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International TECHNOLOGY AND TRENDS FOR THE PRO-AUDIO PROFESSIONAL www audiomediainternational com

May/June 2017 December 2016

SCARE TACTICS We get up close and personal with three professionals tasked with constructing an atmospheric score for Alien: Covenant p24




A trio of monitor manufacturers discuss driver design p18

A look around Distant City Studios p28

Simon Allen assesses iZotope RX 6 p34

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.


Experts in the issue

Adam Savage



Adam Daniel is an accomplished sound editor, rerecording mixer, and one half of Point1Post, which provides postproduction sound services.


Pete Malkin is a London-based sound designer for theatre, film and live performance.

Front cover image courtesy of Twentieth Century Fox

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Joss Worthington is a songwriter, composer, music producer and owner of Distant City Studios.

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ow pleasing was it to see so many people not just from within pro-audio but also the wider world of theatre welcome the recent news that the Tony Awards has decided to bring back its Sound Design categories. Although many, including myself, believe that it was a pretty shocking decision to do away with them in the first place, you also have to give some credit to the committee for making the right choice, even if it has taken a little longer than perhaps it should have done. Some rightly believe that there is a surplus of awards ceremonies in other industries, but in this sector of our business there aren’t that many opportunities for these individuals to get the recognition they deserve, especially when so much of the focus is on the actors’ performances, the glitzy costumes, the complex sets etc., which all deserve their own praise when also done exceptionally well, of course. It can also be challenging to get yourself noticed when you’re only a few years


into your career, but there are some who make it through, and one of these is Pete Malkin, who, along with his associate Gareth Fry, is set to be honoured with the 2017 Special Tony Award. Our interview with Malkin on Page 42 sees him provide us with a brief overview of what the new Harry Potter play has to offer – he’s sworn to secrecy, like everyone else associated with the show, you see – and he also gives us his take on the sound of The Encounter, which the Special Tony is for. You might remember our interview with Fry from last year that looked at the pair’s clever use of binaural, and I have to say it’s about time they were rewarded for their efforts. As for what else we’ve got in this issue, we were lucky enough to be invited to Peter Cobbin and Kirsty Whalley’s new studio to speak to them about scoring Alien: Covenant alongside music editor Tony Lewis and Ridley Scott. From what I saw/heard when I was given a sneak preview, fans of the series are in for a real treat. Then there’s our Studio Profile on Distant City Recording up in Yorkshire, which now boasts an impressively versatile drum room; we’ve already analysed the new version of iZotope RX and we got back in touch with Point1Post’s Adam Daniel to get his thoughts on the big news from Avid and Dolby. Anyway, I won’t keep you any longer. Go have a read!

Adam Savage Editor Audio Media International


Visit to find your local Neve dealer For recording as it’s meant to be heard, it has to be Neve - no question.

May/June 2017






Shure announces Axient Digital


Genelec unveils ‘Ultimate Point Source’ monitors


New flagship thunderbolt interface from PreSonus


STUDIO MONITORS Stephen Bennett speaks to three monitor manufacturers about their different approaches to driver design.


FINAL CUT Pete Cobbin, Kirsty Whalley and Tony Lewis tell AMI how they teamed up once again with Ridley Scott to compile a pulsating score for the new sci-fi horror film ‘Alien: Covenant.’


STUDIO PROFILE Distant City Studios’ Joss Worthington reveals how he turned his vision for the perfect recording environment into a reality.


OPINION Adam Daniel of Point1Post explains why the integration of Dolby Atmos with Pro Tools HD could provide the boost that the format needs


The Park Studios owner Tobin Jones on why it’s important to have good people skills to get the best out of artists.


INTERVIEW Adam Savage talks to sound designer Pete Malkin about securing a Special Tony Award and his work on the new Harry Potter play.





GEO FOCUS: RUSSIA Colby Ramsey discovers how the domestic pro-audio market in the world’s largest country has been dealing with competition from Asia and the West. May/June 2017

TECH FOCUS David Mathew of Audio Precision provides an overview of measurement microphones for accurate acoustic test and analysis.

REVIEWS 32 34 36 38 40

Audix Performance Series Izotope RX 6 and RX 6 Advanced RØDELink Newsshooter Kit Crane Song Solaris PreSonus Studio 192 Mobile

PRODUCT NEWS: WIRELESS SHURE ANNOUNCES AXIENT DIGITAL Shure used this year’s NAB Show in Las Vegas to debut its new Axient Digital Wireless System. The system features a receiver that is compatible with its two transmitter offerings, the AD Series and ADX Series, which incorporate ShowLink, providing real-time control of all transmitter parameters with interference detection and avoidance. The ADX Series also includes the first microbodypack with an integrated self-tuning antenna, enabling greater concealment and comfort. Axient Digital features Quadversity receiver technology, significantly mitigating the potential for signal fades or interference that can cause dropouts.


High Density mode increases the maximum simultaneous system channel count from 17 to 47 per 6-MHz TV band, and from 23 to 63 per 8-MHz TV band. Digital audio quality is maintained via Dante and AES3, along with a 20Hz to 20 kHz range with a flat frequency response and accurate transient response. It also boasts wide dynamic range, AES-256 encryption, and 2ms latency from the mic transducer to the analogue output. Its wide tuning range of up to 184 MHz for all transmitters and receivers covers an extended range of spectrum and simplifies inventory. Axient Digital is compatible with the Shure Battery Rack Charger (SBRC), which supports up to eight rechargeable batteries in a single, compact rack space. The AD Series will be available in late summer 2017, and the ADX Series will be available in early 2018.

A-T EXPANDS SYSTEM 10 PRO RANGE Audio-Technica is now shipping two new packaged configurations of its System 10 PRO rack-mount digital wireless system (reviewed in the September 2015 issue of Audio Media International): the ATW-1366 with boundary microphone/transmitter and ATW-1377 with microphone desk stand transmitter. The ATW-1366 configuration includes a ATW-RC13 rack-mount receiver chassis, two ATW-RU13 receiver units and two ATW-T1006 System 10 boundary microphones/ transmitters. The ATW-1377 configuration consists of a ATW-RC13 rack-mount receiver chassis, two ATW-RU13 receiver units and two ATW-T1007 System 10 microphone desk stand transmitters. Previously, the ATW-T1006 System 10 boundary microphone/transmitter and ATW-T1007 System 10 microphone desk stand transmitter were available as individual components, as integrators/contractors generally

source these pieces separately to configure a system to spec, but AudioTechnica has now decided to offer the transmitters and receivers together as turnkey packages. The ATW-1366 and ATW-1377 System 10 PRO configurations are now available at the following prices: ATW-1366 System 10 PRO rack-mount digital wireless system with two boundary microphones/transmitters: $1,349; ATW-1377 System 10 PRO rackmount digital wireless system with two desk stand microphones/ transmitters: $1,249.00.



Riedel Communications introduced its latest wireless intercom solution at Prolight + Sound in Frankfurt. Bolero is an expandable, full-roaming, DECT-based intercom system in the license-free 1.9GHz frequency range. Fully integrated into Riedel’s Artist digital matrix intercom platform, Bolero offers connectivity that can be applied three ways: as a wireless beltpack, as a wireless keypanel, and – in an industry first – as a walkie-talkie radio. Bolero runs over a standards-based AES67 IP network. Decentralised antennas connect to AES67 switches and then to Artist frames equipped with AES67 client cards, providing a fully integrated point-to-point

Lectrosonics’ new digital wireless monitor system, the Duet, is made up of the M2T dual-stereo half-rack transmitter and M2R diversity beltpack receiver. The Duet covers the UHF frequencies of 470-608 MHz in a single range, uses digital modulation for transmission, and can accept analogue or Dante digital inputs. The M2T half-rack transmitter houses two independent stereo transmitters allowing for up to four stereo or dualmono transmissions in a single rack space. The analogue input connectors are full-size XLR/TRS combo types for balanced line level analogue signals, while the input preamp circuits use a special balanced amplifier with very high common mode rejection to minimise hum and noise. Additionally, a Dante Ultimo interface via dual RJ45 connectors accepts Dante networked audio inputs and can cascade the digital stream

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intercom ecosystem with ‘seamless’ roaming capabilities. According to Riedel, the Bolero voice codec provides both higher speech intelligibility and more efficient use of RF spectrum, supporting twice the number of beltpacks per antenna for the same audio bandwidth as other DECTbased systems. Bolero also features Riedel’s ADR (Advanced DECT Receiver) technology, a diversity receiver solution specifically designed to reduce sensitivity to multipath RF reflections, making the system useable in challenging RF environments where other systems may experience difficulty. The beltpacks support Bluetooth 4.1, allowing either a Bluetooth headset or a Smartphone to be connected. Users can also make calls and then connect that person into the intercom matrix, eliminating the need for a telephone hybrid.

to additional units via CAT6 cables. An additional RJ45 jack provides an Ethernet connection for programming and control via Wireless Designer software and a USB jack on the front panel allows for firmware updates. The M2R bodypack receiver employs advanced antenna diversity switching during digital packet headers along with a 24-bit digital audio stream. The headphone jack is fed from a stereo amplifier with 250 mW available to drive headphones or earphones to sufficient levels in noisy environments.

PRODUCT NEWS: LOUDSPEAKERS GENELEC UNVEILS ‘ULTIMATE POINT SOURCE’ MONITORS Genelec has introduced the 8331 and 8341 three-way coaxial monitors, which join the existing 8351 to create a complete compact coaxial range. The new models, nicknamed ‘The Ones’, are housed in enclosures no larger than a traditional two-way 8030 or 8040 and using Genelec’s Minimum Diffraction Coaxial (MDC) driver, are designed to eliminate limited frequency range, low SPL and uneven dispersion issues. The dramatic reduction in unnatural imaging means listener fatigue is greatly reduced, while Genelec’s Directivity Control Waveguide (DCW) combines with the monitor’s concealed dual woofer design for less coloured reflections.


Both the midrange and tweeter in the new offerings share the same compact magnet system, reducing size and weight with no reduction in response. The midrange coaxial driver cone is now comprised of concentric sections, optimising midrange linearity, as does the DCW which covers the entire front face of the enclosure. Each unit incorporates three stages of dedicated Class D amplification, including ecofriendly Intelligent Signal Sensing (ISS) energy saving technology. The coaxial design allows for ‘ultranear-field listening’, creating a dramatic improvement in the direct sound-toreverberant sound ratio and reducing the room’s influence. The listening distance may be as short as 40 cm, with no loss of precision. At a listening distance of 50 cm, the maximum SPL capability is raised by more than 5dB. Finally, The Ones utilise SAM (Smart Active Monitoring), the technology based on Genelec Loudspeaker Manager (GLM) 2.0 software for PC and Mac, incorporating AutoCal.

QSC INTRODUCES K.2 SERIES LOUDSPEAKERS The next generation of QSC’s K Family line of powered loudspeakers comprises the 8in K8.2, 10in K10.2 and 12in K12.2 full-range loudspeakers. The company has also announced the KS212C, a single-box powered cardioid subwoofer. Each loudspeaker model in the K.2 Series is equipped with a 2000W power module, with DMT (Directivity-Matched Transition) that ensures smooth coverage across the entire listening area. On-board DSP provides Intrinsic Correction voicing and advanced system management to further optimise performance. The K.2’s provide a library of preset contours for common applications such as Stage Monitor, Dance Music, Musical Instrument Amplification, Handheld Mic and more, while also offering storable Scenes to recall user-configurable settings such as input type, delay, EQ, cross-over and selected contour via the loudspeakers’ LCD screen and control panel.

Meanwhile, the new KS212C cardioid subwoofer is designed to complement both the new K.2 loudspeaker line as well as legacy K Series full-range models, and is most suitable for mobile entertainers, AV production and rental professionals as well as modestly-sized performance venues. Dual 12in long-excursion drivers – each arranged in a 6th order bandpass chamber – are powered by a 3,600W amplifier and controlled by the system’s DSP to produce 15 dB more output at the front of the cabinet than at the rear. The QSC K.2 Series will be available in select markets in midMay 2017.



