International TECHNOLOGY AND TRENDS FOR THE PRO-AUDIO PROFESSIONAL www.audiomediainternational.com
May 2015
SOUND ALL AROUND With virtual reality and mobile technology enabling new and exciting experiences for users, we explore the latest developments in immersive audio p22
STUDIO PROFILE
REVIEW
INTERVIEW
Unlike its former home, all is well at Windmill Lane Recording p20
Merging’s Hapi gets the test treatment p34
Star Trek sound designer Frank Serafine p42
WELCOME
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EDITOR Adam Savage asavage@nbmedia.com
Experts in the issue
CHANGES AFOOT
MANAGING EDITOR Jo Ruddock jruddock@nbmedia.com STAFF WRITER Matt Fellows mfellows@nbmedia.com ADVERTISING MANAGER Ryan O’Donnell rodonnell@nbmedia.com
Eddy Brixen is a the owner and senior consultant at EBB-consult, as well as an audio specialist at DPA.
COMMERCIAL DIRECTOR Darrell Carter dcarter@nbmedia.com ACCOUNT MANAGER Rian Zoll-Khan rzoll-kahn@nbmedia.com HEAD OF DESIGN Jat Garcha jgarcha@nbmedia.com PRODUCTION ASSISTANT Georgia Blake gblake@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: Iosono for Björk’s Black Lake Room (credit: The Living)
Dr Adam Hill is a lecturer in Audio Engineering at the University of Derby in the UK, and a live sound engineer for Gand Concert Sound.
Jerry Ibbotsen has worked in pro audio for more than 20 years, first as a BBC radio journalist and then as a sound designer in the games industry. He’s now a freelance audio producer and writer.
Carl Tatz Is an award-winning studio designer as well as an ex-engineer, producer and studio owner. He is also is the creator of the PhantomFocus System
I
know I spent most of last month’s Leader blathering on about Prolight + Sound and what we could expect to see, but with arguably the biggest talking point this time being the future of the event itself, rather than the latest flashy console from Company X – although there was plenty of that too, it has to be said – it would be wrong not to revisit the 2015 Frankfurt fair again here, at least for a moment. It’s fair to say the industry is still digesting the news that 2016 will bring separate dates for Prolight and Musikmesse and an entirely different layout, with PL+S moving from its usual Hall 8 to Halls 1, 3, 4 and 5 (see bit.ly/1CVjzcn if you’re late to the party) and Messe Frankfurt probably needs to clarify matters before decisions are made about whether the changes should be welcomed, as a few questions remain: How will the audio exhibitors be divided across the four halls? Will they get to keep similar stand positions? Those are just a couple of the concerns I’ve heard raised,
but it’s certainly made things interesting, and good luck to the organisers. But enough about that, what have we got for you in this issue? Well, I’m pleased to say we’ve got a fine feature on immersive audio, and no, it’s not just another piece on Dolby Atmos. As brilliant as the technology is, we know you’re probably fed up with hearing about it, so we’ve gone for something a bit different. Turn to Page 22 if that piques your interest (it should!). Also, you may have seen our online story last month about the demolition of the old Windmill Lane Studios building in Dublin, where U2 recorded The Joshua Tree, and Van Morrison was also a familiar face. Although I’m no U2 fan, I, like many, was saddened to see it go, but I was also surprised to see so many other news publications seemingly reporting that the studio is now gone for good. It isn’t. Windmill Lane actually moved to a new facility decades ago and has been growing ever since. To prove the studio’s alive and well, we spoke to the team to find out whether it’s still a strong attraction. We’ve also been chatting to a couple of pro-audio veterans this month – I’m sure they won’t mind me calling them that – in studio design expert Carl Tatz and Hollywood sound design legend Frank Serafine, on Page 28 and 42, respectively, so if you’d rather read some real words of wisdom, you should look there.
Adam Savage Editor Audio Media International
May 2015
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CONTENTS
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42 FEATURES
PRODUCT NEWS 6
PreSonus shows Studio 192 interface
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Audio-Technica strengthens mic line-up
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Allen & Heath adds to GLD line
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Yamaha unveils TF series consoles
PEOPLE 12
OPINION Dr Adam Hill advises on how to get the most from your subs
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Jerry Ibbotson investigates the evolution of recording equipment for news gathering
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Jeremy Borum discusses the importance of time management
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INTERVIEW Frank Serafine talks Star Trek, Tron and educating the next generation
ALSO INSIDE 10
SHOW PREVIEW: PLASA North: What to look out for in Leeds
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GEO FOCUS: SOUTHEAST ASIA We speak to firms in Thailand, SIngapore and Indonesia about the state of their market
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BROADCAST FOCUS: DPA explains why its bodyworn mics are performing strongly at present May 2015
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STUDIO FOCUS: We visit Dublin’s Windmill Lane and discover how it continues to thrive
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IMMERSIVE AUDIO: VR and mobile are encouraging immersive experiences but standards are lagging behind, finds Erica Basnicki
TECHNOLOGY 28
HOW TO Studio designer Carl Tatz delivers his monitoring recommendations
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EXPERT WITNESS Nathan Barrios highlights what you need to know when specifying small-format PA systems for schools
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AMI RECOMMENDED Small-format PA
37 REVIEWS 34 36 37 38 40
Merging Technologies Hapi SSL Live L300 Waves Butch Vig Vocals Sennheiser ew D1 JBL LSR308
PRODUCT NEWS
NEW AVX SYSTEMS FROM SENNHEISER Sennheiser introduced its AVX wireless microphone systems for video cameras at NAB 2015. AVX operates in the licence-free 1.9GHz frequency range. The compact receiver plugs directly into the camera’s XLR, where it switches on using the phantom powering of the camera and automatically pairs with the transmitter. The system immediately sets the correct audio level, matching it to the camera’s input sensitivity and selects a free frequency. If a source of interference appears, AVX will inaudibly shift to a new frequency. The system is equipped with adaptive transmitting power, so it always uses the power that is required to maintain a reliable connection between the mic and the camera receiver. This not only ensures a stable link, it also reduces battery power consumption – as does the automatic on/off function. Power is supplied by lithium-ion batteries, which can be recharged via a USB port, and for DSLRs with a jack audio input the AVX systems include an XLR-3/mini-jack adapter cable and the mounting accessories needed to attach the receiver to the camera hotshoe. NAB also saw the launch of new digital clip-on microphone models from
Sennheiser and Apogee, the ClipMic digital and its professional counterpart, the MKE 2 digital. The new range comes just months after both manufacturers first announced their partnership at NAMM. ClipMic digital and MKE 2 digital promise optimum sound quality for recording, while both microphones feature a Lightning connector for iPhone, iPad and iPod touch. Apogee’s mic preamp and A/D conversion circuitry are built into both models – designed to provide maximum gain with minimal noise – while the A/D converter provides a resolution of up to 24bit/96kHz. www.sennheiser.com
AVID ANNOUNCES VENUE S6L Avid has introduced the Avid VENUE S6L live sound mixing system, featuring a touch-based interface designed to enable audio professionals to easily handle the largest tours and events. Integrated with Pro Tools, Avid VENUE S6L functions as a front-end to the company’s MediaCentral Platform for media management and distribution. The S6L is a fully modular, scalable live sound mixing system. It features the VENUE E6L live sound engine and delivers processing at 96Hz and support for higher sample rates. The system aims to enable professionals to achieve better sounding mixes with greater speed and efficiency, thanks to the console’s touchscreen workflows, ergonomics and
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May 2015
visual feedback. The S6L was created to provide greater onboard plug-in processing than previously available, providing engineers with the same sound processors used in recording studios, fully integrated to deliver responsive control. Pro Tools integration provides recording and playback functionality without the need for a separate audio interface. Furthermore, the S6L is a networked system and can be customised to meet the needs and budget of professional productions. www.avid.com
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PRESONUS SHOWS STUDIO 192 INTERFACE The new 26 x 32 Studio 192 USB 3.0 interface from PreSonus records at up to 192kHz and combines eight digitally controlled XMAX Class A solid-state mic preamps and BurrBrown converters with StudioLive Fat Channel signal processing. PreSonus’ UC Surface touch-ready control software for Mac, Windows and iPad is included, for management of all Studio 192 functions and creation of eight stereo monitor mixes with full Fat Channel processing on every analogue input, plus reverb and delay. Users of PreSonus’ Studio One DAW will find full preamp control and zero-latency monitor-mix functions integrated into their recording environment. Studio One’s hybrid Fat Channel plug-in runs on the computer’s processor and switches to the Studio 192 DSP. This allows the user to monitor with Fat Channel EQ and
dynamics while switching between playback and recording. Mixes and the onboard talkback mic can be routed to any output, including the two headphone outputs. The outputs can be used for monitor mixing or for speaker switching, and onboard Dim/Mute and Mono options are included. Studio 192 offers two front-panel mic/instrument inputs and six rearpanel mic/line inputs; 16-channel ADAT Optical In and Out (eight channels at 88.1 or 96kHz); coaxial, stereo S/PDIF I/O; and BNC word-clock I/O. The unit features eight balanced TRS outputs, balanced stereo Main outputs, and two headphone amplifiers with independent outputs and level controls, while Burr-Brown A/D-D/A converters offer 118dB of dynamic range. www.presonus.com
ALCONS LAUNCHES LR28 LINE ARRAY Alcons Audio announced the LR28 larger format pro-ribbon line array for the concert touring and large-scale installation markets at Prolight + Sound in Frankfurt. Based on the LR24 line-array, the LR28 promises an increase in HF and LF output over its predecessor, combining ‘HiFi-grade sound quality with the highest SPL capabilities’. Featuring Alcons’ proprietary proribbon technology for mid and high frequencies, the LR28 offers ‘very fast impulse response with full-dynamics, and up-to-90% less distortion under the lowest power-compression’. The LR28 also features the proprietary RBN1404rsr 14in proribbon driver, quad slot-loaded 6.5in mid-range transducers and double, purpose-customised 14in woofers. In addition, the allnatural (Isophasic) cylindrical wavefront of the pro-ribbon HF
transducer enables pattern control, without any distortion-inducing horn constructions. The LR28 is one element of a complete application-configurable package, including the LB28 bass array-extension, BC543 triple 18in cardioid subwoofer system, Sentinel amplified loudspeaker controller with ALControl remote-control application, ARC 3D-simulation tool and system connection and transport logistics. “Until now, this high level of precision in sound quality was not obtainable in sound reinforcement, especially not at these sound pressure levels,” commented Philip de Haan, head of Alcons R&D. www.alconsaudio.com
PRODUCT NEWS
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A-T STRENGTHENS MIC LINE-UP Prolight + Sound 2015 was chosen as the setting for Audio-Technica to reveal its new BP40 large-diaphragm dynamic microphone (pictured) and AT2020USBi USB condenser mic. The BP40 is a dynamic vocal microphone with natural condenser-like sound for a variety of applications including radio, overdubs and post-production. Its hypercardioid polar pattern provides isolation of the desired sound source and off-axis rejection when working in close production environments. The mic’s 37mm diaphragm features patented floating-edge construction that promises to maximise diaphragm surface area and overall diaphragm performance, while the humbucking voice coil prevents electromagnetic interference (from video monitors, AC power, etc). The microphone also comes with a switchable 100Hz high-pass filter to provide further pop protection. Combining high-resolution audio with increased connectivity options,
the AT2020USBi, modelled after the AT2020, features an A/D converter with a 24-bit/96kHz sampling rate to deliver high-resolution articulation and intelligibility suitable for home studio recording, field recording, podcasting and voiceover use. The AT2020USBi works with both USB and Lightning cables (included), so users can employ it straight from the box to digitally capture audio on the device of their choice – PC, Mac or iOS. The integrated gain control adds the ability for users to adjust the input level directly on the microphone. www.audio-technica.com
STUDIOMASTER DISPLAYS DIGILIVE DESK Studiomaster launched digiLiVE, a new ultra-compact digital audio mixer at this year’s Musikmesse and Prolight + Sound simultaneously. digiLiVE 16 is a 16-input, 16-bus, eight-output digital audio mixer sporting a control surface integrating a 7in Android-driven high-resolution touchscreen and eight motorised 100mm fader operation, alongside fully remote (iOS and Android) tablet operation. The 16 inputs include 12 professional mic input channels; A/DD/A conversion is 24-bit/192kHz and there are up to eight internal effects busses with reverb, delay, 15-band EQ and modulation. The eight ‘smart’ XLR analogue outputs are assignable, and outputs also include digital AES/ EBU ands SPDIF. Two USB interfaces are included (front and rear panel) supporting playback and recording, scene storage and system updating,
while WiFi and Bluetooth wireless connectivity are supported. www.studiomaster.com
Open & Closed The new flagship studio headphones ATH-M70x PROFESSIONAL MONITOR HEADPHONES
ATH-R70x PROFESSIONAL OPEN-BACK REFERENCE HEADPHONES Audio-Technica introduces the new leaders in studio headphones: ATH-M70x and ATH-R70x. Both models offer extremely accurate audio reproduction along with the comfort, durability and convenient features that are bound to make them studio mainstays. So whether you prefer the sound isolation of the M70x or the spacious sound of the R70x, deciding on your next pair of studio headphones is an open and closed case. ATH-M70x
ATH-R70x
www.audio-technica.com
May 2015
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PRODUCT NEWS
A&H ADDS TO GLD LINE Allen & Heath has released Chrome Edition models of its GLD series mixers with new features and FX. GLD Chrome firmware introduces auto mic mixing capability and further additions to the processing suite, including new FX and channel-based compressor emulations. The automatic mic mixer (AMM) can be configured to work across all 44 microphone sources, allowing the user to select which inputs should be automixed without the usual restrictions of a 16-channel insert-based system. The AMM can be set to work in two modes – ‘D-Classic’ dynamic gain sharing for a quick set-up, and ‘Number of Open Microphones’ (NOM) logic gate technology for a more flexible and sophisticated auto mix. A best mic lock system detects cross talk between different sources and limits open microphones. GLD Chrome firmware includes a new plug-in architecture that allows users to select from a number of different processing units on every input and mix channel as well. Two new RMS-VCA inspired compressor
models, the 16T and 16VU, are included. Integrated within the mixer’s channel processing, all six compressor models can be selected on any of the input and mix channels on the fly, without burning valuable FX slots or adding latency. The new version 1.5 Chrome firmware also features several additions to Allen & Heath’s on-board FX suite, including a new Stereo Tap Delay, with 2.7-second maximum delay time, split LR beat fraction control, millisecond mode and Tap Tempo functions. www.allen-heath.com
L-ACOUSTICS EXHIBITS X SERIES L-Acoustics launched the X Series line of coaxial speakers for the installation and rental markets at Prolight + Sound. The new models offer high-excursion neodymium drivers, ellipsoid directivity, laminar vented ports and up to 30% weight reduction. The live monitor of the X Series, the X8 promises pristine sound, a high SPL and extended bandwidth for operation at FOH position or in control rooms. Its wide conical directivity pattern imparts a sense of spatialisation, with no minimum listening distance. The X12 is designed to cover
allows users to switch between different EQ Overlays with the same type of compensation. A speaker manufacturer or rental company can pre-define these paths for distribution inside the Lake LoadLibrary. The new software release also includes updated button placement for the MESA EQ section, in addition to a number of other new features that provide loudspeaker manufacturer partners with an expanded toolbox for protecting and distributing their Lake-related presets and contents. www.labgruppen.com
