Resolution V5.4 May/June 2006

AUDIO FOR POST, BROADCAST, RECORDING AND MULTIMEDIA PRODUCTION

V5.4 MAY/JUNE 2006

Alex Silva and Groenemeyer’s FIFA World Cup Anthem Tom Elmhirst on sculpting sound for cool new bands How China is ramping up for multichannel broadcast Studio Hilversum’s diversification for a diversified market Getting the most from restoration Meet your maker: Rupert Neve — Rupert Neve Designs REVIEWS: CharterOak E700 • Lynx Aurora 16 • TC Fabrik C/R • Speck X.Sum Focusrite Saffire • Tascam DM-3200 • Summit FeQ-50 • Eventide Anthology II

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May/June 2006 V5.4

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

News & Analysis 4

Leader

16

Products

4

News

78

Headroom

Studio Hilversum

57

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

New introductions and announcements.

More words on Tens and answers to Tens.

Craft 14

40

44

52

One studio’s plan to create a diversified group of facilities to serve a diversified market, all under one roof.

Tom Elmhirst

Sculpting the sound for some of the UK’s coolest new bands and talking Neves, playing on the songs he mixes and the politics of stems.

Alex Silva

Muso, engineer, producer and writer with a big handle on the German market and Groenemeyer’s FIFA World Cup Anthem.

58

62

66

Sweet Spot

JBL’s case for a smarter and better thought-out monitor that doesn’t have to cost a fortune.

72

Katz’s column

Optimising levels in an analogue processing chain and getting it running sweet and right.

Building your own PC

Tips on the assembly of your prospective DAW — it’s all about the processor, balance and low latency.

Getting the most from

Restoration — how to get the best from the experience.

Meet your maker

Rupert Neve — Rupert Neve Designs.

The man talks shop.

Ten

Practical installation tips.

Business 48

China’s multichannel focus

Hosting the Olympic Games at Beijing 2008 has put a heavy HDTV focus on the country’s infrastructure and its sound engineers.

76

Your business

If the studio business is dead, how come big investments are coming back into the industry from people who really ought to know better?

Technology 68

HDMI

A new digital connector with all sorts of implications for the future — including the forced deactivation of analogue outputs.

74

Slaying Dragons

Watkinson is undecided on whether to give an appraisal of dither.

Reviews 24

Tascam DM-3200

32

TC Electronic Fabrik C and R

26

CharterOak E700

34

Speck Electronics X.Sum

28

Lynx Studio Technology Aurora 16

36

Summit Audio FeQ-50

30

Focusrite Saffire

38

Eventide Anthology II

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, Kevin Hilton, Dan Daley, John Watkinson

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

PRODUCTION AND LAYOUT Dean Cook Dean Cook Productions Tel: +44 1273 236681 Email: dean@resolutionmag.com

news Appointments ANDRZEJ SOSNA has been appointed MD of Tannoy. He joined the company 6 years ago sales and marketing director. The company organisation and management team was restructured when TC Group bought the TGI group of companies in 2002 and since then Andrzej has been product development director.

CALREC HAS appointment Greg Siers as its exclusive representative for sales in the Western US states, Alaska and Hawaii. He is sales and market development manager with Calrec distributor ASG/Bexel.

RF SPECIALIST BBM Electronics — renowned for its Trantec brand of wireless radio mic systems — has appointed Steve Baker Director level by its parent company the TOA corporation. As one of the original founders of BBM, Baker will consolidate its UK R&D operations and spearhead a pan-European marketing drive of Trantec’s S6000 series.

CHARTEROAK ACOUSTIC Devices has appointed REPP Italia as its distributor in Italy. REPP Italia joins Helios Recording & Broadcast in the Netherlands, French distributor Imporaudio, Soundware Sweden, and Mastering Mansion in Spain.

Leader

I had cause to notice that a family relation was flying half way around the world to attend a trade exhibition in the UK concerned mostly with the business of crowd control (turnstiles and fencing, etc) and large scale security issues. The dryness of that industry was eclipsed by the knowledge (imparted by a taxi driver as is traditional) that while we may think NAB and CES (Consumer Electronics Show) are big for Las Vegas, so is another concerned mostly with the matter of concrete and its constructional implications.

So really, as an industry, we are extremely fortunate.

The music, post, TV, film and multimedia sectors that we are involved with are dynamic, high profile, current, fashionable, and exciting and they’re full of fabulous gear for all palettes in fantastic rooms driven by people who are really glad they don’t have to work for a living. If you don’t believe me run it by someone who sells bulk concrete or bent tubing for a living and hear what they think. Yet would they still think that if they attended one of our trade exhibitions?

There is a distinct lack of commercial awareness among our audio trade show stalwarts and unless it is corrected soon it could see them drop off into oblivion because they will no longer be relevant to the end users they would deem to serve. The importance of the academic content of Conventions is overplayed at the expense of the exhibitions that run along side and ultimately pay for the pleasure. Academic content is vital for the development of the science of pro audio and our understanding of it and ultimately it can manifest itself in physical product. However, it cannot be allowed to eclipse the commercial realities of exhibitors who are selling product today to that exciting industry I described further up the column. In fact you could argue that practical issues and technological demonstrations should be concentrated on the exhibition floor leaving the hardcore academic content to be run as a separate event somewhere else even at another time.

I say this because in this increasingly commercially pressured business of ours I believe we should be looking at strengthening and adding value to our audio trade shows not diluting their effectiveness. If we don’t ramp up the commercialism and relevance of our exhibitions then we will drive away newcomers who may have been initially drawn to us by their enthusiasm and expectation of our exciting industry. They will think it’s just another concrete show.

Zenon Schoepe

KLOTZ DIGITAL has appointed Larry Howard as VP sales for South, Central and North America. He has held positions at Universal Studios, Planet Hollywood International, Soundelux Showorks and Tannoy and Community Loudspeakers.

DIGIGRAM HAS given Remi Oudinot responsibility for all marketing communications within Digigram. He previously worked for a software vendor in business process management and for a company providing the technology to produce audio guides for museums.

©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

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HHB’s post division Scrub has supplied three Pro Tools HD3/Icon systems to M2 Television in Soho, London. A 32-fader surround spec D-Control and 8-fader D-Command are already installed in Dubbing Suite A and a prep room respectively, while a 16-fader surround spec D-Control is ready to go into Dubbing Suite B.

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.

S2 Publications Ltd. Registered in England and Wales.

Company number: 4375084.

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

resolution

EMI Studios Group seeks tape owners The EMI Studios Group has not been able to establish ownership of approximately 900 music tapes that it has. More than 10,000 tapes were retained by EMI following the sale of Townhouse Studios (which EMI owned until 2002). These tapes would have accumulated from recording, mixing or mastering sessions prior to 2002. EMI managed to return more than 9,000 of them to their rightful owners but have been unable to identify who owns the remaining 900. Due to space restrictions at the EMI archives, storage of these tapes will not be possible after 1 January 2007 and any tapes remaining will be destroyed.

If you were a client of Townhouse Studios, London before 2002 and believe that you may have left tapes at Townhouse you should contact Cary Anning with proof of ownership on +44 20 7266 7233 or email cary.anning@emimusic.com

China 100% 5.1 first China Movie Channel is launching China’s third HD channel, the China Hometheatre Channel HD Movie Channel with Dolby Digital 5.1. It will be the first in China to broadcast 100% of its programming in surround. Chinese viewers with suitable HD set-top boxes and home theatre systems need only subscribe to local cable broadcasters to receive the new HD channel.

In January, China Central TV announced China’s first HD channel, CCTV HD C h a n n e l , w h i c h b ro a d c a s t s s e l e c t programming in 5.1 from Dolby. Last year Dolby worked with the Shanghai Media & Entertainment Group to launch China’s first standard-definition cable TV channel with 5.1.

• Dolby is providing expertise and support to enable the 2006 FIFA World Cup to be broadcast in Dolby Digital 5.1. Dolby is working with HBS, the host broadcast organisation at the event, to ensure that broadcasters will have access to 5.1 feeds of every game from 12 stadium venues around Germany. Dolby 5.1 audio will be available for HD coverage of the matches.

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May/June 2006

news FCS welcomes Ofcom’s UHF flexibility In response to FCS (Federation of Communications Services) and the business radio industry requests, Ofcom has agreed to allow four blocks of 2 x 500kHz to be offered in the forthcoming auction of spectrum between 412 and 424MHz (exDolphin spectrum).

FCS welcomes these important changes, as it felt that the original plan to offer just one single block of 2 x 2MHz would be restrictive for many of the smaller users.

‘FCS congratulates Ofcom on the material and positive changes to the allocation of spectrum in this band, which will now be more suitable for business radio use,’ said Jacqui Brookes, FCS CEO. ‘We welcome the fact that Ofcom has obviously taken note of input from industry following the consultation held in autumn 2005.’

In its own submission to Ofcom, FCS had explained that to take account of the diverse needs of the business radio sector and to stimulate innovation, it was necessary to offer smaller spectrum packages.

Pro-Bel IPO plans Pro-Bel has announced that advisors have been appointed to plan an IPO in Q4 of 2006.

‘Pro-Bel has demonstrated very high levels of organic growth over the last three years, virtually doubling our revenue,’ said Graham Pitman, chief executive of Pro-Bel. ‘Last year’s acquisition of Vistek has already been a great success and our technological choices are meeting with firm market approval. We have one of the best operating profits in the segment and this year we anticipate revenue to grow to around US$60M. We have always been an innovative company and our customers are excited about our vision of the future. We believe an IPO is the most effective means of enabling us to pursue that vision.’

Rahman opens Chennai facility

Appointments AURALEX ACOUSTICS has appointed New Musik to handle distribution in Scandinavia and Stollas SA to cover Greece and Cyprus.

RIEDEL COMMUNICATIONS and CP COMMUNICATIONS — a leading rental company for broadcast, film and event productions — have announced a partnership, which includes a financial interest of Riedel in CP. CP Communications will have the largest rental inventory of Riedel intercoms in North America and the partnership guarantees unilateral support for all customers, including other rental companies.

One of India’s leading film composers and musical film directors, AR Rahman, has opened his own 3,000 square-foot recording studio renamed AM Studios in Chennai.

The facility, formerly known as Panchathan Record Inn, took three years to complete and houses a recording suite as well as a film mixing stage.

Acoustic design and architectural plans for the studios were conceived by Studio 440 Architecture & Acoustics, Hollywood. The two main spaces are a 380-square foot control room and the 1,300 square-foot recording studio/mixing stage. The recording studio offers flexibility for music recording and film mixing, thanks to variable acoustics. The recording area, which doubles as a mix stage, can accommodate 30 musicians for large recordings. Rahman opted for a 140-channel Euphonix System 5-M recording to Euphonix 48-track R-1. The bass system in the recording studio/mixing stage has four Bag End Infrasub-18 PRO subwoofers.

BEYERDYNAMIC HAS been appointed UK distributor for Australian company Violet Audio, manufacturer of the ADP61 5.1/6.1 preamp.

GENELEC IN the US has made three key appointments. Paul Stewart is now product specialist. A graduate of the Berklee College of Music he has more than 10 years experience in pro audio sales. Dan Pye has joined as sales support. He is a graduate of the University of Massachusetts Lowell. Amy Tran has been named sales administrator, making a smooth transition from her prior role at an authorised Genelec reseller.

Panamax acquires Furman Sound Panamax has acquired Furman Sound in a move that will expand the family of companies in the Home Technology Group of Linear LLC. Known for its AC power conditioning and distribution products, Furman also makes audio and video signal processors as a line the company was founded on in 1974. Panamax makes a range of power conditioning products for home theatre electronics, cable and satellite TV systems, household appliances, and office equipment. Linear is a pioneer in engineered radio frequency products.

Leading New York City audio post facility The Audio Department recently added a video editing suite equipped with an Avid Adrenaline and Genelec 8030A monitors.

‘Better sound often results in improved video editing,’ commented David Prentice of supplier VCA Fusion. ‘For instance, better monitoring means you’ll be able to hear differences in ambient sound that could make a tremendous difference to the quality of an edit and in the overall sound.’

SONIC 8 has taken on UK distribution of i3 software, coinciding with the release of the Italian company’s DSP-Quattro 2 software. DSP-Quattro 2 is the updated version of the Mac-based audio editing, plug-in hosting and CD mastering program.

AUDIX CORPORATION has appointed Power Group as its distributor in Canada.

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May/June 2006

resolution

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news Appointments

DPA minis boost Belgian stock

Glasgow City Halls completes refurb

APT HAS appointed Guy Gampell to the position of Asian sales manager. He joins from Sadie having previously worked for AMS Neve and Akai.

APT has appointed Indigon as its distributor in South Korea.

MIDAS AND KLARK TEKNIK have appointed Moscow-based Theater Technics and Technology (TTT) as its Russian distributor. Formed 15 years ago, TTT also represents Midas/KT fellow brands Electro-Voice, Telex and Dynacord.

LINEAR ACOUSTIC has named J+C Intersonic as exclusive dealer for its digital television processors throughout Switzerland, Germany and Austria.

CERWIN-VEGA and KRK have appointed Rick Richardson as marketing manager. He has marketing experience in consumer electronics at Niles Audio and more recently as director of marketing of ScheduALL software.

Mike Newman has j o i n e d C e r w i n - Ve g a and KRK as marketing p r o d u c t m a n a g e r.

He has previously w o r k e d f o r S t u d e r, Digidesign, TimeLine Vista, Gibson Labs and Mackie/EAW. PRO-BEL CTO Neil Maycock has been appointed as president Pro-Bel Americas.

Since the UK management team’s MBO in 2003, Pro-Bel has restructured its US operations. Maycock joined Pro-Bel in 1989 and has stepped into the newly created role following successes as CTO and prior to that as VP of engineering.

Pro-Bel has announced the creation of a Solutions Group and the appointment of Karl Mehring as its department head.

Belgian rental company RF Transmission, owned by former sound engineer Bernard Scyeur, has one of the largest inventories of DPA mics in the country, purchasing some 400 miniature mics since 1998 from Belgian distributor Amptec.

RF Transmission has recently acquired an additional 40 each of DPA 4061 omni miniatures and 4066 omni headbands for its rental stock, bringing its current total to 120 4061s, 125 4066s plus 15 4088 cardioid headbands.

‘I use DPA because they simply make the best and most reliable microphones for the job,’ said Scyeur. ‘Having DPA mics on the equipment list is always reassuring for the customer, in terms of quality, reliability and performance.’

Scyeur has developed a software package to manage stock levels, customer quotes and invoicing, and makes a point of testing all equipment three times before it goes live on a production. Each mic has its own barcode, giving its rental history and dedicated frequency response, measured before and after each job. * Producer Kevin Wallace and director Matthew Warchus recently premiered their 37 million-dollar version of Tolkien’s The Lord Of The Rings to an audience at the Princess of Wales Theatre, Toronto, Canada.

The production is a play with music and is scored by Indian AR Rahman and Finnish contemporary folk group Värttinä and played live by 19 musicians. Sound designer Simon Baker and associate Sten Severson used DPA IMK4061 miniature instrument miking kits and 4021 compact cardioids on strings, 4011 cardioids for horns, 4041 large diaphragms on percussion and more strings, while 4015 wide cardioids looked after more unusual ethnic instruments.

Fox Rox with Millennia

1,000th Icon sold

LOUD TECHNOLOGIES has appointed Moray McMillin as UK Pro Audio sales manager for EAW. GML HAS appointed Studiocare as a UK dealer for its line of products, which are also available to hire. Studiocare’s hire department has taken stock of a further eight Neve 1081 modules, four 1073 modules and four 1084 modules.

MARC PHILLIPS has joined the ENCO UK operation as head of sales and marketing to manage the growing demand for its products. He has held key positions at two radio stations and has worked across media, new media, design, web and IT.

6

A Studer Vista 8 has been chosen for the new recording studio at the refurbished G l a s g o w City Halls.

Perfor mances given by the venue’s principal re s i d e n t , t h e BBC Scottish Symphony Orchestra, in the Grand Hall will be recorded in the new studio for broadcast on BBC Radio 3 and Radio Scotland.

The four-year project to revamp the City Halls into a world-class concert venue was completed at the start of 2006. A priority was the construction of a recording and production environment primarily to service the new complex including the Grand Hall, Fruitmarket and the Recital Room.

Designed by Arup Acoustics to AES-EBU 5.1 spec, the control room’s ‘bare aesthetic’ includes B&W monitoring, a SADiE and Stax electrostatic headphones.

‘Three or four engineers are using the Vista 8,’ explained BBC SSO recording engineer Graeme Taylor, ‘and we all have our own settings in memory, which makes it very easy to come in and reset the desk.

I’ve been impressed by how easy it is to build the desk the way you want it, and, at any time, you’re only one button-press away from the operation you want.’

FOX Channel 6 in San Diego, US has installed Millennia HD-3D mic preamps and TD-1 channel strips for its Emmy award-winning live music show Fox Rox. Fox Rox is a late-night show dedicated to the local and underground music scene that features international musicians and bands, CD reviews, concerts, behind the scene segments, and interviews.

‘When we crack open the mics on the mixer now, our jaws just drop.’ commented

engineer Mike Klowas. ‘Vocals through our Audix OM-7s are so detailed and have such sparkle, that the mics sound like large diaphragm studio mics now.

‘We can actually lower volume levels and still get a better sound now, which is critical when recording to digital videotape,’ he said.

Millennia says that more than 25,000 channels of HV-3 preamps are now installed worldwide.

resolution

Digidesign Icon|D-Control and Icon|DCommand digital console sales have passed the 1,000 mark worldwide since they began shipping in 2004. The 1,000th unit — a 48-fader dual-operator D-Control — was sold to Walt Disney Pictures and Television for installation on the dub stage used to mix the television show Lost.

‘Choosing an Icon was the right decision for us. In fact, we chose three,’ said Gil Gagnon, vice president of postproduction services for Walt Disney Pictures and TV. ‘We’re currently upgrading three of our stages with Icon consoles as the right tool for the demands of those particular s tages. We have had tremendous success with our ProControl stage over the last five years, and this will only further complement the variety of solutions we can offer our clients.’

Disney has a dual-operator, 48-fader D-Control, a dual-operator, 48-fader DCommand, and a 24-fader, single-operator D-Command.

May/June 2006

news Appointments

Hospital moves with S2 Solution

SWITCHCRAFT HAS appointing Madrid based Magnetrón to stock and supply its connectors, audio and video patchbays and cable assemblies to the broadcast and professional audio markets in Spain.

LECTROSONICS HAS opened a sales and service office in Toronto to support the Canadian market. Colin Bernard has been appointed director of Canadian operations, with Joe Burtinsky in charge of the new office’s warranty repair facility.

AUDIO EXCHANGE International’s (AXI) Reso Audiotronics product line is now available in the Japanese market through Heavy Moon.

(l-r) Behrens, Andersson, Mittelmann, Helbo, Christensen, Chapman.

SWEDEN’S LAB.GRUPPEN has expanded its sales and marketing team.

Claus Behrens joins as European sales manager and Tim Chapman has been appointed head of marketing.

Behrens’ experience was gained while at TC Electronic and TC Works.

Chapman’s experience includes stints at Turbosound, Crest Audio and Meyer Sound. They join Martin Andersson (coordination and domestic sales, Sweden), Thomas Mittelmann (international sales manager), Niels Helbo (product manager), and Michael Christensen (VP of sales).

STANTON MAGNETICS has named Sarah Lombard as marketing m a n a g e r. S h e j o i n s from Audiobahn where she was director of marketing and public relations.

COMMUNITY HAS promoted Scott Deegan to the position of inside sales and operations manager after three years with the company.

SHOWTIME

Broadcast Asia, Singapore................20-23 June IBC, Amsterdam .......................8-12 September PLASA, London ......................10-13 September AES, San Francisco ......................... 6-9 October SATIS, Paris ..................................... 7-9 October SBES, Birmingham .................. 15-16 November Interbee, Tokyo....................... 15-17 November

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Hospital Radio Reading, known as Blast 1386, has long occupied the site at Battle Hospital in West Reading and when the hospital authorities told them that they would be moving to the site of the Royal Berkshire Hospital they were delighted. However, HRR engineers Charles Coultas and Jeff Pritchard decided that the existing equipment could not be removed from the studio complex and relocated; it was too old and some pieces of equipment were built into the fabric of the building.

They opted for a Sonifex S2 Solutions package together with Myriad automation system from PSquared, which consists of the S2 digital I-O analogue mixer mounted in modular technical furniture and surrounded by broadcasting equipment integrated to provide an easy to operate studio. The cabling is preprepared so installation consists of simply connecting the various pieces of equipment together.

The new studio was set up temporarily at the Battle site and the actual studio move to the new premises took four days. The service is distributed to the patients via the Patientline system. Stereo audio is delivered via Cat5 cable and the return signal is monitored locally.

Royal Marines sign up SADiE

Chevron Recordings, based at the HMS Nelson Naval base in Portsmouth, has purchased two SADiE systems — a SADiE LRX for portable location recording and a SADiE PCM8 for postproduction and mastering.

Chevron Recordings is the recording department of the Royal Marines Band Service, specialising in recording the Royal Marines Band Service (RMBS) and other service and civilian symphonic wind bands.

Chevron Recordings produces military band music for CD along with music for television, festivals and tattoos. Its largest in-house facility is the purpose-built studio at HMS Raleigh in Torpoint, home of HM Royal Marines Band Plymouth, which can

accommodate 65 musicians.

‘The studio is funded entirely by producing and selling Chevron titles, so the two most important aspects of all our equipment are high sound quality and affordability,’ explained Cpl Andy Deacon. ‘We consulted many other recording engineers about their experiences with hard disk systems, and were consistently told that SADiE would be the best choice, not only for sound quality but also editing and mastering flexibility.

‘The experience of working in the Royal Albert Hall with Phil Wright of Sound by Design and their Digico system, has taught us that with the introduction of affordable MADI in the H64 and soon the LRX, makes Sadie an even more attractive acquisition,’ he added.

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French town halls add recording Acoustic designers AiA are currently involved in a number of projects throughout France that involve the renovation of existing performance spaces and most of the projects include the addition of a control room for recording and mixing multichannel, along with a meeting room, and storage space.

‘Over the last two years in France there has been much talk and many inquiries into the cost and consequences of adding a music recording activity to an existing performance hall,’ said AiA’s Sooch San Souci. ‘Towns looking to sponsor activities for the energetic 16-26 year-old crowd have seen the virtue in the expansion of performance hall activities.

Music is recorded or mixed day and night, while live recordings have now become regular fare. Schools are now recording and distributing the results of the class musical performances and this has given a new meaning for younger generations as performers. Schools have also organised visits for students to observe the work of recording music.’

Many small concert venues are becoming attractive partners for those eager to setup a music recording activity and a social community niche, according to San Souci.

Post house XT trio

Michigan-based post houses Postique and its sister facility, hdstudios, have selected Dream Constellation-XT consoles. Fairlight has provided three Constellation-XTs. At Postique, one of these is an enhancement of the existing Dream Station while another is a new addition. The third ConstellationXT is slated to enhance an existing Dream Station at hdstudios.

‘We’ve been very satisfied with Fairlight,’ said Steve Wild, president of owners Grace & Wild. ‘We are convinced it’s the highest quality product and best for our needs. It’s also excellent for our customers and ideally suits our operating structure.’

Also in the US, NASCAR Images, exclusive rights holder of NASCAR footage, recently installed a Dream Constellation-XT. The company also offers video and commercial production and postproduction, marketing and sales videos, custom DVDs, and turnkey television production. The facility covers more than 35,000 square feet with a 1200-squarefoot studio space featuring 17 nonlinear suites, 22Tb of storage and a Digital Asset management system supporting the last 20 years of NASCAR history.

• Fairlight Japan has been awarded the 2006 Australian Importer Special Recognition award for ‘introducing and expanding the use of Australian made audio equipment in the Japanese market’.

May/June 2006

VCube SE HD-SD VIDEO SYSTEM

Simple - Reliable - Versatile The new cost-effective HD or SD Video solution for non-linear playback and recording in post-production

www.merging.com/vcube

AES Booth 2005

news

Wildtracks wild about Isis

The Big Picture

Scottish TV upgrades to Zeta

BIZ BITES — EMI is considering a second bid for Warners, writes Nigel Jopson. An initial bid of US$4.23 billion from EMI was rejected, though a higher offer is rumoured soon. Warner chairman Edgar Bronfman Jr declared: ‘Consolidation for consolidation’s sake doesn’t make a lot of sense,’ and with an opinion many in the music industry would argue with he continued: ‘Ours is not a business that requires scale economics.’ Meanwhile, a better-thanexpected quarterly earnings report will likely drive the value of WMG, and an EMI bid, upward. Both companies have courted and rejected each other several times during the past 5 years.

BBC Post Production has transferred HD media between BBC sites for the first time without the need for dedicated circuits. Using a combination of Avid DNxHD compression, the existing SD infrastructure inside each building and new fibre between Bristol Broadcasting House and London Television Centre, HD content was moved between BBC Post Production’s Bristol machine room and London eQ Suite. The experiments carry implications for future tapeless workflows and may form the foundations of a practical method of moving HD between sites at short notice.

Rock music became the UK’s favourite genre as its share of the albums market reached 36.2%, according to new figures from the BPI. British artists claimed 49.4% of last year’s UK sales totals, the highest share since 1998. Although Pop remains the dominant genre in the singles market, digital sales helped Rock music increase its share by 8.4% compared to 2004. Gnarls Barclay became the first to get a Number 1 in the UK charts Continued

Total Audio Solutions has supplied a Pyramix and HD VCube system to Soho audio post facility Wildtracks. The installed system includes Merging’s Isis tactile remote control.

‘We needed to replace the DAR system that we have used for 15 years and we found that the Merging system gave us the speed and tapeless production we wanted,’ explained MD Graham Pickford. ‘Pyramix and VCube can handle any format including providing the BWAV files that there is increasing demand for.’

Wildtracks has seven digital studios featuring surround mixing and a Dolby license for cinema commercials.

Bristol University’s 1-year MA course in Film and Television Production has bought two TV post specified Pyramix systems from TAS. ‘We were looking for systems that would do 3 or 4 major operations seamlessly and with minimal troubleshooting,’ said the University’s Peter Milner. ‘These are syncing of film rushes, autoconforming using an audio EDL supplied by the picture editor, track-laying and mixing. As regards the latter, though, we usually take the project to an outside dubbing mixer for the final mix, but this adds an extra requirement: the ability to export smoothly to other systems.

The fact that there are already a number of Pyramixs in Bristol postproduction houses influenced our decision.’

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SMG has ordered two 32-fader Calrec Zeta consoles to equip both studios in Scottish TV’s new Pacific Quay complex in Glasgow.

Studio 1 is for regional news and sport, broadcasting two hours of live TV every day while Studio 2 is for SMG’s Setanta contract, consisting of Rangers and Celtic TV and live football every Sunday.

The sale follows the installation of a 48-fader Zeta in SMG’s OB Unit 1 last year. The truck, owned and operated

by SMG’s facility arm Broadcast & Event solutions, represented the broadcaster’s first excursion into digital mixing. It is used exclusively for Setanta live football.

‘The existing Calrec Q in the truck was in need of replacement, and the operators like using Calrec consoles,’ said SMG technical manager Gary Welsh. ‘When it came down to the new sound desks it was a simple decision. Once we saw the Zeta we knew it was the one to buy.’

‘At Pacific Quay the Zeta’s extensive GPI interface is ideally suited to the demands of automated news coverage and having continuity of control surface across the broadcasting divisions means that operators can be deployed throughout the business with confidence,’ added senior sound supervisor Robert Stephen (pictured).

• NEP Broadcasting, based in Pittsburgh, PA, has bought two 72-fader Alpha consoles for installation in its SS25 and ND3HD mobile trucks. The new units will provide broadcasting services primarily for NEP’s network clients showcasing golf, auto-racing, and NFL games.

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news The Big Picture

Six MCs for Henniger

on downloads alone with the single Crazy.

The project is a collaboration between rapper Cee-Lo and producer Danger Mouse — who shot to fame on the back of an illegal mashup posted online (a crazy mix of Jay-Z’s Black album and the Beatles’ White album named ... the Grey album).