Funktion-One has added two supplementary mid-high and midbass enclosures to its Evolution Touring Series. The Evo 7TH and the Evo 7TL-215 come with integrated flying gear, enabling rapid, seamless configuration with all the components in the Evo 7T flying clusters. Evolution 7TH is the midhigh section of the Evo 7T. It features a 10in mid-range and a 1.4in compression driver for high frequencies. The Evo7TH is significantly smaller than the Evo 7T, making it adaptable for a number of configurations.

DIVA M² is the latest middle-format line source system from the company’s DIVA Series, which utilises a three-way enclosure, bi- amplified design, and two active drivers. The enclosure features one direct radiating 8in neodymium LF transducer, mounted in a bass-reflex enclosure, and two neodymium diaphragm compression HF drivers, coaxially-mounted with a single acoustical output, and coupled to an individual proprietary waveguide. Its unique diffraction horn achieves vertical dispersion of 10° and a horizontal dispersion of 100°, ensuring uniform coverage and spectral quality throughout the whole listening area. The DIVA M² offers a high frequency ensemble tweeter featuring two voice coils, each with its own magnet assembly and its own specialised phasing plug, which transforms the

May/June 2017

Evolution 7TL-215 also features two Evo 7T horn-loaded 15in drivers, providing mid-bass reinforcement for flown and ground-stacked Evo configurations and delivering even coverage to audiences of up to 10,000 people. The company has expanded its range of bass reflex speakers as well with the release of the BR132, which utilises Powersoft’s M-Force linear transducer and M-Drive amplifier along with Funktion-One’s own cone, surround and enclosure technology, and is around 40% smaller than the horn-loaded F132. The 32in bass reflex speaker – with a frequency range of 25Hz - 65Hz – delivers elastic bass and super-low frequency extension and is particularly effective in mid-size live and touring applications requiring deep, strong and well defined subbass at close range.

circular planar wavefront emitted by a compression driver into a rectangular planar wavefront. The tweeter features a combination of extended frequency response, high efficiency, and wide dispersion pattern, while the driver is a two-way system that utilises two concentric annular ring diaphragms. The larger annular midrange diaphragm, featuring a 3.5in voice coil, covers the frequency range between 750 and 6,500 Hz with a smooth, linear response. The ultra light annular diaphragm for the high range, featuring a 1.75in voice coil, offers a transient response with high efficiency from 6,000 to 22,000 kHz.

PERFECT BALANCE XY SE R IE S P R O F E SSIO N A L SP E A K E R S Versatile professional speakers that deliver superb sound and consistent coverage throughout venues of every size.

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NEW FLAGSHIP THUNDERBOLT INTERFACE FROM PRESONUS PreSonus has announced a new 26x32channel ‘Quantum’ audio/MIDI interface, which utilises a high-speed Thunderbolt 2 bus and direct-to-DAW signal path to achieve ‘extremely low latency.’ The new interface combines 24-bit, 192 kHz converters with 120dB of dynamic range and PreSonus’ recallable XMAX microphone preamps. For more I/O channels, four Quantum interfaces can be stacked via Thunderbolt to create a 96x96 system. The Quantum offers two combo mic/ instrument/line inputs and six combo mic/ line inputs, each with a digitally controlled XMAX preamp and +48V phantom power. There are also two 1/4in TRS main outputs, the aforementioned eight 1/4in TRS line outputs, and two independent headphone outs with dedicated volume controls.

With ADAT Optical I/O and S/PDIF stereo digital I/O, users have access to 18 additional digital inputs and outputs for a total of 26 in and 32 out. BNC word clock I/O ensures Quantum and other digital audio devices operate in tight sync, with additional MIDI I/O. Designed to be the central hub for a recording studio, the Quantum offers two main and eight auxiliary balanced line-level outputs for monitor mixing. An onboard talkback mic can be routed to any mix, and users can listen to the mix through either of two headphone amps. The interface includes PreSonus’ Studio One Artist DAW, while most of its features can also be controlled in PreSonus’ free UC Surface control software. The Quantum interface will be available in the second quarter of 2017 at an anticipated MAP/street price of $999.95.

DPA DEBUTS D:VICE DIGITAL AUDIO INTERFACE DPA Microphones has diversified its product offerings further with the new d:vice MMA-A digital audio interface. The d:vice offers ‘unprecedented’ quality for live and mobile journalists capturing or streaming audio in the field. It is a high-quality, two-channel microphone preamp and A/D converter offering mono, dual and stereo capabilities, with interchangeable lightning and USB cables for connectivity. MicroDot inputs allow the d:vice to be connected to all DPA miniature microphones, including the d:screet miniature, d:fine headset, d:vote instrument and d:dicate recording microphones with the optional MMP-G preamp. Measuring about two inches in diameter, the d:vice digital audio interface is ultra-compact, easily fitting in the user’s pocket, and is controlled through a remote app on any iOS device. The app allows users to store gain settings and low-cut filters for ongoing


May/June 2017

and future personalised use in dedicated presets. Third-party applications can be used to accomplish specific tasks, including but not limited to live broadcast and high quality recordings. “This is the first pocket-sized digital audio interface that can deliver sound that rivals studio recordings,” said René Mørch, product manager at DPA Microphones. “The d:vice digital audio interface was designed with broadcasters and mobile journalists in mind, but can also be used by other content makers for recording music, filming on location and during both live and recorded broadcasts. With this interface, users can portably record and stream clear and professional audio from the connected microphones.”

35 Years


d&b is 35. Vier is d&b. Werner ‘Vier’ Bayer is a Product Manager at d&b. He’s been on board since day one. “You dream of being somewhere, and then‌ most of it comes true. d&b is my life. Full of lunatics.â€? In 35 years d&b has evolved from a small garage venture to a worldwide standard in professional sound systems. It’s people like Vier who make WKLVVWRU\SRVVLEOHDQGMXVWWKDWELWGLË HUHQWIURPWKHUHVW

Welcome to System reality.


ARE WE ABOUT TO SEE ATMOS TRULY TAKE OFF? Adam Daniel of Point1Post explains why the full integration of Dolby’s immersive sound technology with Pro Tools HD could provide the boost that the format has needed since its inception.


n my last article for Audio Media International in January 2015 I shared some insight into how we approach Dolby Atmos mixes and their impact on the viewing experience. We have now been using Dolby Atmos for three years and I would like to describe how I have seen the format evolve and where I feel the immersive market is going next. There are now 2,442 Dolby Atmos screens and a further 354 pending. In England there is even a ten-screen multiplex where every auditorium features Dolby Atmos. But what’s really exciting is that the format is no longer just limited to cinemas. It is now available on Blu-ray, streaming TV, Windows 10, Xbox One, tablets, smartphones, linear VR and even nightclubs. This represents a huge worldwide audience. The majority of the world’s best mixing stages now feature Dolby Atmos. The leading commercials and trailers facilities offer the service too. However, it has remained difficult for the majority of sound editors and re-recording mixers to get their hands on the tools. This all changed at Avid Connect when it was announced that Dolby Atmos has been fully integrated into Pro Tools HD. I feel this is a real turning point for the widespread adoption of the format.


Bedding in Avid and Dolby have worked together to fully streamline an already efficient 12

May/June 2017

workflow. Pro Tools HD will support the wider stems required for Dolby Atmos Beds, and the Avid Panner has been upgraded to replace the Dolby Atmos Panner. You can choose whether sound is routed to a Bed or an Object and automate the switching. The RMU and Software Renderer are now set up as peripherals within Pro Tools, while the I/O Setup allows direct mapping to Objects. There is even the option to convert legacy Atmos Panner automation with a single click. Dolby has also released two software suites to complement these improvements. Everyone will now have the opportunity to use the same tools that we use. Subsequently we have decided that now is the perfect time to make a fundamental change to our workflow. The aim is to open up the immersive formats to as many productions as possible. 5.1 will now be regarded as a legacy format. We will be incorporating Dolby Atmos metadata into every project’s premix. This will allow clients to actually hear how their project sounds in Atmos. They can then make an informed decision whether to complete their final mix in Dolby Atmos, 7.1 or 5.1. The metadata will remain in place regardless of how the cinema mix is monitored. Due to the downstream nature of the metadata, clients will then have the opportunity to create a Dolby Atmos master for home theatre and the additional supported devices. This workflow could really suit independent filmmakers, as the option to create and license a Dolby Atmos DCP remains available to the distributor. If the decision is made not to use the Atmos metadata then their legacy mix will still benefit from the fantastic 7.1/5.1 re-render.

As for the other formats… IMAX, DTS:X and Auro 3D are also helping to cement the legacy of 5.1. Sometimes the biggest budget movies are released in all formats. However, in these situations it

is my understanding that Dolby Atmos is commonly used for the primary mix. The Dolby RMU’s ability to generate multiple re-renders makes it ideal for this purpose. The alternative formats are subsequently created using re-renders and the Atmos Object panning as a starting point. We have recently installed DTS:X and this order of working makes perfect sense. DTS:X can be used for cinema mixes, Blu-ray and headphone mixes on supported hardware. It has become standard for sound editors and sound designers to work in 5.1 before reaching the mix stage. This process can now include laying the foundations for a native Dolby Atmos mix. The preservation of the automation data between the cutting rooms and mix stage will be exactly the same. There is a learning curve to Dolby Atmos but widespread availability of the tools, enthusiasm for the format and creative time in the cutting room should lead to some amazing projects. One of the nicest things about working on Dolby Atmos projects has been the enthusiasm from musicians. For example we worked on the short film Escape. The music was composed and performed by Imogen Heap. It was fantastic to work with her on the mix stage. In a bold move she made every musical element an Atmos Object. The only exceptions were some of the reverbs and delays. It was great to

have so much creative freedom without being restricted to conventional channelbased stems. In a completely contrasting musical style we recently completed a mix for Hardcore Underground called ‘Blu The Roof. ‘The film is an HDR graded 4K Ultra HD Blu-ray that creates the sense of being at a Fracus & Darwin DJ set in Phoenix Arizona. It was a really exciting project to work on. The producers now want to start producing their music in Atmos and are very excited about development of the format in nightclubs. The evolution of audio in cinema, music, games, 360 Video and VR are all driving consumers to demand more immersive sound experiences. It is very exciting that more directors and producers can now tell their story with the most creative tools. I’m really looking forward to working on a wider range of productions as we move on from 5.1. The audience is ready.

Adam Daniel has been a re-recording mixer for 18 years. He started his film career in the sound department at Shepperton Film Studios, and after six successful years, Adam left the Pinewood Shepperton Group and formed Point1Post. In addition to mixing, Adam is an accomplished sound editor and also provides consultation services for studio installations and post-production workflows.


THE REAL ROLE OF THE PRODUCER/ENGINEER The Park Studios owner Tobin Jones on why it’s important to have good people skills and not just technical know-how if you want to get the best out of the artists.


ets face it, we engineers love to geek out about equipment and yes, knowing how to use the equipment is vitally important, but the thing that supersedes the technical aspects of engineering is knowing how to relate to artists and musicians – making them feel relaxed and in a creative mental space. An artist friend of mine who also runs creative workshops recently posted a tongue-in-cheek comment: “I’m going to be launching a new course: ‘People Skills for Sound Engineers’.” It’s a joke, but serves to point out that engineering is so much more than how you use equipment to capture sound; it’s just as important if not more so to have techniques at your disposal to help the artists create amazing performances and to ease any tensions that may arise. If you can help maintain a creative environment and inspire creativity, your job as an engineer will be easy – well, easier! So how do you get them to deliver an amazing take or performance? How do you get the artists to that place where they produce something magical? How do you inspire creativity? How do you get an artist to a position where they inspire their own creativity? Traditionally this would be the role of the producer – who sits between the artist and the engineer and leaves the engineer to just worry about the technical aspects of the recording process – but today, with



May/June 2017

shrinking budgets, the roles of producer/ engineer have become more intertwined.