RME unveiled its new 24-channel 192 kHz bus-powered Babyface Pro compact audio interface at Musikmesse/Prolight + Sound 2015. The unit features a redesigned XLR socket as the main I/O, integrated into the housing to save space. The two headphone outputs, offering TRS and mini-jack sockets in parallel, have separate driver stages to match low and high impedance headphones, promising sonic results no matter what type of headphone is used. The user interface of the Babyface Pro is designed to be informative and clearly laid out, aiming to make access to every feature and configuration mode of the Babyface Pro intuitive. Even in stand-alone mode, routing and mixing of the inputs to outputs directly
May 2015
on the device ‘opens a whole world of possible applications’. Its two digitally-controlled preamps provide individually switchable 48V phantom power and feature a gain range of 71dB, adjustable in steps of 3dB, plus a relay-driven PAD, designed for EIN (Equivalent Input Noise) performance as well as line overload protection with enough gain for the low level mics. www.rme-audio.com
NEXO INTRODUCES ID SERIES
Prolight + Sound provided an opportunity to view the new family of low-profile, highoutput loudspeakers for the installation and commercial markets from Nexo, which also used the show to announce Orbital Sound as its principal dealer and rental partner in the UK. 8
multiple applications in rental and installation. An ellipsoid directivity of 90° x 60° makes it adaptable to FOH application. The X15 is the ‘powerhouse’ of the series, offfering high power in beamwidth and acoustic isolation. Its 40° x 60° directivity is designed to provide immunity to feedback. The L-Acoustics X Series is set to ship in Q4 2015. www.l-acoustics.com
RME REVEALS BABYFACE PRO
LAKE UPDATES CONTROLLER SOFTWARE Demonstrated at Prolight + Sound in Frankfurt this year, the Lake Controller v6.4 software program brings a number of features designed to facilitate faster and more accurate optimisation of complex, large-scale sound reinforcement systems. Specifically, v6.4 adds new elements to preset and EQ handling with the introduction of Recallable EQ Overlays. Additionally, when an EQ Overlay is either stored or recalled to a Module or Group, the store/recall path is also stored in the actual Overlay for the module. This feature
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At launch, the ID (InSpace Definition) Series comprises three products, the ID24 full-range compact speaker – using twin 4in drivers in a V formation combined with an HF compression driver offering two preset directivity options – and two low-profile subs, ID S110 (1 x 10in) and ID S210 (2 x 10in). Nexo has created three versions: ID24i for installation, ID24t for touring and AV applications, and ID24c or the ‘a la carte’ version, which allows the customer to custom-configure the speaker to match their requirements. www.nexo.fr
PRODUCT NEWS
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YAMAHA UNVEILS TF SERIES CONSOLES Prolight + Sound 2015 saw the launch of Yamaha’s TF series digital consoles designed for small to medium-scale mixing requirements. The TF series comprises three compact Yamaha digital mixing consoles. The TF5, TF3 and TF1 respectively feature 33, 25 or 17 motor faders, along with 32, 24 or 16 rear-panel analogue inputs, which feature recallable Yamaha D-PRE preamplifiers for the first time in a digital console. There are a total of 48 inputs on the TF5 and TF3, with 40 on the TF1, including dual stereo analogue/USB digital inputs and dual returns. Live recording features include up to 34 x 34 channel recording and playback via USB 2.0 and 2 x 2 with a USB storage device. All of this, plus operation with I/O racks, is said to make the TF series suitable for a wide range of applications. TouchFlow Operation is a new approach to console operation that combines elements of Yamaha’s user interfaces with an advanced touch panel system, allowing users to respond to the music and artists on stage ‘with unprecedented speed and freedom’. One-knob COMP and one-knob EQ features allow users to dial in the ideal sound via a single rotary control – on inputs for adjusting individual sources and on outputs for overall
D&B RELEASES D20 AMPLIFIER d&b audiotechnik unveiled two new products and a major software update at Prolight + Sound. The announcements keep d&b “at the forefront of the touring and installation markets”, according to the manufacturer. The new arrivals include the D20 (pictured), the latest addition to the company’s four-channel amplifier family, and the MAX2, a flexible 15in stage monitor. The software upgrade is a new ArrayCalc feature for d&b’s line array systems. “The only way leading companies such as d&b can retain their market position is through constant innovation, meeting the demands of the market and more importantly the satisfaction of their customers,” said Sabina Berloffa, director of marketing and product management at d&b audiotechnik. Visitors were also able to explore the d&b workflow, comprising the new ArrayCalc simulation software and d&b R1 Remote control software. www.dbaudio.com
sound control. A GainFinder feature facilitates gain setup for individual input signals. Also provided is a range of input and output channel presets created in co-operation with microphone manufacturers such as Audio-Technica, Sennheiser and Shure, made to match a range of musical instruments and voices, covering parameters such as head amp (HA) gain, EQ, dynamics and more. www.yamahaproaudio.com
GENELEC DEBUTS 7040A SUBWOOFER Genelec used both the NAB Show and Prolight + Sound to showcase its new 7040A ultra-compact subwoofer. Designed to complement the 8010, 8020 and M030 active monitors, the 7040A promises accurate sound reproduction and monitoring of lowfrequency content. With a compact form factor, the 7040A is built for music creation and sound design applications, as well as audio and video production work in small rooms and improvised monitoring environments. The 7040A features Genelec’s Laminar Spiral Enclosure (LSE) technology, made from a spiralshaped strip of steel to provide mechanical stability for the large pressures generated inside the subwoofer. The spiral forms the bass reflex port as well, enabling linear airflow at high SPL outputs. This construction provides extended low frequency capacity and low distortion for accurate bass articulation. The 7040A produces 100dB SPL using a 6.5in woofer and a Genelecdesigned Class D amplifier. At the lowest frequencies this SPL is radiated from the bass reflex port by channelling large volumes of air movement through it.
Calibration of the Genelec 7040A subwoofer to the listening environment is done using DIP switches located on the subwoofer connector panel. These controls address typical monitoring placement configurations. With external dimensions of 410mm x 350mm x 205mm, the Genelec 7040A has a footprint that is smaller than that of your average small practice guitar amplifier and is narrow enough to fit in a 19in rack, weighing 11.3kg (25lb). An optional carry bag allows professionals to work on the move, with portable recording devices and in non- purpose-built monitoring environments with an accurate and flexible monitoring tool. www.genelec.com
AKG EXPANDS STUDIO MIC RANGE
AKG has introduced its C314 dualdiaphragm condenser microphone (pictured) and D112 MKII kick drum mic. The C314 promises best-in-class selfnoise and dynamic range, with a flat, high linear frequency response and neutral sound, suitable for a variety of different environments. The new model features a dualdiaphragm capsule with four selectable polar patterns. Each of the diaphragms is designed for the highest polar pattern accuracy, with the same C414 XLS gold-plated membrane for high linearity and neutral sound. An Overload LED Detection Display indicates overly high sound pressure levels, allowing the user to make adjustments to the microphone setup rather than in the recorded track. Integrated capsule suspension aims to reduce mechanical noise and resonances, while a double-mesh, all-metal grille protects the capsule and promises high RF immunity without affecting the microphone’s acoustical performance. A 20dB attenuation pad and bass-cut filter is also featured, enabling close-up recording and the reduction of the proximity effect. The D112 MKII builds upon its predecessor with a new integrated flexible mount, while the cardioid dynamic mic can handle more than 160dB SPL without distortion. Its large diaphragm is designed for very low resonance frequency to deliver a solid response below 100Hz. A narrowband presence boost at 4kHz is also featured, alongside an integrated humcompensation coil, which aims to keep noise to a minimum. www.akg.com May 2015
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SHOW NEWS: PLASA FOCUS
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EDUCATING YORKSHIRE
PLASA has been planning another packed Professional Development Programme for the 2015 Focus: Leeds event, but there’s also plenty of new product to look forward to once again.
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LASA Focus: Leeds returns this month, bringing a host of new products, networking opportunities and a robust programme of seminars, product demonstrations and training sessions to the Royal Armouries on 12-13 May. Funktion-One, which is exhibiting for the first time at the show, will highlight its new SB210A model, an active bass unit with two spare channels for a pair of satellite speakers. And, in a UK exhibition debut, its Evo 6E will also be on show. This high-intensity loudspeaker combines horn-loaded 15in mid-bass, 10in Funktion-One signature mid-range and a 1.4in compression driver into a compact enclosure. Sennheiser will be out in force at the event, showcasing its AVX wireless microphone system for video cameras (see Page 6 for more information); the evolution wireless D1 (reviewed on Page 38), a range of digital wireless microphone systems that promises to ‘let bands go wireless the easy way’, and the SpeechLine Digital Wireless, a model claiming to be the world’s first digital microphone designed exclusively for speech applications. Alongside Sennheiser’s own gear are a selection of products from the brands it distributes in the UK, including K-array’s Khameleon loudspeaker wraps and Apart’s Mask 12 loudspeaker. Source Distribution will be showing products from a range of its distributed brands, including the first-ever UK demo of the PreSonus StudioLive AI digital mixer linked via AVB to an RM series rack mixer. The StudioLive AI active PA loudspeakers will also be on show. Promising studio monitor accuracy on stage, wireless or wired remote control and iPad/laptop monitoring, the fullrange series comprises 312AI, 328AI and 351AI configurations with a matching 18sAI powered subwoofer. The distributor will also spotlight the WorxAudio V5 Compact Line 10
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Array, a two-way line array loudspeaker designed to deliver music and speech in portable and fixed applications. The TrueLine X115 sub, which boasts a direct radiating 15in 1,000W continuous power handling cone driver, and brand new Universal Audio Apollo interfaces featuring A/D and D/A conversion, will also feature. Orbital Sound will be focusing attention on the Nexo Geo M6 allpurpose AV and musical theatre system, in addition to showcasing products from Shure’s wireless range, Clear-Com/ HME’s wired and wireless intercom solutions, and Yamaha’s QL series of digital mixing consoles – plus advance information on the Rivage PM10 desk. Dynacord will present a range of its products at the show, including the AXM 12A, a 12in-coax powered monitor, which features a max SPL of 128dB and an integrated DSP to deliver performance on loud stages. Also in Leeds will be the PM 502 mixing desk, which includes five mic/line inputs and three stereo inputs, as well as a master section with three Master Outputs – each with an individual sevenband EQ. Finally, the manufacturer’s Vertical Array Systems, a loudspeaker system
comprising the TS 400 full-range vertical array cabinet, PSD 25/218 powered subwoofers and PSE 215/218 passive subwoofers will be on the stand. It is designed to provide smooth and even coverage in mobile and fixed applications, in full-range or in active two-way systems. Electro-Voice is set to bring a varied offering of loudspeakers to the show, including the EKX and ETX portable systems. Promising performance and reliability in a compact package, the EKX features single-knob QuickSmartDSP digital signal processing control, Signal Synchronized Transducers waveguide design, and Cardioid Control Technology for subwoofers. The ETX portable loudspeakers are said to offer versatility with a number of options, available in two-way models (10in, 12in and 15in with HF titanium compression drivers), a three-way model (15in with a 6.5in MF driver and HF titanium compression driver) and subs (15in and 18in drivers). A LOT TO LEARN The show also hosts a Professional Development Programme, with speakers addressing topics across the audio, lighting, video and rigging sectors.
This year’s sessions to watch include ‘Sound intelligibility, quality and impact – design, verification and operation’ at 1.15pm on Tuesday 12 May, presented by Funktion-One’s John Newsham and Jim Cousins of Sound Improvement Services. The session will offer an introduction to room impulse response measurement to verify as-installed performance parameters such as intelligibility compliance, clarity and bass ratio, in addition to maintaining attention-grabbing bandwidth, dynamics and impact by using sensible sound check procedures for a bold and uncluttered mix. Also worth looking out for is ‘How to get the best from a live female vocal’, hosted by Justin Grealy of Soulsound, who will demonstrate the creative use of reverb, delay and compression with a live singer on stage. In addition, Martin Audio’s Robin Dibble will present ‘The shape of things to come’, a discussion of brand new patent-pending technology that promises to deliver “superior coverage and performance”. www.plasafocus.com/leeds
OPINION
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GETTING THE MOST FROM YOUR SUBS
Dr Adam Hill, sound engineer and lecturer in Audio Engineering at the University of Derby, delivers his low-end theory.
DR ADAM HILL
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hether it’s a bodyshaking live sound system, a danceinducing club PA or an all-immersive home theatre, we crave bass. There’s no such thing as a one-size-fitsall method for tuning subwoofer systems. Unlike with high frequencies, lowfrequency sound reproduction is equally – if not more – affected by room acoustics as it is by the loudspeakers. Without a good understanding of the loudspeaker-room interaction, it is difficult to get the most out of your subs. We’ll take a quick look here of the basics to deliver intestine-churning, but tasteful low-end to an audience, or whatever experience you’re targeting.
Home theatre In small spaces like home theatres, room modes are king. These are standing wave patterns at various frequencies, dependent on the room dimensions, that cause severe seat-to-seat variations in the frequency response, as well as unwelcome transient smearing. While room modes exist throughout most of the audible frequency range, they are only perceptible below the Schroeder frequency, which is calculated based on reverb time and volume. In small rooms, this comes to around 200Hz. So how can we reduce the negative effects of room modes to provide a 12
May 2015
consistent listening experience across a room? Most home theatre systems come with a measurement mic and an automatic calibration procedure. These methods typically employ an inversefilter, based on a single measurement to give a flat frequency response. These achieve precisely that, but only at the measurement location. In terms of bass coverage, they provide no benefit whatsoever. The same goes for spatially averaged measurements. You cannot EQ away room modes. Something different is required. A good first step is subwoofer positioning. Sticking a subwoofer in a corner excites all room modes, resulting in a boomy bass response as well as severe seat-to-seat variations. If you have one subwoofer, try moving it to a wall midpoint. This prevents certain modes from being excited and gives a less boomy sound with more even coverage. If you can get away with it, get a second subwoofer and place the two at opposite wall midpoints. This further improves the situation. In essence, the more subwoofers you have, the less room mode issues you’ll encounter.
Live sound Low-frequency reproduction in live sound is a different beast than in small rooms. In venues we typically work in, the Schroeder frequency is usually below 20Hz, therefore room modes are not an issue. So what are the challenges in live sound? Instead of room modes causing the issue, it’s comb-filtering between subwoofers and early reflections (for indoor venues) that is the culprit. When you send the same signal to spaced loudspeakers, the signals recombine constructively or destructively, depending on the phase relationship between the arriving sounds at a given location. When subs are deployed as left and right stacks, this causes dead spots in off-centre areas of the audience and a power-alley down the centre. This means the FOH engineer receives more bass, which can result in a bass-light mix for off-centre areas of the audience.
