International TECHNOLOGY AND TRENDS FOR THE PRO-AUDIO PROFESSIONAL www.audiomediainternational.com
December 2016
BRINGING TOGETHER THE BEST NEW TECH OF 2016 p20
GAME AUDIO
HOW TO
REVIEW
Recreating the sound of a warzone for ‘Battlefield 1’ p26
Preparing the perfect control room with GIK Acoustics p30
On tour with Roland’s M-5000C console p32
ULTRA-COMPACT MODULAR LINE SOURCE Packing a 138 dB wallop, Kiva II breaks the SPL record for an ultra-compact 14 kg/31 lb line source. Kiva II features L-Acoustics’ patented DOSC technology enhanced with an L-Fins waveguide for ultimate precise and smooth horizontal directivity. WSTŽ gives Kiva II long throw and even SPL, from the front row to the back, making it the perfect choice for venues and special events that require power and clarity with minimal visual obtrusion. Add to that a 16 ohm impedance for maximized amplifier density and a new sturdy IP45 rated cabinet, and you get power, efficiency and ruggedness in the most elegant package. www.l-acoustics.com
WELCOME EDITOR Adam Savage asavage@nbmedia.com
Experts in the issue
THAT WAS THE YEAR THAT WAS
MANAGING EDITOR Jo Ruddock jruddock@nbmedia.com STAFF WRITER Colby Ramsey cramsey@nbmedia.com ADVERTISING MANAGER Ryan O’Donnell rodonnell@nbmedia.com
Etienne Corteel is CTO of 3D sound specialist Sonic Emotion Labs.
ACCOUNT MANAGER Rian Zoll-Khan rzoll-khan@nbmedia.com HEAD OF DESIGN Jat Garcha jgarcha@nbmedia.com DESIGNER Tom Carpenter tcarpenter@nbmedia.com PRODUCTION ASSISTANT Warren Kelly wkelly@nbmedia.com CONTENT DIRECTOR James McKeown jmckeown@nbmedia.com
Press releases to: ukpressreleases@nbmedia.com
Audio Media International is published by NewBay, The Emerson Building, 4-8 Emerson Street, London SE1 9DU, England. Editorial tel: +44 (0)20 7354 6002 Sales tel: +44 (0)20 7354 6000 Audio Media International ISSN number: ISSN 2057-5165 (Print) Circulation & Subscription enquiries Tel: +44 (0)1580 883848 email: audiomedia.subscriptions@c-cms.com Printed by Pensord Press Ltd
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Sadie Groom is MD at Bubble & Squeak, a PR, marketing and events agency specialising in broadcasting, entertainment technologies, AV and business services.
David Shevyn is general manager of GIK Acoustics Europe, which manufactures a complete line of acoustic treatment products for various environments, as well as professional design consultation services.
Milos Stojanovic is a sound editor/ designer and managing director of Sound Professionals, a new talent agency for the audio post industry.
lthough I’m not sure I agree with Andy Williams that “it’s the most wonderful time of the year” – don’t worry I’m not a Scrooge-type, it’s just I’m more of a summer guy and plus my boiler at home is busted at the moment so please forgive the grumpiness – I do love December, not just because it’s mince pie season but because it’s a chance to look back on the previous 12 months and reflect on their various highs and lows. Newspapers, blogs and social networking sites have all been awash with comments from people all over the world saying how 2016 has been a year to forget, mainly due to unexpected political outcomes and a staggeringly high number of celebrity deaths – and let’s not forget that we’ve lost some big names of our own in this industry – but it’s not been all bad. You might also feel that 2016 has offered up more disappointments than positives, but hey, at least we can always rely on the manufacturers of the pro-audio industry to keep things fresh on the product front, and so keeping
A
with tradition here at Audio Media International, we’re once again bringing the year to a close with our annual gear round-up for you to have a flick through. And it’s not just us looking back on recent developments in this issue – our monthly Opinion section sees audio producer Jerry Ibbotson and Bubble & Squeak boss Sadie Groom respond to other news that has left some concerned: Adobe’s fascinating-butkinda-scary Project VoCo and Sound Women’s unfortunate closure, which has led to further discussions about whether women are being given a fair chance industry-wide. Then there’s the team at game developer EA DICE, who were required to go much further back in time when they began work on Battlefield 1, a project that saw them tasked with recording World War 1-era weapons and vehicles to take in-game realism to new heights. Bringing things back up to date over on the live sound side, though, we have an in-depth review of the Roland M-5000C console, as well as an interview with the brains behind Sonic Emotion Labs 3D audio technology that had me and a roomfull or other audio pros intrigued when we were given a preview recently. So there we have it. However it’s gone for you, 2016 is now in the rear view mirror already, and I’m certainly looking forward to seeing what 2017 brings.
Adam Savage Editor Audio Media International
December 2016
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CONTENTS
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www.audiomediainternational.com
20 FEATURES
PRODUCT NEWS 6 8 10 12
Lectrosonics reveals PDR recorder Symetrix expands Prism Series New dLive apps from Allen & Heath Eighteen Sound adopts Powersoft IpalMod
PEOPLE 16
OPINION Jerry Ibbotson considers the technical and ethical questions raised by Adobe’s Project VoCo
18
Bubble & Squeak’s Sadie Groom on how we can encourage more females into the audio industry
42
INTERVIEW Adam Savage speaks to Sonic Emotion Labs’ Etienne Corteel about Wave 1, the company’s innovative system for bringing 3D listening to a variety of venues
42 ALSO INSIDE 14
NEWS IN DEPTH We take a look around Genelec’s impressive West End demo facility
TECHNOLOGY 30
4
HOW TO GIK Acoustics’ David Shevyn offers his top tips for creating a quality control room December 2016
20
GEAR OF THE YEAR After a bumper year for entries, and with help from our talented team of reviewers, AMI reveals its pick of the technology launched over the past 12 months
24
BROADCAST FOCUS Community station Fylde Coast Radio is on a mission to realise its full potential and better serve its audience but it’s struggling to acquire audio gear. Colby Ramsey asks how we can help
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GAME AUDIO The team behind Battlefield 1 talks about creating authentic sound for this World War 1-based game
28
COMPANY PROFILE Milos Stojanovic introduces us to Sound Professionals, a talent agency that exclusively serves the audio post-production domain
32 REVIEWS 32 36 38 40
Roland M-5000C Drawmer MC3.1 Grace Design m108 AKG K872
THE LEO FAMILY TRUE SOUND IN LINE ARRAYS.
The LEO Family provides power and clarity for nearly every application, from intimate performance spaces to the world’s largest outdoor festivals. LEOPARD, the smallest in the family, is gaining a following for being the most lightweight and versatile line array in its class. From small to midsize to large-scale, this family of line arrays has you covered.
meyersound.com/believeit
PRODUCT NEWS
NUGEN INTRODUCES MASTERCHECK PRO MasterCheck Pro is a new loudness, dynamics and codec toolset for music professionals, which enables producers to hear and visualise the playout encoding and loudness matching techniques now in use by music streaming service providers such as Spotify, Apple Music, YouTube and Digital Radio. Ideal for both mix and mastering engineers, MasterCheck Pro lets producers audition different codecs in real time to help avoid codec distortion or over-compression, resulting in perfectly tuned masters for each digital music streaming service, Nugen says. The plug-in lets users instantly hear how their tracks are affected by loudness matching algorithms on different devices and services. Real-time codec auditioning helps users understand how masters respond to different codecs and bandwidths by giving them the ability to preview the mix through different encoding schemes and quality settings. Industry-standard meters show loudness and peak-to-loudness ratio (PLR or dynamic range) and how these factors are affected when the music is played on the target platform. Producers can also A/B the master FX chain with other reference tracks or with the uncompressed mix.
MasterCheck Pro’s loudness auditioning also reveals any level changes that may be applied by the streaming service. Its true-peak metering ensures the mix is within the delivery spec and guarantees the music won’t be clipped at later encoding stages. For a limited time, MasterCheck Pro is available at an introductory price of $149 (regular price $199). The plug-in supports AAX, VST, AU and RTAS for both Windows and OS X operating systems. www.nugenaudio.com
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SENNHEISER LAUNCHES DIGITAL 6000
The newest addition to Sennheiser’s Professional Wireless Range uses the same long-range mode and proprietary Digital Audio Codec as the Digital 9000. Comprising a two-channel receiver in two different versions, a bodypack and a handheld transmitter as well as a rack-mount 19in charging unit, the Digital 6000 series will be available from March 2017. The two-channel receiver works across a switching bandwidth of 244MHz (470 to 714MHz) covered by three transmitter versions (470-558MHz, 550-638MHz, and 630-718MHz). For larger systems, up to eight receiver units can be daisychained without an additional antenna splitter; the multi-channel system will work with a single pair of antennas.
The receivers are fitted with a Link Quality Indicator that ensures issues get seen before any drop-outs occur. The intelligent error concealment of the Digital 6000 employs learning algorithms to replace the corrupted signal, enabling it to continue transmitting. The transmitters on the Digital 6000 use the same rechargeable accupacks as the Digital 9000 units and the SKM 6000 handheld transmitter is fitted with the company’s standard capsule interface. The SK 6000 bodypack can be used as a high-end solution for wireless instruments such as guitar and bass, and is also ready for use with Sennheiser’s range of clip-on microphones. www.sennheiser.com
LECTROSONICS REVEALS PDR RECORDER
NEW DANTE ANALOG OUTPUT MODULE Audinate has announced the immediate availability of the Dante Analog Output Module – a complete, small-form-factor PCB that allows for rapid development of Dante-toanalogue endpoints. Manufacturers can use this module in a variety of small footprint enclosures with appropriate analogue connectors to rapidly bring finished products to market. The module supports one RJ45 Dante input and one or two balanced analogue outputs. It is able to receive audio channels from a Dante network and provide studioquality, low-latency audio via balanced output connectors to analogue audio equipment.
www.audiomediainternational.com
Any audio available on the Dante network can be routed via the outputs to an amplifier, powered speaker, mixing console, DSP, or other analogue audio device. Adapters (for example XLR-to-RCA and XLR-to-phono) can be used to connect to audio equipment without the required builtin connectors. It features a digital-to-analogue converter and supports a range of sample rates and bit depths, also providing a hardware master clock for a Dante network. www.audinate.com
The PDR recorder is designed to capture professional quality audio – synchronised with timecode – for situations with extreme distances or where using a wireless microphone is not practical. The tiny size and rounded shape make the PDR unobtrusive and easy to place in garments and costumes, and easy to conceal when used as a “plant” microphone to capture environmental or location sound. The PDR records in 24-bit, 48kHz digital onto a Micro SD card (HC type) in .wav (Broadcast Wave File) format. The PDR can be jammed to external time code via the standard five-pin LEMO connector. A headphone output jack can be used to monitor the signal input or listen to previously recorded files.
