Live Sound International 1407

I N S TA L L AT I O N | C O N C E R T | T H E AT E R | C O R P O R AT E AV | W O R S H I P | C L U B | R E C O R D I N G THE JOURNAL FOR LIVE EVENT TECHNOLOGY PROFESSIONALS

July July 2014 2009||www.prosoundweb.com www.prosoundweb.com| $10 | $10

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DO YOU SPEAK GEEK? The unique language of audio analysis.

PLUS: THE INS AND OUTS OF DI BOXES MICROPHONES IN LIVE RECORDING MAKING TECHNOLOGY TRANSPARENT

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www.yamahaca.com

QL5

CL1

CL3

QL1

CL5

A Strong Bloodline Two series, one family. Both represent the evolved sound quality and innovative functionality of today’s digital age while embodying the rich heritage behind the Yamaha name. The CL Series is comprised of 3 models with unique built-in features including Rupert Neve Designs Portico 5033/5043 EQ and compressor, Yamaha’s VCM analog ™

circuitry modeling technology and Centralogic operation. The 2 models of the QL Series take the best of CL’s advanced features and combine a few additions such as a built-in auto mixer from Dan Dugan Sound Design to provide a simplified user-friendly all-in-one mixing experience. Connected by the Dante audio network, the CL and QL Series work seamlessly together to provide complete solutions for a variety of sound applications.

Yamaha Commercial Audio Systems, Inc. • P. O. Box 6600, Buena Park, CA 90620-6600 • ©2014 Yamaha Commercial Audio Systems, Inc.

IS LIVE PUSH YOUR SOUND AS FAR AS YOU WANT

The LYON linear sound reinforcement system is designed to faithfully reproduce your sound even when the system is pushed to its limits. Live sound venues and tours around the world rely on LYON for the most consistent sound at all levels.

PART OF THE

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Pa al Li Partia List stin ing: g: Adap poe • Ame mexAu udi dio o•• Ba Barth haAudio ioV Vis isual • Ba Baye yerrMe Medi dia a • Bazelm man a sAV VR • Brig ght htN Nor orwa w yAS S • Die iezzyCíaLttda d • DigitalConsoleR Renta t l D sh Du how • Equip ipos os Apo polo • Geo eorg rge e Re Rellles Sou und • GMB M Pro-Sou und • Media Resource Group • Most High Produc cti tion o s P Profe PA f si s onal • Pro Sou und and Video e • Spe peci c al Eve vent nts s Audio • Skynight • VER ER TourSound n • Victory Tour Production n • Use Sonido Photto Creditt: R Ralph Larmann n

BEFORE: 96 tracks at 48KHz with a HUGE rack and many peripherals. NOW: Digigrid technology recording 128 tracks of 96K (times 2 - record and backup) at FOH every night flawlessly to a small flyable rack. Amazing!” Mixer/FOH/Ken “Pooch” Van Druten: Linkin Park, Kid Rock, Kiss

DiGiGrid MGO/MGB

DiGiGridMGO 128ch Optical MADI-to-SoundGrid Interface

Find out what DiGiGrid MGO & MGB interfaces can do for your MADI console at digigrid.net For U.S. sales: www.waves.com

DiGiGridMGB 128ch Coaxial MADI-to-SoundGrid Interface

IN THIS ISSUE

JULY 2014

FEATURES 16 | Focus On The Knobs? Making technology transparent in the quest of art. by Karl Winkler

24 | 48 Hours In Las Vegas Upgrading the PA for Blue Man Group at the Monte Carlo. by Marcus Ross

36 | Prepared To Manage Steps to a successful pre-production process. by Danny Abelson

48 | And They’re Off... An audio makeover at historic Churchill Downs.

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by Live Sound staff

DEPARTMENTS 8 | Loading Dock

44 | Spotlight

54 | Real World Gear

Recording options of digital consoles.

EQUIPMENT

ers, networking and more. by Live Sound staff

by Live Sound staff

medium-format line arrays.

18 | Clear Path

50 | Road Test

Going direct – the ins and outs of DI boxes.

Evaluating the Shure GLXD 2.4 GHz wire-

by Gary Parks

less microphone system. by Craig Leerman

30 | In Focus

52 | Road Test

Microphone choice and application for live

Checking out the new QSC Audio amplifier/

recording. by Craig Leerman

processing platform. by Danny Rosenbaum

EQUIPMENT

New subwoofers, amplifi-

38 | Tech Topic

Focusing on the latest

by Live Sound staff

6 | From the Editor’s Desk 60 | NewsBytes 63 | Advertiser Index 64 | Back Page

The unique language of audio analysis. by Pat Brown

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18 Live Sound International (ISSN 1079-0888) (USPS 011-619), Vol. 23 No.7, is published monthly by EH Publishing, 111 Speen Street, Suite 200, Framingham, MA 01701 USA. US/Canada/Mexico subscriptions are $60 per year. For all other countries subscriptions are $140 per year, airmail. All subscriptions are payable by Visa, Master Card, American Express, or Discover Card only. Send all subscription inquiries to: Live Sound International, 111 Speen Street, Suite 200, Framingham, MA 01701 USA. Canada Subscriptions: Canada Post Agreement Number 40612608. Send changes of address information and blocks of undeliverable copies to Pitney Bowes International, PO Box 25542, London, ON N6C 6B2. POSTMASTER: send address changes to Live Sound International, PO Box 989, Framingham, MA 01701. Periodical Postage paid at Framingham, MA and additional mailing offices. Reproduction of this magazine in whole or part without written permission of the publisher is prohibited. Live Sound International® is a registered trademark of EH Publishing Inc. All rights reserved. 2014 EH Publishing. Check us out on the web at http://www.prosoundweb.com.

A division of Jam Industries Ltd

For more information email [email protected]

From the Editor’s Desk… So, as we ask on the cover: Do you speak geek? If not, no worries – that’s why we’re presenting Pat Brown’s Measurement Glossary, beginning on page 38 of this issue. When I first received the article, I had a feeling it was something special, and that was quickly confirmed. As usual, Pat takes a complex topic and breaks it down into an excellent primer that builds to further understanding as you go. It’s a great example of why he’s a renowned educator, and why the pro audio industry is so fortunate to benefit from his efforts and talent. Also in this issue, we get a behind-the-scenes look at a fast-paced system upgrade project at the Blue Man Theater at the Monte Carlo in Las Vegas. Marcus Ross, resident audio supervisor for Blue Man, provides the details on how the whole thing happened within a 48-hour time window, including the back story on the system design and the major planning work it took to pull it off. Another interesting project covered in the issue, this one at historic Churchill Downs, offers further evidence of the resourcefulness and expertise of audio professionals. Coming off a hectic InfoComm show in late June, I wasn’t sure if Craig Leerman would have time to put together the article on microphones for live recording that we’d been discussing. Turns out that he handled it with no problem, as you’ll see beginning on page 30. His decades of experience working in virtually every type of live audio situation serves him well in being able to quickly and clearly communicate some very effective approaches. Karl Winkler checks in with a thoughtful column, while Danny Abelson continues his discussion with noted engineer Dave Natale, this time focusing on key aspects of pre-production. And as always, there’s much more. Enjoy the issue…

Keith Clark Editor In Chief, Live Sound International/ProSoundWeb [email protected]

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VOLUME 23 | NUMBER 7

Publisher | Kevin McPherson | [email protected] Editor-In-Chief | Keith Clark | [email protected] Senior Contributing Editor | Craig Leerman | [email protected] Senior Technical Editor | Ken DeLoria | [email protected] Church Sound Editor | Mike Sessler | [email protected] Europe Editor | Paul Watson | [email protected] Technical Consultant | Pat Brown | [email protected] Art Director | Katie Stockham | [email protected] Associate Art Director | Dorian Gittlitz | [email protected] ProSoundWeb.com Editor-In-Chief | Keith Clark | [email protected] Product Specialist | Craig Leerman | [email protected] Webmaster | Guy Caiola | [email protected] Karl Winker | Gary Parks | Danny Abelson Marcus Ross | Bruce Bartlett | Danny Rosenbaum Live Sound International 111 Speen Street, Suite 200 Framingham, MA 01701 Phone: 800.375.8015 www.livesoundint.com Jeff Turner | Account Executive 415.455.8301 Fax: 801.640.1731 [email protected] Mark Shemet | Associate Publisher Online, ProSoundWeb.com 603.532.4608 | Fax: 603.532.5855 [email protected] Manuela Rosengard | Ad Production Director 508.663.1500 x226 | [email protected] Jason Litchfield | Ad Production Manager 508.663.1500 x252 | [email protected] Circulation and Customer Service inquiries should be made to: Live Sound Customer Service EH Publishing Phone: 800-375-8015, ext 294 (Outside the U.S.: 508.663.1500 x294) Fax: 508.663.1599 [email protected] 111 Speen Street, Suite 200 Framingham, MA 01701 EDITORIAL AND READER SERVICE RELATED EMAIL ADDRESSES Circulation & Subscriptions | [email protected] Loading Dock Submissions | [email protected] World Wide Web Inquiries | [email protected] Advertising Rate Information | [email protected]

ON THE COVER: A fun introduction to our cover story by Pat Brown, beginning on page 38. Our thanks to Rational Acoustics for the Smaart screen image.

.com

REPRINTS: Erica Halloran 508.663.1500 x265 | [email protected]

LOADINGDOCK Allen & Heath Qu-32 @

Adamson Systems E219 @

A 32-fader, 38-input/28-output digital mixer incorporating

A subwoofer loaded with two lightweight, long-excursion

capabilities such as total recall of settings (including faders and

19-inch SD19 Kevlar neodymium drivers utilizing proprietary

digitally controlled preamps), Qu-Drive integrated multi-track

Advanced Cone Architecture. The drivers, mounted in a front-

recorder, dSNAKE for remote I/O and personal monitoring,

loaded enclosure, employ dual 5-inch voice coils for enhanced

multi-channel USB streaming, Qu-Pad control app, and iLive

power handling, designed to reproduce clean, musical low-

FX library. It comes with a 7-inch touch screen to drive “Touch

frequency information. Integrated rigging permits a 0- or a

Channel” access to channel processing, as well as 33 motor-

3-degree angle, allowing for compatibility with the company’s

ized faders. I/O includes 32 mic/line inputs, 3 stereo inputs, 24

Energia full-range line array modules. The E219 is specified for

mix outputs including 2 stereo matrix mix outputs and 4 stereo

use and packaged with the Lab.gruppen PLM 20000Q ampli-

groups with processing, patchable AES digital output with

fier, and four E219s can run from a single amp. The cabinet,

a further 2-channel ALT output, dedicated talkback mic pre

measuring 23.5 x 56 x 35 inches (h x w x d) and weighing 249

input, and 2-track output. The Qu-Drive integrated 18-channel

pounds, is constructed of marine grade birch plywood as well

USB recorder can record and play back multi-track and stereo

as aircraft grade steel and aluminum. It is equipped with three

audio .wav files to a USB drive. The USB interface can also

Speakon NL8 connectors, two parallel in/out plugs, and one

be used to store scene and library data for archiving and later

dedicated output connection point. www.adamsonsystems.com

recall. The free QuPad iPad app gives instant wireless control of the mixer’s key parameters. www.allen-heath.com

™ Shure QLX-D

PreSonus SL-Dante-SPK ¤

A digital wireless system providing

A card for the company’s StudioLive AI-series (Active

24-bit digital audio, networked control,

Integration) active loudspeakers that includes an Ethercon

and compatibility

connection for Dante audio networking and remote control

with Shure’s intel-

via the free SL Room Control application.

ligent rechargeable

It allows users to create a networked

battery technol-

audio system with any Dante-enabled

ogy. It transmits

mixer using a standard 1 GB Ethernet

accurate audio

switch. Users can also connect

with extended, flat frequency response. The system’s automatic

non-Dante mixers, such as

channel scan and IR sync make finding and assigning an open

a first-generation PreSonus

frequency fast and simple. AES-256 encryption comes standard

StudioLive, to the analog

and can be enabled for secure wireless transmission. The

inputs of a Dante-equipped

system’s intelligent lithium-ion rechargeable power options can

AI loudspeaker and then

provide up to 10 hours of continuous use and report remaining

broadcast the signal over the

runtime in hours and minutes. QLX-D transmitters can also run

Dante network using Cat-5

on standard AA batteries for up to nine hours. QLX-D works

cable. The upgrade works with

with networking tools, including Shure Wireless Workbench 6

StudioLive 312AI, 315AI, and

control software, third-party control systems (AMX/Crestron),

328AI loudspeakers, as well

and iOS devices for control and monitoring with the new Shure-

as the StudioLive 18sAI sub-

Plus Channels mobile app. www.shure.com

woofer. www.presonus.com

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Live Sound International July 2014

www.ProSoundWeb.com

Products Fresh Off the Truck Audio-Technica ATND971 ¤

Celestion CDX1-1010 ¤

A cardioid condenser boundary

A lightweight, low-profile ferrite magnet compression driver

network microphone that

with 15 Wrms (AES standard) power handling and 107 dB

transmits audio and control

sensitivity. Frequency range is stated as 1.5 kHz to 20 kHz.

data together over Dante

Finite Element Analysis (FEA) techniques are used to optimize

network protocol. An

both the magnetic and acoustic design. The CDX1-1010 is

Ethernet connection

designed for entry-level 2-way and

allows the ATND971

3-way loudspeakers. The CDX1-

to communicate across an

1020 variant is supplied with

existing network of Dante-enabled devices and, with the mic’s

partial phase plug assembly,

programmable user switch, control any of those devices at the

for applications where the

push of a button. Because Dante can support up to 512 bidi-

outer phase plug and horn

rectional audio channels, the mic offers a scalable solution. The

form a single moulding as

ATND971 is powered by network PoE. It’s also outfitted with pro-

part of the front baffle.

prietary UniGuard RFI-shielding technology and UniSteep low-cut

http://celestion.com

filter. www.audio-technica.com

Eastern Acoustic Works (EAW) Otto ¤

DiGiCo V685 ¤

The first subwoofer in the

The latest software upgrade for the company’s range of digital

Adaptive Performance

consoles. It provides an increased bus count for the SD9 from 16

Series, Otto is loaded

to 24 Flexi buses, with the SD11i/B input channel count increased

with two 18-inch woof-

from 32 to 40

ers, with acoustic energy

Flexi channels.

exiting from four spaced

The upgrade also

apertures in the corners

supports Optocore

of the enclosure. It is rated

DD4MR, DD2FR,

to deliver output of 131 dB

X6R and DD32R

(1 meter, continuous, full-space)

devices in audio

and response that extends down to 22 Hz (-10 dB). Each Otto

I/O. Further, any

transducer is separately powered and processed, allowing

SD5, SD8, SD9,

multiple directivity patterns to be created from a single module.

SD10 and SD11

It can readily be combined in arrays to provide increased

running Waves 9.5

pattern control and output. EAW Resolution software gener-

will now have 32 stereo Waves racks. There’s also support for the

ates DSP parameters to simultaneously adapt the complex 3D

D-Rack AES input card and the addition of the D2 Rack as an I/O

wavefront surface and optimize frequency response to match

device. For theatrical environments, the Relative Faders in cue

the requirements of any venue. The Otto G24 package sup-

groups are now a macro command; auxes, groups and matrix

ports two single columns, each with 12 Otto modules, that can

channels can now be added to channel sets; and channel cues

be suspended from a single flybar. The columns are joined by

now default to showing names. V685 is being provided free of

distribution racks flown adjacent to the flybar of each array as

charge for an introductory period. www.digico.biz

well as full network redundancy. www.eaw.com

™ Eighteen Sound ND4015Ti2 A 4-inch neodymium compression driver outfitted with a next-generation titanium diaphragm that provides higher sensitivity and extended high-frequency performance, resulting in enhanced HF clarity. It has a 4-inch aluminum voice coil and 1.5-inch throat exit, 4-slot phase plug, and is also available in 1.4-inch and 2-inch throat configurations, making it a flexible design and “platform-agnostic” choice. www.eighteensound.com www.ProSoundWeb.com

July 2014

Live Sound International

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:: Loading Dock ::

Lab.gruppen D Series @ A 4-channel amplifier/DSP platform for installations available in three power configurations (8,000, 12,000 and 20,000 watts total power output) and two variants – Lake or Tesira (Biamp

JBL Professional EON610 & EON612 @

Systems). The Lake variant offers Lake Processing with analog, AES and dual-redundant Dante networking. It is supported by

Joining the EON600 Series, the EON610 (10-inch) and EON612

the development of new custom software to provide extensive

(12-inch) 2-way loudspeakers incorporate built-in 1,000-watt

integration with most key system manufacturers. The Tes-

power amplification. Custom JBL high- and low-frequency

ira variant is equipped with Tesira DSP and AVB audio and

transducers deliver high sound pressure levels with low

control. The D Series also includes proprietary Rational Power

distortion throughout the frequency range. JBL examined the

Management (RPM) technology, providing flexible power allo-

radiation characteristics of the HF and LF drivers at 36 differ-

cation across all channels to foster efficient and rational use of

ent points, employing proprietary measurement techniques,

total amplifier inventory. http://labgruppen.com

then designed individual waveguides for both components that control the sound radiation at the high frequencies, the crossover point, and at the low frequencies. Proprietary fluting is designed into the structure to guide the frequencies through the full range of the system, resulting in consistent response. An iOS- and Android-supported interface can be paired with the Bluetooth Smart Ready 4.0 for controlling master volume, adjusting the 5-way, user-definable parametric EQ, and saving and recalling user presets. The cabinet includes four handles and indexed feet for secure stacking. www.jblpro.com

XTA APA-4E8 @ The first model in the APA Series (Adaptive Processing Amplifi-

Aviom A360 Display ¤

cation), providing power and DSP platforms designed to interact

An iOS application that provides a visual display of mix infor-

intelligently and adapt to prevailing conditions, protecting driv-

mation on A360 personal mixers, allowing performers to view

ers, and significantly enhancing performance of loudspeaker

volume levels, stereo placement information, tone and reverb

systems. The APA-4E8 provides four channels of power totaling

levels, as well as signal levels for each mix channel of the A360.

