AMI March 2015

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

March 2015

STRIKING A BALANCE The popularity of practical, costeffective touring kit continues to rise, but has this made sound quality less of a priority for manufacturers? p26

STUDIO PROFILE

REVIEW

INTERVIEW

Step inside Santa Monica’s Windmark Recording p22

Audio-Technica’s ATH-M70x p36

Ben Wilkins discusses his work on the Oscar-winning Whiplash p46

WELCOME

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EDITOR Adam Savage asavage@nbmedia.com

Experts in the issue

FORM OVER FUNCTION?

MANAGING EDITOR Jo Ruddock jruddock@nbmedia.com STAFF WRITER Matt Fellows mfellows@nbmedia.com COMMERCIAL DIRECTOR Darrell Carter dcarter@nbmedia.com ACCOUNT MANAGER Karma Bertelsen kbertelsen@nbmedia.com HEAD OF DESIGN Jat Garcha jgarcha@nbmedia.com PRODUCTION EXECUTIVE Jason Dowie jdowie@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.

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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: Nexo/Ben Kriemann

Rob Bridgett is audio director and producer at Clockwork Fox Studios in Canada.

Inspired by the work of the Sony ASWG group and the lectures of sound advocate Julian Treasure, the Clockwork Fox team have been working on developing adaptive loudness switching and runtime LRA attenuation for child-friendly mobile ed tech products for the last two years.

Stephen Bartlett Currently based at Wisseloord Studios in the Netherlands, producer/ engineer Stephen Bartlett is the former owner of The Docking Station in Australia, where he won various industry awards.

Since moving to Europe, Bartlett has gained four top 10 records, a top 10 single and a platinum certification.

Ben Hammond is a FOH engineer and tour manager for bands such as Deaf Havana, Saxon and Young Guns, and owns Reel Recording Studio in Yorkshire, UK.

David Davies has been writing about professional AV and broadcast for 15 years. He is currently managing editor of Sports Video Group Europe and has been a member of the ISE Daily, IBC Daily and AES Daily teams.

I

t wasn’t too long ago that we journos were regularly bombarded with news regarding the launch of the “best sounding mixing desk ever made”, the “most powerful loudspeaker system on the market” or the “microphone that records with such clarity that you’ll weep with joy when you hear it”. I might have made that last one up, but that’s the sort of thing we’d hear on a regular basis.

Although we’re still not free from exaggerations such as these, we’re now far more likely to hear about the release of “the most lightweight, compact console currently available” or “the line array that’s so easy to assemble your head will explode with astonishment”. OK, I twisted the truth again slightly there, but hopefully you see what I’m getting at.

Practicality now appears to be the primary goal for an increasing number of companies when it comes to new product development, but is the progress of better sound quality suffering a result? Is ease-of-use being prioritised over pristine

audio? Of course, many manufacturers will argue that they offer both in equal parts – and I’m inclined to agree with some of them – and yet we think it’s certainly worth pondering the effects of this recent trend. That’s just what we’ve done in this very issue, so make sure you turn to page 26 for more on this topic, where you will also find an exclusive first look at the new Vero system from Funktion-One. We’re also delighted to be able to offer an interview with Ben Wilkins, a member of the sound mixing trio that in the last month has picked up both the Sound Mixing Oscar and BAFTA for Best Sound. Wilkins tells us how he was given just 10 days to carry out his work on the acclaimed drama

Whiplash, but also how incredibly tight schedules like these are more common than you might think.

And with GDC 2015 – that’s the Game Developers Conference for those of you who don’t know – coming up this month, we’ve also been chatting to the audio team at Creative Assembly about the sound of Alien: Isolation, and our game audio readership will surely be pleased to see another Opinion from our new Canadian contributor Rob Bridgett on page 17.

Finally, if you’re looking for a new studio microphone, don’t make your purchase until you’ve perused our Tech Focus on page 32 and the preceding piece from producer and engineer Stephen Bartlett.

So plenty for all of you to get stuck into this month!

Adam Savage Editor Audio Media International

March 2015

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CONTENTS

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32

PRODUCT NEWS 6

Behringer announces XR18 availability

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Powersoft’s new Ottocanali amplifiers

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Sennheiser strengthens SpeechLine series

SHOW NEWS 10

BVE entertains at ExCeL

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OPINION Ben Hammond asks whether monitor engineers should be concerned about their future

FEATURES 22

STUDIO PROFILE Adam Savage finds out why California’s Windmark Recording is a studio complex like no other

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TOURING SOUND David Davies looks at whether the pressure to produce increasingly compact, cost-efficient gear is impacting on sound quality

PEOPLE Rob Bridgett debates the issue of mobile loudness

TECHNOLOGY 31

EXPERT WITNESS Stephen Bartlett offers advice on choosing the most suitable mic for your recording session

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FOCUS Studio Microphones: We highlight the mics you need on your kit list

Aston Fearon on the importance of accuracy when mixing a show 46

INTERVIEW Ben Wilkins talks Oscars success with Whiplash and reveals the secret weapon in his kitbag

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ALSO INSIDE 12

NEWS ANALYSIS: What happened when Roland Hemming demoed auto-mixing at ISE

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GEO FOCUS: SWITZERLAND Does the pro-audio market reflect the strength of the country’s overall economy? We find out.

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GAME AUDIO: Matt Fellows meets the Alien: Isolation sound designers from Sussex-based Creative Assembly

March 2015

REVIEWS 36 38 40 42 44

Audio-Technica ATH-M70x Sonarworks HD Reference Plug-in Foote Control Systems P4S ME Zoom H5 Cerwin-Vega P1000X P-Series

PRODUCT NEWS

BEHRINGER XR18 MIXER NOW AVAILABLE Behringer has revealed the availability of its new X AIR Series XR18 digital mixer.

The XR18 is a remote-controllable 18-input/12-bus portable mixing solution housed in a sturdy stage box form factor, and is the first mixing system of its kind designed for use with iPad and Android tablets, as well as smartphones and PC, Mac and Linux computers.

It is said to be suitable for both live and studio applications, with its 16 Midas-designed mic preamps, integrated tri-mode WiFi module for wireless control and fast, bidirectional USB audio/ MIDI interface.

Other features include four high-end X32 stereo FX processors, with close to 50 studio-grade X32 FX ‘Plug-ins’; X AIR Gain Sharing for hands-free auto-mixing; a high-res 100-band Real Time Analyser with full bar and spectrograph views, and Ultranet connectivity for Behringer’s P-16 personal monitoring system.

“For the first time in audio history, Behringer has created a digital mixer integrated into a stage box and combined with a wireless router, recording interface, personal monitoring system and even an auto-mixing function,” stated Music Group senior engineer of conception Jan Duwe.

“Competing products only work with iOS-compatible tablets, but the XR18... works beautifully with Android tablets and PC, Mac and Linux computers, as well as mini-notebooks – and yes, even iPads of all sizes. The XR18 not only changes the game, it changes the very playing field by accommodating all of the world’s most popular operating systems.”

The X AIR XR18 has a suggested United States minimum advertised price of $699. www.behringer.com

NUGEN LAUNCHES LOUDNESS TOOLKIT 2 Nugen Audio has released Loudness Toolkit 2, an update to the company’s loudness measurement and correction suite.

Developed in consultation with the post-production community, Loudness Toolkit 2 includes upgrades designed to streamline workflow for NLE and DAW users. Nugen has also made its DynApt dynamic adaptation technology available as an extension for LM-Correct, enabling streamlined LRA targeting.

As part of the release, the VisLM-H visual loudness meter has been enhanced with a Timecode and Loudness Profile Memory Function that tracks and retains the timecode and loudness profile; a Program Loudness Over-dub Mode, which memorises the program loudness profiles, integrating them automatically to provide accurate program loudness readings on the fly; new zoomable and scrollable history and log file capabilities, and a fully resizable, configurable interface.

A number of notable improvements have been made to the ISL Inter

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Sample Limiter too, such as an Intelligent Dynamic Release Mode to enhance algorithm transparency by automatically adjusting to program content in real time; a new-look user interface workflow and display, and variable Center Channel link settings.

Finally, new features for the LMCorrect loudness measurement and correction tool have been added, including single-click analysis and correction; combined program and short-term or momentary loudness correction and a new LRA targeting option to automatically adjust audio to meet specific LRA targets.

The Nugen Audio Loudness Toolkit 2 will be available in April. www.nugenaudio.com

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PRIMACOUSTIC UNVEILS VOXGUARD UPGRADE Primacoustic has announced the VoxGuard VU, an update to its nearfield absorber featuring a window to improve sight lines between a vocalist/musician and engineer/producer.

“The VoxGuard is one of the most popular products in our line of IsoTools,” commented Primacoustic’s Jay Porter. “With thousands shipped worldwide we heard feedback from customers who suggested the inclusion of a viewing window would allow for eye contact and improve the recording experience. This simple solution can now be found on the new model that is now shipping.”

Renamed the VoxGuard VU, the product features a 3in x 7in window but retains all of the benefits of acoustic isolation for use in a

recording studio. The lightweight design also makes it suitable for mobile recording.

The VoxGuard VU’s high impact ABS outer shell is designed to keep unwanted noise out with slotted ports preventing excess bass buildup, while simple mounting hardware and a rear access cable port allows any handheld or studio microphone to be articulated and directed at a specific source. www.primacoustic.com

NEW 1073 UNIT FROM NEVE AMS Neve has announced the new 1073DPX dual 1073 microphone preamp/EQ unit.

Using the same circuit design and genuine components as the original 1073, matched with modern manufacturing techniques, the 1073DPX is designed to be a great-sounding microphone preamplifier unit with convenient extra features and an “unbeatable” price.

The model is equipped with two independent, Neve 1073 transformercoupled microphone preamp/EQ circuits, in a single 2U 19in rack mount enclosure; Neve Marinair transformers on each input and output stage; convenient connections for mic, line and DI inputs on the front of the unit, with switchable mic/line input connections on the rear; switchable +48V phantom power, Hi/Lo impedence, -20dB pad and DI ground lift on the input stage and switchable Phase, and EQ in/out.

Insert Send and Return features allow external equipment to be connected easily to the 1073DPX audio chain, and there are also independent channel output level controls; seven-stage LED signal level meters for each channel, which show mic/line/DI input-stage level, EQ-stage level and output-stage level up to +24dB; a headphone output with adjustable volume control, allowing users to listen to channel 1, channel 2 or both channels combined and external multivoltage PSU. The Neve 1073DPX is available now for £2,749 each. www.ams-neve.com

Powersoft revealed three additions to its Ottocanali install amp range at ISE. The eight-channel Ottocanali DSP+D amplifiers offer a wide range of system control and monitoring functions, as well as sound shaping options and a total of up to 12,000W output power at four ohms for lo-Z or distributed line systems – all neatly packed into a double rack unit.

With two redundant Dante digital streams, the Ottocanali 4K4 DSP+D, 8K4 DSP+D and 12K4 DSP+D are designed to fit any venue as a stand-alone unit, making them suitable for multi-zone applications in mid- to large-scale installs.

These new versions also feature switchable main and aux signal inputs per channel, adequate GPIO, alarms, Lo-Z and Hi-Z output connections, along with mono-bridgeable channel pairs. The two RU models implement two redundant universal switch mode power supplies

with PFC (Power Factor Correction) and patented SRM (Smart Rails Management) technology that maximises the efficiency of the system, while reducing power consumption. www.powersoft-audio.com

OPTOCORE AND FOHHN ANNOUNCE PARTNERSHIP

Optocore’s Tine Helme and Jochen Schwarz of Fohhn Optocore has entered into an agreement with Fohhn Audio to provide multichannel digital connectivity to Fohhn’s Linea Focus range of loudspeakers.

Explaining the background of the new partnership, Fohhn Audio’s public relations and marketing manager, Daniela Bohl, stated: “We were looking for a solution to provide our speaker systems with a digital signal and control data; up to now we only had an analogue solution. We chose Optocore because they are the pioneers of that technology and there is a perfect fit between our two companies, as we are both innovation-driven and Germany-based.”

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In order to provide integration onto the Optocore networks, the fibre specialist has created a card, while Fohhn has developed a new enclosure that can be attached to its Linea Focus series loudspeakers to provide optional input extensions.

This will enable digital audio and control data to be distributed simultaneously, either via Cat5 or optical fibre.

The new cards will be made available in fibre Optocore FX and Cat5 SANE TP versions. The Cat5 SANE TP option can function as a local extension to the fibre node, enabling users to build local daisy chains – with a distance of 100m between the loudspeakers.

“Both companies will benefit from this co-operation – Optocore will raise its profile in the installed sound sector, while Fohhn will be able to offer the best signal transport solution as well as distributed routing and redundancy,” added Optocore’s applications engineering manager Maciek Janiszewski. www.fohhn.com www.optocore.com

TANNOY’S TUNNEL VISION The new THL-1 tunnel speaker from Tannoy is designed to take the company’s horn heritage into a whole new area of application.

With the THL-1, Tannoy aims to provide a robust, durable and costeffective solution for emergency voice alarm/communication systems in the “physically and acoustically hostile” environment of road tunnels.

In an application notorious for high ambient noise levels and excessive reverberation, the THL-1 transmits crisp, controlled and intelligible voice announcements throughout the space from a minimum number of loudspeaker locations, according to the manufacturer. The THL-1 is also built to withstand the corrosive atmospheric contaminants, high moisture levels and temperature extremes typically encountered in these spaces.

To project an effective voice alarm notice over distances of up to 50m, the THL-1 incorporates an

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POWERSOFT PRESENTS OTTOCANALI DSP+D AMPS

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PRODUCT NEWS

asymmetrical AFEX (Architectural Flare Extension) horn, built to leverage the acoustical properties of the tunnel wall to effectively extend the waveguide effect for greater directivity and increased propagation along the horn axis.

The THL-1 is designed to carry intelligible voice notifications over greater distances, meaning fewer loudspeakers are required, resulting in significant cost savings.

The loudspeaker also offers six mounting holes, allowing it to be fixed to threaded rods or bolts attached to the tunnel soffit or unistrut. www.tannoy.com

JBL OFFERS ‘WALL-TO-WALL COVERAGE’ JBL Professional’s new Control HST wide-coverage loudspeaker, featuring Hemispherical Soundfield Technology (HST), marks the introduction of an all-new loudspeaker design, the manufacturer says.

Intended for indoor and outdoor use, Control HST is designed to use the wall it’s mounted on to its advantage.

“Our new Control HST not only overcomes the inherent acoustic reflections that typically occur with wall mounting, but actually makes the wall an integral part of the loudspeaker’s operation,” said Rick Kamlet, senior manager, commercial sound, JBL Professional. “A single JBL Control HST can provide wall-to-wall coverage, which can reduce the number of loudspeakers needed for covering a space, lowering the overall cost for the sound system.”