Setup and adjustment is made through an interface provided by the keypad and LCD. The housing is machined from a solid aluminium billet then hard anodised for the ruggedness needed in field production. The PDR is able to run for over six hours on a single lithium AAA battery. The Lectrosonics PDR is available now with a US MSRP of $900. www.lectrosonics.com
CONTROL COMMUNICATE
INTERCOMANDMEDIORNETCONTROL
TOGETHER IN ONE DEVICE
SMARTPANEL RSP-2318
RIEDEL widens its SMARTPANEL app portfolio with the new MEDIORNET CONTROL app. Switch video, audio or combine in macros… all while using your intercom.
www.riedel.net
PRODUCT NEWS
CLEAR-COM EXTENDS HELIXNET LINE Clear-Com’s new beltpack (HXII-BP-X4) for the HelixNet digital partyline intercom systems is now shipping. In addition to a lighter weight and a tougher, more ergonomic design, the HXII-BP-X4 wired beltpack uses power over Ethernet (PoE) to access any two of the 24 channels on the system at once. The new HelixNet beltpack – designed and manufactured in the USA – will succeed the first-generation HelixNet beltpack. The molded polycarbonate case of the new beltpack is 33% lighter and 13% smaller than its predecessor, and each user can assign and access any one of the 24 intercom channels to a single intercom key or multiple channels to a single key set thanks to the ‘Stacked Key’ feature.
The HXII-BP-X4 operates over a single microphone cable for digital powerline or a single PoE cable. It features one set of Talk and Call buttons for each of the two intercom channels, and up/down arrow buttons for programme audio level adjustment. Three-pin XLR and RJ-45 Ethercon intercom connectors are located on the bottom panel for connection to the intercom station and a four-pin XLR connector is provided for intercom headsets, with a five-pin XLR connector also available as an option A micro-USB connector is provided to update firmware in the beltpack, and this can also be achieved automatically whenever the HXII-BP-X4 is connected to a HelixNet Main Station running a more current version of the firmware. www.clearcom.com
www.audiomediainternational.com
SYMETRIX EXPANDS PRISM SERIES Offering expansion via 64 channels of bidirectional Dante networking when used as a DSP core or co-processor, the new Symetrix Prism 0x0 delivers processing, mixing and routing for Dante-enabled endpoints, while providing ‘exceptional’ sound quality, the company says. The Prism 0x0 is identical to the other Prism DSPs in terms of processing power and delay memory; it is unique however in that it has no analogue inputs or outputs and no external control inputs or logic outputs. Housed in a 1/2 rack x1U rack-mount chassis, the DSP includes flexible mounting options and power over Ethernet (PoE+). “The Prism 0x0 has many applications, including feedback elimination, auto-mixing and FIR filters for processing loudspeakers,” explained Symetrix senior product manager Trent Wagner. “Simply insert the Prism 0x0 between the remote stage-input boxes and the
console, if Dante-enabled, or use the Prism 0x0 as an insert or bus processor. For outputs, relay the signals through the Prism 0x0, if the amps are Dante-enabled, or insert it on the console’s busses.” Like other Prism series DSPs, Prism 0x0 can be customised, connected and configured using Symetrix Composer, a Windows-based CAD programme. Prism 0x0 can be controlled from Symetrix ARC wall panels, ARC-WEB browser-based interface and thirdparty touchscreens. www.symetrix.co
PRO-R REVERB PLUG-IN FROM FABFILTER WINDOWS DRIVER FOR FOCUSRITE CLARETT FabFilter Pro-R is a high-end algorithmic reverb plug-in with musical controls and what the firm describes as ‘a very natural sound.’ The plug-in features Space Control, which enables it to seamlessly switch between dozens of room models with automatic decay time adjustment, as well as an industry-first Decay Rate EQ concept that introduces free adjustment of the decay time over the frequency spectrum. FabFilter Pro-R also features built-in Post EQ and a range of high-quality factory presets, while its algorithms have been designed to blend well with the source and fit easily into a mix. FabFilter Pro-R takes a musical approach to reverb, offering user-friendly, non-technical controls like brightness, character, distance and stereo width to tweak the reverb sound.
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The plug-in’s Decay Rate EQ with low/high shelf, bell and notch filters offers free adjustment of the decay time over the frequency spectrum. As it uses parametric EQ bands instead of a traditional crossover system, it provides much more flexibility to shape the decay time characteristics. FabFilter Pro-R is now available for $199 or £149, supporting both Windows and Mac OS X in VST and VST 3, Audio Units, AAX, RTAS, and AudioSuite plugin formats. It also supports common Pro Tools hardware control surfaces. Discounts are available to existing FabFilter customers. www.fabfilter.com
Focusrite has rolled out the first version of a new Windows driver for its Clarett range. Windows users will enjoy sub 3ms round trip latency and the same reliable operation that Mac-based Clarett owners already experience. The low latency allows users to record and monitor while using their favourite plug-ins in real time, along with other DAW workflow benefits. The first release will support Thunderbolt connectivity via Display Port for all Clarett interfaces, and the driver will be compatible with the latest 64-bit updates of Windows 8.1 and Windows 10. Future iterations will add support for Thunderbolt 3 Type C connectivity and the Red range of interfaces.
The driver also supports Focusrite Control, Focusrite’s included software mixer designed for easy configuration of monitoring and routing setups, which is now available in a Windows version, again with identical functionality to the Mac version, including support for Focusrite iOS Control. Focusrite’s Clarett audio interfaces feature specially designed highperformance, low noise, low distortion mic preamps, which include the analogue “Air” effect, emulating the sound of Focusrite’s heritage transformer-based mic preamps. The interfaces include precision 24-bit/192kHz conversion and offer super-low latency. www.focusrite.com
Inspiring Every Moment Audio-Technica’s In-Ear Monitor Headphones Bringing the worldwide critically acclaimed sonic signature of the M-Series to an in-ear design, the Audio-Technica in-ear monitor headphones have been designed to fully answer the needs of demanding sound professionals and musicians from the studio to the stage and the DJ Booth.
www.audio-technica.com
www.audio-technica.com
ATH-E70
ATH-E50
ATH-E40
PRODUCT NEWS
www.audiomediainternational.com
ALLEN & HEATH DELIVERS NEW DLIVE APPS Allen & Heath has designed two new iPad apps to ‘further enhance the dLive digital mixing experience.’ dLive MixPad provides comprehensive remote control of the live mix, while dLive OneMix focuses on personal monitoring. MixPad offers extensive control capability over channel processing, including filters, gates, parametric and graphic EQ, compressors and input/ output delays, and gives the user instant access to any of the mixer’s channel faders and mutes, DCA faders and mutes, pan, aux sends and assignments as well as mic-pre control and full metering. Other features for system setup include a real time analyser (RTA) to assist EQ’ing monitors and PA, the ability to name and colour channel strips, custom layers with drag ‘n drop strip setup, and channel PFL control for monitoring. dLive OneMix is a version of the dLive MixPad app that locks control
to a single aux monitor mix to give the musician customised personal monitor control without affecting the other monitors or FOH main mix. Multiple iPads can be set up by an admin user to give artists a custom set of controls. It is also possible to connect multiple MixPads to run in conjunction with a dLive Surface or Director software to provide simultaneous control, including independent PAFL buses. MixPad and OneMix run on dLive firmware V1.3, which is available to download from the website below. www.allen-heath.com
WAVES ABBEY ROAD VINYL NOW AVAILABLE Waves Audio has introduced the Waves Abbey Road Vinyl Plugin – a precise model of the iconic studios’ vinyl cutting and playback gear designed to give music the vintage warmth of vinyl records played on classic turntables and needles. Created in collaboration with Abbey Road, the plug-in is designed to capture every stage of the vinyl production and playback process. Users can choose between the sound of a pure acetate (lacquer) cut or the print master vinyl pressing from the factory. Records can be played on two distinct turntable types with a choice of three classic cartridges, and tracks can be sent through Abbey Road Studios’ legendary TG12410
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mastering console on the path into the vinyl lathe. Abbey Road Vinyl also lets users move the position of the tone arm across the record, changing the frequency response and distortion like in the ‘real world’. Users can also add vinyl noise, pops and crackles, apply a gradual slow-down/stop turntable effect and introduce wow and flutter effects for extra analogue warmth. The plug-in is included in the Waves Abbey Road Collection bundle. www.waves.com
C R E ATI V E M A STE R I N G. R E I N V E NTE D. WaveLab is today’s leading mastering and audio editing platform, favored by mastering facilities, music studios, sound designers, journalists and broadcasters. Its comprehensive set of features, customizability and outstanding audio quality are the reasons WaveLab became the world’s most popular professional platform for audio reďŹ nement. WaveLab Pro 9 reinvents creative mastering once again by providing a revolutionary new user interface, full M/S mastering support including editing and processing, the superior MasterRig plug-in suite as well as direct exchange with Steinberg DAWs, such as Cubase, among many other features.
www.steinberg.net/wavelab
PRODUCT NEWS
www.audiomediainternational.com
EIGHTEEN SOUND ADOPTS POWERSOFT IPALMOD MODULE Powersoft has announced that Italian speaker manufacturer Eighteen Sound is the latest transducer specialist to implement the company’s patented IPAL technology. Eighteen Sound has designed its new iD Series transducers to couple perfectly with Powersoft’s Differential Pressure Control (DPC), the core technology behind the company’s patented IPAL (Integrated Powered Adaptive Loudspeaker). To achieve this, Eighteen Sound has adopted the Powersoft amplifier hardware module, IpalMod, which overcomes the limitations of traditional transducers to produce what the company describes as ‘unparalleled’ output at low frequencies. It does so by featuring a zero latency pressure-sensor feedback, applying real-time correction
RTW BEGINS SHIPPING MM3 MUSICMETER
to maximise Eighteen Sound high efficiency 18in and 21in transducers. Powersoft’s IPAL features a low frequency-dedicated system that integrates an 8.5kW high-power, highperformance switching mode amplifier module (the IpalMod) together with ultra-high motor force electro-dynamic transducers – the 18iD and 21iD Eighteen Sound speakers. Both IPAL-compatible transducers have a nominal 2-Ohm impedance. In order to fully integrate them into the IPAL system, and optimise performance, a pair of either model should be connected in parallel when matched to an IpalMod amplifier module. www.powersoft-audio.com www.eighteensound.com
RTW, manufacturer of visual audio meters and monitoring devices for professional broadcast, production, post production and quality control, has announced that the latest addition to its loudness metering family, the MM3 MusicMeter, is now available. The MM3 is designed to provide flexible, user-friendly operation for customers looking to implement loudness metering into musicbased applications.
It features vectorscope, PPM/ TruePeak or VU, real-time analysers, Loudness vs. Time charts, as well as numerical and graphical loudness display and zoom modes, all accessible with a simple swipe. The display can be adjusted for vertical or horizontal modes to accommodate the needs of its users. Supported audio inputs include analogue, SPDIF and USB, and an SPDIF output delivers a buffered stereo signal or downmix from a 5.1 stream. The MM3 MusicMeter is compatible with RTW’s USB Connect software package, which is available at no cost from the company’s website. This software equips its user with a direct audio signal transfer and additional control from within a DAW environment, further expanding the capabilities of the MM3. www.rtw.com
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NEWS IN DEPTH
www.audiomediainternational.com
AN INSIDE LOOK AT GENELEC’S WEST-END DEMO FACILITY We were recently given a tour of Scrub’s new HQ in central London to see what its world-leading demonstration space for the Finnish monitor brand has in store for visitors.