The desire to inspire Being a great engineer is far more than knowing how the equipment works or how to mic up instruments; it’s about how you use that knowledge to inspire and capture bursts of creativity. Creating a relaxed, creative vibe and environment even when time is limited is paramount. Many engineers focus too much on whether a studio space has equipment they would like to use – I too am guilty of this one. More time needs to be spent choosing a space that is relaxing, not intimidating, easy to use and puts people at ease. Budget for this is unimportant; I have worked in many high-end studios which have felt sterile and uninspiring and likewise many smaller spaces that have felt dark and dingy. If you are using a studio space on a budget, which might not feel particularly great, spend some time making the space look nice with lamps or decorations to help artists feel at home. It may seem like a gimmick but having lights you can change the colour of depending on the mood for the song makes a big difference and if you have been in the studio for 12 hours straight, altering the lighting makes the whole studio feel refreshed. If things are not working out take a break, move on to

something else, have a tea break and don’t talk about what’s not working. Come back to it later and nine times out of ten the result will be infinitely better. With limited budgets, recording sessions can often feel rushed and artists are under pressure to suddenly become creative; more often than not this sort of pressure actually has a detrimental effect on performance. A trick of mine is to have a stack of funny hats lying around the studio, so when things feel a bit too serious I just put on a silly hat to lighten the mood. People feel relaxed when you make them a tea – don’t just leave it to the assistant (if you are lucky enough to have one) – and hang out with them in the kitchen/lounge even if only for the start of the day; it takes five minutes and really helps break the ice, especially if it’s your first time working with an artist.

Communication is key We have to make time to talk and chat with the artist to help them feel at ease; take interest in what they’re talking about, have a laugh, talk about their music and what they want to do. If the artist is at ease with you they are far more likely to be at ease with themselves and their performance. Be relaxed yourself; if you seem rushed for time or panicked the artists around you are never going to be relaxed and you

will never get a good take. Get in touch with your emotions too. Music is about emotion regardless of the genre – the artist is reflecting something of his or her own emotional being – so take a bit of time to understand what this emotion is. If you can’t relate to it you are never going to know if you have captured a good take. A trick I use when recording vocals is after about four takes I ask the singer what the song is about, what they felt when they wrote it and how they would like the listener to feel when hearing it. Throughout the production and recording process artists can sometimes become disconnected from the original emotion behind why they wrote the song; getting them to remind themselves of this really helps them reconnect with that initial emotion and creative spark. Above all, sort the business first. Get prices, deposit and contracts sorted before the session begins. I’ve lost count of the amount of projects that I’ve heard go sour because expectations were different on both sides. If everyone knows where they stand beforehand, the whole project will feel more relaxed and therefore more creative. Tobin Jones is owner and head engineer at The Park Studios, a recording studio in Wembley, London.



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BALANCE OF POWER Colby Ramsey discovers how the domestic pro-audio market in the world’s largest country has been dealing with the competition from Asia and the West.

ussia is a country that perhaps gets overlooked sometimes when it comes to the prowess of its pro-audio business. The complex and unstable condition of the global economy has caused those in the domestic market to witness a prolonged period of stagnation. Yet 2017 looks to be the year that this turbulent phase subsides for the world’s largest nation, after AMI spoke to a number of companies who, for the most part, maintain positive outlooks for the future. From the viewpoint of Irina Gromova, Russia and CIS area consultant at used pro-audio gear supplier GearSource Europe, the market is developing albeit not to its full potential. Fortunately Russia’s geographical positioning acts as somewhat of a saving grace, sandwiched between China – one of the largest manufacturers of pro audio gear – and Europe, a stable hub of pro-audio activity. However, “the same factor dictates the geographic limit for operations on the market,” says Gromova. “Our main target audience, rental companies, are limited by local regional events, meaning local budgets and limited buying ability.” Gromova also believes tighter borders are a hindrance when it comes to GearSource’s offering, resulting in complicated logistics and additional import duties and



May/June 2017

taxes: “This factor limits the trade opportunities and as a result, gear application on the local market,” explains Gromova. “It determines certain limits of involving new products in our market and certainly limits the value of total turnover. “Our distributors and/or users have to invest around 1.5 times more funds at every purchase due to import duties, and because of the longer logistics shoulder the return of such investments is substantially slower.” Despite operating in the same information field, Russian buyers are believed to be quite conservative when compared with those in markets further west. This trend has significantly aggravated the competition between brands and has forced marketers to come up with especially creative solutions, as Alexander Khorev, managing director at Major Sound, Meyer Sound’s recently appointed Russian distributor, explains. “Almost all European, American and Asian brands are represented in one way or another on the Russian market, and competition here remains very strong,” he says. “But if the product is good, it will certainly meet demand in Russia – we are approximately one to two years behind some of the current buying trends in Europe.” Headed by industry professionals with more than 25 years of experience, Major Sound was only

GEO FOCUS: RUSSIA Major Sound MD Alexander Khorev (left) with chief technical officer Yaroslav Udovik.

Population: 143 million

recently formed, and provides sales and technical support of Meyer Sound products in Russia. It has also launched a new education program, translates marketing materials into Russian and represents the manufacturer at shows.

Money Talks While Russia’s highly diverse territory and ability to transcend distribution channels makes its pro-audio offering an attractive one, its recording studios are almost exclusively located in Moscow, the country’s capital. Here, four or five large facilities dominate the recording space for music, TV and film due to their access to high-quality equipment such as AMS Neve or SSL desks. Music producer Konstantin Matafonov cites Vintage, Cinelab and the huge, historic Mosfilm complex as three examples of where the majority of recording takes place in Russia. “There are also many well-known independent studios in SaintPetersburg but they work mainly for local listeners without being into ‘everyday pop production’,” says Matafonov. “The overall quality of audio production for the music industry is not so good because the competition for quality releases still does not exist here. Heavyweight artists usually fly to well-known engineers in the USA, England or in Europe. “And, if the average producer has money, they will tend to work with a mixing engineer, almost all of which are on Pro Tools HD with a ton of plug-ins at their disposal. This poses a problem, because all well-financed projects are being outsourced to foreign producers and adequate sounding rooms prove hard to come by while overall listening

tastes/experiences are not so critical.” Despite this fact, Mosfilm sits at the top of the pile as the largest and oldest recording complex when it comes to the Russian film industry, producing almost all the country’s motion pictures as well as a vast array of TV programmes and videos. UK acoustic consultancy White Mark and Moscow-based systems integrator Sfera-Video recently completed a major project to give Mosfilm its first Dolby Atmos dubbing theatre. An entirely new isolation shell was built for the new theatre in Studio 7 – replacing an earlier Dolby Premier studio – which now houses an Avid Pro Tools S6 control surface. Pavel Bazdyrev, general manager at Soyuz Microphones, whose headquarters is located 120 miles south of Moscow, has seen very few top-end recording studios being built in the last five years: “More and more music is being created in the home studio environment,” he notes. “Since the currency devaluation in 2014, customer preferences have moved towards more affordable products and selling highend gear has become challenging.”

The Right Ingredients Due to the country’s sheer size, Russian recording artists are automatically granted a huge audience, meaning the average artist must remain vigilant when selecting good value-for-money equipment in order to stay competitive. Yet quality electronic components are not often made within Russia, resulting in additional import/customs charges that are difficult to ignore. Matafonov reveals that Thunderbolt

interfaces, compact monitors and unusual mixing/live controllers are trending at the moment, with the new generation of UAD plugins also expected to be popular. Gromova concurs in that she does not see an essential growth of local production to compete with European gear going forward, and expects demand to be more focused on “optimal, more economically effective solutions.” Gromova believes that this will in turn affect import turnover, “primarily in diversification of turnover between Europe and China in favour of the latter [new gear in the domestic market],” she explains. “Today’s buyers are looking for versatile, cost-saving, multipurpose solutions that will provide them flexibility in the use of gear and quick ROI. Used gear certainly holds a growing position here. “Then there are of course pending financial conditions and customs regulations,” Gromova adds. “Higher banking rates and complicated, expensive import routines mean longer and costly fund investments, which has always been and still is the main challenge for local businesses including manufacturers, distributors, resellers and end users.” Khorev is seeing many turn their business from import to domestic production: “Producing in Russia is now profitable again, as a number of different industries including lighting and stage equipment develop very quickly,” he says. “I think in the next five years the pro-audio market will mature further and show a good dynamic.” For manufacturers, it has become almost impossible to outsource most of their operations, as Bazdyrev

explains: “Very few companies can offer the level of quality we’re looking for with our relatively low volume, which is why we have to make 95% of operations in-house. “I think it’s a Russian’s strong perception of local product to be not as good as imported ones, stemming from the late 90s when local manufacturing was in decay and Western/Asian brands took advantage of building a very strong reputation for themselves here,” he continues. On the recording side, Matafonov would like to see more affordable, quality production studios open their doors, offering large-scale format recording services in a more compact environment. As a result, he expects acoustical engineering services to grow while the new generation of artists opt for more new and unusual tools and mobile setups. While most of those operating in the Russian pro-audio market seem poised for future challenges, these are likely to change due to the shifting, complex nature of the global economy, unstable political developments and “natural demand evolution process” that Gromova describes. “If we speak about new prospects and opportunities, we have to thank the global information exchange development, new consumer generation and new technologies that businesses cannot neglect and need to implement today,” Gromova urges. “I hope it will develop further in terms of openness, transparency, and further integration into the global trading economy.”

May/June 2017



Genelec’s ‘The Ones’ were unveiled at Metropolis Studios in early May

IN THE DRIVING SEAT Stephen Bennett speaks to three monitor manufacturers about their different approaches to driver design. he first magnetic driver speakers were developed in the 1920s and by the middle of the twentieth century sophisticated full-range monitors were available that any music lover would be happy to call ‘hi-fi’. You’d think then that innovation in driver development might have stopped at that point – we’ve had moving magnet and moving coil, electrostatic, compression, dualconcentric and ribbon drivers coupled with sophisticated enclosure designs, powerful and dynamic amplification and low-noise electronics all aided by Computer Aided Design (CAD) and Digital Signal processing (DSP). Hasn’t the discerning engineer and listener been able to purchase monitoring systems capable of recording or reproducing the ‘best sounding album of all time’ <insert personal favourite here> for decades? So what is there left to discover about the good ol’ loudspeaker? Well, plenty, if the latest



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launches from some of the sector’s leading manufacturers are anything to go by. For example, the three companies questioned for this article offer quite different technologies and design philosophies with their monitors and these choices are the product of a long history of technological innovation.

A bit of background “The historical line of development for Genelec has been to use stiff radiator diaphragms instead of ones that flex significantly and are subject to modal resonances within their intended frequency range of use,” says Aki Mäkivirta, director of research and development at the Finnish firm. “Stiff diaphragms can allow for better control and are more predictable in production. This translates to a more exact control of directivity and more even quality in production.” Pioneer DJ has been developing speaker products for many years, operating under the sub-brand

Technical Audio Devices (TAD) and Takashi Mitsuhashi, general manager, Engineering Department IV, says that the company has a strong history of designing coaxial drivers. “The Coherent Source Transducer (CST) driver, the Acoustic Filter Assisted System Tuning (AFAST) technology and noise suppression processing inside the RM range of speakers are important elements that have been based on technology developed by TAD,” he explains. ADAM Audio is best known for its X-ART tweeter, a third generation version of the Air Motion Transformer (AMT), according to chairman David Angress. “The AMT was created and first marked in the 1970s by Dr. Oskar Heil, a German physicist who had emigrated to the US.”