Finally fulfilling t h e f re q u e n t l y made heady predictions about unknown independent musicians and the Internet, a struggling UK singer managed to get herself signed to major label RCA/ Sony-BMG after performing for free using a webcam to broadcast live from her basement. From an initial audience of 60

people, more than 60,000 viewers were eventually estimated to have watched Sandi Thom’s ‘Twenty One Nights From Tooting Tour.’ Thom offered a poignant insight into the current obsession with an era remembered by many Resolution readers from their first experiences as production professionals, when, in her song I Wish I Was A Punk Rocker With Flowers In My Hair, she yearns for 1977: ‘...When the head of state didn’t play guitar, Not everybody drove a car, When music really mattered and when radio was king, When accountants didn’t have control And the media couldn’t buy your soul And computers were still scary and we didn’t know everything.’

Arlington, VA-based Henninger Media Services is to take delivery of six Euphonix DAW controllers as part of an extensive facility upgrade of its audio services division. This includes three 16-fader System 5-MC controllers, two 8-fader System 5-MC controllers, plus a standalone MC Media Application Controller all integrated with Nuendo via EuCon. ‘We handle a wide range of projects — everything from industrial sales and training pieces through commercials and documentaries — so we wanted a system that was not only capable of meeting today’s audio production challenges, but one that would be able to grow and adapt to an ever-changing production environment,’ explained David Hurley, senior mixer/manager. ‘With its open-ended architecture and the ability to be configured in so many different ways, we believe the System 5-MC represents the best return on investment over the long term.

‘These controllers are solidly built; the buttons feel nice in your hand, and the pots turn and respond the way you expect them to,’ he added. ‘These factors were all important to us.

‘The System 5-MC is very easy to learn and get started with, and yet it’s very comprehensive,’ said Hurley. ‘It comes with a great initial setup, and the ease of assigning different functions to various knobs and buttons makes it possible to tailor the system exactly as needed.’

Rio’s room runs VS2108s

WhiteChalk Music was established in 2004 by Manchester United and England footballer Rio Ferdinand with business partner Courtney Richardson, and is specialising in bringing emerging young talent to a wider audience. Its studio was fitted by Andy Parry of Electroacoustic System Specialists but the owners chose monitors that Andy felt they wouldn’t ultimately be happy with.

‘If I’m designing a studio and the client chooses a monitor I don’t think he’ll be happy with, I’ll usually get him to try the nearest equivalent Quested model.

More often than not, the speakers sell themselves,’ said Andy. WhiteChalk has now installed Quested VS2108s.

‘You need monitors that will tell you the truth. You don’t want to hear any of the speaker’s own character,’ he added.

‘If you’re putting a lacklustre mix in, you need to hear a lacklustre mix coming out, not have a speaker that flatters the sound, or adds components to the mix that aren’t going to tape.’

Nick’s ISA moment Sound engineer Nick Blundell has set up his own facility with Pro Tools and two Focusrite ISA430s and a Compounder. ‘The ISA430s are used mainly for recording as a frontend for Pro Tools,’ he said, ‘but they have also proved themselves useful for mixing and mastering. I wanted a versatile frontend that was warm with a nice transient response for all types of recording.’

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NOTHING ADDED. NOTHING TAKEN AWAY.

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Totally uncoloured and gifted with a breathtaking dynamic range, the remarkable HV-3 microphone preamplifier circuit fronts an extraordinary range of outboard processors from Millennia.

Developed with a singular passion by the visionary US recording engineer John La Grou, devices including the TD-1 Half Rack

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An ultra-transparent, two channel Twin Topology® optocompressor/limiter offering switchable 100% Class-A biased solid-state operation or 100% Class-A biased all triode vacuum tube operation.

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A complete Twin Topology® single channel music recording system offering a selection of vacuum tube or solid-state Class-A circuits at virtually every function, including mic pre, compressor/limiter and parametric EQ.

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Recording System (pictured above) empower the user with unprecedented signal processing control, with features including Twin Topology® selectable vacuum tube or fully discrete solid-state input amplifiers, variable input impedances, fully parametric EQ, dual REAMP® outputs and Speaker Soak® technology.

For the ultimate in high-end audio processing, talk to HHB about Millennia.

Studio Hilversum Take a studio with a history, start again and expand it into a complex that offers a complete suite of facilities with multichannel all under one roof. ZENON SCHOEPE is impressed with the results.

T

UCKED AWAY BEHIND a rather unassuming residential street in Hilversum lies one of the best kept secrets in the recording community of the Netherlands. It’s not some producer’s private room but a complex of studios that has only recently been completed and revealed itself as the full-service place that it is.

It’s the brainchild of owner producer/sound technician Maarten Derksen who will tell you that things have changed dramatically in the Dutch market: studios have closed, a generation of studio owners and engineers have retired and there’s a new order coming in that better represents the commercial reality. He’ll also tell you that the fragmentations in the client base — once based predominantly on a hardcore of big artists and now peppered by many more smaller acts — means that there should be more potential clients. And there are, but they’re not all doing the same thing anymore — recording albums — they now want their own DVDs or CDs for concert sale and promotion, having broadcast concert footage repackaged, uploading to the Internet. All the things that they can’t do at home in their own rooms.

In his own words, Maarten saw a ‘diversified AV industry’ and figured it needed a diversified AV facility to serve it.

He took over Studio Jules, as it had been known, in 2001, renamed it as Studio Hilversum and turned it 14

around. Studio Jules had been a leading post facility in the 1980s and 1990s and the studio site has historical connections with Radio Veronica, which played such a vital part in the evolution of broadcasting in Holland.

However, the turn around was done in stages with post from Maarten and CD mastering from mastering engineer Hay Zeelen, and adding rooms as time and budget allowed. Studio Hilversum now has six rooms all with PMC monitoring.

‘What I had bought,’ says Maarten, ‘was a building designed in 1977! There were two studios, one with a Screensound in it for post and a mastering studio; and on the basis of the mastering activity we were

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able to continue.’ While they knew everything had to be changed at the studio they didn’t know where to start. Hay had tried to encourage the former owner to develop into surround and DVD mastering and they knew they wanted to combine picture. But as Maarten points out, at that time there was no infrastructure in place to allow them to build the sorts of rooms they wanted.

‘We didn’t have any examples of how to build the company that we’ve just described or drawings of surround studios,’ he says. ‘When you read the manuals for Dolby and DTS you get two stories and when you read interviews with surround engineers you again get different stories. We had to find our own way here.

‘Eventually we started with the toilets and worked our way out and in the first year we made the basis for the place with things like the electrics. Hay continued with his mastering, I did some postproduction and it just grew.’

They designed their own acoustics and integrated their own personal favourites — Hay’s bank of Weiss bw102s, for example — and because they kept working during the building it meant there were situations where they were earning in one area and designing the build in another and when that was complete they’d move over to the new room and start the build of the old one. The voiceover booth was May/June 2006

facility

Maarten’s studio for a year.

Building was finally finished at the end of last year as the culmination of three years’ of step-bystep improvements — every studio has daylight and fresh air and air con. And you know that the clients wouldn’t have suffered during the building, such is the professionalism at the place.

What they’ve ended up with is Hay’s mastering studio, a video studio with Avid Adrenalin, a stunning music studio, a DVD studio and an audio post studio with voiceover booth. The beauty of the arrangement is that the rooms can work independently or can be employed as part of a bigger scheme when the project requires it. It means they can take on a concert recording by employing a 6-camera mobile video unit with 72-track hard disk recording and follow a DVD project right through the mixing, mastering authoring and on to replication. They can offer complete packages. ‘We’re geared up here to make the most of the hype associated with a TV concert and all the publicity it gets,’ explains Maarten. ‘We can get the completed video done in 5 days — the video editing, mixing, overdubs and out to market — and we can do that in that sort of time frame if we are involved from the beginning.

‘A big market is the local singer market — there are more local amateur singers than there are professional singers in Holland. They have gigs lined up for every weekend around the country and they can make as much money as a professional singer. They want to have DVDs of them in a boat singing their songs that

May/June 2006

they can sell to their own audience at gigs. A lot of people forget that industry but they are very serious people,’ he says.

The facility uses Nuendo throughout with additional Sadies for mastering — Hay employs multiple screens to drive his equipment — it’s networked, there’s a DigiBeta connectable to every room, a webserver, email server and 40,000 sound effects all housed in machine cupboards.

It’s a fully PMC 5.1 facility and the mastering room sports MB2S XBDs used in full-range mode for stereo but for surround the bass cabinets' filters are changed to act as the 0.1 by physically unplugging them.

‘The association with PMC started with me in 1994 when I was mastering at Studio Jules with B&W speakers, which I had been using for a long time and they were starting to sound dated,’ explains Hay. ‘I tested a lot of speakers but I arrived at PMC and Bryston — I’d been using Bryston with the B&Ws. I started with the smaller PMCs and I was really astonished by the sound and the stereo placement.

When Maarten found out how much it would cost to do a studio in surround with PMC we decided to do it passive and make a hard cut to change the filters.’

Smaller models are dotted throughout in 5.1 configurations and Maarten says that being a PMC facility means that a project that goes all the way through starts on small PMCs, ends up on big PMCs for the mastering, and is then finished on small PMCs in the authoring room.

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The layout is cleverly integrated and interconnected — it looks great and clearly works well. The music recording studio has a custom-built ADT mixer, which was on order at the time of my visit, that will be fully surround capable. So there’s no control room ‘statement’ desk in the traditional sense because they’ll tell you that the requirement is predominantly for monitoring control with the way they work in Nuendo and with the outboard racks. It’s radical stuff and while a lot of people talk about it very few actually do it commercially — Studio Hilversum does, and in multichannel too with a great live area. But then it’s indicative of the ‘can-do’ attitude of the place staffed with experienced operators running state-ofthe-art and sensibly integrated gear in great rooms.

It is still early days for the completed complex but all the indicators say that this is going to be a successful operation because it has grown organically with its clients in tow and each phase of expansion has added value and features, with a commensurate increase in efficiency for the facility as this has happened. You get the distinct impression that Studio Hilversum takes on the client's problems and makes them their own and then provides them with a solution — all under one roof. ■

Contact STUDIO HILVERSUM, NETHERLANDS Website: www.studiohilversum.nl Tel: +31 35 6211920

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review gear

Products

Platform news: Digidesign

Equipment introductions and announcements.

Studer Vista 5

The compact-format Vista 5 digital console brings Studer’s Vistonics to a new price bracket and is the first Vista console to be produced with PCB assemblies from the Soundcraft Studer Group’s manufacturing facility. The console’s small size and simple connectivity make it portable and it employs a new DSP core.

The 32-fader desk has 20 channel strips optimised for input channel operation and 12 additional strips for operating output and input channels. Vistonics put 52 outputs under immediate control and a total of 240 channels can be accessed. The total I-O capacity through cards for mic/line, ADAT, TDIF, AES-EBU, SDI and MADI can exceed 1700.

Broadcast facilities include N-1 outputs, off-air conferencing, GPIO and extensive monitoring including 5.1-to-stereo downmix functions. Virtual Surround Panning is also included. The console’s internal matrix may be controlled from third-party controllers and video routers.

Hardware redundancy options include power supplies, DSP cards with instant switch over as well as redundant link to DSP core and MADI links to D21m remote I-O boxes.

Vista 5 is Harman HiQnet-compatible and has static automation (snapshot filtering and editing) and cue list functionality. www.studer.ch

SADiE V5.5 and PCM-H16

SADiE software version V5.5 for the Series 5 range of DAWs includes improvements to the user interface that introduces a new colouring scheme to ease identification of individual streams and groups in the playlist and relate them to their mixer strips. Enhanced support for the LRX Location Audio Workstation and PCM-H64 multitrack platforms now

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At the AES Convention in Paris, Digidesign is presenting its latest product releases and previews via its Technology Suite and providing information and education on realworld situations in its dedicated Seminar Area.

In the Technology Suite (Stand: 1209 section 5.2 — by appointment only) Digidesign is showcasing forthcoming Pro Tools HD 7.2 software for Icon and Pro Tools|HD systems that includes improvements for integrated video, new automation modes, enhanced support for multichannel field audio, and destructive punch recording capabilities.

It will also be demonstrating workflows between Avid and Pro Tools systems, including the DV Toolkit 2 software expansion option that transforms Pro Tools LE into a powerful audio post system and Virtual Katy 2, which it describes as ‘the ultimate autoconform and change management tool’.

The Digidesign Seminar Room will offer English and French language presentations on Mixing In The Box, Pro Tools HD 7.2, Pro Tools in music production, film mixing with Icon, and Avid workflow for 5.1 audio.

Digidesign’s Pro Tools HD 7.2 software upgrade is described as one of the most significant software upgrades for Icon and Pro Tools|HD systems to date. 7.2 software is expected in the second half of 2006, and will offer efficient new mixing and automation features and new SignalTools for multichannel metering and analysis. www.digidesign.com

includes a dedicated MTR multitrack recorder interface to enable simpler setups and operation of systems in the style of traditional multitrack recorders.

A metadata interface has been introduced for inserting metadata and handling of file and take names during the record process. Support for recording to a mirrored disk, in addition to the primary recording drive, provides simultaneous back-up of files as standard in V5.5.

Enhanced support for consoles, such as the SSL AWS-900, as a hardware controller is included together with Variable Nudge for moving clips by an exact amount.

The PCM-H16 is a high-power DSP-based workstation.

Specification includes 16 channels of AES and A-D and D-A I-Os, 64 streams of 24-bit 48kHz audio

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with real-time crossfades and punch-ins, plus similar DSP processing power to the flagship PCM-H64 multitrack system.

For TV post studios the stem controls enable different versions to be track-layed and mixed within one EDL with DSP capacity to spare. In its postproduction guise, the PCM16 ships with integrated AAF, OMF, Pro Tools and AES-31 interchange facilities, and tape and diskbased autoconform packages, including audio and video reconform from a change list. It also includes integrated non-linear video record and replay, which allows the video stream to be cut and edited in the same EDL alongside the audio. www.sadie.com

May/June 2006

review gear API A2D

API A2D is a stereo mic preamp with A-D convertor featuring a pair of signature 312 mic preamps, API 2520 op amps, 20-segment LED metering on analogue and digital sections, mic and line inputs, polarity, Phantom power, 2:1 transformer routing and input pad.

It offers six sample rate choices, external Superclock input, an insert point between analogue and digital sections and multiple A2D units can be slaved together. www.kmraudio.com www.apiaudio.com

New Euphonix control surface

and for US$1,695 delivers 5.1 with no external mixing or signal manipulation needed.

It has a multidirectional pickup pattern with 20Hz-20kHz frequency response on five perimeter channels and a discrete LFE microphone located inside.

The microphone is phantom powered with an LED indicator and comes with a 5m cable that terminates in a 6-pin Neutrik XLR connector.

A Windscreen and Pistol Grip are optional accessories.

The H4 SuperMini is

described as the first discrete 5.1-channel cameramountable surround microphone. Combined with an integrated multichannel preamp, monitor, and encoder, the SuperMini is based on the same patented Holophone Surround Audio capture technology.

At US$2,495, the SuperMini offers six microphone elements with a bandwidth of 20Hz-20KHz, a matrix encoded stereo analogue output, and six line-level analogue outputs that are available at one per channel from three stereo 3.5mm jacks.

It includes an audio zoom button that increases the forward bias of the pick-up pattern. In addition, the mic has a virtual surround headphone output with gain control, an auxiliary centre channel mic input (XLR) for attaching an external shotgun microphone, and a tri-coloured LED Holophone monitor that indicates sound level and direction. www.holophone.com

Euphonix has a new control surface for the System 5-B console. The new modules are operationally compatible with previous versions of the System 5-B but have higher resolution displays at the top of each module and new touch-sensitive knobs that include colour-coded LED rings at the base for identification of the knob’s function -– EQ-blue, dynamics-purple, aux sends-red. Euphonix has also changed the fader scale to give finer resolution around 0dB. The new modules include faster embedded microprocessors for quicker response and boot times.

The new modules are available for the three System 5 module types: CM408T channel strip modules (8 strips per module), CM401T master module and the CM402T central assign module with 8 assignable faders.

As well as being supplied with new systems, existing clients may upgrade to these new modules, which are available for all consoles in the System 5 range.

The EuCon Hybrid option for System 5, which now also includes Mac OSX compatibility, brings DAW tracks onto the console surface for mixing. The DAWs communicate with the console via the EuCon protocol and any DAW running the EuCon Client Application on the network can have its tracks selected for placement and control on the System 5 surface. Multiple tracks from separate DAWs may be placed in any order on the surface simultaneously. www.euphonix.com

Holophone H3-D and H4 SuperMini Holophone’s H3D microphone is based on t h e c o m p a n y ’s patented H2-PRO, and is a portable and economical 5.1 surround sound microphone.

H3-D is said to eliminate the need for difficult, e xp e n si v e , a n d time consuming m u l t i p l e microphone surround set-ups

May/June 2006

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review gear user-friendliness. Advanced features include infrared wireless synchronisation with the NET 1.

The switching bandwidth of 36MHz is freely selectable in the UHF range from 450 to 960MHz, and the transmitter frequency can be set in 5kHz steps.

NET 1 offers standalone operation via a front panel interface A single AA battery allows the SK 5212 to work for or connection via Ethernet to a PC for comprehensive six hours at its full 50mW output power, and the control of frequency co-ordination and all parameters vital transmitter power can be reduced to 10mW to to the operation of multichannel RF systems. The unit offers adapt to conditions of different frequency scanning and automatic frequency coordination regions. of all connected components, with wireless synchronisation Wind or handling noise can to transmitters and wireless monitor receivers from the front be filtered out with a two-stage panel, and real-time monitoring of frequencies, RF level, low-cut filter (60 and 120Hz) AF level and battery telemetry. Software may be updated and the input sensitivity is via the Internet. adjustable in 1dB steps The SK 5212 bodypack transmitter is said to surpass between -30 and +40dB. its predecessor the SK 5012 in terms of size, weight and www.sennheiser.co.uk junior page 142mm wide x 194mm high H OLOPHONE AD

NET 1 and bodypack transmitter

So Simple a Child Can Use It.

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Surround. Simple.

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A new version of Sonifex’s combined stereo RB-ADDA A-D and D-A convertor is 192kHz, 24-bit compatible and has all the features of the original RB-ADDA plus an optical input and output, front panel pushbutton controls and separate AES-EBU and Word clock synchronising inputs.

The RB-DS2 stereo audio delay synchroniser can provide up to 10.5 seconds of delay at 96kHz, 24-bit and internal Compact Flash allows for 2Gb of memory to be accessed permitting the shifting of a broadcast programme by one hour for a satellite rebroadcast.

Three new utility products have been announced. The RBPA2 is a dual stereo phono RIAA gram amp for converting pick-up cartridge signals on a turntable to line input level.

It has adjustable output gain and the frequency response is held to within 0.5dB of the RIAA equalisation curve.

The RB-ML2 is a stereo mic and line level limiter. It can operate in stereo or dual mono mode. The RB-MM1 is a mix-minus generator and has analogue I-Os, output level control, two null cancellation control pots and a switch for a bandpass filter. www.sonifex.co.uk

Waves bundle expansion

Waves has new expanded versions of its Gold, Platinum and Diamond bundles of plug-ins. Four new plug-ins for the Gold Bundle are Tune LT pitch-transformation, the IR-L Convolution Reverb, a mono/stereo Doubler effect and the Rennaissance Axx compressor. Waves has also added the L3 Ultramaximizer multiband auto-summing limiter as one of the 32 plug-ins comprising the Platinum Bundle.

The Diamond Bundle now includes 41 plug-ins including the complete Platinum, Transform and Restoration bundles, the aforementioned five plug-ins, and the Q-Clone processor.

Waves has also announced Universal versions of its complete line of plug-in products for Intel-based Macs. www.waves.com

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Enhanced Reds

K

Calrec Bluefin technology

"You should get a Holophone. They sound really cool and they are really easy to use. You should go down to the store and buy one. I can't wait to show my 3- movie to my Kindergarten class for "Show 'n' Tell"... It's gonna freak 'em out, big-time... Imagine what it will do for you!!"

Later. - Big J.

Visit us at Stand # 1016 AES Paris May 20-23 2006 Surround Sound Microphone Systems www.holophone.com 01.416.362.7790

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D e s i g n e d to meet the increasing need for 5.1 production, Calrec’s Bluefin High Density Signal Processing system provides 200% more processing power in 92% less space and at no extra cost. It provides 480 equivalent mono signal paths and is capable of providing 78 x 5.1 surround channels.

Bluefin processing offers full EQ and dynamics on all channels and also allows for 8 x 5.1 groups with full EQ and dynamics, 4 x main outputs, 48 multitrack outputs and 20 auxes.

Existing System Plus consoles may be upgraded

by Bluefin. www.calrec.com

May/June 2006

review gear Colour coding connectors Neutrik’s Translucent Coding Ring is a clear ring for the XX Series that allows individual customised labelling, branding and coding of connectors. An individual coloured and/or lettered paper or tape can be placed inside the ring, which provides protection for the label. The Translucent Coding Ring can be put onto a fully-assembled cable; desoldering is not required.

The XX-Series Right Angle Cable Connector is available in 3-, 4-, 5-, 6- and 7-pole configurations, and features a rotatable insert.

The Lighted EtherCon provides the components of Neutrik’s Shielded EtherCon but offers two light pipes for 3mm standard LEDs to indicate data transmission and status. The LEDs (typically mounted to a horizontal PC board) fit into an opening at the bottom of the light pipes; the light pipes transmit the light to the front panel. www.neutrik.com

Maselec MMA-4XR planned

Pitch ‘n Time LE

Leif Mases, the designer behind Prism Sound’s Maselec Master Series range, is developing a successor product for the MMA-4 4-channel mic preamp (pictured). The MMA4XR will be launched in June and each channel will have controls for gain (60dB in 3dB steps), phase invert, cut and 48V power.

Input and output headroom will extend to +28dBu and the MMA-4XR frequency response will extend beyond 200kHz. www.prismsound.com New Shoes 01|05|2006 11:23 Side 1

sound engineering

Serato Pitch ‘n Time LE is available as a plug-in for Logic Pro for pitch shifting and time stretching. Features include the ability to modify tempo independent of pitch and pitch independent of tempo without loss of timing accuracy and the ability to process Dolby matrix-encoded tracks without losing surround information. www.serato.com

NEWS FOR 2006

Fairlight Dream Factory package Fairlight’s HD Dream Factory provides broadcast and post communities with a complete production and HD finishing toolset for US$99,950. It comprises a twobay Constellation-XT large-format mixing console and Fairlight’s Pyxis HD nonlinear video system for recording, editing and mixing to picture in an HD compliant environment.

The Constellation-XT is expandable to 240 channels into 72 buses and supports all mix formats to 7.1 with a simultaneous multiformat mix creation system with automatic bus reduction capabilities. The basic package provides 56 channels into 24 buses, an inbuilt 48-track disk recorder and editor (expandable to 96 tracks) and an HD Tri-Level Sync Separator. Also included are 24 AES-EBU I-Os and 16 analogue I-Os, eight of which are relay protected.

Input and output configurations are expandable up to a maximum of 192 physical I-Os. www.fairlightau.com

KE 800 camera receiver Beyerdynamic’s KE 800 UHF camera receiver uses its ACT (Automatic Channel Targeting) to search for an interferencefree frequency to transmit via infrared to the matching handheld or pocket transmitter of Beyerdynamic’s Opus 800 and Opus 500 Mk II wireless systems and then locks in the frequency.

The KE 800 comes with an LCD that indicates battery level of the transmitter, frequency and channel, squelch, AF and RF level.

Accessories, such as adapter plates allow connection of the receiver directly to camera slots of Ikegami or Sony cameras. www.beyerdynamic.co.uk

May/June 2006

INGRID GOT NEW SHOES! - BOOTH 1305 -

[FOOTNOTE: OH, WE'RE ALSO SHOWING THE NEW SSA-2B]

LYDKRAFT

www.tube-tech.com resolution

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review gear SSL goes for video and MORSE

SSL has unveiled products based on its MediaWAN technology platform that encompasses all aspects of digital video acquisition and storage technologies. It’s the result of the merger of the operations of Broadcast Devices LLC (BDL), developer of MediaWAN, with those of SSL. BDL was founded in 2004 by Dave Engelke, one of SSL’s principals.

MediaWAN Gravity is a scalable broadcast production system that uses proprietary software running on standard high-performance PCs and servers. The software provides encoders to import video from a variety of sources including HD, a preview and job allocation interface for production coordinators, editing software for journalists, and management tools for news producers. An asset management database and user interface also provide information management in the video domain.

MediaWAN Backhaul allows media networks to install encoders at remote locations and automatically ingest video and audio content to central servers where it is

immediately available for use. The system operates over a private MPLS network with very low latency.

SSL’s MORSE — Modular Resource Sharing Engine -is designed to partner C100 consoles and enables any studio floor to be routed to any control room, allowing broadcasters to mix and match their source rooms to their available consoles. The MORSE system is comprised of modular I-O, stageboxes and router hardware, with low power consumption, and convection cooling. Optional PC control software can be used to create password-protected user accounts to control ownership of assets and the system is expandable.

C200 console V3.2 software includes GPI control possibilities that offer remote control of a wide range of external equipment and a DAW control option that provides workstation integration using technology from the AWS 900+. www.solid-state-logic.com

Neumann TLM 49 The retro design of the new TLM 49 is said to be optimised for vocal and speech recording and employs the renowned K 47 capsule that also featured in the M 49 and the U 47.

It is supplied as a set with an elastic suspension f o r a m a n u f a c t u r e r ’s suggested retail price of Euro 1,560. www.neumann.com

TC UpCon

UpCon is automatic 5.1 up-conversion audio software for TC’s DB8 and DB4 DTV processors and is based on TC’s UnWrap algorithm.

UpCon continuously monitors the format of the incoming audio, and if the signal falls back from true 5.1 to stereo it seamlessly crossfades into a ‘convincing’ 5.1 surround upconversion. UpCon may be inserted directly in the main HDTV transmission feed. One DB4 or DB8 processor can simultaneously, and in real time, condition SD and HD outputs. Latency through this up-convertor is less than 4 milliseconds.

The MD3 Stereo Mastering package is available for Pro Tools|HD. It includes MD3 Multiband Dynamics and BrickWall Limiter in a multiband 48-bit internal resolution structure.

The new BM 14S active subwoofer from Dynaudio Acoustics has an LFE output for daisy-chaining several subwoofers, and a high-pass filter that provides bass management in a 2.1 system. Delivering 300W and a claimed frequency response of 18-250Hz, the 12-inch woofer is housed in a closed compact cabinet. www.tcelectronic.com

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review gear Audition V2.0

Adobe Audition V2.0 software for mixing, editing, mastering, and restoring includes a new low-latency mixing engine, the ability to record parameter changes in real time, and new tools for audio visualisation and frequency space editing. It supports ASIO drivers, has flexible routing with sends and buses, punch-ins on the fly, and ‘unlimited’ tracks.

The new user interface permits you to dock and group workspace panels quickly and preview saved panel arrangements as custom workspaces. Royalty-free music beds and thousands of 32-bit uncompressed music loops are included. Other new features include Adobe Bridge for file access and management, Adobe XMP (Extensible Metadata Platform) support for embedding metadata into files, and broader video and audio format support. www.adobe.com

Focusrite Saffire LE Focusrite has brought FireWire Saffire power ‘to the masses’ with the LE version, which has six inputs and eight outputs (including stereo digital I-O) with comprehensive monitoring.

While Saffire LE lacks the onboard DSP of the original Saffire, it comes bundled with compression, reverb, amp modelling and EQ VST/AU plug-ins.

It also comes with F X s p a n s i o n ’s B F D ultralite, a demo version of GURU (Novation’s B-Station soft synth) and Ableton Live 5 Lite plus 470Mb of royalty-free loops and samples.

As with the original Saffire, input and output levels and latency-free monitoring can all be controlled via SaffireControl LE software. www.focusrite.com

Smart V1.5.1

Portable Circle Surround Encoder

S m a r t - A V V1.5.1 software includes stage one of Nuendo integration as well as a host of new console functionality and improvements, which are applicable across all Smart Console supported platforms.