The JBL Control HST is ideal for a wide variety of applications, including retail stores, restaurants, educational facilities, leisure venues and hotels. It is IP34 water-resistant rated, which enables the unit to be deployed as an outdoor loudspeaker.

The speaker also uses JBL’s new HST Hemispherical Soundfield Technology to achieve its wide coverage of nearly 180º and eliminate the primary wall reflection that often causes inconsistent sound when loudspeakers are mounted to a wall.

By employing a wall-facing woofer, two horizontally-splayed tweeters, a bracket that spaces the loudspeaker at the required distance from the wall and a tapered cabinet that provides proper loading and dispersion of the sound, the Control HST substantially reduces these inconsistencies to deliver a flatter frequency response and a more even sonic character throughout the room, JBL claims. www.jblpro.com

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PRODUCT NEWS

EV’S NEW EVID CEILING SPEAKERS

Electro-Voice (EV) has launched a new range of EVID (EV Innovative Design) premium ceiling speaker systems.

The PC6.2 (6.5in two-way) and PC8.2 (8in two-way) models were designed to be “the most advanced ceiling speakers available, with superior performance across every detail”, according to the manufacturer.

Each self-contained unit features a dedicated EV-engineered true compression driver built for enhanced high-frequency performance and coverage control in comparison to the manufacturer’s current EVID ceiling speaker offerings. Simplified wiring and new mounting designs allow for quicker installation into a wide range of ceiling cavities and construction formats, the company says.

Both new speakers provide increased power handling (100W), higher output (115dB/116dB), and wider coverage angles (120/105º),

as well as user-friendly installation and low-profile looks, complementing the range as new high-end options.

Features include a highperformance high-frequency compression HF driver; highexcursion woofer; four-point clamp mechanism and integrated mounting legs, enabling quick setup; detachable pass-thru Phoenix-style signal connections to simplify installation wiring; tap selector with 8-Ohm/70V/100V operation and a reinforced ABS UL-rated baffle to ensure long life. www.electrovoice.com

MERGING DEMOS OVATION 6 WITH 3D PANNING

Merging Technologies’ Ovation 6 was demonstrated with a full 3D audio system at ISE 2015.

When it came to upgrading its media server and sequencer, Merging found itself tasked with identifying the key requirements of sound designers, who are creating ever more realistic soundscapes. With immersive sound now spreading to live events, product launches and even museums, the need to accurately place mono, stereo or multichannel sounds in a given space has become paramount, according to the Swiss company.

Merging’s creative team looked at a number of solutions that were already

commercially available and concluded there was an opportunity to improve upon ease of operation and realistic sound placement.

The firm believes it has something new and different to offer with the latest versions of Ovation and Pyramix – the DAW that is incorporated into the Ovation suite. The result of this work was on display in Amsterdam, where an array of 12 Tannoy Di6 Dual Concentric loudspeakers, driven by IPD Series amplifiers from Lab.gruppen, showed how customised virtual rooms can be created quickly and easily using the new technology from Merging.

The setup featured a TC Electronic System 6000 running the new Reverb 8 algorithm; the first reverb tailored for multidimensional 3D sound.

Visitors were also able to check out Pyramix 10, a digital audio workstation designed for sound design, as well as recording and mastering at any sampling rate up to DSD256 and DXD. www.merging.com

SENNHEISER SHOWCASES SPEECHLINE SERIES

Sennheiser showcased SpeechLine Digital Wireless, a new digital wireless mic system designed exclusively for speech applications, at ISE 2015.

SpeechLine Digital Wireless comes with Automatic Frequency Management, which automatically searches for free frequencies on-site, as well as an Integrated Audio Level Management feature that adapts the system to the application scenario and the speaker.

Network integration allows the system status to be remote controlled and monitored with ease using the Wireless System Remote (WSR) app, AMX or Crestron.

With lithium-ion battery technology, the system provides a battery life of more than 15 hours, and can be recharged

via USB if required.

Users are also offered various selectable sound profiles and processing algorithms, such as presets for more bass-intensive male voices or higher female voices.

SpeechLine Digital Wireless operates in the licence-free 1.9GHz frequency band, with no registration required, and also features 256-bit AES encryption for added security.

The system, which will be available in June, comes with a handheld transmitter or beltpack that accomodates lavalier or headset mics. All versions include a mobile transmitter unit, comprising a handheld or a bodypack system, battery and receiver station that can be integrated. www.sennheiser.com

CROWN EXPANDS DCI AMPLIFIER RANGE Crown Audio has announced the addition of three new models to its DriveCore Install (DCi) Network Series amplifier lineup.

The DCi 4|2400N, DCi 2|2400N and DCi 2|1250N power amps incorporate Harman’s DriveCore technology to deliver a compact, energy-efficient form factor and networked system control capabilities.

The models deliver 1,900W into 8 ohms, 2,400W into 4 ohms and 2,100W into 2 ohms. All power outputs are minimum guaranteed power ratings and offer direct-drive ‘constant voltage’ capabilities for 70Vrms and 100Vrms amplification without the need for a step-up transformer.

The DCi 2|2400N and DCi 4|2400N incorporate Harman’s DriveCore amplifier chip for a lighter-weight, installation-friendly 2RU form factor.

The Class D PWM DriveCore circuitry combines the amplifier driver stage into the power output stage along with additional audio-signal functions.

Both amplifiers provide networking and control features including continuous load monitoring, comprehensive error reporting, pilot tone monitoring and programmable general purpose input/output (GPIO) ports. The amplifiers also offer DSP features including LevelMAX limiters, an input compressor, input and output delay, input and output EQ, built-in crossovers and a built-in signal generator. www.crownaudio.com

March 2015

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SHOW NEWS: BVE

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ENTERTAINMENT AT EXCEL BVE has cemented its place as a crucial date in the calendar for the content creation and consumption industry, and this year was bigger and better than ever.

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here was certainly a buzz around BVE this year following the announcement that next year’s event will be part of London Entertainment Week – seven days of events designed to celebrate the achievements and ambitions of the UK entertainment industry.

Back to this year and more than 15,000 visitors from 60 countries descended on ExCeL in London to take part in the 120 hours of free seminar content and see the latest industry technology. They were not disappointed. Glensound’s focus was on its new commentary box for Dante audio networks – Inferno. This single user box can be used on its own, or networked in multiples for a multi-user system.

It has seven inputs for headphone monitoring and eight audio outputs, while seven illuminated talkback buttons route the audio input to seven different destinations and remove the audio from the main programme output. The routing of this I/O is all controlled by the free Dante Controller software, allowing seamless point-to-point and pointto-multi-point routing within a Dante network. All routes are stored locally on the Inferno whenever a change is made, so Dante Controller is only needed for making changes. Audio is uncompressed and at super low latency levels.

RØDE’s new RØDELink Digital Wireless System, NTG4 and NTG4+ shotgun microphones and the NTR ribbon mic were shown for the first time 10

March 2015

Glensound’s Inferno commentary box for Dante audio networks on the stand of Source Distribution.

RØDELink utilises next-generation 2.4GHz technology to provide a high-res 24-bit/44.1kHz digital audio signal at a range of up to 100m. The system will consist of a number of receiver and transmitter options, and will be available in a selection of kits targeting film, news gathering, presentation and stage use. The first available kit will be the Filmmaker Kit, which consists of a beltpack transmitter, an on-camera or beltpack receiver and RØDE’s broadcastquality lavalier microphone.

Also making its UK debut at BVE was the Genelec 8300 family of Smart Active Monitors (SAM), specifically the 8320, 8330 and 8351 models, and the associated 7350 subwoofer. All SAM models marry DSP hardware and software to automatically optimise and align themselves for relative levels, time of flight, subwoofer crossover phase as well as individual room response compensations, resulting in accurate, neutral imaging and low distortion, whatever the acoustic shortcomings of the room.

Cedar Audio was on hand to demonstrate the DNS 8 Live dialogue noise suppressor, as well as the DNS One plug-in as part of the Cedar Studio 6 suite, which now includes classic restoration capabilities in addition to Debuzz, Declip and Adaptive Limiter processes. Alongside these was the DNS1500 dialogue noise suppressor, which is used in various post-production and live broadcast environments all around the world.

Systembase demonstrated its new EasyMon Codec Network Monitor Tool, which enables a single browser window to simultaneously display status information of multiple C500xr codecs in a tiled format. The EasyMon Tool is available for free download from the Systembase website and is designed to work with C500 codecs running the latest firmware release.

Aspen Media made its BVE debut as a Bel Digital Group distributor,

Systembase demonstrated its new EasyMon Codec Network Monitor Tool

showcasing a wide range of the company’s products including the new BM-A1-4SHD audio and BM-AV2-4SHD audio/video monitors. Both units feature Bel’s intuitive front panel designs and come in a space-saving 200mm depth chassis, enabling the monitors to be easily mounted in outside broadcast vehicles. Housed in a 1U 19in rackmount unit, the BM-A1-4SHD features a threespeaker system providing high clarity.

The BM-AV2-4SHD is housed in a 2U 19in rackmount unit and comes with a 5in video monitor and 52-segment LED bargraph meters.

“We are delighted to be the new distributor for Bel Digital Group’s range of monitoring, delay and MADI products and it’s great to be handling a British brand,” commented Aspen Media director Chris Collings.

New from Kramer was the Dolev family of biamplified studio-grade speakers, which promise clear and balanced flat response with low THD even at high volume. Dolev speakers are built for control rooms, editing studios, sports bars, music clubs, high-end facilities, and home entertainment.

The family currently consists of three models: Dolev 5, which comes with 5.25in fibreglass woofer and a 50W amplifier; the 6.5 in/75W Dolev 6 and the 8in/100W Dolev 8.

Sonifex showed its new Redbox RB-AEC Acoustic Echo Canceller, a DSP-based acoustic echo canceller. The RB-AEC is used to remove the entire control room monitor speaker output from the presenter’s feed by adapting to the environment in which the control room mics are placed. Although acoustic echo cancellation is more commonly implemented in telephony systems, the Sonifex RB-AEC is designed to produce broadcast-quality cancellation.

Making its debut in London was Dedleg, a two-stage carbon fibre monopod which will fit in to any sound bag. Designed by a UK soundman, Dedleg is deployed in seconds in three easy moves enabling the sound recordist to take all the weight off his or her shoulders and back. It weighs in at 740g and can stay folded unobtrusively under your bag when on the move. www.bvexpo.com

Open & Closed The new flagship studio headphones

ATH-R70x PROFESSIONAL OPEN-BACK REFERENCE HEADPHONES ATH-M70x PROFESSIONAL MONITOR 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.

www.audio-technica.com

NEWS ANALYSIS

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RISE OF THE MACHINES?

ISE delivered a wide range of quality educational content for all manner of audio professionals this year, but a highly unusual session on auto-mixing hosted by Roland Hemming stood out for us. Here’s why.

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here were a lot of surprises in store for visitors to ISE this year, but not many bigger than the one Roland Hemming (pictured) of London-based audio consultancy RH Consulting had planned for those attending his talk ‘Better than human music mixing – live music auto-mixed for the 21st century’.

Attendees were probably expecting just another presentation on the possibility of advanced automatic music mixing equipment arriving some time in the future, but what they can’t have envisioned was a demonstration of fully-functioning artificial technology capable of mixing to the sort of standard you might expect from a decent human engineer – right there in front of them.

Understandably, there were a number of interesting reactions from the floor.

The participants, comprising manufacturers, end users and sales professionals, were then asked a series of questions about what they felt the implications of this revolutionary new tech could be, such as what effect will it have on engineers and whether they would consider using it themselves if it became available in some form.

Next came the bombshell – it was all a massive hoax. The system was an elaborate fake and the Korean Advanced Technology Research Group supposedly behind the whole thing was made up.

So what was the point of it all, then?

For Hemming, who says he knows of a number of manufacturers that are currently working on this sort of technology, it represented a chance to provoke a much-needed reaction and move the ‘will it happen or not’ debate forward by pretending effective automixing has already arrived – therefore prompting people to think about it more seriously – as well as enabling the collection of some useful data.

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As you would imagine, much of the preparation process consisted of ensuring absolutely everything was 100% believable, to avoid accidentally blowing their cover before the big reveal.

“If we just had a discussion on automixing and asked people what they think, you would get a very different set of responses to ‘here it is, deal with it’ and my understanding from the people who were there is that they really enjoyed it and felt challenged by it,” Hemming told

Audio Media International.

“Beforehand, we went through some of the ideas involved, as it had to sound plausible. Some of these technologies actually could happen – we talked about using traditional gain sharing, which other systems use, so there’s no surprise there.

The second thing was spectral analysis, and again there’s another system out there originally developed at the Queen Mary University of London that uses that, which basically looks at the whole spectrum and places instruments in the mix depending on where they sit in the spectrum.

“The next thing we made up was that it would have a library of timbres that it would know, so it would begin to learn what kind of instrument it is hearing and therefore you could then give it a statistical map of what an engineer would normally do with an acoustic guitar, kick drum, etc. You could put down 50 different setting for each of those things.”

They also created something called Heisenberg pre-processing – named after the uncertainty principle, rather than Walter White’s alter ego in Breaking Bad – where the signal is split, analysed and then processed before being output.

So how were they able to create a convincing system? It seems Hemming and co had thought of everything when it came to the setup, which would explain why so many readily accepted what they were being shown.

“We had a 24-track music playback.

We kept those 24 channels and pre-mixed it into stereo. We played it three times and there was no mixing taking place live whatsoever – it was all done in advance,” Hemming explained. “The first mix was deliberately bad and raw, with the levels all over the place, and that was playing it without it doing anything. Then we started doing the mix and the idea was that you could hear it taking 30 seconds to put the mix together. The third time we ran it we said ‘well it’s learnt now so it will sound good from the start’, which it did.

“And that was all coming off one laptop. From there we output Dante into a second laptop, which offered a view of the mixer and we created some display using Max MSP. “To make it seem like it was doing something musical we triggered that program with MIDI notes from the sequencer, which we timed against certain cues. That gave the whole thing a degree of realism. It wasn’t a trivial task to make it look real.”

The experiment also threw up some intriguing results. For example, although most people predicted ‘small bands without an engineer’ as the main customer/application for this sort of technology, there were a fairly high number of votes for ‘small/medium music venues’ and ‘theatre’.

And as for whether engineers should be worried about their roles changing or being phased out entirely should this sort of gadgetry come into play, there was a bit of a contradiction.

“The general impression was that it won’t replace engineers but will be a helping hand for them, but then also quite a lot of people think engineers would lose their jobs. So which is it?” commented Hemming.

“The question here isn’t whether it’s better than an engineer; it’s whether

it’s good enough, and the whole notion of ‘good enough’ is an interesting idea.