One direction Genelec has also developed two new coaxial designs, nicknamed ‘The Ones’, that they say overcome some of the

limitations the company believed was fundamental to previous designs. “The front surface of the transducer is continuous,” says Mäkivirta. “There is no discontinuity-related diffraction colouring the audio. We have achieved controlled neutral response in the off-axis directions too. This is possible because we are building the coaxial transducers in a unique way, using flexible sections that conform to the global shape needed for the directivity waveguide.” Though different companies may utilise many of the same kinds of physical parts for their drivers – cones, coils and magnets – and all are attempting to achieve the same sonic ends, the way they are implemented in practice can be quite different. “Controlling directivity in loudspeaker system designs has been one of the longterm goals for Genelec,” says Mäkivirta. “Using a stiff or rigid transducer diaphragm is therefore the logical choice in applying this design principle. Not only

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FEATURE: STUDIO MONITORS do we want to control directivity but we want to do this in a way that this enables the loudspeaker to radiate neutral audio on and off the acoustical axis in order to make sure that the audio going into the room’s reverberant field remains neutral in addition to the sound radiated directly toward the listener.” Angress explains ADAM Audio’s approach: “The technology we use includes a ‘voice coil’ that is printed onto a substrate that is then folded into a diaphragm that looks like a [set of] bellows and is put into a magnetic field. When an audio signal is passed through that voice coil the diaphragm ‘squeezes’ the sound out rather than moving like a piston as a conventional speaker does. The result is a four-times-faster acceleration of the audio, giving the listener excellent transient response, extended high-end response – up to 50kHz – with extremely low distortion.” Pioneer DJ chose to utilise a different – yet familiar – transducer. “The coaxial driver and the noise elimination technology inside the cabinet are the speakers’ main features,” states Mitsuhashi. “Many speakers can’t reproduce spatial information accurately, but in the RM range the sound that’s reproduced becomes wider around the speakers because the sound source includes spatial information. When the sound reproduction is accurate, the sound fields are reproduced in front of you and you don’t ‘feel’ the speaker’s physical presence. Additionally, Pioneer DJ’s AFAST technology drastically reduces standing waves that can produce a muffled response in the low to mid ranges. An acoustic tube in the cabinet absorbs only the standing waves, ensuring true reproduction of low to mid frequencies.”

hard work pays off Mäkivirta is also keen to mention Genelec’s philosophy of systematic improvement through better and deeper understanding of driver design over four decades of continued research and development. “Transducer engineering and system design have resulted in the unique possibility to optimise the system performance for the stringent needs of professional audio monitoring,” he explains. “While Genelec monitors are excellent for enjoying music and high quality recordings, our starting 20

May/June 2017

point is not to tailor the loudspeaker sound character in a certain subjectively pleasing fashion, but to offer as accurate a presentation of the electronic input signal as possible.” And although ADAM’s ribbon tweeter is the most visible feature that defines its range of monitors, it is only one element of speaker design, as Angress notes: “It is important to create an integrated and well-balanced design that considers the other drivers in the system, the correct amplifier power, modern crossover design and properly dimensioned and constructed cabinets.” Mäkivirta agrees, saying that complete systems are more important than just the drivers. “Transducers cannot and should not be designed in isolation. To truly optimise the system for the best performance it is vital to consider the three main components simultaneously: the transducers; the acoustic-mechanic system design, including the loudspeaker cabinet and the transducers; and the electronic system, including the power amplifiers and the signal processing, be it analogue or digital. Poor transducers do not turn to gold with any amount of DSP magic.” Mitsuhashi says that the audio from the point source in Pioneer DJ’s coaxial driver is integrated with the cabinet technology, which then “suppresses noise, creating accurate unequal reproduction.” The company’s monitors feature AB class amplification in a High frequency/Low Frequency bi-amp configuration. “We optimise materials by making hybrids,” adds Mitsuhashi. “For example, in the driver diaphragms we use Aramid fibre combined with paper, which is commonly used to make helmets.” Genelec manufactures the key transducers in-house to have, as Mäkivirta says, “the best possible grip on quality with a very rapid feedback cycle back with R&D if things are not working in production as expected.” He adds that it helps the company to design products with high endurance and a long expected lifetime. ADAM’s products are designed in-house in Berlin, Germany, while Pioneer DJ’s Research and Development arm is in Japan. Angress says that ADAM’s new S-Series flagship range feature woofers that are a new ‘ground-up’ design that incorporate a novel symmetrical magnetic circuit, new low-mass

honeycomb sandwiched cone material, and cooling technology. “These features result in woofers that have extremely light yet stiff cones for excellent transient response and low distortion, as well as the ability to move more than three times further than their predecessors, giving higher maximum SPL capability at lower frequencies. The bass and midrange components are driven by Class D amplifier modules, and the S-ART tweeter is driven by a Class A/B amplifier of ADAM’s own design that pairs particularly well with this tweeter.”

All part of the process Mäkivirta says that the healthy way to design high performance systems is to start with well performing transducers, a correctly optimised enclosure and to perfect the system with careful application of the signal processing. “Proper acoustical design is the healthy platform for utilising DSP to further optimise the system performance and offer tools for active compensation of the acoustical influences of the installation room to maintain the neutral system characteristics.”

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FEATURE: STUDIO MONITORS Mitsuhashi explains that Pioneer DJ has applied acoustic simulation for the basic design of the monitor’s waveguide, the Harmonized Synthetic Diaphragm Optimum Method (HSDOM) diaphragm in the coaxial driver and the shape of the enclosures. “We’ve also used 3D printing for prototyping,” he continues, “which allowed us to achieve a precise design, which we fine-tuned by testing and experimenting with the technology.” Angress says that ADAM spends a lot of time listening to customers on how their workflows are adapting as recording formats are changing. “These conversations, along with research into latest technologies, drove the decision to provide crossover and equalisation options via DSP using the latest generation of SHARC chips,” he says. “This also ‘future proofs’ these speakers by allowing new configurations to be set up as workflows continue to evolve and new networking standards emerge.” “Driver design is still an area of innovation, since there are always new and exciting technologies being developed,” states Mitsuhashi. “This is crucial in delivering products that can be lighter in weight and possibly smaller in size. As new materials and driver configurations are designed, innovative products emerge which provide multiple application options for audio professionals.”

levels, long-term heat management, and long-term stability of materials and structure. While the fundamental principles of transducer design have been known for a long time, better modelling tools and new materials offer new possibilities of improvement. Mitsuhashi believes that optimum designs will be developed according to the needs of various applications. “Smaller and lighter drivers are always being developed,” he says. “This, in conjunction with the DSP technology of today, means, that driver development will always be a crucial part of speaker design philosophy. “ “There has been some study on improving the efficiency of the electroacoustic transducer during the last few years,” reports Mäkivirta. “This research may result in some radical ideas that may lead to novel designs, but not without changes in the amplifier technology as well. Loudspeaker arrays and systems for wavefield synthesis transduction are developing some very interesting solutions.” He also says that with the development of advanced modelling methods, these trends will hopefully fuse into more innovative control designs and a better understanding of the optimisation of the variables that create high-quality wideband transducers.

Trial and error

In conclusion

In Mäkivirta’s opinion, transducers turning electrical drive into audible pressure variations have been, and still are, in many senses, the limiting factor in loudspeaker design – and all manufacturers must face these issues. Angress then goes on to say that: “Over the past decades many different technologies have been tried and eventually abandoned. The dynamic loudspeaker is still the mainstay of our industry. How the design theory is implemented, which materials are used, and how the designer integrates all the elements are keys to their success.” “Today, signal processing can have extreme linearity and inaudible noise level,” notes Mäkivirta. “The transducers have, relatively speaking, rather more stringent limitations on neutral and controlled frequency response, linearity at maximum output

It is clear that while the underlying technology and physics of driver design is well understood, modern design and manufacturing technologies are pushing the envelope of what is possible with transducer design. As ADAM’s David Angress concludes: “We will continue to enjoy incremental improvements in both sound quality and customer value over time. As far as new technologies go – well, we’ll see!”


May/June 2017

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///////////////////////////////////////////////////////////////////////////////////////// Pete Cobbin, Kirsty Whalley and Tony Lewis talk to Adam Savage about teaming up once again with Ridley Scott to piece together a pulsating score for the new sci-fi horror film Alien: Covenant. or fans of the original and the 2012 prequel Prometheus, Alien: Covenant is surely one of their most eagerly anticipated films of 2017. Promising another dose of heartthumping deep-space terror that made Ridley Scott’s 1979 masterpiece a sci-fi classic, the latest entry in the series not only saw Scott hook up once more with some of his most trusted crew members in the music and audio

F 24

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department, but also bring some new faces on board too. With the movie already out in UK cinemas and due to hit US screens on the day this issue is due to land, we reached out to music editor Tony Lewis and score mixing/production duo Peter Cobbin (former chief engineer at Abbey Road Studios) and Kirsty Whalley – all longtime affiliates of Scott’s who were tasked with sculpting a score that would offer both subtle nods to the legendary

Jerry Goldsmith’s work from nearly 40 years ago, and something totally new for the modern cinemagoer, which also led to the appointment of experimental composer Jed Kurzel (Macbeth, Assassin’s Creed). As is often the case in the early stages of these projects, coming up with a direction for the score became a bit of an experiment itself when sharing ideas early on with Scott and film editor Pietro Scalia. A temp score from Lewis was

required, and although it was his job to come up with suggestions of what might suit the picture, there were a couple of interesting requests from the director that set the wheels in motion. “We had a phone call from Pietro saying Ridley would really like to try some of the original 1979 score and I initially raised my eyebrows and thought ‘OK we can try that,’” Lewis recalls. “I reached for it in my library and it is a great score, but it sounded very 1979! I


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Kirsty Whalley and Peter Cobbin at their new studio in Islington, London

//////////////////////// thought how is this going to work? We started to put a bit of that on the film and it kind of did. “The more we put on the film the more it was really soaking up this Jerry Goldsmith thematic vibe and the brief evolved into how can we make these themes work with what audiences today consider to be a modern film score with exciting action music?” Cobbin adds: “Ridley also wanted something different and latched on to a few things that Jed Kurzel had done. I think Tony threw some of his music up as temp music as well; Ridley loved

it and said ‘that’s what I want.’ You could see how both of those elements could work. I think the responsibility of making that blend and make it seamless fell more into our camp with our experience working with all sorts of composers and scores. We could potentially see a way in the manner in which it could be orchestrated.” For those not familiar with Kurzel’s work, his musical style couldn’t be more different to the traditional orchestral approach that Goldsmith favoured, and he arguably doesn’t follow the film scoring rulebook as closely as others, which has led to a portfolio of work that has seen him become a real rising star in his field in recent years. “Jed works with soundscapes and – I don’t think he’d mind me saying – grooves and riffs. It’s a really interesting hybrid score,” Lewis says. “I think it’s unique for Jed, this one, and he had to do some unique writing that he’d never really attempted before, and there’s some pretty sweeping stuff in there.” “Jed’s approach to writing would be very different to Goldsmith. A lot of his music comes from performing – he plays guitar and all sorts of other instruments – and nearly all of his sounds come from an acoustic source, but he does a lot of processing and interesting things with pedals,” Whalley explains. “It didn’t take us long to realise that some of this was going to play into

Jed’s strengths as a composer with his lovely indie, eclectic sound world and I think that was largely driving a lot of the scenes to do with tension, atmosphere and terror,” Cobbin adds. Kurzel’s preference for playing some material live into Pro Tools and then experimenting with the results – rather than writing a tune and orchestrating it in the usual way – is a bit of an unconventional one, and also meant that Cobbin and Whalley would have to expand their roles somewhat, offer more support to the composer and organise a small team of orchestrators themselves due to Kurzel initially having his hands full with temp demos. Kurzel was also a latecomer to the team – he only started seeing the film in November and the first scoring sessions were planned for mid-late January. “Jed and Matt [Lovell, engineer/ programmer] would come up with ideas and in the early days they would go directly to Pietro and Ridley via Tony, but as the schedule was looming for the scoring they sent some of their ideas to us and we would line it up with picture and sometimes embellish their ideas because by then we had a pretty good respect for what they wanted to do,” Cobbin explains. “We were very keen not to make their music at all sound like anyone else’s, so it challenged us to ensure that what we were doing was still working within their sound world.”