To avoid this, it is fairly well known that spacing your subs across the front of the stage gives more even coverage due to the small phase differences from the arriving waves. Typically, you want to aim for no more than ½ wavelength spacing of your subs (two metres @ 85Hz). Without any other DSP, this gives even coverage across an audience that is between the horizontal array of subs, with a gradual roll-off of energy outside of this area. To widen this coverage, you apply delay to the outside stacks. To narrow the coverage, delay the inside stacks. If using cardioid subs, it is essential to give adequate free space behind the units – at least two metres or so. We’ve found at Gand that if you don’t have this, you lose system efficiency as well as the cardioid pattern. If you want to further reduce comb filtering, you must find a way to decorrelate the signals from each subwoofer. Based on my experience, I see more and more system techs configuring PAs so the subs are fed by a mono aux. This means that the signals coming from all subs are highly correlated and will therefore cause severe comb filtering.
Cinema sound In cinema, there are similar issues to live sound. What I’ve found, though, is that cinema is lagging behind live sound in its subwoofer calibration methods. They rely largely on single or spatially-averaged measurements to tune their systems, which, as I’ve mentioned, does nothing to improve consistency of bass coverage.
Cinemas have the benefit of speakers surrounding an audience. The system naturally exhibits a degree of decorrelation, but comb filtering is still a serious issue, much more so than room modes – they’re only a problem below around 35Hz. While it’s possible to use an optimisation routine, requiring precise measurements and signal processing, I’m developing a method of automatically decorrelating the low-frequency signals feeding the loudspeakers – a method known as Diffuse Signal Processing. This technique applies an imperceptible, low-level noise-like decay onto source signals, which when applied correctly avoids significant comb filtering. This requires no on-site calibration. Although work is ongoing to perfect this method, I see applications not only in cinema sound, but also in live sound and home theatres. The ultimate goal is to deliver an equal listening experience to all. Clearly, the challenges at low frequencies are very different from those at high frequencies. While this article only scratches the surface of the tools available, I hope it gets you thinking about simple fixes you can apply to whatever system you work with to give the best possible listening experience to as many people as possible. Dr Adam Hill is a lecturer in Audio Engineering at the University of Derby in the UK, and a live sound engineer for Gand Concert Sound.
Boom festival photo: esweb
OPINION
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WHAT’S NEXT FOR NEWSGATHERING GEAR? Jerry Ibbotson reveals how smartphones are becoming an increasingly popular choice for radio news recording, and why manufacturers need to catch up.
JERRY IBBOTSON
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ext time you’re watching the TV news and you see someone being interviewed by the press pack take a good look at the recording equipment being thrust into people’s faces. As well as the ENG mics and windshields bobbing about at the bottom of the screen (much to the annoyance of the cameramen) you’ll see an increasing number of devices that many audio pros will not consider ‘proper’ recording gear – mobile phones. I realise this is sacrilege to many of those reading this magazine but smartphones are now more and more likely to be part of a radio journalist’s arsenal. From reel-to-reel recorders or Marantz cassette machines, through Minidisc and more lately all-in-ones like the HHB FlashMic or Nagra Ares, we’ve come a long way. But phones? Actually, if you listen to the radio and to news in particular, you’ll have been hearing actuality captured on a phone for a couple of years now. BBC radio reporters are issued with iPhones as standard kit. An old friend of mine, Nick Garnett at BBC Radio 5 Live, is a pioneer of this and uses his iPhone and a range of apps including Luci Live (reviewed in Audio Media a couple of years ago) and Voddio to record and share all his material. It’s the ‘s’ word – share – that is important here. As Nick has told me (a relative Luddite in this field) before: it’s 14
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no good if you have the best audio in the world if you can’t get it to air. And that’s where phones (and tablets to a certain degree) come in. Having recorded interviews using the surprisingly good microphone on an iPhone, it’s a doddle to then file that material back to base for broadcast. Email will do the job in many circumstances but specialist apps are also available and better for larger files. The BBC has its own software, PNG, which combines recording and editing with the ability to drop the sound files directly onto the corporation’s own servers. You can conceivably have material on air in minutes without recourse to a big expensive satellite truck or radio car. I’ve used PNG myself, recording a succession of radio interviews at the scene of a developing story and filing each one back to the studio while I walked to the next. By the time I arrived ‘home’ at the newsroom, each interview had already been edited and put to air. And all of this was done with a piece of hardware with a fruity-sounding name. It’s not just a flash in the pan. I’ve spoken to a well-known manufacturer of portable recorders that was looking at how to commercially answer the increasing need for ‘capture and forward’ technology – recording and getting on air. The conclusion we came to was that if the software doing the file transfer was kept separate from the recorder (i.e. on a phone) it would be easier to update and modify. A while later I spotted a machine from Olympus, which does a pretty good range of portable audio machines that take a first step in this direction. The DM-901 actually sits not in the Olympus Audio Recorder range but among its business hardware. It’s a voice recorder – a 21st century dictating machine. It sits in the palm of your hand and is meant for note-taking, interviewing for nonbroadcast and business presentations. Not the usual review fare for Audio Media International but bear with me. It can record in .wav format at 48kHz 16-bit or as MP3 and has a decent range of options to choose from, including a
Low Cut Filter and a mic ‘zoom’. As well as the built-in capsules it also has a 3.5mm minijack input for a separate microphone – reminiscent of the MiniDiscs of the mid to late 1990s and earlier 2000s. In fact I think I still have a minijack-to-XLR lead rattling around somewhere. I played around with various controls and did some test recordings with the onboard mics and the results were perfectly acceptable; certainly no worse than a lot of smartphones. But what had drawn my attention to the Olympus was not the hardware but the software. There is a separate Audio Controller app for both iOS and Android phones that hooks up directly to the DM901 via WiFi. I downloaded it from Google Play to a Motorola Moto G phone and after a bit of jiggery-pockery, synched the phone to the recorder. One tip: make sure the WiFi on the Olympus is actually activated before you start cursing and swearing. And make sure your phone is disconnected from your usual WiFi source as well. What you have in the app is a remote control for the DM-901 that lets you stop and start recording. This could be useful at press conferences, where the controller has to sit at the front of the event while the reporter loiters at the back. You can index recorders and even add images to them if necessary.
But the real trick up the Audio Controller’s sleeve is its ability to upload recordings from the recorder, via the smartphone to Dropbox. It can only do this with MP3 recordings at 128kbps but the procedure is pretty straightforward. The downside is that it only works with Dropbox, which is great if that’s your Cloud storage of choice but less so if it isn’t. To me, it’s a case of so close but no cigar. I have to admire Olympus for taking this first step down the transfer road. But when I can download a free voice recorder app for my phone that lets me use Android’s range of Share options, then the bar is set quite high. Also – why use WiFi? That’s what my phone should be using to upload the audio, but it can’t if it’s hooked up to the recorder. Why not use Bluetooth for the phone/ recorder link instead? Those things aside, at least someone’s actually made the first move. The Olympus Audio app and hardware may be flawed but they have to be saluted for getting the ball rolling. Jerry Ibbotson has worked in pro audio for more than 20 years, first as a BBC radio journalist and then as a sound designer in the games industry. He’s now a freelance audio producer and writer.
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OPINION
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SURVIVING PERFORMANCE AND SCHEDULE PRESSURES
Working in this industry can be stressful at the best of times, so what can you do when the workload gets too much? Composer and author/producer of Guerrilla Film Scoring Jeremy Borum shares some of the tips he’s picked up over time, which pro-audio professionals should also take on board.
JEREMY BORUM
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s technologies for hardware and software become less expensive and more productive, schedules get shorter and expectations grow higher. As a result, time management skills become a very important part of creative productivity. Creative people usually have a particular context and mindset in which they can be at their most creative, but most have to handle a vast number of other tasks and responsibilities that pull them out of that space. Time is free but in limited supply. When the pressure is on you need to take conscious and concrete steps to make sure that every minute is used well. Our experience of time is very subjective, just like art is subjective, and every person has a certain perception of time that helps to put them into a creative zone. When you are in that zone you have the right feelings, energy and state of mind to go about your artistic work effectively. A feeling of urgency is helpful for some and stressful for others. Some are early birds and others night owls. Seeing continuous evidence of productivity is important to some and others want slow reflection. Some composers write in long blocks and others in short sprints. Before 16
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sitting down to write, one person needs quiet and another needs stimulus. Your mindset can greatly affect the quality of your work, and the ideal context will be different for everybody. When the workload is daunting and the schedule is pressing down on you, remember that you have just as much time in the day as Oprah, or the President of the United States, or any other influential figure you care to substitute. Whatever workload you have doesn’t compare to theirs, and it’s completely possible for you to succeed in a way that makes you feel happy, healthy and fulfilled. Oprah found ways to maximise her time and has done it continuously for years. That’s why everybody knows who she is. She has probably always had a very intuitive understanding of how to manage her time. Time management is just as important to success as passion, intelligence and opportunity.
delivery prep. Put all of your wrap-up time commitments into your schedule, and then look at the time that remains. That is your block of time for creative work and planning, and it will have to work no matter how long or short it is.
Plan Your Attack When you begin a gig you usually have a deadline and a top-level schedule. As soon as the contract has been signed, you should immediately begin to detail your schedule. Knowing that you need to deliver a hard drive full of recordings in a couple of months isn’t a precise enough goal to direct your daily work well. It doesn’t make much sense to dive in and start working until you know your short-term goals. Everybody does this in different ways, but there are some common threads in how the successful ones approach their schedules. The most common approach is to schedule backwards, because the most predictable time commitments are often towards the end of the process. For example, if you’re writing a film score then with a little experience it’s not hard to guess how much music you’ll record in each session, how many sessions you’ll need, and how much time you’ll need for editing, mixing, mastering and
Manage Time Tactically A core time management principle is that you need to assign times to things that have importance. Because creative people are usually self-employed and often work from home, they can very easily fall into a free-form routine of unstructured work. That can lead to doing the tasks that feel urgent rather than the ones that are actually more important. Scheduling time in which you focus only on important things is critical if you want to stay on target. Although it may seem like a good idea, writing a to-do list is not a time management solution. A composer’s to-do list is about a million miles long, and writing a score is an endless pile of details. Some of them are creative and subjective like the phrasing of an instrument, and some of them are dry and calculated like determining the cost of paper for sheet music. To-do lists are nothing but a reminder of what is unfinished. They have
nothing to do with time management and won’t help you meet your goals in a more timely fashion. A list that will help is a priority list, and that is something very different. A priority list is a battle plan. Priorities define your most important tasks, but more importantly, they help you say no to all the distractions that pull you away from those things. If your goal is to write six minutes of music every day and you know that you’re going to crash and burn if you don’t pull it off, then you must focus on those six minutes before all else. Priorities are about focus, and when you set your sights on one thing, the others move to the periphery. Saying no is not an intuitive skill for most people, and needs to be learned and practiced. Successful people say no to a lot of things. Very successful people say no to almost everything. Jeremy Borum is author/producer of Guerrilla Film Scoring: Practical Advice from Hollywood Composers. Both a book and a documentary, it helps composers at all levels to create the best-sounding scores quickly and cost effectively – without jeopardising their art. www.guerrillafilmscoring.com
GEO FOCUS: SOUTHEAST ASIA
EASTERN PROMISE
Picture: Alter
Another up-and-coming region in the industry, Southeast Asia is drawing more and more interest from some of the biggest international players in pro-audio. Matt Fellows investigates the markets in Singapore, Thailand and Indonesia to find out why.
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here have been more than a few shake-ups in the industry with new managerial and directorial appointments in the Southeast Asia region recently, as companies appear keen to mine a sleeping potential in the area. The cluster of countries is increasingly catching the gaze of some of the industry’s big names; so what exactly is the situation in the region that makes it such a focal point of interest? First and foremost, Singapore reports stability, according to Joe Fong, deputy managing director at distributor Electronics & Engineering: “Being a mature state, Singapore has good infrastructure that has been developed and established over the past two decades. We are seeing a shift from mass-market experiences and are now heading more towards a higher quality, technology-driven business environment. The demand for AV/IT is the forerunner to deliver media to audiences. Consumers
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will still flourish but in a defined technology-driven direction, particularly as a leader in the IT-driven environment where Singapore strives to position itself.” But with multiple markets, each with their own climate, the wider focus may not be as straightforward as simple market prosperity. Andy Yulianto Jahya, consultant at HEAR Pro-Audio Consultants of Indonesia, puts it simply: “At the moment the pro-audio market in Indonesia is still not too good”. Fellow Indonesian Rudy Winarto, director of Melodia Sound & Lighting Systems, explains: “A lot of Chinese manufacturing is coming to Indonesia, and there are also a lot of copy/fake products in the market. Digital mixers are now very affordable as a number of manufacturers are developing new digital consoles. Speakers are also becoming more affordable, with more variety in model/type from each brand, so people can choose according to their needs, budget and expectation.”
Bumps in the road And the situation is not all that different across the sea in Thailand, as Fuzion Far East’s director Siri Wongkamolchun remarks: “Thailand has been stuck in a political situation for a long time. The atmosphere has been unstable. However, I would say the Thai pro-AV market is doing OK despite those problems.” These issues, entirely extraneous from the industry, are pervasive throughout the region, and stifle development that could otherwise flourish. “Our market has suffered from financial problems and political issues. Last year is just another bump in the road,” Wongkamolchun continues. “Our domestic demand went down, trade and tourism shrank and so on. Our market seems to react and bounce back quicker than before. In the end, everyone is trying to move forward.” Thailand isn’t the only market in the region suffering as a result of issues unrelated to the industry. Indonesia’s pro-audio sector is feeling the effects
of ‘uncertain political turmoil’ according to Jahya, creating an economy where disposable income is at a premium. “People are looking for affordable products as economic growth in Indonesia is not yet stable due to the political conditions in the country,” adds Winarto. “It still seems unpredictable even now; we have our new president, but there are still lots of internal political conflicts all over Indonesia.” These struggles could be further attributed to a gradual move into the global lens. Juggling the duress of political uncertainty, Indonesia’s move into the global market is further exacerbated by exchange rate woes. “The exchange rate of USD to Rupiah went up a few months ago, which makes it difficult to market US products in Indonesia,” explains Winarto. “Fortunately the exchange rate of Euro to Rupiah seems a little bit lower and more stable, so that brings advantages for European products in the Indonesian market compared to US products, but still, the economic
GEO FOCUS: SOUTHEAST ASIA
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Population: 618 million Annual GDP growth, 2013 (%)
Singapore
Indonesia situation in the country does not yet support it due to political issues.” Singapore, however, appears to be enjoying much greater stability. “We have been able to balance serving the masses and attending to quality, while not forgetting to keep updated with future trends,” notes Fong. “The climate is healthy as long as the company is equipped with technology-savvy human resources. Products can be purchased anywhere but the need to design, build and maintain these sophisticated systems are very much still in demand.”