20 kW peak output into 4 ohms and continuous power available

In addition, the app will allow users to name channels and

of 3,400 watts per channel into 4 ohms. Four audio inputs allow

presets as well as see the customized network slot map for the

all four (class D) amplifier channels to be individually utilized (if

selected A360. The app is designed for iPhone or iPod touch,

required) with a suite of XTA DSP, including dynamic EQ, FIR

which fits in the built-in tray on A360 personal mixers. Also

(and phase linearization) and IIR filtering, mix matrix, and the

required is a D800 or D800-Dante A-Net Distributor,

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Live Sound International July 2014

manufacturer’s limiters and soft-knee compressors. It can route audio from analog, AES or network sources with automatic fall-

which communi-

back. USB and internal SD cards offer additional audio choices.

cates with the iOS

It also includes GPIO and remote control covered by Ether-

device through a

net, USB and RS485. Software written to run natively on both

connected WiFi

Windows and Mac platforms is available and operable via an

router.

Ethernet or USB connection. The APA-4E8 is housed in a 2RU

www.aviom.com

chassis and weighs 28.2 pounds. www.audiocore.co.uk www.ProSoundWeb.com

:: Loading Dock ::

Grund Audio Design GA-LC9 & GA-LC9P 2

Electro-Voice X1 & X2 ¤

Joining the company’s Gala Series, the GA-LC9

array loudspeakers. The SMX 12-inch woofer in the X1 (DVN3125

(passive) and GA-LC9P (powered) line source

12-inch woofer in the X2) is coupled to a proprietary Mid-Band

column loudspeakers are designed for both

Hydra device that

portable and fixed installation applications.

emulates the acoustic

A 2-way design utilizes nine 3.5-inch trans-

behavior of a double

ducers. Frequency response is rated at 140

line of four 3-inch point

Hz – 20 kHz, while system coverage provides

sources, fostering opti-

120-degree vertical and 10-degree horizon-

mized mid-band cou-

tal dispersion. The passive GA-LC9 is rated

pling of the array while

at 300/600 watts (RMS/program), while the

maintaining efficiency and power.

powered GA-LC9P, is rated at 350 watts RMS.

The HF section of the X1 incorporates two new ND2R

A single LC9P can power an LC9. Enclosures

ring-exit 2-inch titanium compression drivers coupled to a pair

include 2 x 2 flypoints as well as pole mount

of WCH constant energy planar wave generators on a 90-degree

adapters on the top and bottom, compatible

waveguide. The HF section of the X2 matches two ND6A 3-inch

with the GT-LPB-24C subwoofer. Manufac-

titanium compression drivers to the pair of Advanced High-Fre-

tured in the USA, enclosures are made of

quency Hydra constant energy planar wave generators, also on

13-ply Baltic birch, measure 32.63 x 5.38 x 6.75

a 90-degree waveguide. A captive twist-lock multi-angle arraying

inches (h x w x d), and weigh 15/22 pounds

system for both models is designed to simplify the rigging of any

(passive/powered). www.grundaudio.com

size of array. (The X1 is pictured here.) www.electrovoice.com

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Live Sound International July 2014

The first models in the company’s next generation of X-Line line

www.ProSoundWeb.com

www.solidstatelogic.com/live

SSL Live

[email protected]

:: Loading Dock ::

Mackie SRM750 & SRM2850 @

Crown Audio XLC2800 & XLC2500 @

Two new models joining the SRM Series, each with 1,600

Two power amplifiers (both have 2 channels), designed for

watts of onboard power paired with custom transducers

install applications, incorporating proprietary DriveCore technol-

housed within internally-braced, all-wood cabinets. Like all

ogy. They can operate into impedances from 8 ohms to 2 ohms

SRM full-range loudspeakers, the SRM750 incorporates

using stereo, parallel or bridged mono outputs. The XLC2800

proprietary HD Audio Processing, which includes patented

delivers 775 watts per channel at 4 ohms, while the XLC2500

acoustic correction algorithms for high-definition output

provides 500 watts per channel at 4 ohms (and 2,400 watts and

plus system optimization tools like application-specific

1,550 watts, respectively, into 4 ohms in bridged mode). The

loudspeaker modes and an accurate feedback destroyer.

DriveCore IC chip combines the amplifier driver stage into the

It also includes an integrated 2-channel mixer with Wide-Z

power output stage along with additional audio-signal functions,

inputs. The SRM2850 is a dual 18-inch-loaded subwoofer

yet is about the size of a postage stamp. Power, signal, clip and

designed for high-output LF performance, making it suitable

fault indicators are included, along with a range of input/output

for applications such as stacked rigs at festivals, clubs and

connectors. XLC Series amplifiers also have rear-panel volume

other live applications. www.mackie.com

controls for each channel. www.crownaudio.com

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Live Sound International July 2014

www.ProSoundWeb.com

OUTLOOK together, is obviously important as long as the end result is kept in mind. The audience probably won’t know if you used an actual LA-2A leveler or a plugin equivalent on the vocals. But they know when they can’t hear the words or if the bass is overwhelming the mix. Adopting new technology into a system should not be about trying to find ways to use it so we get our money’s worth. Instead, it’s about having the new stuff integrate so seamlessly that we almost forget it’s there, except for whatever benefits it brings to the table in terms of better sound, smoother workflow, or faster set-up time.

FOCUS ON THE KNOBS? Making technology transparent in the quest of art. by Karl Winkler

❯❯❯❯❯❯ AT ONE TIME OR ANOTHER, all of us who have sat behind a mixing console at a show are asked “do you know what all those knobs do?” Of course the answer is “yes” – or at least it should be. What they don’t ask is “do you know anything about acoustics?” or “do you have a handle on power and grounding?” because these subjects are not nearly as interesting or obvious to the novice observer. Maybe the real question is along the lines of “do you know how to bring out/enhance the art using the tools in front of you?” So what about all those knobs? I often wonder if we can relate them to the concept of “if you’re a hammer, then every16

Live Sound International July 2014

thing looks like a nail.” In other words, if we know what all those knobs (and buttons) do, does it mean we’re compelled to twist the knobs and push the buttons? In many cases I’m afraid it’s true, and yet, we can miss something in the process. PRACTICE MAKES PERFECT As an amateur photographer growing up in the days of film and mechanical cameras, I always found it useful to practice with the equipment empty before putting real film at risk. In those days, every exposure cost money, and frankly, I didn’t have much to spare. But more importantly, I wanted to always get past the awkwardness with the gear and get on to the whole point: capturing good images. My friend Pat Moulds, a retired professional upright bass player, used to say that “the point of practice is to get to where you can play a passage without hesitation.” In other words, the technique becomes transparent and the art comes through. Back to our business of sound. Knowing what every knob and button does, and how the sound system is put

VISUALIZE (AURALIZE?) Another photography analogy: Ansel Adams espoused the idea of visualizing the result you wished to have when viewing a scene, to imagine how you would want it to appear in a photographic print. Then, using the technology at hand and the technique to go with it, achieve the desired results. One of the challenges is that a natural scene has levels of light and dark, i.e., dynamic range, that cannot be captured or reproduced with photographic equipment. First, Adams suggested exposing the film in order to ensure that there were details in the shadows (above the noise floor). Then he gave pointers as to how the film should be developed in order to prevent the highlights from blowing out (headroom). Finally, he formulated a precise method of printing so that – although the real-world levels of light and dark could of course not be reproduced – the relative levels could be kept intact, providing the viewer with the impression desired by the photographer in the original vision. With the tools of the day, this was a very involved process, with lots of smelly chemicals and expensive equipment, and it required a whole lot of patience and discipline while stumwww.ProSoundWeb.com

bling around in the darkroom. Sound is not that different. For one thing, the real dynamic range of many instruments or ensembles is greater than what can be reproduced through loudspeaker systems. And yet the listener generally wants to have a bit less than reality for the sake of comfort, especially when it comes to things like vocals. Thus, dynamic compression is routinely used for this purpose. However, let’s get back to the main point: cultivating a vision about the desired end result. What kind of music is it? Do the performers have an idea of how they want to be presented? Is there a recording we’re trying to match or to which the audience is comparing our efforts? All these things affect our choices in technology and technique. That is, if we’re paying attention. WHEREFORE ART THOU, REVERB? What are some other examples of using technology to achieve a “vision” in the mix? Application of reverb to create space, for sure. Applying delay to enhance the rhythmic elements of the music or to create “size” by panning a delayed copy of a source. Drawing on distortion to supply “color.” And certainly, using EQ to carve out space for each instrument or voice, draw attention to or away from an element in the mix, or to create vertical “size.” All these approaches are certainly valid, and there are dozens (if not hundreds) more. One way to learn these and other creative uses of technology is to carefully analyze recordings and performances with disciplined listening. One of my best audio teachers in college would start every class with an analytical listening exercise, where we would make a chart with the relative levels of each instrument or voice, what effects were used, panning and space, etc. After months of doing this with dozens of songs, it was very eye-opening because we realized how each different www.ProSoundWeb.com

producer and engineer had exploited the available technology to achieve certain results, thereby enhancing the musical experience. Once in a while we’d also notice the bad examples where some aspects of the recording or mixing techniques got in the way of the results, and even ruined the recording. One final thought: it’s easy to get caught

up in the technology itself. But really, our jobs are to get past that, figure out what works, get really good at it, and make music. After all, that’s what it’s all about. n

KARL WINKLER is director of business development at Lectrosonics and has worked in professional audio for more than 20 years. Reach him at [email protected].

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July 2014

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CLEARPATH to-output path for isolation. Because the transformer passes the signal from the primary to the secondary coil via induction versus requiring a physical connection, the ground current cannot flow and create hum and buzz.

GOING DIRECT The ins and outs of DI boxes. by Gary Parks

������ GIVEN THE WIDE VARIETY of audio sources that are connected to the microphone and line inputs of the mix console, the availability of highquality DI boxes is a true blessing. Electric basses, acoustic guitars with piezo transducers, other stringed and wind instruments with pickups, effects units, CD players, computers, and more contribute to the overall audio palette of the event. DI boxes (also called direct boxes) are the tools that allow the disparate sources, each with its own distinct functions, output levels, and imped18

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PASSIVE & ACTIVE DI boxes are made in both passive and active formats, and each has its primary uses and advantages. Basically, a passive design requires no exterAndy Heller and Gary nal power to function, Wood, co-owners of and its internal audioAudio Production Group quality transformer per(San Carlos, CA), with their Countryman Type forms the conversion 10S and Type 85S stereo functions. An active DI direct boxes. requires power from a phantom power source and/or from a battery. Electronic circuitry is used for the signal balancing and impedance matching functions. Distinguishing the two types, whether they’re labeled or not, can usually be done by noting ance characteristics to connect with the the absence or presence of a battery sound reinforcement system properly – compartment, an on/off switch, and without adding excessive noise and/or an LED. (Note, however, that there altering frequency response. are active units that do not have all of these elements.) PRIMARY APPLICATIONS A basic rule of thumb is to use a DI boxes provide three basic functions. passive DI with an active source, and First, they convert unbalanced signals an active DI with a passive source. A from sources such as instrument pickstandard electric guitar and many basses ups and electronic instruments into are passive sources, as are most acousbalanced signals that can travel longer tic guitars with under-saddle pickups distances without induced interference and other instruments with piezo pickor signal degradation. Second, they ups. Keyboards, active pickup systems, help with impedance matching, espeeffects and other electronic devices, as cially from high-impedance sources well as audio sources with a battery or like passive guitar pickups being fed an AC plug, are active sources. into low-impedance mic inputs on the Mick Conley, mix engineer for mixing console. Third, while performcountry musician Marty Stuart, says ing the above electronic functions, DIs that when connecting a bass with pasact as an interface to change from one sive pickups, he “usually uses an active connector type to another, typically DI to help with the signal level and from 1/4-inch to XLR. clarity,” while a bass with active pickAs an added benefit, the audio transups will have “a passive DI, since there’s former within a passive DI will break no need for the extra output, and a pasa ground loop. Some active DIs also sive DI can help keep the tone from place a transformer within the inputwww.ProSoundWeb.com

getting too aggressive.” Mikail Graham, sound engineer at the Grass Valley Center for the Arts, tells me, “I tend to use active DIs for bass and guitars, with passive units primarily for keys and various processors.” He adds that “keyboards and rhythm boxes, as well as the ever encroaching array of vocal effects processors, also benefit greatly when used with a DI.” Nick Malgieri, AV manager and audio engineer at Stanford’s Bing Concert Hall, states, “Generally speaking, I prefer active DIs. I find the higher output and high-presence tone to be better for most applications. Passives work as well and are often preferred by rock ‘n’ roll engineers who prefer a softer, rounded tone, although I find them to sound a bit flat and the low output can drive up noise floor.”

The design decision of whether an active DI will allow battery powering or use phantom power from the console relates to the potential limitations that battery power poses to the maximum signal level the DI can handle, as well as the fact that batteries deliver less voltage as they’re used up, and that they can run out of juice in the middle of a show. Units such as the Radial Engineering PZ-DI and the Klark Teknik DN100 deliberately forego the battery option. There may be circumstances where a convenient source of phantom power is not available at a particular location, and an active DI is necessary; for example to connect mixing consoles in different locations while using the ground lift, or acting as a line balancer. The Countryman Type 10 active DI works with both phantom and battery power, and it also

Radial Engineering offers numerous types of DIs, ranging from the passive StageBug SB-5 for laptop computers to the active PZ-DI optimized for orchestral instruments.

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:: Clear Path ::

includes a power monitoring circuit with a pair of LEDs and a power-test switch that tracks the relative levels and can transition between sources to maintain the best performance. I/O IMPEDANCE The output impedance of a passive electric guitar or bass pickup can be in the hundreds of kilohms, and that of an under-saddle piezo pickup even greater – and the impedance varies with the frequency of the note being played. In order to transfer the audio signal correctly, rather than attenuating the instrument’s lower or higher frequencies, the input impedance of the next device in the signal chain must be considerably higher. In this regard there are numerous choices; for example, the impedance of the BSS AR-133 is 1 megohm while the Leon Audio Mk 2A is 33 megohms. (Both are active units.) When connecting an acoustic guitar with piezo pickups, Conley notes, “I like the Radial PZ-DI because of the impedance button on the side of the unit; it helps to match the impedance better and therefore warms the tone.” The PZ-DI has impedance settings of 220K, 1M, and 10M to accommodate devices from lower to very high output impedance. Passive units have an input impedance about an order of magnitude lower, with typical units in the range of 50 to 150 kilohms. The lower output impedances of keyboards, effects, CD players, and the like – ranging from a few hundred ohms to several kilohms – means that the input of a passive DI will be more than sufficient to accept the audio signal without introducing frequency-response problems. Also, the passive DI can be less prone to overload distortion with high signal levels, since transformers saturate at higher levels rather than distort, which can be more pleasing to the ear. The XLR output of a DI box is low impedance, similar to that of a micro20

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The Klark Teknik DN100 foregoes battery operation.

phone, so that it properly interfaces with the mic input of a mixing console. The output level is also closer to that of a microphone, so that the normal range of the channel’s trim control is able to make any fine level adjustments, rather than seeing a signal that is too low or too hot. Attenuation buttons or “pads” on a DI are often available to make larger adjustments to the signal level before it is sent to the console, with a variety of values depending on the DI, in the range of -10 dB to -40 dB. GOING THE DISTANCE An instrument-level signal going through an unbalanced 1/4-inch guitar cable is only going to travel a few feet before the capacitance of the cable will roll off some of the high frequencies. For a typical performing-length cable, this can be part of the desired sound when connected to a nearby amp. However, taking that lower level signal all the way

to the console while unbalanced would undo the tone of the instrument, and open it up to induced electrical noises as it travels by various other cables carrying AC and other signals. A DI box converts the unbalanced signal to a balanced one right at the stage, so that it can be more resistant to interference as it travels the sometimes hundreds of feet back to the console inputs. Along with the balanced XLR output that takes the signal to the analog snake (or digital converter box at the side of the stage) and out to the sound reinforcement system, DIs will usually have an additional unbalanced jack that loops the unadulterated instrument signal to the performer’s on-stage amplifier, so that the guitar or bass amp “sees” the instrument’s pickups. Thus the DI can also function as a signal splitter. Along with filling their functions of signal balancing, correcting impedance mismatches, and breaking ground loops, DI’s are audio devices that are inserted directly in the path between the transducer capturing the audio source and the mixing and amplification components of the sound reinforcement system. The quality and transparency of the signal they output is critical to how the listener will hear the sound of the instrument. The key component in a passive DI is the transformer, and its design and qual-

A look inside the Radial JDI, outfitted with a Jensen transformer (center component).

