End to Fauconberg Court for a while and from there to where we are now. I don’t have much left from the early days apart from mics. I still have my 1-inch 8-track which is in Ray Cooper’s lock-up at the moment. It’s a low hours Studer A80 I bought from Joe Brown. When we moved to Covent Garden, Harry Day and Rob Haggas built a console for me which they called Reading Wood. That worked for a number of years then for Camden Town I bought an Amek with MasterMix, which was great. By that time I’d bought a PCM 60, a PCM 70 a PCM 80 and a Rev 7 and latterly a Rev 5. All of which I still have.

How about workstations? My first was a Doremi Dawn. We were soon using digital for most things apart from feeding in sound effects and certain special effects that I couldn’t do with either outboard gear or the Dawn. Things like varispeeding and reversing things, tapes slowing and grinding to a halt and so on that were much more efficiently done in analogue and in any case difficult or impossible to do in digital at the time. I looked at Sound Designer and just thought it was very complicated and the dedicated things from Roland and Korg. Although the Dawn was much more expensive it was much easier to use so we stayed with it until we finally made the move to Pro Tools at version 4 or 5. Why the Mackie D8B? It’s just a knock-on effect of being old-fashioned really. It’s more organic to actually mix with faders. It was invaluable on Tony Hadley’s new swing album, which I’ve just finished producing. When I wanted to dip something very quickly for one bit and bring it back up again, rather than fiddle about in Pro Tools I 44

just did it manually on the desk. Or if I wanted to add a bit of compression I could just knock it on and turn a knob to get the effect I wanted without messing about loading up a plug-in.

How about sound effects? We’re up to 400 Gigs and as of last week it’s backed up. We have two hard drives, a master and a back-up. We use Workspace in Pro Tools to access the effects. We’ve generated a lot ourselves and there’s the usual library stuff. But often, when you hear an effect in a commercial, you know exactly where it’s come from, so especially where Pythons are concerned, I try to create new stuff. And Foley? We have our own small Foley stage and a great Foley artist called Jerry Richards who, for all of his sins, used to be a guitar player with Hawkwind, so his timing’s very good. He’s worked on 15 or 20 projects with us now. We have various surfaces and our own Foley door. I’m now getting actors to sign it.

Do you do the music inside the computer? We use mostly virtual instruments and occasionally grab hold of a ‘real’ one, a pedal steel guitar or whatever we need. If there’s percussion involved sometimes I’ll give Ray Cooper a ring and ask him to bring his tambourine over. We enjoy it tremendously, but the composing isn’t the key factor in doing it as a career. You’ve got to pay the rent so we still have to do all the other bits of recording around that. We’re not big, we’re very, very small. We have one control room and studio. All these people who’d been expanding their facilities with 5, 6 or 7 rooms just made me terribly nervous. So from Camden, where we had a glorious, fantastic studio, which we’d ploughed hundreds of thousands of pounds into, I went completely the opposite way. Small is good. Our expenses are minimal now. Most of the staff are freelance and I just pull them in when I need them. For instance Jerry Richards has his own Pro Tools so once we’ve recorded a Foley session he can go home and put the thing together. Redwood now is like an agency. For a lot of the bigger facilities it’s almost too late. Any ambitions left? I’m still trying to get my own stuff together but it’s been 30 years working on that one. I’m quite happy doing what I’m doing. One of the great joys is earning a living out of your hobby. I’d like to do a lot more project work on documentaries, but people look at what you’ve done and think you are just too experienced and are going to be too expensive. The composing thing is very important to me because it’s the only thing that will earn your retirement. As an engineer you can only go on so long while your brain’s intact! ■

Do you still use analogue? When I did Tony Hadley’s album at Konk Studios I initially wanted to record the bass and the kit using analogue and the band straight into Pro Tools. But, we couldn’t get hold of any tape! So in the end the whole thing was recorded onto Pro Tools. I was debating whether to get an analogue machine in and dump the whole lot across and back, but it was all down to speed so I just stuck with Pro Tools. People use digital for its purity and then look for plug-ins to make it sound analogue, very odd! What else does Redwood do? Not long ago I was trying to look back and sum up exactly what I’d been through and what I’d been doing and it sounds absolutely impossible but I’ve worked on more than 50,000 commercials would you believe? I used to do all the Our Price, Barret’s Liquormart and Houndsditch Warehouse commercials and we were doing something like 20 or 30 a day. It adds up very quickly. These days we do mostly movie soundtracks and sound design. Take Wind in The Willows. With my writing partner Dave Howman, I wrote and scored the music. We didn’t do the floor recording but we did Foley, sound effects, atmospheres, dialogue editing, everything apart from the final mix. Technology has resolution

November/December 2006

sweet spot

Discrete layered sound PHILIP NEWELL outlines the essentials of a recent paper, co-written with Shelley Katz, that describes an interesting new approach to surround sound*.

ORE THAN 30 years have now passed since the introduction of commercially available recordings with four or more channels. Much of the relative failure of Quadrophonics during the 1970s was attributed to the limitations of domestic vinyl disc and analogue magnetic tape reproduction systems, but even in the 1990s, with the launch of multichannel digital discs, many of the old failings still seemed to be evident. Clearly, with the perfect reproduction ability of the digital discs, the medium could no longer be blamed for the lack of realistic reproduction. Something therefore seemed to be missing from the concept of the recording and reproduction systems, and not just from the distribution systems. After some months of discussions with Shelley Katz, who had been developing his Layered Sound concept, a recording session was arranged to take place over two days in the wood-panelled, Elizabethan Room, at

THE ELIZABETHAN ROOM

Herstmonceux Castle in southern England. The first day’s recordings were of a trio of grand piano, flute and soprano voice, and the second day’s recordings were of grand piano, double bass and a drum kit. The microphone arrangement for the first recordings is shown in Figure 1. What gave rise to the concept of these recordings was a pair of papers presented at the Institute of Acoustics Reproduced Sound 20 conference in November 2004. The first paper was highly critical of the current state of affairs with surround sound (1) while the second suggested a new means of creating much less correlated ambience in the reproduction room (2). One of the main criticisms in the first paper was about the way in which so many surround systems and designs of surround room acoustics were limiting the full potential of the frontal stereo sound stage, and one of the main benefits claimed in the second paper was that distributed mode loudspeakers

Figure 1. Microphone layouts for the first recordings.

THE ELIZABETHAN ROOM

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(DMLs) could reproduce ambience better than conventional pistonic loudspeakers. The authors decided that recordings should be made in such a way that the microphones close to the instruments would be subsequently mixed as a conventional frontal sound stage and reproduced by conventional high-fidelity loudspeakers. The ambience microphones would be placed close to the room boundaries and would be reproduced only via DMLs, which would be initially positioned in the listening room in directions corresponding to the ambience microphone directions relative to a defined listening position in the recording room. Transverse bending wave loudspeakers, of which DMLs are currently the best known commercially available examples, radiate in a manner that is spread out physically and temporally. At least at the higher frequencies, the radiation emanates from multiple points on the panel, and the transient response is somewhat smeared in the time domain. Another characteristic difference between pistonic sources and DMLs is that the DMLs radiate in a generally omnidirectional manner that is relatively equal with frequency. These characteristics of DMLs have led to difficulties in their application in high fidelity music reproduction, where the radiation from DMLs has not been conducive to the traditional goal of reproducing the waveform of the electrical input signal. Smearing transients is, by definition, not ‘high fidelity’ where the accurate reproduction of the impulse response is the Holy Grail. Taking as an example of a high fidelity pistonic radiator a loudspeaker such as a Quad ESL 63 electrostatic loudspeaker, its transient response is exemplary. The loudspeaker is also designed such that as the frequency rises, the source area both diminishes in size and recedes in apparent distance from the front surface of the loudspeaker, to emulate the radiation from the hypothetical point source, or small pulsating sphere. As conventional twochannel stereo loudspeakers, the reproduction from such devices can be extremely lifelike, especially when reproducing close miked instruments. In a well-designed room, the reproduction of a frontal sound-stage via such loudspeakers can almost define the current state of the art. If distributed mode loudspeakers were to be substituted for the electrostatics in listening tests, the ‘failings’ of the DMLs, in terms of conventional reproduction, would be plainly evident. The very concept of the DML is not conducive to the reproduction of such a clearly defined frontal sound stage. Reflections from architectural surfaces do not behave like the visual images in a mirrored room. Even a narrowly beamed source of high frequency sound aimed at a flat wall will not reflect in a manner much akin to what its continued trajectory would have been had the wall not changed its direction of propagation. Even at frequencies as high as 10kHz, a narrowly beamed incident source will reflect from the surface with a broadened conical directivity pattern, with a minimum of around 30 degrees. The concept can be visualised better by a look at Figure 2 (3). Lower frequencies will exhibit wider reflected directivity patterns. If the reflective surface is relatively free to vibrate, such as with wood panelled 45

sweet spot

Figure 2. Conical reflection patterns.

walls, the reradiation from the surface will be even more distributed spacially and temporally. In fact, the musical reflections from many architectural surfaces

behave rather similarly to the radiation of musical signals from DMLs. We therefore appear to have a situation whereby the frontal sound stage of a conventional recording with ambient surround is best reproduced by pistonic loudspeakers, with fast transient responses and small source areas (commensurate with wavelength). Conversely, the surface reflections that form the ambient rear channels of a recording would appear to be potentially better reproduced by loudspeakers radiating with the characteristics of DMLs. The recordings at Herstmonceux Castle were intended to test this hypothesis. Barron’s ‘Cosine Law’ (4) states that the sense of spaciousness created by lateral reflections roughly corresponds to the cosine of the angle of arrival at the listening position. The cosine of 90 degrees is zero,

and the cosine of 0 degrees is one. The spaciousness effect created by the lateral reflections is optimal when coming from 90 degrees to the listening axis, and negligible when coming from 0 degrees — centre front. The cosine of 60 degrees is 0.5, which would suggest that we would lose half of the optimal effect if the reflections came from only 30 degrees from centre front, which would be a typical location for stereo loudspeakers of a pair subtending 60 degrees at the listening position. (The ‘Cosine Law’ relates to the angle from the axis through the ears.) In a recent paper, Toole (5) reported that ‘In both concert halls and listening rooms reflected sounds arriving from the front or rear do not contribute to a positive impression [of envelopment]. [There is] a diminished preference for reflected sounds arriving within about 20-30 degrees of the median plane’. In general, the recorded reflections in a stereo presentation with a 60 degree total subtended angle cannot deliver a sense of full spaciousness. That sense of spaciousness is largely a result of low inter-aural cross-correlation (IACC). In other words, the delivery of significantly different signals to the two ears. Obviously, an acoustic signal arriving from centre front will arrive symmetrically at each ear of a listener on the central plane, so the IACC will be high. In the test recordings, therefore, the reproduction of the reflections was initially set with loudspeakers either side of the listening room at 45 degrees and 135 degrees from centre front. The inherently low IACC of the DML reproduction (as compared to pistonic sources) was also expected to aid the perception of spaciousness. The recordings were carried out in September 2005. The instruments in each trio were miked in a conventional way, using AKG C414 microphones. Similar microphones were also placed in seven positions in the recording room, two at floor level (one at each side), and five along the balcony which ran along the back of the room. Photographs of the room are shown in Figure 3. The intention was that each microphone should receive a unique perspective of the trios. Each microphone would receive the direct wavefronts from the instruments in a unique manner, with the instruments at the left hand side of the room arriving at left-located microphones