Howard Jones
ituated in the Fitzrovia area of London lies a stateof-the-art demonstration facility contained within the new offices of HHB Communications’ post production reseller division Scrub. The demo room – a combined vision designed to serve as a hub for pro-audio activities – is the world’s first global showcase of Genelec products in a controlled environment outside of its factory in Finland. With enough space for 15 to 20 people, the facility is an ideal destination for those who wish to come and learn not only about Genelec and its speaker products, but also about modern audio technology in general. The setup allows visitors to choose source material of virtually any format and evaluate Genelec monitors in stereo, surround and immersive audio configurations. The brilliance of the centre’s audio architecture is the ability to route any source to any destination, so visitors can choose to audition anything from a stereo pair of compact 8010A monitors to a full 11.1 immersive audio system. “We can hit a variety of markets with this room: post guys, broadcast, music, install – it tells the whole story
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really,” said Howard Jones, Genelec’s recently appointed international project manager. “You can rig pretty much anything to anywhere because of the way the room has been set up. It’s very flexible and intelligent.” The 11.1 system in question is connected to an Avid Pro Tools | S6 control surface and combines seven 8351A SAM studio monitors in left, centre, right, left side, right side, left back and right back positions, four 8330A SAM studio monitors overhead and two 7370A SAM studio subwoofers handling the low end. Complementing the Centre’s Pro Tools | S3 control surface is a separate 5.1 configuration of five 8320A SAM studio monitors and one 7350A SAM studio subwoofer, which provides what Genelec describes as outstanding monitoring accuracy and flexibility in a smaller footprint, while the 8010A studio monitors and two 8430A audio-over-IP SAM studio monitors demonstrate both analogue and digital compact loudspeaker formats.
Scrubbing up well John Johnson, Scrub’s chief technical officer, describes the room as ‘chalk and cheese’ compared to their previous demo space, and explains how they
wanted to do something on a larger scale when the new property on Wells Street was located. “It just lent itself to having a whole floor with a proper demo space, broken up with a small-environment system which is more of a tracklay/ premix room along with the main ‘hero’ suite,” explained Johnson. “As a result of this collaboration, we are also able to do presentations, training sessions and product launches, as well as distributor meetings and larger events.” A second 5.1 system showcases Genelec’s architectural installation products and includes five AIC25 active in-ceiling speakers and one 5041A active in-wall subwoofer, while two AIW26B active in-wall speakers can be run in stereo mode or in 2.1 mode in conjunction with the 5041A. Finally, four 4030B installation speakers are positioned in the Scrub sales office to demonstrate small to mid-sized commercial installations. “The location is natural because so much of our business is taking place here in Soho among the post-production companies, which is of course the market that Scrub serves,” remarked Genelec’s PR director, Lars-Olof Janflod. “The other motivation is the fact that we can get people in here to listen to our products before they purchase a top-of-
the-range active speaker. If you’re going to buy something like this you just have to hear it in a proper environment.” Visitors are able to choose material from a range of sources including MP3, CD, DVD, Blu-ray, PlayStation, Pro Tools, Logic, Nuendo, Spotify and Tidal – with Blu-ray, Pro Tools and Nuendo providing Dolby Atmos compatibility. The system is also compatible with Auro-3D and DTS:X, and can support low order Ambisonics and Binaural formats. “We’re focusing more on Dolby Atmos but we’ve still got the capability of having a voice of God speaker above your head when doing a DTS:X playback,” Johnson added. “It’s about being able to demonstrate everything as best we can, as well as tying it all into the products that are popular with our customers.” Extensive use of high-quality signal processing, audio conversion and Dante audio networking – courtesy of products from BSS, Focusrite and DAD – helps achieve the desired levels of flexibility and configurability while also reflecting the growing adoption of this type of technology in the marketplace. Jones concludes by stating how he believes the location of the facility is ideal, and perfectly placed to suit the needs of international customers. “We’ve been talking to the Music Producers Guild about putting on events here,” he concluded. “There are some interesting discussions going on in the pro-audio community at the moment and we want them to take place here.” www.genelec.com www.hhb.co.uk
Vero is a large format sound system, which has been engineered for new levels of audio and operational performance. It has been designed, developed and perfected over the last six years by some of the most knowledgeable and experienced audio engineers in the world. Its meticulously crafted proprietary waveguides and driver technology produce naturally even frequency response and coverage. The result is uncompromised system dynamics, headroom and coherency combining to present an incredibly spacious stereo image, which is why Vero is already gaining the plaudits of sound engineers from around the world.
www.vero-system.com
T H E
N E X T
L E V E L
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T O U R
S O U N D
OPINION
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CAN YOU REMEMBER YOUR FIRST EDIT? Jerry Ibbotson reacts to Adobe’s recent reveal of Project VoCo, a new speech editing application it’s working on that has raised a number of technological and ethical questions.
JERRY IBBOTSON
hat’s kind of a personal question but anyone who has worked with voice in sound will have a moment where they hear someone’s voice say something they didn’t, or at least not quite in that way. For me, it was way back in the last century as a trainee radio journalist working with quarter-inch tape. One of the exercises we were given was tidying up a classic archive of a US radio reporter trying to report on “A fire at the Firestone tyre factory”. Say it a few times yourself and you can imagine the mess he got himself in to. It illustrated the power of a tidy voice edit. Across the years some things have changed but the basic principle is the same. Even with digital audio and a visual waveform on-screen, you still ultimately use your ears. But now software giant Adobe has demonstrated something that could shake that up and which throws up some serious ethical questions. At its recent Adobe MAX Sneaks event in San Diego it unveiled a tech demo of Project VoCo. Beyond a few videos online there’s been no official announcement but this event has thrown up technology in the past that has worked its way into Adobe software such as Audition and Premiere. Putting VoCo through its paces was developer Zeyu Jin. He described it as being like Photoshop for audio. I can understand where he’s coming from and it may make sense for his nonaudio audience, but you could say that any audio editing package or DAW is already that. Adobe’s own Audition (my
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DAW/editor of choice) uses plenty of Photoshop-like tools. Shaky start aside, what Zeyu Jin demonstrated left his audience stunned. He began by playing a clip of someone talking about his reaction to being nominated for an award. “I kissed my dogs and my wife.” VoCo automatically transcribed the words into text. Jin then copied the word “wife” and pasted it over the word “dogs” in the text window. VoCo played the audio back. “I kissed my wife and my wife.” The edit wasn’t 100% perfect – there was a slight clumsiness at the start of the first “wife” (the pasted one) – but it was still worthy of the applause it received from the audience. Bear in mind this is the original voice speaking – it sounded as it would if you edited the waveform but it’s been achieved by manipulating text. There was more to come. Zeyu Jin wanted to put the dogs back in. He just wrote the word “dogs” at the end of the sentence, over the second “wife”. “I kissed my wife and my dogs.” This time it sounded exactly right, with nothing to fault. And this was done by simply writing the word.
phenomes it had collected from the sample speech. It hadn’t just reordered words it had made them up or copied one that was already there. Let’s not pretend that a good audio editor couldn’t re-jig a recording to make someone say something they hadn’t. I can remember hearing a quarter-inch tape edit of the Queen giving a speech in which she described a member of her family in language I can’t imagine Her Majesty ever using. And that was more than 25 years ago. But VoCo takes it to another level. It can literally build words from thin air.
How does it work?
The Implications
Well it needs a decent sample of the original voice – around 20 minutes in this case. It then studies the building blocks of the individual’s speech – the phenomes. These are all the different sounds than make up spoken language, like ffs and esses and so on. VoCo then reassembles them on demand. It’s like a form of speech-sampling with the interface being text, instead of a waveform. The principle of using ears to make the edit has been broken. The demo in San Diego wasn’t quite over. Zeyu Jin went back to VoCo and overwrote the word wife with “Jordan” (Jordan Steele on stage, presenting the MAX event). The audio now said, “I kissed Jordan and my dogs.” This was VoCo building a new word from scratch, assembling it using the
So what do other people think? I asked BBC radio journalist and tech-head Nick Garnett. “Potentially it’s the biggest change in editing since waveforms came to the screen of your computer. It is fraught with post-truth issues. It enables quotes to be corrected after the event. Or changed. “Veracity will rely on time stamped recordings or multiple recordings by different reporters. It’s the equivalent of the reporter huddle where newspaper hacks gather round one another to ‘build’ the quote. Honourable hacks would only manipulate a tense or the odd word. Unscrupulous hacks would make the whole lot up.” This is something Adobe is aware of, as VoCo developer Zeyu Jin told the audience in San Diego: “We have
researched how to prevent a forgery, like watermarking. As we are getting the results much better and making it so people can’t distinguish between the fake and the real one, we’re working on how to make it detectable.” But the reassurance doesn’t cut it with everyone. I showed the video of the Adobe demo to another BBC journalist, my former colleague Huw Williams. He was even more alarmed at the potential for misuse that the concept offers. “I think it sounds alarm bells. I can’t understand why someone has developed it and I’d like to know who they think is going to use it? Why would you want to make someone say something they didn’t? “At a time when Fake News is in the spotlight after the US election, this has the potential to make things worse. It’s clever and impressive but some questions should be asked about how it might be used.” For now it’s just on a test-bed but given that previous MAX demos have made it out into the real world, we could be about to see a revolution in voice editing that goes way beyond swapping words around or cutting down an interview. Jerry Ibbotson has worked in pro-audio for more than 20 years, first as a BBC radio journalist and then as a sound designer in the games industry. He’s now a freelance audio producer and writer.
“After months of planning, testing and finding the perfect sound solution for a venue, for me it’s all about that moment when a project truly comes to life.”
THE SOUNDMAKERS > HK Audio is the German pro audio brand offering the easiest way to the best sound. From portable to professional live sound to install solutions for over 30 years, we build PA systems for those who are fascinated by the energy of sound. Giving them a stage. Giving them a home.
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OPINION
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NO MORE LIP SERVICE
Bubble & Squeak managing director Sadie Groom on what needs to be done to give female audio professionals equal opportunities at all levels across the industry.
SADIE GROOM
n early November this year, Sound Women, the networking group for women and men in radio, announced it was closing its doors. For many of the group’s members this was seen as a huge loss but it felt like the right time due to time constraints on the volunteers running the organisation and crucial funding being pulled by Creative Skillset. As someone who’s been trying to get another self-funded female association off the ground, I know exactly how Sound Women must’ve felt, and it made me think about what I was doing. However, rather than shutting up shop on my plans, it made me look into the industry’s networking groups in both the audio and broadcast space and what rapidly needs to be done to support women in this space. First off, I’d better explain why this is important. Sound Women was incredibly successful at increasing the number of female presenters on air. In fact, due to its work with the BBC, over 50% of its local stations now have a female presenter for their breakfast show. Is this great? Shouldn’t it be this high anyway as females make up 55% of the population, and what about the stats in other roles, especially the more technical ones? Soundgirls.org has identified that women make up only 5% of the workforce in audio production – this figure is not only shocking, it also hasn’t changed, which is disappointing. How does this play out?
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At a recent tradeshow I was on a client’s stand and someone came up and asked a question to a member of the sales team, who said, “I will just get Joanne for you” and the gentleman in question replied “no I need to speak to someone technical”.
Much work to do This is just one example and my main problem is that opinions/perceptions like this are not changing, and they aren’t going to encourage women to come into the industry. Normally when I do talks on women in the industry a lot of people (male and female) run for the hills. But when I ask if any of them have a daughter, niece, granddaughter and wouldn’t they like them to have the opportunity to work in such a great industry and have no barriers to doing it they tend to say ‘yes please’ and start to listen. I ask people to take a look at the awards in the industry, which are often run by the associations and see how many women are shortlisted and how many women are on the association’s boards. Are these two things connected? Well yes I think they are because women encourage other women and use their networks to get them to enter or put them forward, which is something we aren’t naturally comfortable with. Having spoken to some of the associations in the audio, broadcast and
AV sectors they will all say the same thing about getting women on their boards and in their awards and conference programmes – it’s not for want of trying. So what needs to be done about it? I like a good list and the below actions can be taken whatever gender you are, the size of your company, your job role or employment status. • If you’re female get out there – start putting your hand up, volunteering to speak at events, find out who does your company’s marketing/PR and see how you can get involved. Enter awards – you won’t be the only one doing it so enjoy it, join the groups and in the words of Sheryl Sandberg, “Lean In”. • Review the situation in your company internally. How is the management made up? Are traditional male and female roles prevalent and are you flexible on these? What is the path to get on the management team? What policies do you have for maternity and paternity leave? Is the environment and culture conducive to encouraging women to apply? • Review your company externally. What do the images on your website say about your business and views on diversity? If you’re a manufacturer how do you approach tradeshows for your female staff and customers?