Tablet computers aren’t considered as good as PCs, but in some cases they’re good enough to replace them.”

Staying on the topic of engineers, another question tackled whether engineering should be considered art, function, or a combination of the two.

“A live engineer might say ‘a machine can’t replace what I do; what I do is art’.

My response to that is: are you trying to tell me that during a two-hour gig, during every second of every song, everything you do is art? Some of it must be function, some of it must be operation?”

It turned out the audience was split more or less 50:50 on this one, although nearly half believed auto-mixing would help improve an engineer’s capacity and work, while 70% felt engineers would accept it is another tool for them to use.

It was certainly a thought-provoking exercise, and those responsible for its execution surely have to be applauded for their unique, and it has to be said, brave approach. For more on this, keep an eye on the RH Consulting blog via the website below, where a full summary is soon due to be published. www.rhconsulting.eu

OPINION

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AUTOMATED TO EXTINCTION?

FOH engineer and studio owner Ben Hammond asks whether the increase in cases of musicians taking care of their own mixes is putting the role of the traditional monitor engineer at risk.

BEN HAMMOND

H

aving just come off a fiveband-bill UK tour, on which four of the five bands were carrying their own in-ear monitoring system, and not one monitor engineer, it begs the question – are monitor engineers soon to be a thing of the past?

An extreme statement I know, but with the current crop of extremely affordable small-format digital mixing consoles, and easy-to-use but surprisingly effective apps that even a bass player can use, are we going to see monitor engineers face the same fate as the large-format recording studio and become, in some circles at least, almost obsolete?

As time has passed since the rise of the bedroom producer, and much of the novelty has worn off, many acts are now starting to realise (thankfully) that maybe an experienced recording engineer and/or producer would do a much better job of capturing their band on record – this in turn has seen a noticeable boom in the studio business of late. Now things will obviously never return to the glory days of a residential SSL/Neve studio in every town, but the talent of a skilled recording engineer – something that was once held in such high regard – is once again becoming an appreciated and sought after thing.

So with the advent of the ‘app’ and the iPad mixer giving the band member the ability to easily control exactly what 14

March 2015

they hear and not having to communicate through the ‘middle man’ monitor engineer, will this result in the dream ‘CD-quality’ in-ear mixes? Or is it a recipe for disaster? Obviously most musicians’ knowledge will stretch as far as selecting their mix and moving faders, but this then begs the question of whether the subconscious element will kick in, and the band member will believe their monitors are perfect as they themselves have dialled in their exact mix, completely ignoring any kind of channel processing etc. Or will this not wash, and will it be all too obvious that it “just doesn’t sound right”. The next question we then have, in the face of ever-dwindling budgets is, “well is it good enough to get you through, as if you do it yourself, you will save X amount of money…?”

Where to start?

Now obviously I’m not suggesting that monitor engineers will disappear completely. As with everything, when a band reaches a level at which budgets allow things to be done properly, and monitor mixes become more complex and increase in numbers, then the monitor engineer will be brought in, but with the aforementioned current trend of up-and-coming bands taking care of their end of the multicore themselves, where does this leave the up-and-coming monitor engineer? Where do they hone their skills? Could the technology that we are bringing in today – to make for a much slicker, recallable, lightweight, and automated show – be in the long run putting us out of the job? As crew, by introducing these ideas to the band, are we in turn actually creating our own biggest enemy?

It goes without saying that the pros of the self-contained monitor system are many. The band are in their familiar sonic territory wherever they go, and so long as the system has been properly designed and built, it will interface with existing house systems easily and be very much hassle free. But in the event of error, the band members suddenly find themselves in a very lonely situation.

With more and more bands turning to amp simulators and dummy cabs, and striking the wedges completely from the stage, not to mention click tracks, cues etc, if the IEMs happen to go down, it’s not an easy thing to get around. The argument, then, is if you are going to make such radical changes on stage in light of moving to IEMs, surely then you need to bring a monitor engineer to make sure things function correctly, as in the event of loss of audio, the gig simply cannot happen.

So the burning question here is are these systems being seen as a one-off investment that will then eradicate the cost of a monitor engineer for good, over a way to further improve the band’s show if ran properly by the appropriate crew?

Just to clear up, I’m not an angry monitor engineer; in fact I’m not a monitor engineer of any kind, but a FOH engineer who now has the added responsibly of the band’s monitor mix – not necessarily the actual mixing of

the monitors, but being responsible for the RF, the patching of the split, the WiFi connection etc – all simple things in their own right, but thrown into a rushed festival changeover situation on top of your normal job, how long until 10,000 people separate you and your suddenly silent band members?

Apologies for coming across a little doom and gloom, but these are things that need to be addressed in the event of moving to such an unmanned monitor rig. The opportunity for bands to own this kind of system, and the advantages that it gives them for such little outlay is incredible, and once more a testament to modern technology, but in this industry as in so many others, nothing can quite beat the human element.

Ben Hammond is a FOH engineer and tour manager for bands such as Deaf Havana, Saxon and Young Guns, and owns Reel Recording Studio in Yorkshire, UK.

OPINION

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MOBILE LOUDNESS: AN ADAPTIVE APPROACH

Rob Bridgett, producer and audio director at Clockwork Fox Studios, explains why an intelligent solution to the issues associated with mobile game loudness is required.

ROB BRIDGETT

A

s an audio director and mixer, having worked on the console side of game development since 2000, I had become acutely aware, through commissioning and implementing audio content, of the necessity for standardised output loudness recommendations (both average loudness and loudness range).

Fast forward to late 2012, and Garry Taylor and the ASWG (Sony World Wide Studios Audio Standards Working Group) made Herculean efforts to get pretty much everyone in the industry to agree on a loudness target via the ASWG-R001 document (http://gameaudiopodcast. com/ASWG-R001.pdf) finally published and updated in 2013.

The Sony document adopted the -24 LUFS standard, essentially the ITU-R 1770.BS-3 recommendations for broadcast, and, over the last few years, Microsoft, Nintendo and the Independent G.A.N.G. I.E.S.D. body have all subsequently fallen into line with the same recommendations and agreed that these loudness levels made great sense for games in the console-based home entertainment space.

In terms of mixing games, having these recommended levels to aim for

has been really great. Prior to their arrival, it was mostly a guessing game in terms of overall levels. The landscape has changed dramatically for the better over the last few years, and ITU-R/ EBU adherent metering is now not only available, but deeply embedded inside DAW workflows (Nuendo) and audio middleware monitoring workflows such as AudioKinetic’s Wwise.

While loudness has been taking centre stage inside console game development audio circles, another significant change has been happening on the side – the rise of mobile gaming via the success of iOS and Android devices as gaming platforms.

It is in the realm of the indie developer that loudness has not yet gained the traction that it has at the larger publishers and triple A developers. And it is easy to understand why. Loudness recommendations don’t seem to translate well to a mobile platform. It is likely very low on a list of priorities for an indie developer in getting their game to the market, for whom even audio integration may be similarly low priority. It is an area where there is great difficulty in pinning down a standard, but the biggest hurdle is that a single standard for mobile devices seems to make very little practical sense.

The idea of a single standard for home entertainment is, by comparison, relatively easy to pin down because the assumption, in the vast majority of usecases, is that the listener/viewer will be consuming the content, via a television, somewhere inside their home.

With mobile, the gamer could be using either the device speaker, the device headphones, or maybe even mirroring the device’s content onto a larger home screen via Apple TV. Add to this the range of locations that a listener may be in – inside, outside, on a bus, on a plane, in a quiet café, a busy street, a classroom – and you have a range of loudness usecases that one single loudness standard cannot ever hope to accommodate.

Mobile Loudness is not just an issue for video game content, but also mobile movie-player software, such as the Netflix player, and could benefit greatly from an adaptive approach to its loudness output.

This is where adaptive/intelligent loudness solutions need to take over – solutions that not only take into account the user’s current output method (detecting headphones, speaker or AirPlay mirroring) but going even further – solutions that dynamically monitor the levels of the background noise in the user’s environment, and adjust the LRA and levels at runtime to compensate.

Fortunately, the research and work already done in other areas of entertainment on loudness offer solid benchmarks for many of these different situations. Mirroring to Apple TV? You are now in the broadcast domain and -24 LUFS becomes the target. Using the device speaker? The iTunes ‘Soundcheck’ loudness of -16.2 LUFS becomes a legitimate benchmark.

Listening on headphones? Slightly trickier. In an ideal, mostly quiet setting, a -24 LUFS level will still work great – when the environmental noise gets louder, runtime master compression can edge things up to -18 LUFS (Sony’s Mobile target) or even -16.

But, what if you are a young child playing an educational game? Something a lot lower, safer and more comfortable is likely necessary around -28 LUFS (levels I am currently working with on our own ed tech products, and which still feel quite

‘hot’ for kid’s ears at full device volume) – this can be edged up to -24 LUFS with -5 LRA by the runtime compressor for noisier travel or classrooms.

Having the software switch (in the case of the device output selected) and blend (in the case of runtime LRA attenuation) seamlessly between these different standards is in reality fairly trivial to set up using today’s middleware and a Unity plug-in (which you’d currently need to write yourself!) to detect the output route of the device and use a device’s built-in microphone to monitor audio background levels when headphones are plugged in, but it is certainly not beyond the reach of any developer with a firm focus on quality audio.

With this technology in place, and with a range of targets defined, it is far easier to mix for mobile by getting everything to sound how you want it at the ‘idealised’ middle target, say -24 on headphones in a quiet room. Then, using the other states to simply tweak the mix at the other target levels through the various other outputs and simulating various noise conditions.

Rob Bridgett is audio director and producer at Clockwork Fox Studios in Canada.

Inspired by the work of the Sony ASWG group and the lectures of sound advocate Julian Treasure, the Clockwork Fox team have been working on developing adaptive loudness switching and runtime LRA attenuation for child-friendly mobile ed tech products for the past two years.

March 2015

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OPINION

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THE IMPORTANCE OF ACCURATE ENGINEERING

Audio engineer Aston Fearon offers his advice on how to ensure the highest levels of precision when mixing a live show.

ASTON FEARON

I

n our pursuit of exceptional quality sound for our audiences it’s important not only to think about the tools we have at our disposal but also the accuracy of our techniques.

We can achieve good results using rough estimations and making coarse changes but the more accurate we can be with our changes, the more robust the systems we use will become (with more than adequate headroom) and the more we can get each instrument to fit neatly where we want it to in the mix. This might mean avoiding certain tools and techniques as well as realising the limitations of human judgement to produce the precision we need.

of 575Hz. Parametric EQ is usually much more accurate because of its sweepable nature and Q.

Bandwidth Finding the central frequencies that are likely to ring in a room is important but if we don’t seek to see how many frequencies either side of it are also a problem then we won’t be able to define the Q of our EQ bandwidth appropriately.

Too narrow and we aren’t being thorough enough; too wide and we are destroying too much of the signal in the same way that using a graphic does when we need to pull out two adjacent frequencies.

Real-time analysis (RTA) Parametric Equalisation The problem with graphic EQs is that because the bands are fixed they are unable to ring out a room very accurately or efficiently. They come close and can get the job done but they often take other frequencies, which aren’t a problem, down with them. If the offending frequency in question is 575Hz, for example, I would have to pull a notch out of 500Hz and 630Hz to try to achieve the desired notch, which is less than ideal because the notch at 500Hz and 630Hz would evidently be more severe than if I could pull a notch out 18

March 2015

Being able to name a frequency by ear is a good skill for a sound engineer to have but because graphic equalisers only have 31 bands, the accuracy at which we can name that frequency is either limited by these 31 bands or difficult to be exact because of the number of actual frequencies in the spectrum of human hearing. There is often debate about whether to use our ears only or also incorporate visual and numerical elements into our engineering processes.

Personally I think the answer is both. My ears might tell me to take a notch out of 10kHz but Rational Acoustics’ SMAART

might identify the frequency as 9.5kHz. I still need to verify whether I agree with SMAART’s analysis but it can give me a lot of useful information to help me make decisions more accurately.

Phase My ears can tell when the combination of two or more sources affects its tone undesirably. Although I know this is being caused by identical sound waves which arrive at slightly different times within a cycle, my ears can notice the effects but cannot accurately hear phase itself. It’s far more effective for me to use SMAART to distinguish how much the sources are out of phase and where I need to move my boxes or add delay to get an accurate alignment. A wavefront that is phase incoherent will negate accurate EQ choice further upfield at the desk, even if a source sounds great in the headphones.

Referencing Using a reference like a good pair of headphones (or nearfield monitors) can help us to listen to our sound sources before they hit the PA. If we are happy with what we hear but it doesn’t sound the same through the PA then we can use system EQ to correct any discrepancies in the system rather than trying to fix

the problem at the channel EQ. If we solve the problem at the channel level rather than the system level we’ll have to correct multiple channels and when we use the desk file again on a well-tuned system the channel EQ will be void.

Once we know our system is tuned accurately we can also be sure that anything we PFL (Pre-Fade Listen) will be the same as what the audience will hear.

Line checking in this way is only easy if we have an accurate reference point to use.

The more accurate this reference point (hopefully with little coloration) and the more familiar we are with how it sounds; the more accurate we can be when we make tonal changes to individual signals.

All of these considerations can give us an extra degree of resolution here and there, which may seem negligible, but added together the results can create a big difference to the clarity of a mix and the efficiency and headroom of a system – both at the amplification stage and the SPL we mix at. This means a system can work less to produce a result that sounds even better to the ears of our audience.

Aston Fearon is a freelance sound engineer and technician mostly working at the FOH end of the multi.

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GEO FOCUS: SWITZERLAND

SWISS WATCH

Population: 8.1 million

A focal point of production and pigeonholed for efficiency and performance, how is the Swiss pro-audio market currently faring? Matt Fellows investigates.

D

espite difficult economic conditions across the globe, Switzerland has seemingly managed to abandon the economic sinking ship.

The country currently enjoys the highest nominal wealth per adult in the world, according to Credit Suisse, and the eighth-highest GDP per capita. But how does that extend into the pro-audio market?

Roger Roschnik, CEO of PSI Audio, brings us up to speed on the current state of play: “The climate and market are very good and healthy compared to neighbouring countries. People still have money to spend on music and other hobbies and government spending has not been reduced.”

The country also enjoys a strong position in the pro-audio industry as a hub of distinguished manufacturers: “There are a lot of brands in Switzerland that are strongly valued, such as Studer, Revos, Nagra, Thorens, Lenco and PSI Audio.”

And there’s hardly been a better time in recent years to be such a manufacturer; Claude Cellier, president of Merging Technologies, has seen his company perform well over the past three and a half years in part because of the security that Switzerland’s financial climate provides.

“Our company has fared pretty well in the current climate since the Swiss National Bank decided to top the Swiss franc versus the euro currency to 1.2, allowing us to benefit from a relatively stable and predictable competitive environment,” he explains.