Big band theory The score was performed by two orchestras: the London Contemporary Orchestra (LCO) at AIR Studios and another large ensemble organised by contractor Isobel Griffiths for a further batch of recording sessions at Abbey Road. Both were of around 92-piece sizes. “A lot of the textural, experimental sounds that Jed is used to making we did with the LCO and for the parts that were more in the Jerry Goldsmith style, we used Isobel’s orchestra,” says Whalley. The beginning of the film is where fans will notice most of the Goldsmithstyle cues, which are ideal for the more scene-setting moments before – and this is hardly a spoiler if you’ve ever seen an Alien film – the tension really racks up with the more grisly sequences later on. The much quieter opening could almost have one thinking why it was necessary to have such a large orchestra for this particular cue, but there are good reasons. “It was a really big orchestra but we had a lot of them playing very quietly at the beginning so you can feel this sense of growing and swelling into the Jerry Goldsmith section. There’s always this power behind it even if it’s very minimal,” notes Whalley. Cobbin agrees: “There’s something about when a piece of music is placed softly, but with a large orchestra. It really May/June 2017



Sonic weaponry

One of the recording sessions at AIR Studios

gives us the opportunity to create this large, widescreen sound.”

Fingers on the pulse Although Kurzel was able to show off his talent for pushing the boundaries of film music throughout, there is one standout scene (which the main image at the beginning of this article was taken from) that really cranks up the fear factor with a pulsating visual and aural build-up that’s based around a thumping underlying musical ‘pulse’ that gradually swells towards a brutally bloody climax. “This part barely has any orchestra and is more Jed’s sound world. That kind of was the ear candy for Ridley – he heard something similar to this and thought ‘this is going to be good scary music’ without us needing to have [for example] ten double basses and 16 bass trombones,” says Cobbin. “Once that pulse starts it’s not going to waiver; it’s relentless. Jed Kurzel chose his moments when to highlight the picture with a new sound or element coming in so it feels like it’s not trying to overplay the picture. It’s a pretty horrific scene and it’s brilliant the way it’s put together.”

A different approach There may have been a lot of experimenting during the temp phase, 26

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but trying out new ideas didn’t stop once the recording sessions started. “For other scores we’d use a lot of percussion but Jed didn’t really want to steer the score in that direction, which we thought was a good opportunity but we weren’t sure how we were going to build it up,” Cobbin reveals. And that’s when the experimentation started, as Whalley explains: “They did a whole rhythmic section where they were using Oyster cards to strike their strings. So you get this really unusual sound and you also get quite a complicated rhythm, so the cellos might start off and then someone else might come in and add a layer of complexity. We also got some other effects like getting them to put their bows on their tailpieces instead of on the strings.” “Give good guidance to good musicians and they’re really quite responsive to doing new things,” Cobbin adds.

Fight against time Having the freedom to try out new things during a session is all well and good when you’re given a generous deadline, but as is so often the case on mega-budget films such as these, the Alien: Covenant team were not so lucky. “We didn’t have a lot of recording time and we had to make sure that when the music went on the stands, even down

to the way it was copied, if we lost two minutes because of too many errors, that was possibly going to undermine our schedule,” Cobbin says. “That was trickier than you’d imagine as well because Jed doesn’t use a click or a tempo reference,” Whalley adds. “So before an orchestrator could orchestrate it we had a whole process where Cecile [Tournesac, score editor] would take things and impose a click and musical structure to all of the pieces, which everybody could stick to, so by the time it hit all of the music stands, there was no confusion. “There was a real balance between allowing time for all of that experimenting and trying things but also getting through the amount of music we had to record.” The score also needed to be recorded with a Dolby Atmos mix in mind, as Lewis reveals: “I was able to come to the party with elements that we recorded at AIR and Abbey Road that we could use to pan above us. Pete recorded with mics up in the gods in both rooms, and we were able to translate that in the Dolby Atmos mixing room to put this beautiful lid on the music.” Finally, Tony Lewis feels it’s only right that we also mention the “incredibly detailed” sound work that he has seen go into this and every other Ridley Scott movie he has worked on, overseen in

Cobbin and Whalley used a lot of period mics to reference the 1979 Alien sound, such as the Neumann M50 for the Decca Tree, as well as vintage AKG C12’s, Neumann u47’s and Coles 4038’s. String spot mics included the new Chandler/Abbey Road REDD mic, which Cobbin was involved in developing. Cobbin and Whalley have gone on to purchase a matched trio – now in regular use. Other new mics of theirs included the Sontronics Apollo, Brauner VM1’s and a selection of other Neumann and Schoeps mics. During mixing they used Bricasti, Lexicon and Sony hardware reverbs with their own convolutions. For analogue outboard there was a Manley Variable Mu limiter/compressor, Thermionic Culture Vulture, Chandler Curve Bender and GML 8200 EQ’s. There was also a variety of plugins in use – highlights of which are the Waves/Abbey Road REDD and Reverb Plates, Massenburg MDW EQ, The Cargo Cult Spanner, along with various FabFilter and Universal Audio tools. Everything was mixed on a 7.1 Bowers & Wilkins 800D3 monitoring system at their new Henry Light studio in Islington, London. this case by supervising sound editor Oliver Tarney. “Ollie and his team have done a terrific job. We always find ourselves in several different scenarios, be it an enormous plague of locusts and the sea parting in Exodus: Gods and Kings, and in The Martian there was a massive gravel storm in the first reel. All of these sequences have big driving music going through them and sound-wise it’s incredibly detailed and loud. We always get faced with this ‘how are we going to mix this?’ scenario and indeed ‘how are we going to write the music correctly so that it fits?’ This was no different.”

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Building a recording studio from the ground upwards is certainly no mean feat, but when two great visions come together, the results can be spectacular. Colby Ramsey discovers how a rural facility in the North of England has produced just that.

////////////////////////////////////////////////////////////////////////////////////////// t the time of writing, a newly-modernised recording studio tucked away in the idyllic village of Rippondon in West Yorkshire is preparing to reopen its doors to clients following a massive overhaul. Joss Worthington’s Distant City Studios has been closed for the past year for refurbishment, with studio design and build company Studio People taking the reins for the extensive project. Despite its locale and appropriate titling, however, Distant City – which has been in its current setting for nearly 20 years – is still within close proximity to big cities like Leeds and Manchester. For a number of years, local bands have brought the area’s strong folk music



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scene to Worthington’s doorstep, choosing to record their material at his facility. Worthington, who took over the studios around 13 years ago, has in fact been putting out his own material for the last four years as Postcards From Jeff, including an EP and an album. “I’ve got some background in film and have done a bit of soundtrack work, so the music I produce is quite cinematic,” says Worthington. “This actually helped draw some attention to the studios, including that of Blackpool four-piece band The Membranes. I did their album in 2015, which was rather well received.” It turns out that Worthington’s ownership of Distant City came about somewhat accidentally. While visiting the studios on a regular basis as a musician, he quickly formed a partnership with its

previous sound engineer, whose role he eventually assumed. “I just got more and more involved with the scene as time went on, and as a result of that I began to realise some of the acoustic problems that the studio had,” he recalls. “There’s quite a bit of industry in this little village now and it has become a bit of an issue. I’d always dreamed about doing things from scratch in my own way, and was very keen to build the studio with my own vision in mind.” John Holmes, Studio People’s technical solutions manager, adds: “Joss first got in touch with us after he found our website and at the time, we were on projects along the M62 corridor between Manchester and Hull so an initial on-site visit was arranged swiftly and proposals starting appearing from that point onwards.”

“When I got talking to Studio People,” continues Worthington, “I soon realised that this was a once in a lifetime thing to do on such a grand scale, and to do it all at once. I wanted to integrate the downstairs areas as much as possible and that’s when the crazy ideas started coming.”

Raising the roof Perhaps the craziest idea of them all was a motorised acoustic drop ceiling that Studio People specified to allow for a double-height drum room, which according to Holmes “appeared very early on in the process as our acoustic consultant Chris Smout’s way of maximising the volume of the drum room within the given footprint.” Yet there were a number of considerations with regards to health

STUDIO PROFILE Joss Worthington

chill out area and even outside,” Holmes reveals. “If Joss wants to record an instrument in an area, he can. Integrating his 24-track Otari MX 80 tape machine to the system was fun too, enabling him to track to tape, mix from tape, or a hybrid combo of tape and Pro Tools.” “We worked intensely together over the space of a year so would certainly consider them [Studio People] friends now,” Worthington adds. “All the building and wiring guys were fantastic and there were very few things that were turned down as completely crazy. It’s a very versatile space now, utilising every room as a recording environment.”

New heights

//////////////////////// and safety for the new ceiling, which has several microphones installed within it: “We came up with a failsafe mechanism in the control room that means we can record at different heights to hear the change in sound,” Worthington remarks. “It sounds absolutely fantastic – better than I had imagined. I think it is definitely something rather unique about what we offer here.” Worthington’s other primary brief was for the control room to be as flexible as possible to allow for any producer to come in and apply their workflow to the studio’s gear, so it was eventually decided that lots of careful, complex consideration should go into the patch bay layouts and cabling infrastructure in particular: “Each room has mic, line, instrument, speaker and data tie lines to it, including the stairwell, store room,

A major problem that was discovered just weeks into the build, though, was the state of the roof, which on a building over a century old in a high-risk flood area was a big concern. Following the strip-out, Studio People’s structural engineers forcibly advised the roof to be repaired as the joists and purlins were in quite bad condition. “This inevitably put a hold on the studio build element but did allow us to design a new steel-based support structure to gain a little more height on the first floor, and revealed the building previously had skylights so these were re-instated too to allow plenty of natural daylight into the first floor,” says Holmes. “So it did have benefits over the shock of the initial uncovering of the issue!” Three weeks after installing the new roof there was another bout of bad flooding which caused extensive damage in the area, proving the decision was the right one at the right time: “It was a big problem but I think it was dealt with in a very smart way,” says Worthington. “The new control room looks out onto the picturesque village, giving it a particular

country vibe, while lots of wood finishing lends it a very natural, homely feel.”

The Personal Touch Worthington has spent the last couple of months remixing a synchronisation album which is going out on EMI production music, while he also plans to welcome back The Membranes to use the new drum room: “I think that’s got a lot of people excited, especially local bands, and they’re looking forward to getting in there,” he notes. An SSL AWS 948 Delta desk is being used in the control room, which Worthington praises for its handson approach and ability to integrate seamlessly with Pro Tools, while in the speaker department there are Neumann KM120s with a KH810 sub, and Genelec 1037s with 1091A subs. The studio’s collection of outboard gear includes Teletronix LA2As, Universal Audio 1176s, Neve 1073s, as well as a whole host of API preamps and 550/560 EQs. Recording options include a Telefunken U47 – the most frequently used in Worthington’s arsenal – along with a Neumann U87, Royer 121s, and

AKG 414s and 451s. For writing and MIDI-based composing he uses Ableton software, while for mixing and analogue recording he sticks with Pro Tools. “Joss knew exactly what he wanted from the space and has a great design eye, so our design team found working with him to be very straightforward and focused,” says Holmes. “This place does not look like any other Studio People studio, and everywhere you look you can see Joss’ input.” While Distant City Studios’ unique selling point is irrefutably its innovative new drum room, many new possibilities have been unearthed with regards to the new layout and collection of gear. “I think the whole vibe of the place is pretty unique and I’ve had lots of comments about it already,” Worthington concludes. “While there are some great studios in this part of the country and it’s a difficult industry, I’m just trying to do my own thing. I couldn’t really want much more with the space I’ve got now.”

May/June 2017



AN INTRODUCTION TO MEASUREMENT MICROPHONES David Mathew of Audio Precision provides an overview of these convenient tools for accurate acoustic test and analysis. ike all technological devices, loudspeakers, headphones, ear buds, MEMS speakers, sonar emitters and even police sirens need to be measured and tested, both in design and in production. Further, products that make any noise at all (motors, aeroplanes, wind turbines, coffee makers) are often measured for safety or environmental impact, or are continuously monitored, listening for signature acoustic signals that indicate correct performance, or failure. The sensors required to accurately acquire the acoustic signals for test and evaluation are not the rock ’n’ roll mics a drummer has arrayed around his kit. A wide range of measurement microphones from a number of manufacturers have existed for many years, with a special solution for every acoustic test need. A measurement microphone is like an ordinary microphone in terms of its superficial features: it is typically tubular, with a sensor at one end and a connector at the other, and the sensor itself is a lightweight diaphragm that is excited by changes in air pressure, responding in a way that can produce an electrical signal. But at this point the two microphone types diverge – you won’t see a singer’s wireless mic measuring loudspeaker drivers in an anechoic chamber, and you won’t see a comedian using a measurement microphone for the mic drop at the end of his routine.