• Think about the examples you use in business. When pointing out the ‘Top 10 Dubbing Mixers’ make sure you have thought whether there is a female in that list because she could be a really good role model for someone else. • Mentoring – this is important as it is a well-known fact that females are inspired by role models. Women in film and TV and other groups run excellent mentoring schemes but also ask if you can have a mentor or volunteer to be one. • Encourage your staff to join the associations in the industry sector they work in – if it’s something like the MPG encourage them to go for board positions and also make them aware of associations specifically for women out there. • Education – go and speak at your local school/college and tell them about what a great industry we work in and how it is open to everyone. We must encourage more females to join the industry and this is the only way to do it. I hope that these few ideas will make a difference and if you would like to get more involved then please contact me on sadie.groom@bubblesqueak.agency Sadie Groom is managing director at Bubble & Squeak, a PR, marketing and events agency specialising in broadcasting, entertainment technologies, AV and business services.
GEAR OF THE YEAR 2016
SPOILT FOR CHOICE
Compiling a list of the year’s standout launches has always been a difficult job, and 2016 was no exception… es, it’s that time of the year already, where we look back on the gear releases that have really caught our attention over the past 12
Y
months. Once again 2016 has presented the AMI team – and our team of expert reviewers, of course – with plenty of shiny new kit to check out, and after receiving a high volume of entries and
much deliberation, we’ve now managed to put together our top picks. The following products have impressed for a number of different reasons – whether it’s innovation,
functionality, cosmetic reasons, value or all of these things combined – so let us know what you think of our selections via the Twitter hashtag
#GearoftheYear.
MAGIC MICS
AUDIO-TECHNICA ATM-350a The ATM350a cardioid condenser instrument microphone comprises six specially designed systems that provide discreet mounting solutions for a host of instruments, including woodwind, strings, brass, percussion, drums and piano. It is capable of handling 159dB max SPL – a 10dB increase over its predecessor – allowing the mic to cope with an even greater range of powerful, dynamic sound sources. The bottom line: “With so many mounting options available with this product, the ATM350a scores top marks for versatility, which is sure to please both live and studio engineers.” Audio Media International
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SHURE KSM8
The KSM8 is the world’s first dual-diaphragm dynamic handheld microphone. Made for live sound situations where vocal clarity and sound quality are critical, it has the ability to adapt to changing environments without impacting sound performance. The mic virtually eliminates the need for EQ and processing and it also offers the purest cardioid pattern that Shure
has developed to date, providing consistent on-axis performance regardless of the talent’s microphone technique. The bottom line: “Shure has put a lot of very clever engineering and its many years of experience into creating what could be the ultimate dynamic vocal microphone.” Andy Coules, live sound engineer
www.audiomediainternational.com
GEAR OF THE YEAR 2016
SOFT SPOTS
IZOTOPE Neutron Advanced
MCDSP 6050 Ultimate Channel Strip
iZotope Neutron Advanced was developed to help musicians, producers, engineers and postproduction professionals achieve ‘unprecedented focus and clarity’ with their mixes. With powerful new features like Track Assistant and Masking Meter, it can be used to quickly reach an optimal starting point, identify perceptual frequency
Offering more than 25 modules, including gates, expanders, signal saturators and specialised filters, the 6050 also features all the modules from the 6020 Ultimate EQ and 6030 Ultimate Compressor plug-ins. Each new module draws on a variety of designs over the last four decades. Designed to ensure simple and intuitive operation, the 6050 Ultimate
collisions and fully wield creative control over mix decisions. The bottom line: “Neutron is incredibly intelligent and quick to use, it really enhances my skill set and will do the same to anyone who uses it, regardless of their level of experience.” Rik Simpson, producer (Coldplay)
Channel Strip has input and output stages, based around a trio of module bays into which any of the modules can be inserted. The bottom line: “I found the more I used the 6050 the more it started appearing on every channel of my mix.” Alan Branch, freelance engineer/producer
STEINBERG WaveLab Pro 9 The latest update for the award-winning editing/mastering software introduces an overhauled single-window user interface with new ribbon tabs and a multi-monitor-friendly window docking system. The redesigned Master Section provides 12 insert slots – including the Resampler – as well as flexible channel processing and new metering/ monitoring, now with M/S support. Its projects can be managed easier with File Groups, allowing single audio files to be grouped individually. The bottom line: “No one has doubted the power of WaveLab and with version 9 this power is harnessed to a wonderfully effective interface.” Alistair McGhee, broadcast audio engineer
December 2016
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GEAR OF THE YEAR 2016 POWER TOOLS
PIONEER PRO AUDIO XPRS Series
PRESONUS ULT-series
The compact, portable, woodenenclosure XPRS Series brings together Pioneer Pro Audio sound engineering with Powersoft Class D amplifiers, resulting in a versatile plug-and-play system. The range comprises the XPRS15 and XPRS12 two-way full range speakers, along with the XPRS215S dual 15in subwoofer. The XPRS15/12 offers a 15/12in ferrite LF driver, a 1.75in titanium diaphragm compression driver and Pioneer’s AFAST technology, which helps to provide a clear, clean sound with a deep bass range.
According to PreSonus, the ULTseries active loudspeakers combine the widest horizontal dispersion of any loudspeaker in their class (110°) with a focused vertical dispersion (50°) for an ultra-long throw. Featuring a custom low-frequency driver, PreSonus’ rotatable Pivot X110 horn and 1,300W (peak) of Class D power in a lightweight design, ULT12 (12in woofer) and ULT15 (15in) full-range loudspeakers also promise first-rate speech intelligibility and natural music reproduction for mobile use and permanent installations.
The bottom line: “Ticks a lot of boxes for this kind of range, combining high sound quality, durability and easeof-use for a variety of live and installed applications.” Audio Media International
The bottom line: “If it’s power, clarity and even, far-reaching coverage that you’re after then the ULT-series is well worth a look.” Audio Media International
PREAMP PICKS
FOCUSRITE Clarett OctoPre Designed to integrate with the company’s Clarett range of audio interfaces and other units that include ADAT optical I/O, Focusrite’s next generation Octopre enables the rapid addition of eight mic/line inputs and line outputs. It comes with eight ‘Air’–enabled mic pres with 24-bit, 192kHz A-D conversion – the ‘Air’ effect emulates the transformer–
RADIAL ENGINEERING Jensen Twin-Servo 500 based sound of Focusrite’s ISA range of heritage mic pres. The bottom line: “When you consider how expensive branded preamps can be, even without A-D converters, the price point of this Clarett OctoPre is incredible value for money.” Simon Allen, freelance mix and FOH engineer
A new 500 series version of the original Twin-Servo that can deliver up to 60dB of gain. It features a front-panel Neutrik Combo connector with choice of XLR and 0.25in inputs, a dual-gang potentiometer for level adjustment along with a ten-segment LED meter. This is supplemented with a 180° polarity reverse switch that toggles the absolute phase, a gentle high-pass filter to eliminate excessive resonance, a -15dB pad and a 48V phantom power activation switch. The bottom line: “Some consider the Twin-Servo one of the finest
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mic preamps ever made, and so making it available in a 500 series format was a smart move.” Audio Media International
www.audiomediainternational.com
ALL-ROUNDERS
STUDIO SOLUTION
CEDAR AUDIO DNS 2
The DNS 2 is described as the world’s first small, light, portable dialogue noise suppressor with a new DNS algorithm that’s ‘the best yet.’ All users need to do is switch on the Learn function (which tracks the unwanted noise as it changes) and then decide how much noise attenuation is desired. With 12V power, camera mounting points, analogue and digital I/O, phantom power and high-quality
GEAR OF THE YEAR 2016
POP AUDIO Pop Filter
mic preamps, the DNS 2 is ideal for location recording, live broadcasting, confidence checking for large productions and even live sound in various venues. The bottom line: “A compact and straightforward device for wherever background noise is an issue that needs addressing.” Audio Media International
DYNAUDIO LYD Series The LYD Series is a new range of personal reference monitors with low volume precision designed for near-field listening in a variety of recording environments. To offer choices and opportunities for musicians, producers and artists, advanced DSP with two tuning presets offers options for different setups, sound balance filters for adjusting the frequency response and bass extension for moving the bass cut-off to the preferred position. The bottom line: “Put a smile on my face with its lively and informative presentation, which I’d have no difficulty recording and mixing effectively with.” Nigel Palmer, owner of Lowland Masters
Pop Audio’s take on a pop filter for the modern recording studio features a quick-release butterfly clamp that it says is simple and quick to fix to a mic stand without fear of overtightening and/or causing damage to the stand. It comes with a choice of three acoustically transparent filters and a ball and socket arm instead of a traditional gooseneck, allowing users to position precisely and accurately, knowing the mic won’t start to droop down. The bottom line: “Quite frankly I don’t think there is any other pop filter worth considering. This is the one you should buy.” Simon Allen, freelance mix and FOH engineer
December 2016
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BROADCAST FOCUS
www.audiomediainternational.com
EVERY LITTLE HELPS
Skipper of the good ship Fylde Coast Radio tells Colby Ramsey why he and his crew of volunteers need relinquished audio gear to realise the station’s full potential in its new premises.
Fylde Coast Radio is currently accepting donations of audio equipment from locations nationwide. If readers would like to donate any gear to the cause, they can get in touch with station manager Don Keady via don@fyldecoastradio.org or by calling 07931 014665.
ylde Coast Radio is a small, community-oriented volunteer radio station that has been servicing the North-West Lancashire area for around two years, originally established with the intention of getting coverage for Blackpool FC games via web streaming. After working out of a local music school for some time, station manager Don Keady and the team of multi-talented presenters set about locating a studio to expand their rapidly growing operation. Keady, who was heavily involved in one of Australia’s biggest community radio stations for some time, has seen it go from strength to strength, and is proud of what he says is now ‘officially the fastest growing radio station in the country’ due to its impressive listener numbers. “Everyone mucks in and it all works,” explains Keady. “We have a very full-on programme but sometimes there is no way for us to prerecord if people are away.”
