AMI February 2015 Digital

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

February 2015

TURING SOUND We speak to the BAFTA-nominated sound team behind The Imitation Game p26

FEATURE

IEMs: The importance of choosing the right system p28

TECH FOCUS

Loudness audio metering p35

REVIEW

Steinberg Cubase Pro 8 p38

WELCOME

www.audiomediainternational.com

Editor Adam Savage asavage@nbmedia.com

Experts in the issue

Managing Editor Jo Ruddock jruddock@nbmedia.com Staff Writer Matt Fellows mfellows@nbmedia.com Commercial Director Darrell Carter dcarter@nbmedia.com Account Manager Karma Bertelsen kbertelsen@nbmedia.com Production Executive Jason Dowie jdowie@nbmedia.com Production Assisstant Georgia Blake gblake@nbmedia.com Head of Design Jat Garcha jgarcha@nbmedia.com

Press releases to: ukpressreleases@nbmedia.com © NewBay Media 2015. No part of this publication may be reproduced in any form or by any means without prior permission of the copyright owners.

Audio Media International is published by NewBay Media, 1st Floor, Suncourt House, 18-26 Essex Road, London N1 8LN, England.

Editorial tel: +44 (0)20 7354 6002 Sales tel: +44 (0)20 7354 6000 Audio Media International ISSN number: ISSN 2057-5165 (Print) Circulation & Subscription enquiries Tel: +44 (0)1580 883848 email: audiomedia.subscriptions@c-cms.com Printed by Pensord Press Ltd Front Cover: STUDIOCANAL

Andy Coules started his career in the industry as a tea boy in a studio, working his way up to studio engineer before developing a taste for live sound. This enabled him to combine his love of travelling and hotel rooms and tour the world with a diverse array of acts, often in the role of sound engineer/tour manager.

Ryan McCambridge is a freelance audio engineer, writer, producer & programmer from Toronto, Canada.

He has taught audio production in workshops and universities, is the creator of the production blog Bit Crushing and is the frontman of A Calmer Collision. To find out more, go to www.bitcrushing.com or www.acalmercollision.com.

Jim Evans has been involved in and reported on the professional audio and music industries for more than four decades.

Founding editor of Pro Sound News Europe, he has written for Audio Media since its inception and is a regular contributor to a number of leading industry publications and websites. His television credits include collaborations with the late Peter Cook and Micky Most.

Brad Watts has been a freelance writer for numerous audio magaziness, has mastered and mixed various bands, and was deputy editor of AudioTechnology in Australia. He is now digital content manager for Content and Technology.

WELCOME Adam Savage Editor Audio Media International

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very year I think that the appeal of the NAMM Show has reached its peak, and every year I am proved wrong again. NAMM 2014 was a big one for pro-audio – thanks to announcements such as the Midas M32, QSC TouchMix (look out for a review of the mixer in this issue actually, page 40), and Audio-Technica M-Series – but as for the 2015 event, even the more minor stories we reported from the show over the past couple of weeks were hungrily consumed by legions of gear heads eager to take in as many updates from Anaheim as their brains could handle.

So as you can imagine, then, when we got the news out about the major launches from Avid, Allen & Heath and Music Group, the site took a bit of a pummeling. It may have been around since 1901, but there’s no sign of interest in the NAMM Show slowing down any time soon. Quite the opposite in fact – I even spotted it trending worldwide on Twitter more than once over the four days, and I’ve never seen that before with an industry trade event.

Couldn’t make it out to California and want to know what all the fuss was about? Then check out our Product News pages in this issue.

And when you’re done with that, make sure you take a look at our interview with Lee Walpole and Stuart Hilliker of Boom Post, who more than deserve the recent BAFTA nomination for their work on hit film The Imitation Game; an in-depth feature on the in-ear monitoring market; a Geo Focus article on why Italy is a challenging place to be involved in pro-audio at the moment and if you were a regular Audio Pro International reader, you might recognise a few faces when you turn to our Opinion section this month.

I’d also like to say how delighted I am with the response we’ve had so far to our new publication. The feedback has been overwhelmingly positive and we couldn’t have hoped for a better reception, so thank you to everybody who has phoned, emailed, tweeted or spoken to me in person in the last month about the magazine and website. Although many of these comments have been about the smart design of AMI, it was nice to hear that the editorial content is also appreciated! It’s good to know we’re on the right track.

I hope you enjoy the issue, and I’m sure I’ll see some of you at ISE and BVE, let me know if you fancy meeting up!

February 2015

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CONTENTS

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PRODUCT NEWS

MIDAS ADDS TO M32 FAMILY

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ALLEN & HEATH EXPANDS QU RANGE

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SOUNDCRAFT ANNOUNCES UI SERIES

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NEW STUDIOLIVE AI SYSTEMS FROM PRESONUS

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RĂ˜DE INTRODUCES FIRST WIRELESS SYSTEM

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SHOW NEWS

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TRAINING AND TECHNOLOGY AT BVE IN LONDON

PEOPLE

OPINION Ryan McCambridge discusses mixing in the box

Rob Bridgett delves into the changing landscape of game audio

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FINAL CUT Adam Savage speaks to BAFTA-nominated sound team behind recent blockbuster The Imitation Game IN-EAR MONITORING Jim Evans looks at how the IEM market has developed over the years and discovers why choosing the right system is crucial

TECHNOLOGY

HOW TO Harman engineer Brandon Graham offers his advice on the best ways to prevent audio feedback

THE LATEST PRODUCT NEWS AHEAD OF ISE 2015 34

EXPERT WITNESS Ian Shepherd explains why broadcast loudness meters also matter in mixing and mastering

35

FOCUS Loudness Audio Metering: Our round-up some of the most effective tools currently available

Andy Coules on how to deal with a badly-placed desk when mixing monitors

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FEATURES

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INTERVIEW Award-winning sound designer John Kassab on the art of the radio play

ALSO INSIDE

GEO FOCUS: ITALY Economic concerns and government regulation are combining to create tricky conditions for the pro-audio market in the Mediterranean country BROADCAST FOCUS Will Strauss reports on a unique approach to mixing the Grammys

February 2015

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REVIEWS

STEINBERG CUBASE PRO 8 QSC TOUCHMIX-16 DPA IN-EAR HEADSET MIC NEVE 1073LB APOGEE ENSEMBLE THUNDERBOLT

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PRODUCT NEWS

MIDAS ADDS TO M32 FAMILY Midas bolstered its M32 console platform in Anaheim with the new M32R (pictured) and M32C digital mixers, along with the DL32 32 x 16 stage box.

Both the M32R and M32C are 40-input/25-bus systems with a host of connectivity options, aimed at audio professionals in need of “advanced live sound and recording solutions”.

The M32R features the same Midas microphone preamplifiers and ‘1-million cycle’ motorised faders as the manufacturer’s PRO Series consoles.

The mixers also feature ULTRANET connectivity for ‘Acoustic Integration’ with the latest Turbosound active loudspeakers and P-16 personal monitoring systems.

Expanding on the functionality and connectivity of the DL16, the DL32 stage box doubles the I/O in a 3U rackmount chassis with 32 Midas PRO Series microphone preamps, 16 XLR outputs, two AES50 ports, AES/EBU stereo outputs, MIDI I/O and dual ADAT connectivity.

“The M32R takes the elements crucial to our highly acclaimed PRO Series console range and puts them into the hands of new Midas digital customers – at an exceptional price point,” said Graham Rowlands, Music Group vice president of global sales, Professional Division.

The Midas M32R, M32C and DL32 are available at an estimated US street price of $2,999, $999 and $1,999, respectively – and are covered by a three-year warranty programme. www.midasconsoles.com

AVID PREVIEWS PRO TOOLS 12 Avid has announced Pro Tools 12, the next generation of its digital audio software, featuring flexible licensing options.

The company has also introduced new ways for audio professionals and artists to work together more closely through Avid Cloud Collaboration and new services in the Avid Marketplace.

Pro Tools 12 allows customers to subscribe from $29.99 per month, or buy the software outright. With these new options, users can stay current with future software updates as soon as they are released via the cloud, at no extra charge.

Avid Cloud Collaboration enables professionals to compose, record, edit and mix sessions with other Pro Tools users in the cloud, as if they are in the same studio. Others can be invited to collaborate on a session using built-in chat, or find new collaborators through the Avid Marketplace Artist Community.

If the person they want to work with doesn’t already have Pro Tools, they can download Pro Tools First – a free version of the software also new at NAMM – to start working together immediately.

Furthermore, 17 new plug-ins and nine new plug-in bundles are now available in the Avid Marketplace and as in-app purchases in Pro Tools. These include the Eleven Effects Bundle, Pro Series Bundle and First Distortion and First Reverb Delay.

Pro Tools software subscription and licensing options will be available this month. www.avid.com

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FOCUSRITE SEES RED AGAIN

Focusrite used the NAMM Show to unveil its Clarett range of audio interfaces, as well as showcase its latest Red 2 and Red 3 plug-in suites.

The Clarett range, which Focusrite claims are now the best sounding, fastest Thunderbolt interfaces, consists of a selection of audio interfaces combining a new preamp design with Focusrite’s latest Thunderbolt technology, designed to deliver exceptional sound quality with an interface latency of under 1ms.

They’re built to provide clear conversion, 24-bit, 192kHz sample rates and top-class dynamic range, while new Clarett mic preamps replicate the impedance and transformer resonance of the original ISAs. The range includes four devices; the 2Pre (10-in/4-out), 4Pre (18-in/8-out),

8Pre (18-in/20-out) and flagship 8Pre X (26-in/28-out). The latter 2U interface has been designed with the permanent racked studio install in mind, featuring extended ADAT I/O and separate rear panel inputs for mic, line and instrument, as well as dedicated phantom power, phase reverse and high pass filters on every channel.

The Red 2 and Red 3 plug-in suite features 64-bit AAX, AU and VST compatibility, and will be bundled free with all Focusrite Scarlett, Saffire, Forte, and Clarett audio interfaces, in addition to retailing separately.

The new plug-ins are designed to accurately recreate Focusrite’s Red 2 equaliser and Red 3 compressor hardware. The Red

2 EQ offers a six-band design with high/ low pass filters, high/low frequency shelves, as well as fully parametric lowmid and high-mid frequency bands. The Red 3 compressor is a VCA compressor design promising natural sounding dynamic control of drums, bass, vocals, and acoustic instruments.

New and existing registered customers can download the new plugins for free, adding to their Focusrite software bundles. www.focusrite.com

HEADPHONES FIRST FROM AUDIO-TECHNICA Audio-Technica has expanded its studio monitoring headphone line-up and enhanced the functionality of its System 10 wireless solution.

The ATH-R70x (pictured) is the company’s first open-back reference monitor headphone, created in conjunction with Paris-based design agency Arro.

Featuring aluminium mesh earcup housings for a low weight of 210g and breathable fabric earpads, the R70x is designed to be comfortable over long studio sessions.

The ATH-M70x, meanwhile, is intended to reveal detailed nuances in a mix, tuned to accurately reproduce extreme low and high frequency content. Its 5-40,000Hz frequency response is possible thanks to proprietary 45mm large-aperture drivers with earth magnets, and at high volumes the M70x’s 2,000mW maximum power input is designed to keep distortion low while retaining clarity and definition.

Finally, a limited edition version of A-T’s M50x headphones, the M50xDG, comes with the original’s 45mm large-aperture drivers, sound-isolating earcups and high SPL handling, and is designed to deliver natural, accurate sound in high-pressure professional situations.

Also at NAMM was the new rackmountable System 10 PRO digital wireless system, which offers a half rack chassis designed to house one or two receiver units, which can be operated either within the chassis or removed, mounted remotely and connected via Ethernet cable. Receivers can be placed up to 300ft away and up to five chassis can be linked with an included RJ12 cable to build a multichannel system, enabling simultaneous use of up to 10 channels.

Operating in the 2.4GHz range, System 10 PRO provides three levels of diversity assurance: frequency, time and space.

Frequency Diversity sends the signal on two dynamically (and automatically) allocated frequencies; Time Diversity sends the signal in multiple time slots to maximise immunity to multipath interference; and Space Diversity uses two antennae on each transmitter and receiver to ensure signal integrity. www.audio-technica.com

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ALLEN & HEATH EXPANDS QU RANGE

Allen & Heath used NAMM to launch Qu-Pac, a compact digital mixer with a built-in touchscreen and iPad control app.

Designed for live music and installed sound applications, Qu-Pac is a freestanding or rack-mount mixer, offering 16 mono inputs, three stereo inputs and 12 mix outputs on the rear panel. The model can also be expanded up to 38 in/28 out by connecting to Allen & Heath’s family of remote AudioRacks over Cat5.

The mixer offers total recall of settings and preamps, multi-track recording to USB via Qu-Drive, a choice of personal monitoring solutions, channel ducking, multichannel USB streaming and the iLive FX Library.

The Qu-Pad iPad app offers wireless control of the key parameters and settings, freeing the

user to mix the show from anywhere in the venue. Qu-Pac’s high-resolution, full colour 5in touchscreen allows access to all functions, providing a simple interface and a backup in the event of WiFi connection problems.

As well as stereo recording or playback from a USB key, the built-in, 18-channel Qu-Drive can record and playback multi-track and stereo audio .wav files to a USB key or drive. Qu-Pac’s 32 x 32, Windows and Mac-compliant USB streaming interface also makes it “the perfect recording solution for tracking, monitoring and overdubbing” in the studio, the firm says. www.allen-heath.com

NEW PRODUCT DUO FROM RUPERT NEVE DESIGNS

The RNDI active transformer direct interface and R6 Six Space 500 Series were both on show at NAMM last month.

The RNDI utilises a custom Rupert Neve-designed transformer and a Class A-biased discrete FET amplifier to balance an instrument signal – whether straight from a bass guitar or a 1,000W amplifier – with sonics extending beyond 100kHz. The unit can be used in two modes: speaker mode, where the DI is connected post-power amp; and instrument mode, where the instrument is connected to the DI directly.

In speaker mode, the RNDI can handle the output of a 1,000W power amplifier (92 Vrms or 266Vp-p) to capture the full tone of the instrument, preamplifier, EQ, inserts, and the amplifier’s output stages before it hits the speakers, allowing the sound engineer to avoid any bleed or unwanted tone.

In instrument mode, the RNDI’s high input headroom of +21.5dBU is capable of handling not only instruments, but professional line level sources

like interfaces, CD players and drum machines without a pad. This allows the RNDI to serve as a way to ‘re-preamp’ any previously recorded track using an interface and preamp.

The R6 Six Space 500 Series is a rack with more than double the required current for a six-space chassis to prevent the power rails from failing under a higher than expected load. The unit also features LED current metering, a doubleshielded internal power supply and eight channels of balanced I/O options including DB-25, XLR and TRS that may be used interchangeably, all within a solid steel chassis with bumpers to support the unit in both horizontal and vertical configurations. www.rupertneve.com

SENNHEISER LAUNCHES EVOLUTION WIRELESS D1

New from Sennheiser is the evolution wireless D1, a range of digital wireless microphone systems for vocals and instruments. Designed to make frequency setting simple, the units feature transmitters and receivers that automatically pair suitable transmission frequencies and configure gain automatically, with automatic co-ordination between D1 systems.

For multiple systems, receivers will synchronise themselves to accommodate up to 15 channels in an ideal RF environment.

D1 operates in the licence-free 2.4GHz range, continually scanning the RF environment and hopping to another frequency if it detects any interference,

allowing co-existence with Bluetooth and WiFi systems.

Menu controls include a seven-band graphic equaliser, low-cut filter, automatic gain control and a de-esser. In addition, each model employs the aptX Live codec, which promises excellent audio quality and wide dynamics for vocals, speech and instruments over the entire audio frequency range, with an overall latency of 3.9ms.

D1 transmitters can be powered by either standard batteries or USBrechargeable lithium-ion ‘accupacks’. The D1 series also comprises headmic and lavalier systems and will be available from March 2015. www.sennheiser.com

EV STRENGTHENS PORTABLE LOUDSPEAKER RANGE Positioned above the ZLX and ELX series and below the ETX series, the newest member of the Electro-Voice portable loudspeaker range is the EKX series. The compact package is designed for a number of sound reinforcement scenarios, and is ideal for musicians, DJs and live/club/ installed sound.

The EKX series features eight models (four powered and four passive), including 12in and 15in two-way models and 15in and 18in subwoofers. The powered models offer Class D power amplifiers (up to 1,500W), delivering up to 134dB peak SPL via high-sensitivity transducers.

On-board signal processing, four presets, visual monitoring, three-band EQ and more are controlled via a single-knob user interface, with menu navigation via an LCD display. The powered models also have EV-exclusive Cardioid Control Technology, allowing subwoofer output to be steered towards the audience with up to a 35dB reduction on stage when multiple subs are deployed, while a Signal Synchronized Transducers waveguide design on the

full-range models ensures precise and consistent coverage.

Each model is lightweight and built to reduce distortion and accentuate bass extension, featuring compact 15mm wood enclosures with internal bracing, a durable EVCoat finish, aluminium pole-mounts and all-metal handles, proving ideal for portable and installed applications.

The EKX series will begin shipping in Spring 2015. www.electrovoice.com

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PRODUCT NEWS

SOUNDCRAFT ANNOUNCES UI SERIES

The new Ui Series of remote-controlled mixers from Soundcraft feature crosscompatibility with iOS, Android, Windows, Mac OS and Linux devices and can use up to 10 control devices simultaneously.

The Ui12 (pictured) and Ui16 feature built-in Harman signal processing, including dbx AFS2, DigiTech Amp Modelling and more, alongside fully recallable and remote-controlled mic gain and phantom power.

All inputs and outputs feature a noise gate, compressor and real-time frequency

analyser, with a four-band parametric EQ, high-pass filter and de-esser for inputs and a 31-band graphic EQ for outputs.

Dedicated Lexicon reverb, delay and chorus FX busses are built-in, and the mixers feature full show/snapshot recallability. Both also offer independent network interfaces for simultaneous control via WiFi and Ethernet. A twochannel USB media player is featured, compatible with MP3, WAV and AIFF formats, with direct-to-memory device functionality available for the Ui16.