“One of the big bonuses of being a Switzerlandbased manufacturing company is usually the stability that our environment provides. Political stability, economic stability, social stability – all are deep-rooted givens that Swiss businesses enjoy as premises to their operations.”

Grey skies ahead?

So the country’s fertile economic ground appears to contribute to the industry’s recent comparative successes. But just because it may not ostensibly be struggling as much as its neighbours does not

20

March 2015

mean it doesn’t have its own unique challenges; with the lucrative financial landscape left uncertain as a result of exchange rate fluctuations in December and January, Cellier further notes: “Not only is the Swiss market a relatively small market, but on top of that it is further divided into four different language zones, which doesn’t make things simpler to manage.”

However, such a multi-faceted culture does afford the shrewd manufacturer an opportunity to capitalise: “The per capita expenses are probably one of the highest in the world, and due to the different cultures that coexist in our country it makes it an invaluable test bed for new product introductions,” Cellier adds.

Facing the music The markets perhaps closest to Switzerland’s heart, however, are the live and studio sectors, and music has always been a large part of its culture, even when compared to the rest of Europe.

“Switzerland has always had a vivid music scene,” states Christian Müller, studio manager at Powerplay Studios. “Because of the strong financial situation here, young amateur bands can afford to record music.”

And on the live scene, Keith Watson, global marketing director at Studer, comments that the country “has a vibrant festival culture”, with some of the biggest festivals in Europe such as Paléo Festival Nyon and Montreux Jazz Festival.

Now, a love for the medium is a great thing, but does it keep the money flowing?

“Music has never been very lucrative in Switzerland,” explains Roschnik. “So it has attracted passionate people more than greedy people.”

But passion alone doesn’t generate a smooth business climate in the studio or live sectors, as Müller points out: “We are very small, so making a living as a musician is not so easy, even with touring.”

“Individual purchasing power is high but the market is small with less money in the music industry than other European countries,” he

continues. “We are still moving from a market of music professionals to a market of music lovers.

This means many more potential customers, but with less money, less technological knowledge and more emotionally driven.”

Silver linings What you don’t often hear, however, is the sense of tentative optimism from those in the industry as, rather than lament the changing climate as it slides into challenging and unfamiliar territory, companies instead seek to capitalise on the benefits of the new market. ”The drop in margins was less of a shock to the industry than in countries with formerly large and lucrative music industries such as the UK, the US and Brazil,” remarks Roschnik.

Müller explains how his studio has discovered new business avenues to support existing and ailing methods: “We’ve opened new markets, such as the digitising of audio archives. In the next years, all of our tapes need to be safe before they ‘die’. We are definitely the studio with a huge know-how in analogue and how it is connected to the digital world in daily use. So transfer is a strong market for us.

Once, machines seemed to be a risk – now they are an opportunity.”

Quality time A major trend that becomes apparent when examining the Swiss market is the consensus among professionals that quality in audio is making a comeback.

This belief is corroborated across the country in a number of sectors as professionals hold high hopes for a more audio-savvy consumer base.

“The past decades have been an endeavour to provide music to people wherever they are through connectivity, file compression, small devices and so on with little or no focus on quality,” Roschnik details. “People are now rediscovering sound quality and demand for high-quality music, equipment and listening conditions will continue to climb.”

GEO FOCUS: SWITZERLAND

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From the manufacturing viewpoint, Daniel Zurwerra, AV consultant at Virtually Audio, comments on how demand for audio quality is beginning to overtake demand for video quality, and how it may affect production: “I see more and more people realising that sound really affects us. Thus, it’s worth it to design products not only for the eyes but for the ears. For instance, the most annoying thing in a videoconference is bad sound.”

In the studio, demand for high-quality audio is also rising, with consumers seeking out the aid of professionals to secure it, as Müller remarks: “The market is recognising that good sound needs a good ear from a trained engineer. So that service will remain a need to consumers.”

And demand is even burgeoning in the live sector, with Watson commenting: “The expected quality level in general is very high. There are a lot of venues and events often competing within a very small area, and this also applies to rental companies.”

Upon further examination, such coherence across

the industry is further indicative of the key role that audio quality will play in the future of the Swiss market and beyond; it is a key differentiator in the market. And this differentiation maintains a crosssector relevance.

Roschnik argues that quality is a proven incentive to customers and a recipe for success: “We produce loudspeakers that are the last element in the music chain before the ears and can’t be dematerialised. Our company has always produced handmade, innovative and top-quality products that users will typically keep for over 20 years. Our most powerful sales weapon is comparative tests and blind tests that we consistently win. The professional can trust what they hear.”

On the recording side, Müller reasons that Powerplay’s service offerings will help it stay ahead of the curve in the changing climate, embracing emerging trends with a workflow that incorporates home recording for overdubs before final mixing

in the studio, making things more affordable and convenient for clients.

“The musician can lower his production cost and still has a good quality sound. He will spend more energy on the quality, because everybody can work at home now. So, if he would like to stand out from the bulk, he has to bring his music to another level.”

Roschnik attests that this climate may not actually be entirely new, but rather its latest manifestation and that, in the manufacturing sector at least, the same techniques used in the past will secure safe passage forward. ”For many decades Switzerland has had to compete on innovation and know-how rather than price,” he adds. “We plan to continue doing what we do best: design and manufacture the best quality speakers we possibly can, based on scientific evidence and subjective listening.”

And it appears as if other pro-audio companies are prepared to follow suit.

Have there been any significant changes to legislation, regulations or laws that have affected the industry in recent years?

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

‘Last year’s immigration vote will have an impact with the EU.’

No, but at this time the extraordinary strength of the Swiss franc compared with the euro will have an impact.’

Better

Worse

Stay the same 10% 20% 30% 40% 50% 60% 70% 80% 90% 100%

Have you seen any trends in technology purchasing? Is there anything that is unique to Switzerland?

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

Economic uncertainty

‘DAB implementation and loudness compliance are still strong.’

‘Something unique is that each band has its own studio and a lot of them are very well equipped.’

Failing budgets Slow payments Goverment legislation Other 1

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3

4

5

6

7

8

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STUDIO PROFILE

MAKING A MARK

Now situated in sunny Santa Monica and run by former studio boss Michael Marquart’s daughter Samantha, Windmark Recording has experienced nothing but success since moving West. Adam Savage went on the case to find the reasons why.

I

t cannot be easy to establish a highend recording facility in a place like Los Angeles County these days. In an area already scattered with big-name spaces, most new studio owners can expect a tough challenge right from the off, but there were no concerns that Santa Monica-based Windmark Recording would struggle to get off the ground when it opened its doors in 2012.

Initially set up in Virginia almost 30 years ago by Michael Marquart, a successful producer, songwriter and former drummer for ’80s new wave outfit A Flock of Seagulls, Windmark is now making a name for itself in California under the leadership of Marquart’s 22

March 2015

daughter, Samantha, with Michael’s son Mikey also responsible for the upkeep of the facility and its equipment.

Following the sale of the original studio 10 years back, Michael Marquart had been looking for somewhere to build his own private space further West, but Samantha had a better idea.

“My Dad was thinking of building a small studio at his house in Malibu so he had a place to work when he was in town, and I suggested he build something in Santa Monica so my brother and I could use it as well,” she recalls. “In my effort to try to convince my Dad to agree, I thought it would give a little bit more incentive if I threw in the idea of renting it

out every now and again when he wasn’t in town. That’s when it hit me – why in the world wouldn’t we just open up a studio out here?”

The decision was made to acquire a sizeable space formerly occupied by Grammy award-winning songwriting and production duo Jimmy Jam and Terry Lewis, designed with high-quality wiring and a focus on eye-catching aesthetics.

And although they went with the old name to carry on Windmark’s legacy, the Marquarts felt that a fresh start was for the best.

“When we decided to open on the West Coast with me as the manager, we approached it like a new studio

entirely,” Samantha says. “What worked for my Dad on the East Coast wouldn’t necessarily work out here in terms of studio operation, so we needed to start from scratch. However, my father set a standard of excellence when he first opened Windmark, and that remains unchanged.”

A RARE TREAT What also remains unchanged are some of the six-studio complex’s most prized possessions when it comes to gear, including one of only four ultra-rare vintage Neve A4310 consoles ever built and a 1967 Ampex 16-track two-inch tape machine once owned by Les Paul –

STUDIO PROFILE

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Samantha Marquart

both of which can be found in the newly designed Studio F. This particular Neve, which features 24 busses and 32 monitor channels, is equipped with 40 31105 mic pre/EQ modules, Neve 2254 and Shep S2151 compressors, and is quite a beauty.

Other highlights include a neat set of soffit-mounted Augspurger monitors and a raft of outboard gear from the likes of Focusrite, Urei, API and SSL. Designed by renowned Swiss-born architect Peter Grueneisen, Studio F also ticks all the boxes for Windmark assistant engineer Tristan Bott, who didn’t have long to get used to the setup once it was complete.

“The day after it was finished we had a session in there, so I had a pretty short turnaround to learn the room and the board [the A4310],” Bott recalls.

“The guys that put the room together did a great job and made it easy for any recording scenario.”

“Our Neve is fitted with a Sony turntable. So whenever there’s that rare chance nowadays that someone brings vinyl, they can cut it right into Pro Tools from the board.”

The equipment offering is a huge attraction, then, but Samantha Marquart believes the homely feel of Studio F and the professionalism of her team is just as important.

“It has such a comfortable atmosphere.

Our lounges feel like living rooms, but more than that, it’s the people we have working here. Our ‘family business’ attitude is really carried throughout all of our staff and our operations run very smoothly because of it.”

And from a technical viewpoint, even clients with the most complicated of requirements will likely find the solution they need at Windmark, as Bott explains: “Any one of our control rooms, vocal booths, or live rooms can be connected together via patchbays and Ethernet, allowing endless possibilities of utilising a room’s equipment and combining recording spaces together for complex sessions,” notes the engineer. “You could have someone singing in a booth across the building and a full drum kit playing in a live room on the other side while listening to a cue mix in our kitchen.

Anything is possible.

“Our client services are top as well.

Signature lemonade, fruit plates and cookies; I mean, what more could you ask for? We take pride in accommodating our clients and giving them the best studio experience.”

SHE MEANS BUSINESS Despite having an experienced studio owner father to learn from and a strong musical background herself – Samantha is also an accomplished singer –

Marquart sees her business acumen as one of her main assets.

“It’s always important to have a good business sense. To ensure a company’s survival you have to constantly be evolving and adapting, especially with a recording studio – a section of the music industry where you wouldn’t think there is much room for innovation,” she explains.

“That is where I think I have a little bit of an edge. I didn’t plan on being a studio manager. I went to Pepperdine [University, Malibu] and graduated with a degree in Integrated Marketing Communication, and aside from what I learned from my Dad, I knew nothing about running a recording studio. So naturally I am going to have a different perspective than someone else who has been working in studios their whole life.”

As for what her main plans are for the future of the facility, Marquart offers a refreshing outlook.

“The music business is always changing; that’s what is so great about it. People are consistently trying to think of a new way to release music or promote their album. The recording side has remained unchanged for a while.

That, to me, is exciting; there are endless possibilities,” she notes.

“Innovation is king, and that has to be at the forefront of your mind when you are running a business. As a recording studio, we are a breeding ground for creativity. There is no reason why that shouldn’t carry over into the way we do business as well.

“I plan to consistently grow and adapt Windmark over time, but it’s hard to say where we will end up. As long as we maintain our standard of excellence and loyalty to our clients, I don’t think we will have any problem retaining a strong position in the business.”

Perhaps unsurprisingly, considering how she can count acts such as Coldplay and J. Cole among her previous clients, Marquart sees the social side as the main perk her job offers.

“The best part has been meeting all of the amazing people,” she says. “As you can imagine we get an array of people in here on a daily basis – people from all walks of life, every one with a different story and a different worldview.” www.windmarkrecording.com

March 2015

23

GAME AUDIO FOCUS

www.audiomediainternational.com

IT’S BEHIND YOU!

Matt Fellows caught up with sound designers Byron Bullock and Sam Cooper from Creative Assembly to chat about the audio elements of survival horror hit Alien: Isolation .

O

ver 25 years since the Xenomorph from Ridley Scott’s 1979 sci-fi horror classic Alien first crept out of the shadows to cocoon itself in cinema history, those first nail-biting moments haven’t faded as the franchise expands evermore into the gaming world.

Set 15 years after the original film, Alien: Isolation puts the player in the shoes of Amanda Ripley, daughter of Sigourney Weaver’s iconic character Ellen Ripley. Stranded in space and stalked relentlessly by a familiar and ever-present alien, the player must keep their senses alert and their wits sharp if they are to escape alive. And one of the key (and only) weapons in their arsenal is a keen pair of ears.

The game aims to assimilate itself into the Alien universe narratively and aesthetically, as sound designer Sam Cooper notes: “The mantra was authenticity and staying true to that 1970s version of the future, that world of tactile technology with big buttons and noisy computers.”

And the team were committed to staying faithful to their progenitor. ”We did a lot of research and we interviewed Terry Rawlings [editor on Alien] – he gave a lot of valuable insight into what it was like creating the original film,” explains Cooper. “We also managed to get Fox to dig out some old recordings on reel-to-reels from their archive.”

24

March 2015

These original recordings from the set of the film proved to be a cornerstone of the game’s authentic lo-fi aesthetic, used extensively to build the library of sounds heard in the game, as Byron Bullock explains: “We could take that stuff and expand upon it and make more material from the original.” ”We got the stems, so we had those as a guide,” Cooper continues. ”We resynthesized a lot of the telemetry sounds, but we were actually able to pull stuff out of the sound effects stem, stick it into a DAW and remake it piece by piece.

“We wanted to create high-quality source. We toyed with the idea of using reel-to-reels, but in the end we created high-quality source and degraded it using plug-ins afterwards so that we always had the original to go back to.”

alone in the dark In Alien: Isolation the player is forced to rely on their sensory awareness rather than brute strength, and the team consciously crafted the game’s sound to complement these gameplay conventions. ”Sound was planned to be used as a gameplay device from the beginning,” Cooper notes. ”A lot of what goes on in the environment is implied by sounds that aren’t in the frame, so the player can hear where things are, where the enemies are, the alien crawling through vents overhead.

It’s about what the player can see, because they are controlling the frame.

So we did a lot with view-cone scripting to

add rises in tension. If the alien is creeping up on the player and they’re facing the other direction, we didn’t want the music to give the game away, so the sound is based on line-of-sight.”

As with any good horror media, for the team it was as much about utilising the absence of sound as it was about using sound itself. “We tried to be quite minimal with sound,” Bullock notes.