This software includes support for Pyramix V5, and a number of features optimising the console for the Pyramix platform. These include support of Pyramix’s V5 SP1, the monitoring section, Multi-Stems, control of GPS stems and the addition of a number of automation controls. www.smartav.net

The SRS CSE-06P Portable Circle Surround Encoder prepares 5.1- or 6.0 multichannel audio for recording on two-channel recorders and ENG/EFP cameras. The unit includes a surround headphone-monitoring mode, SRS Headphone PRO, that monitors discreet 5.1-channel mixes over conventional headphones.

The upgraded SRS Circle Surround TDM Pro 2.0 plug-in includes a Music Mode for mixing and monitoring SRS Circle Surround-encoded music material. The plug-in also features the Xtract up-mixing feature that produces 6.1-channel surround sound mixes from mono and stereo material. SRS TDM Pro 2.0 now provides a broader front sound stage, enhanced lateral imaging, and better-sounding surround channels with a wider sense of separation. www.srslabs.com

A Range Of All-analogue Building Blocks designed by Mr. Rupert Neve • Available in both Vertical & Horizontal Configurations • Custom Rupert Neve Designed Transformers on Inputs and Output provide Sweet Musical Response of Mr. Neve’s Classic Modules • Highest Grade Components, Carefully Designed and Assembled in Texas by Rupert Neve Designs

McDSP Project Studio McDSP’s Project Studio RTAS includes Revolver LE convolution reverb, CompressorBank LE, Chrome Tone LE amp modeller, FilterBank LE, and Synthesizer One LE. Project Studio contains a p re p ro g r a m m e d McDSP Green iLok and is ready to plug and play right out of the box.

With the exception of Synthesizer One LE, all the other LE versions included support settings file interchange with the full-featured version. www.unityaudio.co.uk www.mcdsp.com

May/June 2006

For more information visit our web site

The Signature of Excellence resolution

www.RupertNeve.com 21

review gear Cadac MADI merge Cadac’s X16 MADI Merge Unit has been designed for integrated operation with the M16 remote controlled microphone amplifier and combines the 16-channel MADI streams from four individual M16s into a single 56- or 64channel MADI stream.

A 1U rackmount unit, the X16 is designed for 48kHz or 96kHz operation, in standard or extended MADI modes.

Each unit can receive four optical SC input 16-channel

MADI streams (from M16s or other MADI devices), combine the channels into a single 56/64 channel stream and route this stream to eight outputs (four SC optical connections and four coaxial BNC connections).

Cadac now offers a series of system packages for 64- or 32-channel systems, integrating multiple M16s, a R M 1 6 re m o t e c o n t ro l l e r a n d t h e X 1 6 i n a shockmount flightcase with power distribution and internal interconnect cables. www.cadac-sound.com

DV-RA1000 DV-RA 1000

recorder: the first to record both standard CD-R and high-resolution formats in one affordable unit. Created in collaboration with Sony, the DV-RA1000 is the ultimate 2track recorder, capable of DSD writing 24-bit PCM from 2.822 MHz 1-bit 109 min.

44.1kHz all the way up to 192 kHz 24-bit 66 min.

192kHz, as well as the ul174.4 kHz 24-bit 72 min. timate digital format, DSD 96 kHz 24-bit 133 min (Direct Stream Digital).

88.2 kHz 24-bit 145 min.

Whatever your final mas48 kHz 24-bit 267 min. tering sample rate choice, 44.1 kHz 24-bit 290 min. the DV-RA1000 does it.

As a mixdown recorder. A full complement of professional I/O, including balanced analog, AES/EBU and word clock, allow the DV-RA1000 to easily integrate into your studio. Mix in confidence to universally-accepted Wired remote DVD+RW media, mountable on a Mac® control or PC. DV-RA1000 is the obvious masincluded. tering choice for pro studios: DVD is as easily deliverable as CD or DAT but with significantly higher audio resolution.

As a live 2-track recorder. DV-RA1000 boasts an epic recording time of nearly 5 hours at 24-bit resolution on a single disc.

Built-in processors such as multiband compression and EQ ensure your live perfor-

mances are perfectly captured.

As an archive recorder. Preserve your analog and digital masters using DSD, the format used by Sony® for Super Audio CD (SACD) authoring. With multiple sample rates that allow simple digital transfers from aging formats such as DAT tapes, TASCAM ushers in the Direct Stream Digital era with the only affordable way to accurately archive your precious recordings.

Meet the New Standard at your TASCAM dealer. Or visit www.tascam.com for more information on the new DV-RA1000.

Rubicon 8 and sub

9 Up to 192kHz/24-bit recording 9 Records to DVD+RW and CD-R/RW media* 9 Multiband compression and 3-band EQ effects 9 ±6% pitch control 9 USB 2.0 port for use as DVD data drive 9 Balanced AES/EBU I/O, running at normal, doublespeed and double-wire formats plus balanced XLR and unbalanced RCA I/O

* CD recording is at 44.1 kHz/16-bit (Red Book spec) only. ©2005 TASCAM, a division of TEAC America Inc. All Rights Reserved. All specifications are subject to change without notice. All trademarks herein, heretofore and forthwith are the property of their respective holders TEAC UK LTD Units 19 & 20, The Courtyards, Hatters Lane, The Croxley Centre, Watford, Herts,WD18 8TE. Brochure hotline 01923 438888 e mail.info@tascam.co.uk

contractor

9 SDIF-3 DSD I/O for external conversion & DSD audio processing 9 Word Sync In, Out, Thru 9 RS-232C serial control 9 PS/2 keyboard connector for title editing 9 Records to CD-DA, DSDIFF & Broadcast WAV formats 9 Headphone output 9 Supports UDF disk format for cross platform computer compatibility

www. tascam .co.uk

dj and producer

TAS-86792A4-DVRA1Krev.indd 1

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Sonic Studio V1.0 Soundblade production package for Mac OS is a complete PCM production application that includes the premastering features of SonicStudio•DDP, along with Sonic’s 4-point editing model, comprehensive cross–platform compatible recording plus optional hardware acceleration. Soundblade also incorporates plug–in support, including AU and VST formats, and Sonic Studio’s time–aware DSP processing. The application provides multitrack editing for stereo delivery.

The Model 305 is an 8-channel, ‘audiophile’ FireWire interface with analogue and digital I-O, on–board DSP and eight channels of mic preamp, monitoring and routing. It can be mains or battery powered.

V2.2 of SonicStudio•DDP adds the ability to create Red Book–formatted CD-Rs that carry CD Text metadata.

It combines DDP playback and validation capabilities exclusive to SonicStudio•DDP and improves the auto–spacing capability while auto–fill of P-Q metadata reduces the drudgery of m a n u a l e n t r y.

To h a s t e n r e – sequencing, V2.2 has a re–engineered re–sequencing mode and the mark handling has also been refined. www.sonicstudio.com

The new DSD-ready mastering standard: TASCAM’s high-definition 2-track recorder.

Introducing the next generation mixdown

Soundblade and Model 305

personal creativity

The Rubicon R8a active ribbon studio monitors from Samson feature 2inch Velocity Ribbon high frequency transducers.

It features a bi-amped power section for 75W to a newly designed 8-inch inverted cone low-frequency driver and 25W to the ribbon. The two-way system is housed in a shelf-ported wood enclosure with phono, balanced 1/4-inch and XLR input connectors. There is also a 4-segment high frequency level control.

The Rubicon R10s is 150-watt active subwoofer with a 10-inch driver and a built-in active crossover, with variable crossover frequency, for connecting to satellite speakers. The rear panel has a footswitch jack to switch on the sub and a phase switch to time-align the monitors.

Balanced I-Os are on XLR and jack. www.soundtech.co.uk/samson

13/09/2005 17:58:02

resolution

May/June 2006

review

Tascam DM-3200 Digital consoles have suffered at the hand of the DAW and if they are to wrestle any control back then they have to offer a package of features that complement those of the workstation. ROB JAMES reports on a compact digital desk that gets the balance right.

T

HE ONE CONSTANT you can really rely on is change. Major upheaval is easy to spot but sometimes the wind shifting a few degrees can have more profound long-term effects. As DAWs have slowly grown in power and confidence and whole generations have grown up without preconceptions about the mixer in the order of things, so ‘conventional’ mixers have become increasingly irrelevant in many applications. Even the more recent generation of compact digital mixers is in danger of being sidelined unless there are changes to fit in with developing production practice. Among these new generations there is acknowledgement that at least some degree of hardware control is desirable — hence the burgeoning

market in ‘dumb’ control surfaces. Hardware machine and monitoring control still has a lot going for it and very few people would say no to some extra DSP horsepower. Get the package and the price right and there is still life in the compact digital mixer concept.

Tascam is no stranger to this market.

Its last mixer offering, the DM-24, was interesting and well engineered but somewhat flawed and mostly ‘me too’.

The DM-3200 builds on the virtues of the DM-24 and avoids many of the pitfalls, but this time Tascam is making a much stronger play to place the DM-3200 at the core of the studio, whether for music, post or live work.

The UK£2245 (+ VAT) DM-3200 is a 32-channel, 16bus digital mixing console, even at its full 96kHz/24-bit resolution. It features 48 inputs at mixdown, 4-band fully-parametric EQ and dynamics on every main channel, and two effects processors.

It mixes in surround, panning to up to 6.1 output channels, and the optional (£378 + VAT) IF-SM/DM surround monitor expansion card adds comprehensive surround monitoring, downmixing and bass management. The stylish meter bridge weighs in at £552 (+ VAT). Unusually, and most welcome, the angle of dangle is adjustable.

Although this console has much in common with its predecessor, the DM-24, it is less restrictive and is an altogether more fleshed out and wellrounded product. For DAW use the killer option is the (£297.02 + VAT) IF-FW/DM FireWire interface and 24 bidirectional channels over a single cable is the promise. The Tascam just shows up in the DAW

In and out

The DM-3200 is generously endowed in the connectivity department. Sixteen mic/line analogue input channels with analogue inserts and input level ranging from –60dBu to +16dBu. Phantom power is available, switched in blocks of four channels. There are four assignable balanced analogue sends and four returns and a –10dBv analogue 2-track input. Other analogue outputs comprise a main balanced stereo pair on XLRs plus balanced Control Room monitor on jacks, unbalanced studio monitors on phonos, and two headphone jacks. On the digital side there are two, 2-track inputs and outputs, one AES-EBU and one SPDIF respectively. The AES input has sample rate conversion.

Multitrack digital I-O is catered for by three TDIF sockets and a pair of ADAT optical Toslinks.

BNCs connect Word clock in, out and through, likewise three DINs do the same for MIDI. A USB B-type socket connects to a host computer for remote control and MIDI. Timecode in is phono. Strangely, there is no timecode out. Two 9-pin D-subs cover RS422 for Sony P2 protocol control and GPIs. A 25-pin D-sub connects the optional meter bridge. Two option slots accommodate a wide range of boards and last, there is a footswitch jack IF-ADDM (8-channel ADAT I-O) £158; IF-AEDM (8-channel AES-EBU I-O) £175; IF-ANDM (8-channel analogue I-O) £308; IF-TDDM (8-channel TDIF I-O) £158; IF-SMDM (surround monitoring card) £378; IF-FWDM (24-channel FireWire interface) £297. (All plus VAT).

24

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application as 24 ASIO or WDM I-Os. Complementing this, there is a remote layer with Mackie HUI and Mackie Control protocol options via conventional MIDI or MIDI over USB. Although this is a tantalising prospect, these protocols still fall some way short of full and convenient control over all the DAW’s functions and if you use a lot of plug-ins you may well still find it less of a chore to use the DAW screen, mouse and keyboard. This is no criticism of Tascam, rather a reflection of the current somewhat chaotic state of control protocols. It costs a lot of R&D time to support a remote protocol, at the controller and DAW ends, and until a more comprehensive standard is universally adopted it is entirely understandable if manufacturers go for the lowest common denominators.

Talkback, slating and machine control with Sony 9-Pin P2 protocol, MIDI timecode (with internal generator) and MMC are all standard. These features along with the optional surround monitor card, which adds full surround monitoring control, go a long way to making the point. This is a console designed to provide all the hardware facilities lacking in a DAW as well as adding considerable DSP horsepower.

The ‘Tascam Mixer Companion’ software application is provided with the console. This allows mixer data to be saved and reloaded and offers a Timecode/Transport window, a Meter Bridge window, a Preferences window, firmware update facilities and the ability to take screenshots of the DM-3200 screen. Arguably the most useful feature of such an application is missing. Inexplicably there is no graphic representation of the state of the entire mixer or remote control thereof.

The operational paradigm follows that of many other consoles with the well-established formula set by the seminal Yamaha 02R of multilayered surface, motorised touch-sensitive faders and assignable controls. The central monochrome LCD screen is the operator’s window into the complexities within.

To aid comprehension Tascam has dubbed logical channel strips as modules. Each of the 32 main input modules has 4-band EQ and dynamics (gate and compressor/expander), phase switch, delay, soft inserts, pan, aux sends, mute, solo and direct, bus and stereo outputs. Each of these 32 input modules can have two alternate inputs — main and return. This may have benefits for music recording and mixdown although when the return paths are used the strips become feature challenged in a similar way to the other 16 input modules that are intended primarily as returns and have just aux sends, mute and solo and bus outputs.

The stereo bus module, the 8 aux and 16 bus modules have compressor/expander, soft inserts, delay, mute and stereo outputs. Bus modules can also send to aux 1 and 2. The stereo bus obviously cannot send to itself and lacks a mute. Loopback allows any of the buses, auxes, direct outs and the stereo bus to be used as channel sources, just watch your tweeters!

Dither is only available on the stereo bus. There is a choice of truncate, dither or noise shaping.

Input bypass allows inputs to be patched directly to outputs, for example for format conversion. You can input an ADAT signal and output TDIF without going through the mixer. When strips are linked for stereo, pan becomes balance and width is available on the POD 3 encoder. Either module can be used as a mono signal or both together. If L or R is selected the balance control acts as a pan. Surround mixing is available in formats up to 6.1 but stereo sources cannot be linked for surround panning.

There are two effects processors, each capable of running the built-in and estimable TC Works Reverb processor as well as Tascam’s own time-domain, May/June 2006

review distortion and dynamics.

Although there is a global choice of encoder gearing between coarse and 1-step, I found coarse too coarse and 1-step too slow. Dynamic gearing would be nice although pressing Shift while turning a knob does invoke whichever option is not currently set.

Control surface design makes or breaks a mixer and, if anything this is becoming truer as consoles fight for their very existence. Tascam has managed to make the DM-3200 appear airy and uncluttered, no small achievement given the number of controls.

Some things naturally belong in certain places, fader strips and transport controls, for example, and the screen and meters. The 16 identical channel strips and stereo master occupy the bulk of the operator’s focus.

Each has a touch–sensitive motorised 100mm fader, mute, solo and select keys, record and indicator LEDs plus a rotary encoder with an annular ring of LEDs. Immediately above each fader is an OL/Status LED. This can indicate overloads on either the mic/line or return channel inputs with overload presettable from Over through 0dBfs down to –42dBfs. Alternatively the LED can show automation status for the channel.

Fader pitch, at 25.4mm, is somewhat more generous than some of the opposition. This contributes to the spacious air and will be helpful in live work.

One key to success or failure in hardware interface design is selecting the most useful controls to place on the surface and then putting them in the right place. Presupposing some familiarity with this type of desk, this is an unusually good example, with many controls intuitively falling to hand. Layer switches are large and exactly where you would look for them, as are the routing keys. The channel strip rotary encoders can be instantly switched between user selectable functions, pan, aux sends, etc. They can also be used as a horizontal channel EQ or dynamics strip. A flip switch swaps the encoder functions with the faders.

Four so-called PODs, rotary encoders with associated switches, are placed under the screen to control various soft functions. Housekeeping keys are to the right and the Compact Flash slot is to the left.

Project set-ups, libraries and automation data can be saved to the Compact Flash memory card. Dynamic automation, always a strength of the DM-24, is even stronger here. Touch-sensitive faders, write to end, variable revert, trim and grouping were already present along with a proper multi-pass mode. Now you don’t even need an external source of timecode.

The internal generator can be used as the master for automation and machine control. I’ve always thought this way around made most sense.

Methodical working is positively encouraged with a project and library model. Projects contain a vast amount of information and are stored on Compact Flash so they can be moved from console to console without the need to set-up from scratch. With this board you do have to remember to save your project at regular intervals and shut down properly before turning off. I would have liked to see a user-selectable auto-save function.

Taken as a whole, this is an easy surface to use with a reasonably flat learning curve.

There are some signs of cost saving to achieve the price point. There are seriously unpleasant fake plastic wood side panels, reminiscent of the worst excesses of the automobile industry, and some buttons that wobble about in a disconcerting fashion. However, the review console had ‘done the rounds’ and given the hard life it’s had, everything still works as it should, suggesting it is tougher than it looks. If the plastic wood offends, you could always remove it and either build the desk into a console or make your May/June 2006

own side cheeks from whatever exotic material slides your faders (Or you could fit them in your car. Ed).

The other major sign of the times is the large Made in China label on the box.

All consoles of this type only really shine when considerable time is spent exploring their possibilities and designing set-ups for various scenarios. With the DM-3200 this is even more important than usual because this console is so flexible.

The DM-3200 largely succeeds in justifying its existence to DAW users who might otherwise decide they don’t need a console at all. This is a highly cost effective way to get all the necessary peripherals in one box with a single FireWire connection to the workstation. The hardware has plenty of scope for future development to fulfil users’ wish lists. The console would certainly benefit from a much more comprehensive ‘helper’ application with a graphic

representation of the state of the entire mixer and the ability to change parameters from this.

As a standalone mixer it is appealing for live work due to its sensible layout, price and specification. This time, Tascam has demonstrated more of its old form and has come up with something that pushes the envelope. The DM-3200 is well worth a look. ■

PROS

Intuitive assignable; 24-channel FireWire option; surround monitor option.

CONS

Rotary encoder gearing; no autosave; no stereo pan linking in surround.

Contact TASCAM, JAPAN Website: www.tascam.co.uk

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25

review

CharterOak Acoustics E700 While microphone technology may not have moved on that much in the last ten years the available choice to users has been transformed by a much wider selection of brands.

JON THORNTON looks at an imposing mic that seems to have defined a new position in the landscape.

I

T’S BEEN SAID BEFORE, not least by me, but the world of microphone manufacturing is a long way from where it was even ten years ago. For sure, the big players are still there updating their product ranges, or in some cases still busily producing designs that have been around for several decades. But added to this have been an everincreasing number of new entrants.

Broadly speaking, they can be split into two categories — those who go down the ‘boutique’ route with highly specified, hand-crafted and usually very expensive offerings, and those who seem largely intent on maximising the lower reaches of the price/performance curve. It’s probably also fair to say that the latter rely heavily on manufacture and assembly in China.

In this context then, it’s not surprising that I hadn’t heard of USA-based CharterOak Acoustics or its microphone range before the E700 turned up for review. Founded in 2002 by engineer Michael Deming, CharterOak has quietly developed a line of microphones that have attracted a significant following.

To begin with, the product range was exclusively comprised of large diaphragm capacitor designs with valve-based electronics, but the E700 marks the first FETbased offering.

Featuring a dual 1.22-inch centre terminated diaphragm, the E700 features a switchable polar response (cardioid, omni and fig-8) and a twoposition pad (-10 and — 20dB). It’s a fairly squat looking microphone, with considerable girth and heft to it, and comes supplied in a hard case together with a basic shockmount. Finished in silver and black, it looks the part certainly, but I couldn’t help but get a sense of deja-vu — the smell of the packaging, the

26

construction of the microphone, even the supplied case seemed familiar and these things reminded me of brands of Chinese origin. A quick call to Michael Deming confirmed this — the majority of the components, including the diaphragms are indeed manufactured in China.

The units are then pulled apart, checked, reassembled and tested in CharterOak’s facility in Enfield, Connecticut.

Michael was keen to point out to me that a great deal of time had been spent with their Chinese manufacturing subcontractors fine-tuning the construction of the diaphragms and selecting quality components for the electronic stage and transformers, which was encouraging — nevertheless the E700 is pitched at a price point that gives it some stiff competition from some pretty established brands (US street price $999). A quick poke around the internals reveals a tidy looking PCB with all discrete circuitry, and a hefty output transformer whose casing forms part of the structure of the microphone.

In use, the E700 sets up quickly, and despite its weight, the supplied shockmount holds it very securely. It sounds pretty quiet — equivalent noise is quoted at 17dBA — and it delivers a nice healthy output level. On a variety of close miked sources, the E700 sounded very impressive — in comparison to a 414-BULS used as a reference it more than held its own, and actually sounded remarkably similar on the cardioid setting. There’s plenty of detail, lots of low frequency extension, and it's ever so slightly ‘hard’ at the top end.

Moving on to drums, and positioned as an overhead above the kit on an omni setting, and the E700 again sounded similar. Good transient response meant that there was plenty of punch to the sound, and the slight lift in the microphone’s frequency response around 10kHz kept things sounding detailed as distances increased. Low frequency extension was good, but in comparison with the 414 sounded a little more rounded in this respect — not a bad thing as it helped curb some less than pleasant hard resonances. This trait was even more pronounced when the E700 was set up as a distant room microphone — preferable in some ways to the 414.

The acid test with any microphone of this type, though, is vocals. Male vocals were the order of the day on the test session, and it’s clear that the E700 has been tuned pretty well to this task. As the 414 isn’t my favourite in this application, the E700 was compared to an Audio Technica 4050, which can deliver stonking results on most voices straight out resolution

of the box.

The E700 sounded a little less mellow than the 4050, certainly bringing out a little more ‘spit’ in the voice, but used close up has terrific presence and depth — a definitely larger than life vocal sound that would suit broadcast and voice-over applications as well as music tracks. Although it never sounds overexaggerated, the E700’s performance didn’t seem to take EQ quite as well as the 4050 though — trying to dip the low-mids slightly and dial in some HF lift started to deliver results that sounded unnatural quite quickly.

This might just be a function of familiarity with the microphone though, and more time playing around with EQ bands might help. Interestingly, for its other (valve-based) models, CharterOak offers a customisation service, where by changing valve types and altering component values in the electronics the response of the microphones can be tweaked to the customer’s preference. Sadly, this isn’t on offer with the E700; as Michael Deming explained, there just isn’t enough scope in the FET-based amplifier design to accommodate the tweaks.

All of which leaves me trying to decide where the E700 belongs in the scenario I outlined at the start.

Calling it a boutique microphone wouldn’t be strictly accurate. And putting it in the category of exploring those lower reaches of price and performance would be distinctly unfair — this is a quality piece of kit, backed up by a lifetime warranty for parts and labour. And it sounds good. It’s not as much of a ‘one-trick-pony’ as perhaps the AT4050 is, but it does have a definite sound to it. And if you like that sound, then it works beautifully. I think that CharterOak has actually defined a new position in the microphone landscape — quality design and Chinese manufacture, together with the attention to detail and the ear of an experienced engineer means that everybody stands to gain. ■

PROS

Good build quality; very nice ‘big’ vocal sound; nice rounded sound to LF; good detail and transient response.

CONS

Might be a little overstated for some applications and be a little hard to pull into shape with EQ.

EXTRAS

As Jon points out in the review, there are other mics in the CharterOak

stable. There’s the SA538 side-address dual-diaphragm valve condenser, which employs a centre-terminated S-1 capsule, the SA538B side-address dual-diaphragm valve condenser, which employs a side-terminated S-2 capsule, and the S-3 capsule S600 front-address valve condenser, which is sold in sequentially numbered pairs.

Contact CHARTEROAK ACOUSTICS, US Website: www.charteroakacoustics.com

May/June 2006

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review

Lynx Studio Technology Aurora 16 Now accepted as a quality influencing component in the computer’s add-on arsenal, convertors are still frequently locked in the box with the rest of the paraphernalia. ROB JAMES investigates an A-D/D-A that takes the convertors out of the computer and adds a whole lot more besides.

I

T IS NOW PRETTY well understood that digital workstations demand high quality convertors to give of their best. With this understanding has come a demand for products that fit the description and manufacturers have not been slow in bringing a sometimes bewildering array of designs to the market. Many convertors have established their bona fides where audio veracity is concerned, but there is more than that to designing and building a successful convertor product. If we take it as read that audio quality comparisons of the better examples have become a matter of theology rather than technology then, once the dross has been eliminated, the decision making process comes down to comparing convenience, productivity and cost effectiveness.

Lynx has a great reputation for producing DAW audio interface cards with some of the quietest analogue convertors ever seen (and not heard) on an internal card. The UK£1995 (+ VAT) 1U Aurora 16 departs from this theme by taking the convertors out of the computer and adding a whole lot more.

Essentially, Aurora provides 16 channels of AESEBU digital I-O (16 I-O in single-wire mode or 8 I-O in dual-wire mode), 16 concurrent channels of analogue I-O with 16 channels of A-D and D-A conversion. All I-Os are 24-bit at up to 192kHz. Although perhaps best suited to operation with a DAW, the device can also be operated standalone as an A-D and D-A convertor, a digital router and format convertor.

Front panel control is via six orange buttons and a forest of indicator LEDs. Multiple key presses cycle through the available options. Thus the sample rate button steps through all available sample rates when the sync source is set to internal and the sync source button goes through all possible source options. If no valid clock signal is available for the selected source its LED will flash. Similarly the To Analog and To Digital Out buttons cycle through Analog In, AES In and LSlot In. A further press lights all three indicator LEDs showing that the Aurora is under remote control. The Meter button switches between analogue and digital metering and the final button’s function depends on which metering mode is selected. When in analogue metering mode it switches all analogue input and output channels between –10dBv and +4dBu modes. In Digital metering mode it steps through the AES-EBU single and dual-wire modes. Single to dual wire conversion in either direction 28

is possible (although obviously the channel count is reduced in dual wire mode).

The meters themselves consist of two LEDs per channel. The bottom orange ones vary in brightness according to signal strength while the upper red ones light to indicate overload.

Lynx has a nifty control application for Aurora, downloadable from its website. I installed this on a PC and used a MIDI interface for communication to test.

I also used it on a PC with the Lynx AES16 Audio Interface card. In this case communication is carried over one of the AES multiway cables. Alternatively, the Aurora has an infra-red transceiver that will work with either a suitably equipped PC or Pocket PC. The software supports PC, Mac and Windows CE. While it is perfectly possible to operate the Aurora from the front panel, the control application makes things even clearer, offers extensive metering, individual channel patching and level control.

Analogue and digital inputs can be mixed for zero latency monitoring and more extensive mixing is available when used with the Lynx AES16 card.

All audio connections are on 25-pin D-sub sockets.

Two connect the 16 channels of AES-EBU I-O and the other four connect the 16 channels of analogue I-O. Word clock I-O uses BNCs and MIDI in and out are on the usual DINs. Analogue I-O is electronically balanced with selectable +4dBu and –10dBv nominal levels. The AES-EBU digital I-O is transformer coupled.

A blanking plate covers the Lslot expansion port.

Currently the only available option is the UK£189 (+VAT) LT-ADAT optical and a very reasonably priced LynxTwo/Aurora Interface Kit. Others are promised soon. The LynxTwo kit adds a 6-foot cable connecting a LynxTwo or L22 to a DB connector on an Lslot mounting plate. An internal ribbon cable is connected to the Aurora. This arrangement adds up to 8 channels of I-O at 96kHz, 4 channels at 192kHz (Aurora 8) and 16 channels of I-O (using Aurora 16) at 48kHz. Sync can be internal or derived from Word clock input, Word clock at half sample rate, AES A or B, or the LSlot. Lynx employs a technology it dubs Synchrolock to filter noisy and jittery external references to provide a stable lock and a low-jitter clock output. The stability claims are impressive. In the absence of any means of objectively verifying resolution

them I will say that I noticed no audible jitter artefacts during the review period. Synchrolock takes a minute or two to get its act together but once the LED is lit, the lock seems rock-solid. Synchrolock can be disabled using the external control software but, except for the terminally impatient, I cannot see any good reason for doing so.