Figure 3. The Elizabethan Room. 46

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sweet spot before the instruments placed in the centre and the right of the room. The opposite would apply to the microphones at the right of the room, while the one, central microphone on the balcony would receive the acoustic signals from the left and right instruments in more or less equal timing. This was intended to be used if overall room reverberation were to be added. The signals from each microphone were recorded on individual tracks of a hard disk recorded, all via good quality A-D convertors, and no equalisation, compression or other processing was used. On the day after the recordings had been completed, the hard disk recorder and a small mixing console were transferred to a relatively large room, sometimes used to accommodate guests at the castle, which had a carpeted floor, a double bed, general furnishings including a sofa, and also several large windows. A stereo mix was made of the close microphones on the instruments. This was reproduced via Yamaha NS10 loudspeakers, which were used because of their familiarity to all the recording personnel involved. Four of the ambience microphones were then selected for reproduction via four Amina ACPT DMLs, positioned at roughly the aforementioned 45 and 135 degrees either side of centre front. The microphones which were selected were fed to the panels which most closely related to their angular distribution in the listening room. It was hoped that the reproduction via the panels would closely mimic the nature of architecturally reflected sound. All microphones, were set to a cardioid response, because it was intended that only the direct sound to the boundary should be recorded, it being unknown at that time whether the rapid reflection from the wall, if picked up by an omnidirectional microphone, would confuse the ear on playback. Obviously, the four microphones which were used for the playback of the ambient spaciousness represented only a fraction of the total reflections within the room, but many acoustic studies have shown that only a few reflections are normally required by the ear to detect a sense of space. The big question awaiting answer at the replay stage related to whether the reflections from the walls of the listening room, about two metres from the rear of the omnidirectional DMLs, would create confusing time response anomalies, or whether they would reinforce the effect of the spaciousness. When the four ambience microphone channels were faded up, the four people in the listening room were immediately aware of the sense of being in the recording room. This was a surprise to everybody, but especially to the two very experienced recording engineers, who had deliberately not been told about the true purpose of the recordings they had been making. At the time of the first playback, nobody in the room had ever heard such realistic reproduction of a space, and attempts were soon made to upset the system. The panel loudspeakers were moved such that heights and relative distances from the listening area were varied, and their relative angles to centre front were also changed. Quite contrary to the situation with conventional surround loudspeaker systems, which should be sited on the circumference of a circle centred on the listening position, all set at the same height, and all adjusted to within a decibel of each other, the Herstmonceux system was highly tolerant to heights, angles, distances from the listeners and level differences. This bodes well for domestic reproduction systems, where loudspeakers are often placed where they are convenient for domestic life rather than being positionally optimised from an electro-acoustic viewpoint. In practice, the reflections from the hard walls of the reproduction room also seemed to have little November/December 2006

effect, thus relieving one of the aforementioned worries. However, this lack of effect was recently highlighted by Toole (6, in 5) who stated that â&#x20AC;&#x2DC;The basic audible effects of early reflections in recordings seem to be remarkably well preserved in the reflective sound fields of ordinary rooms.â&#x20AC;&#x2122; In the case of the recordings being discussed here, the temporal relationships between the time of flight from the different instruments to the different ambient microphones, and the attendant level differences in the left and right hand sides, appear to have coded the acoustic sound stage in a very robust way. The whole system showed a very high degree of positional abuse tolerance and even level changes at the mixing console of up to 5dB in various loudspeakers still yielded reasonable results. In fact, it was even abused so far as to put one of the surround

loudspeakers on the floor, behind the sofa, another on the floor in a corner, and the remaining two at different heights, angles and distances, and still the reproduction sounded recognisably like the recording room. In order to test whether the robustness was a sole attribute of the recording concept, the DMLs were replaced the next day by conventional domestic loudspeakers with approximately the same frequency range and cost. The result was abject failure. With panel loudspeakers on the front channels the results were even worse. The non-flat power response of the piston loudspeakers could not send the same frequency response to the listening position as was captured by the microphones whose signals they were reproducing. The â&#x20AC;&#x2DC;point sourceâ&#x20AC;&#x2122; of the high frequencies â&#x20AC;&#x201D; a dome tweeter of less than 20cm

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sweet spot diameter — could not even begin to emulate the spacially and temporally distributed reradiation that would be characteristic of an architectural reflection. The entire system performed very realistically. With two long days previously spent in the recording room, the sensation of being there was still fresh in everybody’s minds. The realism of the reproduction was uncanny. The frontal stereo sound stage was very clear and well defined, and nothing in the surround channels detracted from its clean and tight response. It all happened in its own time and space, and no reflections were heard before their time of flight from the instruments to the microphones, and from the DMLs to the listeners. The natural reverberation of the Elizabethan Room was, of course, captured to some degree by all of the microphones. A further, unusual experience was perceived by all

present, in that everybody was aware of a sensation of clean, deep bass, way below what any of the loudspeakers were capable of achieving. The reason behind this has still to be investigated, but it was postulated that some reconstruction of fundamentals was occurring in the brain after receiving the unusually accurate stream of harmonic, spacial and temporal information. Mutual coupling between the six loudspeakers seems to be an unlikely cause because of the lack of simultaneous reproduction, due to the time of flight to the surround microphones. Despite the known colouration of the DML reproduction, very little ‘DML sound’ was perceived in the playbacks, probably because neither the colouration of the panels nor the wall reflections in the listening room were colouring the perception of the front, direct sound stage. Indeed, nothing else

was emanating from the same direction. The primary sound stage was clean at all times, exactly as in the case when that stage consists of real instruments. Many modern day recordings are built up track by track, and are recorded in rooms that do not have the ambience required for the finished mix. Obviously, the recording technique described here would not relate directly to such recordings, which make up the majority of current recordings. Nevertheless, surround reflection generation could emulate a large degree of the acoustic effects. By defining the room size, the stage location, the position of each instrument on the stage, the listening position, the wall surfaces, and other such parameters, a group of instruments, recorded separately, and even in different rooms, could be electronically inserted into the desired ‘room’. The virtual microphone locations for the reflection pick-up would then be defined, and would receive the signals from the different instruments with different delays depending on the distance from each virtual microphone to each virtual instrument. Reverberation would then be applied also to the instrument microphones after an appropriate delay. What has been described here is a surround sound system that requires six transmission channels. The frontal stage pair are traditional in every way, except that they do not carry all the ambience information as usually applied in conventional stereo. The principal ambient information is transmitted via DMLs, whose inherently diffuse time and spacial responses are well suited to the reproduction of room reflections and reverberation. Additionally the reflections and reverberation reproduced via these panels are more robust in terms of their use in domestic listening rooms. The permutations of which and how many ambient microphones would be fed through which and how many panels are virtually endless, but the system described using four ambient loudspeakers with only one cardioid microphone feeding each panel has been shown to be both realistic and robust. Hopefully, what has been described in this article is only the tip of an iceberg. If this is so, then the prospects for further work are very promising indeed. ■

References

1. Newell, Philip; Holland, Keith; Surround Sound — The Chaos Continues, Reproduced Sound 20, Oxford, UK, Proceedings of the Institute of Acoustics, Vol. 26 , Part 8 , pp135-147, UK (2004) 2. Mapp, Peter; Katz, Shelley; Layered Sound — A New Approach to Sound Reproduction, Reproduced Sound 19, Oxford, UK, Proceedings of the Institute of Acoustics, Vol. 25, Part 8 UK (2003) 3. Newell, Philip; Recording Studio Design, Focal Press, Oxford, UK (2003) 4. Barron, M, Marshall, A H; Spatial Impression Due to Early Lateral Reflections in Concert Halls: the Derivation of a Physical Measure, Journal of Sound and Vibration Vol. 77, pp211-232 (1981) 5. Toole, Floyd; Loudspeakers and Rooms for Sound Reproduction — A Scientific Review, Journal of the Audio Engineering Society, Vol. 54, No. 6, pp451-476 (June 2006) 6. Beck, Soren; Spatial Aspects of Reproduced Sound in Small Rooms, Journal of the Acoustical Society of America, Vol. 103, pp434-445 (1998) • Newell, Philip; Katz, Shelley; Discrete Layered Sound, Reproduced Sound 22, Oxford, UK, Proceedings of the Institute of Acoustics (2006)

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ten

Musicians who pushed recording boundaries Not the technology, nor the engineers or producers but the musicians who pushed creatively at the perceived limits of recorded sound. Here are ten that left an impression — there are undoubtedly more. EARLY PIONEER — Pioneer is probably never more truly used in our business than in reference to Les Paul. A music career as a guitarist that saw him recording in the 1930s and included development work on the solid-bodied electric guitar and the building of a wax disc cutting system using auto parts for his own recording use. His first home recording was released in 1947 and had been made by copying between 78rpm wax discs while adding a new part up to eight times, sometimes at different speeds. Bing Crosby gave Paul an early Ampex Model 200 and he started using tape for a series of hit records through the 1950s. He modified the machine for overdubbing use and some of this fed back to Ampex’s own machines. In 1954 he commissioned Ampex to build him an 8-track. LIVE IN THE STUDIO — Pushing the boundaries wasn’t always a conscious act. Many of the longer established studios had strict rules about where mics could be placed and how recordings should be undertaken, originally based on limitations of the gear but then out of habit. The late Gus Dudgeon, then an engineer at Decca, used to tell of his encounter with Eric Clapton early in his partnership with John Mayall Blues Breakers in 1965 when Clapton insisted on using his amp at a level close to his on-stage volume much to the studio’s horror (‘It’s distorted’). Tom Dowd told a similar tale of Cream’s Disraeli Gears session at Atlantic Studios in New York. The undoubted power of Clapton’s guitar sound on those influential records helped studios reach recording compromises so when Hendrix arrived we were ready. A CORNER OF THE ALL PERVASIVE FORCE — Covering this topic without mention of The Beatles is difficult but it is generally thought that the technical boundary pushing came from Gorge Martin with the Fab 4 pushing at the musical side. However, in a recent interview with engineer Geoff Emerick and info from Paul McCartney’s biographer Barry Miles, it appears that McCartney had a serious role on the technical side. He had an ‘experimental studio’ of his own in the mid-60s where he apparently experimented with avant-garde tape editing techniques. It was he who assembled the tape loops and played the faders on Revolver’s Tomorrow Never Knows, which has remained a very influential track. Emerick added that ‘Paul could be very precise as to what he wanted.’ Following the demise of The Beatles, McCartney soon had a solo album out on which he played everything, and engineered. While this wasn’t the first ‘solo’ album it was the highest profile one. November/December 2006

THE SYNTHETIC FRONTIER — Synthesisers were still a clever novelty until 1968 when Wendy (then Walter) Carlos’ album of ‘virtuoso electronic performances’ of JS Bach pieces was released. Painstakingly recorded, single part at a time with Bob Moog in attendance to modify his early instrument to achieve Carlos’ musical requirements, Switched-On Bach was perhaps the first all-synth recording that was intended to be more than just a novelty. Despite the genre, the Moog content caused CBS Records to promote this as part of its ‘Rock Machine’ stable making it a hugely influential demo of all-synth capability. However, it wasn’t until 1972 and Robert Margouleff and Malcolm Cecil’s Zero Time, also created entirely on a Moog (Series III), that we heard original synth creativity rather than mimicry. SCRATCH’S DUB — For much of the 1970s Lee ’Scratch’ Perry’s Black Ark studio was the spiritual home of reggae and dub music. Perry was one of the producers who was responsible for the development of dub and it was just one of the techniques used to create unique sounding tracks with fairly basic gear and a simple 4-track tape machine. The sound was driving technique with bass heavy recordings, complex rhythms and dub carried to extremes. He also blurred the lines between engineer, producer and artist often performing on tracks as required. Having an open mind to different music types his influences were widespread. INTRICATE EXTREMES — A handful of albums by 10cc from 1974 onwards largely defined the sound and capabilities of 24-track recording and a dry acoustic. Well crafted songs that depended upon production for their vitality had hundreds of overdubs, half submerged intricacies, tight tempo changes and use of sonic colours. Aided by a new studio built for this type of recording, Eric Stewart engineered, mixed and mastered these recordings as only a musician intimately involved with their creation, as band member and writer, could. The technical achievement stands even if some of the earlier recordings are difficult to appreciate now. ONE MAN BAND — Mike Oldfield’s Tubular Bells broke new ground as a solo album. Two over23 minute tracks, largely played by himself with only the basics of a studio and few effects other than tape speed, compression and a little echo. While it was musically rather than technically resolution