“There’s moments when you can just be in a corridor and hear the creaking of the corridor around you and other Foley, and that all helps the feeling of claustrophobia. It was important to get some dynamics in the mix; there are moments that are really quiet, but there are also big, loud moments.”

But the team’s biggest obstacle came in the form of the salivating villain.

“The biggest challenge was that we have this unpredictable, adaptive alien – he can hear you, he can see the light from your torch – and he uses those senses to hunt you down,” Bullock explains. “We wanted to make moments of suspense and tension by pre-empting what he was going to do.

For us that was very difficult – you never know what he’s going to do, because it’s all down to what the player is doing.”

So to craft a soundscape, the team needed a logic system as dynamic and fluid as the player’s erratic nemesis, as Cooper notes: “We had interactive systems driving the mix as well as scripting, so we’d have variables which could adapt the mix and the score at the

same time to create a dynamic range.”

“We created these values called ‘stealth’ and ‘threat’, which told us how stealthy the player was being or how much threat they’re under, and those values remix the music in real time to help drive suspense,” Bullock explains.

“For example, if the player is hiding under a table as the alien is moving past them, the context-driven system will change the music so you get rising strings as the alien is coming past, and it lowers sounds of the atmosphere to concentrate on the player’s and the alien’s sounds to create focus.”

The game is distinguished in gaming media for its use of sound, and this is due in part to the studio-wide collaborative nature of its development. “I haven’t worked on a game where they wanted sound involved so early and having sound inform aspects of the design or the art,” notes Cooper.

looking ahead And as the next-gen of hardware opens up more avenues for sound design, Bullock opines on its future: “It’s becoming more accessible for more indie developers. Some of the tools to create good game sound are becoming relatively affordable to licence, so I think we’ll start to see more good-sounding games. And triple-A budgets are becoming bigger so the expectation on the sound team becomes bigger, and ultimately that will make the games sound better.”

LIVE FEATURE Funktion-One’s VERO touring rig

THE GREAT BALANCING ACT

///////////////////////////////////////////////////////////////////////////////////////// The onus on touring sound equipment manufacturers to produce equipment that is compact, practical and – perhaps above all – cost efficient is now greater than ever. But is there a danger that these pressures are having an impact on overall system quality? David Davies finds out.

T

he explosion in the number of live music events over the past 10 years has been one of the true success stories of the European entertainment industry. But for audio manufacturers keen to respond to the concomitant change in requirements of live sound systems, it’s a development that hasn’t been without its inherent difficulties.

For instance, as artists’ income from recorded work has declined, the need to optimise revenue from live performance has become more acute. Reducing overheads – not least transportation costs – has led to an obvious demand for multi-functional audio products that 26

March 2015

can deliver extensive feature-sets in compact form-factors. The less gear required for a touring production – particularly one crossing countries and even continents – the better.

With equipment prices simultaneously being driven down, the overall expectations of manufacturers have arguably reached new heights during the past few years – and all this at a time of increased inter-market competition.

“Quality cannot be compromised as competition is high because there are more manufacturers now,” says Dave Millard from Full Fat Audio.

With that in mind, we decided to explore some of the challenges facing

manufacturers as they attempt to juggle these multiple requirements, focusing in particular on the matter of practicality – and whether there is a risk that it might ever be achieved at the price of quality.

THE VIRTUES OF VERO Revealed exclusively for the first time in Audio Media International, Vero is a new loudspeaker system from Funktion-One that is claimed to achieve a balanced equation of quality and feature-set.

“Every detail of its composition has been meticulously considered and developed,” states Funktion-One founder Tony Andrews. “This delivers two key objectives: exceptional

audio performance and user-friendly functionality.”

In terms of practicality, Andrews is able to reel off a long list of pluspoints for the new system. For a start, Vero is a “complete touring system”, comprising Vero speakers, flying system, amplification, cabling and a prediction software package named ‘Projection’, which allows users to achieve optimum array designs for smooth audience coverage and impact.

According to Andrews, Projection has been “designed to make the user experience as free-flowing as possible. The array-building function is straightforward to edit and manipulate.

LIVE FEATURE

www.audiomediainternational.com

Live load information allows for optimum rigging and alerts the user if setup exceeds the flyware’s limits.”

Another key feature, Geometric Energy Summation (GES), helps to eliminate the need for “inconvenient and expensive” delay positions up to distances of 1,000ft or more. “GES enables natural tailoring of coverage pattern and sound pressure levels to keep sound focused on the audience, controlling the off-site environmental concerns,” comments Andrews.

The Funktion-One mastermind also draws attention to the flying system, “which unlike many others, can be adjusted while under load” – and an amplifier rack, featuring Lab.gruppen PLM20k+ units – that is “equipped for the challenges and rigours of touring environments. It is fully shock-mounted,

meaning it protects the amplifiers and power distribution inside and is fitted with retractable doors that protect against the elements. The doors are conveniently stowed within the roof of the rack when the amplifiers are in use.”

This strong emphasis on userfriendliness even extends to the Vero transportation dolly. “It is completely bespoke and is designed for the challenges faced when working under the time-sensitive pressures of live event and touring productions and it self-aligns to the Vero boxes as they are landed. The 8in, solid polyurethane wheels don’t deform under pressure and are designed to carry the maximum load capacity over varied terrain. Its swivel bearing has been specially selected for its ability to handle high levels of sideways shock load.”

While Andrews recognises the pressure on manufacturers to optimise the cost-base, it is clear he believes that this needn’t be at the expense of quality.

“While Vero’s primary aim is to achieve the best sound quality possible, its intelligent design features ensure that the user experience is straightforward, making efficient use of tight production timeframes,” he says.

Another prominent loudspeaker system – Nexo’s STM (Scale Through Modularity) – speaks of a similar desire to balance versatility and feature set.

As international touring manager Stuart Kerrison recalls, the R&D process for STM was preceded by a lengthy process of consultation.

“Before we started designing anything, we spent over a year doing market research with leading international PA rental companies, asking them about their business operations and the challenges they were facing,” he says. “Regardless of territory, the answers were almost always the same – problems with diminishing income from rental, other factors such as video and lighting getting priority over audio on production budgets, and requirement to invest in multiple ‘application-specific’ line array systems to suit different venue sizes – often leading to expensive systems sitting in the warehouse costing money, waiting for the next suitable gig.”

The overriding response, he continues, was in favour of “a versatile system that

Other leading manufacturers consistently managing to balance the practicality and quality equation include QSC, whose new GXD Series processing amplifiers follow the trend for Class D amplification and deliver high power output from a 2U rack-mount configuration.

Sophisticated integrated DSP capability includes High and Low-Pass filters (24dB LR), fourband parametric equaliser, signal alignment delay, and RMS/Peak speaker protection limiting. To allow users to get up and running quickly, 20 preset ‘starting points’ for selected typical systems are provided.

Over at L-Acoustics, weight and ease of use have been near the top of the priority list for product developments including the K2 variable curvature WST line source system. “Efficiency – both in terms of power resources and human resources – is at the heart of the R&D process at L-Acoustics,” says L-Acoustics applications engineer could provide outstanding performance in all sizes of venue. At the same time, system riggers wanted a system they could rig alone without having to involve local hands in the critical operations of flying a PA.” STM is the painstaking result of these deliberations. “The ‘one-man rigging’ system enables one person to carry out all the technical aspects of rigging a system from one position, alone, at the rear of the cluster, regardless of system size,” comments Kerrison. “The system is also fast to rig up and down, saving time and, therefore, money.”

Based around four products, the modular design also means that “a touring production can go from stadium or arena gigs to a one-off promo gig in a club, and use exactly the same cabinets, amps, rigging, etc. There is no need to sub-hire a separate small system for the occasion, saving money and hassle.”

But Kerrison is at pains to highlight the fact that just as much effort has gone into the inner workings of the STM boxes.

“It is important to emphasise that in addition to these innovative rigging and trucking features, an enormous amount of innovation in cabinet and driver design was introduced in the design of STM to ensure

electronics Marc Bénard. “You see it with the K2, which makes no sacrifices in terms of power and quality, but comes in at a scant 52kg and can be rigged in no time. It’s no secret that the weight and ease of setup are what makes the K2 one of the most rider-friendly systems on the planet.”

Helping companies to “reduce freighting costs is absolutely a factor” behind the development of new Lab.gruppen amps, confirms product manager Martin Andersson. In the same mould as the PLM 20000Q, but with “twice the processing power, twice the throughput, and a whole host of additional features and improvements”, the PLM+ range features two models: the PLM20K44 and the PLM12K44. Each model sports four analogue inputs, four AES inputs, four power outputs, four Lake Contour modules, an 8×8 Dante I/O, a two-port 1GB network switch, dual redundancy and more. that audio quality is never compromised in order to achieve practicality,” he affirms.

ROBUST AND RELIABLE With standalone power amplifier makers feeling the squeeze in any case because of the growing popularity of self-powered speakers, it is to be expected that they have had to think more carefully than ever before about their offer simply to stand a hope of retaining market share. One way in which this has manifested itself is increased integration of onboard DSP. Dave Millard, managing director of UK-based manufacturer Full Fat Audio, observes: “We are driven by customer demand and close relationships to distributors, dealers and end-users.

Most amplifier manufacturers now offer onboard processing of one type or another, and have done so for quite a few years now. It is certainly a requirement to now offer a complete package.”

In terms of FFA’s own product line in this regard, Millard pinpoints the FFA6004 amplifier (4 x 1,500W 4 ohms) as “the key product. [Moreover] our DSP has features that other manufacturers don’t; for example, we offer extra DSP channels to allow connections to other amplifiers without onboard DSP.” March 2015

27

LIVE FEATURE

www.audiomediainternational.com

The Geo M6 augments Nexo’s STM series He refutes the suggestion that portability is a defining factor, however, “as the amplifier technology utilised by the majority of manufacturers has at the moment stabilised in the physical sense; most amplifiers are roughly the same size and weight now, whereas 10 years ago some amps could be four times the weight and size for one amplifier of a quarter the power output”.

Of course, integrating DSP inside the amplifier “does save the cost of a second enclosure to house the electronics”.

Powersoft is another amplifier maker to have acknowledged the trend for powerful onboard DSP with products including the X Series, which among other features delivers support for Audinate’s Dante media networking technology.

For Powersoft sales and marketing director Luca Giorgi, the advent of highly efficient Class D amplifier topologies are among the developments to be helping manufacturers balance cost, practicality and quality considerations.

“The raison d’être of the pro-audio industry is to provide a high level of sonic quality and reliability,” he says. “Therefore cost reduction can only be applied up to a certain level before being detrimental to the overall quality of the product and its performance. Maintaining the right balance between cost and performance is not an easy task, but Class D amplification and switching mode technologies embraced and pioneered by Powersoft 20 years ago have helped to reach this goal.”

Over at Yamaha Commercial Audio, Karl Christmas also emphasises the 28

March 2015

and Yamaha Commercial Audio products are legendary for being able to withstand the rigours of the road.”

“Quality cannot be compromised as competition is high because there are more manufacturers now.”

Dave Millard, Full Fat Audio

advantages of DSP-enabled amplifiers such as its TXn models – “along with saving rack space and cabling, the TXn series allows for precise remote control and monitoring [that is] unimaginable with conventional amplifiers” – as well as their inclusion of a mini-YGDAI card slot “which allows them to be configured with literally any kind of system”.

Christmas also points to Yamaha’s reputation for reliability as being a potent aspect of the overall sales brew.

“Of course, one of the most important features of touring amplifiers is reliability,” he says. “Whether indoor or outdoor, touring is a punishing environment for electrical equipment,

(EVERMORE) COMPACT CONSOLES For fixed applications, smaller consoles spell increased audience capacity (and therefore revenue); on the road, they mean reduced space in the van and thus potentially lower transportation costs.

The overriding trend for console design this past decade, then, has been for more compact desks – a development facilitated by the arrival of digital technology that has also paved the way for sophisticated onboard effects and the gradual reduction in need for outboard gear.

Allen & Heath’s Qu-Pac digital mixer occupies a space at the crossroads of these various requirements. An ultra-compact mixer with both a built-in touchscreen and iPad control app, Qu-Pac is intended to suit live music and installed sound in environments including colleges, hotels and bars. It offers 16 mono inputs, three stereo inputs and 12 mix outputs on the rear panel, but that can be expanded up to 38-in/28-out by connecting to Allen & Heath’s family of remote AudioRacks over Cat5.

Features that demonstrate the product’s flexibility include total recall of settings and preamps, 18-track recording to USB via Qu-Drive, a choice of personal monitoring solutions, channel ducking, multichannel USB streaming, and the iLive FX Library.

Allen & Heath senior product manager

Nicola Beretta confirms the desire to offer a highly portable solution suitable for touring acts including those “who are doing a lot of fly-in shows”. Also in play was an awareness that operator skillsets can now vary dramatically, particularly in the house of worship market.

“With more and more bands investing in their own equipment and the volunteerdriven house of worship market, we always need to be conscious of the need to balance ease-of-use with providing comprehensive features our more experienced customers require,” says Beretta. “Built-in touchscreens are a great blessing for newcomers to mixing – a sea of knobs or quirky navigation keys and trackballs are unfamiliar and intimidating, but we’ve found that the smartphone and tablet generation quickly feels at home with well-designed touch interfaces. It’s an inherently intuitive way to work.”

QUALITY WILL OUT…?

Versatility, variable skill-sets, portability, practicality… there’s no doubt that it’s a heady brew of requirements that now confront R&D teams when they sit down to begin deliberations about possible new products. But Nexo’s Kerrison speaks for many in observing that, when all is said and done, any manufacturer who undercuts on quality will almost certainly be consigned to diminishing returns in the long-run.

“It is possible that some manufacturers may have sacrificed certain areas of system performance to achieve tight goals regarding weight and size of cabinet, but at the end of the day I don’t think you can get away with that kind of thing for too long,” he says. “A sound system will always be ultimately judged on how it sounds, particularly by the engineers who have to mix on it and whose next job relies on the quality of sound they are able to produce. If a system doesn’t perform adequately sound-wise, it will have a hard time becoming accepted – no matter how practical it is.” www.allen-heath.com www.fullfataudio.com www.funktion-one.com www.l-acoustics.com www.labgruppen.com www.nexo-sa.com www.powersoft-audio.com www.qsc.com www.yamahaproaudio.com

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TECHNOLOGY FOCUS: STUDIO MICROPHONES

Expert Witness

A

s an industry we stand on the cusp of some incredible technology changes. For the past 20 years it feels that more recreation or emulation products have been released than truly new and unique products. It is rare that a new classic is born, however mics like the Royer R-121, Mohave MA-200, and Josephson e22S are increasingly common in studios around the world and are becoming invaluable.