Both models house four XLR mic inputs, two channels of Hi-Z/instrument inputs and a stereo RCA line input.

The Ui12 has four XLR combo mic/line inputs and two balanced XLR Aux outputs. The Ui16 boasts eight XLR combo mic/line inputs, with four XLR aux outputs and an HDMI display connection output. www.soundcraft.com

CELESTION UNVEILS FTX COAXIAL SPEAKER RANGE

Celestion has introduced its FTX range of cast aluminium, ferrite magnet coaxial loudspeakers, which each promise fullrange frequency response in a single selfcontained driver, acting as a single source to improve signal alignment and off-axis response when compared to a traditional two-way system.

Available in 12in, 8in and 6.5in chassis diameters, each model features fully combined LF and HF components powered by a Common Magnet Motor Assembly for improved signal coherence and time alignment.

Each model features a polyimide film HF diaphragm, enabling greater high frequency power handling, while Sound Castle soft clamping assembly decreases diaphragm stress for reduced distortion and greater reliability.

Both HF and LF voice coils are edge wound using lightweight copper or copper-clad aluminium to increase barrel stiffness and enable closer coil wire packing density, leading to improved cooling and increased motor strength.

Demodulation rings are featured to minimise harmonic and intermodulation distortion as well.

Also new from the British manufacturer is the CDX07-1075 neodymium magnet compression driver, which features a 0.75in copper clad aluminium voice coil driving a polyimide diaphragm.

The device delivers 15W rms power handling and 109dB sensitivity across a frequency range of 1,500 to 18,000Hz, with a recommended minimum crossover frequency of 2.5kHz at 12dB/octave. www.celestion.com

NEW REFERENCE MONITOR FROM TASCAM Tascam is embracing the entry-level monitor market with the VL-S5 system.

The two-way monitors employ a 5in curved Kevlar cone with a hightemperature voice coil and damped rubber surround. A 1in silk dome tweeter produces crisp highs without the fatigue of metal components.

The active, biamplified power section produces 70W per side – 30W to the highs and 40W for the low end. The monitor has balanced and unbalanced XLR and 0.25in inputs with a trim control. A large rear port extends the low-end frequency response from 60Hz to 22kHz. Tascam has also announced updates to its DR-680mk11, which now

TRINNOV UNLEASHES THE D-MON

Trinnov has launched a new range of digital integrated monitoring processors.

The D-Mon Series, which was developed in partnership with the Avid third-party programme, has been designed primarily to meet the Eucon protocol specifications and provide Avid customers with a comprehensive alternative to the X-Mon analogue processor that has been used alongside their control surfaces for monitoring functions. It is also ready to interface with Yamaha, Lawo, Calrec and other protocols.

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Based on Trinnov’s Optimizer acoustic correction software, D-Mon offers users assignable functions (main, alternate, cues, talkback, listen-back, headphones), an 18 x 16 internal mixer, digital and

analogue inserts, fully assignable intercom control and more.

The series currently consists of two units. D-Mon 8 (5.1/7.1 studios) features 24 digital I/Os, eight analogue I/Os and

features improved mic preamps, a more stable clock and dual recording. www.tascam.com

analogue high-definition outputs, while D-Mon 4 (St./LCRS studios) boasts eight digital I/Os, four analogue I/Os and eight analogue high-definition outputs. www.trinnov.com

PRODUCT NEWS

PRESONUS ANNOUNCES TWO STUDIOLIVE AI MIX SYSTEMS

PreSonus has revealed two new StudioLive AI Mix Systems, designed to deliver large-format channel counts, recording, powerful DSP and more for under $7,000.

Available in 48- and 64-channel frame sizes, the StudioLive AI Mix Systems consist of two cascaded StudioLive AI consoles, a joining bracket that locks the two mixers together, a PreSonus PRM1 Precision Reference Microphone and a custom dust cover.

StudioLive AI Mix Systems perform as a cohesive mixer and offer a surfacedriven workflow with one-to-one control over every parameter.

Each channel is routed through the bus outputs of the master mixer and every global setting is controlled from the

Master to provide integration between the two cascaded consoles.

The StudioLive 64AI Mix System combines two StudioLive 32.4.2AI mixers to create a 64-channel system with 80 x 66, continuously bidirectional FireWire 800 recording interface, 26 mix busses including eight effects busses (for four reverb and four delay processors), and 558 discrete EQ and dynamics processors.

Meanwhile, the 48AI comprises two cascaded StudioLive 24.4.2AI mixers, resulting in a 48-channel console with onboard 64 x 50 FireWire 800 recording interface; 22 total mix busses, including eight effects busses and DSP processing on every channel and bus for a total of 438 EQ and dynamics processors. www.presonus.com

NEW PIONEER PRO HEADPHONES Pioneer has introduced its HRM-7 monitor headphones – professional studio tools designed to deliver accurate, neutral sound to dance music producers.

The HMR-7 features a newly developed 40mm HD driver unit with copper-clad aluminium wire, designed to produce neutral, high-res sound up to 40kHz and enhance bass response thanks to dual airflow chambers with a three-layer damping structure to eliminate unwanted resonance for cleaner low to mid frequencies.

The fully enclosed ear housings offer a large space around the ear, providing optimum sound isolation, but are also roomy enough inside to enable a wide sound stage and clear audio separation, as well as reduce any ambient noise.

The model also features large ear pads made from hybrid memory foam created for maximum comfort while maintaining prime airflow for accurate monitoring. A freely adjustable headband and velour covers were chosen to ensure the HRM-7 is easy to

wear over long studio sessions without jeopardising sound quality.

Finally, the Pioneer HRM-7 package includes a detachable 1.2m coiled cable, 3m straight cable, replacement velour ear pads and a gold plated 6.3mm stereo jack. www.pioneerdj.com

FIRMWARE UPDATE FOR QSC TOUCHMIX QSC has announced a new firmware upgrade for its TouchMix line of compact digital mixers.

TouchMix 2.0 offers added functionality including support for password protected, multi-level security access, expanded WiFi options – such as wired connection to an infrastructure router – and programmability of User Buttons.

In addition, the TouchMix iOS app has been expanded to include a personal monitor mix application for both iPhone and iPod Touch, and allows users to limit access of connected external devices to one or a combination of mix busses.

TouchMix firmware version 2.0 is free and available to download now from the QSC website.

To read our review of QSC’s TouchMix-16 – the 16-channel version with 20 inputs in total – including an exclusive interview with QSC vice president, professional product management Gerry Tschetter and product manager Jon Graves, turn to page 40. www.qsc.com 10

February 2015

D.A.S. DEBUTS AERO 20A D.A.S. Audio has revealed the Aero 20A compact powered line array system, a new addition to its line array range first introduced in 2014.

Incorporating a new D.A.S. 12in loudspeaker optimised to provide high output and reliability, a light aluminium voice coil bonded to a new fibre glass reinforced cone, an optimised magnet circuit and a new suspension design, the Aero 20A promises exceptional performance regarding distortion, power handling and maximum SPL. In addition, a new voice coil venting scheme dissipates voice coil heat, providing the speaker with a high thermal rating and low power compression.

The high frequency response relies on the M-75N compression driver, which employs a neodymium magnet structure, titanium diaphragm and 75mm (3in) voice coil, attached to an injected aluminium waveguide.

A compact Class D two-channel (800W + 400W) amplifier powers the system, making use of the latest in

switch mode technology, offering performance, predictability and immunity from intermodulation artifacts thanks to the Pulse Width Modulation controller.

Brick wall FIR filters provide alignment between ways in order to achieve uniform coverage all the way to the crossover point, while A-D/D-A converters allow for improved dynamics, lower distortion and ultralow noise levels.

Remote monitoring and control is possible by way of audio management app DASnet, offering users instant and intuitive view of the system status as well as remote control over a range of parameters of up to 256 devices.

A new captive rigging mechanism provides enhanced ease-of-use, allowing angle selection to be made while stacked on the transport dolly.

A locking system can be triggered to secure angles between adjacent enclosures during the stacking and lifting procedures. Safe rigging and precise aiming is achieved thanks to

the AX rigging system and the lowprofile fly-bar that reduces the space needed between the upper rigging point and the top of the array. www.dasaudio.com

PRODUCT NEWS

UA’S MAJOR APOLLO UPDATE Universal Audio has announced its Apollo Expanded software for UA Apollo audio interfaces as part of UAD Software v8.0, which is due to arrive in March.

With Apollo Expanded, users of Thunderbolt-equipped Apollo Twin, Apollo DUO, Apollo QUAD and Apollo 16 interfaces can freely combine up to four Apollos and six total UAD-2 devices – adding I/O and DSP as their studio grows.

Apollo Expanded also marks the introduction of the manufacturer’s

Console 2.0 software, giving Thunderbolt Apollo users new workflow options, with more than 25 new userrequested updates.

Other features include Flex Driver, which lets users customise Core Audio I/O – name, save and share presets for different hardware setups and DAWs – and Star clocking over Thunderbolt, which distributes high-quality clock to all Apollos. www.uaudio.com

RØDE HOSTS LARGEST EVER GEAR LAUNCH RØDE Microphones held its biggest product launch yet, introducing multiple new products, at an exclusive event in San Diego last month.

The focus of the event was the new RØDELink Digital Wireless System, a fully-digital wireless audio solution utilising a 2.4GHz, 128-bit encrypted digital transmission that is sent on two channels simultaneously, providing a 24-bit/44.1k digital audio signal at a range of up to 100 metres.

RØDE also unveiled two new shotgun microphones, the NTG4 and NTG4+. With a new capsule, the NTG4 and NTG4+

(pictured) are designed to exhibit lower noise and deliver higher sensitivity, while on-board digital switching controls a 75Hz high pass filter, 10db PAD and highfrequency boost.

The final product announcement was the RØDE NTR ribbon microphone.

Designed with the ribbon separate to the microphone frame and body for acoustic transparency and to minimise resonance, the ribbon element itself is made from aluminium just 1.8 microns thick. The unit also features a high output, ultra-low noise, low impedance transformer. www.rodemic.com

SHOW NEWS: BVE

LONDON CALLING

On 24-26 February, the biggest names in broadcast and production will converge on London’s ExCeL Centre for three days packed with the latest technologies and a host of expert-led seminars. Here, we summarise what’s on offer.

SEMINARS Some of the foremost figures in the industry will be offering a wealth of expertise in the form of seminars, panels and discussions at this year’s BVE. Highlights include ‘Location, location, location – the challenges and solutions of extreme shooting’, which gets the show started at 11:55 on Tuesday 24 February in the Production Theatre. A panel of experts, including sound recordist and engineer Pete Lee and extreme location specialist Phil Coates, will discuss issues such as temperature temperaments, essential tools for your kit, and how to acclimatise your gear.

Later, at 14:35 in the Post Production Theatre, ‘Finding pure ambience in creative soundscapes – comparing and contrasting best practice audio work across the genres’ is one not to miss. Featuring key audio figures from Molinaire, Clear Cut Pictures and Encore Post, the seminar will cover the best techniques and challenges in audio editing, and provide insights into sound mixes from leading audio post professionals.

Rounding off the day at 16:15 back in the Production Theatre is ‘Connecting sound and vision to achieve the best footage’, which will tackle the challenges of capturing good audio in small live venues, how to plan for future disruptors and troubleshooting on a budget.

And when the Thursday rolls around, so does another seminar gem – ‘Good sound on the cheap’ starts at 15:30 in the Production Theatre, where Simon Bishop, chairman of the Institute of Professional 12

February 2015

What? BVE 2015 Where? ExCeL London When? 24-26 February

The RSP-2318 Smartpanel from Riedel Sound, will discuss tips on selecting a comprehensive, reliable, quality kit collection that will not break the bank.

GEAR And what would a show be without the gear? Thankfully there’s no shortage this year. Riedel will be present at stand F24, showcasing a range of its products, including the RSP-2318 Smartpanel, which features the world’s first control panel designed to serve as a powerful multifunctional user interface, expandable through apps. Also on show will be the Microsoft-licensed STX-200, which provides a single-box solution that brings live contributions from both reporters and viewers into live programming, integrating Skype into the intercom solution, promising more flexible applications and workflows.

HHB is setting up shop at stand J43, and will be demonstrating the Avid S6 and S3 control surfaces running with Pro Tools and the latest plug-ins. The new range of TC Electronic products will also be on show, including a selection of adaptive loudness processing solutions – the TM9 TouchMonitor metering platform and the LCn Loudness Correction plug-in and app, along with RedNet from Focusrite.

Nugen Audio will be co-exhibiting with HHB at J43, displaying its latest range of software updates and extensions.

The MXF Extension for Nugen’s LMB Processor – an offline, file-based loudness

INFORMATION

analysis and correction program for high-throughput applications – allows automated analysis and loudness correction for the audio essence within MXF files. This MXF Extension has been updated to include DPP AS-11 compatibility with immediate availability.

Visitors will also get to check out the DynApt (Dynamic Adaptation Technology) Extension for the LMB Processor, which promises a solution for the intelligent repurposing of audio for TV and streaming at faster-than-real-time speeds.

Nugen Audio will also preview new upmixer technology for Stereoto-5.1/7.1/9.1 upmixing, designed for film and TV production and capable of producing a downmix-compatible upmix with optional dialogue isolation in the centre channel.

At stand D20 Calrec will showcase its new Dante interface, compatible with Hydra2 across the entire range of Bluefin2 consoles, enabling connectivity to every client on a network via a singlewidth 3U interface card. It will also be displaying its Apollo and Summa consoles networked using Hydra2, illustrating how the technology can be used as a management tool to streamline workflow and setup configuration.

Wohler brings its product arsenal to stand L01, including its Tachyon Wormhole automated file-based retiming solution, which enables automated plus or minus run-time adjustment of media assets,

while preserving video and audio quality and closed caption/subtitle integrity. Also on show is the AMP2-E16V Series modular audio/video processing monitor, which marries high-quality audio and one-touch operation with a feature list that claims to be the longest in the industry. Finally, the AMP1-16M Dual 3G/HD/SD-SDI audio monitor provides monitoring of embedded audio in two 3G/HD/SD-SDI streams. The 1-RU system de-embeds and provides metering and monitoring of any or all of the 16 audio channels in the selected stream.

Finally, NETIA will present its new Media Assist software suite, a multimedia asset management system with a full complement of production tools that enables users to “manage all types of content, in any format, on any platform, anywhere, at any time”. www.bvexpo.com

SHOW NEWS: ISE

ALL EYES ON AMSTERDAM

d&b will introduce the Yi series

We take a look at the latest tech you can get your hands on at this year’s extravaganza.

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ith in excess of 1,000 exhibitors taking their latest kit to the halls of the Amsterdam RAI, ISE 2015 is a great opportunity for pro-audio professionals to get hands on with the latest technology.

Coda Audio’s compact line array systems and line source column speakers will be a focus of the company’s presence at ISE 2015. With the CoRAY4i installation line source and CoRAY4Li bass extension, Coda has taken its line array technology and integrated it into compact, highly directional enclosures, delivering high fidelity sound and intelligibility for small to mid-sized venues. d&b will be showcasing its latest solutions for the installation market on stand 7-C175. Following the launch of the Y-Series in 2014, the manufacturer will introduce the Yi family of installationspecific versions, with the Yi7P, Yi0P, Yi8 and Yi12 taking centre stage.

ISE 2015 also sees the European launch of the I-Series modular loudspeaker family from Community Professional.

The series includes point-source, highdirectivity, compact and floor-monitor models in multiple performance levels, providing modular scalability for a range of applications. Matching-height subs, in sizes ranging from compact single 12in models to dual 18in models, complement the full-range I Series models.

The range features high-output woofers designed to offer smooth response, high intelligibility, low distortion and minimal power compression. The two-way models have large-format compression drivers, a choice of six rotatable horns and individually voiced, beamwidth-matched crossovers with single or biamp operation. The three-

way models offer a choice of three rotatable horn patterns and biamp or triamp operation with a combination of large-format HF compression driver and Community’s M200HP midrange compression driver.

DPA, in partnership with its Dutch distributor Amptec, will be showing its latest installation microphones, including the Podium Microphone range. The family consists of the d:screet Miniature Podium Microphone, which offers a directional mic based on an interference tube technology that promises exceptional directivity and off-axis rejection, and the d:dicate Podium Microphones, which incorporate modular capsules from DPA’s d:dicate Recording Microphones.

The company is also showing its d:facto vocal microphone along with its full range of body-worn microphones.

Fohhn Audio will present its new intelligent active and fully digital audio network, AIREA. AIREA is a network-based active solution developed for sound reinforcement applications. The system is based on the most up-to-date speaker technology with integrated DSP technology and two-channel Class D amplifiers. The AIREA master modules come with AES/EBU inputs and outputs.

All AIREA system components are connected by just one single conventional network cable, which supplies power as well as two channels of digital audio and the remote control signals from the master module to each individual speaker.

Powersoft is presenting three new additions to its installation range – the Ottocanali DSP+D amplifiers. The three models of high-performance eight-channel power amplifiers will boast two redundant Dante digital streams, in addition to the

The DXT 900 is the first fully digital PA/VA system to be designed from the ground up by RCF engineers 14

February 2015

features already present on the other versions of the Ottocanali Series.

RCF is set to showcase its new flagship DXT 9000 and DXT 3000 digital voice alarm systems, certified to EN54-16 and EN54-4 standards. The first fully digital PA/VA system to be designed from the ground up by RCF engineers, the DXT 9000 is said to be suitable for use in almost any kind of application, from the smallest school or hotel to huge campuses. The DXT 3000 is built for wall mounting and offers an integrated solution for small and medium, single or multi-zone applications.

Other eye-catchers on the RCF stand will include a new white installation version of the company’s HDL 10-A active line array.

One highlight on the Riedel stand is set to be the Tango TNG-200, the manufacturer’s first network-based platform supporting Ravenna/AES67 and AVB standards. Along with powerful processing capabilities, the Tango TNG200 features two integrated Riedel digital partylines, two Ravenna/AES67- and AVBcompatible ports, two Ethernet ports, one option slot, and redundant power supplies.