KEITH SPENCER-ALLEN driven, the recording supports the work well. In later works Oldfield perfected the technical although the musical became less inspired. He was one of the first to combine Dolby SR with the dynamic automation capabilities of the Harrison Series Ten console to craft intricate moving sound scapes where there was as much creativity in the positioning and movement of the instruments as in the playing. STUDIO MAESTRO — The late Frank Zappa’s public image was always a little outside the mainstream but he had a dedication to recording that allowed him to develop huge reserves of studio ability. He bought his first studio in the early 60s and lived there, working 12 hours a day, a pattern that allowed him to produce (and frequently engineer) most of the 60 or so albums be released. The Mothers Of Invention recordings were notable for the way that long and short musical items were blended with sound effects and conversation held together by superlative editing technique as exemplified by 1968’s We’re Only In It For The Money. From 1968 he assembled a portable studio system to record many of his live concerts and these were often blended with studio recordings such that there was no obvious join. He developed a technique that he called ‘xenochrony’ where key instrumental parts from concerts such as guitar solos were added to studio recordings — very difficult in the pre-digital era. In the 1980s he became a master of the Synclavier. WHEELS OF STEEL — Grandmaster Flash was one of the pioneers of Hip-Hop, putting together one of the earliest rap groups, The Furious Five, and developing an arsenal of DJ techniques that enabled the record turntables to be played like a musical instrument, using cutting, scratching, back-spinning and the use of duplicate copies of discs — what Flash calls ‘quick mix theory’. In the early 1980s he put the two aspects together on ‘The Adventures Of...’ which was the first recording to feature sophisticated cuts and scratching. THE POWER QUOTE — And finally, a single phrase ‘To me the recording studio is another musical instrument to be played’, quoted widely in the music press of the late 1960s, unified many of the different strands of creative thinking post Sgt Pepper, giving legitimacy to what the older studio generation often referred to as just ‘noodling around’. We’ve attributed the original words to American rock musician Steve Miller but it has been said by many subsequently. This was either recognition of a creative reality or a warning that it was all downhill from then on, according to your POV, of course! ■ 49

business

Brand, image and value The traditional model for breaking a band is now largely outdated yet there are aspects of the old approach that are as crucial now as they were then. NIGEL JOPSON looks at credibility and the need to build a community.

HERE WAS A TIME when a band’s road to the top, while never an easy ride, could at least be counted on to display milestones. These were clearly visible to young musos, whether slogging past Watford Gap services in a rusty Transit van at 4am, or pigging-out on deep dish pizza after a gig at the Chicago Metro. The first huge, shining monolith to illuminate their path was the imperative of Get-arecord-deal — this would be the one to carry them through all the sweaty clubs, home-made amps, dole cheques and broken promises of managers. Having achieved this imperative, the next mark would be: record-the-album-in-a-real-studio — this would sustain them through uncertain tempos, laryngeal malfunctions and stroppy recording engineers. Having obtained a sliver of plastic displaying their art at its angst-ridden best, or at least containing a few not-too-muffled mixes of songs they thought were good six months previously, the next big step was finding a tour ‘to promote the album.’ If the band was lucky, a clued-up manager might persuade someone at the label to spend the band’s recoupable money buying the group onto a genreappropriate support slot. The idea was that the more people that heard the troubadours perform, the more people would buy the record and the more money everyone would make. Hopefully. Regular readers of this column will realise this business model is well out of date. Back in Resolution V2.2 I revealed that major labels now have an artist break-even rate of less than 1 in 20: that means 19 of those bands are going to be dropped. This ever-slipping rate means that those bands whose art remains stuck to the wall after being flung at it have to sell relatively more to justify their presence on the roster: gold is no longer enough. The indie-label model, with it’s 50:50 artist contracts, soft-launch album releases and maverick owners, has far better success rates and retains more of the creative edge that started to evaporate the moment the word ‘business’ was permanently welded to the word ‘music.’ But it’s more than that: the whole idea of playing music — whether live or broadcast — in order to sell a physical album, is now looking very tired. Last issue in these pages I showcased an entire ecosystem of new music distribution channels, including social networking websites like MySpace and YouTube, blogs, live venues online and intelligent web radios. I wrote about how artists like Beck have made the concept of a ‘finished album’ old hat with portfolios of downloads, DVDs, remix tracks, ringtones, vinyl, online demos and unplugged sessions spinning off from a music project. There are now alternative methods of funding outside of conventional label imprints, including webbased efforts such as Sellaband.com and Artistshare. com. But the nice part about the old band milestones, however flawed, was the inevitability of one step following the next. If you flogged your guts out around the US ‘showcase’ circuit (venues of around 1,000 capacity), you could step up to a support slot on a stadium tour. And if you made time to do all the local radio interviews you could sell more records. There was a certain symmetry to this progress, which barring bizarre accidents ‘best left unsolved’, would lead to more records sold. Now it’s hard to 50

know where to start: which website to choose, which download aggregator, and bands need media as well as music skills, and lots of time to design web pages and answer all those IMs and emails. Or so it seems. I wrote in Resolution V5.4 Biz Bites about Sandi Thom, a singer-songwriter who scored a £1m deal with RCA without playing a live tour. She webcast 21 ‘gigs’ from her Tooting basement. Between 40,000 and 60,000 people apparently watched them on the Internet. There was a major-label bidding war. She said the idea ‘popped into my head’ after her car

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broke down travelling from a gig in York to Wales. ‘I drove back to London, literally with my exhaust pipe hanging on with a coathanger,’ she told the Independent newspaper. ‘Those are the kind of dire straits that make you think, Oh, for God’s sake, there must be some other way to do this.’ Her first single I Wish I Was A Punk Rocker (With Flowers In My Hair) reached number 1 in the singles chart in June, and her debut album entered the charts at number 1 as well. So there’s a modern web-based success story ... But wait ... further investigation reveals Sandi

November/December 2006

business was already signed to Windswept Pacific Music (the up-market publishing home to Beyonce and Pete Townshend) and was writing songs with John McLaughlin (Mixing Rooms, Resolution V5.3), and Cambridge-based PR company Quite Great later confirmed they’d been working on her behalf since June 2005. ‘You can’t just rely on good music,’ said Louise Harris, PR manager at Quite Great. ‘For unsigned artists it’s really important to have the fans and the street teams out there talking about them.’ Street teams? Oh yes, those would be the people who received an email from Quite Great saying: ‘Nearly 8,000 people have viewed Sandi’s official MySpace page… I want to make that 10,000 by Friday so could you all send out myspace bulletins encouraging people to listen to Punk Rocker… I have a nice pile of Sandi Thom promo Albums here which will go out to the top teamers.’ A Sunday Times journalist, a US radio plugger who sent 1,000 emails, a mysterious rich Scotsman and producer Calum Malcolm (Resolution V5.2) all turn out to have had a hand in shaping Sandi’s music before the webcasts. The amount of Internet traffic data originally generated is now being questioned by techies. Which part of the deception most upset people? The mainstream press and big record labels have been manipulated by savvy managers and artists since ‘Colonel’ Tom Parker paid teenagers to scream at Elvis shows. Brian Epstein drove around buying Beatles singles from chart return (pre-SoundScan reporting) shops. If the Colonel hadn’t staged so many PR stunts, Elvis would never have got famous so quickly, but only a fool would argue the ‘King’ didn’t have the talent to sustain, or transcend, the hype. If the talent is there these artists will be successful, whether Sandi Thom, Lily Allen or The Arctic Monkeys — who now claim: ‘... we were on the news and radio about how MySpace has helped us. But that’s just the perfect example of someone who doesn’t know what the f**k they’re talking about. We actually had no idea what [MySpace] was.’ Yes, mate. Behind every MySpace/ Internet success story there is a Colonel Tom Parker, whether it’s a member of the band, a friend of the band, or a hired-hand PR. The original community-based rock act were the Grateful Dead, who pioneered the idea way before any media darling came up with the phrase ‘street team’.

November/December 2006

The Dead served as a key icon of the anti-materialistic 1960s counterculture for more than 30 years, but in their heyday the band had gross revenues of as much as $95 million annually. They attracted an army of loyal fans, known affectionately as ‘Deadheads’, they never had a number 1 hit, encouraged fans to make amateur tapes of their concerts, and sometimes went for years without even releasing records. The Dead established a long-term relationship with their audience, they never let money dictate the music they produced — in marketing terms, they never became a ‘me-too product’ — and how many acts can you say that about, once the grey hairs start to show in the shaving mirror? The Dead looked unkempt and shambolic on stage, their jamming concerts went on for hours, but, behind the stage personas, they incorporated in 1976 with the band as board directors, paid meticulous attention to mailorder ticketing and merchandising, spent millions on light and sound systems, and brought in parking lot vendors as official licensees under QOS scrutiny. The four annual Grateful Dead regional tours had, for the fans, the allure of a spiritual crusade. Whatever you think of the Dead’s music, they followed some very clear marketing principles that can be replicated by any product or artist: tight control of the brand, clear brand-image and a well-defined value proposition. Those three imperatives really go together. Skip one and you risk losing your fan base. I can still remember the reaction to the news, in 1981, that the Rolling Stones had become the first band to accept corporate sponsorship for a tour. Jovan Musk paid $500,000 dollars for the right to be associated with the Stones’ tour after a deal with Schlitz Beer fell through (Close thing on which smelt the worst. Ed) Was that really the best Prince Rupert Lowenstein could come up with for the lads? The cheapest scent … available at any 5 and dime store. My American pals, who hero worshipped ‘the greatest band on earth’ far more than I did, were devastated. How many record sales have been lost over the last quarter century because of that ‘sell-out’? From a moral perspective, it’s not for the artist to defend themselves from such accusations — we’ve all got to make a living somehow. However, ‘It can hurt a musician’s credibility if they choose to play the kind of event that crosses their fan base’s moral/ ethical boundary lines,’ says Joan Hiller, publicist for

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Sub Pop Records, an indie-label based in Seattle. David Bowie is one of the most business-savvy musicians around. He’s reinvented his stage persona countless times, he was one of the first artists to own the copyright to all his songs and recordings, and the first artist to securitise his IP (Intellectual Property) assets by issuing bonds backed by their value. ‘Music itself is going to become like running water or electricity…’ he hypothesises. ‘So it’s like, just take advantage of these last few years because none of this is ever going to happen again. You’d better be prepared for doing a lot of touring because that’s really the only unique situation that’s going to be left. It’s terribly exciting. But on the other hand it doesn’t matter if you think it’s exciting or not; it’s what’s going to happen.’ The big money for musicians, in the immediate future, is going to come from live performance, merchandise sales, personal appearance, DVDs, publishing, licensing to film or advertising ... and community memberships. To pinpoint the issue, I turn to Harvey Goldsmith, the UK’s best known rock promoter. ‘We’ve forgotten that what kept our business going was always having demand outstrip supply, which is an important facet of our mystique,’ he said in an interview before being presented with the 15th Music Industry Trusts’ Award. ‘If you can’t have something, you want it. That’s a big thing that’s missing, because people are too greedy. We don’t let raw talent develop enough. We seem to spend more time shovelling it out as fast as we can to get a return on it, rather than letting it develop naturally.’ Harvey’s words are relevant to our topic because membership of an exclusive community, by definition, bestows an element of mystique on the member. Any band that wants to carve a rewarding career in music will have to concentrate on this community aspect, and would do well to take a leaf from the Grateful Dead’s book. Between 1999 and 2003, a tough era for the mainstream music industry, Colorado-based band the String Cheese Incident revenues grew from $2m to $14.5m annually. The band had 45 employees and its own record label. In Resolution V2.4 I described how prog-rock band Marillion persuaded 12,674 stalwart fans to pay £16 for an album that hadn’t been recorded, later the band hired Butlins camps to stage special events and made a DVD of the live performance in a record-breaking 48 hours. The old music industry model is all about distribution, artists are compelled to give up control of their rights to gain access to the marketplace. The access part is becoming less and less of an issue, but control of the ‘brand’ has become more important. It used to cost a lot of money — beyond the means of any young band — to record and promote music. Music was filtered through the label system and only a few artists became stars. The new model enables practically anybody to record and distribute music. For example MySpace, already a powerful promotion tool for the 3 million artists who have pages on the site, is setting itself up as a music marketplace. Under a deal with Napster founder Shawn Fanning’s Snocap. com announced in September, artists who previously could only place four promotional tracks for streaming on their page will be able to set prices for songs and sell MP3s to fans directly through MySpace. The really challenging part is the promotion … musicians and producers need to break through the background noise and stand out from the crowd … without having significant recording revenue. In a world were there is too much information and too many options people will cluster into communities. In an era where a PR stunt can be rumbled in an instant and sprayed across the world via blogs and web-boards, credibility and community have never been more important. ■ 51

broadcast

Multichannel mic solutions for OBs SoundField’s B-Format microphone systems have long been capable of capturing audio in mono, stereo, or surround. Now, as HD broadcasters start in earnest to produce audio in 5.1, PIETER SCHILLEBEECKX of SoundField introduces a new digital system that provides a simple and integrated means of creating surround from a single mic.