A rising storm Thailand has sought to legitimise itself within the global frame by attempting to reproduce the successes of established western markets, as Wongkamolchun explains: “Our industry has been influenced by developed countries such as the UK, US and many countries in Europe. We see more and more specialists/consultants from abroad, but it will take time before our industry is going to get more mature.” But such a fixation on the developed world has led to a unique atmosphere in the attempt to imitate it, cultivating a culture of competition. And in a culture of competition, companies have to be prepared to fight if they want to be left standing. “Developing industries in Indonesia and around the world are all competing to be the best, and that is unique in the Indonesian market because of the diversity of cultures that demand very diverse treatment,” comments Jahya. “My company’s fate is at a stage of survival in the face of competition from increasingly fierce audio business in the country.” In the face of stiff competition, differentiation is vital. Wongkamolchun argues that, in Thailand, those key differentiators are quality of both technical
support and service: “Even though there are many good products available in our market, our technical support and service seems to Thailand be our key to differentiating us from other competitors,” he states. Markets are wavering across the region 1% 2% 3% 4% 5% 6% as it makes its shaky steps over to the global playing field, but industry professionals believe things are set to get worse before they new markets in our territory; there will always be new get better, with competition heating up as proopportunities and new people.” audio companies fight for a leading position in the In Indonesia, future business must carefully developing climate. account for the political climate. “Our strategy is to “In the next five years the market will be more keep progressing with innovation and expansion of the difficult, because a lot of its competition is becoming business while keeping an eye on the progress of the increasingly extensive and rigorous,” Jahya notes. “My political issues in the country,” comments Winarto. “At strategy would be to have a superior product that is the same time we do believe that we have to maintain of good quality with a price that is very affordable for our after-sales service because that is the most marketing in Indonesia.” important key to successful sales.” Fong believes that not only will competition Finally, Fong believes that technological increase, but so will technological development development, not just for Singapore but the entire and demand: “Cloud-based media management region, is the key to achieving global aspirations. and delivery will change the demand for native “Maintaining a good team of talent across all isolated content archives. Virtual, wireless or digital aspects of the organisation to establish teams transportation of media will change the business catering to the cultural and business needs across the environment in such a way that some products will region is the only way to go. Singapore does not have become obsolete.” the sheer mass, but the region does. The technological development for the SEA region has a good two unique solutions decades to go of major infrastructure development As each market looks to the future, their distinctive to cater to the growth of the mass-to-middle income. strategies are indicative of the problems that each The growth of China and India into developed or country faces across the region. mature economies will demand for not just entry-level “I think there are a few things that we need to keep goods and services, but ones with quality.” working hard on. We need to keep our standard on A unique, complex and multi-faceted climate technical support, system knowledge and after-sales slumbers in Southeast Asia; a site of simmering service,” suggests Wongkamolchun. “Our relationship competition, symptomatic of a developing market. with the client is also important. We spend a long time It is little wonder it has piqued attentions around the building trust and relationships with them – we have world; it is fertile ground, but there are challenges to to keep it that way. We are going to keep exploring overcome before its potential can be reaped.
May 2015
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STUDIO PROFILE
A TRIP DOWN WINDMILL LANE
Its former dwelling – known locally as the U2 Studio – may have been reduced to rubble last month, but Matt Fellows finds that Windmill Lane Recording in Dublin remains alive and well in its not-so-new location.
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pon hearing the news in April that the old home of Windmill Lane Studios in Dublin, the facility where U2 recorded their debut album Boy as well as The Joshua Tree, had been demolished, much of the industry realised it had lost another prized relic from an increasingly distant age of recording. “It was put together by myself in 1978 along with three business partners; James Morris, Russ Russell and Meiert Avis,” explains original owner Brian Masterson. “Those three were already successful film editors. I was establishing a name for myself working as a freelance engineer in various Dublin studios. The decision was made to build a studio that would appeal to international clients, and John Storyk 20
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from New York was retained to design the studio.” But that’s not the whole story. While it did indeed witness key moments of music history, the site that is now to be made into multipurpose office, retail and residential blocks has not been the site of Windmill Lane Recording since 1989. “The partners wanted to devote their time and energy to launching Ireland’s first and only commercial TV channel,” Masterson explains. “I acquired the name and the equipment and went into partnership with Andrew Boland who then ran Ringsend Road Studios.” Relocating to Dublin’s Ringsend Road but keeping the Windmill Lane name, Masterson found that the studio stood to benefit from the advantages of the new location: “The new premises were much bigger than what was available at
the original location. This did give scope for expansion and diversification.” Over the years the studio has gone on to see acts such as The Rolling Stones, David Bowie, Metallica, Norah Jones and Lady Gaga cross its threshold; it still stands there today, now staffed by an entirely new team. Current studio manager Niall McMonagle saw the move as an easy one to make: “We took over Windmill Lane about seven years ago. If we hadn’t, I don’t know what the future would have held. The previous owner and one of the engineers/producers approached us to say, ‘we’d love for you guys to take it over. You’re the right people to do it,’ because we owned, and still do own, another studio off Camden Street, another area of Dublin where the company had grown and had reached
bursting capacity. So we were looking to move and then Windmill Lane came up and it was a no-brainer really. We’d dreamed about it for some time, and then the opportunity presented itself.” So what has been their secret to success and longevity when so many studios of the same generation have been forced to close their doors? Like any studio manager, McMonagle is keen to talk first and foremost about layout and quality equipment. “On the ground floor, we’ve got two studios: Studio Two and Studio Three. Studio Two is a typical band-style, rock ’n’ roll size; there are a couple of live rooms and a couple of vocal booths with a standard size control room with the SSL G+ Series in it. Studio 3 is a 5.1 surround sound room, which is used more for mixing, mastering and post-
STUDIO PROFILE
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Niall McMonagle
The Neve VR is the centrepiece of Studio One
comes to survival in the studio sector. And Windmill Lane Recording certainly attributes the stability it enjoys today to these key differentiators and auxiliary avenues of business.
Profitable partnerships
production projects. And the entire top floor is devoted to Studio One, which has a big control room and a huge live room – it can hold between a 70- and 80-piece orchestra. It’s got a Steinway in there, harmoniums and Hammond organs; the Neve VR is the centrepiece of Studio One and is a thing of beauty! We also have some lovely outboard, the usual classic stuff like UREI 1176s, distressors, Summit EQs and we also have a lovely EMT Plate Reverb. “Apart from the equipment, which you can find in other big studios, our recording space is fantastic. As with a lot of studios some rooms just work, and this one is right up there. It’s just a lovely room. Obviously acoustically – there’s no parallel walls and there’s nice absorption and dispersion – but also aesthetically. A lot of bands want to come in and have
a clean, comfortable room to set up all their gear in and relax. The large live area is thunderous for drum kits and we love nothing more than setting up a big drum kit and shaking the building.” “I’d like to say I had a way of working but it tends to change from week to week,” he continues. “Trying a new technique or a new piece of equipment or I hear somebody else’s song and think, ‘let’s try and do that’. Trying to change it as regularly as we can is important but the common denominator for us is that we try to get it right in the room as much as we can, working with the band to get them polished, whether it’s just rehearsals or a bit of pre-production before we hit the record button.” But even when you’re as well equipped as all that, it can’t hurt to have some secret weapons when it
“Without doubt our involvement in education has been a huge part of the company for quite some time,” McMonagle remarks. “As well as the usual recording and mixing, we are also heavily involved in education with Pulse College. We have a range of courses from certificate up to Masters level covering areas such as Music Production, Sound Engineering, Film Production, Game Development and Animation. The college has helped the studio and having facilities like we have here has also helped the college. “Without the education side of it I don’t know how the studio would be getting on,” he says. “Like for any other big studio around the world, it’s been difficult. For us luckily we’ve been getting bigger and bigger. The request for the studio is still as big as ever.” The studio has also embraced online distribution with its Windmill Lane Sessions, a series of live performances and interviews hosted on independent. ie, Ireland’s most visited website. “We’re all big fans of music here, and we love live music,” McMonagle explains. “Studio One is so big it seems a shame when it’s not being used for a session. Why not put on a gig? We’ve got so many film students here and film equipment we thought ‘why don’t we film it?’ We’d
had various ideas of doing music shows and approaching TV stations. We did one or two pilots and roped in favours from everyone we knew in the industry, and then independent.ie actually came to us and said ‘Hey listen, we want to do this music series,’ and we said ‘You know, it’s funny, that – so do we!’ We showed them the pilot stuff that we’d done and they loved it, and it pretty much just hit the ground running. “The idea is that we can give all sorts of artists, from young, never-beforeseen bands to old stalwarts who’ve been around the scene for a long time, another platform and another route to market,” he continues. “It’s generally a two- or three-hour session. We set them up, they record two songs, they do an interview, and out the door again, so it’s nice and quick and snappy, so even bands who are doing a gig in the evening can pop in on the morning and do a session. “We were trying to put together a show that was quality musicians doing their thing live. We record, we film it, we don’t do any of the studio magic – there’s no Pro Tools editing or Autotune or any of that studio trickery. We’ll mix it and tart it up a bit, but what goes into the box comes out. So the band’s have to up their game. They get two or three takes, but there’s no editing between takes, there’s no comping. The emphasis for us is on the bands doing their thing, which is the way we love to record. It’s been going great for the past few months and we’ll be looking to up the level and try to attract more high-profile artists.” While the importance of quality and breadth of service cannot be understated, Windmill Lane has shown over the years that the true key for studios hoping to make a stable leap to the current generation is reinvention. For Masterson, the spirit of the original Windmill Lane is still very much alive: “Speaking as someone who now uses the studio whenever possible, I think the same ethos of providing wonderful facilities with a great bunch of talented engineers and assistants is something that hasn’t changed. And that fantastic Neve VR is still going strong and sounding as warm and musical as ever.” www.windmilllanerecording.com
May 2015
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FEATURE: IMMERSIVE AUDIO
SOUND ALL AROUND
Björk’s Black Lake Room at the Museum of Modern Art in New York City Picture: The Living
Virtual reality and mobile technology are leading the way for immersive sound experiences, but a lack of standard might be holding widespread adoption back, writes Erica Basnicki.
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xperiments in immersive audio have been around since the early 1930s, when Alan Blumlein figured out that listening to two audio channels was much better than one. Yet it has taken the better part of a century for it to capture the imagination, and the ears, of the masses. Before we continue, a point of order: defining immersive audio. For now, and for the sake of this article, ‘immersive’ audio includes object-based, channelbased and scene-based audio. In other words, sound in space – regardless of how that’s achieved. The definition itself hasn’t slowed its adoption. A lack of a technical standard, however, has (but more on that later). The confusion is partly born of a heated competition for surround sound supremacy in the cinemas between 22
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Dolby Atmos, Auro 3D and the recently released DTS-X. At least, the cinema experience is probably the one most associated with surround sound technology. Where the actual demand for immersive audio is increasing is, according to many, from another entertainment sector entirely. Both Google and Facebook have plunged headfirst into the world of virtual reality (VR). The former with Google Cardboard, the latter having acquired Oculus in 2014. With two media giants heavily invested in its success, an explosion in VR entertainment isn’t far off. “A lot has changed since virtual reality became a ‘thing’, all thanks to Oculus for that,” explains Varun Nair, founder and VP of products at Two Big Ears. “All of a sudden immersive audio stopped being a technology that existed for the sake of
it existing, or adding a bit of something extra to an experience. It became something that is ultimately very crucial: you’ve got great 3D visuals creating a sense of realism, and the audio needs to match up to it.” Two Big Ears was founded in 2013 by Nair and CEO Abesh Thakur. The company designs immersive and interactive audio tools, among them the real-time 3D audio and environmental modelling engine 3Dception. Designed with VR and game audio applications in mind, 3Dception recently powered the audio for Björk’s groundbreaking video for Stonemilker. It’s the first official single from her most recent release, Vulnicura, shot on a desolate beach in Iceland using a 360º camera. In order to see the video, you would have had to attend one of the Rough Trade launches held in New
York and London, or the exhibition at the MoMa PS1 in Long Island City. 3Dception is also in place for the ongoing retrospective of Björk’s (mid-) career held at the Museum of Modern Art in Manhattan, Songlines. As part of the exhibition, Songlines presents an avant-guard audio tour of sorts; a location-based augmented audio experience that runs off an iPod touch. “It’s a great opportunity to show off what our technology can do,” says Nair. “We created all this spacialised audio off an iPod Touch, which isn’t exactly the most powerful device out there.” Back on this side of the Atlantic, Heaven 17’s Martyn Ware has been in the business of immersive audio for the past 15 years. In 2000 he and former Erasure keyboardist Vince Clarke formed Illustrious, and along with it Tino Fibaek developed their own three dimensional
FEATURE: IMMERSIVE AUDIO
www.audiomediainternational.com
Fairlight 3DAW 3D audio workspace adds immersive sound capability to existing DAWs sound technology, the 3D AudioScape surround-sound system. As Ware explains, demand for their work remained steady until the hugely successful Tales of the Bridge in 2012, which transformed London’s Millennium Bridge into one giant 3D sound installation. Since then, interest in Illustrious’ work has risen, and Ware is particularly excited about an upcoming project in Liverpool. From 15 May until 5 July, the Liverpool ONE shopping centre on South John Street will surround visitors with the sounds that recall the special relationship between Liverpool and New York, in celebration of the 175th anniversary of the first Cunard transatlantic sailing. It’s a world first for outdoor shopping centres. As Ware explains: “This is a commercial opportunity, which is for the general public. This is what turns me on more than anything else; it’s not done for an ‘art’ audience or a ‘music’ audience.
The exciting thing for me is that this is commissioned by a commercial entity. Now that we have this as an exemplar, it takes away the risk for a lot of other commercial organisations who might wonder ‘Will this work for the general public?’ Proof: it does work for the general public.” As both a successful music producer, and a 3D sound enthusiast, Ware is in a somewhat unique position to assess the potential of immersive audio to cross over to the wider music industry. Björk’s VR experiments aside, can we expect more immersion from our music? “I believe in 3D sound, and it wouldn’t be terribly difficult for us to release something in a binaural format. But you’ve got to look at it from the point of view of what is the actual benefit to Heaven 17. Yes, people might say ‘yeah, that’s cool’ but really it needs a whole swathe of people, or somebody much more popular than us to do it. Otherwise it’s just a curio,” says Ware.
“Even when Pink Floyd released stuff in the ‘70s in quadraphonic, nobody bought that. And that’s when people were buying stuff! From an artistic point of view, definitely interested. With Illustrious, we had two albums over a decade ago that were released in binaural on Mute records. I don’t remember getting any emails saying
‘You’ve done those two albums, please put some more out in that format’. It’s about creating a snowball effect. It would need somebody with a lot of marketing resources or huge popularity worldwide to make this happen, I think.”