F
In limbo Despite its recent success, however, Fylde Coast Radio survives entirely on handouts and contributions from the community, meaning future expansion has become a particularly difficult task. After exhausting the search for a new base of operations, local working men’s club The Marton 24
December 2016
Institute provided FC Radio with an unlikely solution – an old sports pavilion in a field at the back of a bowling green, from which it now operates. Deciding to keep their existing studio, the team immediately set out making the new space fit for operation. “We run it as a business but with a 99% community slant – there’s definitely a family feel in terms of how things are run here,” says Keady. “Everyone has mucked in to refurbish the place and now it looks really good.” While the station has taken advantage of some impressive sponsorship opportunities and some incredible good luck with the calibre of presenters that it has managed to get on board, Fylde Coast Radio continues to rely solely on charitable gifts and cast off equipment. Unfortunately, the attainment of audio gear is one area in which they have not had much luck. Despite owning everything needed for a radio station to operate, the current equipment is old and out-of-date, and so the team have tirelessly appealed to any generous individuals across the length and breadth of the UK to donate equipment to the cause, no matter how large or small. At the moment, an adapted Behringer mixer that was originally designed for a live band is being used for day-to-day
operation along with two KRK monitors taken from the DJ’s stereo system for foldback, “so we could really do with a radio desk of some description, along with some recording equipment, especially microphones”, remarks Keady. “We’ve got an old electric keyboard, a drum kit and a lot of old analogue equipment, and we still manage to keep going. When you’re working at a community station all the tiny things add up.” The next step on the road to expansion would be to get some new instruments for the studios, and Keady is confident that one of FC Radio’s 40 volunteers would be prepared to travel “from John O’Groats to Land’s End” to pick up anything from a guitar or saxophone to a single microphone.
Leaving a legacy In the new premises, Keady plans to transform the away changing rooms into a recording studio while the home changing room is set to become a broadcast studio, with the central area to be used for bands and live recording space. “We’d like to get the studios ready to use them as rehearsal rooms for kids,” Keady explains. “We’re looking at training them in both the new recording suite and broadcast suites and so we are also in desperate need of some telescopic mic
holders and all those accessory bits. If some kind people were to offer us any decent gear, our people would be straight in their cars to go and get it.” There are also plans in place to create a national radio service aimed at young people called Tangerine Radio, as well as a female-focused station and another aimed at UK businesses, which Keady says will be operated out of the new broadcast suite. While FC Radio’s widespread public appeal for unwanted instruments and audio equipment remains its prerequisite for expansion, it already has a lot to be positive about when it comes to its strong foundations and plans for the future. The station’s popular blues programme is close to being syndicated with what Keady describes as an “uncannily great” blues DJ, so much so that it is now in talks with the local council about putting on a blues and jazz festival in the Blackpool area. His overall vision however is to deliver and leave the legacy of a genuine community radio station in the area. “We have two exceedingly good mentors and a great management team; our broadcast and recording engineer Ivor Smith is a fountain of wisdom while our studio manager Tony Scarborough is remarkably good at coaching people, and we’re always looking to use tomorrow’s tools to run today’s business,” says Keady. “I think real community radio is not truly appreciated in the UK, but we blend the best bits of both,” he concludes. “It is nice and community oriented in terms of the way we run, but underneath everything is commercially acuitive.” www.fyldecoastradio.org
official audio partner of
MAGDALENA | BERLIN
GS-WAVE SERIES Bringing energy back to a former power plant in Berlin, the phenomenal sound system at techno club Magdalena comprises eight 3-metre GS-WAVE stacks plus extra subs and lenses, and eight tweeter pods above the dance oor. This huge installation, driven by Powersoft’s K Series amps with built-in DSP, provides total coverage of a modestly sized 800-capacity room. Keeping most of its power in reserve, the system can run effortlessly when the club is open for days at a time.
Pioneerproaudio | pioneerproaudio.com | #madeintheuk
FEATURE: GAME AUDIO
www.audiomediainternational.com
Picture: Battlefield 1/EA DICE
ON THE FRONTLINE
///////////////////////////////////////////////////////////////// The audio team behind the new smash hit shooter Battlefield 1 reveal their techniques for recreating sounds from over a century ago. ideo game developer EA DICE has spent a lot of time in Battlefield games making the world sound as realistic as possible, not only with guns, vehicles and explosions near and far, but also with environmental sound design. Battlefield 1, the latest in the series, required the DICE team to go right back to World War I and needed to sound different to its predecessor Battlefield 4, providing already familiar fans with a new experience. Since there aren’t many games set during this time period, the team almost had complete freedom to set the scene without too many strong preconceptions from the audience of how it should sound, which EA DICE audio director Bence Pajor found “exhilarating and scary at the same time”. “We felt that we wanted to juxtapose the leap backwards in time by doing something that felt very fresh and contemporary,” says Pajor. “World War I through a modern lens you could say.” The sound for the game was different in the sense that sounds of WWI are harder to come by and find reference for, posing a challenge for the team, who
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spent a lot of time tracking down vehicles and weapons from the era and recording them. The sound of the game itself also needed to change and bring the player closer to the action on screen, as lead sound designer Mari Saastamoinen Minto explains: “We wanted it to feel more gritty and brutal, with dust falling down on your clothes and helmet, the taste of dirt, mud and hot metal while the rain falls on your head as you crouch for cover in a trench.”
Sound all around Yet the strength of the EA DICE team lies in how it handled the unexpectedness and chaotic nature of this non-linear experience, unlike a film where every moment can be orchestrated perfectly for the audience, “so our task was much like letting 64 random people play whatever instrument they want simultaneously, and still having to make it sound good”, reveals Pajor. The instruments in this case were machine guns, cannons, tanks, planes, rifles, grenades, exploding buildings, screaming soldiers, barking dogs, revving motorcycles, whizzing bullets etc. One of the rule sets that the EA DICE team invented many years ago to deal with some of the most complex elements
is something they call “HDR Audio”, which in its simplest form tells a sound how loud it can play, if at all, depending on how loudly other sounds are playing at the same time. “This way we can play fewer sounds simultaneously making the mix a lot clearer, almost like a real-time film mixer,” Pajor adds. “This also helps to tell the story of some sounds being much louder than others since we don’t have the extreme dynamic range of the reality that we’re trying to portray, from a falling leaf to a 500lb bomb.” A new system that was introduced in the game however was dynamic weather, which posed a challenge not only from a technical standpoint but also in terms of sound design and time management. “The forest around you, the field ahead of you, the trenches in No Man’s Land and the airship you jump off – all these environments sound different depending on the time of day, year, temperature and weather,” Minto describes.
Banding together Strong collaboration between design, audio, lighting and VFX was key in creating a strong focus on changing weather, as the
team felt that it would have an interesting impact on gameplay. This required the creation of a lot more content but within the same timeframe as previous titles. “Good team spirit and communications is the base, followed by being able to work cleverly and set deadlines for yourself as a sound designer,” states Minto. “When it came to the sound design between the different weathers we set up some basic guidelines and went back and forth a few times before we decided on a design and method.” A range of microphones and recording devices were used for different types of sessions during the project. Sound Devices recorders have remained a staple in the DICE team’s arsenal for many years, according to lead sound designer Andreas Almstrom. “We also use handhelds such as the Olympus LS-5/10/11/100, Zoom H4n, Tascam DR-05, Sony M10/D100 for quick and dirty recordings, and for covering vaster areas during larger shoots,” he says. “Since they are usually fairly small we can put them in the line of fire to capture the sound of bullets and other projectiles whirring by.” For character Foley, the team mostly used the Sennheiser 418-s stereo shotgun rigged close to the chest, pointed downwards to get the sound of the shoes and other materials at a natural distance when running around. “We did experiment with binaural recordings on Foley and voice early during development,” Almstrom adds, “but we found it difficult to mix binaural recordings with non binaural recordings and still maintain clear spatiality.” For larger sessions such as vehicle and gun shoots, the DICE team used as many mics as they could handle, including Sennheiser 8040s, Sennheiser 416 and 418s, LineAudio CM3s, SM57s and a Sanken CSS5. For more specific recordings, Almstrom says that the AKG D112 and DPA’s d:screet 4062 were great for close-ups on weapons and vehicles. A mic that still follows Almstrom along on all shoots however is the Sony ECM-MS907, which, “while the mic itself isn’t all that spectacular, its form factor and inexpensiveness allows you to easily attach it to places where no other mic would be considered”. www.dice.se
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From the Sennheiser Professional Wireless Range. No compromise in performance. Whatever you’ve experienced before – expect more, even under the toughest RF conditions. The outstanding audio quality, rock solid RF wireless transmission and proprietary audio codec from Sennheiser’s wireless masterpiece, the Digital 9000, are now available for the most demanding live productions with the Digital 6000. No compromise for your business. Digital 6000 is ready to be used with a wide range of Sennheiser and Neumann capsules and antenna solutions, increasing the return on your existing investments. Made in Germany, the two-channel digital wireless receiver features an ultra-wide 244 MHz switching bandwidth and a wide range of connection options, making Digital 6000 a smart, reliable choice for years of professional use. No compromise in your workflow. Digital 6000 integrates seamlessly into digital or analogue system infrastructures via Dante, AES-3 or transformerbalanced analogue outputs. Security is ensured by AES 256 encryption. An automatic frequency setup function and dedicated management software (WSM) make for simple, reliable operation.
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COMPANY PROFILE
www.audiomediainternational.com
PROFESSIONAL HELP
Sound editor/designer Milos Stojanovic talks to us about his new talent agency that exclusively serves the audio post-production domain. espite the sheer volume of talent agencies based in London but operating internationally, it is difficult to find one that matches Sound Professionals’ proposition. While the new collective specialises in providing a full range of sound postproduction services for film and TV including consultations, Milos Stojanovic aims to uphold it as an alternative to already saturated and competitive networks in the sector, and hopes it will become the first port of call for anyone seeking out audio professionals in a multitude of capacities.
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What were your original motivations for starting the company? The idea to start Sound Professionals came about a couple of years ago when my colleagues and I started to notice an apparent shift in content production and the way freelancers get work. The rapidly developing technology and increasing affordability of equipment nowadays mean more and more people can shoot, edit and publish material. Although this is a good thing which means there is now more content being created then ever before, perhaps it shouldn’t be a surprise that there are increasing variations in quality not only online but in the broadcast world as well. Together with the fact that sound often serves as the last link in the postproduction chain and is considered very late in the process, it becomes easier to understand why there is often much room for improvement. Like myself, a large number of my colleagues come from the world of feature film and TV audio where budgets normally allow a group of 3-10 people editing and mixing for weeks if not months to deliver a 1-2 hour soundtrack. However even if you are lucky enough to be within a small group of people doing our type of work you are very likely to come across weeks if not months 28
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without a project in sight. So we started to ask ourselves how could we work with our colleagues and support them in their career paths while attracting new clients in need of professional sound services. Do you think it’s harder for freelance audio professionals to find regular work then? Being a freelancer I always felt that audio professionals were underrepresented and finding the next job usually comes down to good oldfashioned word of mouth. Even though there are many websites and talent agencies out there the fact is that very few list audio professionals at all. For many of us finding the next project becomes a job in its own right. That’s why my colleagues and I decided to start Sound Professionals, a collective and talent agency specialising in sound postproduction for Film and TV with the idea of being a professional alternative to low/no pay work sites and international networks of professionals competing with their hourly rates. Could you describe the structure of the company and tell us how it operates? At our core we have five founder members who have all been involved in high-end film and television programmes. They are: Max Hoskins (dialogue editor/sound supervisor), sound designers: Nikola Medic and Ruben Agguire Barba, dubbing mixer George Pinnock and myself. We each have eight or more years of experience in sound post-production while Max Hoskins has been in the business for nearly 30 years and our combined list of credits includes critically acclaimed productions such as Jason Bourne, London Fields and Da Vinci’s Demons. Our approach is to work closely with our clients to deliver a solution that best suits the needs of their individual projects/productions and the budgets available. So a client can get in touch to
discuss an upcoming project or upload their video and get a free consultation with a few pricing options in return. All our members have professional equipment and are able to work remotely in stereo or multichannel setups. We have also teamed up with a number of sound post houses to facilitate our clients during spotting and mixing sessions as well as Foley and ADR/VO recording. This enables us to achieve very competitive prices while still ensuring all our members get industry standard rates. Other than the fact you’re audio-only, what else makes you unique? What sets us apart from similar companies is our flexible business approach. Not being tied to leasing a property for our own studios enables us to keep overheads to a minimum and focus on specific needs of each project. As an agency that specialises in sound, only we have experienced sound editors and mixers in charge of each booking. This way we ensure the right person is assigned to a project based on their experience and area of expertise.