OUNDFIELD MICROPHONES ARE not new and there has been a mic of that name in existence for decades, well before the SoundField company was founded in 1993. The microphone is based on the principles of Ambisonics, conceived by acoustics and recording pioneer Michael Gerzon in the 1970s as a means of going beyond the stereo and quadraphonic reproduction systems of the day, and including height information in the captured audio signal —- a development still not matched by digital surround sound systems today. From the late 1970s, the SoundField microphone has been capable of capturing a three-dimensional soundfield (hence the name) accurately and repeatably without recourse to psychoacoustics. It’s also been capable of rendering the captured audio as a listenable stereo or mono signal without any of the audible phase-related problems that frequently bedevil attempts to render three-dimensional sound recordings suitable for reproduction on conventional hifi systems. The SoundField has made a name for itself, especially among classical recordings engineers, as an accurate microphone capable of producing recordings of great clarity in stereo and in multichannel surround. This impressive imaging is due to the phase response accuracy of the SoundField microphone. However, there have always been more affordable ways of recording in stereo, and for many years the SoundField mic’s appeal was limited by the lack of widespread commercially available systems that could reproduce its three-dimensional audio output without first folding it down to stereo or mono. All this has changed with 5.1. As TV companies 52

The four signals that make up the SoundField B-Format, with the three fig-8 responses X, Y and Z at right angles to one another, and the omnidirectional reference W.

gear up for the move to High Definition (HD) broadcasting, for which 5.1 is required, the number of broadcasters who require a quick and easy method of recording in 5.1 while maintaining perfect mono and stereo compatibility is increasing fast. SoundField microphone systems can produce high-quality mono, stereo, mid & side and surround sound recordings in a variety of formats, all from a single mic, and simultaneously if required. This is useful to broadcasters who need to navigate the transition from stereo to multichannel audio carefully over the next few years if they’re not leave their viewers behind. Although the simultaneous output of all these audio formats has been possible with SoundField systems for many years, the company’s designers have spent five years refining and improving its technology to produce a single-mic surround-ready system for HD broadcasters. This new microphone system, which is also the first SoundField to carry out all its processing resolution

and audio format decoding in the digital domain, is called the DSF-2. For the above reasons, SoundField products are becoming something of a standard choice for capturing surround at large-scale events that are broadcast live in 5.1 and stereo. Examples include high-profile sports fixtures like UK Premiership football matches and the Ryder Cup golf championships for Sky Sports, as well as challenging musical OB events such as BBC Scotland’s coverage of Glasgow’s international bagpipes competition, the World Pipe Bands Championship. This year, OB truck companies like Visions, CTV, Arqiva, and Observe in Eire have equipped their HD OB vehicles with SoundField systems. SoundField systems employ a patented multicapsule microphone, in which four sub-cardioid capsules are mounted in a tetrahedral array, and one of several hardware processors. The 4-channel signal produced is a proprietary form of output known as SoundField B-Format, and its four constituent signals are known respectively as W, X, Y and Z. The last three represent the soundfield around the microphone in three dimensions — as though it were recorded by three fig-8 mics placed at right angles to one another — while W constitutes an overall reference signal as though recorded by an omnidirectional capsule. The initial processing also corrects for the fact that the four signals have been recorded at a slight physical offset from one another because it is impossible to mount the capsules at exactly the same physical point in space. However, because the distance of the four capsules from the centre of the tetrahedral array is small, known and constant, it is possible to process the output of each capsule to generate four signals that correspond to those that would be created by recording at exactly the same point at the centre of the array. This eliminates phase problems between the four constituent channels of the B-Format signal, and is central to the SoundField concept. It allows the four phase-coherent channels to be combined without any unpleasant-sounding phase artefacts, the usual drawback of multimicrophone or other multicapsule systems. By combining the four signals making up the B-Format in different proportions, it is possible to generate an audio output in any format from mono to surround, including any future three-dimensional formats that are yet to be defined. Simultaneous decoding from B-Format to several different audio formats is also possible. If the 4-channel B-Format signal is recorded the output format can be determined at a much later date by running it through the B-Format processor. This makes the B-Format an excellent archive format. The process of combining the four phase-coherent B-Format channels in different proportions can be controlled from switches and knobs on SoundField’s software and hardware B-Format processors, and allows users to change the mic’s pickup pattern continuously from omni through cardioid to fig-8. The virtual orientation of the mic — the direction it appears to face — can also be continuously controlled from the processor. This is useful in OB situations, November/December 2006

broadcast where the mic often cannot be physically accessed or adjusted once broadcasting has begun. Such ‘audio zooming’ and ‘virtual rotation’ of the mic can also be carried out by postprocessing a stored 4-channel BFormat signal with a SoundField hardware processor or software plug-in. This flexibility and format independence has been popular among HD broadcasters of late and the DSF-2 was designed specifically to appeal to the broadcast market, through listening to input from Sky Sports and HBS (Host Broadcasting Services, the company created to transmit the FIFA Soccer World Cup). Specifically, this resulted in the use of 75ohm AES3-id unbalanced coaxial cable to connect the DSF-2 decoder to its broadcast vehicle, rather than a standard 110ohm balanced XLR. This allows the reliable transmission of the audio data over far longer cable lengths — depending on the exact coaxial cable used, the decoder can be up to 1000m from the broadcast truck. Moreover, the cable supplied with the DSF-2 to connect the microphone head to the decoder can be a further 300m long, allowing the mic to be placed reliably at total distances of up to 1.3km from its OB truck. This flexibility is popular with OB companies, who often can’t position their trucks close to venues. Another feature requested by broadcasters was the presence of a simple analogue stereo output on the DSF-2 decoder itself. This can be used to generate the ambient element of a headphone monitor mix for commentators at an event, for example. Although the concepts of the B-Format, the tetrahedral multicapsule array and its phase-coherent signals remain at the heart of how the DSF-2 works, it represents a modern implementation of the original theory and as such produces an even better sense of localisation, phase accuracy, and high-frequency sonic detail. SoundField has completely overhauled the analogue electronics in the DSF-2 microphone; a low-noise, low-distortion JFET signal path is now in place throughout. Relayswitched, low-tolerance gain components are also used for the 4-channel mic preamp, rather than multigang pots or VCAs that would have resulted in The Rotate controls on the DSF-2 decoder higher distortion and noise figures. Newly devised allow the mic to be ‘turned’ without capsule-matching methods physically altering its at the manufacturing stage position, by altering add to this, ensuring that the balance of the Bchannel matching, a vital Format signals with respect to one another. part of the B-Format concept, is even better in the DSF-2 than in previous SoundField mics. Detailed study into the interaction between capsules and the microphone bodywork has resulted in more constant polar patterns from one capsule to the next. And the capsule array itself has been improved, the physical spacing between the four capsules having been further reduced (while maintaining the capsule diameters for reasons of audio quality) so that less correction is required to produce the ideal single-point coincident signals. The mic now uses discrete low-noise voltage regulators for its bi-polar rails and the polarising voltage, and incorporates balanced line drivers on its outputs capable of driving 600ohm loads -— hence the ability to use longer cable runs. The mic has better input tolerances all round. There’s a new JFET limiter at a fixed threshold of 3dBFS, which is designed to work on all four input signals equally to maintain spatial aspects November/December 2006

when it’s active, and the inputs have an extra 6dB of provide 5.1 crowd and venue ambience for its HD analogue headroom once 0dBFS has been reached. coverage of UK soccer premiership matches. Sky As mentioned earlier the DSF-2 is the first recently specified the DSF-2 for use on all of its HD SoundField product to use digital signal processing sports coverage. and employs an Analog Devices SHARC for improved Michael Gerzon’s inventions were ahead of his time. accuracy, channel matching and repeatability. Over-specified for stereo recording at its invention the However, in practice it is the DSF-2’s ease of use as SoundField microphone is only now, three decades a simultaneous 5.1 and stereo-compatible mic (over on, truly coming into its own at the vanguard of long distances) that is making it a success among surround. And the 21st-century improvements made broadcasters. In mid-2006, The Host Broadcast to the hardware during the creation of the DSF-2 are Services installed DSF-2 mics at all 12 German stadia designed to safeguard the viability of the SoundField/ involved in the FIFA World Cup and used them B-Format concept for many years to come. ■ to broadcast ambience in stereo (for SD and radio signals) and 5.1 (for the televised HD transmissions) Contact simultaneously to an audience of some 40 billion. SOUNDFIELD, UK Having had an influence on the development of Website: www.soundfield.com the mic, Sky Sports has 14/6/06 permanently12:52 installed cricketad_RES.qxd PMthePage Tel: 1 +44 1924 201089 microphone at several major UK football grounds to

....laying down the LAW

SADiE LRX Location Audio Workstation

The remarkable new SADiE LRX has redefined the meaning of the LAW – the Location Audio Workstation. It is as effective in capturing original soundtracks for film and television production as it is for producing location audio recordings for release on distributed media such as CD or DVD or for any application where portability is key.

SADiE LRX on location in Mumbai, India with BBC sound supervisor Rob Miles for Sport Relief "All out for India" celebrity cricket match.

The flexibility of the LRX hinges on its ability to utilise a standard laptop running Windows XP® via USB2 as the host computer together with combinations of the same high quality i/o cards as the SADiE H64 multitrack workstation. A tactile hardware control surface is employed, incorporating a small assignable mixer and full editorial interface,plus dedicated transport keys. This powerful combination is supplied with a tailored multichannel version of the SADiE on-screen graphical user interface in addition to the full SADiE V5 software system. Timecode and professional genlock facilities are incorporated and a video stream may be simultaneously captured for playback or on-set ADR. The system supports a wide range of industry file interchange formats, plus a second external drive may be simply attached via USB2 or Firewire to mirror recordings and provide simultaneous safety copies. Contact your nearest SADiE dealer or main office and visit our website for further details and a brochure.

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resolution

DIGITAL PRECISION

broadcast

Compaction and compression A couple of issues ago, the editor drew attention in his leader to a problem that has been occupying Master Audio’s MARTIN SPENCER’s working days (and nights) for well over a year. Why does compressed audio sound so downright annoying, and is there anything that can be done to make it sound better?