In a crowded market Aurora distinguishes itself in several ways. The price point is well aimed for a box that crams this amount of quality conversion into 1U. Operation is simple and intuitive whether using the front panel or the remote software. It is this software control that makes Aurora more desirable in a DAW set-up than alternatives relying solely on buttons. For the golden eared, only a shoot-out in ideal conditions will clinch the deal. For other everyday professional audio applications, Aurora should definitely be on the shortlist wherever 16 channels of 192kHz bidirectional conversion are required. ■

PROS

Sound is up there with the big boys; versatile; logical.

CONS

Runs somewhat hot (but then so do most convertors); a sample rate conversion option would be nice.

EXTRAS

The 8-channel version Aurora 8 is available at £1495 (+VAT). The AES 16 (pictured) card is available at £545 (+VAT).

Contact LYNX STUDIO TECHNOLOGY, US: Website: www.lynxstudio.com UK, HHB: +44 208 962 5000

May/June 2006

review

Focusrite Saffire Audio interfaces may be falling out of the trees now but there’s still a little room for clever packaging and price breaking.

JON THORNTON appraises a box that has ‘be-all’ aspirations for the low end of the market but also has pro applications for location work.

A

LTHOUGH IT’S BEEN available for a while now, the Saffire is Focusrite’s take on a small, affordable audio interface. Using FireWire as its connectivity solution ensures that it supports the largest range of computer platforms, and also makes it eminently useful as part of a portable recording solution in conjunction with a laptop. This is an increasingly crowded marketplace, however, which means that something special is needed if it’s going to stand out from the crowd.

Although Focusrite has been involved in this sector for a while, partnering with Digidesign to develop the analogue side of the Mbox, it’s not altogether surprising that one of the few platforms the Saffire won’t support is Pro Tools. Unsurprising because the Mbox serves as much as a dongle as it does as an audio interface for Pro Tools LE. Users of Logic, Cubase, Nuendo or indeed any DAW platform that supports ASIO or Core Audio shouldn’t have a problem though. Saffire ships with a version of Cubase LE included, and it was this that I used for the review.

At its simplest level, the box is just an audio interface, offering two analogue inputs and an SPDIF digital input, and eight analogue outputs together with a single SPDIF digital output. Sample rates up to 192kHz are supported. For UK£350 this isn’t an altogether bad deal, and eight outputs even means that surround mixing up to 7.1 would be a viable proposition. There is, though, much more to the Saffire than this, at the heart of which is a certain amount of onboard DSP, a GUI that runs on the host computer called SaffireControl, and the additional hardware controls on the front of the unit.

Two XLR sockets allow microphone level signals to be input to the device. Phantom power is available on these inputs via a front panel switch, and gain controls are available at the top of the unit together with a three-stage LED meter. Plugging in a TRS jack to either of the two line-level inputs on the front panel overrides the microphone input, and these inputs are individually switchable between line level and highimpedance inputs for electric guitars. Two headphone outputs, each with their own level control, and a monitor level control with dim and mute buttons complete the front panel.

And it is these that start to illustrate the intention of the device. Focusrite’s assumption, I guess, is that most prospective purchasers will either be looking for a convenient location recording solution, or won’t necessarily have the additional studio ancillaries in terms of a mixing console, mic preamps or additional FX to make music tracking an easy proposition, and so they have built this functionality into the unit.

To really get to grips with this, we have to look at the SaffireControl GUI. Launching this brings up a window that looks relatively crowded at first glance, but once you get to grips with it is fairly straightforward. At the top left are controls for the two analogue inputs. The onboard DSP allows the insertion of a compressor and EQ or amp simulator plug-in across these inputs prior to the signals being fed into the DAW. Because the DSP is based in the Saffire itself, this means there are no CPU overhead or latency issues. Processing order here can be switched so that the compressor is pre or post EQ/Simulation, 30

and all of the processing has the option of working in one of two modes.

The first allows full control of all parameters, and the second toggles through a selection of presets for the compressor or EQ based on typical sources. In this mode, the full parameters are greyed out, and some more ‘user-friendly’ parameters appear, which in effect just adjust certain parameters. For example, picking the acoustic guitar preset for EQ brings up four parameters labelled ‘body’, ‘boom’, ‘clarity’ and ‘lightness’, which really just adjust the gain of each of the four EQ bands whose centre/turnover frequencies and responses have been preset. Selecting this option for the compressor presets threshold, ratio and time constants, and leaves you with a single parameter labelled ‘amount’. In practice, these preset options work rather well, and are certainly good starting points for less experienced users — personally I preferred to just dive in.

The quality of the EQ, compression and simulation on offer here is also good, as are the mic preamps.

They aren’t necessarily going to replace your favourite hardware and software solutions, but they are perfectly adequate, and in the case of the preamps nicely quiet.

A further section of the GUI window shows stereo faders, which address the levels of the first ten tracks of the DAW. These effectively overlay their gain settings onto the levels already set in the DAW when the Saffire is connected, and form the basis of setting up a playback mix for monitoring purposes. The bottom half of the control window allows inputs 1 and 2 (analogue) and 3 and 4 (SPDIF) to be balanced and mixed with this monitor mix, and then sent to a pair of outputs. Along the way, the analogue inputs can also have onboard reverb (individual reverbs for each input) applied to them, again with no host CPU overhead. The monitor mix from the DAW can either resolution

be the generic mix set up earlier, or a custom mix of the first ten tracks of the DAW can be set-up for each of the output pairs.

Although the output pairs feed the logical outputs on the back of the unit, outputs 5&6 and 7&8 also feed the front panel headphone outputs. Any or all of the analogue output pairs can also be switched to have their output level controlled by the front panel monitor level pot — useful for surround monitoring.

In all, this adds up to a very flexible tracking and monitoring system, if one which is a little confusing at first glance. Focusrite has thoughtfully provided a mode switch, labelled Sound Card and Track, which while it doesn’t alter the basic functionality of the unit, sets routing and mix levels in Sound Card mode such that each output is simply a direct output from the DAW. A ‘Float’ function also ensures that the GUI is constantly floating over the DAW application, meaning that constant switching between applications isn’t necessary in normal use.

There is a lot for the money here. A slight disappointment is that the onboard DSP is only available for input sources — you can’t access it in mixing if your CPU is maxed out although you do get the same processing supplied as VST or Audio Unit plug-ins for use at the mix stage if you want them.

And while many professional users will probably not need the routing and mixing capabilities for tracking purposes, this is a very flexible system for location work. Adding a stereo mic pre with a digital output option that could feed the SPDIF input would double the number of useable inputs, which would really make it an attractive proposition. ■

PROS

Flexible routing and mixing options; onboard DSP and processing for tracking; useful number of outputs; lots of functionality for the money.

CONS

SaffireControl GUI a little busy at first glance; no way to access onboard DSP in mix stage; only two analogue inputs might be a little restrictive for some users.

EXTRAS

Focusrite’s Saffire Pro 26i/o FireWire interface has 24-bit/192kHz processing and software control for 52 channels

of I-O (26 in, 26 out). Combining eight channels of preamp with eight analogue outputs and 18 digital I-O (two SPDIF and 16 ADAT at 48kHz), as with the original Saffire the unit can be buspowered or powered via an external PSU. SaffireControl Pro provides a control application of a similar format to the original SaffireControl but also has talkback. The front panel provides two independent headphone outputs.

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

May/June 2006

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CineFile Integrated Film Recorder & Dubber

review

TC Electronic Fabrik C and R Many believe that while a lot of energy has gone into the development and harnessing of processing power, substantially less has been invested in the development of the interface. Now there are plug-ins with a difference and a certain minty freshness about ROB JAMES.

T

HE POWERCORE PREMISE is a simple one, the computer’s CPU has quite enough to do without overloading it with processing plug-ins.

So, provide extra DSP horsepower specifically for the purpose but present the results to the user as a normal plug-in. Although this extra power is only available to plug-ins specifically written for PowerCore, TC has kept its promise to provide a steady flow of quality plug-ins from its own and third-party-developers.

This has helped to establish the PowerCore family as a force to be reckoned with.

The Fabrik plug-ins are just the latest in a long line from TC: Fabrik C is a plug-in channel strip with EQ, de-esser and compressor; Fabrik R is a reverb with four basic algorithms, Live, Hall Plate and Club. Both are thoroughly worthy and the army of TC fans will desire them for their characteristic sound. However, these Fabrik plug-ins bring something new to the party, presaging the long overdue dawn of a new age.

An age where user interfaces do not rely solely on an on-screen simulacrum of knobs and faders, borrowed from ancient analogue technology.

TC has dubbed the new interface ‘Meta Intuitive Navigation Technology’ a MINT with a whole new approach to controlling processing parameters. Meta is one of those buzz words, much beloved by geeks and hackers. Although now in common use, it is less well understood than it should be. It comes from the Greek (and should not be confused with feta, which is also from the Greek. Ed) and has at least four meanings. Used as an adjective prefix it can denote a change of position/condition or after, beyond/behind, or, especially in anything to do with computers, meta can denote a higher level of description. Thus, meta language is language used to describe language.

However, in the current argot, meta also has an existence as a noun meaning self-referential, often in the sense of parody. E.g. ‘Star Trek is just so meta.’

It’s a user interface Zen, but not as we know it.

(Rob you’re just so meta. Ed) Laudably, TC has brought this innovation to market at the very reasonable price of UK£205 (+ VAT) for each plug-in.

Fabrik R combines four new TC algorithms in a single reverb plug-in. The interface is roughly divided into four sections: Algorithm, Level, Tweak and meters. The Algorithm page gives you a choice of Live, Hall, Plate and Club. Each algorithm has a distinct character. Live is bight and breezy, Hall 32

adds space without imposing too much of its own character. Plate has the bright, diffuse sound of springs and plates. Club manages to add space while retaining intimacy.

The Tweak section is organised in four pages, accessed by clicking on the lettered buttons. Eschewing conventional knobs and faders per parameter, MINT presents groups of parameters attached to a single movable button. This takes account of the interactivity of certain controls and the need to adjust several using a conventional interface to achieve the desired results.

Parameters can be changed by clicking and dragging the buttons, clicking in a parameter field and dragging up or down, or double clicking and entering a numeric value.

Reverb controls Decay and PreDelay, Modulation offers Rate and Depth, and Color affects the spectrum of the diffuse field with three parameters LoColor, HiColor and HiFactor. HiColor determines which frequency area is in focus and HiFactor boosts or cuts.

Distance defines the balance between Dry, Early and Reverb levels. Of course there are factory and user presets and these can be organised into folders. Any section of the interface can be locked when a preset is recalled.

Fabrik C is a 4-band parametric EQ, simple de-esser and 3- or full-band compressor in a single plug-in — in effect all the building blocks of a channel strip in one. The EQ ranges from surgical notch filtering to gentle rolling slopes with a goodly choice of filter types. Here, TC has missed a trick and could better exploit the MINT aspects, for example by using rightclick + drag for Q in the EQ graph. The De-esser is controlled by a single fader and is effective enough but not on all material. The compressor section uses the MINT principle to greater effect. Source selects the type of material to make settings of ratio, knee attack and release and in 3-band mode there are three pages of parameter Tweaks. Used in Full-Band mode the only control is Gain and this balances Input Gain and Gain Make-up. In 3-band this is accompanied by Compressor Control which balances the amount of compression between the three bands.

Level compensation allows you to compensate for the effects of the Compressor Control settings.

The MINT benefit is most apparent with the reverb but also evident in the compressor section of the channel strip. By letting go of some preconceptions and simply tweaking by ear it is possible to achieve some eye opening effects quickly and painlessly. The resolution

speed advantage should not be underestimated in either plug-in. The quicker adjustments are made, the less need there is to compensate for your hearing becoming too subjective. Initially, when inserted in Wavelab’s master section, I cursed the Fabrik R roundly for not having a dedicated wet/dry mix fader. Later, once I became more familiar with the MINTerface, it all began to make sense.

We’ve been banging on about lack of innovation in the user interface for years. Despite the unconventional appearance, MINT is less radical than it looks but, although it won’t warp the fabric of the entire universe, it is a somewhat cautious step in the right direction. TC has largely succeeded in producing an interface that allows the user to get results quickly and painlessly. The effect of the controls is instant and clearly audible, subtly shifting the emphasis in favour of listening. You can try things in an instant that would take considerable time to set up with a more conventional set of controls.

Of course, all of this would be pointless if the effects themselves were not of an extremely high standard. Fortunately, as you would expect from TC, they are well up to scratch. The reverbs and EQ are especially enticing. As users become familiar with this unconventional interface I hope TC and others will be encouraged to take the concept further. ■

PROS

Instant gratification; great results with little effort; at last some original thinking.

CONS

If you want to get your hands dirty tweaking every possible parameter, look elsewhere; won’t work for all material but at least you will discover this quickly; could be more radical.

EXTRAS

The PowerCore range has been expanded with a PCI Express (PCIe) version. TC will continue to offer the existing PowerCore PCI mkII and PowerCore Unplugged versions.

Contact TC ELECTRONIC, DENMARK: Website: www.tcelectronic.com

May/June 2006

review

Speck Electronics X.Sum Mixer It’s encouraging to still find outboard that offers something different and this summing unit offers more than your typical hard left and right pan arrangement. JON THORNTON discovers a box that he wasn’t sure he’d need until he used it.

T

HERE IS A CURIOUS category of audio equipment that seems to be determined by the needs of a few, and which when actually manufactured elicits one of two responses. Either ‘nice box, can’t see why I’d ever need it’, or ‘I’ve been looking for something like that for years, where do I sign?’ In the case of Speck Electronics’ X.Sum, you can only imagine that someone, somewhere decided that it would be exceptionally handy to have a 32channel, 4-bus line mixer in a 1U package, and the rest is history.

Leaving aside your own determination of needs for a while, the X.Sum (UK£945 +VAT) certainly represents a tidy bit of packaging, cramming 16 dual-concentric pots onto the front panel, each of which allows adjustment of level and pan for a stereo source. Said sources interface to the rear panel via individual 1/4-inch TRS sockets, with separate balanced inputs for the left and right leg of each stereo pair. Also on the rear panel are the main mix bus outputs — again electronically balanced and appearing on TRS sockets, and an RJ45 connector that allows a number of additional output options via breakout boxes — or the ability to ‘daisy-chain’ more than one X.Sum together (more of this later). The power supply is an external box, but a reassuringly hefty one — always a good sign I feel.

Returning to the front panel, there are some thoughtful design features that make the unit more flexible than it might appear at first sight. Although there are 32 individual inputs, arranged as 16 stereo pairs, you have a choice about how to deal with them.

In normal operation, centring the pan control (outer ring) routes left inputs to the left bus, and right inputs to the right bus. Moving the pan control to the right will progressively remove the left leg of the pair from the left bus, and vice versa. Each of the 16 channel pairs also has a mono button, and pressing this generates a mono summed signal from both inputs.

Now the pan control effectively pans this mono signal between the stereo buses.

There is a choice of buses as well. A routing switch is provided for each stereo pair, and pressing this will route the signal to the main bus. Not pressing it means that the signal is actually routed to the secondary ‘MixB’ bus. This is a strictly one or the other proposition — there’s no facility to send a particular channel to both sets of buses, but it does allow a degree of flexibility.

34

For example, using one bus to act as a monitor for a DAW while the other provides the front end.

The main stereo bus has a dedicated master level control, and the option of switching it to become a mono mix. A further level control sets headphone monitoring level, with the headphone socket located conveniently on the front panel. An associated switch allows the headphones to monitor the ‘MixB’ bus, but this somewhat curiously also makes the headphone level control act as a master fader for the Mix-B output. While this saves a little space on the front panel, there may well be occasions where you might need to have independent control over bus level and headphone monitoring.

While the main mix bus has its outputs available on the rear panel, the Mix-B outputs only appear courtesy of the ‘multi’ connector that appears on a RJ-45 socket. The review model was provided with a breakout box that brought Mix-B outputs onto 1/4inch jacks. Other permutations of breakout boxes are available from Speck, or you can roll your own as a full pin-out is provided in the documentation. Another function of this connector is to allow daisy-chaining of multiple units by accessing the post-summing but pre-fader main mix outputs of one box and presenting them to the summing mixer of another unit.

In use, things are pretty straightforward. There’s an indication on each level control for unity gain, but that’s about it in terms of signal metering — you need to exercise your ears to determine if anything nasty is happening. Thankfully, this isn’t a problem because the X.Sum seems to have huge reserves of headroom, and sounds very neutral, clean and quiet. Even with all inputs routed to the main bus at maximum level, background noise was minimal, as you’d expect with such a compact topology — certainly quieter than my mid-range analogue recording consoles. This makes it equally suitable for use as a flexible analogue summing box for DAWs (if that’s your thing), or as a keyboard sub-mixer in studio or live applications, or indeed for that application you didn’t realise you had until you actually see the box.

In summary, despite its unassuming appearance, this is most definitely a professional product — it’s nicely built, well thought out and makes no compromises on overall audio quality and signal path. It’s not without competition, certainly in the resolution

area of dedicated summing mixers, but its additional flexibility in terms of routing and panning options gives it a definite edge. ■

PROS

Good neutral sound; bags of headroom; flexible panning and routing options; scalability with multi I-O.

CONS

Slightly strange configuration of monitoring options for second mix bus; no form of metering.

EXTRAS

Three breakout boxes are offered.

Mix-B Output Breakout Box — connect

this to use the additional stereo MixB outputs. Combo Breakout Box has a set of 1/4-inch TRS jacks to use the additional Mix-B outputs and another set for a second set of Mix-A outputs. The External Preamp Breakout Box allows you to add your own external preamps to the X.Sum.

Contact SPECK ELECTRONICS, US: Website: www.speck.com UK, ASAP: +44 207 231 9661

May/June 2006

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review

Summit Audio FeQ-50 A new product from Summit is always of interest and this passive equaliser sports some fine features and an iron-based constitution.

TERRY NELSON finds he needs very little persuading.

T

HE FEQ-50 IS THE latest addition to the halfrack series from Summit, with the chassis very much in the same style as its predecessors while still managing an uncluttered layout in spite of quite a high control density.

The unit (US$995) features four EQ bands, with a top row of four black level controls (+/-14dB) and a bottom row of four staggered 6-position rotary switches with typically Summit-style red knobs.

Each rotary switch also has an associated miniature toggle switch for peaking or shelving operation in the high and low bands and two Q settings for the two mid bands.

The front panel is completed by a high-pass filter toggle switch situated in the centre between the gain controls and a vertical group to the top right of the chassis that houses the red power On indicator and power toggle switch and an EQ In/Bypass toggle.

These controls line up with those of the other halfrack modules, which means that they look very tidy when stacked vertically.

The rear panel has an IEC mains socket with fuse; two output sections, each with XLR and TRS jack connectors, and a Neutrik Combi input for XLR and TRS jack.

The FeQ-50 circuitry features a discrete transistor input stage and buffered solid state and tube output channels, together with balanced +4dB and –10dB outputs for a total of four separate outputs from the one unit. This flexibility means that the equaliser can be used as an active splitter and level interface before getting into the EQ functions.

The fact that the unit can be used as a line amplifier, i.e. the circuitry is active in bypass mode, encouraged me to see what effect could be obtained by just running signal through it without any further processing. The results were astonishing, to say the least.

The test set-up consisted of a CD player feeding a stereo channel of a Neotek Elite console set to mono and routed to a multitrack bus to feed the FeQ-50. The two outputs of the latter were patched into the line inputs of two channels on the console to enable A/B comparisons. The source material included my trusty Sound Check 2 test CD, which features raw tracks of vocals, instruments, ensembles and final mixes; great for testing and setting up equipment.

Even though levels were carefully set up on the metering, the subjective response of the solid state output of the Summit was such that it seemed a bit 36

louder than the tube output. However, I soon put this down to the fact that the solid state signal seemed to be more ‘upfront’ than the tube output, which, to my ears, had a more natural and rounded response.

Great, two signals for the price of one!

I went through all the CD test tracks and found an interesting variety of different responses. Which response is better is down to personal taste but the tube output definitely smoothes out tracks that could be considered to be a bit aggressive or lacking in depth — it also felt more musical. The solid state output puts things very much ‘in your face’, which could be helpful for making tracks pop out in a mix when applying more level is not the solution. That said, the real icing on the cake is combining the two outputs — talk about pseudo stereo and all stations in between: the spread and depth of the signal has to be heard to be believed.

The EQ characteristics are those of a passive LC resonant circuit, which Summit claims provides good phase coherence, low transient intermodulation distortion and a musical harmonic structure, so let’s switch in the sections and see what happens.

The low and high bands have peaking or shelving characteristics and six switched frequencies at 33Hz, 60Hz, 100Hz, 150Hz, 270Hz, and 410Hz and 5kHz, 7.2kHz, 8.2kHz, 10kHz, 12.5kHz, and 18kHz respectively. The low and mid bands have switched Q settings of just under 2 octaves and just over 3 octaves and switched frequencies at 390Hz, 470Hz, 580Hz, 680Hz, 820Hz, and 1kHz and 1.2kHz, 1.5kHz, 2.2kHz, 3.3kHz, 4.7kHz and 5.6kHz respectively.

All of the centre frequencies in the LF band were found to be useful and it could be said that the peaking response adds focus whereas the shelving adds depth or removes mud and unwanted content.

In the LMF and HMF both of the Q settings are quite broad but the tighter Q is definitely more ‘surgical’, whereas the wide response tends to expand the signal. The HF band was found to be very dependent on the programme material when choosing the centre frequency. When harmonics are present, the upper ranges of 12.5kHz and 18kHz can add a lot of ‘air’ and open up the response, particularly in shelving mode. As with the LF band, peaking mode provides a ‘focus’ function.

The 30Hz high-pass filter has a 6dB slope and is subtle but effective, cleaning up vocal and instrumental signals very nicely without removing essential LF content. It also provides a useful ‘hinge’ effect when resolution

used in conjunction with boosting of the lower ranges of the LF band.

At this point it should be mentioned that it is easy to overdo it — this seems to be a quality of all good equalisers — and you need to compare EQ in/out to realise the extent of the modifications.

The FeQ-50 provides two very different responses for the price of one; the solid state and tube outputs are that different. Subjectively, the tube output has more depth and what I can only describe as the Summit ‘halo’ around the sound.

Switching to bypass after setting an EQ definitely makes you feel that something is lacking and use of the unit on live male and female voice showed that a lot of sound tailoring is possible. I even used it on two piano tracks that could be best described as ‘nasty’ and the FeQ-50 saved the day and proved itself to be a vital tool to restoring musicality. Compliments to the chef on this one! ■

PROS

Lots of flexibility; sounds great; high value for money; completes a ‘channel strip’ for the half-rack series.

CONS

No in/out switches for individual bands; I will have to buy one; you will want multiple units!

EXTRAS

Other half-rack format Summit units include the TLA-50 tube levelling amplifier, the 2BA-221 mic and line module, and the TD-100 instrument preamp and tube direct box.

Contact SUMMIT AUDIO, US: Website: www.summitaudio.com UK, Stirling Trading: +44 208 963 4790

May/June 2006

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review

Eventide Anthology II Eventide has as much history as anyone when it comes to digital processing and it’s now taken its Legacy and arrived at an Anthology with the addition of some important plug-ins. GEORGE SHILLING unbundles the produce.

E

VENTIDE’S ANTHOLOGY II plug-in bundle brings together previous releases with a few new goodies in one easy-to-digest package.

The Clockworks Legacy bundle is included; this was covered in Resolution V2.5 and recreates the original hardware H910 and H949 Harmonizers, Instant Phaser and Instant Flanger, and Omnipressor. In addition to those five plug-ins Anthology II includes H3000-based Factory and Band Delays plug-ins, Octavox harmonizer, Eventide Reverb, and new for this collection are the Quadravox, the Ureiinspired EQ45 and EQ65, and the E-Channel and UltraChannel plug-ins. Additionally, Precision Time Align allows ultra-small delays using a simple slider —- negative amounts can be achieved thanks to Pro Tools’ Delay Compensation feature.

H3000 Factory is based on the mod factory|one and mod factory|two algorithms that made later H3000s so much fun. Eighteen effects blocks allow patching anything to anything, and there are sweeping delays, pitch shifters, filters, and amp modulators. Modulation sources include two LFOs, envelope generators and so on. Creating effects is still a complicated business, but there are a huge number of presets to get you going. However, I missed some of the original H3000 mod factory presets, and it’s a shame these aren’t all included.

Playing around with this plug-in is terrific fun, and with the greatly expanded display, including virtual patchcords and level sliders, it is much easier to program than an H3000, although there is still a bewildering array of parameters spread across Expert and Function tabs. Although you can automate parameters by the standard method, an additional bonus is the 2 x 16 memory Snapshot section. The two banks operate as Global and Current memories, allowing MIDI control firing off presets for an individual song as well as a universal set. Presets vary from the downright bizarre fuzzed-up digital distortions, wahwahs and alien noises, to lovely stereo widening delays, pitch changers and modulations.

H3000 Band Delays provides eight voices of tempo-based filtered delays. The filters are fully parametric and their frequencies can even be MIDI controlled. Plenty of off-the-wall effects are possible here, great for sound design, ambient textures and unusual atmospheres. Again, Snapshot memories are usefully provided.

Although not presented as such, the two EQ plug-ins designs are obviously copied from vintage Urei hardware. The EQ45 is based on the 545 parametric EQ, while the EQ65 is inspired by the 565 ‘Little Dipper’, apparently a popular filter unit with US postproduction engineers. These boast very low latency. The 45 is a fairly conventional parametric, with high, mid and low peaking bands, plus a confusingly named ‘multiband’ that simply means that the frequency knob covers just about the entire audio spectrum. The EQ character is fairly surgical —- the variable bandwidth seems quite narrow at the default centre position, without being nasty — and Low and High Cut filters are steep and wide ranging. I didn’t miss having a graph, but an overall gain control would have been handy.

The 65 is rather unusual, with similar 38

variable from 0 to 24dB and conventional but wideranging controls, and it can be sidechain-controlled.

The EQ includes two different styles of peak EQ, as well as shelves and filters. The display is clear and the EQ is a pleasure to use, sounding pure and clean.

The output section includes a Transformer button for subtle added character.

The Ultra-Channel adds a de-essing section to the compressor, but there is also an Omnipressor for more characterful compression. A Micro Pitch Shift section adds instant stereo harmonizer width, and stereo delays. Eighth-note tempo sync is available although the manual doesn’t explain properly how to achieve this (it’s not difficult). Oddly, there are no presets provided for this plug-in.

Quadravox and Octavox provide easy to set harmonizers with four or eight voices respectively.

Intelligent key-related pitch correction is possible, as well as rich chorusing effects. Presets are bountiful, and displays are clear with level and pan sliders, individual delay settings, and a musical stave with key and scale selection. A selection of algorithms is provided for different instruments and voices. These are very simple to use and great results are easily achieved.

The Eventide Reverb is the jewel in the crown, with some extremely rich and smooth sounding algorithms.

There are nine basic types, two each of Hall, Room, Chamber and Plate, plus an Ambience setting, all with the same selection of editable parameters.

Encouragingly, parameters are not too bewildering and numerous, and have been logically ordered, with the most often used Decay Time at the top of the list, then Predelay, Diffusion, Room Size, and so on.

There are additional sections with separate stereo delays (which act on the dry signal), and separate parametric EQs for tweaking the signal pre and post reverb; the reverberator itself and delays additionally each have separate high and low shelving EQ. This gives you a huge amount of tonal control. A Lo-fi parameter supposedly emulates bit-reduction —- it just sounded like a filter to me. Additionally, there is a fully featured compressor that can be placed pre or post reverb. Snapshot memories are again provided. All these add-ons are most welcome, but even disregarding the extras, this is an excellent reverberator, with an overall warmth and sheen rarely matched by plug-in reverbs. Reverb tails disappear smoothly, and this plugin seems to strike a nice balance between the richness and sparkle of digital reverb and the smoothness of convolution plug-ins. ■

High and Low Cut filters, but featuring a pair of extremely narrow Notch/Peak filters, each with six modes: 5%, 10% and 50% of centre frequency for Notch or Peak. This is an unusual and useful tool for problem solving, and both EQs sound remarkably analogue, thanks to 48-bit double precision filters.