We offer a full range of sound post-production services as well as consultations and our clients can choose anything from a single task such as dialogue editing to a full package deal including a dedicated sound supervisor. In terms of memberships we are open for new applications and will always be on the lookout for creative and likeminded individuals. What are your plans for the future? Going forward we are hoping to become the go-to place for anyone looking for audio professionals. This could be anything from hiring freelancers to helping with busy post schedules, entire sound post teams or recruiting for a full time position. At the same time we will be helping our colleagues achieve their career goals. We also believe we could prove to be an important partner to production companies, agencies and video post houses who would benefit from the pool of creative talent we offer and our flexible working approach. www.soundprofessionals.co.uk
M-5000 & M-5000C LIVE MIXING CONSOLES
A NEW ERA IN LIVE MIXING CONSOLES MAD I In a rapidly changing world, the ability to adapt is needed to excel. OHRCA brings the power of adaptability to the world of live audio mixing. It conforms to the needs of both the application and the operator by delivering 128 freely definable audio paths, flexible user interface and workflow, expandable protocols, and multi-format I/O choices -all delivered at a pristine 24-bit / 96kHz sound quality.
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TECHNOLOGY: HOW TO
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SETTING UP YOUR STUDIO
David Shevyn, general manager of GIK Acoustics, gives his top tips for preparing a control room environment. Crooks Hall Studio in Suffolk – one of GIK’s previous projects
aking the time to set up your control room correctly makes a world of difference and helps to create the perfect control room. What people often miss is the fact that picking the right seating position, the right room and the right furniture are all critical to that listening environment. If you are lucky enough to be building your own room, it is generally agreed that using a guide like the Sepmeyer Ratio will give the most even response throughout the room. You also need to carefully consider the positioning of any doors, windows and air conditioning units. Ideally, a door would be to the rear of a room – away from a corner – you would not have any windows in your first reflection points and air conditioning
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units would not be placed in a corner, thus restricting any treatment that could be placed there at later stages.
Don’t be a square Advising on the room itself can be tricky. In general we would always choose a larger room over a smaller room and avoid, if at all possible, square rooms. Axial Modes Based on the length, width and height of your room.
Low frequency sounds tend to spread out and permeate the room. When the wavelength of a sound is the same as a room dimension, an effect called a room mode will occur (when the volume is high enough to excite it) and this will happen when you play or listen to music due to the natural resonance of the room itself.
There are three types of room mode: Axial, Tangential and Oblique. Room modes create a series of peaks and dips throughout the room – areas where the volume is actually higher or lower than that coming out of the speakers. In square or cube-shaped rooms where the room dimensions have common denominators this is amplified even further.
Tangential Modes Based on the dimensions of four room surfaces.
Oblique Modes Based on the dimensions of all six room surfaces.
TECHNOLOGY: HOW TO Testing, Testing There are four concepts to really consider here. For clarity, I am defining the front wall as being the wall you are facing when seated at the mix position and the rear/ back wall as the one behind you. However, not everyone has the obvious set up position in their room. Maybe the room is an L-shape, is square or has several choices – none of which are obvious. Firstly, you need to see in each of the possible set up positions how many of the following four points can be properly covered and secondly you need to test and test again. To do this you need an omnidirectional mic and software like the Room EQ Wizard. Interpreting the results is an art in itself, and users should take advice from forums and reputable companies with regards to this. 1. A lot of room mode issues are created by the back wall (the length modes) so ideally the back wall is as far away as can be. You also need to be centred on that wall if possible.
2. Look to be sat at 38% off the front wall (this is a good starting point as this is where the nulls and peaks affected by the length modes are the weakest). This is only a guide and in fact you can also apply the same theory to sitting off the back wall. Experiment, move a few inches forward and back as there may be many other factors in your room which affect the seating position, i.e. placement of doors, windows, alcoves. 3. SBIR (Speaker Boundary Interference Response). This is how the speakers respond to the environment around them. We will not go into detail here, however, as this was covered by Carl Tatz in the May 2015 issue of Audio Media International. 4. Symmetry – In an ideal scenario we would have symmetry in the listening position and the materials on the walls would be made from the same material, but this is very rarely the case. Lack of symmetry can create low-end issues, lack of stereo imaging and comb filtering issues.
other things to consider • Alcoves, bay windows, chimney breasts. A lot of rooms have nooks and crannies – avoid placing any of your speakers or desk position within them. If you have alcoves either side of the chimney breast, ensure the speakers are in front of the chimney breast. Similarly, with bay windows, if facing into the window, ensure you are not actually sat in the bay window area. • Sloped ceilings. If the ceiling is sloped in a V shape then you want to most likely sit in the V part. If the ceiling is just sloped one way then you sit under the low part. That will push the early reflection behind the listen position. • Desk size. One of the most common problems we come across in studios is early reflections caused by the mixing desk itself. It is advisable not to use a desk bigger than what is necessary for what you need. It seems like simple advice but a large desk with lots of reflective surfaces creates a lot of early reflections – often seen on an Energy Time Curve graph at 5ms and less – and this can be incredibly difficult to solve.
• Placement of racks, synth racks and other essential equipment. Try not to place any of this equipment in areas that are essential to treat, like corners or the early reflection points. • Client seating positions. If you have a separate playback area, or a sofa for clients to listen from then they are not going to be in the best spot in the room to hear your work. You have two choices: either sit the client in your seat (after all, yours is the best seat in the house and has been set up for the perfect listening experience) or treat the secondary position, making sure the reflection points are covered and also tested from here, effectively creating two sweet spots in the room. www.gikacoustics.co.uk David Shevyn is general manager of GIK Acoustics Europe, which manufactures a complete line of acoustic treatment products for various environments, and also provides professional design consultation services.
Reinventing the large sound system
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ROLAND M-5000C
DIGITAL MIXING CONSOLE
rom an end-user’s point of view, the digital console market is really exciting today. While technology follows its natural progression, thankfully prices have fallen considerably in this sector. This has led to the market moving extremely quickly and now many of the low to mid-priced consoles have feature sets compared to those that would’ve cost tens of thousands of pounds just a few years back. As a result, manufacturers having to up their game while keeping costs to a minimum must be quite a challenge. Of course, less moving parts and fewer analogue electronics are required, which keeps costs down, but a lot of research and development must now be poured into software programming and how that will interface with the user. Touchscreens aren’t exactly cheap either, and the M-5000 from
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Simon Allen heads out on tour with the more compact model in the range to see how it performs in a busy live setting, and whether it could give the major players in this sector a run for their money.
Roland Pro A/V has quite a large but highly detailed one.
Overview When introduced to the M-5000 for the first time you are greeted with the acronym ‘OHRCA’, which stands for ‘Open High Resolution Configurable Architecture’ and highlights a few principles that the console is built upon. ‘Open’ alludes to the variety of input and output formats that it can operate with. Roland is known for its own ‘REAC’ protocol that all of the brand’s consoles and digital stage boxes use, and it was one of the first manufacturers to develop the digital snake concept. As well as this connectivity, however, there are two expansion card slots, which enable the use of other protocols such as Dante, MADI and even Waves SoundGrid. This is great news of course – and
something I made good use of during my review – but it has almost become an expectation for modern consoles. Even without any option cards installed, though, the M-5000 is very powerful. There are two main REAC connections for stage boxes, and a third for hooking up to a monitor console, other monitoring options or for redundancy. There’s also a 16 x 16 USB audio interface for laptop recording and playback. The desk has substantial local I/O, which is great to see as some modern controllers can feel limited. The M5000 has 16 XLR analogue inputs and 16 XLR analogue outputs, and the M-5000C, which I used for this review, has just eight outputs. To be honest, besides this small difference in local I/O and eight fewer physical faders on the M-5000C, that’s where the differences end. The processing power is exactly the same. This is quite surprising as the frame
Key Features 128 freely definable audio paths for building a console structure to suit the chosen application Expansion options include Dante, MADI, REAC Waves SoundGrid, XI-SDI, XI-SFP and XI-DVI High sampling rate of 96kHz 28 faders in four banks (M-5000) or 20 faders in three banks (M-5000C) 16 x 16 USB audio interface for 16-channel recording and playback using a Digital Audio Workstation Up to 300 inputs and 296 outputs (and more at 48kHz) are managed in separate patchbays 16 encoders with rings that change colour according to their assigned functions to match on-screen parameters Decdicated app supports remote control from an iPad, as well as Retina displays RRP: £16,299 proav.roland.com
TECHNOLOGY REVIEW size of the M-5000 is much larger and yet I managed to get the M-5000C, plus it’s very substantial flight case, into the boot of my estate car. Next up in OHRCA is ‘High Resolution’. This refers to everything
console. Unfortunately, I think this is a message the industry has missed. ‘Configurable Architecture’ doesn’t stop there either. One of the M-5000 party pieces is the fact you can change the configuration mid-show. If you need
channels soon ended up with around 64 signal patches. Let me just state for my own sanity that I’m not one to over-complicate a setup, but once this was configured for the tour it made everyone’s lives extremely easy.
virtual soundcheck input. Virtual sound checking was really useful on this tour, not just for me but also for the band to make sure they were happy with their own IEM mixes, giving them a chance to work on their personal mixes with
in the M-5000 system running at a 96kHz sampling rate and 24-bit, end to end. All of the Roland stage boxes have been capable of running at 96kHz, but the M-5000 defaults to this. All the processing inside the mixer is also carried out at 96kHz, with a summing mixer running at 72-bit. If you do the maths, that’s a tremendous amount of headroom. Many consoles are now running at 96kHz of course, but the big news here is that there aren’t many offering that, along with this channel count, in this price range. What is the channel count I hear you ask? Well the final part of OHRCA is ‘Configurable Architecture’. Both M-5000 systems offer a total of 128 freely assignable audio paths. From this pool you can build the mixer however you like – you can choose any number of input channels, auxiliaries, groups, matrices and mains. There are also downmix and mix-minus busses, which are features typically only found on dedicated broadcast consoles. Roland supplies a few templates to get you started, but the M-5000 shouldn’t really be considered just as a live sound
to add channels, or swap some of your system resources from auxiliaries to groups at a festival, for example, you can be doing that during the previous act. There isn’t any drop in audio and you only lose control of the mix for two seconds. Again, there are now consoles that can do this too, but typically not in this price range and quite so flexibly.