T’S MY CONTENTION that more than one type of compression is responsible for the awful sound that we all have come to know and detest; further, that the combination of two different types of ‘compression’ is particularly deadly. When we refer to compressed audio, the delivery media that spring to mind will probably be MP3 players, audio streaming via the Internet, and DAB digital radio. But rather tellingly, Zen also includes ‘CD re-masters and a whole catalogue of modern releases’ in his tirade. Last time I looked, CDs still delivered an audio bit rate of some 1411kbps, the same as they did on their introduction to the market back in the early 1980s. So what precisely do we mean by ‘compressed audio’ anyway? Let’s start by eliminating some slippery language (© Bob Katz). What we have here is one word, ‘compression’, used for two entirely different processes. On the one hand, we have good old dynamicrange-reducing compression. The word has been used this way since the 1950s or probably earlier, and there seems no good reason to break with tradition. On the other hand is the modern connotation of stripping allegedly redundant information out of digitally coded audio or video to reduce the number of bits required to transmit it. I’d like to make a proposal. Could we call the bit-rate-reduction process ‘compaction’ instead, so we have a distinction between the two processes? Thus, an MP2 or MP3 encoder becomes known as a compactor. There’s more slippery language to deal with. Bear with me just a little longer — getting our terms right is going to be worth it. ‘Compression’, as a catchall term for manipulation of dynamic range, is still so vague as to discourage proper understanding. Compression, as we all do know really, is when we squash the dynamics but leave a fraction of the original change in place; done by fiddling with the ‘slope’ knob of course. Contrast this with a limiter, which once above its threshold would erase all trace of the original dynamic ‘eventually’, because the time constants that both compressors and limiters use will ensure any changes in gain are applied gradually. Another term commonly encountered in discussion of broadcast audio processing is AGC or automatic gain 54

control -– this is a compressor or limiter with very long time constants indeed. Moving to the weapons-of-mass-destruction department in the armoury of brutal audio processing tools, we have ‘clipping’. This is a compressor or a limiter, but without any time constants to soften its impact. A compressor without time constants is called a soft clipper. A limiter without time constants is called a hard clipper. A relationship without time, Constance, is called a one night stand. Before we look at exactly what can be done to improve the quality of compressed and compacted audio, it is worth having a look at the broadcast processors of the past, tracing the evolution of techniques for radio processing that have now found their way into the mastering house. In the beginning, there were just two processors: a gain rider and a peak limiter. Gain riding was not carried out in the racks room, it was a job for a professional in the control room, who might even be armed with scripts and scores, and who certainly made a judgement of level based on his ears, with metering mainly for confidence. Final peaks had to be controlled to prevent over-modulation, and this was done by a 19-inch rack box, used to control brief peaks that were too fast for the manual operator to handle, and to minimise the effect of human error. This system gives good audio quality, and the dynamic range can be artistically tailored to fit predetermined limits. Although probably preferable to any automatic system, it’s not really a viable option in the multichannel age. The thought of serried ranks of Levellers sitting at Master Control all with their headphones on is a nice one though. In the 1950s and 1960s along came increasingly crowded AM bands and FM with pre-emphasis. This is where the trouble started for the audio. Now don’t get me wrong, FM was a wonderful invention, and we should all be grateful to Edwin Armstrong for pioneering it. It was just an unfortunate side-effect that narrowing AM receiver bandwidths to cope with interference, and introducing de-emphasis on FM to cope with its noise spectrum, both contributed to an eventual need some 30 years later to make the broadcast programme signal brighter. Of course, the root of this was back in the days resolution

when popular music meant big-band, cabaret, musicals, rock and roll, and then the Folk and Blues revivals. In the days when the system standards were developed, high-frequency components of the programme did not often tax the transmission system in the way they later would but the seeds of an unfortunate future problem had been planted. A succession of broadcast audio processing techniques were developed from the late 1950s; we’ll review these very briefly here, since a full history of broadcast audio processing would be an article (heck, a book!) in its own right. Many of us in Europe will remember Radio Caroline first running an Optimod on AM, and the UK landmark came when Capital Radio first installed its Inovonics 250 for FM; exciting times, but outside the scope of this article. Automatic processing for broadcast first started with wide-band AGCs with sufficiently adaptive response to handle a really wide dynamic range at the input; still essentially an ‘engineering’ tool rather than something that radically changed the sound, but very useful nonetheless. Devices of this type, such as the CBS Audimax, were commonplace in the USA during the early years. In the UK the BBC and even the IBA continued to frown on audio processing, insisting that all stations stuck to what Aunty thought best: conscientious manual gain riding at the studio and automatic safety limiters at the transmitter sites. In the 1970s two novel techniques were pioneered in the USA: multiband audio processing (Mike Dorrough) and overshoot compensation (Bob Orban). Both techniques were pivotal in the advance of FM radio. Multiband audio processing was originally conceived to get around the problems of wide-band limiting -– pumping, and bass peaks causing hole-punching and ducking. But multiband processing soon became, perhaps more importantly, a means to add boom and sizzle for those (presumed the majority) listening on poor speakers. Further, it was possible automatically to tailor the amount of EQ to the source material, previously impossible with conventional ‘static’ EQs. This amounted to a form of remastering tracks that were either ‘EQ thin’ or ‘EQ rich’ towards a common standard, namely the signature sound of the radio station in question. November/December 2006

broadcast Overshoot compensation addressed the problems of loudness-robbing overshoots, combining hard clippers with low-pass filters in a way that optimised timedomain (overshoot and thus loudness) and frequencydomain (bandwidth) performance simultaneously. This was the first really powerful loudness boosting technique that did not rely on extending the source bandwidth — otherwise known as splattering the band. Of course, it came with a fairly hefty distortion penalty when pushed hard. In the 1990s, the availability of suitable DSP allowed the introduction of even more powerful techniques for greater loudness (RMS level) with reduced distortion. These techniques mostly revolve around two novel abilities: look-ahead and complex adaptive systems. Both were theoretically possible in analogue, but only really became practical with digital. So what are the problems inherent in ‘traditional’ audio processors? I’m going to simplify here in the interests of brevity but first is the widespread use of limiting (as opposed to compressing) even in slow automatic gain control and compressing functions. This tends to erase programme and musical dynamics; not a good idea. This is presumably because of a perception that loudness is of such dominating importance that we must throw subtlety out with the bathwater. Perhaps, in this age of HD video and audio, it’s an idea whose time has come and now gone. Second is the unholy alliance of the multiband compressor with the clipper. Any consideration of multiband processing must focus on how the signal is treated after the bands are recombined. Unfortunately, it turns out that multiband compression does not generate anywhere near competitive loudness on its own, so it’s invariably followed by a wide-band dynamic limiter and/or a clipper. Sonically, this is a very interesting combination. Multiband and dynamic processes tend to make the sound mushier, reducing impact and transient clarity. However, passing the recombined multiband output through a clipper, a device that can boost the fundamental output levels beyond 100% while also adding harmonics, can go loud while restoring some of the ‘slam’, perceived transients and brightness to the signal, at the expense of gross intermodulation. This is, in essence, what most processors do (even your humble single-band outboard compressor-limiter typically includes a clipper, if you engage the limit function). It’s one reason you can still hear drums

November/December 2006

reasonably clearly despite all the dynamic squashing going on, but it’s also the reason for all the distortion. Backing off the processing to get rid of distortion is often unsatisfactory — the loudness benefit seems to disappear almost as fast as the distortion! Third, it is common knowledge in radio engineering circles that clipping should not be used in an audio processor that is feeding a compactor. The reason usually given is that encoding the clipping distortions (harmonics and intermodulation) will waste some of the bit-budget of the encoding engine in the compactor. The comment was, I believe, originally voiced by Bob Orban many years ago in a technical paper, but has now been used by more than one company in the advertising of their digital radio processors. These digital radio processors use various forms of look-ahead and other ‘distortion cancelled’ techniques allegedly to mitigate the encoder stress effect. However, in many cases the waveform is changed sufficiently rapidly that a following compactor can’t cope without introducing considerable additional audible distortions — the ‘burbling’, washy, smeared sound. This is particularly a problem for the UK DAB system, encumbered as it is with ancient Layer II at, typically, only 128kbps. There’s no doubt that avoiding clipping is a good idea but there is a lot more to the story than this. Taking what is really a 95% traditional radio processor and bolting it into the digital transmission path is the single avoidable act most responsible for the awful sound we are becoming disgusted with. And unfortunately, because ‘loudness wars’ are now being fought in the trenches of mastering as well as broadcast, the poison gas of over-processing is tending to affect every single delivery channel. But let’s be positive. The first part of the solution is to eliminate all unnecessary processes that are exacerbating the problems of compaction. This is a system-wide problem that extends to all links in the chain, and particular attention must be paid to maintain signal quality in contribution feeds, storage, mixing, distribution and transmission -– a lot of places where the audio quality can be impaired. Here we will concentrate on the audio processor, which is one of the more critical links in the chain. Out go multiband compressors and out go ‘main’ (combining) clippers. What is then needed are specific algorithms to deal with the following problems: normalising average level, powerfully if called for; retaining some musical dynamics and natural flow; avoiding hole-

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punching and pumping; and peak control without clipping, aliasing or inter-sample overload. If we pay heed of all the above points, we can have an audio processing system that is capable of smoothing out levels without sounding awful, and that therefore makes the most of whatever delivery channel is available. Of course, the trade-off between the first two of those requirements must be set by the user, and in doing that, best results will be obtained if the operator is asked to be aware of his levels and machine intervention reduced (but not eliminated). The fader position will still have some influence on the final loudness, albeit with extremes of loud and quiet ameliorated, and this can be used to good effect in the right hands. Returning in closing to the more general question of compressed audio, the current situation is indeed awful, but there are also one or two possible rays of hope. One is AAC plus compaction, which does seem to offer a substantially better listening experience than the others, especially at low bit-rates like 128kbps. Even if digital radio broadcasters insist on subdividing their digital delivery channels until they sound dire (the UK is already there, and it’s looking pretty dicey for the US ‘HD Radio’ system with maybe 48kbps of AAC+, due to the likely adoption of multicasting), the competition may not follow suit. There is always the possibility of something better supplied by alternative delivery channels: iPods, digital TV in various forms, satellite radio, and Internet streaming are all capable of supporting at least 128kbps per stereo audio channel. That’s not to mention DMB and other pioneering forms of 3G delivery. As new means of delivery become increasingly accessible, competition may act as a self-regulatory mechanism. The new fear becomes the competitor who offers similar content, with the same accessibility, and better quality. Remember, it first happened that way when microgroove records ditched the 78, and it certainly happened that way again when FM radio was the new kid in town. Most recently, the iPod has made much greater impact than cheap MP3 players. Perhaps digital delivery does have to maintain decent sound quality after all. ■

Contact MASTER AUDIO LTD, UK: Email: martin@master-audio.com

craft

Stereo pair and multimic mix comparison JOHN LA GROU from Millennia Media explains the approach taken on a landmark orchestral recording and gives a rare opportunity for readers to compare the results of stereo pair and multimic mix versions of the same performance directly for themselves.