Tool of an emerging trade As early adopters of the new medium,
May 2015
23
FEATURE: IMMERSIVE AUDIO both Two Big Ears and Illustrious have had to develop the immersive audio technology in-house. This is becoming less of an issue for content creators as major DAW manufacturers recognise the need for such tools. Last year, Fairlight launched 3DAW, the company’s 3D audio workspace. Designed for film and television postproduction markets, 3DAW supports object-orientated audio such as Dolby Atmos and DTS MDA, as well as fixed bus formats from 5.1, Auro-3D all the way up to NHK’s 22.2 format. It’s a large number of formats to embrace. Surely we’d be better off with just one? “Complicated question,” says Fairlight CTO Tino Fibaek. “Fixed bussing is possibly easier to get to grips with and to deliver, but object-oriented has clear advantages in speaker flexibility and being somewhat future-proofed in that it will support speaker formats that are yet to be defined.” Fibaek is focused on the silver lining of so many competing formats – demand for immersive audio is strong, and growing stronger: “We expect the next segment to grow will be home cinema, with at least four strong delivery formats in the running. Following that we believe demand will come from live broadcast. This will be a combination of object-oriented and immersive features, offering the viewer a better sound experience with more control. Finally, we think that the 3D technology created for the above segments will begin to find home in performing arts; opera, theatre and concerts.” Live immersive audio is an area global technology company Barco is keen to develop further. As the drivers of Auro
24
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11.1 technology, the company upped its immersive game with the acquisition of 3D audio experts Iosono in late October 2014. Prior to the acquisition, Iosono’s spatial audio processor had already been central to the world’s first 3D opera, Neither, as well as Kraftwerk’s Catalogue 12345678 2013 concert series in Tate Modern’s Turbine Hall. The company also had a hand in MoMA’s ‘Björk’, where another video – Black Lake – used the Iosono CORE audio processor and 49 speakers by Bowers & Wilkins to engulf a specially built room with sound. Despite its success with live immersive sound, Brian Claypool, Barco’s senior director of strategic business development, sees momentum building in mobile technology. “How many times have you watched movies, TV shows or played games on your smartphone or tablet in the past month? Is this different than five years ago? Of course! Consumers are more demanding of having their content where and when they want it. This presents a great opportunity to take certain innovations in audio technology and create more value to the way consumers receive and view their content.”
The Standard Issue While companies such as Fairlight see competing formats as a sign of healthy demand, others see a lack of an open standardised format as an anchor holding back a widespread adoption of immersive audio. “We are actually expecting the introduction of such a standard by the Society of Motion Picture & Television Engineers (SMPTE) this year and Barco is ready for it,” says Claypool.
“This will put an increased focus on quality of content, so that audiences understand the full potential of what immersive sound can be. Also, content creators will be able to focus on one workflow rather than making sacrifices to create multiple versions of one mix. The arrival of a true standard in the marketplace will help achieve this.” In terms of mobile entertainment, that standard has already arrived. In March of this year, the Audio Engineering Society established the AES69-2015 standard which, according to the website, “provides an important framework for the growing binaural and 3D personal audio industries”. The question of a cinema standard was also addressed at the 57th AES Conference on the future of audio entertainment technology, which included a discussion chaired by SMPTE’s Brian Vessa. “While there are many commercial solutions being offered to movie theatres, the loudspeaker layouts differ and it is believed that a common interchange format will enable cinemas to invest in immersive audio more securely, whatever the system they employ, while being able to receive content in one format that can easily be mapped to the system in question,” explains Francis Rumsey, chair of the AES Technical Council. “A lot depends on the specification of suitable renderers that can map audio to the loudspeaker layout in question.” An August 2014 SMPTE newsletter notes that Peter Ludé, chair of the SMPTE Working Group on Immersive Sound, is “hopeful a standard will be ready or close to ready for publishing before late 2015. “We don’t know what the actual, final standard will be yet, of course, because the work is still in progress. But the expectation is that it will be a relatively minor change from [the technology] that is in the field today. I think, for the most part, people are envisioning relatively minor software upgrades to get them from their current status to the new standard when the time comes. Everyone feels it is worth the time and expense to get to a single interoperable immersive format; that is the main reason this process is moving forward.”
“We expect the next segment to grow will be home cinema, with at least four strong delivery formats in the running. Following that we believe demand will come from live broadcast.” Tino Fibaek, Fairlight
Back to the Future Expect to hear a lot more about immersive audio over the next few weeks, as the BBC is holding an event at Broadcasting House on 19 and 20 May called Sound: Now & Next. Over the course of the two days, those lucky enough to be in attendance will hear talks from industry leaders about new directions in sound production and technology. Nair is among the confirmed speakers covering the dedicated section on “the art and technology of immersive sound for headphones and loudspeakers”. “There are no more tickets available for the event but we aim to put some content online afterwards,” explains Chris Pike, senior scientist – audio at BBC R&D. A 3D audio specialist, Pike was responsible for mixing VR audio for Björk’s Stonemilker video, working closely with Nair and Two Big Ears. “At BBC R&D we are working on immersive audio technology and its application to broadcasting. We work in partnership with academics and industry bodies to advance what is possible with the technology and ensure that appropriate standards are in place for the future. We also work with BBC content creators on production experiments, which when possible are made available to the audience for feedback.” Audio Media International will have its ear to the ground, with details and further thoughts on the event available in our June issue. www.aes.org www.bbc.co.uk/rd/events/ sound2015 www.fairlight.com.au www.illustriouscompany.co.uk www.iosono-sound.com www.smpte.org www.twobigears.com
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BROADCAST FOCUS
BODY BUILDING
Despite making significant additions to its handheld and install lines recently, DPA has identified bodyworn mics as one its main focuses at the moment, but what are the reasons for this? Let’s take a look. As well as offering the best possible sound quality – a fairly obvious requirement for virtually any audio product, not just microphones, of course – today’s body-worn mics need to be capable of withstanding more than a little bit of rough treatment. “They were developed to be less sensitive to humidity, for instance, and making them so it is possible to clean them up once water gets into them,” Brixen continues. “We also knew that they had to survive hard handling sometimes, which is why we introduced heavy-duty versions.” Eddy Brixen
P
erhaps it’s because they’re designed to be heard but not seen, but body-worn mics can hardly be considered ‘sexy’ equipment. These delicate little devices might not have the visual appeal of an expensive vintage ribbon microphone, and unlike the live mic market you never hear about famous actors, broadcasters or reality TV stars signing lucrative endorsements with their manufacturers over these products, but try telling DPA that body-worn microphones are not important and they’ll have something to say about it. And that’s because it’s become one of the most crucial areas of business for the Danish firm. So how did that happen? Well, it seems the manufacturer has been a leader in this field for decades, as DPA’s audio specialist Eddy Brixen explains. “The 4060 was the first miniature microphone that was developed by DPA, and when it was introduced in the late 1990s it was rather amazing because it was much, much better than what else was on the market. “But nowadays there is so much more television being produced than at that time, and we now have all those reality shows, so we’re seeing quite a lot of consumption of these microphones.” 26
May 2015
BRANCHING OUT Now that’s it been around 20 years since its first models were introduced, how has the technology progressed since the early days of DPA’s body-worn mics? Although some features have remained largely unchanged – the phrase ‘if it aint broke, don’t fix it’ springs to mind – there is now a solution to suit a myriad of scenarios. “The basic capsule design is more or less the same but there has been a need for optimisation since then,” Brixen says. “When the first ones were designed it was essential to achieve as high a sensitivity as possible to get as much out of the microphone as you could, but then we found out that some kinds of pocket transmitters will not accept such a high output from a microphone, so we had to make microphones with different sensitivities. “Basically, the microphone has been developed due to the demands and the requirements for the different applications that you now have.” What would Brixen say are the most popular models in the body-worn range at the moment, then, and why does he think that is? “The d:fine headsets because they are so easy to wear – you barely feel them when you have them on – and they offer great sound and come in many different lengths. We have to offer lots
of sizes as some people have very big necks! There’s no ‘one size fits all.’ Another thing that sets body-worns apart from other kinds of mics is the method of positioning. Whereas most types are designed to be placed as close as possible to the source, these products often need to deliver results from a distance – and that’s not the only complication. “The 4071 compensates for being positioned on the chest, and for the loss you get through clothing, so on a filmset it can be used hidden under costumes,” Brixen adds. “We have put a lot of effort into making these linear microphones fulfill whatever the needs may be when placing a microphone on a person. We have different kinds of fittings for hiding the microphone, which is especially useful for the film and TV people.” “If you have a microphone that is not compensated and you put it on the chest of a person you lose something like 10dB in the range of 2-4kHz so it’s really important for the intelligibility,” he continues. “Very often these mics are not compensated – we still see a lot of that.” “Also, you have to think about wind noise when doing outside recording – it’s not just the sound going into the microphone that you have to consider.”
That’s an issue that DPA has been continuing to tackle with many of its latest lines – not just the body-worns – and the brand new d:facto Interview Microphone is a good example. And if the new d:screet Necklace Microphone (reviewed in the November 2014 issue of Audio Media) – which is simplicity itself but doesn’t skimp on sound quality – is anything to go by, it wouldn’t be too much of a surprise to see the company come up with some more inspired ideas within this sector of the market in the near future. www.dpamicrophones.com
TECHNOLOGY: HOW TO
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ACHIEVING THE IDEAL MONITORING SETUP
Speaker placement in studios has always been a topic of much debate, so we invited a true expert, Carl Tatz, to oer us his thoughts on the matter.
W
hat’s the most important element to consider when creating a control room? A console equipped with the latest features? As much outboard equipment as you can aord? According to award-winning studio designer Carl Tatz, it’s accurate monitoring. As well as building spaces for a variety of clients, the ex-engineer, producer and studio owner – he sold his facility to none other than Sheryl Crow in 2003 before heading into the design business – is the creator of the PhantomFocus System, a custom turnkey solution designed to oer highlevel monitoring accuracy, no matter what the user’s choice of monitors may be. The system has been praised by renowned engineers Ken Scott (The Beatles), Elliot Scheiner (The Eagles) and many more. He knows what he’s talking about then, so what are his main recommendations when it comes to setting up a monitoring system? Start o by working out the listening position and where the speakers should go, according to Tatz. “You want to ďŹ nd a way of calculating the modes in your room (see diagram, right) and then catch the ďŹ rst reections. When you decide where you want the listening position to be in your room the ďŹ rst consideration you want is symmetry – at least from the listening position forward – so whatever the right speakers sees, you want the left to see boundarywise, otherwise they will react to the room dierently,â€? he says. “You can sit in the listening position and have someone go along the wall with a mirror and as you’re looking at the mirror when you see the speakers, that’s where you want to put some absorption. The back wall should also be absorptive. I know you see people use diusers – I’m 28
May 2015
AXIAL MODE CALCULATOR Enter room dimensions Feet Inches Height: 9 ft. 0 in. (108in.) Width: 15 ft. 9 in. (189in.) Length: 18 ft. 6 in. (222in.) Volume: 2622ft 3
1st Through 4th Axial Mode Frequencies 4th 1st 2nd 3rd Meters 251Hz 2.74M 63Hz 126Hz 189Hz 36Hz 144Hz 4.80M 72Hz 108Hz 5.64M 31Hz 92Hz 122Hz 61Hz 74.3M 3 Enter speed of sound: 1131.0 ft/S (344.7M/S)
Height: 108 in.
<
>14in. >18in. 251Hz 189Hz
>27in. 126Hz
>41in. 251Hz
not a fan of that really, but itâ&#x20AC;&#x2122;s better than nothing. â&#x20AC;&#x153;Back in the listening position, if you clap your hands and you hear a whole bunch of ambience coming back, thatâ&#x20AC;&#x2122;s what you want to calm down with absorption because all you want to hear is whatâ&#x20AC;&#x2122;s coming from the speaker forward.â&#x20AC;?
ALL ABOUT THAT BASS So what else should we know? Some recording professionals regularly struggle with the low-end, and many will recall situations where a track has sounded perfect in the studio, but then listening back in an alternate setting has led to frustratingly diďŹ&#x20AC;erent results. Why is that, ďŹ rst of all? â&#x20AC;&#x153;At approximately 125Hz there is a huge 10-15dB dip and it doesnâ&#x20AC;&#x2122;t come back up until about 40, 60, 70, depending on your speakers and your room,â&#x20AC;? Tatz continues. â&#x20AC;&#x153;Between that 125 and 70Hz youâ&#x20AC;&#x2122;ve got what I call The Grand Canyon of missing information and thatâ&#x20AC;&#x2122;s why, when people are mixing in any environment, they have a hard time with the low-end â&#x20AC;&#x201C; itâ&#x20AC;&#x2122;s always the hardest thing to determine. â&#x20AC;&#x153;Thereâ&#x20AC;&#x2122;s often that classic case where youâ&#x20AC;&#x2122;ve got the bass really punchy and
41in.< 251Hz
27in.< 126Hz
18in.< 14in.< 189Hz 251Hz
>24in. >31in. 144Hz 108Hz
>47in. 72Hz
>71in. 144Hz
>95in.< 108Hz 36Hz
71in.< 144Hz
47in.< 72Hz
31in.< 24in.< 108Hz 144Hz
>
Length: 222 in.
<
>
Width: 189 in.
<
>54in.< 189Hz 63Hz
>28in. 122Hz
>37in. 92Hz
>56in. 61Hz
>83in. 122Hz
>111in.< 92Hz 31Hz
83in.< 122Hz
56in.< 61Hz
37in.< 28in.< 92Hz 122Hz
>
Tatz recommends a 67.5in space between monitors how you want it to sound, and then you take it out some place and thereâ&#x20AC;&#x2122;s way too much low-end. And thatâ&#x20AC;&#x2122;s because youâ&#x20AC;&#x2122;re boosting â&#x20AC;&#x2DC;The Grand Canyonâ&#x20AC;&#x2122;. You might have that peak at 60, but youâ&#x20AC;&#x2122;re still missing all that low-end information. But can anything be done about it? â&#x20AC;&#x153;Well you canâ&#x20AC;&#x2122;t EQ that up as youâ&#x20AC;&#x2122;d blow the speakers, but if you get a subwoofer or two, cross it over and experiment then you can try and ďŹ ll in that hole, and the PhantomFocus System is perfect for that,â&#x20AC;? Tatz continues. â&#x20AC;&#x153;The alternative is to learn your speakers, so if the bass sounds shy then itâ&#x20AC;&#x2122;s just right or if it sounds just the way I want then thatâ&#x20AC;&#x2122;s actually too much. You have to compensate or second-guess yourself, and thatâ&#x20AC;&#x2122;s what people do all the time, whether itâ&#x20AC;&#x2122;s in a home studio or a professional space. â&#x20AC;&#x153;You could learn to mix in an empty swimming pool â&#x20AC;&#x201C; eventually your mind and ears ďŹ gure it out.â&#x20AC;?