Firstly I soft patched a Y-split of the input channels onto the desk so I could operate a FOH mix independently to the channel settings going to the band’s IEMs. We then ran direct outputs post channel processing to the KLANG system via Dante. The post-gain direct outputs of the FOH channels where then also routed via the Dante network for multichannel recording onto Pro Tools. Why is all this relevant to reviewing the console? Well, there were a number of ways in which the M-5000C facilitated this in some surprisingly efficient ways. Firstly, the M-5000 offers three inputs for each input channel. There’s obviously the main preamp, in this case the preamps on an S-2416 stage box and an alternative input, which could be used with spare wireless microphones for example. This saves copy-pasting settings to a different channel when the spare mic is needed, so all that’s required is just swapping the input source. The third input acts like a tape machine playback input, which in the modern live sound world we refer to as the
everyone playing. The M-5000C made this extremely easy compared to other similarly-priced consoles that still require manual patching. Another benefit the M-5000C brought to this configuration was virtual patching of inputs directly to outputs. The six stereo outputs from the KLANG were patched back to the console via Dante, where the console then routed these Dante ‘inputs’ to physical outputs on the stage box. This enabled me to retain the KLANG unit at the FOH position as there wasn’t a monitor desk, but the wireless transmitters could still be patched on the stage. There are a few key points to note here. Firstly, the M-5000C facilitated all this patching with ease, between what was soon becoming a notable Dante network and its own REAC connections. The KLANG’s Dante outputs that were routed to the REAC stage box didn’t add to the total of 128 assignable audio paths, therefore leaving the console with a huge amount of spare power. Finally, with a GB/sec network switch I could have all the
On the road I took the M-5000C on tour for a week with a band that I work with regularly. This provided me with material I knew well so I could therefore concentrate on the desk and of course the sound. To add to the technicalities, the band was going to be fully on stereo in-ear mixes. This was partly because we’d decided to and partly because I was also roadtesting the new KLANG:Fabrik 3D inear mixing system. We didn’t have the luxury of a monitor console and needed to arrive at each venue with six-way stereo IEM mixes ready to go. This meant I could really put the M-5000C through its paces. What may have appeared to be a simple input channel list of around 20
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Dante in and outs going to Pro Tools and KLANG then back to the stage box with zero complaints about latency. I was quite impressed by this because that’s quite a number of cross-platform digital patches with a lot of processing involved before you even add the latency of the digital wireless system itself. I should also add that the whole system was extremely reliable. Every night the system was powered up and we were ready to go, without any crashes or loss in audio. In terms of typical channel features and the general operation of the console, it’s really intuitive to use with pro features that sound great. Everything you would expect a serious mixer to offer is there and almost every feature is easily found. I have to confess however, that I did have to read the manual once to learn where the matrix mixer was. This might seem silly, but to access this window it is a separate item on the menu, which doesn’t follow how the rest of the console behaves. This is a small detail 34
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though and once I did find it, the matrix mixer was naturally comprehensive. Every manufacturer approaches channel navigation in a different way. Most digital consoles today can deal with many more multiples of channels than there are physical faders. Getting to what you need has to be fast and efficient. With the M-5000, Roland has introduced a truly fantastic feature called ‘Anchor Points’. Along with the usual scrolling and DCA spill techniques, users can set anchor points to any channel, then via two clicks can return to that point in the channel list very quickly. Sonically the whole system was extremely clean and handled transientrich content, such as drums, in a natural fashion. The 72-bit summing at 96kHz sounds fantastic and the basic channel tools such as EQ and compression are very powerful. Perhaps it would be nice to see a selection of high-pass filter curves on the EQ’s in a future update, but otherwise it’s excellent. I think I personally struggle with some of the
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Roland reverbs as I find them busy, but I think that can be put down to personal taste. The console’s effects section is extremely well loaded with many of their signature effects. Once I’d had time to play with the reverbs I was happy with the results. The Dynamic EQ, which I used as inserts over the vocals sounded great and was very useful.
Conclusion Unfortunately Roland faces a problem with this desk. When you think of a console of a similar spec, you don’t initially think of the brand. On top of that, the price could put some people off as they might assume it’s not up to the job. However, the truth is this is a very powerful and reliable system that should keep even the most experienced engineer smiling. The expansion cards and ‘Configurable Architecture’ of the desk should see it in service for many years to come. All-in-all the M-5000C was a clean and professional sounding
system which felt very natural to use. The feature set and some of the operational behaviours hark towards consoles that are several times more expensive than this. The M-5000 will facilitate a huge number of applications with enough power for some of the largest events where alternative solutions will mean a significant price increase. If you do get the chance to use one of these new consoles, don’t hesitate as I think you’ll be pleasantly surprised.
The Reviewer Simon Allen is a freelance internationally recognised engineer/producer and pro audio professional with over 15 years of experience. Working mostly in music, his reputation as a FOH and studio mix engineer continues to reach new heights.
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TECHNOLOGY REVIEW
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DRAWMER MC3.1 MONITOR CONTROLLER
Alistair McGhee takes a look at this new tool for ensuring faithfully reproduced recordings. grew up in monitor controller world in the days when Glensound outside broadcast desks were simple beasts and needed a DK2/20 or DK2/21 to add monitoring and control functions that made life that little bit easier. But time marched on and more functionality got built into location desks while the starlight of monitor controllers began to fade. Yet, the requirement for comprehensive control of what we’re listening to has never gone away and with the rise of DAW mixing, the need has probably grown a hundredfold. Drawmer has three products in this ballpark. The MC1.1 is a monitor preamp while the 2.1 and the new top-of-the-range MC3.1 have the full fat ‘monitor controller’ label.
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the Ins and outs So what do you get in your MC 3.1 tin? You can connect three pairs of speakers to the XLR outputs and one sub, which is a mono output. There are also two independent headphone amps switchable between the main and cue outputs. Talking of cue, there is an independent cue output on balanced jacks with its own volume control and independent access to all inputs. On the input side there are two un/balanced stereo inputs on combi XLRs, one stereo unbalanced input on phonos and one digital input on combi XLR, capable of taking AES or SPDIF digital signals up to 192kHz at 24 bits. These inputs and outputs are all on the back panel while there is a stereo AUX 3.5mm jack input on the front for extra convenience. Let’s take a quick break there to plug the beast up. Start simple by putting the balanced output of your desk into input one, grab an AES out from your DAW and stick it in the digi input. Now you can select between sources to your 36
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mix and/or cue output. The inputs are additive though, and of course buffered, so you can mix on the monitor input buttons if you wish. Now plug up your main monitors to speaker output A, your second favourites to speaker output B, and your checks, near-fields etc. to speaker output C and you’re ready to do some serious monitoring. But first, you need to line up your monitors properly like they do in big school. When I started out in studios, lesson number one was ‘first line up the monitoring’ – if you can’t rely on what you’re listening to, everything else is compromised. The initial procedure involves pink noise, a sound pressure level meter and the output trims for the speakers located on the base of the MC3.1. Check out the Drawmer website to download the line up tones. Now we’re ready to move on!
speak easy But what about the talent sat alone in a lonely room without any love? When plugged into the cue output the studio monitors have access to the cue bank of switching, and you can feed them whatever you want without changing your own monitor selection. If you’re feeling very communicative, talk to them using the built-in talkback mic or plug in an external mic or foot switch if you don’t want to reach for the button. So these are our inputs and outputs, but what else can we do with them? Here are a few of my favourites: Dim takes 20dB off your level. There’s a mono option available, so often overlooked in today’s multichannel world but something that really should be used. Alongside cut A
or cut B (both available on the MC3.1), you don’t really want stereo mono – the cut A/B options have a special effect on the headphone output to simulate their effect in a room. For total silence, there’s a mute switch, which does not apply to the headphone outputs. Beware though, these headphone outputs go loud! There’s also phase reverse and L/R swap. So what sets the MC3.1 apart in a world of ‘me too’ copycats? Well consider the Band Solo function. You have three frequency bands to choose from – low, mid and high. Hitting these solo buttons allows you to focus your listening on one (or more, the feature is also additive!) of these bands. The crossover points are at 250Hz and 3.5kHz and the slopes are 6dB per octave, and this is really about the discipline of listening. I’ve not seen this feature before and you might argue that we’ve mixed for years without it, but that is to miss the point. The more listening we do and the more focus we put on it then surely the better our mixes will be. A more familiar feature is the preset option on the Master Volume, where there is a select function that switches control from the master volume pot to an adjustable preset level. This is absolutely vital in comparative listening sessions, because what we often perceive as differences in quality can be traced to small changes in level and the ability to guarantee level matching is a must-have.
In Use Then I asked the question, ‘What about a stepped volume control then?’ It seems the available 24 stepped switches just
Key Features Three speakers plus a mono sub can be switched individually and simultaneously Two headphone amps with individual level controls and switching between Main and Cue inputs Cue level control adjust the volume for the artist’s monitors Ultra low noise and transparent circuit design Comprehensive mix checking facilities MSRP: $999 www.drawmer.com don’t offer fine enough control. Of course none of this would matter if the MC3.1 failed on sonics. I recently recorded a cut down version of local band Fountainhead for a Spendor speaker demo and used the 24/96KHz files to audition the controller. I drove the analogue input with the output of the Marenius S2 DAC. Not only did the analogue side of the circuitry sound transparent but the digital input was excellent. The MC3.1 is a high-quality device with an innovative array of features. Highly recommended.
The Reviewer Alistair McGhee began audio life in Hi-Fi before joining the BBC as an audio engineer. After 10 years in radio and TV, he moved to production. When BBC Choice started, he pioneered personal digital production in television. Most recently, Alistair was assistant editor, BBC Radio Wales and has been helping the UN with broadcast operations in Juba.
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TECHNOLOGY REVIEW
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GRACE DESIGN M108 MIC PREAMP/ADC
David Bowles finds out whether this new device deserves to be described as ‘a vital addition to any modern audio production environment.’ ’ve always been a fan of Grace Design mic preamps, and have used them often over the years for commercial projects. The company’s m801 eightchannel mic pres were already a part of many established recording studios; later it offered the m802 with remote control and an optional A/D module. With the m802, this still required one more unit to transport and set up for location recordings: a D/A to monitor DAW mixes. Many of the 1RU audio interfaces currently on the market only provide two to four channels of mic preamps, and some observed these preamps were not of the same quality as standalone mic pres. Units with more mic pres expanded to 2RU were quite expensive at the time they were released and the A/D quality varied considerably. My need was for a 1RU unit for simple recording projects and live concert recordings of up to eight channels, so I wouldn’t have to disconnect my 24-channel unit from my studio setup. Grace Designshas now fulfilled this niche with the m108, a 1RU audio interface with eight channels of highquality mic preamps, A/D conversion and two channels of D/A conversion for their headphone amplifier. Mic inputs are via XLR, with the first two channels switching automatically when high-impedance instrument inputs are plugged into the front of the unit. In the rear, mic pre outputs are mirrored via two DB25 outputs (for analogue and AES respectively), as well as two ADAT outputs. In normal operation, ADAT transmits eight channels of 44.1/48kHz audio out of each of these outputs; in SMUX mode (ADAT 88.2 and 96kHz) channels 1-4 are assigned to ADAT
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Key Features output 1 and 5-8 output 2. Otherwise, sampling rates go up to 192kHz, with extensive clocking options. Interfacing with a PC or Mac is done via USB2.