HIS HISTORICAL PERFORMANCE of Handel’s Messiah was recorded live in December 2004 in the Mondavi Center for the Performing Arts located on the campus of the University of California, Davis. The American Bach Soloists Chorus and Orchestra, led by music director Jeffrey Thomas, performed the predominantly unheard autograph score version of 1741, which includes alternate versions of various arias and choruses and extended repeats, previously cut from other hybrid versions over the years. Complimenting the American Bach Soloists ensemble during the recorded performance were four featured soloists: soprano Arianna Zukerman; tenor Steve Tharp; countertenor Daniel Taylor; and baritone William Sharp. Many have asked for a ‘tutorial’ series on large ensemble acoustic recording. Perhaps the best way to convey the performance of different recording techniques is to simply listen and compare for yourself. You’ll be able to listen to samples from a live performance by the American Bach Soloists, playing on period instruments, and it is among the world’s finest ‘baroque style’ orchestras. A mix of this recording was released on the Delos label (DE-3360) for those who would like to hear the full version. Rarely, if ever, in my audio career have all the ideal acoustic production elements shown up in one place: a truly masterful performance in a world-class recording space captured with extremely realistic engineering techniques. Producer and music director Jeffrey Thomas has allowed the ‘intangibles’ to flourish: organic, human, visceral qualities that are so often lacking in ‘overly formal’ acoustic music, especially Messiah readings. This recording seems to have it all: the choir dynamics and intonation are stunning; the ensemble pacing and phrasing is masterful; and the choir and orchestra gel as one living instrument. Twenty microphones were arranged on stage as explained in the microphone chart and the purpose of this exercise is to give Internet links to allow you to compare the acoustic properties of multimic versus two-mic recording technique. The main stereo microphones heard on these recordings are a pair of Josephson 617 bodies with Gefell MK-221 omni measurement capsules. The recorder is a Radar SNyquist sampling at 44.1kHz and 24 bits. A quad of 130-volt DPA 4012 mics are positioned on the choir. All mics are fed to Millennia mic preamps and the mixes were produced on a Sequoia V8 workstation. Word length reduction (24 to 16-bit) is via POW-R. The stereo pair recordings you’ll hear if you follow the links are unprocessed, straight from Radar with the natural room reverb. The main stereo mics were fixed 290cm above stage floor, spread about 45cm wide, pointed about 45 degrees off-centre (left and right), and pointed about 20 degrees down from stage parallel. This effectively splits the orchestra for even acoustic coverage — omni mics are not omnidirectional at all frequencies, hence pointing does matter. The DPA 4012 high voltage mics were stationed evenly about the choir 340cm above stage floor with the choir on risers. The point was nearly straight down into the heart of each choir section and slightly to the choir’s front. Vocal soloists used 56

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November/December 2006

craft Josephson 606A-KM25 hypercardioid mics. Mics were placed about 70cm in front of the vocalist, about 30cm down from mouth height, and pointed up appropriately. Mics were placed down from the mouth plane not for sonic purposes but to give better audience line of sight. Ideally, we would want to be tighter on-axis. The debate over â&#x20AC;&#x2DC;pureâ&#x20AC;&#x2122; stereo orchestral recording versus multimicrophone mixes has raged for decades. Here is a rare opportunity to hear both simultaneously as a controlled experiment. We all know of poorly executed examples in both â&#x20AC;&#x2DC;puristâ&#x20AC;&#x2122; and â&#x20AC;&#x2DC;mixedâ&#x20AC;&#x2122; categories. Often, twomic recordings employ widely spaced (large AB) omnis, offering a tremendous sense of orchestral spaciousness but sacrificing accurate stereo imaging and stage placement. Conversely, coincident two-mic orchestral techniques (XY) can deliver exceptional spatial imaging, but often at the expense of a flat and lifeless ambience and loss of depth. In this Messiah production, we chose closely spaced omni microphones (small AB) as our main stereo pair. We find that this is often the ideal â&#x20AC;&#x2DC;tradeoffâ&#x20AC;&#x2122; between orchestral space and imaging. Even

though we had numerous mics open, I estimate that more than 80% of the mixed programme remained on these two microphones. Alas, this is often a hallmark of convincing mixed orchestral recordings -â&#x20AC;&#x201C; spot mics are used gently and sparingly, like spice. Some of the finest contemporary examples of mixed orchestra are found in the motion picture industry. Todayâ&#x20AC;&#x2122;s leading movie scoring engineers include Shawn Murphy, Bob Fernandez, Simon Rhodes, Dennis Sands, Steve Kempster, Armin Steiner, Alan Meyerson, Michael Farrow, Frank Wolf, and John Rodd. Record companies, such as Telarc International, are also making fine multimic recordings, with engineers like Michael Bishop, Jack Renner, and Tony Faulkner leading the way. This, of course, brings up the subject of â&#x20AC;&#x2DC;surroundâ&#x20AC;&#x2122; orchestral recording, a topic that I hope to cover in future discussions. â&#x2013;

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Recording example links (16/44.1) Here are comparison audio examples of the identical performance captured with raw two-mic and mixed multimic techniques. GLORY Stereo pair only www.mil-media.com/media/Glory.wav Multimic mix www.mil-media.com/media/GloryMix.wav CHILD Stereo pair only www.mil-media.com/media/Child.wav Multimic mix www.mil-media.com/media/ChildMix.wav GOODWILL Stereo pair only www.mil-media.com/media/Goodwill.wav Multimic mix www.mil-media.com/media/GoodwillMix.wav DANIEL TAYLOR â&#x20AC;&#x201D; COUNTERTENOR Stereo pair only www.mil-media.com/media/Countertenor.wav Multimic mix www.mil-media.com/media/CountertenorMix.wav YOKE Stereo pair only www.mil-media.com/media/Yoke.wav Multimic mix www.mil-media.com/media/YokeMix.wav INTERLUDE Stereo pair only www.mil-media.com/media/Interlude.wav Multimic mix www.mil-media.com/media/InterludeMix.wav AMEN-END Stereo pair only www.mil-media.com/media/Amen.wav Multimic mix www.mil-media.com/media/AmenMix.wav

November/December 2006

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W W W D P A M I C RO P H O N E S CO M

31-10-2006 15:10:28

katz’s column

Smoke and mirrors Continuing his mission to bring power to the people, BOB KATZ indulges in a little acoustic empowerment.

Y WRITING PHILOSOPHY could be summarised as Power To The People. I want to empower you, to help you gain control over your analogue and digital audio life without having to learn detailed circuit design, DSP, or college-level acoustics. We do have to know the basics, and so I thought last month’s column would not be too controversial, but apparently I was wrong. A very polite Dane, Elberg of ELT, maker of mixers and preamplifiers (www.elberg-elt.dk) wrote to say that it may not be so easy to diagnose every balanced output circuit from its audible behaviour. I claimed that by grounding and opening pin 3 while listening to a test tone you could tell how to unbalance a circuit. But Elberg pointed out to me that the SSM chip specifically requests that pin 3 be grounded to avoid a degraded noise floor on pin 2. Not having an SSM-equipped piece of gear I can’t tell you whether it’s easy to hear the noise increase by listening on headphones, so please stand by until someone tries and reports. I hope we can still empower you with a short set of listening tests! Still, the SSM chip does not act like a transformer, it loses 6dB headroom; when unbalanced I advise using 0dBu or lower nominal level, as it clips around +22dBu.

Fig. 1. Obstacle course.

ACOUSTICAL EMPOWERMENT: MICS AS THEIR OWN OBSTACLE COURSE — Let’s move on to acoustical empowerment, especially those pesky first reflections. I’ve caught a couple of reviews of a clever little flash recorder by a major manufacturer that incorporates a stereo mic pair attached to the recorder arranged like Figure 1 (You don’t mean the Sony PCM-D1, do you, with its ‘Electret condenser microphones mounted in X-Y configuration’? Ed). I can’t believe that neither the designers of the recorder nor any reviewer has noticed the fatal flaw in this mic arrangement: it makes a terrible stereo image! (But it does add to the visual image. Ed) The head and body of one microphone interferes with the pickup of

Fig. 2. Much better.

the other! The correct way to set up a coincident pair is one above the other, like Figure 2. In this way, the body of one microphone does not seriously interfere with the stereo image in the horizontal plane, though sounds above or below the horizontal axis of the mic will have a different frequency response due to interference from the other mic’s body. Even better is to slide one mike a bit further in a near-coincident way so that the slits in the capsule which determine the cardioid pattern are not covered by the body of the other mic. These classic AKG-C451s are axial or ‘front address’ mics, but the principle remains for radial or side address mics. I remember trying to use a pair of (side address)

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Mike Felton Best Audio on a TV Programme Later with Jools Holland

from

Graham Wild of Films at 59 and Kate Hopkins of Wounded Buffalo Best TV Film Documentary Planet Earth

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November/December 2006

katz’s column Neumann U89s as a main ORTF pair on a big band. I figured that since the U89’s body and capsule is smaller than the U87, baffle effect would not be serious, but I heard problems in the vertical polar pattern caused by the body of the mic itself. According to Lou Burroughs in Microphones: Design and Application (sadly, long out of print, 1974), ‘in this type of design, the [long] section of the case obstructs and deforms the off-axis response to the point where some units are practically omnidirectional in this region, despite the design intentions.’ I had to carefully adjust the U89’s angle to get acceptable high-frequency pickup of the trumpets, who were in the top, back row, and the saxes, who were in the lowest, front row. A front-address mic would not be susceptible to the baffle effect for instruments that are reasonably in front of the mic. Something to think about, though maybe this colouration is desirable as I’ve successfully used large diaphragm side-address mics in distant pickup for years.

console horizontal vs console tilted so the back is 4-inches higher than the front. When the console is tilted, the sound from the loudspeaker is striking a ‘glancing blow’ so the first reflection arrives at the listener’s lower chest rather than at his ears. The results can be seen in Figure 5. In green is ‘no console’, in red the console parallel to the floor, and in blue, the console tilted. The level console surface causes anomalies at some frequencies as much as

4dB! Tilting the console greatly corrects the errors at most frequencies, usually to 0.5dB or less. This 12th-octave display is easy to read, though some blue traces are covered by the green, I can assure you they are very close to the green in all bands, though I cannot explain why the raked console measures worse in one band, around 800Hz. Baffling, eh, or perhaps just reflecting? Until next time. ■

Information

Fig. 5. At 7ft listening distance: console raked, not raked, and no console.

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

NEARFIELD MADNESS — A friend who is an acoustician/room designer tells me he’s getting tired of hearing clients blame their sound problems on his room design, when they insist in placing the speakers in the worst place in the room. Every time I see a magazine photo of bridge-mounted nearfields I vow to mail the engineer a mirror. Because if you can see the tweeters in a mirror placed on the desk surface, you’re in big trouble. The practice of bridge-mounting speakers introduces many frequency response anomalies that engineers try to compensate for in their mixes and which I as a mastering engineer often cannot fix. I decided to measure just how bad that compromise is. In Figure 3, I placed a simulated console surface (a large piece of plywood) just under a loudspeaker and the test microphone at the suspected listener’s ears. Figure 4 shows the measured frequency response

Fig. 3. Simulated nearfield desk. (Nice. Ed)

result in violet, and when we remove the simulated desk, the much smoother response in green. The frequency-response errors are as much as 7dB! Case closed? Not quite. We can never completely remove the effects of first reflections from a console or any hard room surface, no matter where we place the loudspeakers, the desk and the listener, so we must be ever-vigilant. I moved the listening position to ‘midfield’, 7-feet from the loudspeaker, the desk a very low 23 inches high so it’s crowding my knees, and the test mic at ear height on axis with the tweeter. I then compared measurements with no console vs

Fig. 4. Nearfield console vs no console. November/December 2006

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technology

The Crystal Core Engine

Media Highway

While the demand for larger systems, better quality and lower cost grows exponentially, the professional media industry continues to rely on two decade-old technology, says to Fairlight’s STUART DEMARAIS. He says Fairlight has made a breakthrough with new audio and video products incorporating its CC-1 Crystal Core Technology.

AIRLIGHT’S CRYSTAL CORE Technology (CC-1) processes data in a massive Field Programmable Gate Array (FPGA), architected into what amounts to a purpose-built media processing chip. FPGAs deliver power at price points that will ultimately obsolete the established CPU and DSP/Time-Slice-Bus architectures. With very low processing latency and enough speed to provide smooth analogue-feel tactile response, this technology is fast becoming the bench mark platform for the 21st century. Fairlight’s CC-1 is a media-optimised FPGA architecture that harnesses the step change in performance to support improved quality, faster job turnarounds and the development of new creative opportunities for many years to come. Crystal Core is an aggregation of IP cores, the results of 20+ man-years of development combined with Fairlight’s 150+ man-years of experience as a digital audio pioneer. In recent years FPGAs have emerged as the frontrunning computer engine. They can be programmed so flexibly that, aside from a host PC and some standard memory and ‘glue’, one chip forms an entire system. And a powerful one — single-chip applications include a 200-plus channel audio recorder/editing/ mixer with full I-O and plug-ins; colour-grading for uncompressed HD video; integrated audio/video editing systems; DXD, Super HiDef and emerging 3D audio standards. For larger systems CC-1 uses a wide, fast data highway to interconnect across chips, between computers, or from room to room. CC-1 introduces a disruptive new technology by delivering improved quality, unparalleled flexibility, scalability, enlarged system scope, and a quantum leap in affordability. And the future looks even brighter — by employing FPGAs, Crystal catches the next upsweep in computer hardware development, transparently inheriting the power increases of each successive generation of silicon, continuing the promise of Moore’s Law long after its exhaustion in the microprocessor and DSP technology streams. The Crystal Core technology platform CC-1, gets its name from the crystalline interconnect scheme used at every level of system architecture. Processing blocks connect in three dimensions to other blocks, forming an extensible lattice that scales as required to meet the volume processing demands of media applications. The processing blocks themselves are FPGAs, each programmed with the embedded intelligence required to fulfil the demands of the particular product or system being addressed. The flexibility in programming and operation of these chips is crucial in enabling the creation of a universal hardware solution to a great number of media applications. Fairlight’s invention (patent pending) defines an architectural arrangement of IP cores supporting real-time audio and video processing systems. The first products to be released as a result of this invention will support fully featured audio production systems capable of delivering 230 channel paths; each with eight bands of EQ, three-stage dynamics 60

processing, floating insert point with return, onboard HD video, 12 auxiliary sends and upto 72 user definable mix buses: all from a single CC-1 card.