The E-Channel comprises a basic set of processing tools -— a gate, compressor and 5-band parametric EQ. The VCA-style compressor/limiter has a knee resolution

PROS

Characterful but useful plug-ins; a bargain when bought in this package (US$1195); comprehensive modulation and pitch changing; one of the best reverb plug-ins; OS9 still supported.

CONS

No corresponding AudioSuite plug-ins; some H3000 presets missing; E-Channel unremarkable.

Contact EVENTIDE, US: Website: www.eventide.com

May/June 2006

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craft go back ... and it will be running again. There’s no pen or menu and just a very limited PC file system, it’s kind of idiot-proof in that sense. I’m not a very technical engineer, I don’t use a massive amount of outboard, and I’m not hugely bothered if it’s 96k or 8k, if the song feels right and what’s coming out of the speakers is good then I’m happy.

Looking across the console, you haven’t got many EQs punched in ...

I don’t really use a lot of EQ — it’s all done on faders — there’s a lot of automation and rides. A lot of my EQs are getting rid of things, cutting frequencies. We’d all like to have the space in our tracks that Hip-Hop and R&B has, where the bass is usually extremely syncopated, not sustaining with lengthy notes, so nothing is getting in the way.

Tom Elmhirst Tom Elmhirst is making a name for himself in sculpting the sound for some of the UK’s coolest new bands. He talks to NIGEL JOPSON about Neves, playing on the songs he mixes and the politics of stems.

T

OM ELMHIRST IS A LEADING exemplar of the current crop of top engineers — old enough to have learnt the trade in an era of tape reels but young enough to be at the top of the game. He’s a specialist mix engineer, busy working on albums for hot new signings such as Paolo Nutini, Diefenbach, Hot Chip and Lady Sovereign. Tom mixed the Royksopp single Only This Moment as well as singles for Sugababes and Hard Fi. He started his career working for production Svengali Trevor Horn at Sarm Studios, for whom he recorded Wendy and Lisa, The Art Of Noise and Seal.

He experienced the dramatic escalation in recording workflow as Sony DASH PCM recorders were replaced by early Digidesign systems at Horn’s studio, and made the move to freelancing under the stewardship of Sarm’s in-house management team (he is currently managed by Osohso Music Management). A stint assisting Steve Albini recording Bush hatched a good working relationship with the band, which landed Tom a breakthrough job recording and mixing their album The Science Of Things. Working with programmer Johnny Rockstar on this session led to a meeting with All Saint’s producer Cameron McVey, which in turn led to work with Sugababes, for whom Tom recorded and mixed the album One Touch and several singles.

Three years later, Elmhirst was mixing the critically acclaimed Black Cherry album for Goldfrapp, and working on signature albums for major label artists such as the Manic Street Preachers’ Lifeblood.

Resolution spoke to Tom as he prepared to push up some faders on the Neve VR in Room C at London’s Metropolis Studios.

40

Was it your call to mix here on the Neve VR72?

Yes, I like the connection I have with it, it’s very immediate, nothing’s happening that I can’t see and it’s fairly limited in what it can do. The VR is quite a purist mixer, essentially you’re dealing with balance, there aren’t endless pages of menus. I grew up working on SSLs at Sarm, the very first time I worked on a Neve was when I went freelance. I would never dream of mixing out of two outputs in Pro Tools, I like the physical bit of mixing ... I like pushing it, which you can’t really do in a computer, I find you run out of headroom quite quickly! The Neve has a fantastic amount of headroom, I’m quite ill-disciplined, I know that if I was mixing on an SSL things would be going red here and there. I just don’t get that with this board, and if I do get it I rather like it. People have this misconception that SSLs sound hard and Neve is soft and warm and cuddly — I can make it plenty hard if you want! I would mix on an SSL, but I like the simplicity and the headroom of the Neve.

Have you got a system for setting up your mix?

I throw it all up really quickly, I have an extremely short attention span so I won’t spend two hours on the drums. I know what I want to do, it’s just finding how I get there. I use the A and B bus on the VR console a lot: I set up the A bus to cut all the vocals and the B bus to cut the music. Then I can switch between the two really quickly without altering any automation I’ve written.

I’m quite lazy by nature so I just want the quickest, easiest, fastest route to what I want to do. I don’t want to start by picking up a pen to change modes, to save a mix, to undo a mix ... here [on the VR] even while the mix is running I can turn the automation off, undo and resolution

Have you got a favourite piece of outboard you can’t do without?

I find I use less and less outboard gear as I’ve gone on. When I first started engineering I probably plugged every single bit in! The Manley (Variable Mu) compressor over the mix, I’d struggle without that.

Manleys make a mix sound like those old records — when the vocal goes away — back comes the music, and I love that. You can tuck the vocal in the track a bit more. All my drums mostly go through a Neve compressor on a separate bus. Today I’m using a GML 8200 parametric EQ to give a little bit of a smiley face curve on the stereo bus. I used the Avalon EQ for a while and I also like the Manley Massive Passive, it’s lovely but quite coloured. The GML is fairly hard, it sharpens tracks up and makes everything sound more ‘like a record.’

I’ve noticed you don’t use a lot of reverb in your mixes. For example, the Manic Street Preacher’s Lifeblood is very full sounding, but there’s hardly any room on the instruments.

I do use reverb, but I don’t want to hear an ‘s’ going off into a long reverb. The Manic’s album was quite complicated because they made a point, well before I got involved, of not doing a lot of guitar on the record. That presented quite a challenge ... a guitar band without guitars! When I got the tracks they were very bare, I did quite a lot of extra work drum-wise and I played several synth and electronic string parts — they loved it and were almost pushing me to do more. I suppose like most people behind the glass, there’s a part of me that’s a frustrated artist.

Do you often program extra drums or play keyboards on your mixes?

I do have one bass drum I particularly like, it’s on a lot of records. You don’t hear it because there are other things going on in the track, but it just fulfils a certain requirement. Other times I might use loops ... if I was given a track to mix by a label and mixed exactly what they had given me ... I don’t think they’d be satisfied. I’ve done that and had the reaction — they expect more — The Manic’s album is definitely one where I went to town a bit. The song For the love of Richard Nixon was much more organic originally, I put a lot of drums in and some mad synthi sections after the chorus. I liked the track because it was about the Vietnam era, and I thought it was a really bold statement to have that song as the first single.

You’re also fond of quite radical panning, like on the Diefenbach’s Set And Drift.

Trevor Horn was a great one for either left, right or in the middle. I much prefer records that highlight four great parts rather than 14 mediocre ones. A record May/June 2006

craft I’ve just done from Hot Chip has incredible panning.

Panning can be a dynamic thing, although I haven’t quite got to the Beatles stage yet. I have another mixing rule about Pro Tools sessions: it has all got to fit on one screen. Even if I receive a session with 96 tracks, I will bounce until it is on one screen. It’s a mental thing, I need to see it on one screen, and mixing down the tracks is a great way of learning the song. I might get sessions with hundreds of tracks — the first five hours of the mix is on the Auratones — just figuring it all out. On an album project like Set & Drift with Diefenbach, I will often bring in my own Pro Tools rig, I’ll be mixing a track and an assistant can have the next track up, I’ll go over to listen for a few minutes and have them bounce the tracks.

I heard the Goldfrapp album involved rather an abundance of tracks ...

That record was very constructed. There are a lot of noises that are musical, but not a lot of music. It was put together painstakingly by Will Gregory and Alison Goldfrapp over a long period of time, almost jigsaw style.

When you have eight basses, what do you do, pan two left, two right and four in the middle?! They had had the chance to live with it ... the final mixed album was a composite of some of my mixes, some of their stems, some of Dave Bascombe’s stems ... it was incredibly laborious. It wasn’t perhaps the best technical approach to completely deconstruct it to give to me in Pro Tools, because they use Logic. But it worked out fine, the end result was a brilliant record, although there was indeed a certain amount of arithmetic involved.

It’s always hard to jump into a project that’s benefited from a very slow bake at the recording stage, isn’t it?

Before I stepped into the control room on that project, what they had already was brilliant. It’s that situation where you are being asked to do a job and you are trying to validate your contribution ... if I had never been involved it would still have been a very good record, just in certain places it just needed a little more width. I remember with Train I was pretty much left alone to go to town.

Do you prefer to mix without the artist or producer in the control room?

Quite often I will do a couple of mixes for an artist and they will say – great, do the album. Then they get involved ... and then it starts to become more complex. I’m quite militant when I’m mixing. To me a mix takes a day and a bit — for example on this session, people are coming in at one o’clock — I did most of the mix last night. I don’t want anyone around on the first day of my mix: to me, mixing is not a spectator sport, I don’t like someone sitting there watching me do what I do. I’ve had to explain that quite carefully to a few people ... it’s not me being rude, it’s just that I need to do my thing and it won’t make a lot of sense to you or me until quite close to the end, when it all comes together!

You mentioned working from stems — do you find that’s an increasing trend nowadays?

Sometimes I’ve heard my own stems on other people’s mixes. I’ve heard my vocal delays — it’s quite quick to get a track going around a stereo vocal acapella — all the effects and rides are done. But it’s not your own mix if you’ve only done that. I try to be reasonably ethical about that, if I’m sent a multitrack that has been mixed already and there are stems on disk, I won’t use them in my mix. I just feel that if someone has spent five hours going through vocals comping and de-essing it’s a bit unfair to take kudos 42

for their hard work — unless they are going to be credited, obviously. Some people I know don’t give their stems back on the disk, there’s rather a grey area at the moment relating to stems. Similarly if I do a lot of sound replacement and play several extra musical parts for a mix, do I give back the disk I was given, or the session that I’ve done with my work? I do give my own work and instrument tracks back, otherwise I would have gigabytes of music that doesn’t belong to me: there’s a legal title to be considered as well, I don’t know what the copyright implications might be.

Another Resolution interviewee, Joe Chiccarelli, commented that the increasing proliferation of sessions across disks means the true whereabouts of multitrack masters for many newer hit albums is probably completely unknown.

I think this is going to come back to haunt record labels in 10 to 15 years. It is quite resonant with the way we live now ... a sort of bullshit lifestyle. Once it’s been released and it’s in the shops and the label have made their money, nobody gives a shit about it anymore. It’s gone. We spent years being trained to archive properly at Sarm: look at the track sheet ... work of art, a good track sheet! But that’s not my job now, I give the record company back what I’m commissioned to do — the mix — but you figure as the music is their catalogue, their asset, labels would perhaps be looking a bit deeper into how to protect it.

quick turnaround on a mix, you can give me a song and in two days I can add production on it — I can get musicians in or do it myself. I don’t know if I would want to spend three months in a residential studio going through bass parts. I do like the high turnover and spontaneous nature of mixing.

So you’ve no ambition to be the next Rick Rubin?

I think there’s a place for all of us, for every type of recording professional. I get to do a bit of production on tracks, so I get to satisfy that side of me, but I absolutely adore finishing records. However, if the right project came along ... it would be difficult to turn it down! ■

A specialist mix engineer is generally thought of as several notches up from a plain recording engineer. Is this a step towards production for you, or is it an end in itself?

That’s a good question ... there is a part of me that is wary of going into full-on production because of the time demands involved. I don’t really do any recording any more, in fact I don’t even work with many producers anymore, I find that record companies book me when they need a fresh angle on a track. I don’t miss sitting around recording 85 tracks of vocals — it’s much easier for me now to complain about the state of the multitrack if I haven’t bothered to go and record it myself! Because of the resolution

May/June 2006

Success is a matter of adjusting one’s efforts to obstacles, and one’s abilities to a service needed by others.

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Alex Silva From muso, to assistant, to engineer, to producer/writer in a short space of time, Alex Silva has much to recommend him and a refreshing take on technology and production. GEORGE SHILLING meets him at his London den and talks about his Groenemeyer FIFA World Cup Anthem, among many other things.

I

NSPIRED BY IAN DURY, Alex Silva started playing in a band that got plenty of interest but ultimately ran aground. Meanwhile, he found a love of Portastudio recording and got a job at a studio in Cardiff later known as Soundspace. He was an assistant engineer for about three days, then the owner disappeared as his wife was about to give birth.

He left Alex in charge of a MIDI session, with a list of commands for C-Lab Creator. Silva began engineering bands, working fast on low-budget sessions, then worked with the Manic Street Preachers on their hit Theme From MASH. When he later bumped into Nicky Wire in a café, he pleaded for them to come and record their Holy Bible album at Soundspace, and they did!

Moving to London in 1995 he acquired management and found himself working as Dave 44

Stewart’s programmer. Stewart was working on the film Showgirls and Silva filled the programmer vacancy and in a short space of time found himself working with artists such as Mick Jagger, Paul McCartney and Marianne Faithfull. He was able to work with his engineering heroes Flood and Alan Moulder, and subsequently learnt about production from projects with Steve Osborne and Paul Oakenfold. In recent years, he has become Herbert Groenemeyer’s main collaborator and producer, and Groenemeyer’s 2002 album Mensch recently became the biggest selling album of all time in Germany, clocking up 3.7 million units. Resolution tracked Alex down at the room he shares with Groenemeyer upstairs at the Mayfair Studios complex in London. They have recently completed work on the 2006 FIFA World Cup Anthem (Celebrate The Day).

(Photos www.recordproduction.com) resolution

Why is Herbert Groenemeyer based here?

In German-speaking countries he’s a massive star, and it’s just the hassle you get from everyday life, not being able to be in a traffic jam without everyone gawking at you — he wanted a quieter life, so he came to London.

Funnily enough I partly live in Berlin, so it’s almost like we’re swapping places. There are some incredible studios, there’s an old East German radio station. It was built by some of the pioneers of modern acoustics, there are two studios the size of Abbey Road Studio 1, plus a jazz room, a chamber music room, a voiceover room — about six different rooms. The craftsmanship of the studio is phenomenal, so beautiful, with carvings, 20 metre ceilings, and acoustically they are just great. And they still have some great gear. There is a Bluethner piano with an extra set of resonating strings with a beautiful tone to it.

May/June 2006

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craft Are you knowledgeable about recording techniques?

I am, but we’ve reached this point now with record companies with budgets forcing artists to work by themselves, it’s the same thing as artists working with Logic — which I’ll never understand, because I think if you’re an artist you should work with Pro Tools because it’s ten times easier. If you’re using a computer and you’re trying to write music, it’s a disaster, because it’s two different parts of your brain.

If you’re producing a record you should sit down, read a newspaper and just listen. Get a good mood in the studio, put all your ideas in, but once a band is working, you should really disappear, fade into the background, don’t zone-in on anything. I actually play Tetris or something to distract myself, just so you’re not going, ooh, what’s that frequency on the hihat? Get an engineer to do that, they’re great things if you can have them!

Working with Herbert, are you staring at Pro Tools all the time?

Yeah, unfortunately! I’m finding less and less that I want to program. When we listen to eighties music it’s so rigid and covered in reverb, I think some of it was rubbish! I think the same will happen when we look back. Technology always leads popular music, and everyone ODs on it when it first becomes available. I was a programmer for years and I loved it, and was always sensitive to the song. Now I work with bands who want a more organic sound. Now, if you try and put a bit of programming in there, it can sound terrible, like a little paper sound in the middle of these lush 3D sounds.

Does Herbert have a band?

Yes, the last album, we did 80% of it here, and when it came to the band stuff we went to RAK Studio 1 and opened up both rooms, sat them in a circle, which they protested about — they all wanted to be in the control room, but I sat them together where they could all see the drummer. We got these Langevin personal mixers in, RAK is a bit limited with its headphone system, so we gave them each an eight-channel mixer to solve that problem, and once they settled down they all enjoyed the process.

Do you generally work with an engineer?

I work a lot with John Smith, he’s a wonderful engineer, great taste, and incredible knowledge about traditional recording.

46

Did you miss out on assistant training?

I didn’t really start until I was 27, I felt I was playing catch-up. In America assistants are much older, but I thought, I don’t want to be up at four in the morning making tea. Now I’ll experiment and say, what if I do this? Somebody else would say, you can’t do that, but because I didn’t learn that, I might try it — and get electrocuted or something! But at least I’d try it, and that’s when you get your eureka moments in the studio. The accidents are sometimes the best ways of achieving something.

I had a project where I had a harp player. It didn’t occur to me to have an ambient mic and close mic and crossfade. What I actually did was put a U47 on a skateboard with a piece of string and pulled it very slowly towards the harp over about 30 seconds, it sounded great. But putting up two mics and crossfading would have been a completely different thing! If you’re put in a room of people that just want to have some fun making an album, and you’re no wiser than them, then you just go with it. Some people are too stuck on the correct way, and I think that’s fine, but you have to find the right balance.

Are you a people person?

Yeah, I hope so. I learnt that with Dave Stewart, his real mastery is people. It doesn’t matter who walks in, what mood they’re in, they will be having a good time by the end of it. He’s just a nice guy, he doesn’t take himself too seriously, he likes to do the Eric Morecombe thing with his glasses, which is guaranteed to get anyone in a good mood.

How did you approach the 2006 FIFA World Cup Anthem?

When you think of the World Cup, it’s always Brazilian drums. Herbert has a high reputation in Germany, so we needed to be true to him, not just paste him onto a fiesta song. That was the fun part, to mix the African and Latin drums with a Teutonic writing style. Then we thought: ‘what about Celtic drumming, where does that meet African drumming?’, so we found something that works. It has the big moody opening, Herbert had the idea of big Russian, East European sounding choirs, but he’d been also raving about Amadou & Mariam, great African/French artists, we approached them and they said they’d love to do it, so it’s a soup of the two different styles.

Did you use Mayfair’s main studios for mixing?

Yes, we did for the FIFA track because it was huge — I always try and limit to 24 tracks, and even thought of suggesting to Digidesign a fun plug-in that only lets you create 16 tracks. resolution

So do you generally bounce down BVs to stereo?

Yeah, otherwise you never make a commitment. But that FIFA track it just wouldn’t stop expanding, and it ended up with 150 tracks, even with bounces.

We did the choir on about 24 tracks, and listened to it, and thought, it’s good, but let’s try the London Community Gospel Choir. We did about 20 tracks of them. Then it sounded too high quality, so we got our mates in and tracked them up six times. So it just mushrooms, to try and find the right feeling. Then at the end, we got kids on there!

Did you work at 96kHz? We did, but this track got too big. Up here we were always working with stems, but when we went downstairs to mix, we suddenly had to open it all back up and you can only have 96 tracks. So we exported the files at 44.1.

Did you co-write the FIFA anthem?

Yes, I co-wrote the lyrics. I’ve played on quite a few of the records I’ve produced, guitar, bass, usually when it’s a solo artist. I feel that the boundaries of people’s jobs in the studio are blurred at the edges. Sometimes the assistant will be the engineer, the engineer will make a production contribution and sometimes the producer will help to write a few elements of a song and maybe, if you’re lucky, the artist will make the tea! But writing credits can be a delicate subject and discussing that on a session would be a total vibe killer.

Did you have any formal musical training?

No, I play a lot of instruments enough to get things moving. If you’re doing good popular music, sometimes I think it’s more about cool little ideas.

I gather there was some shunting around of the FIFA track’s files across the Internet…

We’ve just ordered a DigiDelivery system because sending the FIFA media everywhichway was a real headache. There were so many companies involved as well as session folders for remixers that at times burning CDs and DVDs would eat half the day away.

Metropolis helped us out a few times by putting mixes on their server for clients to download, that was a big help. So now we’re getting a DigiDelivery for future projects — we’re also looking into watermarking software for the audio so we can keep track of it for protection.

What goes on in here?

We do everything, I like to try different combinations, and even if a song possibly needs a big sound, I also like to try it in here with a drummer, on a little Ludwig kit or something, just to hear what’s possible. It’s a funny room, because it’s not ideal, but it makes you work hard when mixing.

It seems quite ambient…

The bass is the tricky thing in here. I’ve mixed a few albums here, but it depends on the record. If you’ve got a dead sounding record and you know the sound of this room, it’s easier to interpret it, but if you’ve got a big acousticky album with a lot of room acoustics from other studios mingling with this one, it might get a bit lost.

So do you mix in-the-box in here?

We’ve got APIs. I was working with Ronald Prent at Galaxy, and he introduced me to Paul Wolff, so he told me ‘no, you can’t just come out of two outputs from Pro Tools’. So we come out with stereo drums, then high cymbal stuff — cymbals, guitars on the May/June 2006

craft next stereo pair, backing vocals, it’s all to do with gain structure. We’ve got three 8200s so 24 channels, then the 7800 with four full channel strips, then the 2500 compressor inserted across. It’s incredibly quiet, it sounds good. Obviously you don’t have fader control, which is the only downside, I guess it’s OK because it’s such a quiet unit. We use a Tascam US2400 fader controller. We’ve had other Digidesign products which do everything, but all you want are your mutes, solos and faders.

What about recalls?

I write it down, and record it with effects. That’s possibly the greatest thing about Pro Tools. It’s good and bad, because it’s so easy to have real total recall now. It also makes you think, I’m not going to bother using that Drawmer 1960 because I don’t have to write it down, I’ll use my BF76 instead. You’ve got to be very disciplined not to become a couch potato — I’m guilty, but I think it’s not very healthy. People are proud of being able to do it all in Pro Tools, but I don’t think it’s anything to be proud of — it’s like saying I eat ten curries a week!

Do you keep every single take?

Yeah, I think there’s a setting on Pro Tools called Destructive Record, but I can’t bring myself to do that.

We’ve always done that with tape, and yet now… It’s very strange. I think that technical limitation is great.

When we first got Pro Tools, we thought, 30 gig drive, blimey we’ll never have to change it. And now the FIFA song alone is about 12 gig, mostly because of all these outtakes.

And no-one ever listens back to those outtakes…

No, that’s the funny thing, because you don’t have enough time. Companies pay you less because they think you can make albums quicker, but you end up spending more time editing.

What do you think of record companies and downloading?

The value of music is fast going down the plughole.

With iTunes, no-one cares about artwork. It was a point where the band made a direct statement to the fans. The other thing with iTunes, all these people who make records don’t get any credits any more.

What’s next?

I’ve just done some tracks with James Dean Bradfield for his solo album, and I’m working on about four artists for Herbert’s Groenland label. I think working with these bands, the thing is to not get too involved.

That’s the hard thing sometimes, you work with some things that sound so good, and the thing is not to mess it up, just to get it into the right format, or make it easy to mix without destroying what’s great about it.

I think nowadays, if you’re working on your own in your bedroom, it’s good to have someone, just to look into their eyes and say, what do you think? I think that’s important, that’s what music is about, human interaction. It’s a bit sad, if you’re working on a track and it’s the greatest feeling, and the only thing you have to share it with is a mouse — it’s a bit crap!

What else?

I’m starting another album with Herbert imminently for release next year. I want to get more settled in Berlin. It’s going to be an exciting place for music. It’s the cheapest place in Europe. If you still want to play the squeezebox in your underpants, you can, and you can still live in a nice apartment and not worry about life. That’s where interesting things come from. ■ May/June 2006

resolution

47

business

China ramps up for multichannel China’s broadcasting community has many reasons to get into multichannel sound — in-home DVD is establishing a base expectation while the responsibility of hosting the Olympic Games at Beijing 2008 puts a heavy HDTV focus on the country’s infrastructure.

Most importantly its sound engineers have been getting involved and experimenting. ELITEN CHENG gives context to the progress.

E

NJOYING DVDS ON home theatre systems is popular entertainment in China, so people are familiar with surround sound and there is a Dolby authorised cinema in almost all cities and even some towns. Surround sound in the home started in the 1980s with Dolby Logic Pro encoded Laser Discs, and in 1997 Beijing Film Studio produced and released Opium War as its first 5.1 format film in mainland China. In broadcasting, however, even though radio has been broadcasting FM stereo since the 1970s, television has remained in mono and while a few thought about multichannel broadcasting in radio towards the end of the 1990s very little progress has been heard.

When Beijing won the bid to host the 2008 Olympic Games, high definition television became a hot topic in China because the Games must be in highdefinition television. That also meant multichannel surround sound, and China Central Television Station (CCTV) in Beijing and Shanghai Media Group in Shanghai started HDTV broadcasting with Dolby AC3 on their pay TV channels. Since the beginning of the year CCTV’s HDTV signal can be transmitted to 43 cities and towns but local TV stations have to supply related transmitting networks and subscribers have to buy their own HD set-tops or HD TV sets to watch the programmes.

There is a big difference in the development of HDTV between provinces. Advanced provinces and cities, such as Shandong TV, Tianjin TV, Guangdong TV, Qingdao TV and Zhejiang TV, have equipped with HD cameras and video systems but not all do surround. Shandong television has multichannel surround sound live production and music production equipment but Zhejiang TV and Qingdao TV use stereo for their HDTV. A few TV stations still buy analogue mixing consoles for their HD video and still

48

work in mono.

The Orient Movie Channel of Shanghai Media Group tried Dolby 5.1 TV broadcasting in June 2004, and formally started broadcasting in June 2005. At almost the same time, CCTV decided to use Dolby 5.1 for its HDTV pay-per-view channel. Therefore, CCTV’s HDTV set-top and TV set manufacturing partners have had to use the same sound system in their products.

The 2008 Olympic Games is not the only reason for an increased demand for surround sound broadcasting.

People have DVD players and home theatre systems

resolution

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Orchestra live recording in the HDTV studio at Shandong.

at home and enjoy surround sound movies and are predictably unimpressed with the mono TV sound they’ve been getting for five decades. At the same time, some cutting-edge practitioners in television and radio stations have begun studying multichannel surround sound recording and production techniques.

They are working in music recording, sports event reporting, entertainment live broadcasting and TV drama productions and they get their information from wherever they can.

There were some large-scale broadcast surround sound seminars in mainland China at the end of the 1990s with companies like AMS-Neve, Euphonix, SSL, Schoeps and DPA taking part but nothing answers your questions about how to do surround sound recording quite so well as trying it yourself.

In 2003, a Hong Kong-based publishing company released a Chinese version of the Surround Sound Recording Manual written by Sawaguchi Masaki and this book became the Surround Sound Recording Bible. Consequently engineers know about the Decca Tree, Fukata Tree and Hamasaki Tree and they know that surround sound recording can be used on almost any event and that it can be used for classical and popular music.

In 2003, Lou Wei of China TV Drama Production Centre replaced part of the background music of the original mono music TV drama Fruit Girls with 5.1 sound as the first example of surround sound TV drama in China. After this he completed a 2008 Olympic Games promotional video in Dolby AC-3.

Now he’s made another TV drama — Violin Solo — not only producing music in 5.1 but also creating all sound effects in surround.

In February 2004 at the Beijing Exhibition Theatre, Li Dakang of the Communication University of China and Zhen Zhao of the Beijing Film Academy did an experimental surround sound live production of Chinese national music. They compared some recording and postproduction techniques and burned audio DVDs in DTS and DVD-Audio formats separately. He Bangjing of Guangdong TV was the first to use Dolby E equipment for his 5.1 surround sound live concert recording with video. Ms Yang Qingqing of Shandong TV produced a documentary video — Understanding Deng Xiaoping — and picked many old recordings and processed them into 5.1. In October 2004, CCTV transmitted Jean-Michel Jarr’s Beijing Forbidden City Live Concert to France directly in HDTV with Dolby 5.1.

In October 2005, a HDTV studio was completed at Shandong Television station for live broadcasting and was equipped with a Euphonix System 5, 50

Dynaudioacoustics Air25 5.1 monitoring and a Fairlight Satellite. They also bought a Schoeps KFM360 sphere surround sound microphone and with this they produced various live performance programmes, a chamber concert and an orchestral concert. The Spring Concert 2006 covered a symphony orchestra concert shot in HD and with surround sound microphones and was the first large-scale recording in the Shandong TV studio.

Ms Qingqing and her colleagues used a Decca tree plus PZM, IRT, and Schoeps 360 sphere microphone along with more than 30 other microphones (see Figure). All microphone input signals were separately recorded on to tracks in the Fairlight DAW with a live mixdown stereo signal recorded on the video recorder.

While the broadcasting of the concert was in standard definition TV with mono sound, all the programme material was recorded in high definition video and multichannel audio and was later mixed down for surround with Dolby E and recorded to HD video.