You’ve Got Options Furthermore, an option slot in the rear offers two added-cost choices. The first provides Dante AoIP (Audio over IP), with primary and secondary gigabit Ethernet outlets. The second option card is called Control Room Output – this provides two analogue TRS outputs for driving monitors or headphone distribution amps (this was not available at the time of writing). However, both options cannot be used together, nor can the DAW outputs be routed to the DB25 analogue output. I wanted to monitor 5.1 and stereo mixes from my DAW, but can only monitor a stereo feed. Perhaps this can be addressed in the future? The build quality is what one would expect from Grace Design: simple, functional and solid. The unit is very light in weight, compared to the m801/802. This makes for easy permanent mounting in catwalks and attic spaces. On the front of the unit are two large control knobs, with a wide OLED display window in between them. Separate buttons are clearly labelled for phantom power, polarity inversion, pan and setup. A headphone monitor mix can be created from the eight mic inputs – in this case the first three buttons function as mute, solo and pan respectively. The power switch is also on the front of the unit. While this is practical, this switch can be toggled too easily. I prefer recessed switches, which require more pressure to toggle on and off. The OLED display window is designed to be read from a distance, with large numerals on either side to indicate mic pre and headphone level. In between, a graphic display of each channel indicates
channel number, a level meter, icons for phantom power, Hi-Z input, clipping, polarity inversion, gain level and ribbon mode. On the right side of the window one can see clock rate, source and clock output. In monitor mix mode, one can see channel number/name, level, pan, solo/ mute, mix level, master level, monitor source and headphone source. In this mode, the user scrolls left to right, as there is more information than can be displayed in a single window. Finally, in Setup mode, there are 16 options, including display brightness/auto dim and even one’s choice of headphone level when starting up. Control of the unit can also be done via Ethernet, serial (RS422 or RS485, for Grace Design’s m802RCU remote control), and MIDI I/O. With Ethernet, one can use a Chrome, Safari or Firefox browser (after entering the IP address), or the included standalone Remote Control Application. Presets can be saved and recalled easily to one’s recording computer, which makes switching between projects easy and consistent. Wordclock I/O and autoswitching IEC inlet complete the wellequipped rear panel.
In Use Setting up the m108 was straightforward. On my recording laptop, I use static IP. Once I assigned an address to the m108 it was easy to ‘locate’ the unit, control the settings and save a preset. As the mic inputs are XLR, I didn’t need to pack a breakout loom. Over the next week, I recorded two chamber orchestras in concert at 192kHz – one performing on period instruments, and one on modern instruments. The concerts were done in the same two acoustic spaces in which I’ve recorded these groups for quite a few years.
Eight channels of ‘transparent, musical mic preamplification’ ‘State-of-the-art’ 192kHz A/D conversion Reference DAC and headphone amplifier Combination RS 485/422 serial/MIDI ports High contrast, wide viewing angle OLED display RRP: $2,845 www.gracedesign.com The quality of mic preamps and A/D alike matched my DAD AX32 (and exceeded that of the Prism Orpheus I had previously used as my ‘second rig’). The recordings were clear and detailed without being harsh (in spite of the dryness of both concert halls). The headphone amp drove my Sennheiser HD800’s easily. The next time I record these groups I’ll be ready to record much sooner, by recalling the saved presets. The only downside is that Pyramix 10.1.4 reports the latency rate to be more than double that of the RME MADIFaceXT. Perhaps this can be improved with firmware and ASIO driver updates?
Conclusion I’m happy to add the Grace m108 to my rig, not only for projects of up to eight channels, but to route the AES outputs to my DAD AX32 (thus adding another eight channels to that unit).
The Reviewer David Bowles is a freelance audio engineer specialising in recordings of acoustic music in surround-sound and 3D audio. He guest lectures at New York University’s Steinhardt School Tonmeister seminar, resides near San Francisco, and likes good wine and bad puns.
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TECHNOLOGY REVIEW
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AKG K872 HEADPHONES
Alex Goddard gets up close and personal with AKG’s new professional closed-back headphones. he AKG K872’s box is really rather large. So large, in fact, that while removing the outer casing I wondered if there might be two pairs of headphones hidden within. The internal packaging has a sumptuous suede texture – AKG makes opening these headphones feel special – and inside the flight case, the new closedback headphones are snugly protected. On first glance, the headphones themselves are luxurious and nearly big enough to justify the size of the box they come in. They are discernibly lighter than many other headphones of their size, weighing in at 390g minus the cable, which is heavily shielded and fitted with a LEMO connector – a nice touch. The K872 boasts an open-mesh headband with carved 3D-shaped slowretention earpads that feel superb, albeit a little bit loose-fitting even after adjusting for my head size. Underneath the earpads rest the drivers that are the heart and soul of this high-end model. These leverage the transducers used in AKG’s current K812 open-back model, each with a whopping 1.5 Tesla magnet and two-layer, copper-covered aluminium voice coil. Together, these features allow the frequency range to exceed human hearing, according to AKG, extending from 5Hz to 54kHz.
T
In Use In order to determine how well these headphones reproduce audio, I listened to a variety of tracks. First, I put on Wu-Tang Clan’s 36 Chambers. I notice immediately that the bass – the details of which are normally lost to most speakers and headphones – is clear and present. On the other end of the musical spectrum, I play through some Chopin, choosing a lossless recording 40
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of Op-64 No 2. This reveals one of the professional headphones’ best features – a wide dynamic range, and resultant depth of sound. Quieter sections feel intimate and nearby whereas louder sections are present but by no means overwhelming. This sound stage may not be quite as deep or intricate as some of its open-back counterparts but is nevertheless impressive. In my opinion, this tie-off is worth it for the added isolation that the closed-back design offers. The next stage is to test the K872s in an actual mixing environment. Over my many years of electronic music production, I’ve consistently preferred using reference monitors to headphones because they allow the sound to play into an open space. The best compliment I can give this set of cans is that I immediately forgot I was wearing them, and was deeply immersed into my work. Within a couple of hours, I had mixed down two tracks that had been escaping my Yamaha HS8 reference monitors for a couple of weeks, and with almost no ear-fatigue. The headphones allowed me to hear a close-up snapshot of the songs that the monitors just couldn’t expose. In one song, for instance, I was having an issue getting my bass to glue with my kick drum – the headphones quickly revealed an unwanted digital distortion in my sub which had eluded my ears. One adjustment of the EQ and distortion later, the song had found a new, tighter mixdown.
The Verdict Overall, these headphones are outstanding. The K872s reveal aspects of recordings and mixes that other cans and monitors might not – their low-end is detailed even in muddier recordings, the mid-
Key Features range is clear and perfect for vocal sounds, and the high-end is neutral rather than bright so that peaking frequencies are exposed straight away. A minor issue is that they are seemingly not as isolated from outside noise as other closed-back cans are. This is likely down to the 3D replaceable earpads and comfortable loose fit. Others may take issue with this compromise, but my own work is largely based in quiet studios, where I prefer a revealing dynamic range to perfect noise isolation. In comparison to my everyday-use headphones – Sennheiser’s HD25II – the K872s are a whole different ball game. I would certainly consider purchasing a pair just to support the aspects of my production that my reference monitors can’t reveal.
Closed-back design Custom 53mm drivers with 1.5 Tesla magnet systems 5Hz-54kHz frequency range Detachable cable with LEMO connector Replaceable 3D earpads RRP: €1,599 www.akg.com
The Reviewer Alex Goddard has been producing electronic music for the last decade. Based in London, he has performed the role of sound designer and operator for a number of theatre productions and sitespecific installation pieces.
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EMOTIONAL INTELLIGENCE
After being shown an impressive demonstration of the technology in London recently, Adam Savage followed up with Etienne Corteel, CTO of Sonic Emotion Labs, the company behind a new turnkey solution for 3D sound in various environments to get a more in-depth rundown. The transition between preparation and onsite operation is as simple as routing the output of the sequencer to an external soundcard to feed the Wave I.
Designed and manufactured in Paris, Wave I from Sonic Emotion Labs claims to offer a new way of introducing 3D listening within a venue using Wave Field Synthesis, delivering uniform spatial sound reinforcement for the entire audience and enabling users to easily and precisely position and move sound sources for added control via the software. Sounds interesting, no? Then let Etienne Corteel from the firm explain further… What are the standout features/selling points of the Wave I system? Wave I is a combination of post processing functionalities for optimum sound field rendering in an extended listening area. It combines multiple state-of-the-art functionalities (loudspeaker management, system tuning, matricing) with unique 3D rendering in one consistent product. What does Wave I consist of in terms of actual equipment? The hardware processor and two software interfaces? It comes with a complete software suite for system design, system tuning, real time performance and 3D sound content production. The tools are designed to work either online or offline. Everything can be prepared in advance including content production. We offer an offline rendering software that allows 3D rendering in any conventional loudspeaker layout (from 5.1 to 22.2, including all Auro3D configurations) or through headphones using binaural technology. 42
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What do you believe were the advantages of using Wave Field Synthesis (WFS) technology? WFS is a robust 3D sound technology that accommodates well with various room dimensions and shapes. It is also a power efficient technology that uses one quarter to half of the speakers for any target source position. Our WFS algorithms go beyond the traditional formulation that requires a large amount of loudspeakers and is usually restricted to the horizontal plane. We have allowed the use of a limited number of loudspeakers (from five on), the extension of WFS to 3D with a small number of loudspeakers for height, and the combination of multiple arrays for optimum sound coverage. Is this what allows you to eliminate the ‘sweet spot’ and focus the audience’s attention on the sound source (e.g. Musician) rather than the speakers? WFS is the only 3D sound technology that enables the user to create virtual sources at a defined position in space. All other technologies provide direction control but fail at creating a proper wave front emanating from a precise position, including distance. Not just at the location of the nearest speaker. For example, WFS allows you to position the sound at the exact location of the singer and provides a coherent audio/visual experience for any seat in the audience. What applications is it suitable for, and could you give us some examples of where the technology has already been used around the world? Wave I has been used in multiple contexts. It applies to spatial sound reinforcement maintaining sound level coverage while providing spatial positioning of sounds onstage. We have several fixed installations in Europe (e.g. Théâtre
de Chaillot, Institut du Monde Arabe, Stuttgart Schauspielhaus), more than 160 shows over four years in classical and jazz open-air festivals and an arena tour with French artist M Pokora. Wave I has also been installed in clubs, offering 3D sound rendering for DJs even with stereo inputs only. We have also multiple installations all around the world (art centres, museums and universities). How easy is it to install and is there a lot to learn in order to get started? Our tools have been designed in close connection with users and are recognised as self-explanatory. Creating content for Wave I can be learned in a couple of hours. Designing a setup is very simple, providing you have rough loudspeaker positioning information. The remaining aspects concerning system tuning primarily consist of applying loudspeaker presets (if available), aligning subwoofer level and adjusting classical parametric EQs to optimise sound performance. So there are lots of possibilities with loudspeaker configurations? And is the software compatible with all the major DAWs? Wave I can be used in many different loudspeaker configurations. All we recommend when designing a system and choosing loudspeakers is to ensure that any listener will be the in the field of at least three speakers in order to limit the audibility of individual
speakers. Therefore, we recommend using loudspeakers with wide horizontal acoustic dispersion and smooth directivity characteristics. In the session at Rambert [Ed: where the London demo took place] with composer Roberto Rusconi, we used DX12 coaxial speakers from APG that provide such characteristics. The setup may only cover the stage area, or comprise an additional surround system, support arrays and ceiling speakers for source positioning in height. And our software is compatible with all major DAWs including Ableton Live with a custom integration. Are there any future developments at the company that you can tell us about? Sonic Emotion Labs is the proud coordinator of the Edison 3D research project that is funded by the French national agency of research (2013-2017). We gather experts (signal processing, acoustics, human machine interface, psycho-acoustics) and expert users. The goal is to develop tools that offer a consistent 3D sound production workflow between the studio and the live situation. Radio France is a key partner in this project, already using our tools in multiple live and web-based productions. They have successfully created the “cinema for your hears” events in which the audience experiences 3D mixes that are then available as online binaural productions (http://nouvoson.radiofrance.fr/). www.sonicemotion.com