Large system of multiple hosts connected by Media Highway.

The Crystal architecture is based on the use of high-density FPGAs. Each of these chips contain millions of simple logic parts, that can either statically or dynamically upload a program that configures those parts into complex arrangements for high-speed computer applications. These applications will be used to replace media signal processing hardware that was previously composed of dedicated components or partially-programmable digital hardware. FPGAs can be programmed to form a great variety of standard logic components and I-O ports that can be built into complex sub-systems. Larger programs can even embed whole processors exactly emulating commonly-used DSPs or CPUs into a part of the chip while other parts function differently. It is thus possible, with the addition of a few extra components such as RAM and some standard computer bus interface chips, to build a complete system using one programmable device. A chip can upload partial algorithms while in operation, allowing continuous reconfiguration while operating. This is particularly useful in the context of a user controlled system, where tasks change during a session. It is even possible to take a system down and completely reprogram it to change from, say, a large scale audio mixer into a video colour correction device in a matter of seconds. The Crystal Core architecture has been implemented on a PCI Express card (CC-1), which is connected to a suitable host computer. The smallest configuration capable of delivering 230 channels into 72 mix buses is delivered from a single CC-1 card. Each channel in the system is equipped with eight bands of EQ and 3 stages of dynamics. A medium system providing an increase in the capability (960 channels into 320 buses) is accomplished in a host computer with four CC-1 boards. Larger systems may be assembled by using multiple hosts with each additional CC1 card delivering a linear increase in the channel capacity. There is no theoretical limit to the number of processors that can be linked. resolution

The fundamental operations of a media engine are processing and connectivity. Processing means mathematical and logical operations. Processing is carried out in mathematical sub-units called ‘nodes’ that are formed within the chip by loading processing algorithms into it. Once a program is loaded, that part of the chip is temporarily dedicated to its processing function, which may be a series of operations designed to achieve any required media processing outcome. Connectivity means getting the right signals to the right processing node. For example, in an audio system with 200 source channels, there may be more than 1000 actual channels being processed and routed to different destinations, since each channel must be handled before and after each of its multiple stages of processing. This routing requirement consumes a considerable amount of computer resource, so the processing and connectivity requirements must be balanced against each other when designing a media engine. The Crystal Media Engine challenges other systems using acceleration hardware because it can allocate processing and routing resources flexibly to different processing nodes as needed, allowing all its power to be distributed effectively. Each processing node is individually programmed for its specialised function, and the ‘size’ of the node, or its cost in hardware resources is completely flexible. A simple node performing input and output functions may use a fraction of the resources of a complex node like a multichannel equaliser, and the system will allow precisely the right amount of resources to be allocated to each task. This means that resource usage is optimised, so the maximum possible number of signals can be brought to their appropriate processors and mathematically transformed. By contrast, the architecture of previous systems is locked into standard configurations, where resources are hardwired to signals that may not need them. This causes waste of resources and thereby increases system cost and reduces flexibility. Previous systems achieved expansion through a linear connection bus that allocates a specific number of channels to each discrete processor. The speed of the bus also limited the total system size. In general these systems were hampered by having limited processing power per block, which dictates and constrains the algorithms that can be run, and November/December 2006

technology having a fixed architecture, which commits resources to processes even when they are not needed. The Crystal architecture connects in three dimensions, allowing almost limitless expansion. It can connect multiple circuit boards within a single computer, or multiple computers, fusing them into a single system of immense power. In addition, its reprogrammable allocation of channel connections to processing nodes ensures the flexibility to make full use of the processing power. Unlike the technology it replaces, which is based on interconnected circuit boards using dedicated DSPs, the Crystal Media Engine will be a complete architecture on a single chip, with the option of seamlessly linking other chips to the first when extra capacity is required. Using FPGA technology has allowed Fairlight to further increase the quality of audio manipulation by implementing a new processing paradigm known as Dynamic Resolution Optimisation (DRO). The DRO architecture enables the optimal precision needed for a specific task to be used within each of its Nodes. This design means that ultra-precise 72-bit fixed point can be used in CC-1’s EQ node, while optimal 36-bit floating point can be used in the mixing node. In areas where extreme precision is not required, CC-1 adjusts the precision accordingly. For example, audio metering is more than adequately specified at 16-bit fixed point. DRO is unique, and is patented by Fairlight worldwide. The CC-1 media engines powers the Dream II product family. These products form a range of audio production tools for recording, editing and mixing applications. All initial product outcomes will be delivered from a single CC-1 card that will connect with any of the four available surface technologies (SatelliteAV, StationPlus, Constellation-XT and Anthem). The end user’s physical I-O requirement is supported with the SX-48 or SX-20 modular I-O products.

Each CC-1 engine includes a 1U SX-20 Sync and I-O unit that provides analogue and digital I-Os together with machine control and sync capabilities. In addition, each CC-1 card can support up-to seven multiplexed data connections. These may be used in the form of 64-channel BNC MADI connections or as SDI video data streams. If MADI is employed this equates to 448 physical inputs and outputs per CC-1 card. These MADI ports can be connected to Fairlight’s SX-48 modular remote I-O or any other third party MADI-equipped I-O platform. The initial Dream II products provide the capability to connect up to 4 x SX-48s delivering up to 192 physical inputs and outputs.

Fairlight’s SX 20 is a versatile ‘Sync I-O Toolbox’, and is a required component of any base CC-1 system. The SX-20 includes two mic/instrument preamps plus two additional balanced analogue inputs, 12 balanced analogue outputs, four digital inputs and eight digital outputs. In addition, SX20 includes powerful simultaneous independent multimachine 9-pin control. The SX-20 provides for sync at any frame rate including HD trilevel sync, video sync, Word clock, AES and LTC. The unit also generates LTC at any standard rate. When combined with CC-1, SX-20 provides all the capabilities required for a wide variety of audio production and postproduction tasks.

Anthem 230-channel large format mixing powered by a single CC-1 card.

Fairlight’s SX-48 Signal Exchange extends the CC-1 platform with flexible and cost-effective I-O. Up to four SX-48 units can be connected to a single CC-1 card via MADI providing up to 192 channels of discrete I-O per engine. SX-48 is designed to accommodate all standard sampling frequencies from 44.1kHz to 192kHz. Fairlight’s I-O can be installed in 8-channel modular blocks, allowing combinations of up to six cards of analogue and/or digital I-O to be mixed together in each SX-48 unit or added later if required. SX-48 locks to external sync at any frame rate and accepts HD trilevel sync, video sync, Word clock or AES as references. Fairlight’s Total Studio Connectivity Protocol (TSCP) allows intelligent management of all SX-48 I-O resources on the TSCP network. ■ November/December 2006

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slaying dragons

Sampling Although sampling is most widely known for its importance to audio convertors, it shows up in a lot of other places too. JOHN WATKINSON explores.

john watkinson

â&#x20AC;&#x2DC;Shipbuilding offers a perfect analogy for audio reconstruction. The ships lines are infinitely narrow, just as audio samples represent a vanishingly small instant.â&#x20AC;&#x2122;

N THE LOOSEST sense, sampling is any process in which discrete measurements are taken instead of a continuous process. In audio, we are accustomed to samples being taken with respect to the time axis at a fixed frequency, and we become very nervous, even jittery, if the time between successive samples is not exactly the same. It is important to realise that not only can samples be taken with respect to other axes, such as distance, but there is also no fundamental need for the time or space between them to be constant, although it often is for simplicity. In television, images are sampled into lines and in digital television the lines are sampled into picture elements or pixels. The sampling is spatial and the sampling rate is measured in samples per mm or samples per picture height. The temporal sampling rate is obtained by multiplying the spatial sampling frequency by the scanning velocity. The first description of sampling theory is usually attributed to Shannon although, in the former Soviet Union, the same findings were independently made by Kotelnikov. Interestingly, they were both pre-dated by

Whittaker, who worked in a shipyard. It is well known that the shape of a shipâ&#x20AC;&#x2122;s hull is described by a set of lines. These are a set of cross sections that you would get by putting a ship through a giant bread slicer. Each cross section is a sample and what Whittaker wanted to know was the minimum number of samples that would accurately describe the shape. Clearly an aliased ship would be a source of embarrassment. Actually shipbuilding offers a perfect analogy for audio reconstruction. The ships lines are infinitely narrow, just as audio samples represent a vanishingly small instant. The finished ship is shaped by creating a surface that smoothly joins together all of the lines. This is analogous to the audio reconstruction filter that joins the tops of the samples to obtain the waveform. We never see zero-order-hold ships looking as if they were made out of Lego, but we see plenty of incorrect digital audio explanations showing these quite erroneous staircases. Itâ&#x20AC;&#x2122;s a sad reflection on the audio industry that they know more about sampling in shipyards. The sampling rate needed must be at least twice the bandwidth of the signal to be sampled. In the case of a signal going from zero Hz to some maximum, we would use low-pass filters before sampling and for reconstruction, whereas if the signal is band-limited, we would use bandpass filters. We could sample a signal whose spectrum lay between 10 and 11kHz using a sampling rate of only 2kHz. In audio compression, we frequently split the audio spectrum into sub bands. MPEG uses 32 sub bands. After band splitting, the overall sampling rate has not risen at all. Each band is expressed using 1/32 of the original sampling rate. The bands can be recombined using a set of bandpass filters. If we ignore these filtering requirements, the result will be aliasing. This is a word that doesnâ&#x20AC;&#x2122;t translate; it appears to be limited to English. Aliasing causes the spectrum of the original signal incorrectly to be represented. In digital audio, aliasing is usually a bad thing. In other applications, itâ&#x20AC;&#x2122;s very useful. The mathematics of sampling are essentially the same as those of signal multiplication, which radio engineers call mixing. This should not be confused with audio mixing which is, of course, additive. When two signals are multiplied, the resulting spectrum contains new frequencies representing the sums and

the differences of the original frequencies. It is the difference frequencies, or lower sidebands, that have the most interesting effects. In a system having a fixed sampling rate, if the input frequency goes up, the frequency of the lower sideband goes down. In the sampling oscilloscope (not to be confused with the digital storage oscilloscope) an input signal of some appallingly high frequency is sampled at another appallingly high frequency so that the lower sideband, or difference, frequency is low enough to display on the tube. In analogue tape recorders, the contact between the tape and the heads is imperfect and subject to rapid variations. This has the effect of amplitude modulating the audio waveform. The result is that any frequency in the original audio is accompanied by sidebands due to the modulation. Despite the fact that the unwanted signal added to the sound is audio waveform dependant, the term used is modulation noise. In digital audio we divide the frequency range from zero up to the sampling rate in half. The input spectrum lives in the bottom half, whereas in the top half the same spectrum, having been reversed as if in a mirror, comes down from the sampling rate. Normally these two halves are kept separate by lowpass filters, but in the absence of such filters, if the input frequency rose beyond half the sampling rate, the frequency of the lower sideband would fall below it and we would have aliasing. If the input frequency continued to rise until it reached the sampling rate, the lower sideband frequency would come down to zero. The stroboscope is a positive application of aliasing. The flashes of light are very short and sample the scene. By flashing a light at the same speed as a periodic mechanism, the motion can be arrested. Essentially the motion is being sampled at its own frequency, and the lower sideband is at zero Hz. Upmarket vinyl disc turntables often had stroboscopic markings on the rim. Using a neon light connected to the household AC supply, the speed could be checked. Later turntables had quartz-crystal-locked motors and only retained the stroboscopic markings out of tradition. However, these were useful for measuring the stability of the AC supply. The strobe lights gradually became more powerful and were instead used to illuminate Sharon and Tracy dancing around their handbags. A slight reduction of the flashing rate of a strobe causes the lower sideband to have a negative frequency, so the motion appears to reverse. This reversed motion phenomenon is often seen on films where the images contain spoked wheels. The spoke passing frequency is simply too high for the inadequate frame rates that are used. In spectrum analysis, we often wish to know how