Symetrix, co-exhibiting with its Dutch distributor Iemke Roos, will showcase an upgraded version of Radius 12x8. SymNet Radius 12x8 EX offers the same compact 1U footprint as its predecessor, with the added benefit of an expansion slot to future-proof installations and reduce the need to ‘swap out’ DSPs if project requirements change. The expansion slot accommodates all SymNet I/O cards and the newly released SIP-based SymNet 2 Line VoIP Card, which is Cisco- and Asterisk-compatible.

The first AVnu-certified Audio Video Bridging (AVB) audio endpoint reference

INFORMATION

What? ISE 2015 Where? Amsterdam RAI When? 10-12 February

platform has been announced by XMOS and the AVnu Alliance. The XMOS software-defined solution has passed all testing for certification and will now bear the AVnu-certified logo.

The newly certified XMOS AVB hardware and software reference platform is described as an affordable, scalable and production-ready solution that allows customers to easily build a range of AVB-enabled audio products, from single speakers and microphones to complex multichannel mixing desks and multi-port conferencing systems.

Yamaha’s ISE focus will be on the latest additions to its Commercial Installation Solutions (CIS) – the MA2030 and PA2030 compact 2 x 30W mixer amplifiers. Both feature a Class D amplifier, silent operation and direct support for both low and high impedance speaker systems. For maximum flexibility the MA2030 adds two-channel mixing with source EQ (Bass/Treble), plus several of Yamaha’s DSP technologies. www.iseurope.org

OPINION

WHY I MIX IN THE BOX

How a listening session with Grammy winner David Bottrill inspired producer and engineer Ryan McCambridge to go digital.

RYAN MCCAMBRIDGE

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here was a time not too long ago when admitting that you mixed a song in the box was seen as committing high treason to the audio monarchy.

“Off with his head!” the audiophiles would shout. “There is no reverb worthy of mine ears to hear such a noisy tail.” And I was right in there, ready with a rotten tomato in hand.

I own a lot of analogue outboard gear, which I love. I once asked Wade Goeke of Chandler Limited to turn my TG1 into a tombstone for me when I die because I dare not live without it, even in death!

Being the gentleman that he is, Wade agreed. All those knobs and metres and the touching and turning – I couldn’t get enough of it. I used plug-ins but I didn’t really keep up on the advancements.

People will always say “you’ve got to try this new (insert audio thingy) because it’s (insert positive adjective).” I had always known that there were people working ITB, but I didn’t really think too much about it. I just wanted to keep working and my beautiful tubes and transformers were serving me just fine in doing that.

Then one day I heard a mix that changed everything for me. My friend, colleague and mentor, David Bottrill (Tool, Peter Gabriel, Muse, Smashing Pumpkins, etc), asked me over to listen to a mix. I knew that he had built a home mixing room but I didn’t know much about it. He’d amassed tons of drool-worthy vintage gear over the years and had spent much of his career in front of an 80-channel SSL, so I just assumed that we were in the same analogue camp. We listened to this mix in his car and it was incredible.

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I honestly couldn’t believe how good it sounded. Then the punch line came: it was mixed entirely in the box. In that moment I heard the capacity of digital tools and completely understood that they’d come of age.

Then came the soul searching. Cue me wandering the streets in the rain in a saxophone-supported montage where I look off into the distance a lot. There may have been a scene where I was crying in bed with my TG1. I had spent my life collecting gear and now here I was thinking about reducing it to an interface and some monitors. “Off with his head!”

Half-heartedly, I tried it and it was strange and unforgiving and I so wanted my outboard gear. So I started inserting a few compressors and EQs into my digital world and committed to keeping my master bus analogue, and then I breathed a sigh of relief. “I’m progressive,” I thought, patting myself on the back. That continued only for a short period of time though because I quickly started feeling neither here nor there, all while the imperfections of what was left of my analogue world were becoming more and more apparent.

I would be asked to do mix recalls and I’d catch myself being annoyed that I’d have to change settings on my outboard. “I’m in the middle of something else,” I thought. By this time I’d grown comfortable with the different approach

needed to mix digitally. The tools were there, and they were great, but they were all just a bit different. I was re-learning to use my ears, because I constantly wanted to use my eyes, and I discovered that headroom and gain-staging were everything in a digital mix. It was time to take off the training wheels.

I think that at this stage of digital development, debates of analogue versus digital often miss the point. It’s really easy to get dragged into endless arguments about the sonic differences or get lost in lofty proclamations like how having 20 Fairchild emulations in a song really gave it “that magic”. But there are so many more interesting aspects to the debate and they’re all answered by personal preference.

I know, for me, my workflow became faster and more efficient, and I was able to jump between many projects, which was a huge advantage. That made it easier for me to take on smaller gigs, mainly because I could fit them in between other sessions, or I could do prep work from home or on the road. Not to mention that creating templates that suited my mixing and writing workflows meant that I could be up and running quicker than ever. All of these things are important to me. But I know people who are still working on 2in tape who will cite the same sort of advantages within their medium. Sure, I miss turning

knobs and pressing buttons, but I feel that I’ve gained a lot of other things that I really appreciate. And my contact cleaner bills have gone down, which is a nice bonus.

In the end, I saved my relationship with my TG1 – who’d grown quite jealous of my affair – by always reaching for it and all my other analogue gear when I’m tracking. Even in my love for my Mac and my Universal Audio Apollo, I still get that tingly feeling when I see my outboard’s VU metres lit and pumping.

I’ve made my choice through trial and error, and who knows, one day my workflow may change again. The important thing is that we stop debating and each just work however we feel comfortable. As far as I’m concerned, we have reached the promised land of digital audio, that place that we knew was out there but just hadn’t found yet. I love it here and thankfully I can keep my head while I’m residing.

Ryan McCambridge is a freelance audio engineer, writer, producer and programmer from Toronto, Canada. He has taught audio production in workshops and universities, is the creator of the production blog Bit Crushing and is the frontman of A Calmer Collision. To find out more, go to www.bitcrushing.com or www.acalmercollision.com.

Twitter: @RyanMcCambridge

OPINION

WHEN THE DESK IS IN THE WRONG PLACE Technology may be constantly improving, making engineers’ lives easier, but difficult mix positions remain a common obstacle, according to Andy Coules.

when someone is happy from the way they move because if they’re happy then they relax and you can see them enjoying themselves. When they’re not happy, their movement is more stilted and you can usually tell that something’s wrong so you instinctively check their mix to ensure all is well. This ability to pick up on such subtle cues increases the more time you spend with a band on the road.

Things have certainly changed in monitor world in recent years. When wedges dominated, musicians had a

ANDY COULES

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previously wrote at some length about an issue that blights the daily existence of sound engineers all around the world – the mixing desk being in the wrong place (which can be viewed at tinyurl.com/lsm9yu7).

In that instance I was writing from the point of view of the front of house (FOH) engineer, but it’s a problem that also affects monitor engineers albeit in a somewhat different manner. The way in which it affects monitor engineers highlights the differences between mixing FOH and monitors – two roles that utilise a similar skill set yet require them to be applied in quite different ways.

I was recently asked to do the monitors for the support act on a sold out European Arena tour and, while you will typically find me plying my trade at front of house, I’ve always enjoyed the opportunity to spend some time at the other end of the multicore. Monitor mixing is often considered the less desirable sibling of front of house mixing, or a place where people cut their teeth keeping the band happy before they’re promoted to the glory of FOH. I must admit the first time I was invited to mix monitors I considered it a step down from the job I wanted to do, but once I got into it I found it quite satisfying in it’s own right. When you’re on stage, working alongside the band, you feel much more part of the performance – what you do has a direct effect on the band and how they play so it’s thrilling and 18

February 2015

exciting in equal measure.

At FOH your main role is to piece together a single mix and ensure it propagates to the audience in a way that conveys the musical content in the appropriate manner. So when the desk is in the wrong place it impacts your ability to constantly monitor the evolving mix and react in real time.

Over in monitor world you’re required to piece together multiple mixes to be deployed by in-ear monitors (IEMs) or speakers (often both), each of which is tailored to the requirements of a specific individual to enable them to perform and produce the appropriate musical content. If the desk is in the wrong place it doesn’t affect your ability to monitor the mixes in the same way it does at front of house, as you can still put your listening wedges where you need them or use your IEM pack wherever you are. No, the main problem with the monitor desk being in the wrong place is the key issue of communication.

Mixing monitors is all about communication – the musicians on stage need to be able to communicate to you

what they need in their ears (or wedges) in order to be able to play, so a clear line of sight is vital. When the desk is in the wrong place this can be tricky, especially if you’re the support act. The demands of the headliner’s production often require you to set up in less than ideal places, usually behind their monitor desk, but sometimes you’re shoved so far back as to be off the stage completely, making it difficult to maintain lines of communication.

When this happened to me I noticed something about the subtleties of monitor mixing that had not previously occurred to me. The majority of the on-stage communication is done via hand signals, the universal pointing at something then pointing up or down is usually enough to convey what needs to go up or down in the mix. A simple nod or thumbs up in return is all that’s required to confirm the message is received and understood.

But there is an extra layer of communication I wasn’t fully aware of until my ability to watch the stage was taken away. I realised that I relied quite heavily on body language to judge if people were happy on stage. You can usually tell

degree of control over what they heard, simply by moving about or tweaking their own volume they could help in achieving the desired balance on stage. But in the age of IEMs, which are essentially high attenuation ear plugs with speakers in them, musicians are removed from the ambient sound on stage and thus demand a much fuller mix in their ears. And if that isn’t complicated enough the prevalence of digital desks with their advanced recall abilities means more productions are demanding the ability to tweak mixes on a per song basis to produce more accurate mixes, but I digress.

When I wrote about the desk being in the wrong place at front of house it was an open plea to those responsible for the design of venues to take into account the needs of the sound, which is undeniably a key part of the concert-going experience.

However when it comes to the monitor desk being in the wrong place there is little that can be done – if you’re a support band then you have to learn to deal with these difficulties, it’s the nature of the rock and roll beast. The only way to handle it is to get yourself in the right place as much as possible during the sound check and then do your best during the show. If you have a digital desk the ability to remote control can be invaluable but you may have to just make the best of a bad situation.

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. www.andycoules.co.uk

OPINION

GAME AUDIO: THE CURRENT STATE OF PLAY Walk, Device 6, Sailor’s Dream) and UsTwo (Monument Valley) – rich, polished and gorgeous experiences that are every bit as immersive as those that would pursue a cinematic narrative model.

Producer Rob Bridgett discusses how sound quality in video games has improved over the years, and offers his thoughts on what the future may hold.

THE MIDDLE / THIRD SPACE Of course, this isn’t just a simple case of David vs Goliath. There has also emerged a thriving and innovation-driven middleground, the likes of developers such as Double Fine, Media Molecule (Tearaway), E-line Media (Never Alone), Minority (Papo & Yo) and even Playdead (Limbo/ Inside) are occupying a ‘third space’ in between the small, innovative risk-taking mobile-focused Indie teams and the giant triple-A console behemoths. These ‘third space’ developers tend to leverage medium-sized teams to work on innovative game experiences across mobile and console platforms – usually relying on a downloadable distribution model rather than a boxed product on shelves.

Picture: Ubisoft’s Far Cry 4

ROB BRIDGETT

Rob Bridgett is a producer/audio director at Clockwork Fox Studios in Canada, and runs the blog www.sounddesign.org.uk

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s little as five years ago, when pondering the future video game audio landscape, I don’t think anyone could have predicted the extent to which there would be such a huge schism in how we think of ‘video games’.

Ten years ago, most of us would have predicted a steady increase in visual 20

February 2015

and aural fidelity as games inevitably crept towards a future ‘cinematic’ model, foregrounding photorealism and narrative immersion. To some extent, this is still happening. However, things have changed dramatically, and quickly, on two fronts.

Firstly, the unstoppable rise of Indie developers – studios like Playdead, which have challenged what audiences think of as ‘game experiences’ by going back to the drawing board from both the design and artistic perspective. Their game, Limbo, a black and white platformer with no dialogue, stood out primarily for these reasons, but underneath lay no less of a polished player experience. Similarly, from a technology view-point, Limbo could have been realised on a PlayStation 2, but was one of the biggest critical successes of the PS3/Xbox360 console cycle. In doing so, Playdead created something completely against the grain of the firmly established first-person shooters and third-person open world triple-A games.

In addition to this, the introduction of, and subsequent ubiquity of, tablet and mobile phone technology as a gaming platform, not only pushed a massive reset button on the whole game design philosophy of ‘bigger, longer, more’ experiences, but has provided a publishing

model free of publisher intervention, opening up any potential game or app developer who can code up their idea to put it in an App store and reach an audience.

These two disruptive shifts have also had a significant effect on the way games are conceived, developed and, of course, how audio is integrated.

SO WHERE ARE WE NOW IN TERMS OF GAME AUDIO? When we look around at video game culture right now, what is apparent is that the bigger triple-A productions, and the big teams who make them, are still around and are still pushing the envelope ever further to produce some fantastically rich, and ever-more cinematic, experiences.

Naughty Dog’s output is nothing short of incredible in this regard and the GTA franchise, rather than going stale and seeing critical (or sales) declines, continues to go from strength to strength.

Ubisoft is similarly invested in developing huge franchises like Assassin’s Creed, Far Cry and Watch Dogs and there is still a healthy and lucrative ambition to move into and dominate this space.

On the other hand, we see incredibly innovative and successful games being created by tiny teams like Simogo (Year

IS THERE A SIMILAR SCHISM IN AUDIO PRODUCTION AND TOOLS? The mobile devices that these games run on are capable of some incredibly sophisticated audio rendering, able to run plenty of simultaneous voices as well as reliably streaming sounds and having multiple run-time DSP effects. While the output of the smaller devices is limited to a device speaker or stereo headphones, sophisticated run-time mixing, ducking and grouping is also used extensively to allow games to be presented beautifully.

The same can now be said of browserfocused game and audio experiences.

In terms of tools available for both triple-A and mobile audio production and implementation, the choices are often identical, with software like Audiokinetic’s Wwise being used on both large projects, such as Assassin’s Creed, Alien: Isolation, Bioshock Infinite, as well as smaller titles like Limbo and Peggle.

Across the industry, the tools for content creation turn out to look very similar, with middleware companies that have come into existence only over the last 10 years or so finally finding a prominent foothold in the production landscape across all game genres and types and all platforms. Many of the techniques and processes like mixing, dialogue logic, SFX

OPINION

www.audiomediainternational.com

production and implementation are the same from big studios to small ones. The primary difference to understand between the two extremes of the industry is simply one of scale: less content, shorter experiences, shorter development cycles and smaller development teams on the mobile/indie side of the garden.

However the approach, the overall goal of audio is the same on any scale of project: How can sound convey the experience to the player?

And how can it accomplish this in a polished and non-annoying way?

WHERE NEXT FOR TRIPLE-A SOUND? As the sector is so risk-averse due to increasing production and marketing budgets, we can’t expect too much on the horizon in terms of innovation, certainly not on a design, franchise-model or production style level. However, we can expect many smaller incremental changes.

IMMERSIVE AUDIO It is now entirely possible, indeed just around the corner, that the already object-oriented 3D positional audio sources will begin to take advantage of the recent object-oriented surround formats like Dolby Atmos, more specifically in their ‘home consumer’ incarnations.

With the addition of a run-time translation layer, these technologies will allow 3D positional sound sources that already exist in the game engine – say the positional sounds associated with a game object like an enemy sniper – to be localised as audio objects in such technologies as the Dolby Atmos RMU at run-time.

Introducing height speakers (overhead helicopters, footsteps of an enemy overhead as you crouch under the floorboards) and much finer localisation through multiple speaker arrays will certainly be the next big thing in immersion technology and mixing and will be used to produce some incredible moments akin to ‘ride-films’ for those who have this technology installed at home.

From a marketing standpoint, we’ll no doubt be seeing the likes of the Dolby CP850 in theatres being able to receive the objectbased surround output of a video game console’s sound via HDMI and the game experience could be enjoyed in a fully equipped Dolby Atmos movie theatre – perfect for big game promo events and exclusive reveals, and the like.

integrated, improvisational style of performance and writing – overlapping dialogue, hesitations, magnifying all the flaws of natural everyday speech and bringing these performances into a convincing video game scenario, with transparent, real-feeling AI. This could forever change the way narrative -driven games sound. Imagine a GTA-like experience but with The Wire’s (TV) documentary feel, looseness and believability.

WHERE NEXT FOR INDIE AUDIO? As for the mobile and Indie sector, I’m sure we’ll see things continually and very quickly changing in terms of experimental game styles and tools. Available horsepower will certainly increase much more quickly in this sector than in the console sector. The game creation engine Unity’s most recent update, which will provide much-needed extension to the engine’s audio tools, could also be a game changer in terms of built-in audio scripting and more deeply integrated interactive sound, a gap that has so far been ably filled by the likes of Fabric, Wwise and FMOD.

I see game and art style as being the driving force behind these kind of games, as new, novel and quirky experiences seem to continually drive discovery in this segment of game development. In terms of implementation, this is the segment where I see the most benefit from MIDI and procedural sound creation, not only in terms of memory saving, but especially in terms of more stripped back game aesthetics and timbres.

I see developers in the third-space benefiting from both the increments in triple-A console technology and being able to more quickly assimilate rapid design innovation occurring in the mobile/Indie sector – in fact, the studios that are in the fortunate enough position to be able to leverage an Indie approach with console and mobile technology may be where we see the most innovation and interesting applications of game audio over the next five years.

No matter which area of game development you are involved in, the sector as a whole is thriving and growing increasingly diverse. I, for one, am excited to experience what’s next on all fronts.

PROCEDURAL CONTENT Procedural sound propagation (the creation of sounds at run-time by purely synthesized means) remains a hot area for growth. GTA V boasted a large percentage of its sound effects being created procedurally, and as the techniques and tools become more accessible for designers and content creators to work with, these techniques will almost certainly start to take more of a foothold in the everyday lexicon of game sound designers and composers – perhaps even moving into convincing run-time voice content.