Recently manufacturers are turning to new technology, including digital conversion within the mic, or microphone modelling such as the Slate Virtual Microphone System, that could provide engineers with a freedom that we haven’t seen before (when will these geniuses start to work on decent mic stands?). At this stage it’s just beginning, and I still rely on the tried and tested microphone forms of the last 70 years, but I have no doubt that soon we will see changes that will revolutionise how we record and what microphones we use.

Vintage – The good and the bad There is a real exaltation of vintage equipment in the audio industry. I’ve been incredibly fortunate in my career to use several of the highly prized microphone treasures we all chase after. Some of them have provided me with moments of magic and, at times, pure frustration. Last month I was producing three tracks for a band, and on the first day, I placed a pair of original M49 Neumanns to be the room mics. It sounded great and they brought an amazing warmth to the room, with super smooth high frequency response as well. However, after lunch on day two, we started hearing an increase in noise on the left room mic. Within an hour, that microphone was unusable due to the noise (which turned out to be a power supply problem). Even though it had been kept in great condition, using anything that is over 50 years old brings an inherent risk. It was magic, and it was frustrating.

There are certainly microphones that have changed over time, where the

Producer and engineer Stephen Bartlett steps into the vintage versus modern debate and looks at how the choice of microphone can affect a recording session.

modern counterparts do not reflect the microphone’s original design. Neumann’s U67 and U47 (Tube and FET) are obvious examples. There is a certain magic in both of these mics that I haven’t found in any recreation. Neumann’s U87 has continued to be built for many decades, however place an early 1980s version next to a current Ai, and there will be two completely different sounds. Other mics, like AKG’s C414, have changed as well (originally it came with the C12 capsule) and has seen several variations. They are different, and while I do love the EB versions, I would happily use any of them, in slightly different situations and methods. EV’s RE20 and Sennheiser’s Md421/Md441 have all been manufactured continually for over 40 years, and without hesitation I will use any of their variations.

Choosing a microphone Choosing a microphone is one of the most important parts of setting up a recording session, and planning starts well before the session does. As an engineer, I’m often trying to create a mix from the first step of recording, which influences my microphone choices.

Size within a mix can often be determined by the stereo width of the miking technique used. A piano or acoustic guitar that features in a song with the majority of harmonic content could benefit from X/Y or M/S or Blumlein miking to give it size and space in the mix.

If a song is likely to be mixed to surround, then I will employ five-point miking on certain instruments, because using space is always easier and nicer than creating it.

Conversely a tambourine that is just in the chorus will only have a single microphone.

I find that depth within a mix can be hugely aided by mic selection as well. As Michael Stavrou suggests in his book, Mixing with Your Mind, microphones have an inherent ‘hardness’ factor. A ribbon mic would usually be quite ‘soft,’ dynamic mics are often in the middle, and condenser mics are usually ‘hard’. A hard mic would bring an instrument to the front of a mix, while a softer mic tends to place an

instrument further into a mix and blend it with other elements. Just as microphones have a ‘hardness’, so too do instrument/ vocal sources, and by blending between these elements you can find a way to control where an instrument sits. Using a ‘soft’ ribbon on a ‘hard’ tambourine will help blend it into a mix more. Simply changing microphones could help place an acoustic rhythm guitar into a mix, while the riff with the same guitar could be brought forward with a different set of mics (and perhaps a wider stereo field).

The pickup pattern of a microphone can also determine the ‘wetness’ or ‘dryness’ of a source. A direct mic like the Sennheiser Md441 can help keep a snare dry by being very direct in how it captures the source, while an AKG C414 placed further back on a wide cardioid captures more of the atmosphere around the snare, and thus makes it feel more ‘wet’.

Purchasing/Hiring When it comes to purchasing a microphone there are several important factors. Often people feel that they need a condenser microphone first, especially for vocals. But good condensers cost thousands, while cheap condensers often have inaccurate or hyped high frequency response and often-poor dynamic range and unmusical coloring of the source.

Good dynamic microphones are far more likely to give a high-quality recording at a dollar-for-dollar price comparison.

Another consideration is that when microphone selection is small for an entire recording, often one mic is used on several instruments. By using just one condenser

the high frequency range can become cluttered and a mix may contain far too much sonic information in one register.

Parts and company support is another factor when buying a microphone. A company that has been around for 40 years is more likely to supply you with a new shock mount in four years than a company that has just started by emulating a famous mic.

One problem with recording is that to be creative and flexible requires a lot of equipment, most of which doesn’t get used all of the time or even in every session, making the experience incredibly expensive. But to sacrifice this directly relates to a sacrifice in quality or creative options. I mix a lot of records recorded by people around the world, and last year this exact problem was becoming a frustration in my mixing. It was due to this that I conceived The Audio Hunt, which is expected to launch within a few months, and will create a platform for peer-topeer rental of microphones, instruments and recording gear.

Expert Witness Currently based at Wisseloord Studios in the Netherlands, producer/engineer Stephen Bartlett is the former owner of The Docking Station in Australia, where he won various industry awards. Since moving to Europe, Bartlett has gained four top 10 records, a top 10 single and a platinum certification.

March 2015

31

TECHNOLOGY FOCUS

STUDIO MICROPHONES A high-quality, dependable studio mic can quickly become a cornerstone in your recording arsenal. Here is a round-up of some of the latest contenders for your attention.

Audio-Techn nicca

Sontroniics

AT5045

Aria

The hand-built AT5045 large-diaphragm sideaddress condenser microphone is the latest addition to the company’s flagship 50 Series studio range.

It boasts Audio-Technica’s largest single diaphragm, allowing for fast transient response, low noise, high SPL handling (149dB SPL) and a dynamic range of 141dB. At just two microns thick and made from vapour-deposited gold it is designed for use on overheads, percussion, acoustic guitar, strings and other acoustic instruments requiring the capture of dynamic, detailed sound.

Designed by Sontronics founder Trevor Coley and developed with the input of several Abbey Road engineers, the Aria valve mic has a fixed cardioid pattern and features a -10dB pad, 75Hz filter and a European-made 12AX7 vacuum tube.

The Aria is designed to deliver smooth results on sung and spoken vocals, but also to give quality results on piano, cello and acoustic guitar.

Audio-Technica’s premier studio condenser instrument microphone Discrete components for optimised capsule performance Transformer-coupled output is optimised to provide smooth sonic character Hand built and inspected for 100% quality control Advanced internal shock mounting decouples the capsule from the microphone body

www.audio-technica.com

Neuman nn TLM 107

Valve condenser mic with fixed cardioid pattern Specially selected European 12AX7/ECC83 vacuum tube Includes shockmount, power supply, cables and flightcase

www.sontronics.com

The TLM 107 is a universal, standard-setting microphone for studio, broadcast and demanding home recording applications.

It has a modern, high-resolution sound transducer developed for exceptional impulse fidelity, able to capture the original sound without any coloration.

Multipattern large diaphragm microphone with five polar patterns: omni, wide cardioid, cardioid, hypercardioid and figure-of-eight Navigation switch with LED illuminated display for polar pattern, PAD and low cut Transparent sound characteristic for outstanding transient response to naturally reproduce human vocals Balanced sound in all five polar pattern settings

www.neumann.com

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March 2015

TECHNOLOGY FOCUS

www.audiomediainternational.com

DPA A d:dicate 4006 Omnidirectional Microphone

DPA’s d:dicate 4006 Omnidirectional Microphone is designed to deliver precise, natural and detailed sound across a wide range of instruments with total transparency. Ideal for A-B stereo pairs for symphonic concert hall recordings, the 4006 also promises excellence on vocals and closemiking of instruments like grand piano, guitar, double bass and percussion.

The frequency response and directional characteristics can also be transformed by using its acoustic modification accessories, including the nose cone and interchangeable protection grids, which won’t produce phase shift, noise or distortion.

Acoustically configurable via a selection of modification accessories 147dB max SPL before clipping 124dB dynamic range Meets professional recording standards

Sennheisser MK8 The true condenser, double-diaphragm MK8 features five selectable polar patterns (omni-directional, wide cardioid, cardioid, super-cardioid and figure-of-eight) to adapt to the recording situation at hand. The five switchable pick-up patterns exhibit relatively constant characteristics, especially in the low end, to ensure more defined sound in challenging recording environments as the selected pattern is maintained over frequency; patterns will remain precise, providing the same high attenuation of lateral sound sources as originally desired.

Five selectable polar patterns: omni-directional, wide cardioid, cardioid, supercardioid and figure-of-eight Fitted with a low cut/roll-off filter and pad switch Features a low sensitivity to humidity as the impedance conversion is done within the capsule Sturdy metal housing

www.sennheiser.com

www.dpamicrophones.com

Shurre Beta 181 The Beta 181 is a multipurpose instrument condenser microphone that can be used for a wide variety of applications including guitars, bass and drums (overheads and snares). A total of four different capsules (cardioid, supercardioid, bidirectional and omni-directional) are available, allowing users to interchange them for different polar patterns depending on the situation.

Choose from four different capsules: cardioid, supercardioid, bidirectional and omni Consistent polar responses Versatile – for use on a wide range of instruments Tiny form factor – small enough to get close to the source in the tightest conditions

Samso on MTR101A Studio Mic Kit Ideal for use as a room microphone, for recording vocals, close miking instruments and more, the MTR101A offers a smooth, extended frequency response, as well as low self-noise and a wide dynamic range for recording.

Kit includes 1in diaphragm studio condenser microphone, pop filter and shockmount Cardioid pickup pattern for increased isolation Low self-noise, wide dynamic range for recording Handles SPLs of up to 137dB Internally shock-mounted capsule reduces handling noise and external vibrations

www.samsontech.com

www.shure.co.uk

March 2015

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TECHNOLOGY FOCUS

RØDE NTR Active Ribbon Microphone Sonically designed to reproduce high frequency detail and accuracy, the NTR is equipped with a custom transformer and electronics that enable use with a wide range of preamps, without the additional gain many other ribbon microphones require.

Highly sensitive laser-etched 1.8 micron ribbon element Custom transformer and electronics for high output No external shock mount required

www.rode.com

www.audiomediainternational.com

Beyerdynam mic M 160 Ribbon Mic The M 160 is a dynamic double ribbon microphone featuring a hypercardioid polar pattern with a noise attenuation of 25dB at 110°. It is designed for all types of remote and in-studio voice reproduction. In the recording studio, the M 160 is recommended for miking string instruments and for pianos, saxophones, hi-hats and toms. In conjunction with the M 130 dynamic double ribbon microphone the M 160 is ideal for stereo recordings done using the MS technique.

Unique double ribbon microphone transducer Hypercardioid polar pattern Extended frequency response Excellent transparency and transient response

www.beyerdynamic.com

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March 2015

TECHNOLOGY REVIEW

AUDIO-TECHNICA ATH-M70X HEADPHONES

S

ome gear is born great – Neumann microphones, say, or Nagra recorders. You expect great things of a new Neumann mic or a Nagra recorder. But other gear from maybe less fashionable manufacturers has to fight tooth and nail to establish the credentials required for a seat at the top table.

The Audio-Technica ATH-M50 headphones belong very firmly in the latter category. I reviewed them first in 2009 and was immediately struck by their all-round brilliance. Accurate, comfortable, solidly made and reliable, foldable and their closed back design meant a wide range of applications – and all that audio goodness at a price that was a steal. I used the M50s for mixing, recording and location work, as well as the general day-to-day duty of leisure listening. My original pair is still doing sterling service at my son’s house, my only criticism was that I wore out the headband – well, the cover – which now adds interesting flakes of black to your coiffure; it looks fine on him, with hair like a badger, but it’s not a look that works for balding middle aged men.

The M50s, of course, are still available now in the M50x guise and some 50x features have made it over to the new kings of the Audio-Technica headphone hill. Yes, having established themselves as headphone gurus AudioTechnica have come back to the table with a new top-of-the-range model.

Well, actually a new pair of top-of-therange models. The ATH-R70x is the open backed offering, while I’m reviewing the closed-back ATH-M70x, which is very much in the lineage of the M50.

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A fan of the original ATH-M50s, Alistair McGhee was understandably keen to get his hands on Audio-Technica’s brand new flagship M-Series headphones. And, as it turns out, he wasn’t disappointed. coming from the in-house M50x, not to mention the rest of the high-end headphone world?

Opening the ATH-M70x box reveals a very jolly, media black neoprene case, emblazoned with the A-T logo. Inside you’ll find your cans with the ear pieces rotated through 90º for storage and a mysterious little black plastic case.

Physically if you are familiar with the M50 then there are no surprises, or maybe just a couple. The look is black and silver (no fancy colour versions available at the moment), but the yokes that hold the earpieces are now metal, as are the headband end pieces. A pair of my favourite Sennheisers died of broken yoke so I’m all in favour of more ruggedness at this point.

Unlike the M50s the M70x cannot do the neat hinged folding thing that reduces their footprint by about half.

On the other hand you do get a nice travel case.

One thing I nearly forgot to mention was that the promotional literature claims that the headband and the earpads are easily replaceable – so if in five years you do get headband shed, it should be fixable. Nice touch.

The mysterious smaller bag is home to three locking cables for your new headphones: one 4ft cable terminating in 3.5mm jack; the classic 9ft coily cable complete with 3.5mm to quarter-inch jack screw on converter; and finally a mighty 9ft straight cable to 3.5mm jack. All the cables feel good in the hand and are well made. Joy of joys, these cables mate with the left hand earpiece of the M70x with a reassuringly firm bayonet twist lock, a design borrowed from the 50x. Cable management then is excellent in terms of ease-of-use and security. But you won’t be able to use

your custom headphone cables if you are into that sort of thing.

The new design hosts some 45mm drivers, with all the usual stuff about copper clad cabling and rare earth magnets. Actually I don’t think ordinary earth magnets would work! The point is there’s some clever stuff inside.

In use Now the listening. The first thing that struck me was something rather odd – the M70x dries up acoustics. Listening carefully I think this reflects the underlying accuracy of the headphones.

Where you might get a pleasant warmth or bloom on a recording with lesser

TECHNOLOGY REVIEW transducers the M70x is going to get you that bit closer to what is on the record.

Remember records?

The second thing I noticed was the nimble bass – tuneful, well controlled and cleanly defined. This is an area in which manufacturers really walk the line.

I’m willing to bet that shoving a nice bass boost shelf into the response will make a pair of headphones more instantly appealing – especially easy to do with a closed back design. Reviews full of descriptions of bass slam and low-end authority have to be good for business.

But of course what works when impressing your friends with a Decaf mix after six pints of lager and a chicken bhuna, might lead to terrible mixing errors when you’re trying to establish your rhythm section with a degree of accuracy that transcends bombay bass.

Looking back at my very first notes I can see I wrote ‘hf – bright?’ Well, having lived with the Audio-Technicas for a solid week I would say, I don’t think so. I’m usually pretty sensitive to over-egged top ends and any shoutiness in the mid range. Even though, again, this can make an audio product stand out in the short term. A-T claims the drivers work up to 40kHz. This is not a claim I’m in a position to refute, but I found the top end detailed and delicate but certainly not backward in coming forward.