The replacement of current standard TV with HDTV

completely is highly unlikely for some time in China — the HDTV transmitting and receiving equipment is expensive — so standard definition TV and HDTV will exist in parallel for a long time. Based on this reality, many TV stations buying HDTV equipment must think hard about compatibility but as the price difference between stereo and surround sound equipment decreases most of them will buy into surround sound audio systems directly. An influential aspect is the recording engineers’ personal knowledge and skills and whether they have had the opportunity to experiment with surround.

In China, the main TV stations clearly recognise that HDTV must be accompanied by multichannel sound and, because their main business is live broadcast, sports, entertainment, and location recordings will increasingly see the use of it. Surround sound production for HDTV at Shandong TV is a successful example in China and I believe more and more TV and radio stations will be developing their surround sound production techniques soon. ■

HDTV Control room at Shandong TV. resolution

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LSR4300 Series — explained and measured In the last 15 years the world of recording has changed dramatically but other than the addition of internal amplification, speakers haven’t changed all that much. JBL Professional’s PETER CHAIKIN puts forward the case for a smarter and better thought-out monitor that doesn’t have to cost a fortune.

W

HAT ARE THE MOST significant changes in recording? One of the most significant is that the majority of users are not working in professional control rooms any more. They’re working in rooms at home and industrial spaces.

These spaces don’t have the controlled acoustic performance of professional control rooms.

Most recording is now being done using computers.

In the move to computers we’ve given up a lot of the hardware we used to have. In the old days, we had the ability to listen to multiple sources, such as CD recorders and other record/playback devices. Today’s I-O interfaces may not have switching that allows monitoring of multiple sources. Some computer interfaces have no monitor volume control. Those that do, may have only a stereo analogue output.

What about surround and what about digital connectivity? Surround production is on the rise; but systems are difficult to configure, and the user may not have room for all those speakers. Working distances from speakers are routinely compromised.

And control of these systems requires outboard hardware.

At JBL, we looked at the market closely and discovered there is a real need for a technical update, a speaker for ‘modern recording’ and out of this came 52

the LSR4300 Series. The LSR4300 Series is comprised of three models: the LSR4326P and LSR4328P have 6-inch and 8-inch woofers respectively, and 1-inch silk dome tweeters. These systems have integrated 150 and 70W power amps for low and high frequency amplification. The LSR4312SP Powered Subwoofer is based on a 12-inch neodymium low frequency transducer and 450W of power. Resemblance to other

powered monitors ends here.

In addition to the use of digital signal processing, the system’s fundamental enabling technology is Harman HiQnet Protocol. With all the speakers networked using supplied Cat5 cable, the system can do some wonderful things and provide extraordinary functionality to address the points mentioned above.

In addition to simply being well designed and great-sounding loudspeakers, the LSR 4300 Series automatically apply correction to overcome acoustic problems in the mix environment. They integrate console ‘monitor section’ functionality including a system-wide volume control, input source selection, individual solo mute, all controlled right from the mix position using your computer workstation or supplied wireless remote control. It also allows configuration of networked surround systems with up to eight main speakers and two subwoofers.

The RMC Room Mode Correction System in the LSR4300 (JBL Professional’s method of EQing only the portion of the signal that excites the dominant room mode or boundary condition) can be invoked in the LSR4300 simply by pushing a button. With the RMC Calibration microphone positioned where you’ll be sitting (yes, the microphone comes with the system!), a deterministic calibration signal is emitted by each loudspeaker in turn, and each speaker is individually EQed to correct for the dominant room mode and its effect on response as measured at the microphone.

Figure 1. LSR4312SP subwoofer system response, before and after RMC calibration (on same grid). resolution

May/June 2006

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sweet spot

Figure 2. LS4328P studio monitor system response, before and after RMC calibration (on same grid).

At the same time, the distances of each loudspeaker from the calibration microphone position are individually calculated and the drive levels are adjusted so that inverse square level relationships at the listening position are equal to within a quarter of a decibel. Up to eight main channels and two subwoofers can be set up in this manner.

Let’s have a look at the automatic functions in operation. For these measurements, a single LSR4326P and a single LSR4312SP were set up in a small studio space at JBL. The subwoofer was on the floor, away from a corner, and the main channel was placed at ear height on a worksurface. The aim of the RMC system is to remove any low frequency peaks caused by the excitation of dominant first-order room modes for both systems. Considering first the LSR4312SP subwoofer, we show in Figure 1 the before and after curves using the automatic RMC function. (These curves were run using SMAARTLive

54

analysis software.)

From the computer screen we read the following summary of the action that took place: the frequency peak is at 38.3Hz; the Q of the peak is 3.87; and the attenuation is –4.5dB. This action can clearly be seen in the before and after curves in Figure 1.

Our next step was to look at the satellite system by itself. Because of the favourable placement of the satellite there was no objectionable peak, and the system ‘chose not’ to introduce any correction. This is shown in Figure 2.

Now, when we couple the subwoofer with a satellite, the situation will change a bit. First, we have a choice of crossover frequencies: 50Hz, 80Hz, and 120Hz. Setting this value determines the frequency ranges over which each element in the system searches for a dominant low frequency peak. Since the sub and satellite occupy different positions in the room, each

Figure 3. Subwoofer and one speaker system response, before and after RMC calibration (on same grid).

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May/June 2006

sweet spot one seeks out its own problem frequency.

To set a precise balance between sub and satellite, the satellite’s output will be analysed over the frequency range up to about 700Hz. This is well into the range where the system’s output is not influenced by room mode structure, and as such it gives the system a reference level for matching the equalised broadband outputs of both subwoofer and satellite. Each satellite is adjusted accordingly so that all reference levels are equal. The joint operation of these two systems is shown in Figure 3. Note that the reference levels for both sub and satellite are in exact agreement.

Is a computer required to make all of this work? The answer is no. All the foregoing functions, and others, are carried out within the loudspeaker electronics themselves. However, if you want a good picture of what is going on, you can connect your Windows or Mac workstation to the system’s USB port, boot up the supplied LSR4300 Control Center Software and get a nice set of visual cues and controls that are helpful during set up of projects.

For example, each loudspeaker has a built-in equaliser that can be applied for overall spectral adjustments in loudspeaker response to suit individual preference and acoustic conditions in the room. The computer lets you see the actual response curves that you have picked. The software allows you to select input sources (analogue and digital), adjust system gain, solo and mute individual speakers. Complete system configurations and set-ups can be stored on the computer and recalled.

Functions are duplicated on the Remote Control unit so you can remain centred in your seat without having to reach for controls at the loudspeakers.

Let’s turn now to the issues of monitoring in computer recording systems. Affordable computer

interfaces may not have a dedicated volume control for the monitor speakers. Likely, if included, the volume control only regulates a stereo analogue monitor feed and it is not possible to control the volume of a surround system.

LSR4300 provides a virtual monitor section, like the one included on a mixing console, that allows control of system volume, solo and mute of each speaker, and input source selection. These functions can be controlled right from the speaker’s front panel, the user’s computer using LSR4300 Control Center Software, or from a supplied wireless remote control.

Each speaker incorporates an analogue, SPDIF and AES-EBU input and the ability to switch between them.

The user can switch between the analogue inputs and two digital inputs, system-wide — so you can monitor the output of the recording system, a DVD player and the output of a multichannel encoder/decoder or processor. While soloing a channel on a workstation can be dangerous while recording or broadcasting, it turns out the speaker is an ideal place to perform certain functions. The LSR4300 monitoring system allows solo of any speaker without concern of interrupting a feed. Using LSR4300 Control Center Software we can even save complete system configurations for specific locations or projects. So we can return to these settings.

While there are dedicated speaker controllers on the market designed to provide these functions, integrating monitor section functionality into the speaker reduces cost for the user.

In addition to the capability of the system to

precisely adjust the levels of each loudspeaker in a surround system, the LSR4300 Series is designed to simplify other aspects of surround set up. The LSR4312SP Subwoofer incorporates features for 5.1 production. Once the system is interconnected using Cat5 cable, the user sets a switch on each speaker that identifies the speaker’s function in the system — Left, Right, Centre, Left Surround, Right Surround. Once this is done, the entire system can be controlled from any speaker, wireless remote and software.

Up to eight speakers and two subwoofers can be controlled and soloed on the network without the need for an external controller.

When using digital inputs, each speaker knows which channel of the digital stream, A or B, to monitor. So the left speaker automatically plays channel A of a stereo digital input, and the right speaker automatically plays channel B of the stereo digital input. Finally, since space may be at a premium, the speakers can be wall-mounted using industry-standard mounting hardware.

All of this functionality would be meaningless if the speakers didn’t provide exceptional acoustical performance in the first place. During the development stages of the product we devoted considerable time listening to a broad range of programme material, and measuring system performance to make sure we were right on the mark.

We believe the LSR4300 system is something of a breakthrough in the combination of high performance and high functionality -— all of this at a price point comparable to that of previous systems. ■

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katz’s column

How to optimise levels in an analogue processing chain Part II Having set down some ground rules in the last issue BOB KATZ embarks upon the in-depth side of getting the most from your sound chain so it’s running sweet and right. accumulates distortion, so running everything up to the max may not be a good thing if you are looking for a clean analogue chain. But if you are looking to get that ‘max’ sound, either through saturated tubes, solid state, or an overloaded A-DC, next issue I’ll give some suggestions on how to obtain that sound from your processing chain in a controlled, repeatable manner.

Let’s get down to the setting of basic analogue levels. Figure 1 shows a simple D-A-D processing loop, with a D-AC connected to a compressor in

Fig. 1. D-A-D Processing Loop

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AST ISSUE I HELPED some of you get off the voltmeter-phobic wagon. To further you on that journey, just think of a voltmeter as a way of measuring the output of an analogue processor that doesn’t have a meter of its own, or for doublechecking the accuracy of the internal meter (usually a VU). For example, a classic Urei LA-4 compressor has a VU meter that can be selected between Output and Gain Reduction. Put it in the Output position and the meter measures the voltage level of the output in decibels. 0 VU on the LA-4 is the same as 1.23 volts (+4dBu). Put a test tone into your analogue gear, connect the external voltmeter, and prove that to yourself.

series with an equaliser, and returning to an A-DC. But before we build a complex chain, we must first calibrate the levels of the D-AC and the A-DC to each other.

Figure 2 shows a D-AC connected directly to an A-DC with two Y cords feeding your monitor and the AC voltmeter. Use the monitor to make sure you have

Fig. 2. D into A for Measuring

Good audio-certified voltmeters do not have to cost a lot. Here’s a picture from my test bench of another classic, a Ballantine True RMS AC Voltmeter I bought surplus for US$35, attached to a Hewlett Packard attenuator/balanced bridge I picked up for $10. It can measure signals from 300 volts down to the noise floor in microvolts.

If you read between the lines of last issue’s column, you’ll remember that I advocate a lower analogue voltage level for 0 VU (where to calibrate -20dBFS on the digital side). This is because a lot of cheap gear simply does not have good headroom. So, running that LA-4’s meter up to 0 VU may not be the solution when it is used in conjunction with some lesser equipment; or consider that your whole chain May/June 2006

a clean signal and no bad connections. Feed a 1kHz -20dBFS sine wave digital test tone from your DAW to the D-AC and adjust its output to 0dBu (0.775 volts). This conservative level will guarantee your DAC or the rest of your chain will never run ‘ratty’ on peaks and it is not so low that it will cause an audible decrease in signal-to-noise ratio.

Next, adjust the input gains of the A-DC while reading an accurate digital meter on its return to also read exactly -20dBFS (within 0.1dB). This makes the system have unity gain, and it also sets known and repeatable values for the analogue side of your chain.

Most DAW digital meters do not provide accurate readouts to 0.1dB, but some plug-ins do, including the Inspector from Elemental Audio, the PAZ from Waves, and the excellent Digicheck utility that comes with RME interfaces.

SETTING LEVELS WITH MUSIC — Now repatch your analogue chain, for example like Figure 1.

The best way to set levels for music at this point is to know the unity gain settings of each piece of analogue gear and start at those positions. When in doubt, get out the Y-cord and voltmeter and find the unity gain points on each processor’s controls and mark them with a piece of tape. resolution

At unity gain in each processor, and with a -20dBFS = 0dBu level, the VU meters of any piece may never come up to 0 with normal musical signal.

That’s a good sign if you like to have a clean analogue processing chain. Signal-to-noise ratio should still be excellent with this level calibration. It’s a myth that you must be constantly concerned about signal-to-noise ratio, ‘maxing’ each piece’s levels or especially the level of the A-DC. Just operate the active controls of the compressor or equaliser while occasionally checking the peak level on the A-DC. Let the A-DC’s levels fall where they may; there is no such thing as ‘too low’ because you’ve adjusted your analogue system’s gain to be noise-free for any normal monitor gain.

Of course, a significantly low level would force you to turn up your monitor gain, but if you follow the Max Peak Between -10 and -3dBFS rule, all will be well.

It’s highly unlikely you’re A-DC will overload unless you are really pushing the equaliser, and if so, then attenuate the output gain of the equaliser below its taped unity mark until the A-DC is once again running below full scale. Or, if the equaliser has no output gain, attenuate the digital level within the DAW feeding the analogue chain. The noise and distortion of a 24-bit processing system is significantly below the noise floor of your analogue chain, including the convertors, so this practice will not cause any audible degradation; dither the DAW’s output to 24 bits to be safe.

There’s so much headroom in this calibration that an over is highly unlikely. If you do hear hiss at the listening chair at normal monitor gain with no signal, then it’s probably caused by a defective or improperly-adjusted analogue processor. Disconnect the input to the A-DC and make sure the hiss goes away to isolate the problem to the analogue chain.

In my system, I can raise the monitor gain by 10dB above my standard 0 cal and barely hear noise from the loudspeakers.

OPERATING THE ANALOGUE COMPRESSOR MAKEUP GAIN — The compressor will have an output or makeup gain control. Common practice is to adjust the makeup gain so the compressor does not change loudness switching between bypass and engaged (in) while listening to music. But this is not a requirement; as I mentioned, it is not necessary to max out the A-DC. Use the makeup gain only when you need to evaluate the sound of the compressor.

In Part III next issue, we’ll talk about how to purposely get distortion from the chain. Now why would people want to do that? And in the last installment, Part IV, we’ll discuss the nitty-gritty of interfacing balanced and unbalanced analogue connections and their many permutations. ■

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

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know how

DAWmagedon — Best practices for building your own PC Most of us attempt it at some stage of our lives and some even get quite good at it, but building a PC for audio is not a task to be taken lightly. CHARLIE BOSWELL, director of Digital Media and Entertainment for AMD, says it’s all about the processor, balance and low latency.

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HEN WAS THE last time you were inspired to create some music? What was the first thing you reached for to harvest your muse? Guitar? Piano? A computer? The PC has become a critical part of studios and people are recording and producing without going into high dollar commercial rooms. The computer tuned for this task has been dubbed the Digital Audio Workstation and the availability of affordable computer hardware and software has brought this within the reach of most.

Some technology companies reckon that musicians will be every bit as naïve and long suffering as general consumers when it comes to using a computer and be resigned to re-booting the machine when something fails. An artist cannot tolerate flakey technology...I certainly won’t. For me, technology must behave like that guitar and piano. I must be able to perform with it rather than work or labour over it. The technology must allow me stay in the creative moment with my tempo unbroken rendering my performance as ‘live’.

But this does not happen by accident and putting DAW software on any home PC does not guarantee your success.

Inspiration arrives in our minds as a single thought and this leads artists to take action to express it. This single human thought is the most delicate quantum in the universe and is truly unknowable. Brute force technology as put forth by a multitude of computer companies rarely considers the difference between moving from productivity applications, like word 58

processing, spreadsheets and slide presentations, to the world of the creative.

For those undaunted by the challenge of building their own DAW, today’s choices bear good news and bad news. The good news is that the number of available components and software for building your own DAW has increased exponentially. The bad news is that more choice often leads to confusion and misinformation. For some, the lack of a definitive solution can be undesirable, but clearly people who want to create, record and mix music on their computers prefer options. Freedom of choice is part of the creative spirit that underpins the music production industry.

The spirit of AMD’s approach is that the technology must be in the service of unleashing the creativity of the user. It is no longer acceptable to be handcuffed by low overhead, clumsy workarounds, or restrictive platforms. Guided by this spirit, we set out to design a system architecture around a new processor technology called AMD64 that is designed to help remove traditional technology barriers that have hindered the creative process in the past.

What are the priority considerations for putting together a DAW? It helps to understand that digital audio is the most unforgiving of all media applications.

This is true because in a sampled data application like audio, every sample is critical. Dropping a single sample among millions will result in an audible pop or click that will kill a project or a great take. Also, it resolution

goes beyond merely recording original data. Adding effects, plug-ins, multiple tracks, higher resolution and mixing various elements together requires balance in your PC platform architecture.

To achieve balance, the relationship between the processor, I-O and memory are critical factors. The bigger the workload, the more important balance becomes. Put another way, the bigger the workload, the easier it is to find the limitations of a poorly designed system architecture. Latency is the result of poor system architecture where resources are forced to compete with each other over the same set of wires, like that found in a traditional front side bus PC architecture. For optimal performance, reliability and sonic integrity, look for the system architecture not with the highest processor MHz rating, but with the lowest system latency.

AMD64 technology helps achieve state-of-theart system balance and low latency while reducing the bottleneck found in a traditional front side bus that media creators have long suffered with. AMD’s design is called Direct Connect Architecture whereby processors are directly connected to memory on independent wires, processors are directly connected to I-O on independent wires and processors are directly connected to other processors on independent wires. When resources don’t have to compete with each other, system balance, low latency and loading levels with confidence can follow.

Currently, this is best represented by Dual Core AMD Opteron processors in a 2-processor configuration, which is unmatched in terms of performance on nearly every type of benchmark measuring 2P system performance in nearly all applications. Dual core essentially means two engines in the same engine block, so two Dual Core AMD Opteron processors in tandem will perform like four.

Adding greatly to processing demand in today’s digital audio are software plug-ins. As the plug-in revolution continues to ramp up, audio media creators are often unaware that plug-ins are computationally bound and need to be threaded. This means to use plug-ins effectively and seamlessly, you need a system that will scale.

Make sure you use licensed software for the backup and support; it takes time but it’s worth it. If you do a plug-in update give yourself time before you book a client to make sure it’s up and running. Make sure you have storage backups with a defined system together to do that.

An increasing reliance on plug-ins requires lots of dynamic range in your computational arithmetic or you may be forced to throw data away, dither, or round off, in any case resulting in deteriorated sound.

This brings us back to low latency, as other types of architecture that don’t have balance and don’t load level well can cause you to drop samples. To avoid this you need double precision floating point and a processing platform that is powerful and scalable to help you stay in the creative moment without fear.

This is important because it helps you avoid the standard fixed point arithmetic DSP approach, which can exacerbate the arithmetic problems. Further, the May/June 2006

know how

floating point, multiprocessor approach enables software hosting that is in the future of audio. A good example of this is Nuendo or Cubase.

In the end, you’ll be disappointed if you don’t understand what’s under the hood; so when you pick a motherboard with AMD64 Direct Connect Architecture you get the fundamental building block for a robust and scalable DAW. On motherboards, we mention the Tyan 2885 or, for advanced graphics capabilities, the Tyan 2895. Remember, you can take a 2P motherboard like this and expand it to four processing cores by using Dual Core AMD Opteron processors.

So you’ve chosen a great motherboard, now each choice you make from this point on can have an impact not only on the quality of the audio signal, but on the ability of the system to operate properly over time. A standard 4U rack is the preferred chassis because the added space inside the 4U form factor makes a better fit for sound cards. For cool and quiet operation, we mention the Zalman CNPS 7000B-AICU heatsink and fan solution. You can use an Antec True Power 550 for a power supply. For DDR SDRAM memory, we selected 4Gb ECC Corsair memory. Please note that memory must be ECC registered for use with AMD Opteron processors in DAW applications. You often hear of people going through troubleshooting procedures of swapping memory sticks to correct audible problems and it’s often because they have not chosen the correct memory in the beginning.

Hard disk drives, too, can be a source of problems if not chosen and configured properly. We believe the best practice involves using a separate drive for system software and then a series of disk drives for audio writing only.

We’ve found Western Digital or Seagate to be very good brands that should be configured in the following way.

For OS, software (this is also where the digital audio applications are installed along with all the plug-ins): 1HDD x 80Gb, 7200rpm or 1HDD x 36Gb, 10,000rpm For writing audio only (the speed needs to be at least 7200rpm but we recommend 10,000rpm): 4HDDs x 250Gb SATA array for 1Tb RAID 0 or 500Gb RAID 1/0.

Note: both Tyan motherboards have onboard SATA controllers.

For FireWire(1394), a common I-O approach is to use Presonus Firepod, RME Fireface 800 or the MOTU 896HD. (For PCI or PCI-X based audio cards) Anybody putting together a high quality system like this wants easy access to I-O. We recommend RME Hammerfall or Lynx for the audio cards.

In choosing graphics cards, it depends on which of the two motherboards you choose. You need AGP graphics for the Tyan 2885 and PCI Express graphics for the Tyan 2895. ATI, Matrox or Nvidia will fit the bill.

A popular operating system for this hardware/software configuration is Windows XP Pro, but note that there are certain switches to be turned on or off to optimise the system for digital audio use. Also, with a Windows platform, make sure your drivers are updated. Don’t wait to do this in the middle of a project. Have the most current ones and have back up.

Other things you’ll want are DVD R/W, which can be any one you like, and of course video monitors. Dual monitor displays make it easy to see what you are tracking and that’s what it’s all about: what you are tracking. That and having the most robust, balanced system with the best architecture to set your creativity free. Remember, it’s those little things that can break or limit aesthetic or break your muse...pick your architecture wisely. ■ 60

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getting the most from MONITORING — Accurate and colourless monitoring is essential. Too often monitoring is chosen for its ability to handle certain types of music but for audio restoration and transfer, the cleanest, least coloured sound is required. If you cannot hear exactly what is coming off the record or tape, you stand little chance of being able to make a successful re-recording. You need a system that will show up the flaws, so that you can decide what to correct, and what to not. My own favourites are Professional Monitor Company transmission line loudspeakers because they are so neutral, but there are others, including several by ATC and Dynaudio. Good power amplifiers are of course, a sine qua non, as is an uncoloured, but perhaps not anechoic acoustic environment. In this regard, it is interesting to observe how large shelves, full of cardsleeved records and NAB spools can do an efficient job of breaking up standing waves and absorbing other acoustic problems. But, as Tom Lehrer said, I digress.

Also under this heading, I put level monitoring and spectrum analysis. Accurate digital level monitoring is essential. Noise frequently has very high but short term energy levels so it is easy to overload the A-D. If you stick to a peak of around -3dBfs with a one sample setting, measuring the output of the A-D convertor, you should have no trouble. Incidentally, the various articles in Resolution on level measurement should be compulsory reading for every engineer.

Spectrum analysis is so useful in that you can easily visualise where the overall response may be deficient or where there are unnatural peaks or resonances, and thus home-in and correct more accurately and more quickly.

For many, LCD screen-based measuring tools, such the RTW Portamonitors, are ideal, in that they combine level measurement and spectrum analysis in one unit.

Restoration It can be a long and convoluted process, a frustrating exercise against the clock, or a pleasure of rediscovery, but restoration of audio is something we all get involved with in some way. ROGER BEARDSLEY does it every day and puts forward the case for treatment and how to get the best from the experience.

A

T THE TIME OF WRITING, there is more than 100 years of sound recording available to us. During that time, methods, formats and inevitably quality of recording, have changed.

Increasingly engineers are being asked to transfer earlier material to modern digital formats, and to do whatever is necessary to present these historic recordings in acceptable sound. Acceptability, of course, depends upon what is wanted as the end product. On the one hand you have the archive that wishes to preserve recordings in as accurate a sound as possible — warts and all — and on the other you have the ‘Music & Memories’ type of operation that is issuing what is cheap, out of copyright material of the famous names of the distant and more recent past.

The former will not want any interference whereas the latter wants the sound to be as modern as possible, with no noise, possibly artificial stereo, and added reverberation.

In between these two extremes, there is a path that the responsible engineer will tread. Using the latest and best possible equipment, they will be asked to transfer these sometimes fragile, often unique 62

recordings using today’s technology to produce a result where processing, including noise reduction, is only used to a point where it does not interfere with the musical integrity of the original.

How this should be achieved depends upon the type of recording you are asked to transfer. For the purposes of this article I shall be considering three principal types: analogue tape; mono vinyl LP; and 78rpm disc.

There are some basic parameters common to all these formats and I shall examine these first. Also, I’m assuming knowledge, or at least awareness of, various standards of record and replay equalisation, proper level measurement and monitoring in the analogue and digital domains, and a sound general knowledge of music in its many forms. A good ear is de rigeur.

The basic chain should consist of: analogue source with appropriate equalisation; A-D conversion; noise reduction; digital processing tools; editing/recording; and monitoring. It’s important to appreciate that while most studios will have some of this equipment, the needs of audio restoration can be different from those of the normal studio. resolution

A-D CONVERSION — Next to consider is the AD convertor. A good rule is that the better the A-D convertor, the more accurate is the signal for noise reduction systems to deal with. In turn you will be able to obtain greater levels of noise reduction without audible degradation than with a poor A-D. If you can afford a Prism or similar then you’ll find it a wonderful investment. Smaller budgets should look at the Apogee, which is very fine, and Benchmark seems to be making quite a name for clean conversion.

A-D conversion should occur as soon as the signal leaves the format’s equalised source. That will ensure that the noise reduction system gets its best chance of recognising and dealing with noise.

NOISE REDUCTION — Older finished product or master formats all have noise of one type or another and this may be tape hiss, crackle, clicks, hum, or buzzes. For the professional, the first port of call has to be CEDAR. Despite its price and relative age (in terms of concept), it is still the only system for serious consideration. How you get into it depends upon your preferences. Essentially there are two methods.

Standalone, whereby you purchase the modules such as De-Click, De-Crackle and De-Hiss and run them in real time between you’re A-D convertor and the processing or editing system, have the advantage of flexibility and ease of use. The alternative is to go for the full computer-based systems, perhaps as part of a package with a Sadie editor. This approach gives the maximum ability to restore and clean up damaged recordings in a way that the standalone modules do not. The cost is greater, so it may well depend upon the sort of material you will be working on and how often, but if it requires heavy intervention then the full system is the one for you.

May/June 2006

getting the most from Individual preferences also play a part; I personally prefer the standalone, but that is probably because I do not use a computer-based editing system and I like to be able to configure my system in a variety of different ways, depending upon needs. Either way, it’s not cheap.

One thing to always remember is that any noise reduction system, even CEDAR, has limits. If you go too far with any of the processes you will damage the signal. Over-use of De-Click can produce a distortion akin to growling and it will catch on the attack of brass instruments. De-Hiss when used inappropriately can give some very strange bubbling or watery effects.

Then there are the ice-skaters who will turn up for a bit of practice! So keep checking between ‘before’ and ‘after’ to make sure that the lowered noise benefits are not being compromised by unpleasant effects, or artefacts like Torvill and Dean.

PROCESSING TOOLS — This refers to the ability to make adjustments to the sound once it has been denoised. You can forget all those wonderful analogue processors that make the sound so ‘you’, with restoration you are not the painter you are the picture restorer — you clean away as much dirt (noise) as possible without affecting the image and allow the original to shine through. A chain of analogue processors will cloud the image.

You need high, and low pass filters, shelving at both ends of the spectrum, variable Q notching and a parametric function. These could be standalone units like the Z-Systems Z-q2, in-built facilities in the digital mixer or part of the computer-based processing package that you use — or a combination of all three. Indeed, it is useful to have them all but as with noise reduction, less is often more and it is recognising when you have gone too far with the process that is important. Providing your source has its correct replay equalisation, laterstage EQ should only be for shaping to iron out any deficiencies in the response envelope, plus gentle filtering above the usable frequency response. If you find yourself adjusting by more than a few dB either way, think again and ask yourself why.

Tight notching is necessary for the removal of problems like hum, heterodyne whistles and other types of steady-state narrow-frequency interference.

There are CEDAR modules that will work on hum and buzzes if you prefer to do it that way, but they can cause problems with some types of material.

replay them, and one will not cover all tape formats.

I am ignoring multitrack tapes on half inch or higher since they are likely to be recent and easily handled.