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:: Clear Path ::

ity is an important differentiator between a passable and a high-performance professional unit. This choice can have audible results. Companies such as Lundahl and Jensen specialize in manufacturing transformers with excellent audio characteristics – and with prices to match. The quality of the circuitry within an active DI is also a critical factor in its audio response characteristics, as well as being a differentiator of the higher-quality boxes. Key measures are frequency response and flatness across the audio spectrum. ROAD READY Because DI boxes are distributed around the stage in unprotected locations, they’re usually ruggedly built devices, often weighing a pound or two, though a few more diminutive (yet still rugged) units can also be had. Connectors and switches are typically recessed

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offering some protection to switches and attached cable connectors. Internal durability is also a factor. The quality of the switches, connectors, electronic components, and circuit boards directly affect how well the DI performs its functions, how long it lasts, and its immunity to induced noise. In most cases you get what you pay for, and since the cost of even a relatively expensive DI is inconsequential compared to the price of a good instrument or mixing console, investing in quality is wise. ■ Some makers offer DI boxes with single and dual channels, as seen here with the Countryman Type 10 and 10S (“S” stands for stereo).

within an extruded chassis, with perhaps 1/8-inch-thick metal surrounding the more heavy-duty units. Some models also include thick rubber side bumpers that function as non-slide feet while

GARY PARKS is a pro audio writer who has worked in the industry for more than 25 years, including serving as marketing manager and wireless product manager for Clear-Com, handling RF planning software sales with EDX Wireless, and managing loudspeaker and wireless products at Electro-Voice.

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:: Blue Man Group::

IN L AS VEGAS

A

t 9:30 pm on a Tuesday in December, I had five teams totaling more than 30 people poised to descend on the Blue Man Theater at the Monte Carlo in Las Vegas to implement a new house system. With less than 48 hours, including meal breaks, turnarounds and hopefully a nap or two, everything had to be ready for a show by Thursday at 7 pm. The organized chaos worked even better than imagined and the sonic improvement is nothing less than stunning. The choice to make the system upgrade at the 1,500-seat, three-level venue began only a few months earlier when the artistic and production team from Blue Man Group were at the Sydney Lyric Opera house in Australia, building a new production. As resident audio supervisor for Blue Man, I was tasked to design an entirely new system from the ground up for the production down under. After a lengthy evaluation process I chose L-Acoustics KARA line source arrays for their sonic quality and compact size. From the beginning of tech process, the sys24

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������������ ������������

48 HOURS

Upgrading the PA for Blue Man Group. by Marcus Ross

tem’s elegance and efficiency was evident to everyone, with the musical director, the technical staff, Blue Men and local producers all remarking that the sound was the clearest, full-range system they had heard. After the Australian production finished, the company started looking at other opportunities to recreate this experience. The Las Vegas production was a natural fit.

Excitement & Impact In September, 2011, when I joined Blue Man, my goal was to bring consistency www.ProSoundWeb.com

to our productions around the world. In this quest I’ve experimented with everything from microphones and console choices to loudspeaker selections. My ultimate goal is to convey the excitement and impact of Blue Men performances at more than 64 shows a week at eight sites around the world, as well as numerous special events. Shows are presented at venues ranging from the 300-seater Astor Playhouse in New York City to a special one-off production at the Hollywood Bowl for more than 17,000. With similar content across all productions, but different needs in each venue, I find that I need to keep an open mind and consider the unique needs of each space. With a PA system, my focus is on SPL, bandwidth, coverage, polar stability and any physical limitations that might exist. The musical composition for Blue Man shows is very dynamic and contains a significant amount of bandwidth. Choosing a loudspeaker system that has the ability to translate the entire bandwidth clearly throughout a large dynamic range is necessary for the blue men to convey both their subtle humor and high energy to the audience. The transient response of the mix is what makes these shows far different than other artists I’ve worked with. This is instantly noticeable with Blue Man instruments, ranging from the acoustical PVC instruments to the electronic MIDI-triggered backpacks. The amount of drums in the show also determines the need to be able to handle transients. The main system also needs to be able to reproduce unique and typical string instruments: zither, stick, guitar and bass. With such complex musical content, it becomes very important that every aspect of the system supports the show and does not impede the performance.

insure sufficient headroom, the specification for the new PA was established as 108 dBA (RMS) at front of house with a peak output of at least 118 dBA. Establishing a target before selecting the box count or type helps me make sure the system has enough resources for the content of the show. To me, selecting a specification for output of the PA is the same as selecting a mic for an instrument: knowing in advance the SPL capability of the system means clipping or limiting of the system will not happen.

The coverage target for the theater was +/- 3 dB from front of house in both A-weighted SPL and response. With a long throw of close to 100 feet and a short throw just under 20 feet for a differential of five times from front to back, this venue is very well suited to a line source array. This is just over two doublings of distance; it would be possible to have no more than 6 dB of loss from front to back, and with proper angle selection, less than 6 dB seemed doable without breaking the line into segments.

Blue Man Group performing with the new system at the theater at the Monte Carlo that bears their name.

Knowing In Advance The mix volume for the Las Vegas production of Blue Man Group spans from 80 dB to 105 dB (A-weighted). To www.ProSoundWeb.com

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:: Blue Man Group::

Left to right, Tony Pittsley (head of audio for the Orlando production), author Marcus Ross (resident audio supervisor), and Jesse Stevens (head of audio for the Las Vegas production) at the DiGiCo SD7 console at front of house.

The combination of coverage and size limitations really dictated that a modular line source array would be the right solution. By segmenting the lowfrequency component from the main element, it allows for a reduction in element size and an increase in resolution. To deal with the limited vertical space and coverage demands, it was a far better solution than a large-format enclosure with a 12- or 15-inch LF driver. Further, having already successfully deployed KARA in Australia, I knew it would be a good fit. It was more than capable of delivering the dynamic range and response the show requires, and being smaller than the previous system, it would not be limited physically in the space. A nice byproduct of the smaller form factor is that I was able to move the arrays further offstage and rotate them more toward center, keeping reflections off the architecture and allowing for a larger stereo field.

Meeting Goals Using L-Acoustics SOUNDVISION modeling software, I determined that 15 per side of KARAi elements would be able to achieve the SPL target and 26

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almost perfectly meet the coverage goals. This approach reduced the overall vertical size of the arrays, allowing them to be positioned slightly lower overall to cover almost every row in the theater with minimal shadowing from scenic and architectural elements. It also meant a reduced need for fills. With the limited time for the transition, it was essential that all variables were considered in advance of the install. The software modeling also helped me ease the concerns of the production team. I wanted the main system to be responsible for the entire effective musical bandwidth. By doing this, the coherence problem of the mains and subs in different locations becomes a non-issue, and the buildup of LF in the first few rows of seating with ground stacked sub woofers is avoided. With any traditionally deployed system, having the subs in a different physical location makes it impossible to time align the two across the entire space. This can be less of a problem if the audience is only on one plane, like a ballroom or a festival, but it becomes exceptionally difficult to find a decent

compromise in a theater space when the seating sections are on multiple planes. By using the main arrays to reproduce the full range it becomes possible to have a uniform tonal balance across the entire audience as opposed to a significant buildup of LF in the first few rows, due to the proximity of the seats to the loudspeakers. To achieve these two goals, lines of six SB18i subwoofers are flown directly beside the main arrays, extending response down to 32 Hz. And with the SB18i having as much output as most double 18-inch subs, we’re able to produce more than 95 percent of the show’s musical content from the arrays and subs.

Homogeneous Coverage The few fills needed to supplement the mains are L-Acoustics enclosures, either coaxial point source or constant curvature line source boxes. Having the same voicing across all the loudspeakers in the system reduces complexity in the tuning process, and not having to spend as much time unifying the response of the fills allowed me to increase the time spent on the creative portion of the show. Within SOUNDVISION mod-

A look at the main arrays as well as the split center cluster deployed in the main system overhaul at the Blue Man Theater.

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:: Blue Man Group::

eling, I was able to ensure that loudspeaker resources and arrival times between the mains and fills would be supportive of each other and not pose a problem in headroom or imaging. For front fill, five very-compact 5XT loudspeakers are fit into the lip of the stage, also with a pair of 8XTs just offstage. The goal was to pull the image down from the flow arrays to the stage and

support the SPL in the first few rows. Ten more 8XTs function as underbalcony fills, while dual ARCS FOCUS enhance coverage to the last three rows of seating in the rear of the balcony. In the end, the delay times, gain settings and EQ provided by the software were almost perfectly matched to what was measured onsite during the calibration of the system. The result is very homogeneous

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coverage across the entire venue with the SPL difference well within the target. For enhanced effects purposes, four SB28 dual-18-inch subwoofers are positioned beneath the stage, along with a dozen of 12XTi coaxial point-source boxes overhead, a pair of ARCS II upstage/center, and a split center cluster of six ARCS IIs. The SB28s are housed in custom bunkers directly attached to the floor, and with the ability to reproduce down to 25 Hz, they provide the infrasonic portion of the show, really focusing on the 25-50 Hz region.

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One of the two KARAi and SB18 flown array sets.

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The upstage/center pair of ARCS IIs, which are flown, foster imaging effects. Thanks to the razor-sharp coverage of the ARCSII, I was able to get greater SPL without affecting the performers that are located beside and below the array. The dozen 12XTi for effects purposes supply very high output and are also passive, reducing the need for additional wiring in our marathon install. The split center cluster of six ARCS IIs, which handles many of the vocal channels, posed an interesting problem. The center location in the theater is not available due to a scenic element, so the choice was to either shoot sound through several truss elements or split the cluster into two parts. Again, due to the very tight coverage pattern of the ARCS IIs, I was able to segment the coverage of www.ProSoundWeb.com

the center cluster while avoiding comb filtering between the two arrays. On the electronics side, all amplification and DSP was simplified into one platform and re-located, with 18 L-Acoustics amplified controllers deployed at three locations around the theater. LA NETWORK MANAGER provides the ability to compensate the response for array size, curvature and atmospheric conditions, while also providing plenty of EQ for room-specific issues. In the end only a couple of filters were needed in any part of the system.

Group, this has been the single biggest leap in performance the show has seen. I’ll close with the reaction of senior music director Byron Estep: “The L-Acoustics system handles high volume and density perfectly, without losing clarity in the transients or changing the tonal character of the mix as the musical dynamics change. Our performers and mix engineers have been

extremely happy with them and feel that they accurately translate the choices they make during a performance. With more than 20 years of experience working on shows and listening to different systems in different rooms, I can say without hesitation that L-Acoustics loudspeakers sound the best and provide the most musical listening experience for our audience.” ■

Go Time Starting at 10 pm on Tuesday, we removed the previous loudspeakers, amplifiers, DSP and cable, and working with our integrator, Clearwing Productions (Phoenix, Milwaukee), the new system elements were all installed and tested by 8 am. Following a rest period and fortified with plenty of coffee, we were back to work by 5 pm that same day, calibrating the system with the help of Scott Sugden from L-Acoustics we finished before break. As noted, the measured results of the system were almost spot-on with the predictions from SOUNDVISION. Within three hours, the system response was unified and the adjustments required were limited to small time-alignment changes and managing the response of the system in the room. By 9 am Thursday morning, it was time sound check. In less than 35 hours, the team had installed 87 loudspeakers at 44 locations in the theater, wired and tested 72 amplifier/controller channels, tuned the PA… and gotten some sleep! From the first downbeat in sound check it was noticeable that all of the hard work was worth it. Every fader I brought up felt as if I was mixing on nearfield monitors in a studio, not a PA at 70 feet away in a 1,500-seat theater. Having made many adjustments to shows over the past several years for Blue Man www.ProSoundWeb.com

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INFOCUS

QUALITY CAPTURE Microphone choice and application for live recording. by Craig Leerman

������ WHEN DONE WELL, a live recording captures the energy and personality of the performance, along with the ambiance and (if desired) audience response. There are many different ways to record a live show, but regardless of the approach, a good recording starts with the right microphones, correctly placed. By “right” I’m referring to mics that fit the particular application, taking factors such as pickup pattern and SPL handling into account. Mics tend to be categorized as “live” and “studio.” Yet while it’s true that certain models are too delicate for live use, and other certain models lack the sonic characteristics sought in the studio, today there’s 30

Live Sound International July 2014

a plethora of models tough enough to handle the rigors of the live realm while delivering the desired audio quality. There’s also a wide range of types. Large and small diaphragm. Dynamic, condenser and ribbon designs. Cardioid, supercardioid, hypercardioid, and figure 8 patterns. Vocal mics, except when they’re used on instruments. Instrument mics, except when they’re used on vocals. Drum mics, except when they’re used on other instruments... If you’re not up to speed on mic types and technology, I recommend a visit to Microphone World on ProSoundWeb, which provides dozens of articles on these topics. STAGE STRATEGY Before even thinking about mics and their placement for live recording, take a look onstage and see what can be done to maximize separation and isolation of instruments and amplifiers – from each other as well as the house and monitor systems. A good multi-track recording consists of clean isolated tracks, and we can use a few studio tricks to help. Separate the backline amplifiers away from acoustic instruments and each other. Try pointing the amps in

a different direction (like offstage) to minimize bleed. Better still, spend time and convince the musicians to actually turn it down (“just this once” for the recording). We can isolate between loud sources with damping materials, and they don’t need to be fancy or expensive. For example, one trick I use is to set the boom of a mic stand to a “T” shape and then drape a packing blanket from my truck over the T. Viola! It’s a portable, adjustable-height gobo that can be placed between loud amps and other mics. Plexiglass is another common way to isolate instruments onstage; a plexi shield around the drums and/or percussion can help keep the drum sound out of stage mics, while keeping the loud amps out of the drum mics. Position stage monitors that are close to mics so that they play into a null spot in the pickup pattern of the mic(s). Better still, try to eliminate stage wedges and get the performers to use in-ear monitors. Try to closemike instruments as much as possible

Factor in monitor location when placing mics, or, consider a wedge- and fill-free stage.

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in order to only pick up the intended sound. In the live world, we tend to like cranking the gain up until it’s close to the red, but studio engineers often use only as much gain on a mic as needed to ensure a good dynamic range. The lower the gain, the less chance of picking up unwanted sound (and noise – remember, this is live). Overheads on drums tend to pick up a lot of sound we don’t want, so bring them in as tight to the kit/cymbals as possible. On a loud stage I tend eliminate the overheads altogether and just close-mike the cymbals from underneath. This technique can also work well for straight-up live sound.