MIDI RESURRECTION On a similar theme to procedural sound, MIDI controlled music is already making a comeback. Games like Peggle recently made excellent use of this with their re-appropriation of older MIDI controlled sound instruments to accommodate lots of musical variety in a small memory footprint. Expect to see more of these older systems undergo a similar resurrection as the game teams on mobile platforms rediscover these forgotten memory-saving techniques.

PERFORMANCE From an aesthetic/style viewpoint, the triple-A sector is still crying out for a revolution in how voice-over is written, approached and performed. Even the most carefully produced and meticulously directed efforts still somehow manage to feel stifled, flat and instructional. A game changer here would be a well-executed, deeply February 2015

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GEO FOCUS: ITALY

WHEN IN ROME Italy is home to a host of distinguished names in the pro-audio industry. But how has it held up in the recent turbulent climate? Matt Fellows investigates.

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ith one of the world’s largest economies and a prodigious cultural presence, Italy would be among the ďŹ rst countries you would imagine to have its head held high, resolute against the eroding waves of ďŹ scal inďŹ rmity. But it looks like we have to continue relying on our imaginations for such a valiant image, because in reality it seems no one has escaped the grasp of the global downturn. “Generally speaking Italy is currently still suering from the global economic crunch,â€? says Giorgio BiďŹƒ, CEO of Italian loudspeaker manufacturer Outline, painting the grim picture that is corroborated by professionals across the industry and beyond.

Val Kolton, CEO of headphone maker V-Moda, echoes the same sentiment on his country’s ďŹ nancial standing: “Italy’s overall economic environment is not good. The unemployment rate is rising on a monthly basis while GDP is steadily decreasing.â€? But he is quick to note that this is not necessarily indicative of the status of the country’s residents. â€?The fact is that Italians are not poor. They are only afraid of the future and therefore spend less.â€?

But while the global recession has shown little regard for geography or sector, the source of the Italian pro-audio slump is no mystery, with industry personnel agreeing unanimously. “Basically the problems come from the budgets,â€? says Stefano Amerio, chief sound engineer and producer at Artesuono Recording Studios. “In the last ďŹ ve years the size of the pro-audio market in Italy has reduced dramatically with low levels of investment, particularly from the public sector,â€? agrees Luca Giorgi, sales director at Powersoft. “This lack of investment has resulted in excessively aggressive competition that has also produced a drastic reduction in the quality of delivered services and installations.â€?

Andrea Guerranti, technical manager at distributor Sisme, says the same, arguing that this problem has arisen directly from the greater economic downturn: “GDP has decreased, and a high unemployment rate and high taxation, combined with the strong credit reduction of the banks, has caused a reduction in public and private investments, and therefore sales. “The professional audio market has always been ďŹ nancially very weak because it is made up of many

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Population: 61 million

small companies which have grown up thanks to the low margins and credit oered by the suppliers and by the banks. It was obvious that this system would start to have problems as soon as sales and ďŹ nancial support decreased.â€?

These slashed investments are only exacerbated when you delve into the issues faced by speciďŹ c sectors of the industry. Amerio explains: “The studio market isn’t good like it was years ago. Most of the well-known studios suer because artists in the last few years built their own recording spaces due to the reduced budget of productions and the global change with the download market. Commercial studios are having problems surviving, especially big ones.â€?

Roberta Ferrari of Forum Music Village concurs: “The Italian market for studios is in a bad position nowadays. Huge and legendary studios, like Forum Music Village, are having a critical period in economic terms. Even top artists are not investing in big productions anymore. The music business has lost its power, with a huge fall in sales and low interest in underground, small or large artists and labels.�

And this situation all comes down to progressing technologies and their eect on the sector’s status quo. “Labels don’t invest money because of less and less income from downloads. No royalties, no investments, no productions start, so no work for studios and for the related companies,â€? Amerio adds.

Ferrari expands: “Ten years ago a soundtrack production lasted two weeks minimum, covering recording, editing and mixing. Today I can conďŹ rm that even the most famous composer of all time will only book studios for six to seven days at most.â€?

The situation may be none too bright for studios, but what about the rental, install and manufacturing sectors? BiďŹƒ puts it simply: “In the domestic rental industry the situation is pretty bad, while for the private install market it’s slightly better.â€?

Elaborating on the situation, Giorgi explains: “The Italian market is characterised by the presence of a lot of audio manufacturers, with several industry leaders, but as is the case with other industries, the Italian market is keen on importing foreign brands rather than using local products.�

How would you say the Italian professional audio market is currently faring?

„

‘In Italy the pro-audio market suffers continuous difficulties, due to the economic crisis and its cultural approach. In particular, locations where Italian bands can record and play on stage are becoming less common, along with fewer musical productions. There has been a worrying increase in studios with below average-level technicians’

Have you seen any trends in technology purchasing?

Is there anything that could be unique to Italy?

„

‘Nothing really new or unique to the market.

Digital consoles and line arrays seem to keep getting most of the attention, with mid-price products attracting prospective buyers previously belonging to both the higher and lower end of the market’

„

‘In Italy there are some brands, like IK Multimedia, Gem and Proel, that are planning to grow their income in 2015. Musicians still continue to buy their items in the EU, where prices are lower’

GEO FOCUS: ITALY

www.audiomediainternational.com

This reinforces the sentiment that many Italian companies vocalise, of Italy’s problematic economic sustainability when taken in isolation from the rest of the world.

Guerranti is one of many who echo this, stating: “Industry developments are highly inuenced by the size of the market and a lot of Italian companies have been forced to look for business outside of our borders for growth, or, in some cases, to survive.â€?

Another frequently cited obstacle standing ominously in this sector is not strictly ďŹ nancial, but regulatory. “EN54 is one of the latest regulations that has aected our industry, even if in Italy the lack of investments has not created many projects where this change could have been applied,â€? explains Giorgi. BiďŹƒ agrees: “As far as the ďŹ xed install market, in my opinion the EN54 rules are signiďŹ cantly aecting the jobs.â€?

EN54 is a mandatory safety standard implemented across the European Union relating to ďŹ re detection and alarm system compliance. With this new set of requirements that installations must now adhere to, many Italian companies are struggling to follow a safety

code that was hardly constructed with the pro-audio industry in mind.

Guerranti explains: “The introduction of EN54 has generated a price increase in the realisation of systems and projects, from equipment and assembling through to testing, modifying market demand and directing more and more clients towards certiďŹ ed products.â€?

A common theme that arises through discourse with Italian industry professionals is that of survival, which in itself gives a vivid sense of the industry landscape. And when the landscape is harsh and unforgiving, you have to be resourceful if you are to survive.

Giorgi attests that the key to prosperity in the Italian market lies outside its borders. “Powersoft has historically worked more abroad than in the internal market and this has guaranteed the growth path of the company in the past 10 years. Only recently has our domestic market begun to produce signiďŹ cant results.â€?

However, Amerio argues that quality of service is what has enabled them to remain an industry competitor: “Artesuono Recording Studios survives because we oer high quality at reasonable rates. My idea is to oer the

best quality at every stage along with good equipment.

Quality and knowledge make a big dierence.�

BiďŹƒ shares this view, even going as far as to assert that quality of product is the very reason why Outline does not look beyond Italy’s borders from a manufacturing point of view. “We have never indulged in considering our products to be manufactured in full or in part in those countries that enjoy and oer cheaper costs or reduced taxations because we could not guarantee our usual high quality standard,â€? he explains.

Despite these beliefs, the future still looks uncertain, making it diďŹƒcult for anyone to forge a solid game plan. “In the current economic and political situation, making a plan for the future is a gamble. Moreover, the Italian economic situation has been showing no signs of improvement,â€? says Giorgi.

In the current climate, patience may be a virtue and a cautious strategy the way forward, says Kolton: “Italy’s time to change is always longer than countries like the US,â€? suggesting the situation cannot be compared to other markets under similar duress. After all, there may be hope yet for the conditions for recovery and growth to bloom. As BiďŹƒ remarks: “The political situation looks promising as well as stable, so we are conďŹ dent.â€?

What’s having the biggest negative effect on the market at the moment?

Economic uncertainty

How do you expect your sector to perform in 2015 compared with 2014?

Falling budgets Slow payments

Better Worse

Goverment legislation

Stay the same Other

1

2

3

4

5

6

7

8

9

10

0%

10% 20% 30% 40% 50% 60% 70%

February 2015

23

BROADCAST FOCUS

EXPANDING HORIZONS

Inside Eclipse – one of two M3 trucks used at the Grammys

Traditionally, a DAW-based system would never have been considered for mixing live broadcasts. When it comes to music shows, however, things have changed, as Will Strauss discovered when he met Music Mix Mobile.

I

n the US, the Grammys are a big deal.

When the 57th Awards is broadcast live from Los Angeles later this month, somewhere in the region of 28 million Americans will tune in to watch it on CBS. And they will see some of pop music’s biggest acts, from Pharrell Williams and Beyoncé to Sam Smith and Ed Sheeran, picking up awards and performing.

Interestingly, despite being a live show, where reliability is everything, the performances that the audience will hear back home, rather than being mixed using a dedicated broadcast console, will actually be handled by a control surface and DAW running on a Mac Pro. Although it goes against industry tradition, that is the preferred set-up for Music Mix Mobile (aka M3), the remote facilities company that will provide the broadcast 5.1 music mix, and is a great example of how DAW-based mixing has evolved to the point that it is now closer to the gold standard of DSP-based consoles.

As with many outside broadcasts and live events, a lot of the hard work on the 24

February 2015

Grammys is done in advance, in this case during rehearsals.

Two of M3’s five trucks – Eclipse and Horizon – will be used, with mixing handled in an Avid Pro Tools HDX environment on a D-Control surface via Avid HD MADI I/O interfaces and DirectOut routers and A-D/D-A converters. At the front-end, a dedicated technician will dial all the acts via recallable Grace Design m802 preamps, viewing high-resolution metres on a Chromatec MADI-xx monitor.

A Grace Design A-D convertor will turn the signals into MADI and RMEs before they are brought up to the Eclipse truck.

From there the live mix will be routed up to the console and into recorders that can capture 192 tracks in 48k.

The sessions will then be moved digitally into the Horizon truck, which acts as an offline vehicle, allowing fine-tuning and tweaking before the finished notes and plug-in settings are moved back over to Eclipse. Template sessions will then be loaded in order of performance ahead of going to air.

Joel Singer, chief engineer and cofounder of M3, says a DAW-based mixing set-up like this is as reliable as one based around a traditional broadcast console.

“Basically, most broadcast consoles are running UNIX shells or running on a Windows system, or similar,” he says. “If you investigate what a computer actually does you realise that they’ve just chosen a platform similar to what we’ve done. You do have to understand the limitations of the computer though.”

By that he means removing all the unnecessary functionality and applications from what is an off-the-shelf product. “We strip off everything from the Mac that we don’t need and run it just as a music computer,” he continues. “We turn off things that cause issues such as automatic back-ups. If you understand that, you can build a system that is just as reliable as anyone else’s in the market place.”

If modifying the computer is all that is needed, why isn’t everyone doing it? Force

of habit, perhaps, but equally, because there is some integration to do, argues Singer. “Mac, interfaces, D-Control, cards, MADI router: you have to put them all together to make it work,” he says. “Some people just want to spec what they want to spec and let the manufacturer deliver it. It will come in one rack, they will plug it in and [off they go].”

Singer acknowledges that there isn’t huge cost saving to be made from working with Pro Tools but along with access to a huge array of plug-ins, there are logistical and creative benefits: “With 160 channels of Grace preamp you’re looking at $80,0000 to $100,000 of front-end. Add in everything else and you’re still spending $300,000 to $350,000 on a system. But in our world this means that I can do a show, which might go out live and is then re-mixed, and I can give the tracks to an engineer who can boot up the session on his way home on the plane on his Mac Book Pro laptop and edit right there and then.”

Importantly, having started out on the road to DAW-based mixing in 2004, and as they are now using it across various projects, not just the Grammys, the founders of the company feel that they haven’t made a loss on their investment.

“That is the most important thing,” adds Singer. “Previously, we’d spent $800,000 on a console only to be told that it wouldn’t be supported anymore. We had two, at a cost of $1.7 million, and two years later they were being sold online for $32,000.

We decided we could not do that anymore.

Conversely, every time Pro Tools changes, we have increased our feature set.”

While no one is expecting a mobile Pro Tools system to be used for live news and sport, could it be used for other programme genres? Singer thinks so: “We get into some situations with 5.1 broadcast where we have to do multiple stems and I look at other broadcast consoles with their hefty price tags that won’t even come close to that. There are certain things, like on a Calrec Alpha or an Apollo, for sports or other things that need dedicated workflows. Pro Tools doesn’t have that built into it. But it is very easy to approximate what goes on.”

As far as Singer is concerned, M3 trail blazed the use of DAW-based mixing for live TV music and is now reaping the rewards. It’s not for everyone, of course, but others have followed. “It’s not a lone incident,” concludes Singer. “Since we’ve been doing it for 10 years, if it had faults, we wouldn’t be as successful as we are.”

This month’s Grammy Awards coverage will be testament to that.

www. musicmixmobile.com

FEATURE: FINAL CUT

TURING SOUND

It’s always nice to be rewarded for hard graft in audio, especially when it’s a BAFTA nomination for Best Sound. Adam Savage spoke to Lee Walpole and Stuart Hilliker about their work on The Imitation Game.

T

his time of year is probably best known for its trade shows and big product launches, but for one area of the pro-audio industry it’s a period to look forward to for another reason.

For the world’s top film audio crews, February is awards season, and nominated for a prestigious BAFTA this year is the widely acclaimed period drama The Imitation Game – also a finalist in a whopping eight other categories.

The film stars Benedict Cumberbatch as mathematics genius and computer pioneer Alan Turing – the man responsible for cracking the thought-to-be-unbreakable codes of Nazi Germany’s Enigma machine during World War II, and therefore significantly shortening the length of the conflict, according to numerous historians.

Cumberbatch’s portrayal of Turing, Morten Tyldum’s direction and Billy Goldenberg’s editing explain most of the 26

February 2015

film’s multiple nominations, but its sound quality can be largely attributed to the work of Soho, London-based Boom Post.

Crucial to the success of this project were supervising sound editor Lee Walpole and re-recording mixer Stuart Hilliker.

At first it might not seem like your usual candidate for a Best Sound gong, until you discover how much effort went into creating the aural characteristics of the film’s main prop.

“A key thing that Morten wanted to get right from the very beginning was the sound of Christopher, the [code-breaking] machine that Turing built. He was keen to find a sound and language for Christopher and he wanted it to become a character in its own right,” explains Walpole.

“He described it to me as the sound of a thousand knitting needles all working simultaneously and creating a deafening racket. The machine they shot with on set was a replica they built themselves, but

it had no real merits for sound – it made a rhythm but that was it.”

Not the easiest of briefs to meet, then, although the crew did gain access to the only existing replica of the original machine – real name Bombe – giving them the ideal starting point on which to build. The replica at Bletchley Park – Turing’s base during the war – was built from the old plans, enabling them to hear what the great man’s actual machine would have sounded like.

“The one they shot with on the film was just built for the aesthetics and had electronic motors, whereas the machine at Bletchley looks very similar to the one in the film, but makes the authentic sound,” Walpole continues.

“Together with Andy Kennedy [sound designer], Joe Beal [sound effects editor] and Forbes Noonan [ADR mixer], I went along to record the sound of that; we took along a couple of sound devices and recorded multi-track. The mics we used

were a Schoeps CMIT 5 U, Neumann KM 104s, a RØDE NT4 and JRF contact mics.

We recorded every whirr, click and buzz from multiple microphone perspectives with the machine on a full run.”

The results of this initial recording were more than satisfactory, but there was still a lot of work to be done to elevate the sound of the machine to a level that would meet Tyldum’s expectations.

“When we got back with our sounds we discovered that the authentic machine actually matched up rather well with the replica they built on set in terms of the rhythm, rotar speed and pace. It was a good mimic of the real thing, but what the authentic recording lacked was detail, complexity and intricacy,” Walpole reveals.

“When we were there [at Bletchley] we had access to the National Computer Museum next door as well, and they’ve got a whole ton of machines appropriate to the era. We were able to record a lot of

FEATURE: FINAL CUT

www.audiomediainternational.com Picture: STUDIOCANAL

these and that gave us a massive pallette of genuine sounds from Turing’s time. “So we constructed an initial version of the Bombe machine primarily using our Bletchley recordings and a little bit of stu from the Computer Museum, and sent it over to Billy and Morten to lay into the cut and get an original round of feedback. “They felt we needed to bring a sense of scale to the machine and suggested a heart-like element be incorporated into the rhythm, implying a life force, which tied into Morten’s desire for the machine to become a character in its own right.

After a number of failed attempts we found our answer in some old clunky elevator server recordings, which had a wonderful natural reverberant acoustic, suggesting the sound was originating from deep within the machine, so this additionally served to provide us with the sense of scale we were looking for. “We also began to pick out and enhance the visual detail of the rotars as they whirr and click on close-up. Some of the recordings of the machines were running at dierent rhythms but we were able to ďŹ t them to work with the visuals, and they could be tailored to ďŹ t the metronome of what the machine was doing. “We also used handheld cranked movie cameras, telephone relays, clockwork sounds, old sewing machines – they all

provided a palette of high-frequency ticks and ratchets. And then ďŹ nally we used the contact mics that we’d used on the machine at Bletchley, which added another bass element to it, and again helped enhance the scale.â€?

Walpole has been involved with several award-winning movies in the past, including Les MisĂŠrables, so he’s used to these tricky jobs, but trying to recreate the sound of a hugely complex device for a sound-passionate perfectionist like Tyldum can’t have been easy. “Everything in sound follows the same pattern – you build it up from the bottom, but it was diďŹƒcult to tailor it because Morten wasn’t entirely sure what he had in his head, but he did know what didn’t work, and so we sent out four versions before it felt like we were on the right path.â€? “He loves sound and encouraged us to experiment, which was liberating. He knew what he liked and that didn’t change throughout the process,â€? recalls Walpole.