One of the main reasons you spend more money on headphones is to get more! More detail, more dynamics, more bandwidth, more of everything really. If you spend twice the money you’ll be wise enough to know that you won’t get twice as much of everything but there should be clear audible advantages. So where’s the extra with the M70x?

Well how about more vocals? Not more level on vocals but more tracks.

With the M70x I could distinguish double-tracked vocals, where on lesser cans there was a vocal ‘smear’ – now resolved and the mix revealed. One step closer to the master tape. Remember tape? Well if you don’t beware the M70x, the extended top end will not hide the tape hiss, or mic amp hiss for that matter. You have been warned. I even picked some hum on recordings that I hadn’t really been aware of before.

All of these are the technical aspect of the M70x’s performance and they are

Tom Harrold, marketing manager EMEA at Audio-Technica, outlines some of the company’s main considerations when designing the ATH-M70x headphones.

How does the ATH-M70x provide a step up in terms of sound quality compared to the existing models in the M Series?

The M70x is an exceptionally accurate pair of headphones, designed for highlevel, critical monitoring duties and so one of the areas Audio-Technica has worked hard on is the reproduction of extreme low and high frequencies.

The frequency response of the M70x is wider than any of the other M Series models, at 5-40,000Hz, to reveal all the details of a mix, while also remaining well-balanced and ‘listenable’.

components to ensure longevity – and design touches like the robust, wellpadded headband, generous ‘around the ear’ pads (which are detachable), 90º swiveling earcups and detachable cables all contribute to excellent wearability and durability.

M70x’s super-accurate sound has gone down very well with them. There’s always some pressure when launching a product that has to uphold (and extend) the reputation of something so well loved as the M50x, but I’m very happy with the reception they’ve had.

What sort of feedback have you had from professional users so far?

The reception has been really great so far – we’ve been delighted with the response. Obviously a lot of our pro users are fans of the M50x and the

We’ve seen quite a number of new headphones from Audio-Technica released recently. Can we expect any further additions in the near future or is the company satisfied with its current offering?

Audio-Technica is very driven by design innovation and as a company we’re constantly looking to improve products and offer better solutions for customers.

So while we’re very happy with the new ATH-M70x – and indeed the new open-back R70x which launched at NAMM 2015 in January too – I can confidently predict there will be developments in the future. Although you’ll understand I can’t say more than that at this point!

What about build quality and comfort? How important were these factors during the design phase?

These are pro-oriented headphones, so obviously they need to stand up to the rigours of a working life – either out on the road or in the studio – and also offer great comfort over long tracking sessions etc.

The M70x features metal for key

“The Audio-Technica ATHM70x are a fine pair of headphones – an excellent combination of monitor neutrality and insightful musicality. And a bit of a bargain. Another A-T hit.”

Alistair McGhee

important and you might buy a pair just based on the A-T’s strengths in these areas. However, there is a different but surely related quality, which for want of a better word I’ll call musicality. That is the ability of equipment to allow insight into the original musical intention of the artists. Obviously gross failings in the technical performance will probably render musicality moot but good gear provides access to the inner logic of the art of music.

For example – and one that is available in every good record collection – spin up

a great Stax classic like Private Number, enjoy the hiss under Cropper’s intro, feel the insistence of the bass and the boldness of the brass – tumble, swoop and fall with the strings, maybe a tear wells up with the swell of Judy Clay’s vocal and a smile at the conversation between Cropper and Duck. And we haven’t even mentioned Al Jackson Jr or William Bell. If it sounds like a record, even a good one – you need better gear, gear with musicality.

A bit of perspective – my work-day headphones (in the absence of the M50s!) are Sennheiser HD 25s. The Audio-Technicas are over twice the price and are comfortably worth the extra. My reference headphones are Ultrasone Signature Pros – they are over twice the price of the M70xs and they’re better again. But value at that point is a deal struck between your ears and your wallet. Listen and reckon.

In conclusion the Audio-Technica ATH-M70x are a fine pair of headphones – an excellent combination of monitor neutrality and insightful musicality. And a bit of a bargain. Another A-T hit.

Key Features Closed-back 45mm driver diameter Frequency Response: 5-40,000Hz Maximum input power: 2,000mW at 1kHz Sensitivity: 97dB Impedance: 35 ohms Weight: 280g without cable and connector RRP: £299.00 www.audio-technica.com

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.

March 2015

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TECHNOLOGY REVIEW

SONARWORKS HD REFERENCE PLUG-IN HEADPHONE CALIBRATION SOFTWARE

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easuring and calibrating speaker systems, particularly studio monitors, is an area of science and technology that has seen many advancements and differences of opinions.

As engineers we talk about the theoretical holy grail, which we call a ‘flat frequency response’. The principle of working in an environment that has a flat frequency response, so that our judgements aren’t influenced by anything other than the source, is a widely accepted practice. However, achieving this known ‘reference’ sound is not only extremely difficult in the real world, but it is heavily debated as to what exactly is a flat frequency response.

Assuming we have agreed upon a flat frequency response and managed to electronically capture and measure a device, how do you go about making corrections and adjustments? As we know, making adjustments can introduce secondary issues.

The Technology Sonarworks won’t go into detail about its measurement technology or its ideas around a ‘human’ flat frequency response, which isn’t surprising. However, its approach and choice of language is certainly on the right lines. The company already addresses each debated area of monitor calibration using its own methods with much success, and now it has found a way to bring the same principles to headphone users.

Firstly, Sonarworks claims that it has heavily invested in developing technology that measures the 20Hz to 20kHz bandwidth in a way that is closer to a human ear, unlike a measurement microphone. The firm states that its method provides more realistic results.

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Sonarworks calls its method PAPFR (Perceived Acoustic Power Frequency Response) and makes its measurements at a normal listening volume of 83dB SPL. While discussing its measuring technology with marketing manager Rūdolfs Putniņš, he explained: “The problem lies in the fact that currently there are no set standards for how to do headphone FR measurements. All of the head-microphone setups are made to measure SPL, not FR, so we had to develop our own, which we feel does a far better job”.

The company has also thought about the flat frequency response, and uses the explanation: “The default flat frequency response reference target is made to match the frequency response of flat sounding speakers, ensuring smooth translation between working on studio monitors and headphones.” Although there is no clue as to what that might be, at least we can have the confidence that this is more than just an EQ service based on theory.

So, if Sonarworks has managed to take a ‘realistic’ measurement and thought about a desired goal, what about a calibration method? In the same vein as its approach to measuring sound, the company has invested in high level mathematics and an ‘adjustable filter engine’.

This is my favourite part about the UI side of this plug-in. Due to the highresolution EQ curve, the user can select the level of resolution and the amount of phase distortion within the EQ, to offset against latency and processing power. For example, if monitoring on headphones in a recording or performing scenario, then low-resolution EQ and a minimal phase relationship will allow for lower latency. Whatever the amount of processing you use, the plug-in will give

Interest in the Latvian firm’s new headphone calibration software is starting to build, but is it really something to get excited about? Simon Allen delivers his verdict.

you a gauge and a reading of the latency you are incurring.

The Plug-in Method The Sonarworks HD Reference plug-in comes as AU, AAX Native, RTAS or VST. The principle is to run the plug-in over your headphone output from within any DAW. From there, you simply load the appropriate headphone profile file from Sonarworks. Once loaded you are able to hear the before and after of the adjustments made, and choose between linear, mixed or minimum phase filters to suit the task at hand.

The plug-in works well and doesn’t draw too much processing power. Other considerations have been taken, such as the output level post plug-in – you can choose to either monitor at a lower level so that the plug-in’s processing will never peak the output, or limit the output to avoid clipping and remain at the same audible level.

Once the plug-in knows which headphones you are using and is loaded with the correct calibration file,

Sonarworks has taken this one step further by providing other targets.

Besides the obvious flat frequency response, there is a selection of other devices it can simulate. For example, you can hear what your mix/master might sound like on a pair of NS10s, or a pair of Beats by Dre. Although alarming, I did find the simulations realistic.

However, there is one drawback at the moment. As this is a new product, there are a very limited number of headphone models for which Sonarworks has measured and released calibration files.

Currently, Sonarworks is selling three options of headphones at varying price points with the calibration plug-in. Other calibration files exist, but are few and far between. However, if your headphones aren’t listed, then Sonarworks says you can send them to its HQ, and for only €20, they will measure, create a calibration file and return them.

Listening Experience Rūdolfs sent me a pair of Sony MDR7506 headphones with the plug-in and

www.audiomediainternational.com

calibration file to try out. I’ve used 7506s many times and already knew that they were far from my idea of ‘flat’, so thought it would be a good test. It was a shame that at the time of this review, they didn’t have profiles for some other headphones I know and use more regularly.

On first listening, I was very impressed.

In fact, possibly like most engineers, I spent some time simply listening to different types of music comparing, with and without the calibration, to try to appreciate the changes. I was taken aback, the sound is vastly different with the calibration on, and it really did feel like listening to flat response studio monitors. I always knew the 7506s were very capable headphones, but now I felt I could actually do some preliminary mixing on them. If I had to be fussy, the bass response was a little too much for my taste, but easily learnt.

The issue now is that I feel I need to hear another pair of headphones from a different manufacturer with the Sonarworks calibration running. I want to know if they have a consistency for

measuring headphones and if they really are able to pull nearly all headphones in-line with one another. After all, besides offering a solution to calibrate your headphones, this is a solution that is trying to bring that ‘reference’ sound across the board. I guess I will have to wait for them to continue to bring out calibration files but I am both intrigued and excited.

My verdict from hearing the response on the 7506s was better than I expected.

Even if it isn’t 100% correct to everyone’s ears, it is an extremely good attempt, and a reference worth learning. What the tech does is achieve a response that is transportable and in line with working on studio monitors.

Conclusion When I first heard about this calibration plug-in, I did wonder if it might have just been a simple EQ curve which was electronically calculated. Although I’m still not clear about the methodology here, it is apparent from the short explanations and what I can hear, that

TECHNOLOGY REVIEW this is a very serious solution. It is a hard area of the market to venture into, which is why there isn’t really anything else out there quite like this. I highly commend Sonarworks for its thoughtful approach and long-term efforts in providing us with a tool that opens up possibilities on a higher level.

Besides the continual growth of supported headphones there are a

Key Features Currently supported headphone models: AKG K141 MKII, K701 and K712 Pro; Sennheiser HD598, HD600 and HD650; Sony MDR-7506; Superlux HD681 Available in AU, AAX Native, RTAS or VST formats Precision above +/- 0.5dB across all correctable frequency range Simulate different audio systems on your headphones to check how your mixes translate to other systems RRP: €69/$79. Extra licences €20/$26.36 www.sonarworks.com

couple of other features I’d like to see incorporated in future releases.

Firstly, I would like to have the ability to cross-feed the left and right signals. This would add to the simulation of working on monitors in the studio. Secondly, this calibration method only runs as a plug-in hosted by your DAW. It would be an obvious move to develop an app for smartphones, tablets and computers, etc.

For example, to listen to anything, you must currently run the audio from within your DAW, which isn’t ideal if you want to quickly hear a mix sent by a colleague, or a production idea from a co-producer.

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 grow.

March 2015

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TECHNOLOGY REVIEW

FOOTE CONTROL SYSTEMS P4S ME MASTERING COMPRESSOR

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’ve been a fan of US-based Foote Control Systems’ products for a while – checking back for this review I was surprised to discover I’ve now owned their excellent P3S ME mastering compressor for three years.

In that time it’s been in regular use for its clean signal path with some colour if required, helped by a Class A/ transformer upgrade. Its stealthy style of compression, where it might not appear to be doing a great deal until bypassing it, is for me unsurpassed. I therefore sat up and took notice when Roger Foote recently announced the advent of the P4S ME, also aimed at the mastering process but with a more ‘vintage’ sound, a superior detector and the use of premium parts (and pricing).

Overview The Foote Control Systems P4S ME is a sturdily built unit occupying two rack spaces and measuring 205mm (8in) deep. The engraved turquoise front panel of the review unit – other colours, including the more usual black and grey, are available – sports six rotary controls, three pushbuttons and a gain reduction meter. From left to right the rotaries and their ranges are: Attack (0.15 to 15ms per dB), Release (15 to 150ms per dB), Threshold (-8 to +15dB), Ratio (1:1 to 4:1), Gain (0 to +11.5dB) and a high-pass sidechain filter with rolloff frequencies at 240, 150, 120m 80 and 50Hz, plus ‘Off’. The first surprise is that all these controls are high-quality Elma switches with multiple positions rather than continuously rotating like a standard potentiometer – while adding significantly to the cost, this approach offers first-class resetability, a feature that’s especially useful in mastering.

The three pushbuttons are firstly A/R, which toggles between RMS mode 40

March 2015

Nigel Palmer finds this new premium offering from the American manufacturer adds versatility to the mastering process.

(an auto time constant mimicking the way we hear) and enabling the Attack and Release controls (good for more specifically set gain reduction, as when you want the music to ‘bounce’ in time).

Next up is Feedback, offering either older-style feedback compression that tends to ‘glue’ more, or the more modern feed-forward method with a tighter feel. The last button is a true hardwire Bypass.

The gain reduction meter is a responsive instrument measuring up to 5dB of GR in 0.25dB increments. I couldn’t hear compression occurring when there was no meter indication – a useful feature in a busy working environment and something I’ve found to be fairly unusual. Finally, a look around the back reveals a spartan rear panel with a multi-pin socket for the switched-mode power supply (the unit passes audio when switched off), I/O on XLRs and jack sockets for external sidechain access, such as with an EQ to tailor the frequency response of the detector for de-essing and other functions.

In Use After a day or so of experimentation I let the P4S ME loose on some realworld projects. The unit is stereo, so single-ganged controls affect both sides; that’s my preferred way of working as it’s quicker, and although it does happen I can’t remember the last time I needed different Left and Right compression settings. A second surprise was immediately apparent in that, unlike the P3S ME, you can’t bypass the large Cinemag transformers – they’re apparently the reason the box had to be 2U – and neither would I want to. Without ever being too colored for my taste, the sound is rich and wide

with just a hint of an old-school feel and offering commendable depth of field and a big low end response – in the latter case, it’s not so much that the Foote adds LF, it’s more that it preserves it in a most attractive way.

While capable of being very transparent at low gain reduction levels, especially in RMS mode, the unit also benefits on occasion from being made to work harder and producing more attitude – the knee used is harder than in its stable mate, but even so I found that more gain reduction than I would typically use in mastering could be useful under the right circumstances.