The most useful machines come from the Studer range. They will handle virtually all professional quarter-inch tapes in full track or stereo/2-track format.

The A80 is an obvious choice, but a good B62, or B67 can be just as effective. You will need one with the option of IEC or NAB replay characteristics. A Nagra IV in one of its various guises can be extremely useful too, especially for the most delicate of tapes. If you can get a QGB 10.5-inch reel adaptor so much the better.

A well maintained machine goes without saying and if you cannot service it yourself then get a specialist to do it. Badly adjusted or faulty tape machines can ruin precious tapes as well as give LRX_cricketadvert2.qxd 7/4/06 11:38 AM poor results.

DOMESTIC AND 4-TRACK — For ‘4-track’ domestic quality tapes running at 3 3/4ips or 7 1/2ips, a Revox will do the job admirably. There are plenty of refurbished machines on the market.

CASSETTE — Now to swear: ‘cassette’. There, I’ve said it! Actually, many cassettes are really not that bad. A first rate machine such as the Studer A721 will work wonders and with judicious tweaking of the EQ, it is amazing what you can get from them. I have mastered a few CDs from cassette sources and had very good reviews. The A721 will extract the last ounce out of them and sometimes that’s all there is to work on.

LINE UP — When did you last line up and set Page 1 up an analogue recorder? Have you ever done the

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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.

EDITING AND RECORDING — Providing you can edit at waveform level and do all the usual tricks, then you should have no difficulty with what ever system you use although you may need to manually edit out some types of damage, such as deep scratches on discs, or drop outs on tape. You have to make sure that the format conversion between the digital audio output of the replay system and the computer is of the highest quality, but that should be included in your studio system anyway.

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.

SOURCES (EQUALISED)

PROFESSIONAL TAPE — Probably the format you’ll come across most often. You will need machines to

Contact your nearest SADiE dealer or main office and visit our website for further details and a brochure.

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May/June 2006

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DIGITAL PRECISION

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getting the most from

EMT 950 turntable with added digital speed readout driven by the tacho output.

job? Thankfully for restoration you only have to line up the replay chain. Ideally you should have NAB and IEC test tapes for the speeds that you are likely to encounter. Don’t forget to clean the heads and demagnetise them (and guides) before using the tape.

AZIMUTH — What you will not know is whether the tape to be transferred was recorded on a machine that had been properly lined up for azimuth. The simplest way to check is to play the tape and then adjust the replay head azimuth a fraction one way, then the other and note the effect on the perceived HF. Adjust for highest HF level — that’s not scientific, but with a little experimentation you’ll soon get to know how to do it. Of course you can/should do it with the proper test kit but don’t forget to re-line up with a test tape after making alterations, or leave a note for the next user! CEDAR does make an azimuth correction module and it’s very useful too.

TAPE HANDLING — Old tapes are likely to have a variety of problems. Those made of acetate are very brittle and can snap easily. Mylar and other

similar ones don’t snap, but they stretch. Never wind/ rewind tapes at high speed; treat tape gently. Before attempting to transfer, wind and rewind at least once and examine any edits as these can become unsafe.

Remake any edits that look to be in poor condition.

If you are presented with tapes, such as Ampex 456, that have gone sticky consult experts. The Vidipax. com site has very useful information.

CASSETTES — Again, wind and rewind before using. No edits, of course, but you may find the pressure pads are loose or even detached. Either put the tape in a new cassette shell (fiddly) or glue the pressure pad back onto the metal strip where it should be, pulling out a loop of tape first.

Compensation for Dolby pre-emphasis can be problematic, especially Dolby B. I usually prefer to play without Dolby compensation, then digitally equalise to get the correct response.

MONO VINYL LP — Microgroove LP began in the US in 1948, followed in the UK by Decca in late 1949, and finally EMI in 1952. Early mono LPs were not recorded to the standard RIAA curve and there

is such variety as to make a sane engineer have nightmares. Your turntable should have variable speed — 33 1/3rpm was the ideal, not always the actual speed. Many of the first LPs were re-recorded from 16-inch lacquer masters, so there is the added problem of (sometimes) incorrect speed of original replay, on top of the two layers of replay equalisation inherent in that system. Speed may vary across the record, and it is simpler to adjust in real-time while the disc is playing than to do it later at the digital processing stage.

The Technics 1200 is ideal for the purpose unless you have the budget to go for an EMT, such as the 948 or 950 on which you will need to bypass the built-in RIAA preamps when necessary. A multicurve equalising preamp is essential and about the cheapest professional unit is the Elberg from Denmark. However if you are really serious, the Ted Kendall famed ‘Front End’ (also known as Mousetrap) is the only choice.

With fully variable lo/mid/hi equalisation using the correct time constants, it is enthusiastically used by many engineers and archives. The Elberg and Front End are ideal for 78s as well as LPs. You will need more than a standard LP stylus

Studer A721 cassette machine. If it’s on the tape, the 721 will recover it.

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May/June 2006

getting the most from degradation. Even if you cannot hear it, CEDAR will notice it!

Learn. Learn about the history and techniques of the formats you transfer. If you know how the signal got there in the first place, you will have more chance of recovering it successfully.

Listen. You really do have to listen intently.

When that happy day arrives when you are sure that you have finally learned it all, you can either retire immediately or wait for the job that will surely arrive that makes you realise you know less than you thought. In my experience, it usually comes the day after. ■

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I have been working with early recordings on disc and tape for more than 30 years and I have learned my trade by experience. What that has done, is to reiterate some fundamentals.

Restore. Do not change or create. You should be opening up a window on a past performance.

Keep it clean. Think about signal path integrity.

From the tape head and tape (or disc and stylus) right through to the point of audition, make sure that there are no unnecessary barriers that might degrade the signal.

Compare. You have an A/B function — use it when applying EQ and noise reduction.

Experiment. Never assume that the settings that should suit a particular recording are enough.

Less is more. With noise reduction, better small amounts of steady state noise, than horrid artefacts: the ear soon adjusts to slight background noise and ignores it. Only use the equalisation that is strictly necessary.

May/June 2006

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78RPM DISCS — Many of the same parameters apply as for LP except that the styli will be different. Here you need a range from 0.0020-inch to 0.0035-inch and the correct stylus will again minimise noise. Speed variation is even more of a problem and although many discs do play at around 78, a range of 65 to 85 is useful. Replay equalisation will usually be different, especially at the top-end, because apart from late 78s, no treble pre-emphasis was applied on recording. So a flat or slightly rolled off curve will be the starting point.

The mid will generally require less boost.

For a suitable turntable, and similar to the Technics, Skytronic has a model with the necessary speeds and variations.

Convert. Get into the digital domain as soon as possible and do not revert to analogue until listening.

Analogue processors (including those that go A-D — process — D-A) in a chain will introduce signal

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so get a few headshell/cartridge/stylus sets. Expert Stylus Company are the people. Stylus sizes from standard 0.0006-inch up to 0.0015-inch will be very useful for minimising noise and maximising signal recovery because groove sizes and shapes vary.

The Shure range, such as the SC35 (obtainable from Canford Audio), are perfect for early mono LPs and will take a higher playing weight — often necessary for discs that have been played with less than perfect original pick-ups.

Experiment when playing back. Choose an EQ curve that is audibly the nearest, then tweak if necessary at the processing stage. Don’t worry about which curve you think or are told a record might be.

Use your ears. As with all mono discs, including 78s, always mono the two channels to reduce rumble and other effects. The Elberg and Front End enable you to do that.

Even if a record appears clean, there will be dirt and grime in the grooves. If you play the dirt surface rather than the groove surface, a muzzy and probably noisy signal will result. Buy a disc cleaning machine like the Nitty-Gritty or Moth. If you have the budget, the Keith Monks is still the choice. Never play a record unless it has been cleaned. At the very least, you will help to grind in the existing dirt, adding to the noise.

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65

meet your maker

Rupert Neve The man who helped define the industry and set the parameters for the appreciation of performance and quality talks shop.

H

E’S ONE OF THE most well-known names in professional audio and has become strongly associated with all that is great and good about sound. His is a legacy that extends back to cover many of the defining quality moments in audio. He started the Neve companies in 1961 and while many would have hung up their soldering iron following the acquisition of the operation in 1985 by Siemens, Rupert Neve in his new life-after-Neve actually went on to do some of his most memorable work. There was Focusrite with the fabulous Forté console and that era-defining outboard range; consultancy with Amek that resulted in the incredible 9098 console (with Graham Langley) and a range of outboard (anyone remember that first product the Medici?) and contribution to other Amek desks. He moved to the US in 1994, consulted for Summit Audio (Element 78), Taylor Guitars (instrument preamp) and launched Rupert Neve Designs last year with its Portico range of modular outboard.

All this from a man who is 80 and has two greatgrandchildren. He was born in Devon, UK, lived his early years in Buenos Aires, and served in the Royal Signals during the War. After the War he ran a PA and disc recording business and designed and built his own equipment. Desk design started with Musique Concrete composer Desmond Lesley with the breakthrough coming with placements at London’s Recorded Sound and Phillips Records. The rest, as they say, really is history.

What is special about Rupert Neve Designs products?

Two parts, I think, to this answer: the first is simplicity.

Rupert Neve Designs Inc. products are based on the experience I gained in the early days — not only in 66

ZENON SCHOEPE terms of sonic quality but also in reliability. One of the most powerful keys to reliability is a low component count — there is less to go wrong! Circuits had to be simple.

Simplicity of the basic designs is the first part of the answer. From my earliest years I was taught that an Engineer is a man who can do for Tuppence what any fool can do for Sixpence. This involves understanding what really needs to be done. What does your client — the person who pays the bills — actually ‘need’ as opposed to ‘want’? How could I meet his needs in the most cost-effective way? This does not necessarily mean it’s cheap!

It was both an engineering and an artistic challenge 40 years ago and it is still so. We are constantly looking for ways in which we can place powerful tools into the hands of the artist — you, the client — that will give you creative fulfilment.

My early designs addressed the issues of specialised EQ; of getting the signal from source to destination without loss of quality, without adding noise or distortion and continuing to deliver the highest quality signals reliably under extremes of working conditions, such as temperature and humidity.

Transistors were just making their appearance.

They tended to be unreliable and suffered from huge spread of performance figures. If they got hot they could all-too-easily enter a thermal runaway mode and blow up! They were noisy and, when used in some of the manufacturer’s published designs, they produced their own peculiar sound (crossover distortion), some of the reviewers even hailed this as the new, ‘accurate’ sound of the future!

The second part of this answer is, I think, innovation.

I’ve spent many years designing consoles for the great names of our industry who do know what they need and are able to evaluate sonic quality to greater depth than any test gear can measure. The music recording industry has developed sophisticated and cost-effective methods of controlling, editing, storing, marketing and reproducing sound, much of which, has been achieved at the expense of quality.

Talented people have been buying low priced gear with inadequate performance, thinking that this was the norm and are often left wondering why their recordings lack depth and perspective.

In spite of this basic loss of ‘musicality’ in the digital domain, we have been able to reveal and enhance hidden depths that lie embedded within a recording.

When it’s processed carefully in analogue, not only professional artists and producers but any appreciative listener, not necessarily knowing how to express audio ‘needs’ in terms of technical specs, can hear and enjoy the difference!

With such an extended backcatalogue, how do you think up new products?

As an enthusiast for more than 65 years and in the fortunate position that my hobby is my business. The ideas often spring from something I’ve wanted for myself or have arisen from excitedly sharing ideas with ‘golden-eared’ friends.

I’m always searching for ways to make something resolution

sound better — perfection is a huge field; there’s no ‘ultimate’. Most designers would admit that the great difficulty is knowing where to stop. You have to work to an agreed spec and when that is met, your design is complete.

I’m convinced that the field of sound is infinite in its complexity. There’s always more waiting to be discovered than we ever dream of. Created by an Infinite Creator, we want to achieve perfection, compete in the race, climb the mountain, paint the picture; the motivation of creativity, the ecstasy of accomplishment, is always there.

So we press on ever onward and upward, seeking fulfilment in sound as our art form and our mission but knowing that what is good today will be better tomorrow!

How do you feel about your name still being associated with products you have nothing to do with?

Astonishment and disgust that humans who are gifted with their share of creative talent should want to misuse their abilities and abandon any concept of integrity by copying the designs of others!

The only people who have any rights to my original designs are AMS-Neve who acquired them under my original sale of the Neve Group of Companies in the mid 1970s. There are several manufacturers who have, under the un-watchful eyes of the recent ownership of the company I founded, been left free to make ‘recreations’, knock-offs and un-inspired products that claim some connection with my name to obtain credibility.

I should say that a few of my original staff are still working with AMS-Neve and I have been happy to support them and the Company with technical information as, under new ownership, they start now to reissue some of the originals and track down the violators.

It’s an unfortunate fact of life that down the ages in any field of human endeavour, be it literature, art, music or finance, there have been those who prefer to copy, plagiarise or commit acts of forgery.

Does digital draw you or do you remain an analogue man?

For many years we’ve been told about wonderful digital quality. It’s a wonderful medium certainly, for storing, editing and delivering sound but resolution is greatly inferior to analogue — a bit like comparing film photography, where resolution is probably limited mainly by lens quality, with pixels where more is always better but expensive (and slow if you are trying to receive an email!)

However, digital is here to stay and seems to get away with ever lower resolution. But I’ve heard nice sounding recordings that have been up-sampled from CDs. We have found analogue ways of counteracting some of the worst sounding digital recordings. For example, by restoring sweetness and apparent resolution. Some of you have seen (or heard!) this in the Portico Range.

The more serious impact of low grade sound reproduction is that, without a real Point of Reference, the internal data bank you build up when you go to concerts, listen to real instruments and voices, is missing and you just don’t know how to distinguish between good and bad! At least two generations have grown up without a perspective because their only access to music, either as entertainment or as an art form is via the ubiquitous Compact Disc.

As long as 30 years ago, famous ‘golden-eared’ friends demonstrated that they could perceive the effect of frequency response anomalies as high as May/June 2006

meet your maker and even beyond 50kHz. Research has shown that when we hear reproduced sound that either lacks musical frequencies we expect to find there or that introduces artefacts that ought not to be there, the brain radiates electrical activity of the sort associated with frustration and even anger. This electrical activity can be measured. Many questions arise as to whether such sound can do us lasting damage or can result in unexpected patterns of behaviour. It’s a vast subject and not one to be argued here. So the answer to your question is ‘I don’t want to be guilty of harming the world: I’m an analogue guy!’

Have you changed your mind about any design principles over the years?

Yes, but more about how to achieve my objective than what to achieve!

I entered the music recording field in the early 1960s with fear and trembling. Would my designs be good enough? Flat frequency response to 20kHz and reliability of transistor circuits were my first concerns. After all these were modular amplifiers that would be cascaded to form a mixing console and any weaknesses would be cumulative. Transistors were pretty new and had a bad reputation. Famous companies had delivered audio gear that simply went into thermal runaway when they got too hot!

As consoles became more complex we had to pack more and more circuitry behind the panel. Amplifier efficiency became important as we reduced size. We designed push-pull amplifiers that were no longer Class A. Oh yes, not every console with my name on it embodied the sacred ‘Class A’. But they still sounded good!

There came a point when we realised that the magical octave above 20kHz was as important as that below it. Extended frequency response became the aim, together with better noise and always with very low Total Harmonic Distortion. That meant new transformers. A lot of work went into designing these and we achieved a full response to over 200kHz. More features were wanted with the consequent need to use Integrated Circuits to save space.

Later, when as Consultant to Amek, I produced the Pure Path range of outboard modules that had incredibly low distortion and noise and with excellent calibration accuracy, they could be considered in almost every way, as about as good as it could get. There are many users who love them but, strangely enough, I was never myself enthusiastic about the way they sound. It would take more space than is feasible here to discuss the design details.

Continuing the upward path, I am fortunate here in Wimberley Texas, to have Billy Stull as a neighbour (www.legendaryaudio.com). Billy runs a riverside mastering studio in the most picturesque setting in this lovely Hill Country village. He has the most amazing gift of sound perception that I have ever come across. We worked together on mastering gear — the object mainly, is to make great music recorded digitally, sound sweet and lovely. Between us we produced the Masterpiece, a substantial piece of rackmounted, very accurate equalisers, compressors, phasing circuits and other effects which go a long way towards achieving our goal! Rupert Neve Designs started trading last year and the Portico range of modular amplifiers based on what I learned working with Billy, is the first new embodiment of simple circuits that sound better.

Yes, there have been circuit design changes but not really a change of principle!

May/June 2006

What were the significant technological advances over the years that changed what you were able to build?

Having got somewhat carried away by the last question, I think the answer to this one has largely been covered.

However, all my designs are characterised by the use of transformers. In my early days I was fortunate to work with engineers and designers of the ‘Old School’ who taught me how to design high performance transformers ranging from very small ones for microphones and hearing aids to output transformers handling several kilowatts of audio power. It is essential in any audio design, to exclude any unwanted signal. Not only noise and crosstalk, but Radio Frequencies, hum and its harmonics and associated spikes. Transformers maintain total isolation, providing the ideal way to prevent ground loops, by which such artefacts are often introduced.

Together with a completely closed ferrous case they do a vital job in excluding those unbelievably microscopic interfering elements that the experienced human ear can perceive.

Significant technological changes? Solid State devices: transistors in the early days and later ICs that enabled amazing packaging of high performance audio circuits.

Improvements in transformer steel have been a great help in making better and more affordable transformers. Availability of very high performance switched mode power supplies has enabled us to improve isolation between modular amplifiers.

Why no valves?

Valves? What a good idea! We’re working towards them again, strangely enough! The more desirable qualities of the big triodes are being achieved with semiconductors. Valves suffered from short life, microphony, inefficiency, and, well, the list is well known. It would be fun to calculate the heat and power that would be generated if valves were used in a module of the Portico range.

The modularity of the RND products permits interconnection to create a ‘desk’ but are you tempted to build a console?

As we all know, the monster recording console is, sadly, a product of the past. Yes, I am often tempted to build a console. However, the console of the future will be a very different beast that allows much greater flexibility, freedom from massive and costly redundancy, and provides for monitoring of the several present formats as well as future ones.

The Portico range will shortly include simple mixing units and these will be followed by the first of a new breed that I can’t say too much about at present. However, it won’t be long! Bear in mind that within our first year six new products will have hit the production lines!

Which of all your designs are you most proud of?

I think the famous Montserrat console I designed for George Martin would qualify together with the Amek 9098 designed on similar principles. But please ask the question again in about 18 month’s time!

Most people would have stopped designing audio equipment by your age, what keeps you going?

What else would I do? There are things that I want to do not directly related to pro audio — training for the Media, for example. With a magnificent team of young engineers, and colleagues, and the support of my wife, Evelyn who has put up with my wild dreams for 55 years, I am working towards it. ■ resolution

technology

High-Definition Multimedia Interface It’s the interconnect that sorts out a lot of problems we thought we had and a few that we didn’t; it’s the High-Definition Multimedia Interface. NIGEL JOPSON discovers a new digital connector that has all sorts of implications for the future — including the forced deactivation of analogue outputs.

O

N THE LAST DAY OF MARCH, Toshiba put on sale the world’s very first HD-DVD player, the HD-XA1. HD-DVD and Blu-ray are the two new formats set to replace the DVD for high-definition home movie viewing and will vie with each other for consumer spending-money in the marketplace. The arrival of players has been eagerly anticipated and trailered by home-entertainment

pundits and vendors for quite some time. However, new software included on both Blu-ray and HD-DVD releases will automatically slash the image quality, making it only marginally better than current DVDs, unless consumers have a new connector and cable called HDMI to hook up players to their screens.

Only one in 20 HD-TVs sold to early adopters has the appropriate version of this connector. Perhaps

just 15% of new sets sold this year will include it and be able to deliver the full 1080 high resolution picture that these devices are being sold on. Unlike the disparate disc formats that have polarised manufacturers into two camps, the HDMI connector has cross-industry support. The founders include electronics manufacturers Hitachi, Matsushita Electric (Panasonic), Philips, Sony, Thomson (RCA), Toshiba, and Silicon Image. Digital Content Protection LLC (a subsidiary of Intel) is providing High-bandwidth Digital Content Protection (HDCP) for HDMI. In addition, HDMI has the very partisan support of major movie producers Fox, Universal, Warner Bros and Disney, and system operators DirecTV and EchoStar (Dish Network) as well as CableLabs and Samsung.

So HDMI is an industry-supported, uncompressed, all-digital audio/video interface. The small connector provides the interface between any compatible digital audio/video source, such as a set-top box, DVD player or AV receiver and a compatible digital audio and/or video monitor. It was conceived as a sort of digital SCART, primarily a point-to-point connector; HDMI grew out of DVI and the picture side is backwards compatible (providing that copy protection is implemented). HDMI adds support for component video, multichannel audio, the so-called universal CD control, and the concept of auto-configuration. The payload on the new cable includes 24-bit video pixels and sync, at a possible 4 different rates up to 165MHz (or 330MHz for type B), information frames for video that inform the receiving device about the resolution and aspect ratio of the picture, and colour information, such as gamma inventory, so the receiving device can do a good job of rendering.

HDMI is, coincidentally, the highest specification single-cable digital audio connection currently available, with v1.3 of the standard adding support for 8 channels of 192kHz audio, native DSD and all the Dolby True HD formats. It’s streaming audio, a vast improvement over packet-based computer protocols, but even so the digital audio is likely to arrive with a lot of jitter in it. As might be expected of a standard designed with the TV metaphor in mind, video is the master clock and the audio clock is derived from it via PLLs (phase locked loops). To transmit audio, a

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technology blank video screen must be output to maintain sync and for an audio manufacturer to acquire a ‘license’ to use the connector. From an audiophile point of view, putting all the very high speed digital silicon inside equipment seems slightly undesirable. However, the audio support in version 1.3 is probably good news for HD player manufacturers. Currently, player makers have no choice but to fit all the decoders and codecs for different protocols inside their machines. In future the consumer can anticipate cheaper and simpler players, with all the multichannel audio output as digital data.

On the actual HDMI wire there is one clock channel and 6 data channels (in the type B specification). The clock design is non-coherent: a 165MHz or 330MHz clock down a 15m twisted cable is hardly conducive to keeping in time, the twist of the cable means that internal cable lengths will differ quite significantly. So when the signal arrives at a destination the clock is retimed for each of the data wires, indeed the clock itself runs much slower than the pixel rate on the wire, which can be up to 10 times faster.

Digital audio is sent at the beginning of every horizontal video line, and there is additional descriptive audio data sent concerning how to recover the audio clock for the PLLs and the type of encoding. Other information frames sent to the receiving piece of equipment carry data explaining pixel encoding, colour, aspect ratio and whether the picture is overscanned. Pixel encodings supported are RGB 4:4:4, YCbCr 4:4:4 (8 bits per component) and YCbCr 4:2:2 (12 bits per component). There is support for legacy-resolution VGA plus progressive formats 720 and HDTV 720p and 1089. Frame rates officially supported include 50, 59.94 and 60Hz, although backwards compatible 48 and 72 can also be used.

The new interface supports interlaced and standard definition by using double clocking. Even quadruple clocking may be used as well — perhaps for situations where audio bandwidth is very high compared to the video — maybe a concert recording from some future Mobile Fidelity Sound Lab-type operation. The audio formats are the areas where there has been most change in upgrading the HDMI standard, as the 1.0 version only supported audio up to 48kHz and the compressed formats found on DVD. But HDMI is far, far more than a neat little digital connector. It is a Compliance Program. HDMI mandates the use of AACS (Advanced Access Content System) via the HDCP license, which is a form of DRM (Digital Rights Management) and for which a fee is payable. HDMI devices have a special 5 volt signal wire that sends something known as a Hotplug event: think of it as similar to a computer USB port’s plug-and-play behaviour. Every time a Hotplug event is detected, something called the EDID (Extended Display Identification Data) is read in each device and the format of information to be sent is agreed between the two devices.

This has all sorts of potentially useful consequences, such as the display being able to adjust itself to match the format of an incoming picture and the audio reproduction chain being able to set itself for the correct sample rate and surround format. The HDMI link is copy protected based on an authorisation code, and the contents are encrypted on the cable. Each device has 40 56-bit keys, the source and the sync generate a shared secret that is changed every 4 seconds (just in case some mischievous consumer pulls the cable out and connects to a recording device), so as the content is travelling the key is changing. The HDCP protection has the goal of preventing transmission of non-encrypted content, with the key-changing data encryption preventing any form of eavesdropping on May/June 2006

the digital content stream.

DVDs also benefited from software protection, the content-scrambling DeCSS system. Resolution readers may recall this protection was famously cracked in a very short time by Jon Lech Johansen (DVD Jon).

His teenage success was not quite as dramatic as first believed, however, as it later turned out that Real Networks had ‘forgotten’ to encrypt a CSS key in one of its software players, and Jon was simply recovering content using the compromised key. Bearing this embarrassing snafu in mind, Intel has built a key revocation procedure into the new HDCP protection system, which means that devices manufactured by any vendors who violate the license agreement, or any spoof-keys used by future ‘HD-Jons’, can be easily blocked from retrieving HD content. How? Well, HD products (the disc bought by consumers,

resolution

for example, or possibly a live HD stream ... from a satellite ... from broadband) may contain a list of revoked keys. So your newly purchased piece of content will just switch off your expensive home

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theatre system if it thinks any part of it might have been too naughty.

The handshaking between HDMI devices, and the AACS/HDCP protection, has significant implications for system installers and facility designers. The 1080 picture can be sustained for runs of 10m, but to go 60m some form of equaliser is required. Clearly, 10m

is a pathetic length for an installer fitting cable into a wall chase. Extending the authentication process using repeater devices currently looks like a bit of a minefield for designers. With two syncs, audio and video, the authentication process can become quite complex. Every time you flip programmes there will be a Hotplug ‘event’.

If you want the audio and video to be rendered by different devices — for example the audio has to be returned to a separate device such as a 6-channel surround amplifier — then the audio sync has to read the video capability of the video sync, patch it into its EDID and make a new EDID, then the source can read the composite EDID, authenticate it and decide what picture and what audio to send. With the new Toshiba HD-XA1 taking over a minute to switch on, and 28 seconds for the ‘loading’ display to change to ‘HD-DVD’ after putting in a disc, it’s not looking good for larger and more complex installations.

Part of the HDMI program includes signing up to the rather sadly named ‘Analog Sunset’ clauses.

According to the AACS Interim Adopter Agreement of 15th February 2006 (Exhibit F - Compliance rules paragraph 1.7.1): ‘2010 Analog Sunset.

Existing Models (as defined in Section 1.7.2) may be manufactured and sold by Adopter up until

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22/12/05 4:46:32 pm

May/June 2006

technology a new digital display interface standard, approved this month, put forward by the Video Electronics Standards Association (VESA). It defines a new digital audio/video interconnect, intended to be used mainly between a computer and its display, or a computer and a home-theatre system. DisplayPort connections support 1 to 4 data pairs on a link that also carries audio and clock signals, with transfer rate of 1.62 or 2.7 gigabits per second. The video signal provides an 8- or 10-bit pixel format per colour channel.

DisplayPort includes DPCP (DisplayPort Content Protection) from Philips, which (surprise!) uses the 128-bit Advanced Encryption Standard with session key establishment plus authentication and an independent revocation system. Displayport is being Quested_S8_Resolution_ad_5-05.qxd 4/5/05 driven by the silicon makers and computer companies,

December 31st, 2011. For any Licensed Player (other than Existing Models) manufactured after December 31, 2010, analog outputs for Decrypted AACS Content shall be limited to SD Interlace Modes only (i.e., Composite, S-Video, 480i component.) 576i component will be addressed in the applicable Final Agreements.’

What price, then, the estimated 5 million nonHDMI compliant HDTV displays sold in the USA? If the analogue sunset is implemented on time, it will also mean that extremely expensive Digital Light Processing projectors — more than capable of displaying 720p and 1080i — will not work with the new players. Perhaps all consumers need to do is purchase a player before the sunset period?

Only with the following proviso: that their keys are not revoked or downgraded by future content purchases. Their so-called ‘Robust Inactive Product’ will need to be ‘activated through a Periodic Update.’