Characteristics Measurement microphones are optimised for superior performance in one or more of these characteristics: frequency response, frequency range, self-noise, maximum level and distortion. Further, some are designed to be robust in harsh environments, or to have characteristics that closely match in an array application. Sensitivity and frequency response are very stable over 30

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time. A measurement microphone is typically shipped with a calibration table or chart documenting its performance. Frequency response – Typical measurement microphones are specified as ± 2 dB from 5 Hz to 20 kHz, but some models have useable response as low as 0.07 Hz, or as high as 140 kHz. Low noise – Most measurement microphones have a noise floor of about 20–40 dBA, but specialised 1in models can spec a noise floor as low as –2.5 dBA. Maximum level – For measurement microphones, 3% THD is considered overload. Typical measurement microphones might overload at 160 dB; specialised models will not overload until 184 dB or more. Engineers with some experience in sound amplification or recording might be familiar with microphone directional patterns such as cardioid, figure of eight, shotgun and so on. These characteristics are accomplished by modifications to the basic diaphragm element, such as acoustic ports, additional diaphragms, or interference tubes. Measurement microphones, on the other hand, are omnidirectional,

without modifications for directionality. Measurement microphones are optimised for one of three acoustic applications: measuring sound pressure, measuring incident sound from one direction in a free-field (anechoic) acoustic space, and measuring sound that may arrive from any direction (random incidence) in a diffuse-field acoustic space. Measurement microphones are offered in four nominal diaphragm sizes: 1”, ½”, ¼” and 1/8”. Generally speaking, the smaller the diaphragm, the greater the self-noise, the higher the frequency response and the higher the maximum level. Most general applications are satisfied with ½” measurement mics.

Sensor design A number of methods have historically been used to convert sound pressure to an electrical signal: piezoelectric, using a crystal attached to a diaphragm; variable resistance, using packed carbon granules in a small container, attached to a diaphragm; dynamic, using a magnet and a coil to convert diaphragm movement to a current; and variable capacitance,

where the diaphragm itself is one side of a capacitor, converting the movement of the diaphragm into a voltage. As it turns out, the capacitive method will, in most applications, provide the most sensitive microphones, largely due to the low diaphragm mass that this method makes possible. A survey of measurement mics over the past 50 years reveals wide use of capacitive microphones. In microphone circles, capacitive microphones are often called condenser microphones, and that is the term we will continue with. The one exception is an application where the sound levels are very high, such as near a blast or explosion. In this case, a piezoelectric measurement microphone is the correct choice.

Powering condenser microphones A dynamic microphone can simply be connected via a shielded cable to an appropriate downstream amplifier and put to work. Condenser microphones, however, require more support: • The capacitive sensor element requires a polarising voltage.


• The impedance of the sensor element is very high. Consequently, the signal current is so small that it must be amplified at the source before it is swamped by noise. Condenser microphones always have a preamplifier either built into the microphone body or connected directly to the microphone sensor capsule. Prior to the introduction of solidstate amplifiers, the preamplifier in a condenser microphone was of a vacuum tube (valve) design. These microphones required custom power supplies and

multi-conductor cables that provided the capacitor polarising voltage and also plate voltage and filament current for the tube. Today, measurement microphone preamplifiers are solid-state and have modest power requirements. Depending upon applications, some microphones are externally polarised and require a 200V polarising voltage; many other designs are pre-polarised, with an electret capacitor as the sensor element, and require only preamp power. Early electrets were not suitable for highperformance applications, but modern electret microphones offer excellent specifications and long-term stability.

Effect on incident sound waves The mere presence of a microphone in an acoustic space disrupts the sound pressure wave as it encounters the microphone. The wave reflects from and diffracts around the sensor element to varying degrees, depending on the dimensions of the microphone and the frequency and angle of incidence of the

sound wave. This effect is avoided in the first case below. Pressure mic – A microphone’s pressure response is flat when its presence does not disrupt the pressure wave. This occurs when the microphone is not in the sound field, but is a component of the barrier containing the sound field. Applications include flush mounting within an acoustic coupler, or flush mounting on a wall or barrier. Free-field mic – A free-field microphone is compensated to produce a flat response when used in an anechoic space where the sound waves are arriving from one direction. Applications include loudspeaker testing, microphone testing, evaluations and monitoring of sound-emitting equipment, and soundlevel meters. The sound field must be free of reflections, such as an anechoic chamber or use out-of-doors. Diffuse-field mic – Compensated to produce a flat response in a reverberant space such as a church, a concert hall, or an aircraft or automobile cabin.

Applications include room tuning, impulse-response testing and ambient industrial or environmental noise evaluation.

Microphone arrays Some applications require a geometric array of two or more matched mics to capture temporal, directional and phase information for mathematical analysis. Array microphones are typically of the free-field type, with careful attention paid to close phase-matching among the microphones. Because a large number of microphones may be required, array microphones are usually of a generalpurpose design. David Mathew is technical publications manager and a senior technical writer at Audio Precision. He has worked as both a mixing engineer and as a technical engineer in the recording and filmmaking industries, and was awarded an Emmy in 1988.

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May/June 2017




Andy Coules shares his thoughts on this versatile range of wireless systems from the American mic company. he Audix Performance Series wireless system comprises various components and configurations, at the core of which are two transmitters – the H60 and B60 – and four receivers: R41, R42, R61 and R62. The 1 and 2 in the receiver names denote single and dual units in halfrack and full 1U rack configurations respectively; the 4 and 6 denote the 40 and 60 series receivers, the specifications of which differ in terms of the width of the tuning spectrum (32 and 64MHz), the range (91 and 137 metres) and internal antenna management (diversity or true diversity). The H60 is a traditional handheld microphone transmitter, which can be combined with one of three capsule: an OM2 or OM5 dynamic or VX5 condenser. There are also various accessories including lavalier microphones, headworn microphones, instrument cables and antenna combiners, which can be combined in over 60 different configurations to cover a wide range of possible wireless use. The receivers are housed in stylish black metal enclosures. On the front is a power button, as well as four control buttons and an LCD display panel for each channel, and at the back there’s balanced XLR and unbalanced 1/4in jack outputs for each channel, a power socket and a pair of BNC connecters (for



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antennas). All units feature two antenna connectors, the dual units utilising an internal antenna combiner. For this review I was supplied with the AP42 VX5 E system, which comprises the R42 dual receiver and two H60 handheld transmitters (with VX5 capsules, and shown on the right). I was also sent a B60 body pack transmitter and a HT5 headworn condenser microphone.

E-asy does it The E in the name tells us that the system operates in the E band of frequencies, specifically 823-832 (channel 65) and 863-865 MHz (channel 70). Channel 65 was freed up in 2015 following the digital switchover and requires a licence for operation in the UK, this being the same UHF shared frequency licence required for channel 38 (I spoke to Ofcom to confirm this as the information online is unclear). Channel 70 doesn’t require a licence for operation and is licence-free across the whole of Europe, which is great news for those of us who are used to our wireless gear suddenly becoming illegal when we cross certain borders. The only issue is that it’s only a 2 MHz slice of channel 70, which limits the number of devices you can use in this range – not to mention the fact that those frequencies are likely to be very popular. When I unboxed the system I was pleased to discover it comes with a

printed manual, which I completely ignored as I connected everything up and turned it on. The clear bright white display on the receiver shows RF and AF signal strength, group and channel numbers, receiver lock, receiver gain, squelch, active antenna and battery level. Both transmitters use a pair of AA batteries with a claimed battery life of 14 hours indicated by a four-segment display (on transmitter and receiver), which denotes nine, seven and three hours of life as each segment disappears.

Key Features „ One-touch auto scan and sync „ Available as 40 Series system (32 MHz of spectrum) and 60 Series (64 MHz) „ Durable metal housings „ All transmitters work with all receivers „ Minimum ten hours of battery life RRP: $1,400


When there are no segments left the battery indicator disappears from the receiver and flashes on the transmitter to let you know you have just one hour left. If the transmitter is turned off or goes out of range a red warning triangle appears at the top of the receiver display to indicate link loss. Audix places a lot of emphasis on the ease-of-use of the system and I found it was indeed very straightforward to use; the menus on the receiver are very intuitive. All you need to do is hold down the up or down buttons and it will scan for a frequency that’s not in use. You then press the sync button and hold the transmitter’s IR window in front of it and you’re good to go, or as Audix puts it: Scan-Sync-Play. The display defaults to showing group/channel numbers but can easily be switched to frequency mode for those who like to know what’s going on under the hood; alternatively a quick tap of the Set button will also briefly display the frequency. There are ten groups of ten channels, which are grouped to prevent intermodulation

issues. One thing that caught me out was the fact that when you change the group on one side of the dual unit it also changes the other side. This is of course completely logical as you would naturally want to operate multiple units in the same group, but that didn’t stop me losing a lock a few times while I was fiddling. The H60 handheld microphone transmitter is black, metal and non-tapered with a stubby plastic antenna at the end. The display is small but conveys all you need to know clearly (i.e. frequency, group/channel numbers and battery indicator); there is a single button – a long press turns it on/off and a short press mutes the microphone. The mute function is completely silent – I couldn’t detect any noise whatsoever as it smoothly transitions. For those who are worried about unintentional muting during a performance, the mute function can be turned off via the receiver menu (which requires a re-sync to transfer the setting).

In Use In operation the signal delivered by the R42 is very clear and free from noise – there was none of the hiss or interference you often get. I must say I was very impressed with the VX5 capsule – it has a very smooth, silky sound, which worked particularly well with a female vocalist, and it admirably handled a wide range of levels and consistently delivered a clear, crisp sound that required very little in the way of EQ. I’m pleased to say that the battery life was also as reported – I got just over 14 hours of use out of it in rehearsals and the display accurately reported the remaining time. This gave me the faith to run it lower than I normally would before replacing the batteries, which should enable more efficient use. The B60 bodypack boasts the same display and features as the H60; input is achieved via a locking mini XLR socket. At first I coupled it with the HT5 headworn microphone (pictured, above). The wire frame is light and

comfortable once fitted correctly – after a while I forgot I was wearing it. In use it’s unobtrusive and it picked up my voice brilliantly with an impressive frequency response, which rolls up from 20Hz to be pretty flat from 45Hz to 2KHz, then dips slightly at 2.5K and more so at 8-10k, which produces a clear, even tone while not being too sibilant. In addition to the same power/mute button as the H60, the B60 also has three buttons that allow access to the settings for RF amplitude (10 or 40 milliwatts) and gain (0, -6 and -12dB) – the same settings are accessed via dip switches inside the H60. Using the instrument cable (a 1.4in male jack to mini XLR) I hooked up a guitar and again I was impressed with the clarity of tone and minimal noise – it was almost as good as a high-quality cable, and definitely one of the better guitar tones I’ve heard on a wireless system. In the Performance Series Audix have produced a versatile high-quality wireless system at a competitive price, which is likely to give the “other two” wireless manufacturers a serious run for their money. It sounds great on a wide range of input sources and delivers a clear noise-free signal while being incredibly easy to set up and maintain.

The Reviewer Andy Coules is a sound engineer and audio educator who has toured the world with a diverse array of acts in a wide range of genres.

May/June 2017




Simon Allen gives us his take on the latest version of the audio repair suite, which promises some significant updates this time.

Key Features

ow on its sixth version and with a price tag for the Advanced option greater than some complete DAWs, this isn’t just a noise removal plug-in anymore. As a daily user of RX since version 2, I was really keen to check out the latest update and put it to the test. There was a lot of hype around this release, giving the impression RX 6 doesn’t just have some neatly presented new features, but offers some fundamental processing advancements. I soon realised this was the case and any audio professional will appreciate the wizardry here.