The place where each mic sounds the best to my old ears, and where it most rejects the other instruments and amps, is where it ends up. Keep stage rumble to a minimum. In addition to rolling off very low frequencies with EQ or high-pass filtering, I also make sure mic stands are in good shape and have rubber feet for isolation from the stage. If stage vibrations entering the mics are a problem, use shock mounts. When recording outdoors, keep windscreens handy. A lot has been written about “correct” mic placement for recording but I follow a simple philosophy. In sound check, I put on a pair of headphones with a long extension cable, and then with the musicians playing, I move each mic about in different positions. The place where each mic sounds the best to my old ears, and where it most rejects the other instruments and amps, is where it ends up. After positioning the mics, a quick listen to a test recording confirms if the placements work, and it provides the www.ProSoundWeb.com

opportunity to ID any trouble spots and make adjustments before the show starts. With that in mind, here are some of my approaches with microphones for live recording. Vocals. Depending on the vocalist, I may use either a dynamic or a condenser but the focus is the narrowest

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pattern I can get away with, depending on the singer’s mic technique. If there are wedges, I try to position them at a 30- 40-degree angle, which is usually the null zone of the mic’s pickup pattern. For singers who hold the mic away from their face or down by their belt, I outfit them with a wireless headworn

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:: In Focus ::

mic positioned near their mouth, or worst case, clip on a lav. The goal is to capture the full vocal between the two mics, which can then be optimized later in the recording mix. Background/Multiple Vocalists. Many of us tend to use the same mic on every background vocalist so we have an easier job doing monitors. But when recording, I try to choose a mic that suits each vocalist’s voice, even if they all end up with different types. Kick Drum. I have two approaches, depending on the style of music. To capture the attack sound of a drum that has a hole in the front head, I place a large-diaphragm dynamic inside the drum, within 4 to 12 inches of the front head, pointed about halfway between the center of the drum and the rim. This is joined by a boundary flat-plate type mic sitting on foam or a pillow inside the drum, which captures more of the shell sound. If there isn’t a hole (common with jazz, for example), I place the large-diaphragm dynamic on the beater side to get the attack sound and use a standard-sized dynamic on the rear head to get some of the ring sound. Hi-Hat. A small-diaphragm con32

Live Sound International July 2014

denser is my “go to” mic, but a dynamic can also work well if it needs to be positioned where there’s a chance it will be hit with a drumstick. Snare. A single dynamic placed an inch or two away from the head, pointed near the rim, is my live approach. For recording I sometimes place a cardioid condenser a few inches from the bottom head to capture more of the “snap,” tailoring the position based on what I hear. Toms. Cardioid dynamics are a good choice, but small clip-on condensers designed for drums can work great.. Overheads. Cardioid or supercardioid condensers are my first choice, placed as low as possible and pointed mostly at the cymbals. Ride Cymbal. This is a must-have mic for me with both live and recording. A small cardioid or supercardioid condenser located about 6 inches under the cymbal halfway, between the bell and the edge, is a good starting point. Percussion. For conga, djembe and other small drums, dynamic cardioid or supercardioid near the rim, pointed toward the middle of the head, works well, making sure the mic isn’t in the

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way of the musician. For bongos, it’s a cardioid dynamic placed in between the heads, about 8 inches away. Clip-on condensers designed for percussion are also a good fit here. Grand Piano. This can be one of the easiest or hardest instruments to mike, depending on who you ask. I go for simplicity and normally deploy two small cardioid condensers. They’re placed over the strings near where the hammers hit, one located at about the middle of the bass single strings and one positioned about one-third inward from the high strings. Both are pointed away from the keys to reject pageturning noises. If the piano is full size, I also opt for a larger diaphragm mic over the low strings. A single boundary plate mic taped to an open lid can also work well in picking up the entire keyboard. Acoustic Guitar. Depending on the guitarist, the choice is one or two mics. I always point a cardioid condenser between the neck and soundhole, a few inches away from the guitar, and if extra tone is needed, a second large diaphragm condenser is pointed below the hole.

A guitar amp mic placement.

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:: In Focus ::

dynamic placed about 6 inches away and off center from a speaker, combined with a DI feed, works well. Acoustic Bass. On a quiet stage, a large-diaphragm dynamic on a short stand, pointed at one of the bass “f ” holes, produces a good result. On a louder stage, a cardioid dynamic “vocal” mic with the body wrapped in foam, stuffed under the tailpiece and pointed

at the bridge, picks up pretty well and does not get in the player’s way. Banjo. A small-diaphragm condenser is the first thing I grab for a banjo, aimed at the sound bridge and placed 6 to 12 inches from the head. Organ w/Leslie. A large-diaphragm dynamic about 6 inches from the bottom rotor joined by two small-diaphragm condensers for the top horn A ball-style mic for harmonica. This beauty is the Shure 520DX, a.k.a., “green bullet.”

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– one at each side of the cabinet about 6 inches from the spinning horn – captures the unique sound of this instrument. Make sure the mics can handle a decent amount of SPL. Horns. Brass instruments get loud, so I choose large-diaphragm dynamics that can withstand the SPL. For most players I simply place the mic in front of the bell at least 6 inches (and often a foot) away. For tuba, I’ve actually used a clamp to hang the mic inside the bell. Harmonica. Many players carry their own mic, but if not, a cardioid dynamic vocal-style ball mic is usually a solid first choice. If the signal is sent to an amp, the approach is the same as with a guitar amp. Audience. Usually we want a live recording to be just that: live. To capture the audience, I place a few shotgun-type mics at the stage wings on stands, pointed at the crowd and positioned higher than the first few rows (or that’s all they’ll pick up). I’ve also suspended cardioids over an audience with good results. Live recording doesn’t require dozens of different microphone models. Just take stock of what you have available, select the best ones for each application, be patient and diligent with positioning, and you’ll be good to go. ■

Senior contributing editor CRAIG LEERMAN is the owner of Tech Works, a production company based in Las Vegas. 34

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PERSPECTIVE PREPARED TO MANAGE Steps to a successful pre-production process. by Danny Abelson

spend a lot of time with, and they’re critical to your success. They understand their artists and can offer a lot of insight. Finally, I speak with the artists directly to make sure I have all of their preferences covered. “Always provide a copy of all documentation you generate – mic chart, console files if you’re mixing digital, etc. – to the production manager. There are circumstances where an engineer may get sick or self-implode, and having a complete set of documentation in the production office can be helpful in maintaining continuity. “Once your research is complete, it’s time to start on a shop order. The front of house engineer generally picks the mics, so talk with the sound company, review gear requirements, and usually you can get what you want. When I started with a number of clients, there was already a mic chart from the previous engineer. “I have rather simDave Natale kicking back at Right Coast ple tastes in mics, so Recording, his typically it’s out with recording studio in the Neumann U87s Pennsylvannia. and in with models from Shure, Sennheiser, and ElectroVoice. Dedicating the necessary time at this stage is essential to insuring you have the gear you need when arriving TALK TO EVERYONE at rehearsals. “When planning for band rehearsals, if “One other important note: you must it’s an act I’ve not mixed before, I start fit in with the backline crew. They were by talking with the production manager, there before you, and will probably be who will usually have a copy of the stage there after you’re gone. They can make and mic info from the previous tour. life easy or miserable, so make friends This is usually an excellent source of and keep them. They have an even more information. Next I usually talk with direct path to the artists than most engithe backline crew. These are folks you

������ WE’RE CONTINUING our discus-

sions with veteran independent touring engineer Dave Natale, this time focusing on pre-production. Dave’s prepared for band rehearsals, production rehearsals, and tours countless times, with preproduction rehearsals a critical process, where many important issues can be resolved before an act hits the road. Here are a few thoughts from Dave to consider when getting ready for a tour and transitioning to shows.

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neers. A mutual respect here will go a long way to your success.” THE PROCESS “During early rehearsals with a new act, I always go out of my way to spend time onstage. It’s the only way to really learn just what is coming off the amplifiers. Truth is, you need real musicians actually playing the music to make this a useful exercise (smiles). Normally, I just hang out and listen. Universally, bands absolutely love that I’m interested in what their instruments sound like. “Also, I always try to get a separate room with some isolation to mix in. I recommend using some big loudspeakers/monitors because if the band decides to come in and listen, you’ll need something that sounds impressive. If they listen to mixes through nearfields on the console, they won’t get the full effect of what you’re trying to do live. We’re not making recordings here, we’re trying to a) learn the material, and b) demonstrate what it should sound like live. “I prefer large full-range boxes like (Clair) S4s because they fit comfortably through doorways and are easy to stack; however, it may be simpler to use a few of the loudspeakers you’ll actually be using on tour. In the past I’ve used a few ( JBL) VerTec 4889s and 4880 subs, and (L-Acoustics) dV-DOSCs and subs. “The key is generating a big sound with enough low end. If you’ve ever listened to high-powered loudspeakers at close range (at a professional level), you www.ProSoundWeb.com

understand why I do this. It’s simply not any fun listening to small loudspeakers. There’s a difference between listening and hearing, and I prefer to hear. This puts me in the right frame of mind. “One time working with a new client, the principals came in to have a listen after the first night of rehearsals. I was a bit nervous but had confidence in my mixes. I rolled the tape, and they looked at each other and said ‘this sounds great.’ Presenting a decent mix on big loudspeakers really helped to earn their confidence and alleviated any questions as to what it would sound like during a show. “To this day, that particular client has never said a word to me audiowise, ever. No suggestions to turn this up or down. Nothing. They trust me,

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and I think it all comes from having made a good impression on our fi rst night together.”

Presenting a decent mix on big loudspeakers really helped to earn their confidence and alleviated any questions. SURVIVE THE FIRST SHOW “Sometimes just getting through the first show takes some ‘calm nerves.’ I’ve had tours where after four weeks of band rehearsals, the first show was in a stadium in a major market. In one instance, my very fi rst time with the

band in front of the PA was the afternoon of the first show. The band came out and sound checked two tunes, and I was asked, ‘Are you OK?’ My response: ‘I think so?’ What else was I going to say? “This was a far cry from the cozy confines of a mix room at rehearsals; more like ‘OK, here we go...’ A very high-profile act, with all of the media in the known universe on hand and celebrities galore crawling all over the front of house platform. Not an ideal circumstance for a first show with a new act, but one we must be prepared to manage if called upon.” ■

DANNY ABELSON enjoys writing about the subjective nature of reinforced sound and the human factors that are so critical to a successful event.

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TECH TOPIC

Measurement Glossary The unique language of audio analysis. by Pat Brown

❯❯❯❯❯❯ IN MY LIFETIME, the size of sophisticated audio analysis systems has evolved from table top, to under-airplane-seat, to computer bag, to cell phone. The cost has evolved from the price of a nice automobile to that of a Happy Meal. As such, there are more audio practitioners than ever equipped to perform sophisticated loudspeaker and room measurements. Those who make the decision to get serious about this are faced with the daunting task of learning the ropes in order to get meaningful data from their measurement platform. It comes down to signal processing, and the theory and principles are not unique to audio and acoustics. They are used by virtually every engineering field, and even spill into seemingly non-technical fields such as accounting and photography. Audio practitioners use software and hardware tools with deep signal processing roots. The good news is that textbooks and websites regarding every part of the measurement process are plentiful. The bad news is that you can Wikipedia-yourself until kingdom come and never cover it all. You will also find that the rigid definitions may not even seem to apply to audio, since they were not developed to measure audio systems. In many cases they are presented in the most concise form possible – as mathematical equations. I decided to cook up a glossary of some of the terms most frequently encountered when working with audio analyzers of all types. Since acoustic analyzers analyze audio signals, this glossary applies to them, too. I have relaxed the rigidity to communicate the concepts, resulting in definitions that, while less general, are more applicable to how audio practitioners use them. It is my hope that this will help you better understand your measurement platform of choice.

Figure 1: The Time and Frequency Domains.

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Time Domain

FFT iFFT

Frequency Domain

Figure 2: The FFT and Inverse FFT.

As one engaged in both web-based and in-person training, I have the benefit of observing first hand how those new to the field wrestle with these principles. I get lots of questions on a daily basis. I learned long ago that “What one is wondering, many are wondering.” In the future I can refer the investigator to this document when they are wrestling with getting meaningful data from their measurement platform. These are ordered in a way logical to learning measurement from the ground up, starting with general terms and then including the more esoteric. I wrap with a concise description of a real-world measurement session, using all of the terms from the glossary. Here we go… GENERAL TERMS ❯ Signal Domain – The X-axis (horizontal) of a 2D plot of an audio waveform, usually a captured impulse response (IR). The strength of the signal is plotted on the Y-axis, either in linear units (pressure or voltage) or as a level in dB. The two domains most often used to analyze signals are the time domain and the frequency domain (Figure 1). ❯ Fast Fourier Transform (FFT) – A mathematically efficient algorithm for determining the spectral content (frequency domain) of a time domain waveform. In measurement work, the FFT is usually performed on the impulse response (Figure 2). ❯ Impulse Response (IR) – A display of the signal amplitude vs. time of an impulse that passes through the system. Acoustic examples include handclaps and balloon pops in a room. It is the inverse-FFT of the transfer function, which can be captured using non-impulsive stimuli (e.g. pink noise or log sweep) (Figure 3).

Figure 3: The Impulse Response.

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❯ Log-Squared Impulse Response – A time domain plot that results from rectifying the IR and displaying it on a log vertical axis. It displays the relative levels of the various events, making it easier to judge whether or not an event is significant (audible) (Figure 4). ❯ Envelope-Time Curve (ETC) – Formerly the Energy-Time Curve, it displays the envelope of the impulse response. It is a sort of “smoothed” log-squared response, and can aid in interpreting it. Note that the term “ETC” is used loosely in measurement, and its exactly meaning is specific to the analysis platform (Figure 5). ❯ Time Window – A technique used to limit the application of the FFT to only part of the impulse response. This yields the transfer function for only part of the time record. A time window can be used to reduce the effects of room reflections on the transfer function, effectively allowing anechoic loudspeaker measurements to be made indoors. ❯ Symmetrical Time Window – Symmetrical time windows are used when there is significant energy on both sides of the direct sound arrival of the impulse response that must be rejected from the FFT. It is usually centered around the direct sound arrival (Figure 6). ❯ Shaped Time Window – A time window that is tapered at its edge(s) to avoid abruptly interrupting the time domain data. A symmetrical time window is tapered at both ends. There are various time window “shapes” available (e.g., Hann, Hamming, Blackman-Harris, etc.) (Figure 7). ❯ Half-Window – A time window that is tapered on the trailing edge only. Half-windows are used when the impulse is near the start of the time record, where there is very low energy ahead of the impulse arrival. Here the use of a symmetrical window is either unecessary or it could exclude the direct field arrival (Figure 8). ❯ Asymmetrical Time Window – A time window that is not tapered (rectangular) at the leading edge, but tapered at the trailing edge. The leading edge of an asymmetrical time window must always be placed before the impulse arrival (Figure 9). The leading edge of a half-window can be placed after the arrival of the impulse (Figure 8). The half-windows implemented by many analyzers are actually asymmetrical windows, and you will not find universal agreement on which is the “correct” implementation. ❯ Dual or Multi-Time Windows – The use of more than one time window length. This allows the room reflections to be excluded from the high frequency response, but allows the room into the measurement for computation of the low frequency response. This appears to emulate the way that humans perceive sound. In effect, at low frequencies (long wavelengths) the contribution of the room reflections become part of the woofer’s response, and cannot be separated from it, either by the listener or the analyzer (Figure 10). In general, symmetrical time windows are used by analyzers that collect the real-time transfer function. Half-windows are usually used when the operator manually selects the portion www.ProSoundWeb.com

Figure 4: The Log-Squared Impulse Response.

Figure 5: The Envelope Time Curve (ETC).

Figure 6: A Symmetrical Time Window (rectangular).

Figure 7: A Shaped Symmetrical Time Window.

Figure 8: A Half-Window.

of the impulse response to be transformed to the frequency domain (using cursors). The objective of all window types is to produce a frequency response magnitude response that is free of comb filters (frequency response ripples caused by the phase interaction of multiple sound arrivals), leaving the part that can be meaningfully improved using an electronic equalizer. July 2014

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:: Tech Topic ::

Figure 11: Transfer Function of a Bandpass Filter.

Figure 9: An Asymmetrical Half-Window.

Figure 10: A Dual Time Window.