Hilliker adds: “We never felt inhibited in the way we approached the mix or the

with them the sort of accents you would associate with wartime England and it just felt so natural. I think that was a huge thing when it came to bringing Bletchley to life in a fashion where you just didn’t question it.�

Hilliker notes: “As well as the ďŹ lm being ďŹ rmly grounded in reality there are moments when we follow the thoughts of Alan into dierent times and spaces. It’s often the subtraction of sound that helps.

When Alan’s world is falling apart and he’s completely alienated from his colleagues

were over here,â€? reveals Walpole. “There was more of a conscious eort on both of our parts to ensure we were consistently on the same page. I was doing a lot of uploading and Skyping and discussing what was and wasn’t working.â€?

This regular back-and-forth will certainly have helped with the development of the movie’s sound, which could see the Anglo-American partnership justiďŹ ably rewarded when the 2015 BAFTA winners are announced on 8 February.

On location with Morten Tyldum

sound design. He was so open to ideas.�

While re-recording mixer Martin Jensen handled the FX in the mix, Hilliker took care of the dialogue and music, and there was plenty of good material for him to get stuck into as well, which he was able to combine with Alexandre Desplat’s music to excellent eect. “We were given some of the most solid dialogue I’ve ever had and from my perspective that was complemented by Alexandre’s score, which was recorded at Abbey Road and has the most beautiful sound to it,â€? enthuses Hilliker.

Due to its strategic importance, Bletchley Park would have been a hive of activity in the 1940s, which gave the sound crew another opportunity to experiment with various sound ideas. “Throughout the exterior-shot Bletchley scenes there are soldiers walking around constantly and bikes coming and going, so the challenge for us was keeping this sense of active oscreen life on the base throughout the often dialogue-driven interior scenes,â€? states Walpole.

The team’s choice of participants for the loop group recording process allowed them to add further accuracy to the soundtrack too. “In addition to shooting traditional studio-based loop group we contacted a local college in Highgate, got a group of sixth form drama students and took them to Waterloo Park, along with Forbes Noonan, to record the students for some exterior crowd,� says Walpole. “We took a list of topics with us and got them to perform wild tracks. They also brought

L-R: Stuart Hilliker and Lee Walpole

the mix subjectively reinforces the despair he is feeling.� “That was another thing for Morten – what is happening when this genius is thinking and how do we tell the story of what’s going on inside his head? One of our key ways of internalising it was drifting the world away and making it very subjective, almost like a reverby, underwater space,� explains Walpole.

Going back to Boom Post’s relationship with their colleagues over in the US, did the distance between them geographically create much of a problem? “It was interesting with them being in Los Angeles and us being here but we quickly settled into a workow and the nice thing about that was I was probably getting more feedback than I would if they

Walpole and Hilliker were understandably delighted when they got the good news in January – it’s also been nominated for an MPSE Golden Reel Award for Best Sound Editing, Dialogue and ADR in an English Language Feature, as well as an AMPS nomination for Excellence in Sound in a Feature Film – but the pair seem more pleased with the standard of what they’ve produced than the fact that they could be adding to their prize collection later this month, as Hilliker concludes. “It’s a soundtrack we’re certainly very proud of. Usually when it all goes through there will have been something that you weren’t quite able to ďŹ x, but this one doesn’t have any of that. When we had test screenings I hit Play and never wanted to press Stop.â€? www.boompost.co.uk

February 2015

27

LIVE FEATURE

THE EDGE OVER THE WEDGE

In-ear monitoring is approaching its 35th year in the business. Jim Evans tunes in to assess how the market has moved forward, and what to look out for when buying a system.

W

ireless technology has pretty much taken over the world of communications and the live music sector is no exception. Wireless microphone systems have become the norm and, as evidenced at last month’s Winter NAMM in California, remote wireless mixers are gaining in number and popularity.

And in-ear monitoring, having kicked the wedge monitor into the long grass, is now favoured by acts across the spectrum 28

February 2015

– from the smallest pub combos to the mega stadium bands.

The advantages of IEM and the history of their development have been well chronicled. In this respect, readers are warmly recommended the paper, In Ear Monitors A Brief History, written by Sensorcom’s Richard Frankson and sound engineer/IEM pioneer Chrys Lindop.

In-ear monitoring provides a multitude of benefits including the ability to hear vocals and instruments in the mix at levels the artist would like. IEM will also help

singers pitch accurately, with the system also doubling as a hearing protection device. For monitor engineers IEMs have been a revolution both in the complexity of the mix and the fight with artists over levels and the constant problem of acoustic feedback.

In fact, monitor mixing has become considerably more complex as the mix has to assemble all instruments as well as ambient microphones to generate the live feel for the performers, which would otherwise be absent as their

ears are occluded. And for the concertgoer the benefits include significantly improved sound as the floor monitors and side-fills, often at high audio levels, were picked up by the stage microphones and are now used as low level backups or somewhere to put feet and playlists.

In the consumer market IEM earphones are part of the iPod revolution where it is possible to block out sound and create your own listening space in substantially better quality than regular non-occluding earbuds aka: Walkman-style earphones.

FEATURE

www.audiomediainternational.com

Richard Frankson: “Customisation has two functions. It forms an earplug to exclude external sound and allows for the IEMs to be made discrete. They will be very comfortable too as being custom there are no pressure points that a generic one will create. And it also guarantees that no one else can use them.�

Mick Shepherd: “Custom ear monitors are a really good idea... but you have to have the budget. Not only will custom ears never fall out, but also if they’re from any of the leading manufacturers they will sound fantastic – guaranteed.

Having said that many people get a very good result from less expensive

universal-ďŹ t earphones – it’s crucial to ďŹ nd the eartip or foam that gives a snug, comfortable ďŹ t to your ear canal. If it does that the sound will follow.â€?

Tuomo Tolonen: “Custom moulds are popular but on the other hand are expensive and diďŹƒcult to replace on the ďŹ eld. The right universal earpieces can give you outstanding audio performance, with the SE846 (quad driver) earpieces being a great example. They include a patent-pending low pass ďŹ lter design that gives the 846s an outstanding low frequency performance. The rest of the frequency response is also customisable with detachable ďŹ lters.â€?

Martin Fischer: “If you want to have the utmost in audio quality you should invest in customised ear moulds. In many cases, however, generic ear moulds will do a good job too. This decision will depend on your budget – professionally made ear moulds can be quite expensive.�

Michael Santucci: “In our view, custom ďŹ t is essential to realising the potential of the IEM concept. In fact, Sensaphonics makes only custom-ďŹ t products for this very reason. There is obviously a great market for so-called universal-ďŹ t products, but the beneďŹ ts of custom ďŹ t are a core requirement for music professionals.â€?

At NAMM 2015, Sensaphonics showcased two advanced IEM systems designed speciďŹ cally for hearingimpaired musicians. 3D-ME is a Music Enhancement method for those with signiďŹ cant hearing loss, while the 3D-CROS addresses the more profound issue of unilateral deafness. These

On stage with Shure’s PSM300 personal monitor system

“The biggest mistake you can make is choosing an IEM based on driver count.�

Michael Santucci

Mick Shepherd, co-founder of North London-based radio microphone and earphone specialist Hand Held Audio, says: “There have been advances and improvements [in IEM technology] but the central technology remains the same.

A number of systems now use diversity body pack receivers, such as the Shure PSM1000 and AKG IVM4500, while Sennheiser has an adaptive diversity technology where the earphone cable neatly doubles as the second antenna. The

lithium-ion rechargeable battery advances are appearing in some systems too.� “RF technology in general has advanced greatly over the past decade,� suggests Tuomo Tolonen, manager – pro audio group, Shure Distribution UK. “Ongoing changes to UHF spectrum availability has resulted in some outstanding products and IEMs are no exception. Use of wireless systems has increased and they need to operate reliably in ever-harsher RF environments. As the IEM receiver is typically on stage they can be prone to RF noise from sources like LED walls.

PSM1000 is the only real diversity system out there as it operates on two identical antennas and this makes a very noticeable dierence in these kinds of environments.

Another improvement over the last few years has been audio quality with advances in dynamic range, lower selfnoise and frequency response.�

Sensorcom’s Richard Frankson notes: “IEMs started by stripping down Walkman earphones and building them into custom acrylic earshells, which was ďŹ ne, but when it came to ďŹ tting smaller ears it became a problem. Typically the drivers were 15mm in diameter so it could be diďŹƒcult to make the IEMs as small as the client would like.

Manufacturers started to look at hearing aid components, balanced armature drivers, and found that with correct matching they would give acceptable results. The manufacturers of these

systems are speciďŹ cally designed to keep musicians on stage and employed despite mild-to-severe hearing issues, and both are based on the patented Sensaphonics 3D Active Ambient technology platform. “These new systems demonstrate the power of Active Ambient technology,â€? components have since made drivers especially for the IEM market. “As IEMs evolved then multiple driver sets were employed to give better frequency coverage. Typically today in the professional markets dual or triple drivers have become the norm, but in the consumer esoteric markets up to eight driver combinations can be found.

However, short of costing a phenomenal amount of money and producing very high sound pressure levels we don’t really see the acoustic advantage.�

Rob Piddington has been involved with in-ear technology since the early days and is currently working as a consultant with Sensorcom whose IEMs are marketed under the ProGuard brand. He comments: “The basics of IEM technology have remained the same over a number of years, originally using dynamic and gradually changing to moving armature drivers. Moving armature drivers were originally designed to be used with hearing aids. They are used extensively in most of the leading IEM products on the market today; they are much smaller and compact in physical size than their dynamic cousins. The big advantage is they can be tuned and utilised in a speciďŹ c frequency band, multiple drivers tuned then coupled using a crossover network.â€?

US-based Sensaphonics focuses on hearing health, using its advanced technology both for accurate,

says Sensaphonics president Dr Michael Santucci. “Our focus has always been on musician hearing health, something that few if any other in-ear manufacturers can claim. We deliver amazing audio, and we do it safely. The goal is to provide longer careers and better quality of life for musicians and sound engineers.� comfortable stage monitoring, but also as a platform to provide hearing solutions for musicians and engineers who otherwise cannot work due to various impairments.

Company founder Dr Michael Santucci observes: “The core technologies of in-ear monitoring have changed very little in the past decade, as indicated by the fact that most new agship models are notable only for adding more drivers to existing designs. Consumer companies are now dabbling in wireless headphone designs, but due to latency issues and bandwidth limitations, that technology remains impractical for the professional musician.â€?

Santucci further suggests: “The biggest technical advance in recent years for us is the Sensaphonics 3D Active Ambient, which eliminates the need for stage mics by providing full-range, controllable ambience to the monitor mix with accurate 3D dimensionality, while maintaining the isolation needed to realise the hearing health beneďŹ ts of in-ear monitoring.â€?

“High ďŹ delity is a given, and is delivered both by us and most of our competitors,â€? says Santucci. “But then comes the hard part – high isolation, consistent seal and long-wearing comfort. “The biggest mistake you can make is choosing an IEM based on driver count.

Our clients, including some of the most February 2015

29

FEATURE demanding artists in the world, will conďŹ rm that a properly designed IEM can deliver full ďŹ delity with just two or three drivers, preferably with only one crossover to minimise phase distortion.

Excessive drivers are certainly louder, usually more expensive, and provide more parts that can break down. “Sensaphonics IEMs are designed for deep insertion – past the second bend of the ear canal – and are made from exible medical-grade silicone, which has been proven to provide more isolation than other materials. Because the ear canal changes shape with facial movement, silicone earpieces move with them, maintaining the seal. An acrylic earphone that ‘pops right in’ may be convenient, but is poorly ďŹ tted and will not reliably provide the needed isolation and seal on stage. “This is important because losing the seal results in a huge loss of bass response (often the cause of artists pulling one earphone out). Hard acrylic cannot reliably maintain its seal under stage conditions, which is one reason so many manufacturers keep adding more drivers. Beyond the marketing advantages, in my view, it’s also an attempt to compensate for a design aw. “Soft silicone also provides a more comfortable IEM experience, which is very important for musicians and sound engineers who regularly deal with long rehearsals and performances.â€?

Mick Shepherd advises against buying an IEM system that operates on a frequency above 694mHz as there’s a very good chance that availability in that spectrum for radio mic and in-ear users will disappear within the next ďŹ ve years or so. “Look for a system with a solid body pack as they often get dropped. A good wide switching window to maximise frequency choice is advisable for European and worldwide use,â€? he says. “The ability to accept two mono inputs at the transmitter and mix them at the pack is very useful – it allows you to get two discreet mixes from one transmitter and two packs. “If you don’t move around in performance (eg, most drummers) look at a hardwire system – much less expensive.

Again look for the facility to mix two mono inputs – you can run a click track separately this way.� www.akg.com www.handheldaudio.co.uk www.proguarduk.co.uk www.sennheiser.com www.sensaphonics.com www.shure.co.uk www.ultimateears.com

30

February 2015

ProGuard IEMs from Sensorcom Tolonen cautions: “Ensure you get the right system for your application. If you are a global touring act, the tuning bandwidth can be a factor in making sure you have enough of a ‘window’ to get all your channels programmed as spectrum will change from country to country.�

“DeďŹ nitely,â€? declares Shepherd. “We wouldn’t have cheap systems in our hire stock and on the odd occasion that we’ve sold them we’ve had nothing but trouble.

If you have a limited budget it’s worth

looking for ‘pre-owned’ decent quality systems in good condition.

monitors as acoustic feedback is not a concern. But like most things in life you

“It depends what you consider cheap,� says Tolonen. “The bottom line is that your IEM system is what you listen to when you are performing. It needs to sound good, needs to have solid RF performance and also have a feature set that suits your requirements. I’ve heard on several occasions that a good IEM system can make you perform better so it’s critical you get the right one for the job.�

Frankson adds: “Even cheap monitors will produce better monitoring than oor

will very much get what you pay for and with well-built IEMs with MA drivers the sound will create a sense of precision and detail which will make them more pleasant to listen to.â€? “Sound is very subjective and you like what you like – the real answer is ďŹ nding the right balance in quality and the sound you desire rather than cost,â€? says Piddington, while Santucci concludes: “For professional musicians on stage, there is no substitute for quality.â€?

Sennheiser was among the ďŹ rst companies to launch wireless in-ear monitoring systems. After a period of time when the company supplied individually manufactured systems to artists, Sennheiser eventually launched its ďŹ rst professional in-ear monitoring series in 1996. These systems largely corresponded to the current state of the art. “However, since then we have seen quite a few improvements regarding the convenience of IEM systems, and product features have been continually reďŹ ned to make the work of monitoring engineers easier,â€? says Martin Fischer, product manager, Live Performance and Music. “To further improve transmission reliability, for example, Sennheiser introduced diversity reception with its evolution wireless 300 IEM G3 systems and the 2000 IEM Series in 2009. Or take the Engineer Mode, which was launched a little later and enables the monitor engineer to tune into the

beltpacks of the artists and listen to their monitoring signals. “Then on the software side and via the Wireless Systems Manager Software, we created a remotely controllable RF co-ordination tool and included adjustable audio settings for IEM systems. Thus, frequency co-ordination of multiple monitoring systems has become a lot easier, while the audio settings ensure that artists get exactly the kind of audio reproduction they need and want.�

On the viability of digital IEM systems, Fischer states: “Quite a few people are asking about digital IEM systems these days but such systems are currently not feasible because digital systems always involve a certain amount of latency due to A-D conversion and the codec used. “The requirements on IEMs regarding latency are quite high – if the signals travel too slowly, the IEM audio signal and the bone-conducted audio signal will interfere, and frequencies will cancel each other out. As a general rule,

the audio signal should not travel more than ďŹ ve milliseconds across the entire signal chain, (ie from the microphone to the monitor desk and back again to the artist.) At any point in this chain, any digital equipment will add latency to the signal.â€?

Fischer suggests: “The most important criterion is transmission reliability. An artist relies on the monitoring, and a signal loss would be critical. Audio quality is also high up on the list, and most often a stereo signal is preferred over a mono one. “Also, many users ask for high levels, which can be attributed to the fact that many people only switch from wedges to wireless monitoring when they notice that their hearing has already deteriorated. “And last but not least, frequency exibility is becoming increasingly important these days. Frequencyagile systems allow users to select alternative frequencies should any given frequency band be congested.â€?

TECHNOLOGY: HOW TO

PREVENTING AUDIO FEEDBACK

Harman engineer Brandon Graham oers his top tips on setting up and EQing a system to avoid this common problem, as well as explaining the best way to use suppressors.

W

e’ve all probably been exposed to far more feedback than we’d like (even one exposure is one too many).

As performers and live sound engineers, it can be frustrating when we’ve done all that we can and still encounter problems with feedback. This article is intended to help you prevent it, and get the most out of feedback suppressors.

There are many things you can do to passively prevent feedback, and it is important to address these issues ďŹ rst before you start maxing out gains and setting ďŹ lters.

Check the polar patterns of your mics.

Make sure that your monitors are aimed at the nulls in the pattern to minimise feedback loop gain. Keep your mics oaxis from your speakers when possible, because the gain is lower there (especially for high frequencies).

Keep as much distance between your mics and speakers as possible. The feedback gain reduces with distance.

Less is more. The fewer microphones and speakers you have in your system, the fewer feedback paths. These feedback paths can build on each other, so you may get more gain-before-feedback by using less equipment.

Watch out for room reections. They can be strong enough to cause feedback.

Now we reach two conicting goals: achieving the ideal tone for your performance and maximising gain-before-

32

February 2015

feedback. If you are still experiencing feedback problems after doing your best with passive reduction, you will have to start using EQ.

For maximum gain-before-feedback, EQ for a at response coming from your mics.

One broad EQ boost can result in several feedback problems. Better to solve the EQ problem now than to deal with feedback later.

A room full of people has a dierent response than an empty room. This is one reason why a feedback suppressor is useful, because you can never anticipate all feedback problems ahead of time.