But surely going out and buying an expensive set of speakers would also help? The more you spend the more you get, right? â&#x20AC;&#x153;It doesnâ&#x20AC;&#x2122;t matter how â&#x20AC;&#x2DC;goodâ&#x20AC;&#x2122; they are â&#x20AC;&#x201C; and Iâ&#x20AC;&#x2122;ve tuned so many diďŹ&#x20AC;erent speakers â&#x20AC;&#x201C; they all do the exact same thing. Over the years theyâ&#x20AC;&#x2122;ve been getting better but itâ&#x20AC;&#x2122;s a slow process â&#x20AC;&#x201C; itâ&#x20AC;&#x2122;s not like digital consoles where there have been quantum leaps,â&#x20AC;? remarks Tatz. And those who spend countless hours ensuring their room sounds perfect from corner to corner are probably overcomplicating things. â&#x20AC;&#x153;The only thing you really need to worry about is one position. People say you need to tune the room but you donâ&#x20AC;&#x2122;t tune the room; you tune the speakers. Iâ&#x20AC;&#x2122;ve done rooms that are seven feet wide and nine feet deep and they sound as good as a million-dollar studio in that position, and thatâ&#x20AC;&#x2122;s all you really want.â&#x20AC;? www.carltatzdesign.com
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 Buyerâ&#x20AC;&#x2122;s Guide supplements focus on key technologies Q DAWs June (copy deadline 19/05) 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
SMALL-FORMAT PA
Expert Witness
T
here are many scenarios around a modern school where the permanent distributed audio system or auditorium system is not appropriate, and a separate, portable PA system must be used. When specifying any audio system there are a number of factors that need to be accounted for, such as audience size, coverage area, audio source style, number of sources and audio quality. In a school environment there are some additional considerations, which will refine the selection of the speakers to be used. The goal of any truly portable PA is to be as light and compact as possible, but this is an even more critical factor when considering a school environment. Often the equipment will be moved and set up by students or teachers, not professional operators. A speaker will need to be carried from a storage location to its destination, and at that point often placed on top of a speaker stand. OHS (Occupational Health and Safety) guidelines often restrict the weight allowed to be carried by one person, so these should be taken into consideration. Another consideration is for the safety of the students, particularly younger students who could accidentally cause a speaker stand to topple over. A lighter speaker and shot bags to help anchor the stand will be a harder setup to push over than a more top-heavy, large speaker setup. Immediately this will eliminate some speaker designs – most quality 12in and horn-style boxes, wooden boxes, some powered speakers and larger subs will exceed weight requirements. Cheaper plastic boxes may allow for larger drivers with less weight, but with poorer audio quality as a compromise. An ideal setup for a 300-capacity audience listening to pre-recorded material, performance and vocal announcements would be a setup featuring two 8in and horn top boxes on stands, or, if budget allows, extension poles on small subwoofers. The Electro30
May 2015
Nathan Barrios of Australia-based The P.A. People outlines what you need to know when specifying a loudspeaker system for one specific area of application.
Voice ZX1A and ZXA1-SUB are a great example of this setup, and are an entrylevel professional product which is light, compact, powerful and reliable.
POWERING UP This setup, as well as those from other manufacturers, are usually powered speakers. By placing small amplifiers in each speaker – adding a small amount of weight – a heavy main central amplifier is no longer required. The system can also be expanded to include more speakers without the impedance matching consideration required in a passive speaker and amp combo system. In a powered system each speaker must be provided a line level source from the mixer as well as a mains power lead, so safety precautions when dealing with mains power should be considered, particularly when dealing with young students. An added benefit to small powered speakers is the ability to use them on their own for even smaller applications. A PE or gym class could use one speaker with an MP3 player and microphone connected in directly, or a similar setup for a school fete, to address a smaller audience. These speakers are often also utilised in a school’s larger installed PA system as convenient fold back wedges. The other major event a PA is often required for is carnivals and school sports days. In this instance, the focus should be on selecting a speaker system to give maximum coverage and sound output. Employing a system that features many lower-powered speakers is ideal, and the speaker of choice would be a horn-style paging speaker. A typical school oval can be covered evenly with four to six pairs of horn speakers on stands when positioned carefully, but the added benefit of this system design is in its constant voltage back bone. Constant voltage systems consist of a central amplifier with passive speakers, similar to a traditional low impedance system. In the case of constant voltage, impedance matching
isn’t an issue. One speaker run can be used to drive as many speakers as the selected amplifier will handle. A 250W amplifier will drive a total of 250W worth of 100V line speakers, regardless of how they are connected or their quantities. The system would deal with a single 250W speaker in the same manner as it deals with 10 25W speakers or 250 1W speakers. Another feature of constant voltage systems is that all speakers can be driven on one speaker cable run, with each speaker or cluster of speakers connected in series, reducing setup time and complexity. This speaker run can be extremely long with minimal signal loss. For example, a typical sports oval would be 400m around on a single cable run with no loss. TOA provides a range of economical, light and robust horn speakers, such as the SC615M and SC530M, and also offers a range
of ‘music horns’ like the CS304, which provide a smoother, higher quality audio while still functioning as a horn speaker. In summary, a small-format portable PA system in a school environment must be lightweight, easy to set up, simple to operate and safe for students and operators. Understanding the user’s needs and the style of events they intend to hold will ensure the correct equipment is selected.
Expert Witness Nathan Barrios is showroom operations manager at The P.A. People, a specialist integrator and contractor in sound reinforcement and AV systems located in New South Wales, Australia. www.papeople.com.au
International
SMALL-FORMAT PA
www.audiomediainternational.com
D&B
Y-SERIES Not just any old workhorse: the absolute Y-Series from d&b.
T
he Y-Series from d&b audiotechnik provides flexible and configurable solutions for any small to medium-sized sound reinforcement requirement, whatever the application style. While the Y7P and Y10P loudspeakers cover point source duties, with the B6-SUB providing extended frequency response, two line source loudspeakers and a matched cardioid subwoofer, the Y8, Y12 and Y-SUB tackle line array tasks. Echoing the d&b V-Series it utilises patented three point rigging. The Y7P and Y10P loudspeakers share the same dipolar 8in driver arrangement centred on a 1.4in compression driver fitted to a rotatable CD horn, facilitating deployment horizontally or vertically. With dispersion characteristics of 75° x 40° and 110° x 40° (h x v) respectively, the Y7P and Y10P offer a multitude of deployment options, individually as a full range system or in combination with other elements from the Y-Series, either ground stacked or flown. An advanced port design delivers extended LF performance down to 59Hz. The B6-SUB complements the Y7P and Y10P, with a single 18in driver built into a bass-reflex design, extending the frequency response down to 37Hz. The Y8 and Y12 loudspeakers are compact, configurable and easily recognisable as the latest members of the d&b line array family. Utilising the same rigging design as their bigger brothers from the J and V-Series, the Y8 and Y12 also share the same 80° and 120° horizontal dispersion characteristics. The Y8 and Y12 can be suspended in columns of up to 24 loudspeakers with splay angles from 0 to 14º with a 1º resolution. The Y-SUB houses a forward-facing 18in driver and a 12in driver radiating towards the rear, producing a cardioid dispersion pattern. Driven by a single amplifier channel, the compact Y-SUB can be ground stacked or flown at the top of a Y8/Y12 array. Both the point source and line source loudspeakers feature two 8in drivers with neodymium magnets mounted in a dipolar arrangement around a 1.4in compression driver.
FACTS & FIGURES Y7P and Y10P: Dipolar 8in driver arrangement centred on a 1.4in compression driver fitted to a rotatable CD horn B6-SUB: Single 18in driver built into a bass-reflex design Y-SUB: Forward-facing 18in driver and a 12in driver radiating towards the rear, producing a cardioid dispersion pattern Yi: Installation-specific versions designed for permanent applications
This driver configuration enables the Y8 and Y12 to offer a horizontal dispersion pattern controlled down to 500Hz. An extensive range of transport solutions and loudspeaker accessories facilitates ease of setup and mobility. While the Y loudspeakers are designed to provide flexible, mobile solutions, the Yi installation specific versions are designed for permanent applications, differing only in cabinet construction and mounting hardware. Special colour and weather-resistant options provide further visual integration and positioning possibilities. The Y-Series assimilates into the d&b workflow comprising the ArrayCalc simulation software, the R1 Remote control software and d&b amplifiers; a process ensuring consistent and efficient results whenever and wherever required. A multitude of line and point source loudspeakers can be combined within ArrayCalc, providing a graphical representation detailing the coverage, level drop and safety aspects of a system setup in a given space. The R1 export function transfers all configurations and settings from this simulation into an R1 Remote control project file, taking into account any system specific functions. The R1 workplace presents an intuitive format for operating systems via the d&b Remote network, through CAN-Bus to access the D6 and D12 amplifiers, as well as the D80, which can also be controlled through Ethernet using OCA protocols. “Since the Y-Series combines both line and point source technologies, the application possibilities are endless, ranging from performing arts to conferences, religious events and live shows,” ponders product manager Werner ‘Vier’ Bayer. “This has been an exciting and illuminating project; we see this Series becoming the workhorse of our industry, on every end of the spectrum.”
CONTACTS
Address: Eugen-Adolff-Straße 134 71522 ,Backnang, Germany Web: www.dbaudio.com
May 2015
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International
SMALL-FORMAT PA
www.audiomediainternational.com
FUNKTION-ONE
F1201/BR118 COMBINATION Funktion-One continues to break the mould, and its small-format PA offering is no different.
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unktion-One doesn’t do typical. The company’s approach to sound design, driven by Tony Andrews, John Newsham and the design team, is centred on creating better audio – regardless of the application. It is no surprise, therefore, that when it comes to the fairly typical format of ‘the small PA’, Funktion-One has created something that it claims is capable of transforming any small to medium-sized space in to an immersive audio environment. The company has a range of speaker and bass enclosure combinations, but in this instance we are focusing on the F1201 and BR118 pairing – two products that are fairly recent additions to the product range. The F1201 is a trapezoidal cabinet, which features a custom-designed 12in driver and a 1in compression driver, which combine to deliver clarity across the 60Hz to 20kHz frequency range. The high passive crossover point ensures the crucial vocal range is reproduced with minimal distortion and no external processing or corrective EQ is required. In terms of efficiency, the F1201 offers 100dB 1W at 1m. Integral M10 fly points and a rotatable 90° x 45° HF horn makes installation straightforward, even in low-ceiling environments. Alternatively, the F1201 can be mounted on to a bass enclosure, like the BR118, using the integral pole mount, which is fitted as standard. The BR118 is a single 18in bass reflex enclosure, which delivers deep, elastic bass – even at close range. Its unique tuning results in a relatively compact size and more definition than is usually available from this type of enclosure. This extra definition means the BR118 can also be used for higher bass frequencies, leading to use in a wider variety of configurations. With the addition of the M20 mounting pole, the F1201 and BR118 combine to create a flexible small PA system, which carries the high-fidelity performance characteristics associated with Funktion-One’s larger format speakers. Tony Andrews explains: “By careful driver and enclosure tuning we have achieved the speed and immediacy of our larger products in our near to mediumfield smaller range. This means excellent coherence and smooth response for good speech intelligibility and tight, 32
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FACTS & FIGURES F1201 Trapezoidal cabinet Custom designed 12in driver and a 1in compression driver Frequency range: 60Hz to 20kHz Integral M10 fly points and a rotatable 90° x 45° HF horn for ease of installation BR118 Deep near-field bass from a compact enclosure Well matched with products including F88, F1201, R1, etc 1 x 18in driver Frequency response: 37Hz to 300Hz Weight: 38kg
meaningful low frequencies, making the combination suitable for all applications and genres of music. “Our customers have found the F1201 and BR118 to be a highly portable and effective combination for use in small music venues and outdoor parties. They have also found the F1201 to be a perfect standalone cabinet for use in speech-oriented applications such as theatre and AV.” The Stove – an arts hub in Dumfries and Galloway, Scotland – recently acquired two F1201s and a pair of BR118s. The venue is used for a wide variety of events, including exhibitions, live gigs and spoken word events. Colin Tennant, a member of The Stove’s curatorial team, explains why they chose the system: “We wanted a sound system that could be used around the building, even outside the front of the building, so the FunktionOne set-up is brilliant for that. But we wanted more than flexibility. We wanted a level of sound quality that matches our ambitions for the building, so that every event that requires amplified sound will have the clarity and impact that people associate with Funktion-One.”
CONTACTS
Address: Funktion-One, Hoyle, Horsham Road Beare Green, Dorking, Surrey RH5 4PS, UK Email: mail@funktion-one.com Web: www.funktion-one.com
International
SMALL-FORMAT PA
www.audiomediainternational.com
PIONEER
XY SERIES
FACTS & FIGURES XY-101 & XY-152: 10in and 15in two-way, full-range PA speakers XY-81: Two-way full-range speaker with 8in LF driver and 1in compression driver XY-122: 12in two-way full-range speaker system
Robust, compact and suitable for a wide range of applications – we take a closer look at the XY Series.
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ioneer’s XY Series is a range of robust, high power, compact speakers designed for multiple applications including DJ booth monitors and PA speakers in bars, restaurants and mid-sized rooms. The recently expanded range offers the widest possible scope for venues to mix and match to meet their specific requirements and enjoy the clear and immersive sound on offer. There are multiple installation options, with brackets, pole mounts, top hats, flying cradles and other accessories to enable a quick set-up. The XY Series is supported by Powersoft K-Series DSP amplifiers, which are supplied with pre-set data to maximise the performance of the speakers. The XY lineup comprises wide-dispersion (90º x 60º) two-way full-range speakers with either 8in, 10in, 12in or 15in LF drivers, all with a rotatable horn for horizontal installation (XY-81, XY-101, XY-122, XY-152). These speakers are complemented by single or dual bass reflex subwoofers with either 15in or 18in drivers (XY-115S, XY-118S, XY-215S, XY-218S) capable of reproducing rich deep bass without distortion. The whole range is available in black or white with optional gold grilles.
XY-115S: Discreet, 15in bass reflex subwoofer for smaller spaces XY-118S: 18in bass reflex subwoofer XY-215S: Quasi-bandpass subwoofer with 2 x 15in drivers XY-218S: A powerful bass reflex subwoofer with twin 18in drivers, ideal for large rooms and temporary outdoor installations
CONTACTS
Email: contact@pioneerproaudio.com Web: www.pioneerproaudio.com
May 2015
33
TECHNOLOGY REVIEW
MERGING TECHNOLOGIES HAPI
Key Features NETWORKED AUDIO INTERFACE Stephen Bennett puts the Swiss manufacturer’s latest piece of Ravenna-compatible hardware through its paces.
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o, this isn’t a review of a piece of equipment that will improve your mood – Hapi is actually the latest audio interface from Merging Technologies. Continuing to name its interfaces after Egyptian deities, the new boy on Merging’s block gets its moniker from one of the four sons of Horus, which is another Merging product, so surely we can expect at least three more siblings spawned from the company’s flagship interface? The naming of Hapi makes a lot of sense really as it shares many of the features of its parent interface, albeit in a smaller, less expandable package. Hapi is unusual in the world of rack mount interfaces, as it utilises Ethernetbased connectivity to communicate 34
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with the computer, shifting audio via a single Cat cable. Hapi talks with the outside world via Ravenna, an AES67compatible layer 3 IP-based audio networking protocol, which provides up to 88 input and 90 output channels over a single Cat5e or Cat6 cable to any other Ravenna-based devices on the network, including the Horus interface. Hapi will happily – see what I did there? – play ball with any existing computer networks. The use of Ravenna enables the interface to work at extremely low latencies and jitter rates and synchronisation information – including Linear Time code (LTC), Video Reference and Wordclock – is also handled alongside Ravenna. Hapi is a 19in one-unit-high modular multichannel, networked audio interface and A/D converter. It is beautifully finished in ivory white and, as you may
expect from a device built by the Swiss, it feels solidly constructed with the minimalist front panel featuring just a data entry knob, an OLED screen a headphone jack and a backlit Merging logo that doubles as an on/off switch. Merging boasts that the Hapi is extremely frugal in power usage, going so far as to provide display of power consumption, and is happy to print detailed technical specifications of the unit in the user manual, right down to the gain behaviour of the direct output section. The rear panel of the basic Hapi features a D–connector for AES channels 1-8, an SPDIF/ADAT connection, BNC connectors for Wordclock, a D-connector for Sync purposes and two Ethernet sockets for connection to the computer and other Hapi or Horus interfaces.