❯ Finite Impulse Response (FIR) – An impulse response of fixed time length. An example is a wave file of the IR of a room or loudspeaker, but it could also be a high or low pass filter used to form a crossover network. ❯ Infinite Impulse Response (IIR) – An impulse response that is generated on-the-fly with feedback of previous sample values. In theory, such an IR would never decay to zero. Analog filters are IIR, as are some digital filter types. In general, IIRs have lower latency than FIRs. ❯ Frequency Response Magnitude – A measure of the relative or absolute level of the signal vs. frequency. There is information about “how much” but not about “when.” ❯ Frequency Response Phase – A measure of the relative or absolute phase of the signal vs. frequency. There is information about “when” but not about “how much.” In the vast majority of system tuning applications, the display is of the phase response relative to a time reference, which is usually the arrival of the direct sound field from the loudspeaker (its impulse response). This time reference is selected by the operator, or automatically determined by the analysis software. ❯ Transfer Function – A display of both relative magnitude and relative phase of the frequency response on the same plot (Figure 11). It is the FFT of the impulse response, and the IR is the iFFT of the transfer function. This allows the observe to exploit the strengths of either domain when analyzing the response. ❯ Absolute Phase Response – The phase response displayed 40

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using “time zero” (the time origin of the signal) as the reference. It always begins at zero degree, and goes negative with increasing frequency. This is required by causality, which says that the signal cannot arrive before it is emitted. The absolute phase response can go negative by many thousands of degrees, depending on the “time of flight” of the signal between source and receiver. It is not very useful for measurement work, but is used extensively in computer room modeling, where the time relationship between virtual loudspeakers in a virtual space must be considered. ❯ Relative Phase Response – The frequency response phase using a user-selected “time zero” which is selected to make the center frequency of the bandpass filter (e.g., loudspeaker) go through zero degrees at its center frequency. The “correct” time zero is the one that produces the least phase shift through the pass band of the filter. Unlike the absolute phase, the relative phase can go positive (Figure 12). The relative phase response is an indicator of how well the system an preserve the shape of an audio waveform that passes through it. ❯ Minimum Phase Response – A given magnitude response can have an infinite number of phase responses, depending on the frequency-dependent delay of the signal passing through the system. The minimum phase response represents the minimum possible phase shift for a given magnitude response. It can be calculated from the magnitude response using the Hilbert Transform. It serves as reference for comparison with the measured phase response of the system. Raw transducers are usually minimum phase. ❯ Non-Minimum Phase – Phase shift in excess of the minimum phase response for a given magnitude response. Loud-

Figure 12: Minimum (top) and Non-Minimum Phase Response of a Bandpass Filter.

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:: Tech Topic ::

speakers with analog or IIR crossover networks are often non-minimum phase (Figure 12). ❯ Group Delay – A “joint domain” display of time (y-axis) vs. frequency (x-axis). The phase response can usually be displayed as “group delay” to make it more intuitive for determining the arrival times of various chunks of the spectrum (such as the woofer or tweeter response) (Figure 13). ❯ Linear System – A 1:2 increase in the input signal level produces a 1:2 increase in the output signal level. Compressors and limiters are non-linear, and should be bypassed for measurement work. ❯ Time-Invariant System – The response of the system under test is stable vs. time, making the measurement exactly repeatable. Moving the loudspeaker or mic during the measurement would produce time variance. Since the Time and Frequency domains are mathematically related in an inverse way (for linear, time-invariant systems) the weaknesses in one domain can become strengths in the other. For example, the time domain is more useful for looking at “when” but the frequency domain is more useful for looking at “what.” Most analyzers let you toggle back and forth between the time and frequency domains, and some display both simultaneously.

TF = 1 F = 1/T T = 1/F ❯ Comb Filter – A series of peaks and dips in the transfer function that are equally spaced in frequency. A comb filter is always result of multiple energy arrivals in the time domain, within the time window being observed. Since it is caused by the time/distance relationship between two sound arrivals, a comb filter will be unique at every measurement position. This makes the response impossible to correct by equalization (Figure 14). ANALYZER TYPES Now, some traditional, common-usage definitions of analyzer types and measurement methods: ❯ Real-Time Analyzer (RTA) – Measures the frequency response magnitude in real-time. The data is usually octavebanded (or fractions thereof ) and results from pink noise being played through the system. There is no phase data, so this

Figure 13: Magnitude and Group Delay of a Bandpass Filter. 42

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Figure 14: A Comb Filter caused by multiple energy arrivals.

response is “time blind.” The RTA displayes the magnitude of each 1/n-octave band, but nothing about their time relationship. An RTA measures the stimulus along with how it is modified by the loudspeaker and room. Early RTAs were basically a bank of sound level meters driven through fractional-octave filters. Modern RTAs measure in the time domain and compute the spectrum using the FFT. The data can optionally be octavebanded for display. This emulates the traditional RTA. RTAs are the simplest “frequency analyzers” to use. Just place the mic, excite the system, and observe the response. But since the measurement is blind to the arrival time (and direction) of the sound, the observed response may have little to do wit what a human perceives when listening at the same position. This is an aspect of measurement that the more sophisticated measurement processes attempt to address. ❯ Dual-Channel FFT – Measures the impulse response (or transfer function) by comparing the signal fed into the system to the signal re-acquired through the measurement microphone. A dual-channel FFT measures what happened to the reference signal, not the signal itself. As such, many different broadband stimuli can be used (even music). ❯ Discrete Method – The system is excited with a stimulus, and the resultant IR is manually time windowed (by the placement of cursors to mark the window edges) and the transfer function observed. Each is a discrete step. If I make changes with an equalizer, I will need to repeat the measurement to observe the effect of the filter. ❯ Real-Time Transfer Function – The process of collecting the time domain signal, comparing it to the reference signal, time windowing, and FFT-ing to the frequency domain, done at such blinding speeds that the transfer function is given in pseudo real-time and updated continuously. It is a “running” measurement that can allow system tuning to be performed during a show. The effect of filtering is immediately seen and www.ProSoundWeb.com

heard, since the measurement is ongoing. ❯ Coherence Plot – A frequency domain plot used with the real-time transfer function that indicates the likelihood that the observed response is valid, based on the input and output signals of the system. High coherence suggests good data. For the discrete method, repeatability is the “acid test” for good data. STIMULUS TYPES ❯ White Spectrum – A stimulus with equal energy per Hz. Examples include an impulse, white noise and linear sine wave sweeps. White spectra have energy that increases with frequency (since there are more “Hz” in the higher octaves than the lower ones), and typically yield poor signal-to-noise ratio at low frequencies (Figures 15 and 16). ❯ Pink Spectrum – A stimulus with equal energy per octave. Examples include pink noise and log sine wave sweeps. A pink spectra voltage contains more low frequency energy (and less high frequency energy) than an equivalent white spectra voltage. Most general sound system testing methods utilize pink spectra. ❯ Narrow Band Stimulus – A stimulus having limited spectral content, such as a sine wave (Figure 15). ❯ Broad Band Stimulus – A stimulus having broad spectral content, typically covering the entire audible spectrum. Examples include unfiltered noise and impulses. PUTTING IT ALL TOGETHER Here is an example of how all of the terms in the glossary might be used to describe the measurement process used for sound system tuning session: I wish to improve the frequency response of my loudspeaker through equalization. I will collect the impulse response using a pink spectrum signal (pink noise or log sine sweep), fed into the system from the sound card output of my dual-channel FFT measurement platform. This signal is also looped back into one of the inputs, allowing the analyzer to compare it to the signal picked up by the measurement microphone. The resultant IR is displayed as a log-squared response to better visualize relative levels of the sound energy arrivals. I bypassed the compressor/limiter blocks of my DSP, to assure that the system response is linear. The mic was placed on the floor plane, and I made sure that no one was moving near the mic or loudspeaker. This, along with turning off the ceiling fans assured that the system was time invariant. A time window is placed around the direct arrival, to exclude room reflections. Performing an FFT produces a transfer function that is free of comb filters. A shaped time window is used to avoid an abrupt interruption of the time domain signal, and the resultant anomalies in the frequency domain. Equalizer filters are placed on bumps in the frequency response magnitude. Since these bumps are minimum phase, the equalization process also improves the loudspeaker’s phase response. The phase wrap at www.ProSoundWeb.com

Figure 15: Spectra of test stimuli.

Figure 16: Energy of test stimuli.

crossover of the 2-way loudspeaker is non-minimum phase due to the IIR crossover filters used in my digital signal processor. I have to live with it. The phase response could possibly be improved by using FIR digital filters, at the expense of increased overall group delay. Since I am tuning the system before the event, I am using a discrete measurement method. While any broad band stimulus could be used, I will use one with a pink spectrum so as to provide equal energy to the system in each octave band. This assures that I don’t smoke my tweeters while trying to energize the subwoofer. Later, I will monitor the response during the event using a real-time transfer function to assess the effect of the increased absorption presented by the audience as the room fills up. Since my earlier measurements used a mic placement that ignored the floor reflection, minimal adjustment of the equalizer is needed for “audience compensation.” The coherence plot indicates whether the data can be trusted. It is needed because the spectrum of the music signal is continuously changing, and there may be time variance caused by people moving around. CONCLUSION That’s a general overview of the terms and their application. The principles are universal and apply to all measurement platforms. Some of the definitions were slanted toward their general use in audio and acoustic measurements. As with all theoretical concepts, some of the definitions could be challenged on the fine details, especially with regard to their general use outside of audio and acoustics. That’s one of the reasons that I love this business. ■

PAT & BRENDA BROWN lead SynAudCon, conducting audio seminars and workshops online and around the world. For more information go to http://www.synaudcon.com. July 2014

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SPOTLIGHT FLEXIBLE CAPTURE Recording options of digital consoles. by Live Sound Staff

������ DIGITAL MIXING CONSOLES provide wealth of capabilities for recording, designed to provide simple onboard 2-track capability to interfacing directly with computer-based multi-track recording systems. Whether configured to do so from the factory or by using optional output cards, many consoles can output MADI, a digital protocol with 64 channels of audio, or ADAT optical, a digital protocol that sends eight channels down each optical cable. Both of these are used by many recording systems for multichannel audio transport. AES/EBU and S/PDIF are two other common digital audio protocols to interface recording and playback equipment. Digital snake systems, stage boxes and networks present further advantages when it comes to recording. Instead of having just one isolated output located at the snake head, many digital transport systems have multiple splits that can be placed anywhere along the network, accommodating remote recording, webcasts, broadcast feeds, and any other sends that may be required. All of that said, we thought it instructive to take a look at some specific capabilities of current digital console series. Note that this isn’t intended to be comprehensive, but rather is a roundup of highlights that can serve as the basis for your own further investigation. Yamaha QL Series. Provides convenient recording capabilities for everything from basic 2-track to multi-track recording and playback. A standard USB flash drive plugged into the front44

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panel USB port serves as media for direct 2-track recording in mp3 format, where, for example, the recording can be handed to performers as soon as the show is finished. Sound files in mp3, AAC, or WMA format saved on the flash drive from a computer or other source can be played back as well for handy background music or sound effects without the need for extra playback equipment. On the other end of the spectrum, with Dante Virtual Soundcard software it’s possible to transfer audio directly to a Windows or Mac computer connected to the Dante network. With an appropriate DAW such as Steinberg Nuendo Live (sold separately) running on the computer, up to 64 tracks can be recorded simultaneously. It’s a great way to capture professional caliber live performances and also is useful in creating the tracks needed for virtual sound checks. DiGiCo SD Series. Capabilities vary depending on model but suffice to say there’s plenty of facilities. For example, the proprietary Stealth Digital Processing engine applied to the

SD7 provides 896 simultaneous optical, 224 MADI, 24 AES/EBU and 24 analog connections. Further, the SD Rack supplies up to192 kHz high resolution analog I/O converters and a choice (via option cards) of multiple digital formats, including MADI, AES, and ADAT. Users can also select other sample rate options for specific outputs – MADI at 48 kHz for recording feeds, for example. UB MADI presents another option, feeding a MADI stream in and out of a PC or Mac via USB. Bus-powered, it uses a USB-B type socket and standard cabling, taking up minimal space while providing quality location recording or a virtual sound check system. In addition, DiGiGrid MGB (coaxial link) and MGO (optical link) interfaces foster plugging in a coaxial MADI-enabled device to Waves SoundGrid for recording, processing and playback of up to 128 audio channels. It can even record to two computers simultaneously. Soundcraft Vi Series. The new Vi3000 provides integration into Dante audio networks and access to DAWs for live multi-track recording and virtual sound checks via MADI. Also included are MIDI, USB and Ethernet ports, along with a DVI output and four channels of AES I/O. And optical MADI interface is fitted as stan-

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dard, allowing direct connection to a Pro Tools HD recording system via a third-party converter box or any MADI compatible device. The ADAT card provides two optical 8-channel ADAT inputs and outputs, with selectable 44.1/48/88.2/96 kHz operation. Optical inputs and outputs are provided on Toslink connectors and can be used to record to, for example, a hard disk recorder or other device with ADAT inputs and outputs, as well as receive playback. In addition, the MADI card offers a simple recording solution for the Vi Series. Additional MADI cards can be fitted by exchanging with other I/O cards. And, both standard and compact stage boxes offer expansion slots for Studer D21m I/O cards, allowing connection to most popular digital formats and also accommodating a MADI recording interface. Allen & Heath GLD Series. Provides the ability to record and playback a stereo signal on a USB memory stick, and at the other end of the spectrum, standard iLive audio I/O option cards for Dante, MADI, EtherSound and Allen & Heath’s ACE protocols can be fitted to foster multi-channel recording/playback. For example, M-Dante, M-MADI and M-MMO cards are all available for GLD to enable integration with other systems, including multitrack recording. These cards can be fit to the I/O module expansion slot in a GLD-80 mixer. The Mini Multi-Out card provides a variety of formats of multi-channel digital output at 48 kHz sampling rate, including ADAT (three optical ports for www.ProSoundWeb.com

up to 24 tracks). and iDR (two 8-channel links to the iDR Series installed product range). Any GLD signal can be patched to any of the 56 outputs for flexible recording. GLD can transport up to 16 signals directly to the iDR-8 and iDR-4 digital mix processors, and also use the 8-channel iDR-out (analog XLR) and iDR-Dout (AES, SPDIF, Toslink digital audio) output expanders for remote feeds. SSL Live. MADI I/O connects the SSL Live-Recorder option, a 1RU device that can record 64 tracks at 96 kHz continuously from the console’s input stage and play back the channels in sound check mode. It exports/imports native (.ptf format) projects directly to/from Pro Tools and to/from Apple XML and Steinberg XML. Connectivity is via standard optical MADI so it can connect over long distances directly to any MADI-equipped digital console, as well as a venue’s audio distribution infrastructure (i.e., Riedel and Optocore), or routers. The Live-Recorder system consists of a fully configured 1U PC outfitted with a 128-channel SSL MADI audio interface and with Soundscape

V6.2 recorder/player software, and it has four front-loading RAID bays prefitted with two SSD drives. It’s connected to the console (directly or via a router) using 2 x 64 channel pptical MADI connections, and supports MTC and MMC via MIDI over Ethernet (or any other MTC/MMC-capable USB synchronizer or MIDI interface) for external system transport control, and it can sync via MADI or word clock (via BNC). The software also offers crash recovery routines which will retrieve incomplete audio recordings on reboot after a host system catastrophic failure such as sudden power loss. Avid S3L. The open networked architecture and modular nature of this platform presents a high degree of flexibility. Simply connect a laptop (with Pro Tools or other DAW installed) to the mixer’s Ethernet AVB network (using a single Cat-5e cable) for up to 64 tracks of audio recording/playback.

VENUE Link makes it straightforward to control live mixing and recording/ playback setups as one, and users can also perform virtual sound checks globally or on a per-channel basis with the input switch feature, which enables performers to sound check live alongside pre-recorded tracks. Further, with complete EUCON support, the S3L can be used as a standalone control surface to mix sessions recorded in Pro Tools, Logic, Cubase, and other popular DAWs. And, it can function as a 4 x 4 I/O device for recording in the studio as well as remote locations like hotel rooms or tour buses. July 2014

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:: Spotlight ::

Midas PRO Series. The DL371 processing engine is loaded with four modules for a PRO3, five for a PRO6 and six for a PRO9 in the standard configuration. The engine has dual-redundant HyperMac ports (both Cat-5e and optical), and there are also eight AES50 ports that facilitate connections to three different stage boxes and/or other AES50 I/O hardware. It is also possible to use a PRO3, PRO6 or PRO9 with a DL431 stage box, Klark Teknik DN9696 audio recorder (up to 96 tracks), and DN9650 network bridge, which offers the ability to convert the Midas AES50 format to just about any third-party platform. Another interesting approach with a Midas PRO2 was presented in LSI (May 2013 issue) by Todd Hartmann, audio engineering coordinator for The Austin (Texas) Stone Community Church. Devised by Jim Roese of RPM Dynamics, the RPM-TB48 I/O is a stand-alone solution with no external interfaces required, providing a 48-channel, 96-kHz, 24-bit recording/playback solution. It utilizes a pair of Lynx Studio Technology AES50 to PCI cards, all mounted in a Sonnet Thunderbolt chassis. Because the processor load of the conversion is being handled by the interface, the load on the CPU of the recording computer is very low. A pair of Neutrik Ethercon cables connects the recording interface to the PRO2 via two of the AES50 ports on the console surface.