Talk of the trade Whenever I find myself in those alltoo-common geeky conversations with other like-minded professionals, once you’ve discussed various EQ plug-ins, the 34

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subject of RX soon comes around. If you haven’t got to grips with this extensive software, well, where have you been? I have used CEDAR hardware products before and other noise removal plug-ins, which still hold their merits, but iZotope has become the household name. On that note, I ought to mention that I still believe there’s a strong argument for DSP-based solutions with external hardware, including the ability for real-time processing in some post or broadcasting scenarios. There are some algorithms inside RX that use a surprising amount of CPU, which could be a consideration for some. That said, although some RX algorithms require heavy processing power, the result is nearly always worth the wait. iZotope has ploughed so much R&D into RX that it not only sounds good, but is more efficient. RX 6 is no exception, with the latest modules benefiting from

a new detection and processing method called ‘machine learning’. I find the developments we’ve seen in RX over the years very timely. Historically, we’ve had noise removal tools, such as Dolby tape noise reduction, from very early on. However, I feel as if this technology is even more relevant today. Recording and filming budgets are generally much lower than ever before, with even tighter time constraints. You don’t always have the luxury of going back to the studio to re-record something, just because a car went past the window at the wrong time. Even working mostly in music as a mix engineer myself, I probably rely on RX more than any other bundle of plug-ins.

For Post It’s all about detection in this game. If the software can accurately determine

„ Incorporates new ‘machine learning’ algorithms „ Sophisticated De-ess module „ De-bleed module for removing signals that have bled into another „ De-Rustle and De-wind for separating dialogue from unwanted noise „ Composite View for editing up to 16 individual audio tracks RRP: £315 (Standard); £939 (Advanced) what’s the noise and what’s the source signal, then steps to remove that noise prove simpler. This is what version 6 is all about. If RX 5 was a software and frontend overhaul, then version 6 brings new and exciting detection algorithms. iZotope has used this ‘machine learning’ to train RX to understand the nuances between the noise and source. My impression, even from my regular musical standpoint, is that post-production professionals are gaining more than they might realise in this release. The new De-Rustle (which was conceived using machine learning, along

TECHNOLOGY REVIEW with the Dialogue Isolate feature) and De-Wind (which wasn’t) modules are perhaps the most impressive. We all know how a rustling lavalier microphone can cause hours of work to become usable. Additionally, we all know how wind on a mic can damage the whole signal, not always being contained to the lower frequencies. Well, I have been very impressed at how RX is deciding what’s the dialogue and separating it from the unwanted noise. This new technology has also given way to the Dialogue Isolate module. Until now, we’ve used noise removal tools to suppress noise, in the hope the dialogue becomes clearer. However, RX can now pick out the dialogue from a recording that might have complex and irregular background noises.

For Music iZotope has clearly recognised the increasing number of users who are also employing RX software to clean up their music recordings. Version 6

adds a couple of refinements to the existing spectral De-Noise and De-Clip, as well as introducing new De-Ess and De-Bleed modules. The De-Ess module, which users in post-production will also find useful, has to be one of the most sophisticated de-essers I’ve used. Using the spectral display, you can be sure RX is processing only the correct frequencies, to avoid the signal sounding overprocessed. Then by setting it to output the ‘esses’ only, you can hear exactly what RX has detected as the offending ‘ess’ sounds. The results leave you feeling reassured that you haven’t sucked the life out of the recording. Finally, there is the all-new De-Bleed module. It wasn’t until I started using and understanding this module that I realised the power available here. This module is designed to remove a signal that may have bled into another. The prime example would be headphone click track bleed into a vocal or instrument microphone.

We may have been able to remove click track bleed before with RX, but not quite as effectively, or anywhere near as quickly, as this module can alone. De-Bleed goes further, though, which is a common trait that you discover when using RX 6 in a creative environment such as music production. The simple operation of the De-Bleed module is to provide it with the ‘active’ track and the source track (the source track containing the sound you want removed from the active track). Therefore, this tool can be used to virtually remove any sound from within another, provided you have the original to offer as a sample. This opens up a world of opportunities such as isolating instruments recorded in the same room as one another for further mixing opportunities. I can see this module being developed further in the future, perhaps even isolating multichannel live recordings into its component parts with the touch of a button.

Conclusion This isn’t just another fancy looking update with some well-presented new features for marketing purposes. Version 6 of this fundamentally important tool for modern production just became intelligent. I was sceptical following what I considered to be a disappointing release from iZotope with Neutron, which also claimed to be smart. However, the detection techniques used in RX6 are truly impressive, paving the way for faster workflow and new possibilities.

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 mix engineer continues to reach new heights.

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„ Series II, 2.4GHz digital transmission „ 128-bit encryption „ Range of up to 100 metres „ Locking XLR input and headphone output on transmitter „ One-touch pairing

Jerry Ibbotson takes a look at this user-friendly wireless solution for news gathering, reporting and other applications. requency Hopping Spread Spectrum, the technology that this product relies upon, is a wonderful bit of tech. It lies behind not just Wi-Fi and Bluetooth but also modern radio control gear. Ever wondered why drones (and RC cars, planes, boats etc.) don’t interfere with each other? It’s because their radio systems are busy hopping around different frequencies – the transmitter and receiver working in unison. It’s a robust, interference-free and easy-to-use system, which lends itself to something like a wireless mic kit. That’s why it feels so at home on the Newsshooter, RØDE’s wireless all-in-one setup that is designed for DSLR video shooting but which could find a home in many more applications. I tested the Filmmaker version, which is based around a clip-on personal mic, just over a year ago. The Newsshooter is aimed at handheld news mics and I’ve been testing it with one of RØDE’s own Reporter mics, though it should work with a wide range of other models.

RRP: £459


Overview What is it? It’s a two-part kit: a transmitter that plugs into the XLR on a mic and a receiver that sits on the hot shoe of an SLR and feeds audio into the camera’s mic input on a mini-jack lead. And that’s it. The two units are bound together out of the box and hook up the minute you turn on. What they provide is a solution to the perennial weakness of SLR’s when shooting video: audio. There are various solutions to this, some of which I have tested for AMI. Many are based around separate audio recorders that also feed a secondary 36

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its cylindrical shaft, making it easy to grasp. The Newsshooter TX clicked into place and I was good to go.

In Use output to the camera. The Newsshooter and Filmmaker leave the DSLR to do the actual audio recording, but they do offer something that most other solutions don’t – they get the mic off the damn camera. DSLR video is popular for good reason but a lot of video producers seem content to work with a microphone mounted on the camera. The Newsshooter has the same receiver unit as the Filmmaker. It’s a rectangular plastic/composite block with a hot shoe attachment. There’s a single rubberised On/Off button and a small LED display that shows the signal strength, synch status and ‘channel’. There’s also a coiled lead running to a 3.5mm minijack plug. There’s precious little to fiddle with and therefore precious little to mess up.

So what’s changed? What’s different with the Newsshooter is the transmitter unit. This is a small black brick with the XLR connector rising from the top with a knurled locking ring. There’s actually more to this than meets the eye. For a start, close to the XLR is a threaded minijack/TRS input. This means you could use the TX with a personal mic if needed.

On one side of the transmitter is another minijack socket, this time for headphones. A lack of any monitoring was my only criticism of the Filmmaker. The receiver with this version still doesn’t have any way to hook up headphones (being identical to the Filmmaker unit) but at least the reporter wielding the mic can listen to their own interview at source. I still think a headphone input on the receiver would get over the lack of monitoring on most DSLRs though. Next to the headphone jack is a micro USB input, to allow you to power the TX externally. On most occasions, I’d imagine you’d be using two AA cells. On the main panel of the transmitter is a small LED screen and three rubberised buttons. One is the main On/Off switch while the other two are arrows. Once the unit is powered-up these three work together to allow you to alter certain parameters such as Phantom Power, which input to use, and the gain level applied to the input. These buttons can be hidden from view by a sliding flap. Once both the TX and RX are powered up, they sync and you’re ready to begin. I tested the kit with RØDE’s own Reporter – an omnidirectional interview mic. This is a solid beast with a flattened edge to

I put the kit through a number of tests, including conducting an “interview” on one floor while the RX was downstairs. The signal was unbroken and as clear as a bell. There’s no hiss, no dropout and no interference. As well as testing on a Nikon DSLR I also hooked the receiver up to my Roland R-26 portable recorder and it performed just as well. The RØDELink Newsshooter works. It’s an easy to set up, easy to use and good quality wireless kit that works not only with DSLR video but in any number of situations. The audio quality is excellent and there are a number of applications I can think of where it might come in extremely handy. It’s a great partner to the Filmmaker and will elevate the sound quality on any DSLR video project, as well as being useful in a range of other audio applications.

The Reviewer 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.



Wes Maebe takes this new D/A design from Dave Hill for a test-drive. first met Dave Hill from Crane Song and Dave Hill Designs around the start of the millennium at an AES show and was intrigued and blown away at the same time by his EQs, compressors and monitor controller. Since then I’ve been using Crane Song hard and software on about everything I work on and one can’t help but be impressed with the precision of the engineering, the immaculate design and the sheer quality of the sound that comes out of the Wisconsin workshop. Let’s have a closer look at Crane Song’s latest: the Solaris Quantum D/A Converter.


Not Just A Pretty Faceplate Straight out of the box, the front panel shows the user those classic Crane Song colours: brushed metal grey and green controllers. As it’s a standalone DAC there’s not a lot to clutter up the front panel. Nicely laid out across the front there’s the Power switch; the Mute button for muting the main output; a Source selector which allows you to choose between AES, SPDIF, Optical and USB; a Gain output level control; a 1/4” Headphone socket with its level control and in the middle sits the LCD display. The I/O options on the back couldn’t be more straightforward. The AES runs on XLR, SPDIF on Phono, Optical goes in via Toslink and lastly, USB. In addition to the stepped main outputs, which incidentally deliver up to +24dBu, the Solaris offers an extra set of fixed outputs, set at +18dBu. Before we dive into the guts of the box, we should have a quick look at what information the LCD display gives us. Once the unit is powered up and has gone through its boot up procedure, the function display shows the Source 38

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(SPDIF, AES, USB and OPTICAL), which is selectable with the Source encoder. This is followed by the auto-detected sample rate (up to 192kHz), Gain – again with its own dedicated controller – headphone level and the Left/Right output meters. When you power up the Solaris and hold down the MUTE button until the red SOLARIS name switches to full black screen, you’ll engage the Setup Menu. This offers you the option to engage a 6dB PAD on the Main output. Simply select ON or OFF with the SOURCE knob and hit MUTE to exit. This pad will reduce the Main output to +18dBu from its standard +24dBu, making it an identical parallel of the secondary outputs. Another function you can change is the GAIN SCALE of your unit by changing the SCALE TYPE from ATT to AVO. When you switch the SCALE TYPE, the display scale for the GAIN will change. In ATT mode the maximum output number is 0dB. When AVO is selected the maximum output number is +12dB, mimicking the Crane Song Avocet Monitor Controller.

Crystal Clear So what makes this DAC so special? It’s all about that magic word: jitter, or lack thereof. The Solaris is based around a 32-bit ASRC converter which up samples to 211kHz in order to reduce that jitter. Jitter is commonly caused by an unstable clock. This unit’s reference clock’s jitter has been measured at less than 1ps (picosecond). When you combine that with the Crane Song proprietary re-construction filter you end up with around 44 femtoseconds within the auditory spectrum of 10Hz - 20kHz. To put it in perspective, this is the lowest published jitter spec in proaudio to date. So, what does all this geek speak actually boil down to? Plain and simple,

after two years of research, the Crane Song Solaris delivers amazing transient response and stereo imaging, tight and super defined low frequencies and a crystal clear high end.