A ‘Periodic Update’ is defined as ‘ ... provision of or replacement of a Device Key Set, via ... download of updated software.’

For those of us using HDMI screens in a professional setting there are some other concerns: so far, I’ve been unable to accurately determine if time-to-render will be correctly reported by the EDID handshaking. Any sort of motion compensation will clearly change the delay and will have to be reported back accurately in order for screens using this connector to be used in a situation where lip-sync is important.

For those who may be thinking ‘they will just play it on their PCs’, I’d like to point out that Microsoft and Hollywood’s aim of a digital-rights-holy-grail moved a step closer when Intel embedded DRM into its latest dual-core Pentium D processor and accompanying 945 chipset last year. And there’s another connection standard on the horizon: DisplayPort. DisplayPort is May/June 2006

who would love to use one twisted pair between the lid and body of a laptop rather than the currently required 30-60 wires. One of the design hurdles with HDMI is the quantity of wires between chips.

I have yet to meet a consumer in possession of the equipment necessary to make a digital copy of a DVD, never mind an HD-DVD. End-to-end DRM has moral and possibly economic ramifications for our industry, as equipment manufacturers are required to push high-definition content through a progressively more tortuous and obtrusive path, just to prevent casual copying. But with the increasing practice of using consumer equipment — especially displays — in a professional environment, this policy of treating the consumer as a potential criminal also has the potential to create some very severe roadblocks in 2:13 pm facilities.

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ten

Practical installation tips You’re not a professional wireman but you can handle a soldering iron and fancy making some changes in the studio.

KEITH SPENCER-ALLEN offers ten tips, most of which you won’t find in the available literature.

NON-LEAD SOLDER — You may not have noticed but the formulation of solder has changed and is now lead-free (under EU regulations for the removal of harmful materials), having a greater percentage of tin. If your soldering experience has been with lead solder you need to get some lead-free practice before beginning a major project. The melting point is higher and so you may need a larger wattage iron to do the same job. With more heat being applied you may have to consider the use of heat sinks. It doesn’t flow as well as lead-based, the joint takes longer to set, and when it does, there isn’t the smooth and shiny appearance — it’s looks more like a bad lead solder joint. You’ll need to ensure that whatever you are soldering is suitably tinned or of a correct material. And stock up on some lead-solder for those crucial jobs.

RACKING — You’re mounting outboard gear into a standard 19-inch rack and about halfway up the rack it stops fitting easily. All rack mounting equipment is supposed to conform to precise dimensions given in IEC and EIA standards but some makers have not looked carefully enough and have produced gear with front panels exactly 1.75 inches (1U) or multiples of it when they should be undersized by approximately 0.8mm to allow easy fitting. Try laying the rack on its back so that the gear hangs vertically by its ‘ears’. This allows the gear to be positioned before fitting any mounting bolts and it should be possible to find those precious few fractions of a millimetre to fit them all in. Some MIorientated gear has panel mounting holes very slightly off position and occasionally this can be accommodated by fitting the rack cage nuts so they grip the rack holes vertically rather than horizontally, allowing more play.

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WHEN YOU GET TIRED — Long repetitive tasks can cause loss of concentration (I don’t think it’s just me). An easy mistake to make is wiring XLR pins 1 and 3 reversed when moving from male to female panel sockets, but starting to wire the rear of a panel from the wrong end or row runs close (Depends how far you get along the row before you realise. Ed).

Wiring a 92-pin EDAC socket using a diagram of the male plug from the connection side is also a challenge when tired. When working on the rear of a panel make temporary labels and notes about the wiring that show what you should be looking at. Don’t try to transpose plans in your head — draw the pins connection for the EDAC socket as a guide. It’s quicker that having to remake your errors.

JACK PLUGS — In most studio situations 1/4inch jack plugs are only to be found at the rear of rackmount gear, plugged in and removed only when the gear is taken out of the rack, once in every few years. While you can use low cost types with plastic shells for this, should there be a need to handle them any more frequently you need a professional design that incorporates strain relief and has wiring tabs that allow reliable cable termination. Professional types have wider bodies with more space internally so that, in the case of a mono plug, it is easy to connect or remove the screen connection independently of the signal wire, without melting the wire insulation, or destroying the plug. resolution

REUSING OLD WIRE — Cable can be expensive and there are attractions in reusing old lengths but approach this with care. The obvious weak point is corrosion at the termination. Worst are the very fine (Litz) stranded cores where atmospheric action can extend oxidation of the cores up the cable under the cable coating, up to 12-inches in one example I’ve found. This all needs to be stripped back to fresh copper before using. If the cable has been positioned in a particular shape with sharp bends for many years, it may be internally damaged at that point when straightened. Externally the cable coat will have become stiffer and more brittle, while internally the insulation around the cores may not have the resilience to heat shrinkage of modern designs, something that becomes more significant with the higher temperatures of lead-free solder.

COLOUR CODES — Colour is a useful guide to which cable core is what but what do the colours tell us? There is an IEC standard that defines the colours used to identify signal pairs up to 48. For multicores of greater capacity than that manufacturers will list their colour suggestions — and it works. However, under this standard, the colour pairing for a twin-core cable would be white and blue. While some follow this, many don’t so you need to find a recognisable method for deciding which will be the hot core when confronted with red/blue, white/black, etc. I tend to work on the principle of the lighter colour being used for the hot signal. Do be aware of the American mains power cable colour code — there are still a lot of older US-built products in use with their original power cords that I frequently find wrongly wired in the mains plug. Correct is black/live; white/neutral; and green/earth.

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KEEP THEM TYPES SEPARATE — Have some respect for your cables and what they are carrying.

Keep cables with different ‘wire content’ apart — so low level signals away from high(er) level and mains power (and timecode). It doesn’t take more than a few inches to ensure adequate separation, and should it be necessary for cable content types to cross, ensure that they do this at right angles. It’s recognised good practice and it is worth noting that a really quiet installation — one where it is difficult to tell if a system is on or off — comes from paying attention to all the minor points.

DON’T MIX YOUR SOLDERS — One of the difficulties that we will face in the future is the problems that occur when you try to mix lead and lead-free solders. Repairing a lead solder joint will require the use of lead solder. Even if you are able to remove most of the lead solder, the objects to be soldered will still be tinned with lead solder and difficult to work with leadfree solder. For the foreseeable future we’ll need both solder types to hand, and the ability to distinguish between them.

XLR CHOICE — There are about a dozen makers of branded XLR-type connectors worldwide, all of which do the same job but there are key differences. XLR designs fall into two camps — those that use screws to grip the incoming cable and hold the plug together and those that have a rear strain relief that screws into the plug body bringing pressure of a plastic insert to grip the cable. In general the latter type is quicker to assemble, can take larger diameter cables and is very reliable. The screw-based designs can sometimes take a wider variety of cable cores, are a little better with thinner cables, and being more metal may have more resistance to being crushed (if that is an issue). Before committing to several hundred, check out a variety of types with your cable of choice. Most importantly, check that your chosen type fits all the sockets it will need to — some mics that have an XLR positioned within the body of the mic are notoriously fussy.

DISCONNECTION FROM GROUND — Never disconnect the mains earth from a piece of outboard gear — it’s there for safety. If there are hums due to an earth loop (two routes to earth) most problems can be sorted by the wiring in the plug that takes the unit’s output. First try disconnecting the screen from the output lead at the unit end if it is still connected — this is true for unbalanced or balanced outputs.

Try the manufacturers for further info particularly if the output is electronically balanced. But leave those mains earths attached. ■

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May/June 2006

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slaying dragons

john watkinson

Dithering about Asked to give an explanation of dither, JOHN WATKINSON replied that he probably would, but then again he might not.

D

IGITAL AUDIO IS today the norm, to the extent that it’s almost unnecessary to qualify it. We might as well agree that it should simply be called audio unless we specify that it’s analogue. The advantage of digital audio is that it’s just another form of data and can therefore be processed, stored, randomly accessed and delivered using IT. Such systems can deliver audio data with no corruption, allowing the highest quality, although the modern idiom is to use lossy compression systems that render digital audio inferior to the best analogue practice. Audio data only differ from generic data in that there is an implicit timebase in the way the data must be presented.

However, in order to reap the benefits of digital audio, the original analogue audio has to be converted.

More importantly, the quality of the audio is limited once and for all by the quality of that first conversion, so it had better be good.

Conversion relies on two orthogonal processes: sampling and quantising. I’m not dealing with sampling ‘The perfect quantiser causes low level here; maybe at another time if my arm is twisted. distortion because the quantising error Samples consist of binary numbers and therefore is a deterministic function of the input there is a fixed number of values possible, depending signal. The essence of dithering is that on the wordlength. The theoretically continuous the quantising error is decorrelated voltage range of an analogue signal is divided up from the input and made random.’ into intervals, where each interval corresponds to one chicken.qxd 3/5/06 5:54 PM Page 1 combination of the binary number.

There is a quality advantage to be gained if these intervals get smaller HE BEST SOUND IN THE WORLD closer to zero volts because we can describe low level Come and see us at AES stand 2234 signals more accurately without running out of The Dream ADA-8XR: range for large signals.

• Flexible 8 and 16 channel modular Unfortunately, all we can configuration do with the resultant non• The XR has direct connection to linear quantised data is Pro ToolsIHD or Mix, AES, DSD, convert it back to analogue Firewire and SPDIF again at some distant • The XR works at sample rates You wouldn’t compromise your location. Because of the from 32k - 192k performance, so why non-linear relationship • The XR features the Overkiller between the analogue compromise on your converter. function a +18dBu Soft Limiter voltage and the binary code, it is impossible to control For more information visit gain by multiplying sample www.prismsound.com values and impossible to mix by adding sample values.

Consequently all audio production equipment uses uniform quantising, in which each interval represents exactly the same analogue voltage increment.

Changing the least significant bit (LSB) of a PRISM MEDIA PRODUCTS INC. PRISM MEDIA PRODUCTS LIMITED sample changes the output USA UK of a D-AC by one quantising Tel : (973) 983 9577 Tel : +44 1223 424 988 Fax : (973) 983 9588 Fax : + 44 1223 425 023 interval. A quantising Email: sales@prismsound.com http://www.prismsound.com interval is measured in Volts and as we proceed it will be NOTE: Digidesign do not endorse or support the Prism Sound ADA-8XR unit. Not all Pro-Tools controls or configurations are supported by the Prism Sound ADA-8XR. Prism Sound reserve the right to amend their own product specifications without notice. Digidesign, Pro Tools I HD are trademarks of Digidesign a division of Avid Technology Inc. clear that the whereabouts of the analogue input

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voltage with respect to the quantising interval is very important. Nevertheless many treatments use the term LSB to mean a quantising interval, even though one is a binary digit that can only have two values and the other is a continuum. Even worse is when locations within the continuum of a quantising interval are described as some fraction of an LSB. This is an oxymoron and makes about as much sense as being three-quarters pregnant.

In a perfect quantiser, the analogue voltage can fall anywhere within the quantising interval, but the same code emerges. The precise location of the voltage within the interval is lost. From that code an ideal D-AC will always output a voltage in the middle of the interval. The D-AC output differs from the actual input by an amount that varies between zero and plus/minus half an interval. This is quantising error. The same sloppy writers who use fractions of an LSB usually call it quantising noise.

In audio we divide corruptions of the waveform into noise and distortion. Distortion is an unwanted signal that is a function of the wanted signal, whereas noise is an unwanted signal that is not a function of the wanted signal. In other words, to be classified as noise, quantising error must first be proven to be decorrelated from the signal.

Unfortunately, using nothing more sophisticated than a piece of graph paper, anyone can draw an arbitrary waveform and then proceed to derive from it another waveform representing the quantising error. As an extreme, but valid, example, if the input waveform is so low in amplitude that it stays within the same quantising interval, the output is silence and the quantising error is the input signal. So much for being noise.

In a perfect quantiser, the error is distortion and the distortion frequencies fold about the sampling rate and re-enter the audio band as anharmonics.

Nevertheless most treatments of quantising then proceed to work out a signal to noise ratio of 6.02n +1.76 dB where n is the number of bits. This is wrong twice. Once because the quantising error is not noise and twice because perfect quantisers are unsuitable for audio: when we modify a perfect quantiser using dither to decorrelate the quantising error, we will incur a noise penalty in doing so that will make the above equation give an optimistic result.

The perfect quantiser causes low level distortion because the quantising error is a deterministic function of the input signal. The essence of dithering is that the quantising error is decorrelated from the input and made random.

In a perfect quantiser, the intervals are fixed and the same input waveform always gives the same output data and the same quantising error. However, imagine a quantiser in which all of the intervals can be moved up and down by an external signal.

If that signal is random, successive samples would find the quantising intervals in different places and the quantising errors would then develop a random element. If the amplitude of the random signal is adequate, the quantising error decorrelates from the signal and we have produced a linear quantiser having a noise floor.

To see this properly requires some understanding May/June 2006

showcase

of statistics. One of the things that can be shown with statistics is that most people don’t understand statistics. The relationship between smoking and early death is clear, but whenever the topic is raised, some hi-fi enthusiast will cite the example of their uncle Fred who has smoked 60 a day and is 90 years old thereby disproving the theory. Actually it does nothing of the sort. Any probabilistic phenomenon requires a small number of cases that are unaffected.

A simple random number generator, such as a twisted ring counter, produces a finite set of numbers and each one is equally likely, whereas outside the set, the probability is zero. Thus the probability function is rectangular. Now imagine two different random generators whose outputs are added together.

We have to combine the two rectangular probability functions in an appropriate way. They have to be convolved. We slide one rectangle over the other and plot the area of overlap as a function of displacement.

The result is a triangle. If the process is continued, so that the number of rectangular generators rises, the result is that the triangle starts to round off at the peak and to grow tails at the ends. Eventually we reach the familiar Gaussian curve that describes the probability of systems having large populations. Uncle Fred and his ashtray live on one of the tails. Most of his contemporary fellow smokers are no longer with us.

Noise in analogue electronics results from countless processes and so is Gaussian. Thus if we want to dither a convertor we are looking to produce something like a Gaussian noise floor.

The same is true if we need to shorten or truncate the wordlength of digital audio samples. Simply ripping off the low order bits we don’t want is as bad as quantising without dither in the first place. Instead we need to create a new, higher noise floor so that the shorter wordlength system remains linear. In fact the process is no more than a digital simulation of an ADC. A low level noise in the digital domain is added to the original data, and each sample is then rounded up or down according to whether the bits to be lost put the level above or below half way up the new larger quantising intervals.

The type of noise required is interesting. If a simple, single, random number generator is used, the re-quantisation is linear, but the noise floor rises and falls with the exact level of the signal, a phenomenon known as noise modulation. However, if two such generators are used, with outputs added, the resultant triangular probability noise works perfectly. This result was first shown by John Vanderkooy and Stanley Lipshitz, who have done a great deal to put digital audio on a scientific footing.

Interestingly enough, dither in quantisers has a forerunner in mechanical computation. Computers based on gearing become non-linear and inaccurate at very low speeds because of stick-slip, a phenomenon in which the force required to get something moving is greater than the force needed to keep it moving. It was found that by applying vibration to the mechanism, the linearity was restored. Mechanical bomb aiming computers were linearised by the vibration from the engines in the same way that early un-dithered ADCs were linearised by the thermal noise from the vacuum tube microphones. ■ May/June 2006

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your business

The wealth of nations The cash and the fun may have been driven out of a lot of music studios some time ago but there are now big investments coming back into the industry from people who you’d think really ought to know better. DAN DALEY suggests that perhaps they’ve spotted something others have missed.

dan daley

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‘The six-figure recording console that was once the technology anchor of a facility is now a boat anchor. This is the BYOC — Bring Your Own Console — approach to the studio business.’

T

HE REFERENCE TO Adam Smith’s seminal work on free-market economics might be a little obtuse for a pro audio magazine, but it’s deadon for the times we live in. There is a hell of a lot of money floating out there these days. I’m not sure where it came from, but it’s apparent that a bunch of it is finding its way into the music business. In fact, a surprising amount of it is even going into the studio business.

There is an old saying that if you want to make a million pounds in the music industry, start with resolution

two million pounds. That’s looking truer than ever as recording studios continue to go out of business or are driven to the brink by shrinking budgets from their putative benefactors, the major record labels, as those dinosaurs shrivel from revenues lost to illicit downloads and idiotic decision-making. Hit Factory and Enterprise and O’Henry are gone and we’ll almost certainly see EMI Music do a merger of some type with Warner Music or Universal Music before the year’s over. Seems like the only people making money in music are the ones we see on MTV’s Cribs.

And half of them are sports stars with Pro Tools (and an engineer on retainer to run it) in their basements.

So how come some pretty smart guys are putting beaucoup bucks into the next generation of sound palaces? Over in Covent Garden, Microsoft co-founder Paul Allen has sunk eight figures into the Hospital, a multilevel media playground that has not only two music studios and the UK’s first High-Definition broadcast studio but also a gourmet restaurant and a private members club — a high-tech Groucho. And 11 hours away by Gulfstream G-5 (with extendedrange fuel pallet option) in San Francisco, Steve Luczo, chairman of computer drive giant Seagate, is about to put the finishing touches on a 4,000-squarefoot, US$5 million music studio complex that comes with John Storyk acoustics and Industrial Light & Magic interiors.

There are other moves by the swell set to report.

In Miami, a member of the Saudi royal family bought South Beach Studios — and the hotel it’s located in — earlier this year. Up in New York, a wealthy Lebanese businessman acquired Quad Recording.

In both cases, according to local scuttlebutt, the studios were turned over to the musically ambitious (though not necessarily musically talented) next generation of the family to be run as businesses.

We’ll see. What we do know is, musical talent or business acumen aside, they all have the most crucial prerequisite to operate a recording studio these days: money to burn.

I was at the opening of SAE’s new Los Angeles school in April where I was handed business cards for two new studio ventures, both eager for media coverage, one of which is built on the ashes of one of the fallen studios in that neck of the woods. I appreciate the fact that some of these former facilities’s brilliant acoustics and layouts are not going to be turned into mini-marts (or Wal-Marts, in the case of some of the larger ones) but I have to wonder what’s behind this sudden spurt of interest and investment in recording studios on the part of people, some of whom are pretty smart cookies.

Looking closely at some of these ventures, it’s tempting to think that they’re likely doomed. Barring an amazing run of luck, they’ll be vulnerable to the same market dynamics that asphyxiated their predecessors. Fewer, not more, recording artists are seeking out large, conventional, commercial recording studios. Assuming that some of these acquisition ventures came with an existing client base, those clients are subject to the same temptations they always were: using the huge amount of available May/June 2006

your business studio capacity to leverage rates or to bail on using commercial studios in favour of getting their own gear and space. Studio owners in large cities always used the logic that their clients could not achieve everything they needed on their own in dense urban markets because of the cost of space. But if you can do most of your work with a £50,000 AWS-900 instead of a £500,000 9000-J, it’s amazing how much more you can pay for rent on space.

But a look at some of the other new studio initiatives reveals some potentially very workable strategies. Seagate’s Luczo has built three control rooms that will host six staff mixers, at least one of whom, Peter Krawiec, a close associate of GRAMMY winner Walter Afanasieff, has a track record of some significance. At the end of the day, Luczo’s Talking House Productions will own content, perhaps the ultimate currency in the digital age. Anyone who has watched the proliferation of the cable and satellite broadcasting industry, or kept apprised of the increasing multiples paid for music publishing catalogues, or of the independent film business, knows that there are entire cosmos that need their audio filled cost-effectively. And what’s more costeffective than owning the means of production as well as the content itself?

In the case of the Hospital, the strategy there plays into the lifestyle-oriented inclinations of the music industry. Assuming anyone can have Pro Tools, a good console, reasonably decent prêt-a-porter acoustics and the wherewithal to run it, what sets one studio apart from another? Or apart from the comforts of home? The comforts of Paul Allen’s home, or at least of his well-evolved and well-capitalised sense of what a musical home should be. Why hang around a dingy lounge when you can lounge instead around a private bar one level down and hob-nob with those whose careers you truly aspire to emulate?

There are other clues at the Hospital that suggest where the upper strata of the studio business is headed. Where the HD broadcast studio is gorgeously equipped, the two audio recording studios are equally bare. They and their business model take into account that artists, producers and engineers today have their own ideas about what technologies they want to record with, as well as the fact that those technology platforms are portable, affordable and subject to change on a monthly basis. The six-figure recording console that was once the technology anchor of a facility is now a boat anchor. This is the BYOC — Bring Your Own Console — approach to the studio business. The attractions are not the recording spaces but the spaces adjacent to them. Both the equipment and dinner menus are a la carte.

As always, the music business, and by extension the studio business, is best judged by looking for clues in the larger culture. The making of music has followed the music itself into the realm of lifestyle. Page Six in the New York Post has a better rate of return than a page in Mix these days, and failing music careers can be kept alive between comebacks by the adroit implementation of felony indictments and barrister-negotiated community service sentences. (Seriously, no one on my side of the Atlantic Ocean would know who the hell Pete Doherty was if he didn’t keep getting arrested for heroin possession. And as for his erstwhile girlfriend, Kate Moss’s endorsements are now worth more than they were before she was photographed being seduced by Bolivian marching powder.) It all makes you realise that in the grand scheme of things, Alan Parsons got far more mileage out of a good haircut and having his band play one gig per decade than he did engineering for the Beatles and Pink Floyd May/June 2006

put together.

So, will the embrace of the studio business by a new generation of wealthy amateurs (and I respectfully use that word in the classical sense) foster some kind of Renaissance in the studio business, and perhaps the music business, too? I don’t know. But I do know that Paul Allen and Steve Luczo are very smart guys.

They also share something else that’s been lacking in the business lately, a force that, when you look back at the successful musical and recording ventures of the past, was always there in abundance: passion.

True, they have the money to fund their obsessions at a very high level. But considering all of the other things they could be putting that money and time into, the fact that they’re investing their personal brands in the music recording business makes others look and think twice.

resolution

That these guys are putting their resources into an industry that’s been crying the blues about money for the past five years speaks volumes. It’s not like they’re movie stars playing in weekend bands — they’re sharp and savvy businessmen who understand the concept of ‘buy low, sell high’. It makes you think: perhaps the music recording sector isn’t a dog, but rather an opportunity whose share price is so low that, like a tech stock in 2001, it’s become very attractive. If Paul Allen suddenly began investing in mousetraps, I’d seriously consider buying into in a rodent farm. Plus, they’re both computerindustry veterans, and wasn’t it a computer guy named Steve Jobs who figured out the new business model for the music business when all those guys at SonyBMGWarnerUniversalEMI couldn’t? Keep an eye on where all this is headed. ■

77

headroom DUAL CONCENTRIC REPLY I’d always expected that there would be more letters about the ‘Ten’ column because making a (sometimes) serious point in under a 100 words can lead to inprecision through brevity. In this case, I think both Tim Lount and I are not disagreeing but just approaching the subject from opposite directions. (Headroom V5.3) The Tannoy Dual Concentric speaker most certainly deserves its place as an ‘enduring design’. It is interesting to note that other co-axial speaker designs such as the Altec 604, (from a similar era) and a little more recently, the UREI 813, also showed a longevity of professional use. Although I don’t select the accompanying images, I consider the use of a modern Dual Concentric application just emphasises the point of the column — enduring.

With regards to the claimed ‘anomalies’, perhaps I should mention that I’ve used Tannoy monitors in the studio and at home since 1970 (including a wide range of Red, Gold and HPD variants plus Dreadnoughts, FSMs, the DMTs, AMS8, etc), frequently as the sole monitor. The uniform dispersion in both the horizontal and vertical planes, and the resultant stability of the stereo image has always been important to my ears, particularly when close monitoring. The genius of the basic design was to integrate the HF horn and the LF unit such that the former was positioned slightly behind the latter and share a similar face to the world.

The back positioning of the HF unit also partially helps compensate for the inevitable crossover delay, and so bring the unit even closer to the theoretical ‘point source’ response, probably closer than any other monitor but not actually ‘point source’ and my column’s description of ‘virtually point source’ seems fairly accurate.

In long discussions with the Tannoy R&D team in the late 1980s, during the development of the DMT series drivers, as it was a then hot topic, I specifically asked about how much awareness there would have been, if any, of driver delay and alignment in the 1940s. The response was that these topics would not have been an issue and that achieving even dispersion was probably the principle being followed and high engineering skills achieved the fortuitous end result.

My statement that Tannoy has placed less emphasis on Dual Concentric technology in newer models is of course an outsider’s perception, and one that has been relayed to me by several other pro users, probably encouraged by the high visibility of the Reveal series which doesn’t use it, and the addition of extra HF drivers in some models such that, although having Dual Concentric technology at their core, they don’t always appear so to the casual observer.

I’m delighted to hear that ‘Tannoy is placing more and more emphasis in this technology’ but as a pro-user I’ve always felt that its advantages in a studio environment have always been underplayed. It is unfortunate then that the best source of Dual Concentric info, history and technology is not Tannoy’s own website despite the fact that Tannoy-originated material is available. In fact keying ‘Dual Concentric’ into the site search facility at www.tannoy.com produces a rather disappointing response.

Keith Spencer-Allen

78

WORD ORIGINS Just reading the latest Resolution. Great to see bus and buss explained!

Gobo is, I thought, something to do with projecting an image — e.g. logo — using a metal cut-out in front of a lamp. I’ve never heard it used referring to an acoustic screen. My Collins Concise says ‘a shield placed around a microphone to exclude unwanted sounds, or round a camera lens etc. to reduce the incident light.’ Oh. Never knew that.

I am currently engaged (as and when, as they say) in editing and entering into the IBS website a vast compendium of audio knowledge amassed by the late Antony Askew (IBS member 001).

Dubbing : The process of copying from one recording to another. It is a corruption of the term which came about in the early days of the cylinder recording industry which could not produce multiple copies by ‘pressing’. By means of a mechanical ‘pantograph’ system, one cylinder could be copied onto another. This appears first to have been employed by unscrupulous companies who would engage an artist, and pay him/ her for so many ‘rounds’ of recording (perhaps four cylinders could be made at one time, so four ‘rounds’ of a song would produce 16 cylinders) and then, by ‘doubling’ these, a far greater number could be issued, to the profit of the company.

It was discovered by chance by Russell Hunting who had recorded and been paid for making 16 cylinders in New York, only to chance upon a boy carrying a tray of 24 cylinders labelled as being the particular piece that Hunting had just recorded. It came to be called Doubling, now ‘corrupted’ to Dubbing. Hunting, by the way, threatened Court proceedings and received payment for all the doubled copies that had been made in the past.

Never knew that, either!

I ‘did’ know about Foley — which in the Beeb we called Spot Effects. ‘Those sound effects which were made in the studio by a “spot effects” SM which were best done live with the actors — tea cups, etc. In the film world this is called “Foley” after their Hollywoodbased originator Jack Donovan Foley (1891–1967).”

John Andrews, chairman, Institute of Broadcast Sound, UK You’ve probably had a slew of emails telling you that ‘omnibus’ is the dative plural of the Latin omnis, hence ‘for all’, but your entry for ‘wrap’ triggered alarm bells in my head. Your suggested derivation is almost certainly what some linguists call a ‘backronym’ — an invented (and frequently inventive) derivation of a word.

Acronyms didn’t come into widespread use until quite late in the 20th Century (indeed the word ‘acronym’ itself didn’t come into use until 1943, according to the OED).

It seems far more likely that ‘wrap’ simply comes from ‘wrap up’, meaning to finish off or complete.

Dom Latter, Cambridge I’d like to thank everyone for their response and input to this topic — all of which adds, or reveals, another layer of the origins of our terminology. Still no input on the subject of ‘jitter’ and ‘glitch’ though. Continued input is welcome as there are still more words whose origins need exploring.

Keith Spencer-Allen

resolution

Proving that there is a life after pro audio publishing, one-time colleague Sam Patel threw away his contact book, tie and some pretty sharp suits to start a life as a Restaurateur in Dubai’s Bur Dubai region with his Cinnamon Indian Vegetarian Restaurant. That’s right, vegetarian, and dry too on account of it being the UAE. But business is brisk and the food is good although his Banter Rating is only averaging a 5 these days because he’s not getting to exercise it.

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May/June 2006

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