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PreSonus StudioLive AI-series. In a straightforward approach, a pair of bi-directional FireWire s800 (IEEE 1394b) ports connect StudioLive AI consoles to a Mac or PC for recording. The largest model, the StudioLive 32.4.2AI, has an integrated, bi-directional recording interface that can send up to 48 audio streams to a computer

and return up to 34 playback streams (48 x 34/40 x 26/32 x 18 streams available) at 24-bit/44.1/48/88.2 (and 96 kHz support is coming in fall 2014, according the company). The FireWire s800 and Ethernet ports come on a preinstalled card that is user-replaceable with optional Dante, AVB, or Thunderbolt cards. Designed specifically for StudioLive mixers, Capture 2.1 software adds proprietary Active Integration networking,

offering fast setup and recording directly from the mixer, with auto configuration. It also provides convenient, automated virtual sound check. Roland V-Mixer Series. Enables three types of recording and playback solutions: onboard stereo recording via USB port, integrated multi-channel recording and playback via the company’s R-1000 48-track recorder/player, and integrated multi-channel recording using the proprietary REAC platform. The R-1000 can be used with any REAC digital snake as well as with any digital console with MADI output capabilities by using the Roland S-MADI REAC MADI bridge. The REAC driver kit is intended for use with all V-Mixer consoles is also compatible with Roland S-4000S, S-2416, S-1608, S-0808, and S-MADI digital snake systems. In addition, up to 40 channels from a V-Mixing console or digital snake can be routed directly into most ASIO-based DAWs via Cat5e/6 connected directly to the gigabit network port on a PC. QSC TouchMix. Yes, they’re a bit on the smaller scale for the purpose of this discussion, but we wanted to point out that these new miniscule mixers are capable of direct recording to an external USB hard drive – no external computer is required. All inputs plus a stereo mix are created in 32-bit WAV format. Tracks can also be played back on the mixer or imported into most DAW software for over-dubs and post production. Mackie DL Series. Also very compact in form factor, these iPad mixers provide the ability to record stereo tracks directly to the iPad. It probably doesn’t get any easier than that. n www.ProSoundWeb.com

StudioLive AI speakers are available in three full range sizes and a matching 18-inch subwoofer. ©2014, PreSonus Audio Electronics, All Rights Reserved. Sceptre, CoActual and StudioLive are trademarks of PreSonus Audio Electronics. Temporal EQ is a trademark of Fulcrum Acoustic. Dante is a trademark of Audinate.

Studio monitor sound quality…

…only way

louder.

Y

es, that’s a bold claim. We back it up with a boldly different, 3-way CoActual™ physical design, and Fulcrum Acoustic™ Temporal EQ™ DSP — much the same as we use in our acclaimed, high-end Sceptre™ studio monitors. Plus 2000 watts per box — the most of power in their class — and the freedom of iPad® adjustment and monitoring with SL Room Control. And Dante connectivity. But enough technical huzza-guzza. Freaking hear StudioLive AI PA at a PreSonus dealer and/or when our traveling Thunder Road Show comes to your area.

StudioLive™ AI CoActual™ from

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SNAPSHOT

AND THEY’RE OFF... An audio makeover at Churchill Downs. by Live Sound Staff

������ CHURCHILL DOWNS in Louisville, site of the Kentucky Derby since 1875, received a significant sound reinforcement system upgrade in time for the 2014 season. Unlike baseball and football stadiums, the grandstand at Churchill Downs has seating tiers that are stacked vertically, straight up and down. In addition, various expansions over the years have added sections on either side of the historic “twin spires,” and all of these sections have slightly different profiles. Further, there are varying ceiling heights, seating depths and column spacing. It’s a unique situation requiring careful planning, and unfortunately, Churchill Downs did not have CAD 48

Live Sound International July 2014

in the adjacent clusters. Both types also include a rear-firing full-range Danley SM60F aimed toward the top of the Level 300 seating tier and a “more-or-less downfiring” SM96 with its woofer removed. “Danley had slightly greater vertiThe new Danley clusters delivering cal coverage patterns coverage to the grandstand, which is in similar box sizes shown in perspective in the inset image. [to those of the other manufacturer considered for the project], which allowed us to modify our design to use one less box per cluster,” notes Dave Marsh, owner of Marsh/ PMK. “That would ultimately be a cost savings. And Danley emphatically stated that the boxes would be delivered drawings of the facility. Further, PDF on time and that sealed the deal. They drawings provided to Marsh/PMK were made good on their promise.” not to scale. Tim Lindstron and MelIn addition, Renkus-Heinz, QSC vin Saunders of the sound team worked Audio and Community loudspeakers together on a solution. were re-purposed and added as necesLindstrom used dimensions obtained sary to other audience/public areas. The during the initial site survey to creexisting QSC Audio Q-Sys DSP, netate re-scaled PDFs, while Saunders working and routing infrastructure was took to Google Earth to confirm/ correct the dimensions and then created a SketchUp model of the grandstands. The SketchUp model was then imported into EASE, which allowed Dave Stearns of Encompass to get into the detailed loudspeaker design. Two basic cluster types alternate along the length of the grandstand, just under the front edge of the Level 300 Marsh/PMK principal Dave Marsh (left) ceiling. There are nearly sixty clusters and Dave Stearns of Encompass at in total. The first type includes a DanChurchill Downs. ley Sound Labs SH50 Synergy Horn expanded to handle the new requirelong-throw loudspeaker and a companments, accompanied by increased QSC ion TH212 subwoofer. The SH50 covpower amplification. ers seating in front of the grandstand “Re-use of existing equipment where building all the way out to the edge of possible was a goal of our design, but the track. Danley now provides the major audio The second cluster type utilizes horsepower for the grandstands at Danley SM96 compact loudspeakers Churchill Downs,” Marsh concludes, to provide near coverage in between the claiming no pun intended. n horizontal cut-off angles of the SH50s www.ProSoundWeb.com

LESS IS MORE IN A WORLD THAT IS HANGING MULTIPLE SPEAKER CABINETS LINKED TO NUMEROUS PROCESSORS PROGRAMMED BY VARIOUS PREDICTIVE SOFTWARE.. ONE COMPANY HAS FOUND A BETTER WAY

The KV2 VHD2.0 delivers clear detailed audio - further and louder than any other single speaker cabinet on the planet. This three-way cabinet consists of a 3” compression driver with NVPD treated dome assembly, two hornloaded 8” drivers featuring AIC Technology, and two horn-loaded 12” midbass drivers. Combining proprietary amplification with various high-output sub configurations, VHD is easy to transport, fast to set up, efficient on power, and can be tailored to virtually any application. VHD simply delivers more for less!

KV2 BUILDS PLUG-AND-PLAY SYSTEMS THAT SAVE YOU MONEY PROVIDE FAST AND EASY SETUP COVER VENUES OF ALL SIZES ...AND SOUND AMAZING

The ES1.0 is a compact, active, three-way mid high cabinet that can be combined with a variety of ES Series subwoofers and is driven by KV2’s proprietary EPAK. ES Series systems have redefined the formula for size-tooutput, reducing transport costs and setup times for nearly a decade. As we continue to strive to make them better, the ES Series is now available with 20MHz on-board sampling for time alignment and phase correction producing sound quality of extraordinary detail and clarity.

Our new standard defines the key elements for building sound systems of outstanding quality from the ground up - incorporating ultra-fast electronics, superb pulse response, 20MHz digital sampling and superior sound design. We call it Super Live Audio.

George Krampera is truly a Pro Audio pioneer. Through his long and successful career, he has designed speaker products for a number of companies that have been used and enjoyed by millions worldwide. At KV2, George’s vision is to eliminate distortion and information loss in the signal path, providing sound reproduction that has true dynamic range and representation of the source. KV2 is the culmination of George’s life-long quest for perfect sound.

FOR MORE INFORMATION OR TO FIND YOUR NEAREST KV2 DEALER VISIT WWW.KV2AUDIO.COM

Shure GLXD24 With Beta87A Capsule Evaluating a 2.4 GHz wireless microphone system. by Craig Leerman

❯❯❯❯❯❯ THE SHURE GLXD digital wireless microphone system combines intelligent technology and smart power options into a compact, cost-effective package. The system operates in the 2.4 GHz band and offers automatic frequency management as well as rechargeable transmitter batteries. Transmitter options include handheld units equipped with SM58, SM86, Beta58A or Beta87A mic elements, belt pack transmitters that can be used with a variety of lavalier and headset mics, and instrument mics and a cable for backline applications. Up to four compatible systems can operate together in a standard setting, with up to eight systems maximum in optimum conditions. The stated operating range of the transmitter for indoor usage is up to 100 feet typical, with a maximum distance of 200 feet from the receiver under ideal conditions. Outdoors, up to 65 feet is typical, with a maximum distance to the receiver of about 165 feet under ideal conditions. The maximum use time for the continuous rechargeable transmitter battery is stated as 16 hours. THE OVERVIEW Out of the box I was pleasantly surprised to find that the system ships with a sturdy foam-lined soft sided zippered case that holds all the components and has room for some accessories. The specific system I evaluated consisted of a GLXD2 handheld transmitter with a Beta87A supercardioid condenser mic element with a standard GLXD4 receiver. (There is also a nifty GLXD6 wireless guitar pedal – an all-metal receiver with built-in tuner that’s avail50

Live Sound International July 2014

Shure GLXD4 receiver and GLXD2 transmitter with Beta87A mic element.

able for musicians.) The GLXD4 receiver has a rugged yet lightweight half-rack-sized body with two permanently attached antennas on the front. A large screen flanked by a few push buttons allow the user to easily configure the system as well as monitor status of frequencies/channels, audio signal from transmitter, battery life of transmitter, and system gain. A battery charging bay is also situated on the front (more on this later). The rear of the receiver offers both XLR and 1/4-inch output connectors, a power switch, DC power input on a locking (thank you!) jack, and a USB port for uploading fi rmware updates. The power cord is of the “lump” style that places the transformer in the middle of the cable instead of the “wall wart” style that places it at the plug end. I prefer this style of cable because it lets me use a regular power strip and also allows for a cleaner rack cable scheme because I can securely mount the “lump” to the side wall of a rack with a few screws. The GLXD2 handheld transmitter has a rugged plastic body with a large side-mounted (and recessed switch), and status LED. The LED changes colors and flashes to indicate different

conditions. During normal operation the LED glows green, while f lashing green indicates the transmitter is searching for or linking with a receiver. A steady red light indicates about one hour of battery life is left, and flashing red tells the operator that only about a half hour of battery life remains. An amber light signals that there is a battery error. A very useful feature to help a user identify which mic is linked to a specific receiver is called Remote ID. Pressing a button on the receiver causes the LED to flash red and green on the transmitter, easily identifying the unit you’re looking for. Another handy feature is that the power switch on the transmitter can be configured to operate normally or be locked so that it will not turn the transmitter off one it has been turned on. GLXD transmitters are powered by Shure SB902 lithium-

Inserting the battery into the charging bay on the front of the receiver.

www.ProSoundWeb.com

ion rechargeable batteries that are stated to have zero memory effects, eliminating the need to discharge batteries prior to charging. Charging can be accomplished when the battery is inside a transmitter or when removed. When installed, the included USB cable can be connected to a computer or wall outlet or car adapter. The battery can also be inserted into the receiver’s charging bay, and a spare battery can be kept inserted in the bay during use. While a spare battery is always a good idea when using wireless equipment, it should not be needed with the GLX system as the manual states that a 3-hour charge using the receiver battery bay results in up to 16 hours of continuous usage. A partial charge of 1 hour provides up to 6 hours of use, and even quick charges of 15 and 30 minutes give respectable usage times of up to 1.5 and 3 hours, respectively. The battery is rated for up to 10,000 hours of use before needing to be replaced. While I’ve not been a big fan of rechargeable batteries in wireless mics, the long run time and remote battery life readout available on the receiver gave me no reservations about using the GLXD system at a show. ON THE BENCH It was easy to connect the system because of the choice of XLR and 1/4-inch connectors. Set-up was simple, and users are provided with a choice of three groups of frequencies to operate in. The initial factory setting is Group 1, and it allows up to four systems to operate together. In this configuration, the systems can switch to backup frequencies if interference is detected. Group 2 also offers backup frequencies in case of interference, and it allows up to five systems to operate together. Group 3 allows up to eight units to coexist, but backup frequencies are not www.ProSoundWeb.com

The GLXD6 wireless guitar pedal receiver.

available, so it should only be used in controlled WiFi environments. All receivers at a location must operate in the same group. Once a group is selected, pressing a link button syncs the transmitter to the receiver. I had no trouble linking the units and getting the system working. With the receiver on my test bench, which is located at the front of my warehouse, I found I could easily walk around my entire warehouse (full of gear and metal pallet racking) with the transmitter without experiencing any drop-outs or audio glitches. I even walked out the back door and it still transmitted clearly until the metal door fully closed. The audio quality was great, even with the transmitter located more than 100 feet away from the receiver. Satisfied that the system was working correctly, I decided to use the transmitter to test a new mic head that just arrived at the shop. It was then that I realized that the transmitter does not allow the swapping of heads, as is the case with many Shure systems. This is certainly not a deal breaker, but worth noting.

IN THE FIELD Packed back into the zippered case, I took the system out on a couple of gigs. The first show, it served as an announce and podium microphone. Looking at my frequency scanner, it was obvious that we were in a pretty crowded WiFi area, but there were no interference issues. (I was using Group 2.) For those not familiar with the Beta87A element, it’s quite simply a fantastic-sounding supercardioid mic, one of my favorite “go to” choices for both male and female vocals. And with the GLXD system, the Beta87A sounds the same as if you plugged a cable into the back of the mic, and that’s exactly what you want from a wireless system. The next event was a community play where the GLXD was used both onstage and in the wings for dialog. Again, it was easy to conf igure the system, and using Group 1, the system performed flawlessly even though (again) the spectrum was crowded. (This is pretty much the case throughout Las Vegas.) Having backup frequencies that the system will silently switch to in case of interference takes away much of the worry about using it where the spectrum can be iffy. With its long-life batteries and multiple options for charging, the GLXD system won me over as a rechargeable battery wireless user. I really like the built-in recharging bay on the receiver. Rock-solid performance in the 2.4 GHz band, a variety of transmitter options, a cool floor pedal receiver/tuner for musicians and no more searching for new batteries – all at a great price point – makes Shure GLXD wireless system a winner. GL X D24 system with Beta87A element – $736 n

Senior contributing editor CRAIG LEERMAN is the owner of Tech Works, a production company based in Las Vegas. July 2014

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:: Road Test ::

QSC Audio PLD4.5 Checking out a new power amplifier/processing platform. by Danny Rosenbaum

The QSC PLD4.5 power amplifier.