An Ocean Of Options As I mentioned earlier, the Solaris DAC is a flexible machine. That versatility allows you to customise its setup to your specific needs. Here are a few of those configurations I ended up using to test-drive the unit. DAW monitor controller: When working entirely in the box, you will still want to benefit from top notch D-A and ultra-low jitter conversion. Hook up the Solaris to your DAW via USB, using the unit as your interface. The Solaris will convert your mix from the digital domain to analogue to feed your monitors. On-the-go conversion and monitor control: There will be situations where you find yourself away from your studio setup needing to work on mixes. Solaris gives you that clear image you’ve been working with in the studio. Just plug in via USB and you’ve got your Crane Song sound feeding your portable rig. Take it a step further and feed the backing track mix to FOH and deliver the pristine sound to be part of the live performance. Digital Source Switcher: You may have a few digital sources, DAW on USB, a playback device on AES, another one on S/PDIF and a third unit outputting Optical. You can now easily switch between the various formats. Output Extender: This is how I ended up running the Solaris in my personal setup. You can use the DAC as your master output from the DAW and it can provide an extra set of analogue outs fed from your interface’s digital outputs. You can make the Solaris your DAC for analogue processing. So if you’re running a bunch of analogue outboard to process mix components

Key Features „ 19” standalone digital to analogue converter „ Jitter reduction up sampling to 211 kHz „ Stepped main outputs that can deliver up to +24dBu; extra set of outputs provide +18dBu „ Source selector for choosing between AES, SPDIF, Optical and USB „ LCD display showing input source, sample rate and level metering RRP: £1,699/$1,899 you can feed them to the Solaris to convert them, process them and then record them back in through your A-D Converter. This way this box can also serve as your mastering DAC. That NY Compression Trick: You’ve probably heard of NY or parallel compression, where you take a parallel feed of your drums and maybe the bass, EQ and compress them together and then blend that signal back into the mix. Of course you can approach this old school trick within your DAW, but there’s nothing like doing this the analogue way. Solaris’ dual output architecture allows you to do that. You buss your rhythm parts to the Solaris so you have level control on one set of outputs and then you can route the secondary outputs to your analogue parallel processing. After I heard the Solaris for the first time, I knew I had to have it in my rack. Comparing the conversion to most of what’s available out there, the sound is so defined that it allowed me to work a lot faster and spend a lot less time EQing. You hear what you get.

The Reviewer Wes Maebe is a UK-based recording, mixing, mastering and live sound engineer.



Russ Long takes a closer look at this new compact 24-bit, 192 kHz USB 3.0 audio interface. he PreSonus Studio 192 Mobile is a USB 3.0 audio interface featuring two channels of XMAX Class A microphone preamplifiers. The 24-bit/192 kHz device measures 1.75in high by 12.5in wide by 7in deep. Its small footprint makes it an easy addition to a laptop case or gig bag. The interface provides two front panel mic/instrument inputs (via a combination balanced XLR/unbalanced TS jack) with an additional pair of balanced line inputs on the rear panel. Also on the rear are six balanced analogue line outputs; two pairs of optical digital inputs and outputs delivering up to 16 additional channels of I/O utilising the ADAT protocol; coaxial stereo S/PDIF I/O; word-clock I/O on BNC connectors; power input for the external power supply; and a USB 3.0 port. The mic preamps are digitally controlled (for gain adjustment and phantom power toggling) remotely or from the front panel, adding to the interface’s flexibility. Remote control is via the included PreSonus Universal Control software or by using a DAW to send MIDI Continuous Controller messages. If one or two PreSonus DigiMax DP88 eightchannel mic preamps are incorporated into the system, they can be controlled as well. Its DAW control allows preamp settings to be stored with a session simplifying recall. It’s important to note that the unit’s I/O specification changes as sample rates are increased; when operating at 48 kHz, there are 22 inputs and 26 outputs. At 96 kHz, there are 14 inputs and 18 outputs. At 192 kHz, there are four inputs and eight outputs.



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The front panel includes a headphone output with dedicated volume control and a large master volume control. The main outputs and each of the four analogue inputs have eight-segment LED metering; channels 1 and 2 include a blue LED phantom power indicator. PreSonus is one of the industry’s premier companies to jump on the USB 3.0 bandwagon and has done it quite well. Harnessing the full power of the 192 Mobile requires USB 3.0 but the interface provides more than enough power for nearly any application when coupled with a USB 2.0-equipped computer. Computer integration with the interface is perfectly implemented with the Universal Control (UC) Surface application that also supports control via iOS and Android apps. Obviously designed with touch screens in mind, the application is void of menus and rotary controls, and the spacing of the controls prohibits accidentally grabbing the wrong control when it’s being used with a touch screen. Controlling the I/O with my iPad via the PreSonus UC Surface was straightforward and easy with no complex configuration needed. The Fat Channel (available on all analogue inputs and the first eight ADAT inputs) is PreSonus’ zero-latency software channel strip. It incorporates a compressor, limiter, gate/expander, high-pass filter and four-band parametric EQ. The input channels also incorporate aux bus and polarity invert functionality. UC also provides global reverb and delay built into the mixer. There’s no way to open a configuration and make adjustments without having the 192 live connected to the computer. This is a personal disappointment; I could see myself doing some setup tweaking with my laptop at the coffee shop or on an airplane. While the zero-latency mixer in Universal Control is easy to configure

and use, I found latency to be too high to monitor through my DAW in most instances. When using PreSonus’ Studio One DAW (Studio One Artist is included with the system) the mixer becomes integrated into the DAW, so the Universal Control app isn’t needed.

In Use For this review, PreSonus sent a Digimax DP88 to use along with the 192 Mobile, and integration between the two is simply fantastic. Connecting two optical cables between the two units gives complete control over the DP88 via Universal Control or Studio One. The computer sees the DP88 as eight additional mic pres and it requires no additional setup or configuration. Up to two DP88s can be connected to the 192 Mobile, resulting in 18 remotecontrolled preamps. Recording through the Studio 192 Mobile/Digimax DP88 combination went extremely well. Its mic preamps are clean, smooth and natural sounding – somewhat lacking in character, but versatile all the while. I had success using them to record drums, percussion, electric and acoustic guitars, bass, synths and vocals; in every instance, I received good results. I recorded vocals with an ADK Z-67 tube large diaphragm condenser through the Retro Instruments OP-6 tube mic pre, straight into the line input of the Studio 192 Mobile and received stunning results; this fully convinced me of the converters’ high quality. The 192 Mobile’s headphone amp sounds good. I spent several hours monitoring with a wide variety of headphone models, including AudioTechnica ATH-MSR7, BLUE Lola, and Focal Spirit as well as with my Ultimate Ears UE Pro Remastered IEMs and the sound was consistently impressive. The headphone amp can provide enough volume to put a smile on the face of

Key Features „ 24-bit, 192 kHz USB 3.0 audio interface „ Simultaneous I/O up to 22 inputs/26 outputs „ 118 dB digital conversion „ Fat Channel processing on every analogue input „ UC Surface monitor mix control software RRP: £569 even the dullest-eared drummer. The metering is good and illumination spill between meters is minimal. Fat Channel processing in Studio One can be applied to inputs that are being monitored with low latency. The Fat Channel can be opened, edited and auditioned from within Studio One, although the processing, in reality, is running on the DSP of the 192 Mobile. It even offers the ability to place it in either the monitor or record path. If you are only monitoring the channel, you can drag it into the main insert so you can continue to process the audio with the same settings during playback. This is simply the best integration of a DAW and interface that I’ve seen on any level. As a portable USB 3.0-equipped recording interface, the Studio 192 Mobile simply can’t be beat. It sounds good, has tons of I/O configurability and is amazingly priced. Anyone in need of a mobile I/O interface should give it top consideration, and if you are a Studio One user, this is a must-have.

The Reviewer Russ Long lives and works in Nashville, engineering and producing a wide variety of music and film projects.

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Up for a Special Tony Award and having already seen success with the new Harry Potter play, Pete Malkin is seemingly in a position where he can do no wrong. Adam Savage meets a sound designer at the height of his career so far. cast, and if you’ve read the story, almost every page is filled with potential for interesting sound design. There was a lot of content to create and a lot of people involved. Our approach had to shift from time to time, but we spent a lot of time in rehearsals working with the creative team and actors in order to build palettes of sound effects and soundscapes that felt part of the world. We had a great time working closely with the Music team (Imogen Heap, Martin Lowe, and Phij Adams) as well to create a cohesive auditory world for the story.

Photo: Sarah Ainslie

First of all, congratulations on the Special Tony Award! How does it feel to receive this recognition? It came as an incredible surprise and it’s a surreal experience. I feel so proud of what everyone on this show has achieved and to have our work as sound designers recognised in this way is a real privilege. What was your reaction when you heard they’re bringing back the Sound Design categories? How important was it to you that the Committee made this U-turn? I’m grateful that the committee listened to the appeal of the thousands who petitioned against their decision to revoke the award as well as for finding a new voting process to go along with it. It’s a great step forward. The signatures of the #TonyCanYouHearMe campaign, led by John Gromada, was a driving force towards the decision and I hope The Encounter has helped in its own way too. It’s an important U-turn as it rightly recognises sound design as a creative theatrical art form, much like the work of our colleagues in other creative theatrical disciplines. Your Special Tony is for your work with Gareth Fry on The Encounter. What is it about that show that you feel has been so well received? 42

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I think audiences attach quickly to Loren Mcintyre’s story and his tale of getting lost in the Amazon rainforest – it’s fascinating. Complicitè are great advocates of using new technology in theatre, but the story is always at the core of the work. We place the audience in the action using a combination of the Binaural microphone and headphones, which provides an intimate storytelling tool for McBurney’s performance to lead you through the show with a huge amount of energy. The audience perhaps responds so well because they are granted scope to use their imagination to form part of the theatrical experience. We suggest ideas of places, atmospheres, people, feelings, etc. using sound and music rather than the literal physical objects or set on stage, and it’s exciting to allow their imaginations to run with it.

to experiment with new technology, the more we’ll find great ways of integrating it into stories, which will lead to more audiences welcoming these interesting forms. There is a huge amount of innovative sound design work going on not just in theatre, which is always inspiring to hear and learn from.

We loved the clever use of binaural and other methods of audio trickery. Do you feel audiences are starting to welcome more experimentation with sound? To a degree yes! The Encounter is a good example of this. There has been a wonderful response to the nature of the audio experience; of course there will always be some who find it difficult to welcome experimental work. Perhaps the more that we as creators are able

Then there’s Harry Potter and the Cursed Child, of course, which won a record nine Oliviers this year, including the one for Sound Design. We know you can’t give much away, but can you tell us about some of the challenges you and Gareth faced? Sure, we do love to keep the secrets, which is a challenge in itself among plenty others. I can’t say too much, but the show is in two parts with a large

How did your partnership with Gareth come about? I’ve been lucky enough to work on numerous projects with Gareth over the past six or so years. I first got in touch as a graduate and he invited me to see his working process on Complicitè’s adaptation of ‘The Master and Margarita’. This swiftly became my initiation into Complictè. Since then he’s been a brilliant mentor and we’ve gone through all sorts of weird and wonderful projects together.

Now we hear there’s going to be a Broadway version coming next year. When will work start on that? It’s already begun! There will be lots of work over the next year to be ready for our official opening on 22 April 2018 and we’re excited for US audiences to experience it. What other projects have you been working on recently? I recently sound designed ‘The Kid Stays in the Picture’ at the Royal Court and ‘Beware of Pity’ at the Barbican, both directed by McBurney, with the fantastic help of associate sound designer Ben Grant. Both shows demanded heavy sound design and, much like The Encounter, we had two operators on each (Yamina Mezeli and Neil Dewar on the former; Sven Poser and Stephan Pinkernell on the latter) in order to react to McBurney’s style of collaboration. Everything is kept as fluid and reactive as possible. There are a couple of interesting shows in the pipeline with other companies: one will be an experimental piece that will again use headphones, but this time accompanied by VR headsets, so that’s a venture into a totally new world for me. I’m also looking forward to working on The Seagull at the Lyric Hammersmith in London, directed by Sean Holmes, later in the year.

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AMI May/June 2017 Digital Edition  
AMI May/June 2017 Digital Edition