As a signal processing engineer at Harman, I contributed to the development of the latest dbx feedback suppression products. During that time I had the opportunity to test feedback suppressors from all major brands, and one thing became clear – everyone has improved their algorithms signiďŹ cantly since their ďŹ rst generation of suppressors. They are better at distinguishing between feedback and music, faster at eliminating feedback, and more careful about how much they notch out of your audio.

In some ways, we often ask automatic suppressors to do the impossible. We can imagine what our high-gain setup would sound like without feedback, so we think that it should be possible to maintain perfect sonic integrity at those levels.

This is like expecting a refrigerator to preserve your food without allowing it to lower the temperature. There may be some tricks it could try, but when it gets really hot something is going to spoil. If we understand the limits of automatic suppressors, we will be more capable of using them to their full potential.

Note: if you operate near maximum gain, be aware that you are adding strong comb-ďŹ lter-like EQ to your sound. Wellplaced feedback suppressor notches can cancel some of that and help you get closer to your desired response.

Your feedback suppressor should be used for two things – revealing and eliminating potential feedback before your performance, and quickly eliminating any feedback during the performance.

Every feedback suppressor has some form of ‘Setup Mode’, which is intended to help you address your main feedback problems before the performance starts.

Use it! It is far better to deal with it now than during the show. Here are some useful tips to make your suppressor setup more eective.

Rerun your setup mode for every show you do. Don’t expect to save a preset and have it be totally eective the next time you set up. Feedback frequencies are dependent on many factors, including mic and speaker positions, where the performers and audience are, and even the room temperature.

As you ring out your system, have the performers stand in their positions (with ears covered) if possible. Feedback often occurs as a performer nears a microphone or speaker.

You can tap the micropones or clap to elicit feedback. Be aware that in setup mode, feedback suppressors don’t distinguish well between feedback and music. Ensure your performers don’t talk or sing into the mics, as you may get some ďŹ lters set on them.

If all of your notch ďŹ lters are lumped together in a small group, this is an indicator of a larger EQ problem. Do a second round of EQ to address the problem and then try again. It’s likely that the ďŹ rst handful of ďŹ lters will address the worst problems and give the most beneďŹ t.

After that, you may ďŹ nd that you can’t increase the gain much before running into several new feedback tones. This is a good indicator that you are nearing your maximum operating gain.

Every feedback suppressor has a ‘Performance Mode’ where it works hard to distinguish between music and feedback.

This is meant to monitor your performance and only kick in if feedback happens. If you’ve set up your system correctly and aren’t pushing the limits with your gains, this should only be an issue when the unexpected happens (ie, when the vocalist points the mic at the monitors).

For the fastest response, use the widest ďŹ lters possible. This ensures that each ďŹ lter tackles the broadest potential feedback region possible. If you’re worried about tone, use the narrowest ďŹ lters and operate at the lowest practical gains.

You can also set your ďŹ lters to autoremove based on a timer. This ensures that they are only active for the short time they are needed during a feedback emergency.

International TECHNOLOGY AND TRENDS FOR THE PRO-AUDIO PROFESSIONAL

New monthly magazine for the entire pro-audio industry covering the latest news, views and technology across the live, studio, installation and broadcast sectors.

Sent by subscription only to 7,500 audio professionals and end-users across the globe.

Quarterly Buyer’s Guide supplements focus on key technologies Q MICROPHONES April (copy deadline 16/03) Q THEATRE SOUND June (copy deadline 12/06) Q MONITORS AND HEADPHONES September (copy deadline 10/08) Q LIVE CONSOLES October (copy deadline 16/09)

New responsive website and newsletter service, bringing you the latest technology news via a series of clickable headlines sent direct to the inbox of over 30,000+ pro audio professionals and end-users, and new custom-built responsive website offering the best possible viewing experience from desktop monitors, smart phones and tablets.

Contact the team for more information: Managing Editor Jo Ruddock jruddock@nbmedia.com Editor Adam Savage asavage@nbmedia.com Commercial Director Darrell Carter dcarter@nbmedia.com

TECHNOLOGY FOCUS: LOUDNESS

Expert Witness Engineer and Perception plug-in developer Ian Shepherd explains why broadcast loudness meters matter in mixing and mastering as well.

W

e’ve all heard about the new loudness standards by now – official, internationally agreed methods of measuring ‘loudness’ and a set of guidelines for broadcast – even legally binding requirements, in US TV. But none of this matters for mixing or mastering, right? There are no rules or restrictions on how loud a CD can or should be mastered, and there never will be. The new ITU regulations only apply to broadcast, so we don’t need to worry about them… right?

Well – yes and no. The new standards and regulations are going to have an impact on the way the music we work on is heard in future, whether we like it or not – and if we don’t pay attention it could seem to suffer as a result. Here’s why.

To cut a long story short, the world is changing. Loudness standardisation is going to be everywhere before long.

Situations where people hear music at the exact level we mix or master at will be the exception, not the rule.

In fact this is already the case in many situations. Loudness regulation has been in place on radio and TV for decades, it’s just that from now on it will be using LUFS instead of peak level to set the standards. Online platforms like iTunes Radio and Spotify already make sure we hear everything at similar loudness by default, and I’m sure the same will be true for YouTube and everywhere else before long. It makes sense – loudness has always been the number one reason for complaints about audio. That’s why the CALM Act was introduced in the US, to regulate the loudness of adverts, because people found them so annoying. And even when people do hear music at the exact level it was mixed or mastered at – played straight from a CD, for example – the first thing they do is adjust the playback volume to whatever they’re comfortable with. So the actual ‘loudness’ of a digital recording is becoming less and less relevant in the real world.

But even if we take it as read that the levels people listen to our music at are going to be messed with, one way or 34

February 2015

another, who cares? There’s nothing we can do about it, so we may as well just ignore it and worry about more important issues, right?

Wrong. Now, and in future, our music’s loudness is going to be measured and controlled by computer algorithms, and this affects the way it sounds in comparison to everything else. So we need to measure it and listen to the result.

And this is where things get interesting.

For example, if two songs were played side-by-side direct from CD, one measuring an ‘integrated’ loudness of -8 LUFS, and the other a more conservative -12 LUFS, the first song would sound 4dB louder – although in reality the end listener would probably turn it down. But on radio, or TV, or iTunes Radio, Spotify, or any other loudness-matched replay scenario, both songs will be played back at the same perceived level. Neither will be significantly louder than the other. So how does that affect how they sound in other ways, compared to each other? The only way to know for sure is to try it ourselves and listen, and that’s why it’s essential to start getting familiar with the new LUFS loudness meters.

We can make some broad predictions though, just by thinking about the numbers.

On TV, where the standard requires an overall playback loudness of -23 LUFS (to allow for very dynamic material like film soundtracks) our -12 LUFS song will be turned down by 11dB, whereas the -8 LUFS song will be reduced in level by a whopping 15dB (!) – meaning our more conservatively mastered -12 LUFS song has the advantage of peaking up to 4dB higher than the ‘louder’ example. It has 4dB more ‘crest factor’, or peak-to-loudness ratio (PLR).

That’s 4dB less limiting, or compression, or clipping, with all the sonic advantages that can bring – or even more, if we choose to mix and master at even lower levels.

And, perhaps most importantly, the extra 4dB of extra crush in the ‘louder’ -8 LUFS master has no benefit, because the playback loudness is standardised. In fact, more dynamic, higher PLR material often sounds noticeably better when loudness is balanced. And all this applies in exactly the

same way on Spotify or iTunes Radio, even though they use different algorithms and higher reference levels – in fact it applies in any situation where playback loudness is standardised.

If you’re interested in experimenting with loudness metering and comparisons for yourself, the key is to match the overall integrated loudness measurements of each song to hear how they will stand up against each other – I recommend adjusting them all to a nominal reference level like -16 or -23 LUFS. More and more DAWs have LUFS loudness metering built-in these days, but there are also plenty of great plug-ins available, too. TC Electronic has been instrumental in much of the research underlying the new units and standards – its LM2 and LM6 Radar meters were among the first and best. I also really like the Nugen Audio VisLM, and MeterPlugs’ LCast, both of which are firmly targeted at pro users. All of these plug-ins feature a loudness history graphic display, which is a feature I find invaluable.

Just to be clear, though, no one is telling us or our clients how loud we should mix or master the music we work on. We can still make things as dense and aggressive

as we like. But we need to understand how those decisions affect the way it sounds when loudness is standardised. The truth is that in future there will be no advantage to mixing or mastering at really high levels, and therefore no need to be ‘competitive’ – and if we go too far, the music could suffer in comparison to more dynamic material.

And actually that’s great news, because it means we can stop worrying about the so-called war, and concentrate on finding the perfect balance between loudness and dynamics instead – doing what’s right for the music. Which at the end of the day, is what we’re all in this crazy business for in the first place, isn’t it?

Expert Witness Ian Shepherd is a mastering engineer at Mastering Media and runs the Production Advice website (www. productionadvice.co.uk). He is the founder of Dynamic Range Day, and has developed the Perception plug-in with MeterPlugs, to help people find the loudness sweet spot for their music. www.perceptionplugin.com

TECHNOLOGY FOCUS

www.audiomediainternational.com

LOUDNESS AUDIO METERING Keeping one eye on the loudness levels in a mix can often be a stressful experience, but it doesn’t have to be, as the metering tools in this tech focus show.

RTW Smart Series The Smart range adds extra functions, facilities and flexibility to the meters in the TM3 series and the dual-stereo TMR7. The ultra-compact TM3S is a six-channel meter offering moving coil emulation of broadcast PPMs and loudness scales. The four-channel TMR7S has two AES I/O, a full range of loudness measurement tools and a 7in screen. The TM3-3GS features 16 audio channels of 3G-SDI with a maximum of eight channels displayed at once, offering DPP-compliant loudness measurement, a 16-channel 3G-SDI de-embedder and a monitoring controller with onscreen fader.

Touchscreen and flexible user interface enables the selection of presets quickly with just a finger swipe Timecode reader on TM3S and TM3-3GS models Loudness conforms to all relevant international standards: EBU R128, ITU-R BS.1770-3/1771-1, ATSC A/85, ARIB

Wohler AMP2-E16V

Wohler’s AMP2-E16V series 16-channel modular audio/video processing monitor provides a complete suite of tools for analysing and managing audio quality, level and loudness, metadata and more. Dual high-resolution LCD screens auto-detect 3G/HD/SD-SDI video and enable monitoring, metering, mixing and routing of embedded AVB, SDI, AES, analogue and TOSLINK audio.

www.rtw.com

Displays critical measurements: loudness (LKFS or LUFS) within window, maximum true peak, maximum loudness, programme loudness and LRA Compatible with all worldwide loudness standards Interface provides convenient colour-coded alarm indications to speed up identification of loudness issues Start, pause, reset and stop functions facilitate manual monitoring, while pause and reset controls support continuous monitoring Adjustable tolerance allows users to enable and disable alarms according to their loudness monitoring requirements

www.wohler.com

DK Technologie es DK-T7 The DK-T7 delivers metering precision and clear read-outs across a full range of audio metering tools, and includes eight-channel AES I/O, HDMI and headphone out and true peak logging and readout.

Bargraphs or Moving Coil – choice of metering tools Real time True Peak PPM – no hidden ‘overs’ SMPTE reader Loudness automation – instant loudness results Features SDI 3G I/O, AES3 eight channel I/O, analogue two channel

www.dk-technologies.com

February 2015

35

TECHNOLOGY FOCUS

Nugen Audio VisLM VisLM is a visual loudness monitoring and metering plug-in that provides detailed, objective loudness measurement, history and logging facilities. It provides an ITU, ATSC and EBU standardcompliant way to measure, compare and contrast loudness during production, broadcast and post-production.

International standards compliant – ITU-R BS.1770, EBU R128, CALM Act Native NLE/DAW operation (PT, Avid MC, FCP, Premiere, etc) Automatic mono through 5.1 compatibility and ASWG 7.1 support Maximum True Peak Level meters Compatible with the Nugen Audio LM-Correct plug-in, a stand-alone tool for automatic, faster-than-real-time loudness analysis and correction

www.nugenaudio.com

iZotope e Insight

iZotope Insight provides audio analysis and metering tools for visualising changes made during mixing and mastering, troubleshooting problematic mixes and ensuring compliance with broadcast loudness standards. Fully customisable and scalable, Insight allows users to visually monitor all relevant information from a mono, stereo or surround mix in a convenient floating window.

Loudness meter readouts show Momentary, Momentary Max, Short-term, Integrated and Loudness Range calculations, as defined by BS.1770 metering guidelines Loudness History Graph retrospectively analyses the loudness of a mix, and monitors loudness trends as they occur BS.1770-3-compliant True Peak Meters ensure no audio will clip during analogue playback Offline Loudness Calculation quickly assesses if current or past audio projects are in compliance Write automation to a DAW to capture when the loudness level is above the target with Loudness Overflow Tracking

www.izotope.com

36

February 2015

TC Electronicc LMn Meters

TC Electronic’s LMn Meters are described as the latest in native plug-ins to measure loudness and true peak on mono, stereo or surround (LM6n) tracks in new versions of Pro Tools and other major DAWs. TC has loaded the native LM2n and LM6n plugins with new dedicated settings and viewing options, improved off-line measuring in Pro Tools and a host of other features.

Radar Display allows users to capture the essence of a loudness landscape at a glance Compliant with all loudness standards Runs natively in Pro Tools and other DAWs, including AAX, VST and Audio Units Off-line measuring in Pro Tools measures audio loudness faster than real time by running as an AudioSuite plug-in

www.tcelectronic.com

TECHNOLOGY FOCUS

www.audiomediainternational.com

Orban Loudness Meter

The free Orban Loudness Meter for Windows and Mac provides real-time monitoring of BS.1770 loudness, CBS loudness, VU, PPM and True Peak for up to 7.1 channels, plus offline, automated file analysis for dozens of audio and video file formats.

Full support for BS.1770-3 and EBU R128 Oversampled peak measurement accurately indicates the peak level of the audio after D-A conversion

www.orban.com

TSL PAM PiCo

PAM PiCo is a compact standalone audio and loudness meter, designed for use in operational positions where ‘at a glance’ audio metering is required. It provides reassurance throughout the production chain that loudness levels are consistent. Loudness logging enables key operators to maintain a database of ‘show by show’ loudness history as a definitive record of compliance to any regional and international standard.

Loudness measurement to EBU, ITU, ATSC and many other recommendations AES and analogue audio inputs EBU Digital, BBC and EBU PPM, DIN, Nordic, VU and many other regional and international scale options Moving Coil Meter emulation including M/S

www.tsl.co.uk

Jünger Audio D*AP Processors

All Jünger loudness control processors incorporate loudness logging for review and archiving, focusing on automatic and adaptive loudness management, and are compliant with all current broadcast audio loudness recommendations. The graphical view of the logs includes an informative and clear before-and-after processing analysis of loudness distribution and range.

Two-, four- and eight-channel integrated loudness, metering and transmission processors Loudness logging with before and after-processing overview Detailed presentation of true-peak level and loudness histories Centralised logging server for multiple broadcast channels Dedicated Continuity and Voiceover processing with loudness matching

www.junger-audio.com

February 2015

37

TECHNOLOGY REVIEW

STEINBERG CUBASE PRO 8 DAW

Alistair McGhee takes a look at what adding the ‘pro’ label means to this audio production stalwart.

M

an on Fire is a standard Hollywood revenge film made watchable by Denzel Washington. As Denzel blasts his way through the layers of corruption every bent copper claims in mitigation: ‘I’m a professional’. But Mr Washington is not impressed by the pro label.

What would he make of Cubase Pro 8?

Yes, the latest version of one of the icons of audio production has added the ‘Pro’ tag and now you have a choice of Cubase Pro 8, Artist 8 and Elements 7.

High on the list of pro features is the addition of VCAs to Cubase mixing. For perspective consider when hardware VCAs arrived on good old analogue desks you were talking mortgage money. And it wasn’t all plain sailing. My colleagues told a story of a live opera broadcast from an OB in Glasgow with our A-Type vehicle 38

February 2015

boasting a desk proudly furnished with VCA groups – cutting edge at the time.

All was going well until halfway through the gig when the control volts fell off the VCAs turning everything up to a Spinal Tap-style 11. Never has Wagner’s Ring been more painful.

Look how far we have advanced – now in 2015 with the latest release of Steinberg’s Cubase Pro 8 you get VCAs for a few hundred notes. In fact, my review setup, which paired a Prism Sound Lyra 1 to Pro 8 is less than £1,500 all in – now that the Lyra has been reduced to under a grand – and what’s more Cubase Pro 8 has no control volts to worry about.

So why the excitement about VCAs? Don’t we already have that functionality in the form of groups and the linking feature?

And the answer to that is yes and no and, well, maybe. A VCA (the acronym

standing for voltage controlled amplifier, reflecting the idea’s origin in proper mixing desks) does not sum together a number of inputs to make a new output.

So you cannot insert a processor into a VCA group output – there isn’t one. In that sense VCAs are examples of linking.

If you want summing to apply uniform processing, then a group works better.

But if you have a range of post fade sends to effects processors, for instance, then a VCA group is a better bet. The sound you are working towards is a mix of ‘dry’ audio, direct from the track, and the ‘wet’ contribution coming back from the processing. Now assuming your FX sends are post fade (and mostly they are) then if you were using a group and you nudge the group fader down, the level of your dry drum sound will decrease, but the amount sent to reverb from a given track will remain the same. This is because the

change in level has taken place at the group stage while the individual track fader is in the same place, sending the same amount of snare to the reverb. What to do?

Use a VCA. The VCA doesn’t sum the track outputs, it changes gain by ‘moving’ the faders of the members of that VCA group up and down. When you push back

The Reviewer Alistair McGhee began audio life in Hi-Fi before joining the BBC as an audio engineer. After 10 years in radio and TV, he moved to production. When BBC Choice started, he pioneered personal digital production in television. Most recently, Alistair was assistant editor, BBC Radio Wales and has been helping the UN with broadcast operations in Juba.