I’LL ADMIT: sometimes it’s hard to get excited about power amplifiers. If they do their job correctly, they make signal louder without creating a lot of mess or fuss. But if they do their job poorly, somebody doesn’t get called back for the next gig. In fact, we really only tend to notice power amps when they’re doing something wrong. When they clip or thermal, the result can be a jarring reminder of how bad things can be. Sometimes the evidence of an amplif ier’s bad behavior can be quite dramatic. Square waves at the front end will eventually cause a fire at the back end. (Time to pull out the marshmallows!) Of course, I can’t remember a QSC amplifier ever leaving me stranded, so when I received the opportunity to check out one of the new PLD Series models – specifically, the PLD4.5, the most powerful model in the family – I jumped at it. THE ESSENTIALS All PLD Series models for portable/live sound (and CXD Series for installed sound) are available in a 2RU configuration with four channels, with independent DSP for every channel and a preset wizard that streamlines the setup process. DSP includes crossover 52

Live Sound International July 2014

filters, parametric EQ , alignment delay and other dynamics processing, and when paired with QSC loudspeakers, proprietary Intrinsic Correction processing techniques help optimize sonic performance. This is the kind of amp that can replace several pieces of gear in your rack. USB interfaces on the rear panels allow firmware updates as well as the ability to upload loudspeaker tunings and configurations and transfer data between amplifiers. Multiple DSP-controlled power saving modes combined with the Class D design increase efficiency, lower weight, and save money. The most significant aspect of the platform, however, is Flexible Amplif ier Summing Technology (FAST),

developed to improve power allocation by actively distributing total amplifier power across one, two, three or all four outputs, enabling channels to be combined for maximum current or voltage output, with the largest models capable of up to 5,000 watts continuous (8,000 watts burst power). This was the aspect about PLD that most captured my attention. OPERATING PARAMETERS Pulling the PLD4.5 out of the box at the shop, my first thought was construction quality. Hefty metal, solid build, and the overall feel of a piece that won’t let you down at very high SPL. A surprising thing was finding that the 8,000 watts of (burst) power it could deliver tipped the scales at a mere 22 pounds. (My road crew is going to be happy about this.) The menu structure was easy to get around and revealed some of the real power under the hood. Equalization, limiting, filters and delays can be recalled via presets that provide the ability to set your sound system parameters at the turn of a knob. The LED display gave me metering and key info, as well as access to all of the programming functions. I could read the display in all types of lighting and from any angle. Setting up the amp for a gig is quick and simple. Mute buttons for each channel as well as input gain and output

Front and back panel views of the PLD4.5.

www.ProSoundWeb.com

have for busy production companies. attenuation make it easy to maximize There are 20 factory presets that gain staging. It also sounds great, with primarily go through the different exceptionally solid low-end perforpossible system variations. The front mance even at high output. panel clearly shows these structures by I wasn’t surprised to see lots of procolor coding the channels to indicate cessing presets for QSC loudspeakif you have them set up for LF, HF or ers built into the PLD4.5. What did full range. Note that recalling the presurprise me, however, were all of the sets does take about three seconds, and presets for other brands. This is an amp the outputs are muted as you do this in that’s comfortable making a pair of biorder to keep the amp from outputting amp JBL cabinets go to “11” on a live any strange noises. music gig and then Once the amp driving four QSC It’s hard to imagine is dialed-in for the ADS12Ts covering specific needs of a a political convena gig that would tion the next. It’s a present a problem that given show, there really flexible amp. this amp won’t solve. are 50 user preset slots available It was also great for storing these to find that I could settings. If the capacity to store more do everything needed via the front than 50 is required, the free Amplipanel. Sometimes you need to tweak a fier Navigator software facilitates the parameter without hooking up a lapstoring and managing of an unlimited top, and that ability is really apprecinumber of presets. ated here. The user interface is driven by turning the data knob and pushing TIMES HAVE CHANGED an enter button. So if you can run an The back panel is also well-equipped, iPod, you can run this amp. with four XLR inputs with passthrough’s to go to another amp as well FLEXIBLE TEMPLATE as a USB port for preset management. The most impressive aspect of the Outputs are six Neutrik connectors – two PLD is the abilit y to completely for bridged outputs and four for indireconfigure the four amplifier chanvidual outputs to make connecting to nels. They can be split or combined in loudspeakers in any configuration simple. whatever format meets the particular Even the standard IEC power cord needs of the day. If you need to drive is a locking version to add a little bit four full-range side fills, you can use of security. It’s also worth noting that all of them as individual channels. If this is a 15-amp cable, probably due to the next day you’re in front of a subthe that these amps incorporate power woofer that needs piles of power, comfactor correction (PFC), so the whole bine all four channels to push a single multichannel amp, processing, and output of up to 7,500 watts – all the reconfigurable structure runs off of a way down to 1 ohm. single 15-amp circuit. That’s a pretty Three channels to push a sub and a impressive use of technology. single channel for the HF? No problem. It used to be that four channels of Need to push two biamp cabinets the amplif ication and processing would next day? Again, no problem. It’s hard to tie up eight or ten spaces in my road imagine a gig that would present a probrack and easily weigh more than 100 lem that this amp won’t solve. It’s the pounds. Now I can have all of it in two kind of flexibility that makes it a mustwww.ProSoundWeb.com

A look at the PLD4.5’s Preset Wizard.

rack spaces weighing 22 pounds. That’s the kind of improvement that makes a big difference for sound techs, engineers and sound companies alike. U.S. MSRP: PLD4.5 – $2,929.99; PLD 4. 3 – $ 2, 329.99 ; PLD 4.2 – $1,729.99 n

DANNY ROSENBAUM serves as a vice president for Morris Light & Sound, a leading touring company based in Nashville. July 2014

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>>>>

REALW RLDGEAR Lesser Scale, Big Sound Focusing on the latest medium-format line arrays. by Live Sound Staff

WE DEFINE “MEDIUM-FORMAT” line arrays as those with 10-inch to 8-inch low-frequency drivers. The definition is somewhat arbitrary, but no more so than the various designations that manufacturers use for their loudspeaker/ line array systems – one entity’s “miniature” is another’s “midsize” and so on. We’ve also included the NEXO STM, even though the “main” boxes are outfitted with LF drivers measuring just 6.5 inches. It’s because the overall concept of the system, quite modular in approach, couples dedicated low-frequency enclosures loaded with 12-inch woofers to the identically sized main enclosures within array structures that can range from quite small to quite large in terms of physical scale. When the modern version of line arrays first hit the market, the vast majority were of the large-format variety (12-inch and larger LF drivers), with subsequent introductions primarily defined by ever-decreasing footprints. Smaller enclosures are not only less expensive, but also weigh less and can bend more rapidly without breaking their coupling, due to the smaller diameter of their woofers. The physics of coupling dictates a limit to the angle from one enclosure to the next, beyond which beaming and spotty coverage occurs. While a line array with 15-inch LF drivers has a limit of about 5 degrees, enclosures based on 10-inch cones can bend by 10 degrees from one cabinet to the next. Lesser scaled line arrays can therefore provide a greater angle of vertical coverage in a shorter height, especially important at smaller venues. In addition, line arrays lose pattern control at frequencies whose wavelengths are longer than the array’s height. To provide pattern control down to 100 Hz, for example, an array must be 11 feet tall and, with a typical cabinet height of a foot or less, control down to 100 Hz requires 11 or more cabinets in an array. Increasingly, we’re seeing more loudspeakers available in self-powered versions or with custom external amplification and processing platforms, and this genre is no exception. Indeed, more models fit that criteria this time out than ever before. The same goes with companion, flyable subwoofers. The advantages are many, for numerous applications, when talking about line arrays of a reduced scale. Enjoy this look at medium-format line arrays. n 54

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Adamson Systems SpekTrix www.adamsonsystems.com Configuration: 3-way Weight: 62 pounds Size (h x w x d): 8.6 x 27.9 x 18.9 inches LF: ND8-L 8.5-inch Kevlar neo cone MF: ND8-M 8.5-inch Kevlar neo cone HF: B&C DE900 1.4-inch-exit compression driver Frequency Response: 100 Hz – 18 kHz Dispersion: 120 x 5 degrees; SpekTrix W is 120 x 15 degrees Power: 250/1000 W each for LF & MF (AES/peak), 110/440 HF (AES/peak) Companion Sub: SpekTrix Sub dual-18, flyable

Martin Audio MLA Compact �� www.martin-audio.com Configuration: 3-way Cellular Drive Weight: 109 pounds Size (h x w x d): 11 x 31 x 19.7 inches LF: Dual 10-inch neodymium cones, Hybrid slot-horn loaded MF: Dual 5-inch neodymium compression drivers, horn loaded HF: Quad 0.7-inch-exit neodymium Frequency Response: 65 Hz – 18 kHz Dispersion: 100 x 10 degrees Power: Self-powered, 500/180/40 W (AES) Companion Sub: DSX dual-18, flyable version available

Meyer Sound MICA �� www.meyersound.com Configuration: 3-way Weight: 150 pounds Size (h x w x d): 13.4 x 41.4 x 17.8 inches LF/MF: Dual 10-inch neo cones HF: Dual 1.2-inch exit neo compression drivers with REM Frequency Response: 60 Hz – 18 kHz Dispersion: 100 degrees horizontal; vertical array dependent Power: Self-powered, 4 channels – 2 x 950 W (LF/MF), 2 x 560 W (HF) Companion Sub: 700-HP dual-18 integrates into arrays www.ProSoundWeb.com

NEXO STM M46 �� www.yamahaca.com Configuration: 2-way Weight: 130 pounds Size (h x w x d): 13.8 x 22.6 x 28.1 inches LF: Quad 6.5-inch flat membrane LF-MF cones. B112 Bass module has dual 12-inch neo cones in a hybrid horn design HF: Quad 2.5-inch neo compression drivers (with Keton Polymer diaphragms) Frequency Response: 85 Hz – 19 kHz (M46); 55 Hz – 200 Hz (B112) Dispersion: 90 degrees horizontal; vertical array dependent Power: NEXO Universal Amp Rack feeds 12 modules in any combination in groups of three. Companion Sub: S118 single-18, flyable

Renkus-Heinz VARIA VA101 Series www.renkus-heinz.com Configuration: 2-way Weight: 64 pounds Size (h x w x d): 13 x 23.7 x 15 inches LF: RH model SSL10-7 10-inch cone HF: Dual RH model SSD1750-TN-8 1-inch compression drivers Frequency Response: 60 Hz – 20 kHz Dispersion: Horizontal – 90 degrees standard, 60 and 120 degrees available; transitional 60 to 90 and 90 to 120 degrees also available. Vertical – 7.5 degrees; 15 and 22 degrees also available. Power: Self-powered, RHOAN DSP/control; externally powered models also available Companion Sub: VA/VAX15S dual-15

d&b audiotechnik Q Series �� www.dbaudio.com Configuration: 2-way Weight: 49 pounds Size (h x w x d): 12.1 x 22.9 x 16.1 inches LF: Dual 10-inch dipolar neo cones HF: Single 1.3-inch exit compression drivers on toroidal waveformer Frequency Response: 60 Hz – 17 kHz Q1 Dispersion: 75 x 15 degrees Q7 Short-Throw: 75- by 40-degrees via rotatable CD horn Q10 Short/Wide: 110- by 40-degrees via rotatable CD horn Power: Passive, d&b D6, D12 or D80 amplification Companion Sub: Q-SUB single-18, integrates into arrays

EAW KF730 �� www.eaw.com Configuration: 3-way Weight: 77 pounds Size (h x w x d): 13 x 28.5 x 17.5 inches LF: Dual 10-inch cones, proprietary Phase Aligned MF: Dual 7-inch cones, horn-loaded HF: Dual 1-inch-exit neo compression drivers, horn-loaded Frequency Response: 80 Hz – 20 kHz Dispersion: 110 x 12 degrees Power: Bi-amped 700, 350 W (LF, MF/HF, both at 16 ohms); custom Powercube amplifier/DSP package also available Companion Sub: SB1000z, SB1001, SB1002 (all dual-18, SB1002 is flyable)

JBL Professional VTX V20 �� www.jblpro.com QSC Audio WideLine-10 WL2102 �� www.qsc.com Configuration: 3-way Weight: 83 pounds Size (h x w x d): 10.8 x 27.4 x 20.7 inches LF/MF: Dual 10-inch cones with 3-inch voice coils HF: Single 3-inch, 1.4-inch-exit neodymium compression driver Frequency Response: 55 Hz – 18 kHz Dispersion: 140 x 10 degrees (patented waveguide) Power: Bi-amped – 800/80 watts (LF/HF, AES); tri-amped – 400/400/80 watts (LF/MF/HF, AES) Companion Sub: WL218-sw (vented) and GP218-sw, both dual-18

www.ProSoundWeb.com

Configuration: 3-way Weight: 88 pounds Size (h x w x d): 11 x 35.4 x 15,8 inches LF: Dual 10-inch Differential Drive cones MF: Quad 2164H Ultra-Linear 4-inch cones HF: Triple D2415K Dual Diaphragm compression drivers Frequency Response: 53 Hz – 22 kHz Dispersion: 110 degrees x array dependent Power: Tri-amped, 1500/800/225 W (LF/MF/HF), Crown ITech HD or Crown VRACK recommended Companion Sub: S25 dual-15 and S28 dual-18 both flyable, G28 dual-18 July 2014

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:: Real W rld Gear ::

Bose Professional RoomMatch �� www.bose.com Configuration: 2-way Weight: 123 pounds Size (h x w x d): 16.9 x 39.1 x 23.6 inches LF: Dual LF10 10-inch cones HF: Six EMB 2 (2-inch voice coil) compression drivers on continuous-arc diffraction-slot manifold Frequency Response: 60 Hz – 16 kHz Dispersion: Numerous coverage pattern choices Power: Bi-amped, 500/150 W (LF/HF, long-term), PowerMatch Series recommended Companion Sub: RMS215 dual-15, flyable

D.A.S. Audio Event 210A �� www.dasaudio.com Configuration: 3-way Weight: 74.8 pounds Size (h x w x d): 10.6 x 28.7 x 14.4 inches LF/MF: Dual 10-inch cones, each operates in a specific frequency range HF: M-75 compression driver on waveguide Frequency Response: 70 Hz – 20 kHz Dispersion: 90 x splay dependent Power: Self-powered, 180/360 W each for LF, MF, HF (continuous/peak) Companion Sub: Event 218A dual-18

Montarbo RA16 �� www.montarbo.com Configuration: 2-way Weight: 39.6 pounds Size (h x w x d): 11.1 x 20.7 x 17.3 inches LF: Dual 8-inch neo cones HF: 3-inch voice coil titanium compression driver Frequency Response: 70 Hz - 20 kHz Dispersion: 120 degrees horizontal, vertical is array dependant Power: Montarbo PLM6800 amplifier/DSP recommended; otherwise, LF: 800/200 W (LF/HF) Companion Sub: RA18 single-18, flyable; RAB1815 single-15/ single-18; MOL1818 SubOne dual-18 (active version available)

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Live Sound International July 2014

WorxAudio TrueLine V10 �� www.worxaudio.com Configuration: 3-way Weight: 159 pounds Size (h x w x d): 10.5 x 48 x 18 inches LF: Dual 10-inch vented cones MF: Dual 8-inch cones HF: Single 1.4-inch exit compression driver on Flat Wave former Frequency Response: 47 Hz – 16 kHz Dispersion: 140 x 10 degrees Power: Self-powered, 1000/1000/500 W (LF/MF/HF) Companion Sub: TL.218SS extra-long excursion dual-18

RCF HDL 20-A �� http://rcf-usa.com Configuration: 2-way Weight: 63.9 pounds Size (h x w x d): 11.5 x 27.7 x 17.5 inches LF: Dual 10-inch (2.5-inch voice coil) cones HF: Dual 3-inch voice coil compression drivers Frequency Response: 55 Hz – 20 kHz Dispersion: 100 x 15 (max) degrees Power: Self-powered, biamp, 500/200 W (LF/HF, RMS), 1000/400 W (LF/HF, peak) Companion Sub: HDL 18-AS single-18, flyable

VUE Audiotechnik al-8 �� www.vueaudio.com Configuration: 3-way Weight: 76.6 pounds Size (h x w x d): 10.2 x 29.4 x 17.5 inches LF: Dual 8-inch (3-inch voice coil) neo cones MF: Quad 4-inch Kevlar neo cones HF: 1-inch-exit compression driver with beryllium diaphragm Frequency Response: 75 Hz – 18 kHz Dispersion: 90 x 10 degrees Power: Driven by rack-mount V6 Systems Engine Companion Sub: hs-20 dual-10, hs-25 dual-15, and hs-28 dual-18

www.ProSoundWeb.com

Alcons Audio LR16 �� www.alconsaudio.com Configuration: 2-way Weight: 63.9 pounds Size (h x w x d): 8.9 x 30.5 x 17.1 inches LF: Dual AMB8 8-inch cones, vented HF: RBN601 6-inch ribbon driver Frequency Response: 70 Hz – 20 kHz Dispersion: 90 x 40 degrees Power: 560/70 W (LF/HF, RMS), 1600/1000 W (LF/HF, peak) Companion Sub: LR16B dual-15, flyable

Carvin Audio TRX3210A �� www.carvin.com

Grund Audio Designs GA-2021N www.grundaudio.com Configuration: 2-way Weight: 90 pounds Size (h x w x d): 12.2 x 37 x 13 inches LF: Dual 10-inch neo cones HF: Dual 1-inch neo compression drivers Frequency Response: 50 Hz – 18 kHz Dispersion: 100 degrees horizontal Power: 600/1200/2400 W (RMS/program/peak) Companion Sub: GA-L15 single-15

Electro-Voice XLD281 �� www.electrovoice.com

Configuration: 3-way Weight: 60 pounds Size (h x w x d): 11.5 x 23.5 x 14 inches LF/MF: Dual 10-inch neo cones HF: Dual 1-inch-exit Mylar compression drivers on a PurePath lens Frequency Response: 85 Hz – 16 kHz Dispersion: 100 x 10 degrees Power: Self-powered Companion Sub: TRx3218 dual-18, TRx3118A single-18 active/flyable

Configuration: 2-way Weight: 48 pounds Size (h x w x d): 9.9 x 28.6 x 14.5 inches LF: EV DVN2080 8-inch cone LF/MF: EV DVN2080 8-inch cone HF: DualND2S 2-inch compression drivers Frequency Response: 65 Hz – 16 kHz Dispersion: 120 degrees (or 90 degrees, model XLD291) x 10 degrees Power: 200/80 W (LF/HF, continuous), 800/320 W (LF/HF, peak) Companion Sub: XCS312 triple 12-inch (cardioid), flyable