A standout feature, especially for dance and bass-heavy music, is the high-pass filter’s ability to operate at different frequencies. The 80Hz setting made a good start point, but depending on the material I found a use for all the others, including bypassing it altogether on those occasions when I wanted the low frequencies to deliberately pump the whole track.

The Attack and Release controls were speedily set for tasks such as allowing more or less transients through, although much of the time I found RMS mode in combination with feedforward compression did what I needed, making this one of the easier compressors to operate that I’ve seen. The P4S ME was at home with any musical style I tried it on, with notably good results in the acoustic and electronic genres that can quickly reveal compression deficiencies.

Conclusion While I’m probably among the first to say that compression in mastering is often overrated, I think the Foote P4S ME mastering compressor is an unusually good and versatile unit of its type. Easy and fast in operation but with the potential for great results on a wide range of music, it’s worthy of serious consideration by anyone (and not only mastering engineers) looking for a high-end compressor. The entry price may be relatively high, but in my opinion it’s a not unreasonable exchange for the generously classy performance you get in return.

Key Features RMS or manual peak (attack/release) modes Feedforward and feedback modes Cinemag input and output transformers Eight VCA, four per channel SL2520 op amps Introductory pricing: : $4,750 www.footecontrolsystems.com

The Reviewer Nigel Palmer has been a freelance sound engineer and producer for over 20 years.

He runs his CD mastering business Lowland Masters (www.lowlandmasters.com) from rural Essex.

TECHNOLOGY REVIEW

ZOOM H5 HANDHELD RECORDER

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s a live sound engineer I’m often asked to make reference recordings of live gigs for the scrutiny of the musicians involved.

While the purpose is a quick and simple recording I’ve never been fully satisfied with the results from either recording the desk mix or capturing the ambient sound using microphones. I got good results using a four-track to combine the desk mix with ambient microphones, but this proved too unwieldy to take on the road.

So I was quite excited in 2005 when I heard news of the Zoom H4 handheld four-track recorder. Unfortunately that first model was unable to record four channels simultaneously so I had to wait until the H4N (the ‘N’ denoting the ‘next’ version) in 2009 to attain the functionality I craved – and I’ve used mine to record hundreds of gigs since.

Now we have the H5, which seeks to improve on the H4N, and brings some of the functionality of the top-of-the-range H6, so let’s see how it stacks up.

My first impression is that it’s slimmer, taller and slightly lighter than the H4N. At the base it has the familiar pair of XLR/TRS combo inputs (which can provide phantom power and a 20dB pad) and the front panel features the transport controls, four input selector buttons (and indicators), input volume controls (protected from accidental knocking by a metal bar) and the display screen. The most obvious departure from previous models is the detachable mic capsule (a feature borrowed from the H6). By default it comes with the XYH-5 X/Y microphone capsule, which features two matched unidirectional condenser microphones set at a 90º angle, as well as a protected volume control and a 3.5mm mic/line input jack (which supports plug-in power). There are four other capsules available: the MSH-6 mid/sides microphone capsule, the XYH6 X/Y microphone capsule (which closely resembles the H4N microphones with its 42

March 2015

Promising power and flexibility, the Zoom H5 claims to have set a new standard in portable recording, but does Andy Coules agree?

switchable 90/120º pick-up pattern), the SGH-6 shotgun microphone capsule and the EXH-6 dual XLR/TRS combo capsule – which gives two additional balanced inputs as well as separate volume controls and a pad. This provides a useful range of recording options, which can be swapped as quickly and easily as changing a camera lens.

The left side panel has a 3.5mm line out jack, a 3.5mm headphone jack, a volume rocker, mini USB socket and a power/hold slider switch. On the other side is the menu button, the scroll button, the remote control jack socket and the SD card slot. On the underside is the speaker, battery compartment and screw mounting hole. They’ve done away with the power supply input, opting instead to power it via USB (via an optional AC adapter), however a pair of AA alkaline batteries should last up to 15 hours, even when recording continuously. The USB socket can also be used to connect to a computer so the H5 can act either as a card reader or an audio interface.

In Use Basic operation is straightforward – you turn it on, select which inputs you want to record, set the level and press record, all of which is done using the front panel controls. The level meters start working as soon as you select the inputs and the protected volume controls make it easy to set the levels and get going.

The screen is compact but uncluttered with all the information you need readily available, such as available recording time, battery status, file name, level meters (from -48dB to 0) and the chosen recording format (which defaults to 16bit, 44.1kHz wav files). They’ve done away with the multi-track record mode and the effects unit, which, when used together, made the H4N a powerful multi-track recorder with various amp models and effects. My initial reaction is that this is a pity, but on reflection it’s sensible as very few people utilised this functionality.

You have a choice of two common file formats – MP3 and Wav. Wav files can be 16 or 24-bit with sampling rates of 44.1, 48 or 96kHz, although 96kHz is only available in stereo recording mode, as are MP3 files (which allow bit rates from 48k to 320kbps). Wav files are automatically time stamped and are thus Broadcast Wave Format (BWF) compliant. The H5 utilises SD and SDHC cards for storage and supports anything up to 32 gigabytes.

The included 2GB card can store about 14 hours of stereo MP3s at 320kbps, three hours of stereo 16-bit/44.1kHz or an hour of stereo 24-bit/96kHz Wavs.

The pre-record function, when enabled, is continuously recording such that it can capture the two seconds before you press the record button, as well as an auto record function that can be set to record when the volume exceeds a user-defined level (and stop when it drops below a set level).

And then there’s the potentially lifesaving Backup-Record feature, which automatically records a duplicate set of stereo tracks that are 12dB lower than the original pair – although this only works when recording stereo Wav files at 44.1 or 48kHz. In operation the recordings are clean and clear. The default X/Y microphone capsule includes rugged rubber shock mounts, which help isolate handling noise and the X/Y configuration enables the capturing of a detailed stereo image while ensuring centre sources are clear and well defined; it can also handle sound levels up to 140dB SPL. The line inputs use the same highquality preamp as the H6, and support professional line level input (i.e. +4dBu), which is an improvement on the H4N, whose balanced input was configured for -10dBV operation, which often meant you had to pad the output from professional gear (such as mixing desks) so as to not overload the inputs.

So overall it’s a very capable and straightforward portable recorder,

equally comfortable perched on a mixing desk, in the palm of your hand or sitting on top of a DSLR camera (via the optional hot shoe mount adapter). The interchangeable capsules provide new levels of versatility and flexibility and the lengthy battery life make the H5 a very worthy successor to the much-loved H4N and a powerful handheld recorder in it’s own right.

Key Features Five interchangeable mic capsules, including a detachable X/Y option Four-track simultaneous recording Large backlit LCD display Records directly to SD and SDHC cards up to 32GB Up to 24-bit/96kHz audio in BWFcompliant WAV or a variety of MP3 formats RRP: £219.zoom-na.com www.zoom-na.com

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

TECHNOLOGY REVIEW

CERWIN-VEGA P1000X P-SERIES LOUDSPEAKER

Strother Bullins considers how this portable PA compares with the company’s older offerings.

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he P-Series is one of two powered live loudspeaker lines by Cerwin-Vega, the other being C-V’s flagship CVA Series – quite possibly the best all-around PPA (portable PA) gear I’ve ever used.

The P-Series incorporates some notable CVA-influenced features, as well as some unique and useful touches.

For this review, I received a P1000X pair (at $699 street, each), a two-way biamped full-range bass-reflex speaker featuring a 10in LF driver and a 1.75in HF driver, polypropylene cabinet with steel accents and handles incorporating a flush waveguide and 18 gauge perforated steel grille; and a Class D amp providing 1,000W of peak power.

The P1000X is compact for what it offers, with dimensions of 25.5in x 15in x 15in and a weight of 37lbs. Frequency response is relatively flat and balanced for PPA gear, at a reported 61.0Hz to 20.6kHz, (±3dB), with a 128dB maximum SPL (1W @1m).

I/O for the three-channel mixer includes Neutrik XLR/TRS inputs on channels 1 and 2, switchable mic-DI or line plus an XLR throughput each; dual 0.25in TS line inputs on channel 3; and a mix output on XLR. EQ-based parameters include Enhanced EQ (contour adjustment with feedbackreducing midrange attenuation); Vega Bass Boost (LF gain); and a High Pass Filter (attenuation below 80Hz), each with on/off switches.

In use After several months of use in several applications – club/theatre, outdoor venue and rehearsal use – I found the P1000X to be a great stage monitor and main speaker (when used as a pair coupled with a powered subwoofer).

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Key Features System configuration: Active two-way full-range, bass reflex Frequency range (-10dB): 51Hz-22.5kHz Frequency response (±3dB): 61.0Hz20.6kHz Nominal coverage (-6dB): 90° horizontal, 65° vertical Maximum SPL (1W @ 1m): 128dB Amplifier type: Class D Amplifier rating: 1,000W dynamic Low frequency driver: 10in High frequency driver: 1.75in diaphragm, 1in exit Connectors: channels 1 & 2 – Neutrik XLR/TRS inputs, channel 3 Dual 0.25in TS THRU 1&2 – XLR, MIX XLR Indicators: Signal/Clip, Power, Limiter, Protect Enclosure: Polymer cabinet, flush waveguide Grille: 18 gauge, perforated steel Mounting & suspension: 35mm pole socket 6 x M10 suspension RRP: $699 It’s plenty powerful in most PPA applications and seems to be physically overbuilt in typical Cerwin-Vega fashion, with an overall better and thicker grade of polypropylene than I find in the vast majority of portable PA products.

However, if shopping for active portable PA gear today, I’d honestly choose an older Cerwin-Vega fullrange model – the more compact, tour grade-built Active Series CVA-28 dual 8in speaker (at $599 street, each,

with 800W peak power) over the P1000X; there are several products very comparable to the P1000X in the market yet nothing out there quite like the CVA Series.

That said, for those looking for a more traditional PPA powered enclosure with Cerwin-Vega’s well-earned reputation for quality sound and rugged construction, the P1000X is a very worthy choice and notably user-friendly to boot.

www.cerwinvega.com

The Reviewer Strother Bullins is reviews editor across NewBay Media’s AV/Pro Audio division.

He was also editor of sister title Pro Audio Review. www.prosoundnetwork.com

INTERVIEW

www.audiomediainternational.com

MAN OF THE MOMENT

A veteran of sound editing for the big screen, Ben Wilkins has worked up a formidable list of credits including Twister, 2 Fast 2 Furious, Terminator 3 and Star Trek, but he’s also been in the news lately for being part of the Oscar-winning Whiplash sound team. Matt Fellows quizzes him on the secrets to the film’s success.

W

hiplash tells the story of a talented young jazz drummer who studies under the caustic tutelage of an obsessively perfectionist music teacher, whose brutal methods he must endure if he is to achieve greatness.

The film recently scooped the Oscar for Achievement in Sound Mixing and a BAFTA for Best Sound; feats made even more impressive by the fact the sound team completed the work in just 10 days.

How did you approach a project with such a tight schedule?

Everything had to be prepped ahead of time with that short deadline in mind.

A great deal of levelling and panning went on inside a Pro Tools environment.

We had to eliminate the need to make decisions – we had to start narrowing our choices. We didn’t use a scattergun approach to sound effects and Foley – it was more a sniper approach to make everything as efficient as possible. There were a few sections that I did and sent to the music editor and the director to get signed off ahead of time, but for the most part we kept it very straight-ahead.

Is such an efficient and expedient approach one that you’ve employed in your earlier career?

Yes, it wasn’t the first time we’ve done this. It’s starting to become standard.

We’re definitely at the bleeding edge of technology as far as finishing films as quickly as possible is concerned.

Schedules are getting shorter and I’m not sure how much shorter they can sensibly get. We’re starting to come up against the physical limitations of how long it actually takes to record and play back a film in real time.

What did you find most challenging about this piece of work?

Getting sounds into the movie that weren’t drums. The drums were a character; the drummers were all 46

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reliable and familiar becomes a real asset because you just don’t have the luxury of testing and experimenting.

Also the Euphonix Series 5 Fusion, which allows you to both use the traditional mixing console and control Pro Tools, was invaluable and a real time-saver.

interacting with these very loud drum kits, and to get any sounds in under the music or around the music was the biggest challenge for me because the music represented these monolithic blocks of sound that I had to try to pierce and shape sounds out of.

The drums are the main focus of the film, so how did you go about capturing them?

The main philosophy was just to make them sound as real and accurate as possible without any coloration. The director had a very strong desire to connect with that visceral feeling of how deafeningly loud they can be when you’re sitting at the kit and playing them.

To achieve that we created impulse response recordings: we took loudspeakers with all the impulse response tones that were necessary out to the set and we captured fourchannel quad recordings of the impulse responses, which we then sent to be rendered into usable reverb programs.

What was your methodology for the wider film?

My approach was to enhance the storytelling and help the audience to emote and feel what the director wanted them to feel, but actually in this case, a lot of scenes were made more

powerful through the absence of sound.

The film takes place mostly in New York, a very rich and busy environment with a ton of extraneous sound always happening. There are times in Whiplash where you do get that sense and come out into the world, but for the most part we’re in these womb-like subterranean basement practice rooms that are very warm, but very sterile. And we tried to create this feeling that you were trapped in these cave-like surroundings with a monster.

What would you say were the key items of audio equipment that you used on Whiplash?

The classic Lexicon 480L Reverb.

Believe it or not, that reverb unit is still as useful to me as the day I first used it in the 1980s. It was the first reverb unit I ever used, and the first where anyone sat me down and taught me how to use it. It’s become a real collector’s item, and it still sounds better than 90% of other reverbs out there. I use all the Avid Pro Tools-based reverbs as well, but there are certain sounds that the 480 makes that are still very dear to me.

The fact that it’s such a predictable reverb saves so much time that when you come up against a schedule that’s as challenging as Whiplash was, having equipment that is predictable and

Why do you think this particular project succeeded with such little room for error?

I think it was a rare situation where a lot of very talented people came together and we had a lot of technical support.

The sheer scale of technology that we were able to utilise was designed for much bigger, grander scale projects; all of the sound was played off one central server and there was no downtime from having to move from one room or machine to another.

What was your favourite part of the whole process?

The most fun was putting back the sounds that the recording engineers worked so hard to take out of music recordings. There’s a scene where an upright bass player is playing – to put the sound of his fingers back on the frets and strings was a private joy for me; to add all that music paper, all that rustling, all the setting up. Those quick-cut scenes where people are adjusting music and unpacking instruments, blowing spit out of the trumpet – all of those sounds were really good fun. I had to take advantage of any time where there wasn’t music playing, to put the reality of the physicality of playing instruments back into what otherwise would have been very sterile playback.

How proud are you of the finished product?

Whiplash is a great film; even if someone else had done the sound, they would have had just as much fun with it. We were really lucky to have such a rich canvas to put our mark on.