Eight AES/EBU I/O (on DSub 25) Wordclock Input and Output (on BNC) Two Option slots for AD8D/AD8DP, DA8/ DA8P and MADI option card AC and DC power connections Web-based control interface Detachable rack ears RRP: £1,574 for base unit (exc VAT) www.merging.com The two slots for the optional input and output modules sit next to the IEC mains connector – the Hapi’s switchable power supply will work at 240V or 110V AC and there’s also a DC power supply option that can be used to directly power the unit on location or act as a backup supply in situations where you think you may lose mains power. All the inputs and outputs on Hapi are handled by D-subs, which makes sense as most professional installations use the Tascam connectors these days, although Merging can provide you with breakout cables if you’re so uncool as to be still using XLR and TRS connectors.
TECHNOLOGY REVIEW line cards enable the user to split off the analogue signal immediately after the mic preamplifier, which can then be used to feed monitors or another recording device. The A-D converters on the premium cards feature extended headroom for DSD work and the stated technical specification of both types of standard module is very impressive and comparable with the best audio interfaces on the market today. This modular flexibility means that you can specify precisely what connectivity you require from your interface and, as Hapi is perfectly capable of working as a standalone unit, I can see it getting a lot of love from those of us who create permanent audio installations. The MADI module provides
indication of actual input and output signal levels. The Preamp menu displays an on screen mixer, showing the available analogue inputs. Here, users can switch between mic or line level inputs, engage the +48V phantom power and swap the phase of the channel. Pad and 80Hz high pass filter buttons sit alongside a low impedance input setting, while the long throw input level faders can easily be grouped together. The I/O and Sync menu allows the user to adjust various synchronisation settings, while the Headphone menu has the routing options for the front panel headphone jack. The Setup menu opens a page where you can view and control most of the Hapi’s other parameters, including system and Wordclock output sample
powered monitors and got to work right away. Setup of the unit went swimmingly and my first port of call was to listen to some familiar mixes through the ATCs. It was obvious from the word go that the D/A converters and analogue electronics on the Hapi are of an extremely high quality and easily up to par with my current beau, the two-channel Metric Halo ULN-2 interface, while also being audibly superior to the Apogee Duet II, RME and Digidesign 192 I/O interfaces I compared it with. The microphone inputs are extraordinarily clean with bags of headroom and again, are similar in quality to those in my Metric Halo unit, which are themselves considered exceptional, so the Merging unit is impressive in this respect. I used the Hapi to record
users with up to 64 channels of I/O, and both coaxial and optical interfaces are available simultaneously and configurable in blocks of eight channels – the one limitation being that Hapi can only handle one MADI module at a time. Modules can be changed by “The Hapi is a compact, flexible the user so it’s conceivable that and relatively inexpensive you could purchase more than high-quality audio interface two and swap them out for that will be equally at home in specific tasks. the studio, on the road or in a The supplied software fixed installation setting.” installs the Ravenna control Stephen Bennett and MTDiscovery applications and the ASIO and Core Audio drivers. MTDiscovery displays all of the Hapi is a modular unit, working happily Ravenna-compatible devices that you – look, I’ll stop now – with Apple’s Core have connected, including Hapi and Audio and Steinberg’s ASIO 2.2 audio Horus, ASIO, Core Audio and MassCore, protocols. The interface can handle two the real-time engine used by Merging’s modules at a time and are the same ones Pyramix DAW. The Merging/Ravenna that can be fitted to the Horus, providing Easy Connect application allows you to quite a few operational options. The connect and view the various Ravenna base unit provides eight channels of AES streams on your network. digital I/O and eight channels of ADAT Once your physical connections have interfacing, which can then be expanded been set up, Hapi can be controlled from by filling the two module slots. Most the front panel display via a ‘push and modules are available in ‘standard’ and turn’ menu system or a web browser on ‘premium’ versions – the former are the computer. The browser interface able to work at sample rates of up to presents a graphic view of the interface 192kHz, while the latter are DSD/DXD on the screen, with various function compatible. Merging produces modules menus located under the Hapi logo. The that offer eight-channel microphone and Meters menu provides an onscreen line level input cards with direct outputs display of the audio coming into, and (AD8D and AD8DP) and eight-channel out of Hapi but the latency in the display line output cards in both versions (DA8 makes this feature only useful for ‘signal and DA8P), a standard eight-channel present or absent’ duties – although microphone/line digital to analogue it does enable you to easily see if the (D-A) card (ADA8) and a MADI I/O audio is coming in too hot. The meters on card. The direct outputs on the mic/ the unit itself provide a more accurate
rates and a section where the user can override the default Network settings to let Hapi play nicely with existing infrastructure. The Presets sub-menu enables the user to store and load up to five preset banks of Hapi configurations at one time. The Routing menu requires some extra explanation as Ravenna offers some quite complex routing options – Merging has applied a ‘route to’ instead of the more common ‘route from’ philosophy in its connection methodology and the company says that, after an initial learning period, this presents a more efficient process for connecting Ravenna inputs and outputs – and they are probably right! You choose where you want your audio to go, decide what input you want to feed to that output, and the software offers up all the possible connectivity options. To avoid confusion – they hope – Merging has limited the routing options to blocks of eight channels so you can, for example, route eight AES input channels to eight MADI outputs. It sounds a tad complicated when written down, but it’s much simpler to grasp in operation. The front panel of the Hapi can also be used to set these options, so you don’t need to use a computer to adjust the settings. The OLED screen is clear and the menu system relatively simple to use.
acoustic guitars, a string quartet and drums using a brace of AKG 414 XLS and Neumann KM84 microphones. I have a lot of experience of recording similar instruments using these microphones and the Hapi did not disappoint. Merging’s interface also passed my ‘if it’s a good preamp it’ll make the Shure SM57 microphone shine’ test on snare drum and Celestion-filled guitar cab, which is not something that all microphone preamplifiers that I’ve used can do. The Hapi is a compact, flexible and relatively inexpensive high-quality audio interface that will be equally at home in the studio, on the road or in a fixed installation. The Ravenna audio network means that latencies are ultra low, and the flexibility of the connectivity with other devices using the same protocol, even over standard networks infrastructure, means complex systems are easy to manage. Merging has taken the technology it premiered in its Horus interface and created an impressive offspring that any self-respecting audio engineer would be Hapi to own.
In Use The Hapi that was supplied for the review by UK distributor emerging was loaded with both standard and premium mic/line cards, so I connected the monitor outputs to my ATC SCM50A
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.
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TECHNOLOGY REVIEW
SSL LIVE L300 DIGITAL CONSOLE It may be a more compact alternative to the L500, but has downsizing taken away any of the original’s appeal? Andy Coules investigates…
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hen Solid State Logic announced its entry into the live sound arena a lot of people were understandably excited about the prospect (myself included). SSL is well known for providing high-quality pro-audio gear and has an impressive history of innovation, particular in the area of mixing desks. The manufacturer’s first live digital console, the L500, was well received, and now along comes a (slightly) smaller brother, the L300, complete with a version 2.5 firmware, which delivers better FX optimisation and 27 new features and updates to the platform. V3 of the firmware was released at Prolight + Sound last month. Operationally the two consoles are very similar, they run the same software and the control surfaces have the same five basic components: two fader tiles with 12 channels each (although the L500 has an optional third fader tile), a 19in multi-touch screen, a channel control tile and a master section (which houses focus and master faders, shortcut keys, numeric entry keypad, solo and scene recall controls). The layout differs from the L500 insofar as the two fader tiles are on the horizontal surface while everything else is on the angled surface. The main differences between the two models are physical size (the L300 is slightly smaller), available channel paths (192 on the L300 whereas the L500 Plus has 256), screen brightness (the L300 is capable of producing 600 Nits whereas the L500 does 1,500) and available I/O. The L300 comes with 16 local mic/line XLR inputs, 16 XLR line outs, four pairs of AES/EBU, eight coax/optical MADI 36
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Key Features ports (grouped in redundant pairs), a dedicated MADI FX loop and an option to add a redundant pair of Blacklight connectors. This means that it can handle, with the appropriate stage boxes, up to 568 inputs and 568 outputs (compared to the 962 ins/outs of the L500). If the optional Dante interface is added the maximum inputs and outputs for the L500 is 962 + 32 and the L300 is 568 + 32. The I/O architecture is based on ‘paths’ which can be assigned as channels, stem groups, auxes or masters and can be configured as mono, stereo, LCR, 4.0 or 5.1 where a mono channel uses one path, a stereo two and an LCR three, etc. The L300 has 192 mix paths (at 96kHz), 144 of which have full processing and 48 are dry. A full processing path has nine blocks: filter, EQ, gate, compressor, insert A, insert B, fader, all pass filter and delay whereas a dry path just has three: insert A, insert B and fader (although EQ and dynamics can be added as inserts). One interesting feature is that the order of these processing blocks can be changed on a per channel basis by muting the channel and dragging and dropping the blocks. Each fader tile has five scrollable layers of five banks and each fader has a detailed level meter next to it as well as gate and compressor state indicators. There is also a query button (denoted by a capital Q) which, when pressed, shows what is routed to and from the fader.
In use Basic operation revolves around the central touchscreen with logical touch controls as well as double taps and holds. The touchscreen behaves a lot like an app which encourages you to grab, pinch and manipulate things. The channel control tile features a 5.7in touchscreen surrounded by 15 knobs, which gives quick access to key channel parameters such as EQ, gate, compressor, all pass filter, etc. It includes all of the usual features you would expect on a console but there are a number of interesting ones too. The high pass and low pass filters have three choices of slope values: 24, 18 and 12dB per octave which is a nice touch, the 12dB slope is perfect for negating the proximity effect. The compressor has a nifty ‘tube’ button which kicks in an analogue tube emulation that adds noticeable warmth to the signal – it works well on bass guitar and kick drum and I can see it being one of those things you end up adding to everything. Then there’s the all pass filter, something I’ve never seen implemented on a live sound console. An all pass filter passes all frequencies equally in gain but changes their phase relationship to compensate for any undesired phase shifts that arise in the signal path. It’s a subtle tool but it’s inclusion on every full path signifies a key commitment to sonic precision. There are a couple of minor things that bugged me, the shortcut keys are not programmable, the top six are dedicated
Maximum I/O – 568 in/out (600 with Dante) 192 paths (144 full, 48 dry) 96 kHz or 48 kHz sample rate 48 FX slots 36.3in width, weighs 81kg RRP: From £36,000 www.solid-state-logic.com to the tap tempo of the six FX banks and the bottom four select different screens (Ed – they are programmable in V3). Also, the on-screen knobs have no level indication (Ed – a level meter has now been added to the small OLED screen within each channel strip). SSL’s entry into the digital live console market is well timed. The company has had time to check out the competition and see what works and what doesn’t. The fact that the amount of DSP now available makes it possible to satisfy the most rigorous demands of live sound engineers can’t be a coincidence. The combination of these two factors have enabled SSL to deliver a relatively mature console with a host of excellent features that encourage a fluid and intuitive workflow with the kind of sound quality you would expect from SSL.
The Reviewer Andy Coules is a sound engineer and audio educator who has toured the world with a diverse array of acts in a wide range of genres. www.andycoules.co.uk
TECHNOLOGY REVIEW
WAVES BUTCH VIG VOCALS PLUG-IN
Simon Allen sees whether the latest addition to the company’s Signature Series is more than just another product backed by a big name.
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here is no doubt about it, the graphical user interface (GUI) of this plug-in caught my eye and, being a Waves plug-in, I couldn’t wait to try it. It’s such an unusual design that only a wellrespected plug-in developer, alongside a well-known manufacturer, could release such a thing without facing potentially unfounded criticism.
The Concept Like other plug-ins from the Waves Signature Series, this one is designed to deliver that Butch Vig vocal sound, drawing on Waves’ classic gear algorithms and extensive knowledge of a modern pro-audio plug-in. For me, having Butch Vig collaborate with Waves is very exciting and I expected this to be something special. The purpose of this plug-in is to deliver a fast and creative way to achieve a legendary vocal sound. Yes, it does include tools we all know and use, but the key point is the way these common tools are used and implemented in the vocal processing chain. As a large degree of the scientific engineering work is already taken care of, the user experience should be a creative one. Therefore, I feel the artistic GUI is a great move. I love the artwork, but whether or not it’s to your taste, it does remind you that you’re not pulling a channel strip together that ‘might’ suit the track, but simply deciding upon the creative direction for your mix.
Overview Once you’ve got your head around the clockwork-like design, the controls are laid out in a familiar way and there’s plenty to discover. The level meter dials can be set to show input, output, gain reduction or de-essing and there’s the addition of the commonly found sensitivity LED as on many Waves plugins. The sensitivity is more relevant in
many ways than the input metering, as this gives you an indication of how the plug-in will react upon your source, as several thresholds are hidden from the user interface. There’s an input control on the left and an output on the right, which allows you to compensate for gain changes and clearer AB comparisons. Moving down, we find the EQ section. This consists of low-cut and high-cut filters, with frequency value read-offs, which allow you to be quite accurate. Staying with EQ, there are three controls labelled Low, Presence and Air. These are simple boost or cutting controls with frequency and filter shapes predetermined. Lastly in the EQ section there is a ‘mid-dip’, which actually provides you with a sweepable -6.5dB cut in the mid frequency range. Then your eyes are drawn to the dynamics section, which comprises a de-esser and compressor. You operate the de-esser by dialing in the amount of de-easing you want. The frequency of the de-esser is set at 6kHz. The compressor is again just the one dial, but as you increase it, not only is the threshold changing, but so are the other functions and characteristics. This means there are a number of different compression flavours behind this one control. The last two tricks Butch has built into his plug-in are a ‘Focus’ control – in effect a mid-range compression at either 1kHz or 2kHz – and a saturation section. Like other features of this plug-in, the settings for the ‘Focus’ control are behind the scenes; the user is simply left turning it on or off at either frequency position. The saturation section offers two types: tube or solid state. There are also hi-cut and low-cut filters here, which only have an effect on the parallel saturated sound that you are blending against the dry signal. A really significant part of the design that Butch focused on is the
processing order of all these elements and their ‘mix’ amount.
Key Features
In the studio My first reaction when I tried out the plug-in on a selection of vocals was the sensitivity of the parameters and how they change the sound. I was expecting massive alterations to the sound, and although these are possible if the controls are set to extreme, they are in fact quite subtle and accurate. The compressor has a lot to offer, with what feels like a change in ratio and style of compressor as you move up through the range. My favourite feature though has to be the hi-cut and low-cut filters on the saturation section. Driving the saturation quite hard, and only effecting the mid-range of the voice allows it to sit in the track prominently, with the highs left unprocessed for clarity. The scope to create different vocal sounds with this feature alone is huge. The more you use the plug-in, the more you realise how much time it has taken to develop. It is sonically clear that the quality of the processing is some of Waves’ best, and I understand Butch has worked very closely on the features. This is a very serious tool, which shouldn’t be under-estimated. I love the creative approach, which helps you sit the vocal into the track, especially the ‘Focus’ feature. As the plug-in offers a multitude of tools and possible sounds that are available in one window, it can also inspire you to
Butch Vig’s vocal sound via a single plug-in Three-band EQ with ‘distinct frequency character’ LoCut, HiCut and MidDip filters Compressor and de-esser Focus control for the 1kHz and 2kHz range MSRP: $150 www.waves.com try other creative directions for your mix that you may not have otherwise thought about. I found this helps speed up workflow during the mixing process, which is commonly where plug-ins like this have a hidden benefit, in what is currently such a time-conscious market. Don’t think for a moment, however, that this is a quick-fix tool for vocals. It is fast to use, but it also offers the chance for sensitive movements to reach some great results on almost any vocal, including rap.