Turbosound FlexArray TFA-600H �� www.turbosound.com

ISP Technologies HDL 4210 �� www.isptechnologies.com

Configuration: 3-way Weight: 90 pounds Size (h x w x d): 12 x 28 x 22 inches LMF: Dual 10-inch neo cones on Turbo devices HMF: Single 6.5-inch cone on Poly-horn device HF: Single 1-inch neo compression driver on dendritic device Frequency Response: 90 Hz – 18 kHz Dispersion: 75 x 16 degrees Power: Bi-amp, optional tri-amp or DP self-powered Companion Sub: TFA600L single-18, integrates in arrays

www.ProSoundWeb.com

Configuration: 4-way Weight: 145 pounds Size (h x w x d): 11 x 43.5 x 23 inches LF: Dual 10-inch reflex neo cones LMF: Dual 6.5-inch horn-loaded cones HMF/HF: Dual 2-inch and 1.75-inch neo drivers Frequency Response: 67 to 20,000 Hz Dispersion: 100 x 6 degrees Power: Self-powered: quad-amped, 1000 W total Companion Sub: VT4881 single-15, integrates into arrays

July 2014

Live Sound International

57

:: Real W rld Gear ::

PK Sound VX10 �� www.pksound.ca Configuration: 2-way Weight: 63.5 pounds Size (h x w x d): 12.6 x 28.1 x 15 inches LF: 10-inch (2.5-inch voice coil) neo cone LF/MF: 10-inch (2.5-inch voice coil) neo cone HF: Dual 4-inch planar wave drivers Frequency Response: 110 Hz – 18 kHz Dispersion: 90 degrees horizontal, vertical array dependent Power: Self-powered, 2500 W total available for two cabinets Companion Sub: Klarity 18 single-18, Klarity 218 dual-18

Ramsdell Pro Audio LA10-2 www.ramsdellproaudio.com Configuration: 2-way Weight: 29 pounds Size (h x w x d): 12 x 19.8 x 16 inches LF: 10-inch neo cone, vented HF: 1-inch-exit neo compression driver Frequency Response: 65 Hz – 18 kHz Dispersion: 90 degrees horizontal, vertical array dependent Short-Throw Box: 120 degrees horizontal Long-Throw Box: 60 degrees horizontal Power: Passive, optional bi-amp Companion Sub: LA12-S (1 x 12-in), integrates into arrays

Alto Professional SXA28P �� www.altoproaudio.com Configuration: 2-way Weight: 48.1 pounds Size (h x w x d): 10.7 x 24.4 x 16.7 inches LF: Dual 8-inch (2-inch voice coil) ferrite cones HF: Dual 1.4-inch (1-inch-exit) neo compression drivers Frequency Response: 77 Hz – 18 kHz Dispersion: 100 x 7.5 degrees Power: 400/1600 W (continuous/peak) Companion Sub: SXA18P single-18-inch

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Live Sound International July 2014

Outline Butterfly C.D.H. 483 �� www.outlinearray.com Configuration: 3-way Weight: 75 pounds Size (h x w x d): 9.4 x 29.6 x 23.6 inches LF: Dual 8-inch neo cones, bandpass-loaded MF: Dual 8-inch cones, horn-loaded HF: 3-inch-exit neo compression driver on Double Parabolic Reflective Waveguide Frequency Response: 110 Hz – 18 kHz Dispersion: 90 degrees horizontal, vertical array dependent Power: Biamp, 800/120 W (LF & MF/HF, continuous RMS, mid-bass section mechanically filtered); tailored AMPRACK 12 HE also available.

dBTechnologies DVA T4 �� www.dbtechnologies.com Configuration: 3-way Weight: 29 pounds Size (h x w x d): 9.6 x 23.2 x 13.1 inches LF: RCF 8-inch neo cone MF: RCF 6.5-inch neo cone, horn-loaded HF: Dual RCF 1-inch-exit neo compression drivers Frequency Response: 80 Hz – 19 kHz Dispersion: 120 x 15 degrees Power: Self-powered Class T, 420 W total Companion Sub: S10 single-18, flyable; S20 dual-18

Sound Bridge XYON 7208XY �� www.soundbridge.com Configuration: 2-way (configured as alternating upright and inverted units) Weight: 86 pounds Size (h x w x d): 8.5 x 38.7 x 20.2 inches LF: Dual 8-inch neo cones with QUAD-One focusing MF/HF: EXO-LD neo compression drivers on Q-Drive WaveShaper horn Frequency Response: 68 Hz – 22 kHz Dispersion: 80 or 120 x 10 degrees Power: Crown iTech 4000 recommended Companion Sub: 5118 SW single-18, 5218 SW single dual-18

www.ProSoundWeb.com

TAKE THE STAGE

AND GET SET FOR GREAT SOUND PA SPEAKERS • POWER AMPS • OUTBOARD GEAR • MIXERS • MICROPHONES • WIRELESS SYSTEMS

Call 800-356-5844

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NEWSBYTES

:: The latest news from ProSoundWeb.com ::

RCF World Tour At The Cotton Bowl In late May, RCF USA hosted live demonstrations at the 85,000-seat Cotton Bowl in Dallas, showcasing a range of its line arrays and stadium loudspeakers over the course of two full days. Attendees could roam throughout the entire stadium, including the furthest point from the loudspeakers, the extreme sides, and even behind the flown and stacked systems. Flown systems included the TTL55-A and TTL33A active 3-way line arrays and HDL20-A and HDL10-A active 2-way line arrays, as well as the new HL Stadium Series, a line of passive 2-way horn loaded arrays designed for long-throw stadium applications. A range of subwoofers were stacked beneath in a variety of configurations. And, there were additional demos of ground-positioned loudspeakers, including the TTL11-A, an active steerable column array composed of two modules – one for mid-high and one for bass frequencies, which is stacked on top of the TTS26-A dual-18-inch subwoofer. “It was great to be able to provide the opportunity for consultants, prospective customers and our dealer partners to see and hear all of our line array and stadium solutions in one place at the same time,” says John Krupa, RCF USA director of sales. n

mix engineers

reauthorization and full funding of the

Ken “Pooch”

Elementary and Secondary Education Act

s PreSonus recently acquired WorxAu-

Van Druten

(ESEA) and to continue the designation of

dio Technologies, which was founded

(Linkin Park, Alter

music as a core academic subject.

in 1979 by CEO and director of engi-

Bridge), Peter

s COMPANIES

neering Hugh Sarvis, who is taking a

Keppler (Katy Perry, David Byrne, St.

s Ashly Audio is now shipping Dante

leadership role in the new direction. With

Vincent) and Jim Ebdon (Maroon 5, Aero-

digital media networking with all of its

the acquisition, WorxAudio is expected

smith), who demonstrated how they use

nXe, nXp, Pema and NE Series network-

to significantly augment core elements

Waves plugins in their live mixing.

able amplifiers. NE Series rack-mount

of PreSonus’

system processors (4400, 4800, 8800) are

professional

s Jonathan and Susan Lipp, owners

loudspeaker

of Full Compass Systems, recently

development

attended the National Association

s Nomad Sound of

of Music Merchants (NAMM) Advo-

Austin, TX recently

efforts, including drivers, system and

also now available with Dante.

mechanical design, system tuning, and

cacy Fly-In in

added new NEXO STM

rigging system design. (Pictured above,

Washington,

line arrays and Yamaha

left to right, are PreSonus chairman

D.C., joining

Commercial Audio CL1,

Kevin Kouhig, WorxAudio founder Hugh

musicians,

Sarvis, and PreSonus CEO Jim Mack.)

researchers, congressional leaders and other NAMM members to press Congress

CL3, and CL5 digital consoles to further bolster its rental inventory. Damon Lange, principal owner of

s Waves Audio recently held the latest

on the importance of providing music

Nomad, notes, “When I heard there was

installment of its WavesLive Master Class

education to all children. This year’s

an STM demo scheduled at the Alamo

series at Cirque du Soleil’s Resident

fly-in was the largest ever, with NAMM

Dome in San Antonio, I went down with

Shows Division in Las Vegas. The event

members and artists attending more than

Joel Hume, one of our lead engineers,

included a mixing workshop with live

100 meetings with Congress to urge the

and we spent two days listening and

60

Live Sound International July 2014

www.ProSoundWeb.com

system and the clarity at 350-plus feet,

Wireless Update

as well as examining the rigging and

Following the completion of a multi-year exploration of the role of wireless

listening to a small ground stacked rig, it

microphone systems in modern content production and the technical challenges

was simply a matter of figuring out what

surrounding their use, the FCC recently adopted new rules that expand wireless

configuration to start with.”

mic license eligibility to include professional sound companies and venues that

examining the rig. Once I heard the

routinely use 50 or more wireless mics. After consideration of technical and regulatory factors, the FCC concluded that the use of wireless mics (and related equip-

s APPLICATIONS

ment such as in-ear monitors and production intercoms) at major productions and

s The new

events could be effectively protected by expanding license eligibility.

Astana State

Wireless mic users may request protection from interference from TV Band

Auditorium,

Devices (also known as “white space” devices) at the time and location of these

located in

events by registering in one of the FCC-approved TV Bands Devices Databases.

Astana, the

Licensed users are able to obtain protection in a more streamlined and efficient

second largest city in Kazakhstan, has

manner. Previously, only broadcasters, cable networks, and TV/film production

unique fan-shaped wood ceiling panels

companies were eligible for licenses.

that create a flexible acoustic signa-

“Shure applauds the FCC’s thoughtful decision regarding expansion of entities

ture to accommodate a wide variety of

eligible for wireless microphone licenses,” states Mark Brunner, senior director, Global

events, from classical music to rock,

Brand Management at Shure. “As spectrum demand is increasing from a wide variety

pop, cinema, and spoken word. The

of technologies and services, protection of professional audio operations is becoming

venue’s sound reinforcement system,

more critical. We are pleased that the commission has recognized the wide impact that

implemented by A&T Media, is headed

wireless microphones have on our daily lives – in broadcasting and media production,

by a main array of 16 Renkus-Heinz

sports, live entertainment, and in corporate, civic, education, and worship venues.” n

PN102LR line array loudspeakers, along with a secondary array of five more PN102LR boxes to cover the upper balcony behind the stage. For added versatility, the main array can be hoisted up into the ceiling when not in use, and a second system with 12 IC Live ICL-R digitally steered column arrays on custom brackets can be put into play.

s The Killers are back on the road this summer in support of “Direct Hits,” a greatest hits compilation, with appearances at North American and U.K. music festivals. The band’s front of house engineer, Kenny Kaiser, is delivering his mixes via an SSL Live console. Kaiser, who works with Delicate Productions of Camarillo, CA, came on board with The Killers last year and took over FOH for this tour.

s Riedel Communications gear is providing flexible and reliable signal www.ProSoundWeb.com

July 2014

Live Sound International

61

:: News Bytes :: transport and

Peter Magu-

product manager, previously served as

communica-

ire, and Jonas

global product marketing manager at

tions for the

Domkus to

D&M Professional.

Red Bull Air

bolster the

Race World

company’s

s Meyer Sound has

Championship season that started in

customer support infrastructure. Pelli-

promoted Luke Jenks to

February in Abu Dhabi. For these events,

cano and Salmon join the EAW Applica-

the newly created position of

Riedel engineers built a signal distribution

tion Support Group (ASG) as application

director of product manage-

infrastructure for audio, communications,

support specialist and application

ment. Most recently the

video, and data with the MediorNet real-

support coordinator, respectively, while

company’s loudspeaker product manager,

time media network comprising seven

Maguire has been named customer sup-

Jenks is now responsible for defining

MediorNet Compact frames and three

port manager and Domkus takes on the

market needs, articulating customer

MediorNet Modular frames. Technicians

role of U.S. field application engineer.

requirements, and representing customer

interfaced audio signals via MADI and

feedback throughout the lifecycle of a

Riedel’s RockNet digital audio network in

s Martin Audio

product. He also assumes direction of a

combination with MediorNet to integrate

recently bolstered its

newly formed product management team

the event’s main mixing console, located

staff with Bash Akhtar

that encompasses market requirement

joining the company

definition, new product introductions, and

in the OB van that provided the bigscreen mix for the live audience.

as operations director and David Morbey

technical documentation. n

taking the role of product manager. Akhtar

s Eastern Acoustic Works (EAW) has added Gino Pellicano, Dave Salmon,

ProSoundWeb provides all of the latest pro audio news, and follow PSW on Facebook and Twitter - just go to www.prosoundweb.com and click on the icons at the top of the page.

previously served as operations and

.com

s PEOPLE

manufacturing director of RLC Callender, Smiths Detection, and Harmonic. Morbey, Martin Audio’s first dedicated

800•203•5611

www.SoundPro.com YOUR DIRECT SOURCE FOR

ULX-D

ZLX-15P

ALLEN&HEATH Qu-16

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SOUND BETTER. SAVE MONEY. SOUND PRODUCTIONS 62

Live Sound International July 2014

www.ProSoundWeb.com

ADVERTISERINDEX Adamson Systems Engineering Inc. . . . . . . . . . . . . . . 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IFC . . . . . . . . . . www .montarbo .com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — PreSonus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47 . . . . . . . . . . . www .presonus .com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Radial Engineering Ltd. . . . . . . . . . . . . . . . . . . . . . . . . 41 . . . . . . . . . . . www .radialeng .com . . . . . . . . . . . . . . . . . . . . . . . . . 604-942-1001 Ramsdell Pro Audio . . . . . . . . . . . . . . . . . . . . . . . . . . . 31 . . . . . . . . . . . www .RamsdellProAudio .com . . . . . . . . . . . . . . . . . . 727-823-8037 Renkus-Heinz, Inc. . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 . . . . . . . . . . . www .renkus-heinz .com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Riedel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 . . . . . . . . . . . www .riedel .net . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Shure Incorporated . . . . . . . . . . . . . . . . . . . . . . . . . . . .7 . . . . . . . . . . . www .shure .com/americas . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Solid State Logic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13 . . . . . . . . . . . www .solidstatelogic .com/live . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Sound Productions . . . . . . . . . . . . . . . . . . . . . . . . . . . 62 . . . . . . . . . . . www .SoundPro .com . . . . . . . . . . . . . . . . . . . . . . . . 800-203-5611 Telefunken . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37 . . . . . . . . . . . www .t-funk .com . . . . . . . . . . . . . . . . . . . . . . . . . . . . 860-882-5919 Waves . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3 . . . . . . . . . . . www .waves .com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Yamaha Commercial Audio Systems, Inc. . . . . . . . . . .1 . . . . . . . . . . . www .yamahaca .com . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . — Live Sound International provides this index as a service to advertisers. We assume no responsibility for errors or omissions.

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Splitsville An approach for location recording. by Bruce Bartlett

������ LET’S CONSIDER A DIFFERENT WAY to make a multitrack recording. Plug each microphone into a mic splitter, which sends the mic signal to two destinations: the PA mixer and recording mixer. The splitter has one XLR input and two or more XLR outputs per mic. Some splitters have a third output which goes to a monitor mixer, and a fourth output might go to a broadcast mixer. Splitting the mics is the most expensive method, but is the most professional. It gives you and the PA operator independent control of each microphone’s recording level and signal flow. Pros: Ultimate sound quality. Independent control at each mixer. Consistent sound. Cons: Complicated. Expensive if transformer splitters are used. Equipment: Mic splitters, maybe mic preamps, mic cables, mic snake, recording mixer and multi-track recorder or audio interface and laptop, mixer-to-recorder cables, headphones or powered monitors. There are many advantages of splitting the mic signals. You use your own mic preamps, so you’re not dependent on the quality of the PA console mic preamps. Also, you’re not hassling the operator about adjusting gain

trims. Each mix engineer can work without interfering with the others. The FOH engineer can change trims, level, or EQ and it will have no effect on the signals going to the recording engineer.

Another plus: a splitter provides consistent, unprocessed recordings of the mic signals. This consistency makes it easy to edit between different performances. What’s more, splitters let you use mic preamps on stage if you wish. That way, the cable from each mic to its preamp is short, which reduces hum and radio-frequency interference. As shown in the illustration, connect the outputs from all the splitter channels to the PA snake and to your recording snake. Connect the recording snake to your recording mixer mic inputs. This mixer is used to set up your own monitor mix and to set the recording levels. Connect the recording mixer’s insert sends to a multi-track recording system of your choice. n

This is an excerpt from the just-released Second Edition of Recording Music on Location by LSI/PSW author BRUCE BARTLETT and JENNY BARTLETT, published by Focal Press and also available from Amazon and Barnes & Noble. 64

Live Sound International July 2014

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