TECHNOLOGY REVIEW

www.audiomediainternational.com

the level of your VCA group, the individual track faders move down and therefore the post fader sends to your effects chains will be proportionally reduced.

I’m sorry to have laboured the point and I’m sure many audio grandmothers are throwing their eggs out of the basket by now, while others are thinking, ‘Ah but I can achieve that effect by dedicating drum processing and sending it to the same group or by using channel link.’ Welcome to the world of cat skinning (don’t try that at home!), but a VCA is quicker and remember, of course, competition is the name of the game and VCA implementations are available in other high-end software, so it is only fitting that Cubase joins the party. Nuendo will get VCAs (and possibly a more powerful implementation) when version 7 arrives later this year.

And, being super flexible, Cubase Pro 8 allows you to ‘VCA’ more than just gain.

Pan, EQ, dynamics, sends, inserts, routing and even automation can all be included in the VCA group. So you have automation on your snare and toms and kick tracks but you also want to apply automation to the drum mix as a whole – the VCA lets you do that. And suppose you have a drum VCA and a ‘glock and marimba’ VCA you can then nest them inside a ‘master’ percussion VCA. So flexible, but enough VCA already.

Frank Simmerlein, Steinberg’s director of marketing, talks to Adam Savage about the latest version of the software, and what it offers both new and experienced users.

There have been a lot of Cubase enhancements over the years and this is a pretty big one. Is this up there with the most significant updates you’ve made perhaps?

Each iteration of Cubase has had its wealth of new features. I believe that maintaining a good balance between improvements to existing features and adding new capabilities makes all the difference. From that perspective all versions of Cubase released in the past 25 years are equally significant.

How important was existing customer feedback when it came to deciding what the new version would offer?

Actually, customer feedback had top priority. We’ve always had an ear for our users but this time it had the utmost priority when it came to enhancements and new features. We hosted surveys, monitored community channels closely and collaborated with producers and musicians working with Cubase. And the result is a wellreceived Cubase Pro 8.

What about newcomers to the software? Are you confident they will also be able to find their way around the software easily?

Cubase is a professional DAW with an extensive feature set.

Our approach to implementing new tools and technologies always keeps the user in mind but admittedly newcomers will have to prepare themselves for a long learning curve. If Pro is too overwhelming I propose going with one of the smaller packages – Cubase Artist or Elements.

What sort of response have you had from users so far?

The feedback we’ve been getting from users has been very positive. And this reflects the fact that we have been carefully listening to the requests of our customer base. A big thanks goes out to all Cubase users.

Why should audio professionals pick Cubase over all the other DAW options available to them?

This question has to go to all the pro users running Cubase in their studios. I’m sure they’ll have the best answer for you.

New Features Quickly backing this thing up. As someone said about Cubase 7.x – ‘the only thing missing is render in-place’. Well the Steinberg feature fairy was listening and Pro 8 has a render in-place function.

You can render tracks, or events, MIDI parts or range selections. There are options for how much of the signal path (processing) you want to include.

Rendering some mono keys into stereo was straightforward and on completion you have the option of a nice shiny new stereo track. However there’s no render to mono – but there are workarounds, of course.

One of my favourite new features is ‘wave meters’ in the mixer. This turns your meter bridge (which normally hides in the ‘Set Up Window Layout’ button top left of your screen) into a set of live scrolling audio meters – only for audio tracks though.

It looks fantastic and maybe having ‘lookahead’ built into the mixer will be a boon for those mixing against the clock who don’t have the screen space for the project screen as well as the mixer.

Actually having audio wave forms directly above the mixer channel is not just cute

but more importantly will impress the client no end. Sometimes that is the definition of ‘Pro’.

Alongside these new features there are two other areas Steinberg has been working on, the first is the most visible – new instruments and effects and the chord pad feature. The second is the stuff that is more mundane or hidden from view – window management and the audio engine performance.

To assess the import of these I spoke to Mal Pope and Andrew Griffiths, the musical powerhouses behind Jack to a King, the story of Swansea City football club, which is currently tearing up the DVD charts. Football fans have come across it I’m sure. Both Andrew and Mal are longtime Cubase users, with track records dating back to the Atari days.

For Andrew it is definitely the lower key stuff that gets the plaudits, he’s using a three-monitor setup with a touchscreen to mix on and the window management in version 8 is something that makes his day-to-day workflow that bit easier and quicker. Similarly, VCAs speed up the daily business of mixing drums. Mal

and I focused on the new audio engine, discussing the importance of squeezing every bit of power and speed out of your setup. Two votes for Pro 8.

Funnily enough the new instruments and effects here might be less attractive simply because most long-term or Pro users already have racks and racks of plug-ins and banks of instruments to call on. The new chord pad feature, however, did remind me of a great story. Andy Partridge of XTC fame once recounted that his search for chords on the piano was very labour intensive – not being a reader of music, in order to remember a chord shape he liked, he would draw round his hand on a piece of cardboard and cut out the chord shape. Andy, with Cubase Pro 8 your cardboard-cutting days are over, because even if you don’t have a degree in composition then the new chord pads will hold your hand, cardboard or otherwise, making musical life that little bit easier. And, of course, if you are starting from scratch the new instruments and effects are nice to have.

So does Cubase justify the Pro tag or is Mr Washington going to come round with

his shotgun and rocket launcher? I think Steinberg can sleep easy in their beds New pro features like VCAs are welcome and the honing of the underlying audio engine and the tweaks to the workflow are keeping the professional user base more than happy.

Match Pro 8 with some top class hardware like the Prism Sound Lyra and you have a high-grade audio system capable of really professional results.

Denzel says yes to Cubase Pro 8 and he, after all, is a true pro.

Key Features VCA faders for complex mixing and automation workflows Render in-place Plug-in manager to arrange, sort and group effects and instruments New Virtual Bass Amp, Quadrafuzz v2, Multiband Expander, Multiband Envelope Shaper effects RRP: £448 www.steinberg.net February 2015

39

TECHNOLOGY REVIEW

QSC TOUCHMIX-16 COMPACT DIGITAL MIXER

QSC’s first foray into the digital live sound mixer field shows you can have professional quality in a compact package, writes Morten Støve.

I

t seems like there is a smorgasbord of new small digital consoles on the market, including the QSC Touchmix-16, which I’ve been using in recent months.

It’s a 16-channel digital console with practically everything you can imagine built-in.

The past few weeks have been a whirlwind: the Kristin Korb Trio tour in Sweden and the United States; a tour with the vocal group Lines for Ladies; and finally, a jazz cruise with JazzDagen from Costa Rica to Miami. Manning the helm at FOH, I’ve tried to travel light with a number of the microphones I need – all DPA and Neumann – a TC Helicon monitor and a small digital console. I typically only use around 10 to 12 channels, so it is not a major production.

You’ve surely experienced this as I have: every venue seems to have a different console. If it’s one of the new digital consoles, it takes a minute to find your way around (while you keep telling yourself that you really are a sound engineer). In touring scenarios like mine, why not travel with your own mixer?

Features The TouchMix-16’s main features include 16 input channels, eight to 10 auxes, four effects channels plus basic recording capabilities. Using its preprogrammability – setting parameters before the gig – setup is very quick; users then simply name main channels, monitor channels and effects.

TouchMix’s built-in Wizard allows effects to be set up swiftly and easily.

[Visit qsc.com/products/Mixers/ Touchmix_Series/TouchMix-16 for QSC’s comprehensive rundown of features– Ed]

In Use I have now been on tour with the TouchMix-16 for several weeks, first with the Kristin Korb Trio – vocal, bass, piano and drums – using a 10-channel set up with only one monitor. My normal procedure is this: I begin by playing some tunes through the PA just to get used to the room; I rig everything else up while the band warms up; and finally, I play my PA test tune – Bill Cantos’ Don’t Say A Word – for EQing purposes. The EQ on the TouchMix-16 – a

1/3-octave graphic EQ with four-band parametric on stereo auxes – works great.

I quickly discovered the Touch-Mix-16’s intuitive nature. On purpose, I did not read its manual before I turned it on. Just like arriving at a venue with an unfamiliar console, I wanted to see if I could figure it out right away. I did, yet when I hit a wall and wanted to flick through a manual, it was right there at the press of a button: I simply clicked INFO on the right side of the mixer. How cool is that?

After the Sweden tour, we did two weeks in Germany for the Lines for Ladies tour featuring five vocals, four monitors plus piano and bass. The shows ranged from a 500-seat church to a variety of small jazz

Joanne Ruddock speaks to Gerry Tschetter, QSC vice president, professional product management, and Jon Graves, product manager, mixers, two of the driving forces behind TouchMix. What was the thinking behind the move into the digital live sound mixer space?

WIth the success of our K and KW loudspeakers it became clear that there are lots of users who have high aspirations for live sound production quality. The majority of these users do not, however, need speaker systems that can cover arenas or mixers with huge input and output count. So we saw an opportunity to bring these users products that provide the quality and tools the top pros rely on but with a cost and scale that makes sense for them.

We felt that we could offer systems that delivered better results to musicians and production people working in small and mid-sized venues.

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Was the intention always to go with a compact offering – what challenges did this create and how were they overcome?

When the decision was made to pursue TouchMix, we went all in and aimed for a laptop-sized form factor that would be extremely powerful. Essentially, the two TouchMix models are about as small as they can be while still having enough space to fit the controls and connectors.

Probably the biggest challenge was getting the sensitive analogue circuitry (mic-pres), the DSP and the digital control circuitry to get along with each other. We have some engineers who are really good at ‘mixed signal’ design and they worked hard to get it right. On the other hand, relegating so much of the user interface

Key Features Twenty total inputs: 16 mic, (four with XLR/ TRS combo connector) and two stereo line (TRS) Dedicated talkback mic input Stereo Main L/R outputs (XLR) Ample monitor outputs (eight in total) with six mono auxiliary mixes (XLR) and two stereo auxiliary mixes.

Class A microphone preamps 32-bit floating point processing 24-bit A-D/D-A convertors RRP: $1,299 www.qsc.com

Gerry Tschetter

to the touchscreen allowed us to forego a lot of hardware controls that would have expanded the footprint.

How important was ease of setup and use when designing the TouchMix? Other than sound quality, nothing was more important. There’s a story about a guy who attached a rocket engine to a Chevy coupe and ended up becoming one with the side of a mountain, thus demonstrating that lots of power with no control is a dangerous thing. We spent many long days mocking up our user interface and making sure that it made sense to users. Then we did an extended field beta test with real-world users and refined the firmware based on their input.

Was it difficult to balance this ease of use with giving pro users the functionality they expect?

That is a fine question but I’d add one more element – how to balance these factors and help the novice get near-pro results. There are some areas where there really isn’t a conflict, for example the workflow on TouchMix seems to function well for pro and novice alike. But in other areas it’s a bit more tricky. For example, effects processing is the secret sauce

TECHNOLOGY REVIEW

www.audiomediainternational.com

that pro users apply to really polish their mix. But less skilled users can be puzzled by what effect to use and how to route signal to and from the processors. So we developed an FX Wizard that helps the user select the right effect, send an input to it and route the output where they want it – all without having to even understand the concept of signal flow. We also created an extensive input channel preset library. Unlike many mixer presets, these were made for live sound. We used real musicians, common microphones, live stage monitors, multiple PA systems and decades of experience in live sound production and performance. Then we added a ‘Simple’ mode that allows users to select a basic four-knob channel EQ as well as single knob compressors and clubs. One gig included a ‘PA’ mounted on the back wall; that was interesting, to say the least, with the five vocal mics up front.

Again, the onboard EQs and notch filters helped me. I finally disconnected the two frontline monitors (since they just made it all sound like an undefined cheese sauce). I used two DPA d:facto II and three Neumann KMS 105 live vocal mics; the preamps in the console complemented the mics very well.

Speaking of preamps, a common

gates. But they can still recall the presets with all the under-the-hood sophistication we dialed in. Pro users say they like the presets and FX wizard as time-savers that can get them close in a hurry.

What feedback have you had from users about the console?

We are getting lots of comments about the audio quality of the mixer. Our goal was to have audio quality comparable to concertlevel mixers so we’re really gratified that owners appreciate the effort. We’re also getting comments from users who had been struggling with their live mix finally getting the results they were looking for, whether it’s a better drum mix or being able to get the stage monitor performance they’ve been missing. concern with modern small live mixers is preamp and effects quality. In general, preamps are getting better, yet effects remain questionable from mixer to mixer. That said, I like the TouchMix-16’s effects; its range of parameters allow for some fine ’verbs. Yes, I did connect a TC Electronic reverb to the TouchMix’s aux channels before the tour – and it was nice – but I had no space for more gear on this tour and what QSC offers was fine.

Finally, are there any new additions/ updates to the range in the pipeline?

By the time this is being read, we will have released our 2.0 firmware (see page 8) and an updated iOS app. The 2.0 firmware implements a lot of the ‘why don’t you’ user input with things like multi-level security, more WiFi connectivity options, programmable user buttons, expanded options for auxiliary mix pick-off points and lots more. The iOS app now includes personal monitor mix capabilities for iPhone and iPod touch users. In the near future we’ll be offering an app for remote control by Android devices. Beyond that there’s not much we can share except to say that we didn’t get into the mixer business just to develop two models.

In every venue I used the console, QSC’s iPad app was used as the control surface. While you’ll miss the beautiful dial knob on the TouchMix, it works fine with the iPad. I rarely touch the app’s ‘faders’, but instead use the Nudge +/- buttons on the right side of the controller. You can set it in Normal or Fine Control mode for smaller adjustments. It took me a couple of shows to discover that this was the thing to do.

TouchMix also offers a number of recording features – all individual channels via multitrack capabilities – so I bought an SSD with converter cable and connected it to the USB port on the backside of the console. I’ve recorded a few shows and it sounds very clean; I can’t wait to get home and try recording my grand piano with this unit.

On the back panel, there is an Ethernet port; whether it is for a future LAN stage box connection, I don’t know. If I could hook a router up to it, I believe the network connection would be stronger.

SUMMARY For what you get, the TouchMix-16’s price (at $1,299 street) is amazing. I will be travelling with the Touchmix-16 wherever I go from now on.

The Reviewer Morten Støve co-founded DPA Microphones and is front-of-house/recording engineer for jazz vocalist/bassist Kristin Korb (kristinkorb.com)

February 2015

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TECHNOLOGY REVIEW

DPA D:FINE IN-EAR BROADCAST HEADSET MIC HEADWORN MICROPHONE

Alistair McGhee explains why he sees the company’s new in-ear mic as “a piece of fine art audio engineering”.

P

icture the scene – a much loved uber-confident radio presenter, a packed theatre, a waiting audience, a live broadcast, radio headphones and hand held radio mics. What could possibly go wrong?

Well let’s say the radio headphones’ cabling is not as reliable as it could be and at a crucial point in the evening when the whole shebang is live, it fails. Now the much-loved presenter can’t hear himself, or the people in the gallery yelling at him to carry on. No, he thinks the mic has failed, so he picks up the second hand held, ‘is this working?, still can’t hear himself, so he picks up the third and then the spare. Bellowing furiously into four working mics that ‘none of these mics are working.’ You had to be there, it would be funny, unless of course you happened to be the engineer (hi Mark!) or the producer that night.

What we need, of course, is a superreliable headset system from one of the world’s top manufacturers. Something with a high-quality mic you can sing into and some well engineered cable management. And maybe a remote controlled presenter would be nice.

Enter stage left DPA and its new d:fine In-Ear Broadcast Headset Microphone, complete with single or dual earpieces.

The d:fine In-ear mic will be available in two mic versions – an omni based on the existing d:fine 66 and a directional version based on the d:fine 88; both are available in brown, beige and black.

The new d:fine In-ear headset mic is hard to describe, especially if you haven’t seen the mic-only versions. Superficially delicate but actually sturdy and beautifully finished. It’s a piece of fine art audio engineering.

Two slender but tough wires form the backbone, which is expandable to fit the size of the talent’s head – in broadcasting usually XXL. To this wire frame are attached all the tricky bits. At your right ear a rubberised curly fixing while the left ear has an earpiece and a boom complete with the DPA mic. The mic and earpiece are cabled to the midpoint of the retaining wire and then exit down what should be the back of your neck – if you are wearing the wire like a tie, you have it on the wrong 42

February 2015

way round. About five inches down the wire there is a small clip to retain the cabling to a suitable anchor point on your clothing. The signal cables are about four feet long – with radio systems in mind – and terminate in very discreet connectors. Micro dot for the microphone and 3.5mm jack for the ear piece feed. The 3.5mm headphone jack is metal finished and the cable entry point has some nice strain relief, as does the microdot microphone connection.

“All in all the d:fine In-ear Headset mic is a remarkable microphone, a credit to DPA’s engineering.”

Alistair McGhee

By definition a head-worn microphone has a tough life. Stuck on a presenter’s head, under the lights, that’s a hot sweaty environment. These things get bent and pushed into and out of shape on a daily basis. On first inspection at least the d:fine In-ear mic looks made to take a beating. The finish on the mic boom is not going to shed and there’s a ‘drop stopper’ to keep moisture from running down onto the capsule.

The boom is adjustable back and forwards, which is a neat trick and works in conjunction with a cabling assembly that allows slack to be taken up out of the trailing leads to make fine adjustments to the amount of cable you have available to the microphone boom. And, in a miracle of

engineering, the boom is removable, which is very impressive.

In use It kind of goes without saying that a DPA mic will be good, the question is just how good. I plugged the d:fine In-ear mic into my Micron radio kit and gave it a whirl.

I normally use a Sanken COS 11D, which is also an omni, so that was a natural starting point of comparison. Clipping the Sanken high up on my collar I ran a set of vocal tests. Listening back the DPA easily bested the Sanken. A top end sparkle and a less boxy lower mid range. Checking the frequency responses the DPA’s peak is a bit higher than the Sanken’s, which accounts for the top end but didn’t explain the mid range contrast. As something of a champion for the 11D I was a bit disappointed. I then mounted the Sanken on the boom alongside the d:fine in-ear mic to test mounting position as a factor.