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November/December 2006

slaying dragons much energy exists in a large number of different frequency bands. The approach of making one bandpass filter for each band is too expensive. Instead we can use a single fixed filter, typically a low-pass. We take the unknown signal to be analysed and multiply it by another known frequency and see if the result will pass through our single filter. If the input signal is close to the known frequency, the lower sideband will have a low frequency and can thus pass through out LPF. If the process is repeated with a set of known frequencies, the spectrum can be obtained. A simple extension of this process is where we take a set of audio samples and multiply them by a set of samples representing a sine wave of known frequency, which is called a basis function. If the input frequency is the same as that of the basis function, the samples will

be taken at the same point on each cycle and the sum of all the samples will be non-zero. For all other frequencies this will not happen and the sum of the samples will be zero. The only difficulty is if by chance we sample on the zero-crossings of the input signal and also get zero. To overcome this, we can use two basis functions, one a sine wave and the other a cosine wave. Thus if one basis function falls on the zero crossings, the other will fall on the peaks. If we call the sum of the samples in each case a coefficient, the ratio of the coefficients will reveal the phase of the input signal. This sounds fairly simple, but it is actually how a discrete Fourier transform (DFT) works. The other approach to the problem is to time reverse or mirror the input samples and place them directly before the original samples to obtain a block of

double length. This mirroring will cause any sinusoids in the original block to be inverted in the mirrored samples and cancel out, whereas any cosinusoids will double. This is how the discrete cosine transform (DCT) works. I can hardly raise the subject of sampling without being asked my opinion on the high audio sampling rates that are now supposed to be necessary. As noone has shown that sampling theory is wrong and as no one has shown that the bandwidth of human hearing has suddenly increased, the case doesn’t seem proven. I checked with the shipyard: ‘Do you now use four times as many lines as before to build hulls?’ You can imagine that the answer was in the negative. As I said earlier, shipyards know more about sampling than the audio industry. ■

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V I S I O N

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your business

Here today, here tomorrow Fame is a fleeting thing and especially so if you’re a record producer. DAN DALEY asks if the pursuit of it is a viable career strategy.

dan daley

AST NIGHT, JUST before nodding off, I saw Scott Storch. It would be the closest I would come to him in what had been a two-week pursuit in search of an interview with him for a magazine. I didn’t see him in person, which is what I had been trying to do a few weeks earlier, driving to Hit Factory/Criteria in Miami, where one of his two Bentleys was usually parked out in front. Unfortunately, he chose not to show up that day, a pattern that would get repeated a couple more times despite appointments set up by his publicist in New York. The publicist would be an easy target to unload on as I drove back home, but I resisted the urge, sensing she had been cratered more than once on this issue before. Instead, I would see Storch weeks later, more than once, on any of a series of Hollywood Insidertype programmes on the TV — Storch hanging with his buds like Dr Dre, who mentored the phenomenal producer. Or Christine Aguilera, with whom Storch has had a highly publicised running feud, played out in the slick magazines, she dissing him for not showing up to finish an album reportedly because she would not send a private jet to ferry him back to Los Angeles where she was working. She then cut a track, FUSS, an acronym you can probably figure out on your own, castigating him. He responded by dissing the rest of the album publicly and asserting credit for millions of her sales. He’s not reticent about venting his anger when he feels slighted. When he was passed over for a Grammy nomination in February he complained not only about the snub but also about his competitors that were nominated: Jimmy Jam and Terry

Lewis, the Neptunes, Nigel Godrich, Steve Lillywhite and Danger Mouse. With the exception of Mouse, losing out to producers that have been making hits for two and three decades might be taken as a compliment rather than a slight. And that’s where I come to praise Caesar, not to bury him. While I wasn’t pleased with having to waste my time chasing Storch (which might not be a bad name for a reality TV series), it’s hard to fault him too much in an era when fame is as valuable a currency as it’s ever been. In fact, as a producer — and give him his due: the guy did produce 13 hit singles in 2005 alone, for artists like Aguilera, Chris Brown, Terror Squad and 50 Cent — he’s gone about as far as you can towards becoming as known for being a celebrity as for what he’s being celebrated for. Indeed, there are as many references on the Internet linking him romantically with Paris Hilton as with being the producer of her first record. This is a rarified realm for record producers. Within the industry, we have a pantheon of names that are instantly recognisable to us. Who among us does not know who Ed Cherney, Al Schmitt or Elliot Scheiner are? Their high profile within the industry of music has been a foundation for the continuation of their careers. In a business based on word of mouth, they have been able to leverage their accomplishments into revenue and the ability to keep on doing what they love to do and are most talented at. But you could hardly call that fame. I mean, the technical Grammys are relegated to a pre-primetime broadcast ghetto that shoots by as a lightningfast video crawl while the cameras are focused on Beyoncé’s dress or Courtney Love’s parole officer. There is no solder on the Red Carpet. Nor is there supposed to be. This is a cohort of people who prefer to labour in the background. Fame — actual stalk-your-ass-at-home, intrusive paparazzi, gimme-your-autograph-or-else fame — just gets in the way of making records.

‘Lack of fame has never hindered a legitimate production or engineering career. It’s arguable that fame, or its dark twin, notoriety, are actually stumbling blocks in this end of the music business.’ 64

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November/December 2006

your business Lack of fame has never hindered a legitimate production or engineering career. It’s arguable that fame, or its dark twin, notoriety, are actually stumbling blocks in this end of the music business. Consider the few producers who are household names. Phil Spector might have the highest Q factor, and he’s as well known for waving guns in people’s faces (and, depending upon the outcome of a pending trial in Los Angeles, actually using them) as he is for making records with John Lennon and Darlene Love. Rappers often straddle the line between being producers and artists themselves, but those known for their work producing other artists end up with larger headlines in the tabloids, like Sean Puffy Combs. There are exceptions. Sir George Martin has benefited from his fame with The Beatles. Like Phil Ramone, he’s become a benign presence in the business, perfect for the occasional ribbon-cutting. Then there’s Mutt Lange, whose fame or notoriety, as you prefer, derives as much from his reclusiveness as from his achievements. Then there are the dead ones, like Sun Records founder Shelby Singleton, who produced early Elvis Presley, Johnny Cash and Jerry Lee Lewis incarnations, or pioneering blues producer and field recordist Allan Lomax. They achieve a kind of founding fathers status and are invoked for the purposes of gravitas and when you want to impress musically astute girls. But for the most part the producer flies under popular culture’s Radars, and as mentioned earlier, perhaps that’s all for the better because at least they get things done that way. But is fame a viable career strategy for producers in the 21st century? It depends on what you might want to do later. Production careers are only slightly less volatile than those of artists, and nothing lasts forever. Eddie Kramer has leveraged the time he spent with Jimi Hendrix and Led Zeppelin into a very viable career in photography — his pictures of Plant and Page aren’t just the byproducts of serendipity but are small, vibrant works of art –- and as a talking head for part of a generation we only saw on stage and in Rolling Stone. True, his best work might be 30 years behind him, but then, what he did with a kick and snare is still worth listening to and admiring. Records may have changed

but Isaac Newton hasn’t, and Kramer’s ability to place a microphone is still worth emulating. Then there’s Don Kirschner, that avatar of the Brill Building in the 1960s who produced lots of hits, not least of which were the Monkees, and in doing so revolutionised what a mere record producer might attain in terms of empire building. Once Kirschner figured out that television and rock ’n’ roll were a good fit, he created the concept of the prime-time broadcast rock show with the Don Kirchner’s Rock Concert series in the 1970s. Simon Cowell is the latter-day saint of this type, turning the American Idol TV programme into a highly lucrative factory. Rick Rubin and Dr Dre both managed to parlay their record production careers to the next step, owning record labels, fashion lines. Then there are the names that the aficionados recognise — Nile Rodgers, Jeff Lynne, Brian Wilson, Brian Eno — which is like knowing the difference between Kierkegaard and Nietzsche over an exotic coffee. The test as a producer comes when you leave your base elements behind and see what you can pull off in the larger world. Rick Rubin may have cut his teeth on the Beastie Boys but he was also the best producer Johnny Cash ever had. Paris Hilton is not a useful comparison. When celebrity rivals talent, and when that talent is predicated on a transient phenomenon — and all pop music is a transient phenomenon — the true value of the producer, as a medium and muse for the creativity of others, diminishes. That doesn’t mean you don’t sell millions of records for a while to come, but it does mean you belong to the moment, not to the ages. One last point. I would have written the Storch story for a pro audio trade magazine, not Hello! Cherney, Schmitt, Scheiner, et al may not be household names outside of our industry, but they know how to give back. They’re all over articles in the trade magazines not because they want to promote themselves but because they’re willing to share the techniques and knowledge that got them where they are. Had I been writing for US or People that day, maybe the Bentley would have been in the parking lot. Then, we all would have learned a key piece of knowledge: where do you place the microphone for Paris Hilton? ■

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PASSIONATE ABOUT TUBES

headroom A MATTER OF CHOICE I just finished reading through the October issue of Resolution. I can confirm that John Watkinson’s conclusion in the last but two paragraph of his ‘choice’ article is correct. He’s not alone. As usual with John’s pieces I found this article amusing, educational and moving in equal measure. Keep up the good work. Huw Price, Corntown, UK Huw Price’s kind thoughts made my day. I don’t think I’ve been referred to as moving before and I think I like it. If moving the reader means effective communication I’m all for it. A thread that runs through so much of what we see today is that the goods we can buy are quite devoid of taste and so far away from what technology permits that it is almost criminal. Lack of space in the article prevented me from adding that I don’t drive an SUV and I don’t have satellite navigation in my car. The ‘choice’ I referred to extends to modern cars. We can choose between vulgar and bland. I choose to drive the last car Malcolm Sayer ever designed before his untimely death. Automotive styling died with him and automotive engineering was sacrificed on the altar of commoditisation shortly thereafter. John Watkinson

ABSURD WATTS This letter may seem to be nit-picking, but the standard of articles in Resolution magazine is usually so high that I think it is worth making the point. On page 10 of the September 2006 issue it stated that the re-opened Empire Cinema in London’s Leicester Square now had 56kW of sound output. This is plainly absurd, as it would deafen the entire audience. The amplifier power does not define the sound output, the latter of which is dependent on the electro-mechanico-acoustic conversion efficiencies, the way in which the loudspeakers are mounted, and a number of other factors. I make the point because I noticed in another magazine, the same month, an advertisement claiming ‘250 watts of clear, powerful sound’ from a small, box loudspeaker. This is misleading advertising by ignorant people in a company that should know better. This lackadaisical misuse of

terms needs to be checked, to stop an entire industry de-professionalising itself. The tone of your editorials (Leader articles) suggests that you might be in general agreement. Philip Newell, Spain OK, you’ve got me on this one Phil. It’s sloppy editing and I apologise. But it does happen because a lot of stuff gets processed and the occasional bit of the absurd does get through. I don’t believe that the original press release intended to mislead it just used a little slippery language — you know what they mean even though it’s not quite right. Mind you, what a result if it was true — you could probably smell the cabling in Camden Town as well as hear it. I agree with you on the advertising because apart from anything else it doesn’t inspire confidence in the integrity of the product. Ultimately it’s probably their loss. ZS

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