The Reviewer Simon Allen is a freelance, internationally recognised sound engineer and pro-audio professional with over a decade of experience. Working mostly in music, his reputation as a mix engineer continues to reach new heights.
May 2015
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TECHNOLOGY REVIEW
SENNHEISER EW D1 WIRELESS SYSTEM
I
t was only after unpacking the new Sennheiser Evolution Wireless D1 digital radio mics that I realised how much Sennheiser radio kit I had. Two VHF diversity kits with mains powered receivers, three VHF location kits, an Evolution G2 personal system and ew 500 plug-on, oh and a 300 IEM transmitter. I guess all that is testimony to the success (and longevity) of Sennheiser’s radio product. So a new departure from Sennheiser in the world of wireless is always going to raise a lot of interest. And when that product carries the Evolution label it has a lot to live up to. There have always been cheaper alternatives to Evolution kit but I’ve never found any that were compelling purchases. Of course if you have more money, Sennheiser has more expensive options. However, today we’re concerned with the Evolution D1 radio mics operating license-free in the 2.4GHz band. Out of the box the finish is very good and all the gear (I had a handheld and a beltpack system) felt good in the hand. The mains powered receivers are not housed in quite the bulletproof sleeves of the old G series I have, but they are well finished and offer some significant technological advances. First, 2.4GHz is license-free. Hooray, I hear you cry. There is a word of caution of course – 2.4GHz is home to the WiFi party and the Bluetooth bonanza so it’s a busy old bit of spectrum. Don’t sit your ew D1 receiver on top of your wireless router – that is a bad idea. In fact, keep your D1 receivers at least five metres from your WiFi transmitters. On the other hand I tried out the ew D1 on gigs in suburban houses of worship – pretty much bang on the target market – and despite the plethora of networks flying round I had no problems at all. The digital Evolution tech involves time and frequency diversity and redundancy to protect against burst disruption. Sennheiser advises that 38
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When a new Sennheiser carries the Evolution label it has “a lot to live up to”, our reviewer Alistair McGhee says. So is the new D1 system worthy of the title?
under ideal conditions you can run 15 systems, but I reckon you’re ahead of the game at anything over seven – more than you can run license-free without intermod in channel 70. Second, tuning is a thing of the past; pairing is the new tuning. All parts of the system have a ‘Pair’ button. A long press on Pair at the TX end brings up ‘Press Pair on RX’ and indeed a long press on Pair at the receiver links you up. Once a pairing is established a short press on the pair button at either end flashes the LEDs or both RX and TX to show you which channel is which. Nice touch. The handheld and the lavalier have small squinty screens – I guess there’s no need now to show frequencies and other tuning info. However it will show the name of the receiver the mic is paired with. On the receiver you get a lovely OLED screen, which will even display QR codes linking you to the Sennheiser help site! The SK-D1 body pack is a beauty – really nice ‘Made in Germany’ finish with the battery sled – which has come down from the higher range – in which both AA batteries fit in the same direction. What an idea . The mic input on the SK-D1 is a locking 3.5mm jack and the supplied mic is the decent-enough ME 2-2. Sadly my Sanken Cos 11 is wired for Micron and my AKGs to XLR, which is a shame as it would have been nice to try a mic upgrade. One thing that will strike you straight away is the stubby little aerial – higher frequency = shorter waves = shorter aerial. This is great, but there is a price to pay. Shorter waves = more directional and less bendiness round corners. And due to EU restrictions the 2.4GHz system is restricted to 10mW and therefore range is lower than its UHF brethren. I tested the D1 against my older Evo kit and the analogue does indeed best the D1 for range. Sennheiser quote 60-odd metres for D1 against 100 metres for the G3, best case and line of sight.
Round the back of the receiver, in addition to the XLR and jack outputs you also get a network port, which allows firmware updating and wireless control of the receiver. There’s an app for that – well there will be soon, Sennheiser says June. Of course you want to be careful with the amount of 2.4GHz you are chucking around – set your network to 5GHz or remember the 5m rule! The great thing about wireless control is it helps overcome the reduced range of 2.4GHz systems like the D1, when you might have to site the receivers nearer your talent than you would otherwise. I’m hoping for a windows or OSX version because I want wired control of my wireless! Which brings me to one other shortcoming of digital wireless – latency. The ew D1 system has about 4ms of latency, which is not a problem for performers but will be a problem if you want to combine analogue and digital mics on the same stage. The D1 system uses the AptX audio codec and in my tests the sound was good. The handheld capsules are industry standard and off-axis rejection is comparable to an SM58. The closest match I had to the e835 capsule in the handheld was a Sennheiser MD46. Running that through the ew500 plugon the sound quality was very close, and even down a cable I wasn’t getting significantly better quality than the ewD system on a rock vocal. So where does this all leave us? I think there are three points to ponder: do you
Key Features Operates in license-free 2.4GHz band Automatic frequency management ‘Intelligent’ channel back-up enables coexistence with WiFi and Bluetooth Robust metal housing iOS and Android app to control and monitor the system remotely RRP: ew D1 sets from £519 www.sennheiser.com have a hostile 2.4GHz environment, is maximum range your biggest concern and do you have to run a mixed analogue and digital environment? On the sunny side the ew D1 is really well made, highly featured, has long battery life and sounds good. It is relatively inexpensive and with the promised network control of the receiver, very flexible. I think the 2.4GHz R-Evolution has landed.
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
JBL LSR308 STUDIO MONITORS
Featuring a ‘revolutionary’ Image Control Waveguide originally implemented in the high-end M2 model, these new 3 Series speakers promise power and performance in an affordable, compact package. Strother Bullins takes them for a test run.
I
n January 2013, JBL Professional unveiled its large reference studio monitor, the M2, to intrigued proaudio types at the Winter NAMM Show. In private listening sessions, the M2 certainly was impressive to me, due to a number of new JBL technologies, one of which, its patented Image Control waveguide, provided truly superb imaging plus flat and even frequency response across a wide listening area, creating a very pleasing, smooth listening experience. On the opposite end of the price spectrum, JBL soon after unveiled its 3 Series of powered nearfield studio monitors with a very familiarlooking waveguide for its soft dome tweeter – one that, to my eyes, was clearly influenced by the M2’s Image Control. I wondered, would the 3 Series be anything like its large and pricey ancestor, especially considering the largest 3 Series monitor, the LSR308, costs well over $9,000 less than the M2? I received a pair of LSR308 monitors ($249 street, each) and their accompanying subwoofer, the powered LSR310S ($399 street), for review, and what I discovered was surprising. JBL’s 3 Series powered monitors are available in two sizes: the aforementioned LSR308 with an 8in woofer and the LSR305 ($149 street, each) with a 5in woofer (a 41W/41W Class D biamped design). The LSR308 features biamped Class D power (56W/56W for LF/HF, respectively), a 37Hz-24kHz frequency range, 112dB SPL (C-weighted) maximum SPL, and a ported design. Inputs include XLR and quarter-inch TRS with switchable +4dBu/-10dBV input sensitivity, and the cabinet weighs in at a comparably light 19lbs. Rear panel adjustments include three-position LF and HF Trim parameters, each with -2dB, O, and +2dB settings. The front of the speaker is made of attractive polypropylene while the rear-ported 40
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more in music ranging from EDM to even orchestral/chamber music. I’ve auditioned many solid nearfield studio monitors over the past 15 years – most in the 8in woofer category – used within my own workspace. That said, the 3 Series truly impressed me. I could use them as my main monitors indefinitely. Knowing their low price points before starting this review, I suspected I might hear things I wouldn’t like, based on its composition (of largely materials such as MDF, plastics and low-cost Class D amplifiers). But I discovered that the design trumps materials. To me, the Image Control waveguide obviously makes the LSR308 a low-cost marvel of detailed imaging and controlled frequency response across horizontal and vertical planes, as the tweeter itself seems to be a rather common soft dome. The fact that a recordist can own a pair of large JBL near-fields, clearly borne from M2 R&D and this good sounding, for under $500 per pair makes the LSR308 perhaps the best bargain available in powered studio monitors today. cabinet is constructed with a lighter weight MDF. The LSR310S is a down-firing powered subwoofer featuring a heavy-duty 10in driver, 200W of Class D amplification, and a low frequency range of 27Hz with a maximum SPL of 113dB. It features a compact design (less than 18in x 15in x 16in and 35lbs in weight), the patented rectangular Slip Stream double-flared front panel port “for accurate bass response at low playback levels”, and JBL’s proprietary XLF Extended Low Frequency – a 10dB boost at 60Hz engaged via switch. XLF is essentially a custom frequency response to simulate club/large PA playback – especially helpful in modern-leaning music productions. Overall, 3 Series build quality is impressive and attractive even if a bit spartan, yet refined. In a business
overly aware of brand names, these monitors also have the value-added benefit of being JBL – a name associated with professional studio monitors for decades. The LSR308 pair satisfied my low-end needs by themselves in most applications. Yet, while I don’t typically use a sub during production, the LSR310S came in handy, like when listening to an ultra-low frequency sound source in a mix – a bass ‘drop’ for emphasis. The sub allowed me to accurately gauge where these extreme low-end elements sat in the mix. But largely, I didn’t notice the sub, and that’s to its credit; the transition between the low-end from the LSR308 pair and LSR310S was seamless. I expect that the sub would be more crucial/useful when paired with the smaller LSR305s, or when working less in pop/rock and
Key Features ‘Revolutionary’ Image Control Waveguide Slip Stream port design for deep bass response -10dB/+4dB sensitivity switch LF Trim and HF Trim controls for both the low and high frequencies Balanced XLR and 6mm TRS inputs RRP: $249 www.jblpro.com
The Reviewer Strother Bullins is reviews editor for NewBay Media’s AV/Pro Audio Group. sbullins@nbmedia.com www.prosoundnetwork.com
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ttracting over 250 MI retailers and pro audio resellers from around the UK and organised by NewBay Media, MIRC 2015 is solely targeted at maximising sales. The morning conference programme is designed to increase business and margin, with an afternoon expo featuring a selection of pre-built stands and private meeting rooms geared towards meetings and conducting business.
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INTERVIEW
www.audiomediainternational.com
FRANKLY SPEAKING
A legend in sound design whose work has shaped hundreds of movies, adverts and video games, Frank Serafine has decided to share his wisdom via a 33-date Sound Advice Tour of the US and Canada. Matt Fellows chatted with him about the past, the advance of technology and the importance of education. because they understand not only the creative process that goes into doing sound design but also the experimental techniques that it takes. Sound design is very much like orchestrating a big orchestra; there’s a lot of layers, a lot of emotion involved – it’s composing with sound instead of musical instruments. We understand that you’re often cited as a driving force behind several
So tell us a bit about the Sound Advice Tour. How did you get involved? I worked with Zoom on their tours; they just called me up, and I was very interested because I’d been teaching at UCLA Extension for 13 years and the LA Film School for a year and a half. I love educating, and the opportunity just came along and I took it. Do you think it’s now more important than ever for professionals in your position to pass on knowledge? Of course, because there’s not a lot of people from the whole Hollywood system who have learned the new techniques. A lot of the greatest people who’ve done well in the industry in Hollywood end up teaching at UCLA [University of California, Los Angeles] or USC [University of Southern California]. I’ve worked in the industry for close to 40 years now and I learned from all those guys from Disney, so I’m passing on what I’ve learned from my mentors over the years. Now things are getting very complex, like with 3D audio, which never existed back then. People have tried to convince me of 3D audio over the years, but it never clicked until now. 42
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What’s your opinion on 3D audio then? Where do you see it going? What’s so great about it is that those 48 speakers in a Dolby theatre translate down to your headphones, so what I predict is that people are going to start watching 3D audio films on their iPhones and iPads. And what about the home cinema market? Do you see it taking off there? That’s where it’s going. You don’t need 48 speakers – for home theatres it’s 11. The home theatre market is going in that direction and I’m sure in the next five years it will be a very big part of home cinema. You’re an accomplished composer and musician. How does that relate to your sound design project? All the greatest sound designers in the film industry that I’ve loved over the years, like Timmy McDonald from Disney who did all the original cartoons, Ben Burtt who did all the Star Wars films – these are all musicians who made their way into sound design. Some of the greatest sound designers I know have come from being musicians
software and hardware innovations? My first movie was Star Trek: The Motion Picture. I was a young kid in my early 20s and I was able to make sounds on a synthesizer that people who had been in the industry for many years didn’t know anything about, because they were all cutting on 35mm. So I started out as a special sound designer/effects creator, and as time went on I didn’t want to cut on traditional 35mm magnetic film because it was like trying to make a major motion picture soundtrack on a sewing machine. So I kind of brought in a lot of this technology. I was the first to lock in a videotape recorder with a multitrack machine on Tron, and we used Pro Tools on Hunt for Red October, which won the Oscar for Best Sound Effects Editing. Lawnmower Man was the first film that was cut 100% digitally. We were still using analogue tape recorders but we were way beyond anybody. We kind of revolutionised the way sound editing was done in the film industry. What are your thoughts on the evolution of professional audio technology? How has it changed the way you work? What it did was it allowed me to become more of a supervisor because I didn’t have to spend the time; I had more control over it because I could actually cut the dialogue myself, I could do a lot of the editing on my own – it’s allowed me to do a ton more work for the same amount of money. The profit margins have also gone down because there’s a lot less labour, less lab room
expense, machine room, transferring – you don’t even need the real estate rooms anymore, you can cut out of your bedroom. The industry has changed a lot as well as the technology, and the way things are going I think someday we’re going to be cutting on our iWatches. How has the role of the sound designer changed over the years? I’m finding that you can’t just be a sound designer anymore – you’ve got to be a sound designer, a cinematographer, a Foley walker. It used to be very compartmentalised back in the day. Now I can run a camera, I can edit – who would have ever thought I would be an editor? That’s what’s happening. Of the hundreds of projects that you’ve worked on, do you have any favourites? How could I not? I was a sound virgin in films, and Star Trek was my first romance. I loved Tron, and the experimental work that went into Hunt for Red October, and it won an Oscar. And then I’ve done other beautiful musical projects, like I produced Ravi Shankar’s fifth album Tana Mana and I got to work with George Harrison. And then I worked with Peter Gabriel on his Kiss That Frog project and went on the road and toured with him live. Another fun one is that I got to work with Matt and Trey from South Park on Orgazmo. I had so much fun working on that. They’re all different, it’s like saying ‘Who’s your favourite child?’ You love them all. Have you got any projects lined up in the near future? There’s quite a few things that are in the pipeline, but one of the biggest things is that I’m building my ultimate studio – a little Skywalker ranch up in the hills above Los Angeles. It’s going to be a self-contained environment where you can just get out of LA, come to this little capsule, work on your stuff and not be distracted by the city. soundadvice.mzed.com
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