A much closer result, the 11D’s mid range opened up and now the differences were all about the top end. Both mics were good enough to hear the wireless link.

Despite the fact that the Micron is very good indeed. If I cabled the DPA direct to the mixer it had even more to give and throughout the testing I was struck by the low handling noise and pop resistance. All in all the d:fine In-ear Headset Mic is a remarkable microphone, a credit to DPA’s engineering.

The earpiece And so to the earpiece. I had the single earpiece but a dual version offering stereo will be available. In all, the options are single-ear mount and single in-ear; dual-ear mount and single in-ear; and dual earmount with dual in-ear. DPA has outsourced the units which are designed for comms,

a feed of programme with talkback would be the normal send. This is the bit that allows you to shout, ‘no the mic is working, just keep going’, straight into your talent’s lug hole. It’s not designed for quality monitoring, more the essentials of keeping the show on the road. The earpiece comes with a removable rubber boot, making it easy to clean. One area the industry does need to think about is a standard locking connector for our headphone feeds – a magsafe one would be brill.

To sum up DPA has yet another top-end audio product. It’s not going to be cheap but let’s face it you’re worth it, even if your talent isn’t! Hopefully they’re still working on the remote controlled presenter bit.

Key Features Available in two mic versions: an omni based on the existing d:fine 66 and a directional version based on the d:fine 88 Single or dual earpiece options Launch date: 17 February RRP: €587-704 (excl local VAT) www.dpamicrophones.com/in-ear

The Reviewer Alistair McGhee began audio life in Hi-Fi before joining the BBC as an audio engineer. After 10 years in radio and TV, he moved to production. When BBC Choice started, he pioneered personal digital production in television. Most recently, Alistair was assistant editor, BBC Radio Wales and has been helping the UN with broadcast operations in Juba.

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TECHNOLOGY REVIEW

NEVE 1073LB MONO MIC PREAMP MODULE

I

t might seem slightly perverse to begin a review with praise for a competitor’s product, but if it were not for the development of the innovative 500 series format by API – arguably the most successful open source audio technology since the Musical Instrument Digital Interface (MIDI) – the product under review, the Neve 1073LB mono microphone/line level preamplifier, hereafter known as the 1073LB 500, would not be sitting snugly (even smugly) in my API ‘lunchbox.’

A ’70s design, the original 1073 module rapidly became one of the preamplifiers of choice for discerning engineers and is one of the main reasons for the continuing popularity of Neve’s eponymous large-format consoles. Several versions of the 1073 series modules have been sold over the years by Neve – and cloners, of course – and the model under test has been designed to retain the sonic heritage and characteristics of the classic preamplifier, but packed into the 500 series format.

The 500 series chassis provides both power and the required input and output connectivity to the British-made 1073LB 500, which consequently means that the preamplifier’s cost is significantly less than would be the case if you were to buy the full monty version in a 19in rack format, or indeed, one of the repackaged vintage units. Because of the small front panel footprint of 500 series modules, the best are usually those with fewer controls and the 1073LB fulfils this criterion admirably.

A useful Neutrik combination connector sits at the bottom of the front panel and provides XLR microphone and balanced 0.25in TRS connectivity, while a large, rotary, stepped input gain control sits under a two-colour LED that flashes green when the signal is between -25dB and +26dB and red at +26dB or greater. I would have preferred to see a proper input meter, but, in practice, the LED allows you to easily set up the correct input gain into your digital audio workstation (DAW) or desk.

The input control allows gain settings 44

February 2015

Originally launched in the 1970s, the 1073 is now available for your 500 series rack.

Stephen Bennett puts this reworked classic to the test.

that run from -20dB to +10dB in 5dB steps for the line input (the right hand side of the control) and -20dB to -80dB for the microphone input (the left side), along with an off setting. The 1073LB features a phase button with associated LED and a Lo Z button that switches between two input impedances, 1,200 ohms (Hi Z) and

at line settings from 22Hz to 22kHz. For a piece of equipment like this though, the importance of this data is moot; most experienced engineers will know what a 1073 preamplifier should sound like and so will choose it for its provenance rather than its specifications. Suffice it to say that I had no problems with noise or distortion over the period of the review.

In use

“If you’re after the ‘Neve sound’ I can unreservedly recommend the 1073LB 500 – it’s an awful lot of beautifully built Neve for the money.”

Stephen Bennett

300 ohms (Lo Z) – the latter setting indicated by a yellow LED. The final button on the 1073LB 500 switches in the front panel input, disabling the API lunchbox’s input connectors. A rotary Trim control allows you to tweak the input gain for both microphone and line signals and also switches on the +48V phantom power when depressed – phantom power is automatically disabled when the stepped gain control is set to line input settings. Finally, the signal is fed into a transformer-balanced Class A output stage and thence to the lunchbox output XLR. Neve’s rather grandly named Audio Processing Insert Technology allows other Neve 500 series modules housed in the same lunchbox – such as the 1073LBEQ Equaliser – to be inserted directly into the 1073LB 500 pre-output stage signal path via the preamp’s rear panel INS LINK connector.

Neve publishes distortion figures for the 1073LB of 0.07% from 50Hz to 10kHz @+20dBu output with an 80kHz bandwidth, a frequency response of 20Hz to 20kHz with a ±0/5dB deviation and -3dB down at 40kHz. Equivalent Input Noise (EIN) is specified as better than -125dBU @60dB gain and noise at -82dBu

I tested the 1073LB 500 in the real word by using it to record bass and snare drums, piano, vocals, cello and acoustic guitar using a variety of microphones including the venerable Shure SM57, a Neumann U87 of ’80s vintage and a C12 be-capsuled AKG 414. I compared the performance with the preamplifiers in my Metric Halo ULN-2, a Focusrite ISA 430 Mk I channel strip and some recordings I did with the same company’s Red 500 series preamplifier.

As I’d expected, the Neve was noticeably more colored than my other preamplifiers – but in a lovely way. Allowing the input gain to hit the red occasionally didn’t seem to detrimentally affect the sound in any way – in fact I got into the habit of letting this happen probably more than was absolutely necessary, as I was so impressed with the results, especially with a Fender Rhodes and MiniMoog fed into the line inputs. The 1073LB exhibited the same characteristic I’ve often noticed in good preamplifiers – it allows microphones to really shine and even humble performers to produce excellent results. I didn’t have another flavour of 1073 available during the review, but I’ve used Neve desks extensively in the past and comparing the results obtained from the 1073LB 500 and some archive master tape recordings leave me in no doubt that the 1073LB 500 series is basically sonically the same as other versions of this classic Neve design – bearing in mind, of course, that a ’70s vintage 1073 rescued from a defunct console is unlikely to sound the same as a pristine new baby from Neve’s workshops!

However, if you’re after the ‘Neve sound’ I can unreservedly recommend the 1073LB 500 – it’s an awful lot of beautifully built Neve for the money.

Key Features Classic transformer microphone preamp module (Class A design) Neve designed hand-wound transformers +5/-20dB level Trim control with integrated phantom power on/off switch Phase, Impedance and Front Input selector switches Front combi-XLR connector with intelligent switching of phantom power Microphone input: Gain -80dB to -20dB in 5dB steps Line Input: Input impedance 4k ohms bridging, gain -20dB to +10dB in 5dB steps Distortion: Not more than 0.07% from 50Hz to 10kHz at +20dBu output Output is transformer-balanced and earth free Frequency response: ±0.5dB 20Hz to 20kHz, -3dB at 40kHz RRP: £695 (excl shipping and taxes) www.ams-neve.com

The Reviewer Stephen Bennett has been involved in music production for over 30 years. Based in Norwich, he splits his time between writing books and articles on music technology, recording and touring, and lecturing at the University of East Anglia.

TECHNOLOGY REVIEW

APOGEE ENSEMBLE THUNDERBOLT AUDIO INTERFACE

Brad Watts got his hands on the new 30 x 34 audio interface for Mac, and found a lot to get excited about.

A

pogee isn’t an entity to rest on its laurels, although it could easily do so. The company has been at the forefront of digital audio conversion since the very beginnings of the format, having been founded by Bruce Jackson in 1985, along with Christof Heidelberger and Soundcraft USA president, Betty Bennett.

For many years, Apogee DACs and ADCs were the bastion of the professional sphere, with devices aimed at the upper echelon of recording. Yet from 2007, with the release of an audio interface aimed at the project studio market, the Ensemble FireWire-based interface brought top-shelf Apogee audio conversion to a wider audience. The Ensemble then went on to win a TEC Award, chosen by the awards group’s 100-plus audio industry professionals.

Eight years on, Apogee is offering a continuation of the Ensemble heritage.

Gone is the original FireWire connectivity, supplanted with wider, faster and vastly more versatile Thunderbolt connectivity.

This fact leaves the Ensemble steadfastly in the realm of Apple computers – a tactic Apogee is more than comfortable with, having renounced development for Microsoft operating systems in early 2009.

Physically the new Ensemble is one of the sturdiest pieces of audio gear you’ll come across. The casing is constructed of 2mm-thick steel – no aluminium to be seen here. Equally as reassuring is the fact that all I/O ports are firmly attached to the chassis, so there’ll be no fear of bending connectors held to the board by way of solder connection alone. Plus, in somewhat of a recent departure for Apogee, the Ensemble is black.

I/O to the unit includes eight of Apogee’s high gain (75dB) mic preamps.

The preamps include Apogee’s Soft Limit, 48V power and high pass filters – all switchable via software. Connection is via the rear, with the first four being combo inputs for either XLR or jacks, and inputs one and two including inserts. The inserts

offer separate jacks for in and out as opposed to a single TRS jack. Outputs one and two are presented as balanced jack outputs with the remaining eight balanced analogue outputs presented via a 25-pin D-sub connector. A further 16 I/O points are available via four TosLink/ADAT optical connectors, which can be configured for S/PDIF or S/MUX. Alongside are coaxial S/PDIF in and out, wordclock in and out, and two Thunderbolt 2 ports. All up you’re looking at 30 x 34 I/O points (including the two headphone outputs) should you add additional ADAT interfaces.

Out front are two additional high impedance instrument inputs. These incorporate Class A JFET circuitry for more realistic harmonic character when recording guitar and bass. What’s exciting about this section is the additional instrument outputs. These are provided for re-amping duties, or indeed, strapping high impedance stomp-box effects across a track. Top marks Apogee – a creative gold mine built right into the interface. I’ve seen many an audio interface, and nothing else can pull this trick. Exemplary.

Visual feedback such as metering and headphone output levels is delivered via OLED displays, or of course, via the downloadable Maestro 2 control panel software. In fact, Ensemble has a hybrid creative user interface whereby most controls can be accessed from both software and hardware. Pressing and holding the desired input selection buttons on Ensemble lets you access input settings such as 48V power, high-pass filters and grouping from the front panel.

Speaking of software, you’ll need to be running version 10.9.3 of OS X (a.k.a.

Mavericks). But more on that in a moment

while we peruse the Ensemble’s front panel. Either side of the OLEDs are large detented potentiometers. The left side pot adjusts the selected input’s gain, and the right side pot adjusts monitoring level. A push on the right post will mute or un-mute the main outputs, and a push on the left pot scrolls through the 10 primary input channels (one through to eight and the two guitar channels). To the left of the input level pot are dedicated buttons for quickly selecting input channels for adjustment. Off to the right are four assignable buttons, which can be assigned via the Maestro 2 application for dozens of functions. These can include options such as clearing meters, through to toggling the guitar channel outputs from the DAW software or the guitar inputs.

What’s interesting is the ability to assign a button to engage talkback via the talkback microphone planted in the front panel. The talkback destination can be set to go to either or both of the headphone outputs, and outputs 9-10.

That’s right, the Ensemble Thunderbolt entirely negates the need for an external monitoring device. This ensures you’re hearing every drop of sonic goodness from the Ensemble without sullying the fidelity with a sonically inferior monitoring unit.

You can even assign outputs to three sets of monitors and choose to use an external mic for talkback. Absolute gold.

Speaking of fidelity, Apogee has upped the ante again, with a definitive upgrade from units such as the Duet and Quartet.

The Ensemble incorporates an ESS Sabre32 32-bit Hyperstream DAC with ‘Time Domain Jitter Eliminator’. And while the Ensemble doesn’t outshine Apogee’s flagship Symphony I/O, it comes extremely

close in specification. The Ensemble pulls THD+N of -114dB (at 96kHz) and a dynamic range of 123dB for its D-A, while the Symphony I/O manages -117dB and a dynamic range of 129dB. Equally as impressive is the device’s latency performance, with a 1.1 millisecond round trip delay at 96kHz. The resulting sonics are something to behold, with Apogee’s usual smooth and crisp high frequency reproduction, but with a bottom end that’s simply glorious. You’ve never heard your kicks captured and replayed like this before. All in all, you’re going to have to look hard to find an interface to offer all the Ensemble can at a price like this. Truly a watershed moment for Apogee.

Key Features Plenty of I/O Great mic preamps Versatile routing Re-amping built in Sublime sonics RRP: $2,495 or £1,999 (inc VAT) www.apogeedigital.com

The Reviewer Brad Watts has been a freelance writer for numerous audio mags, has mastered and mixed various bands, and was deputy editor of AudioTechnology in Australia. He is now digital content manager for Content and Technology.

February 2015

45

INTERVIEW

BACK TO THE RADIO DAYS

Once a proud and popular source of entertainment, the radio play has arguably become a bit of a relic of tradition in modern times. Award-winning sound designer John Kassab talks to Matt Fellows about his first foray into the medium, and why he is keen to see a resurgence of the format.

A

ustralia-born John Kassab has worked on a number of award-winning projects, including the Sundance Grand Jury Prize-winning short film Deeper Than Yesterday and the Oscarwinning animation The Lost Thing.

His most recent project, King of the Egg Cream, is a 10-part serial radio play starring Richard Kind (Curb Your Enthusiasm), Bobby Cannavale (Boardwalk Empire) and Justin Bartha (The Hangover).

Set in Brooklyn during the prohibition era and the golden age of radio, the project is a fitting way to bring the medium back into the limelight.

How did you get involved with the project, and what was its primary aesthetic goal?

I was invited onto this project by the writers, Emil and Sigmund Stern, who had heard my work on The Lost Thing. They were keen to make it cinematic and take advantage of advancements in audio technology, but they still wanted to retain an old-world feel. During the peak of this medium in the 1920s, the recording, transition and reproduction frequency bandwidth was very limited and creatively restrictive. Given that this project was created for electronic/web distribution, is pre-recorded and benefited from all the modern advancements in audio technology, it presented an opportunity to create the ultimate radio play production that our golden age predecessors could have only dreamed about.

How did you manage to retain that old world feel while also utilising modern audio technology?

We continuously walked a very fine line between old and new-world sound production aesthetics, as we still wanted to retain the charm and sensibility of the older programmes. This was achieved by being conscious of the stereo image and frequency band. We went wide and full band on the big stuff. We had to gently guide the audience from the narrow comfort of mono with a limited frequency width to the bombastic full width of stereo with all tops and bottoms included, then gently strip away at the layers of sound to contain things back to where they were, leaving the audience none the wiser of this stylistic transition.

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February 2015

Photo by Solomon Belfort

Very early on in the process there was the temptation to narrow the frequencies to that of a traditional radio play, but this made for a very boxy listening experience and, given our contemporary listening expectations of what narrative sound should sound like, it was a bit too much style-overkill.

The solution was to make it feel old but make it sound new. This was achieved by limiting the use of bottom-end frequencies. The use of sub bass has very much defined the overall contemporary theatrical film sound aesthetic; given that we were going for an old-world feel, and the plain fact that most people have the bass in their cars turned up way too high, I really had to pick my low frequency moments to retain a bygone aesthetic.

Can you tell us about some of the equipment that was used to achieve this?

The vocals were recorded in isolation booths at Goldcrest, Argot Studios and CDM Studios in New York, and ST5A in LA. At Goldcrest, the vocals were recorded using a Neumann U87 into a Millennium Pre; at Argot Studios a Neumann U89 into a Rupert Neve Portico 5032 was used – applying light compression with a Manley Stereo Variable MU tube compressor – and at CDM recording engineers used a U87 into a Focusrite ISA 220, as well as a Rupert Neve Portico 5015.

The sessions were cut by Emil and Sigmund Stern using Pro Tools, then sent to me for sound design with a list of sounds they would love to hear at the various cues.

We spoke a lot about the technology of the day, which led to a lot of independent research about soundscapes. I collated a library of era-specific sounds, and for stylistic consistency, pre-designed locations to make sure they matched and were stylistically consistent and historically believable. I recorded using a Sound Devices multichannel portable recorder and preamp solution, using a range of instrument or super cardioid microphones for mono effects and small diaphragm match pairs or stereo mics in XY placement for stereo recordings. I also carry a Sony PCM-D50 portable recorder, which I love because of its build quality, low noise, tonal transparency, stereo image flexibility, weighted silent gain control and its wonderful ‘divide’ button, which creates a new file mid-recording with a five second lead up. This is a life saver, particularly when recording stubborn wildlife and other sound events which require patience to capture.

How did you account for the average listener experience?

Given that this project was most likely going to be heard on headphones and

through car speakers, I sound designed it on my laptop monitoring though headphones and then cross-referenced it on domestic stereos and in cars. As there was no picture to sync to and no theatre acoustics to replicate in the studio, the portable studio solution was perfect as the session can be taken straight to the car, home stereo, and other listening environments for cross reference and adjustment. Once the designs were complete, the tracks were then delivered for mixing and mastering.

What do you think the future holds for the radio play medium?

I am very excited about the future of radio plays and I am keen to see a resurgence of this long-forgotten medium. Obviously the audiobook and podcast are now with us; these mediums have created more distribution channels and broadened listening audiences, which have readied the market for an onslaught of radio plays.

However what excites me most about this medium is that it will allow a greater number of filmmakers to tell their stories through sound. Radio plays give rise to a much cheaper mode of production to tell ambitious stories which would otherwise not be heard.

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