live sound international 1407

Two series, one family. Both represent the evolved sound quality and innovative functionality of todays 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, Yamahas VCM analog

circuitry modeling technology and Centralogicoperation. The 2 models of the QL Series take the best of CLs

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.

www.yamahaca.com

A Strong Bloodline

CL1

CL3

CL5

QL1

QL5

Photto Cro editt: Ra: R lph Larmannn

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IS LIVEPUSH 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.

P A R T O F T H E

F A M I L Y

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

Find out what DiGiGrid MGO & MGB interfaces can do for your MADI console at digigrid.netFor U.S. sales: www.waves.com DiGiGridMGB 128ch Coaxial MADI-to-SoundGrid Interface

DiGiGridMGO 128ch Optical MADI-to-SoundGrid Interface

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.

IN THIS ISSUE

FEATURES16 | Focus On The Knobs?Making technology transparent in the quest of art.

by Karl Winkler

24 | 48 Hours In Las VegasUpgrading the PA for Blue Man Group at the

Monte Carlo. by Marcus Ross

36 | Prepared To ManageSteps to a successful pre-production process.

by Danny Abelson

48 | And Theyre Off... An audio makeover at historic Churchill Downs.

by Live Sound staff24

JULY 2014

8 | Loading DockEQUIPMENT New subwoofers, amplifi-

ers, networking and more. by Live Sound staff

18 | Clear Path Going direct the ins and outs of DI boxes.

by Gary Parks

30 | In FocusMicrophone choice and application for live

recording. by Craig Leerman

38 | Tech Topic The unique language of audio analysis.

by Pat Brown

44 | SpotlightRecording options of digital consoles.

by Live Sound staff

50 | Road Test Evaluating the Shure GLXD 2.4 GHz wire-

less microphone system. by Craig Leerman

52 | Road Test Checking out the new QSC Audio amplifier/

processing platform. by Danny Rosenbaum

54 | Real World GearEQUIPMENT Focusing on the latest

medium-format line arrays.

by Live Sound staff

6 | From the Editors Desk

60 | NewsBytes

63 | Advertiser Index

64 | Back Page

DEPARTMENTS

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For more information [email protected]

I N T E R N A T I O N A L

VOLUME 23 | NUMBER 7

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Publisher | Kevin McPherson | [email protected] | Keith Clark | [email protected]

Senior Contributing Editor | Craig Leerman | [email protected] Technical Editor | Ken DeLoria | [email protected] Sound Editor | Mike Sessler | [email protected]

Europe Editor | Paul Watson | [email protected] Consultant | Pat Brown | [email protected]

Art Director | Katie Stockham | [email protected] Art Director | Dorian Gittlitz | [email protected]

ProSoundWeb.comEditor-In-Chief | Keith Clark | [email protected]

Product Specialist | Craig Leerman | [email protected] | Guy Caiola | [email protected]

Karl Winker | Gary Parks | Danny AbelsonMarcus Ross | Bruce Bartlett | Danny Rosenbaum

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From the Editors DeskSo, as we ask on the cover: Do you speak geek? If not, no worries thats why were presenting Pat Browns Measurement Glossary, beginning on

page 38 of this issue. When I fi rst received the article, I had a feel-

ing it was something special, and that was quickly confi rmed. As usual, Pat takes a complex topic and breaks it down into an excellent primer that builds to further understanding as you go. Its a great example of why hes a renowned educator, and why the pro audio industry is so fortunate to benefi t 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 wasnt sure if Craig Leerman would have time to put together the article on micro-phones for live recording that wed been discussing. Turns out that he handled it with no problem, as youll see beginning on page 30. His decades of experience working in virtually every type of live audio situ-ation 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 Abel-son continues his discussion with noted engineer Dave Natale, this time focusing on key aspects of pre-production. And as always, theres much more. Enjoy the issue

Keith ClarkEditor In Chief, Live Sound International/ProSoundWeb

[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.

LOADINGDOCK

8 Live Sound International July 2014 www.ProSoundWeb.com

Adamson Systems E219 @A subwoofer loaded with two lightweight, long-excursion

19-inch SD19 Kevlar neodymium drivers utilizing proprietary

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

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

power handling, designed to reproduce clean, musical low-

frequency information. Integrated rigging permits a 0- or a

3-degree angle, allowing for compatibility with the companys

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

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

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

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

pounds, is constructed of marine grade birch plywood as well

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

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

dedicated output connection point. www.adamsonsystems.com

Allen & Heath Qu-32 @A 32-fader, 38-input/28-output digital mixer incorporating

capabilities such as total recall of settings (including faders and

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

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

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

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

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

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

mix outputs including 2 stereo matrix mix outputs and 4 stereo

groups with processing, patchable AES digital output with

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

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

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

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

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

recall. The free QuPad iPad app gives instant wireless control

of the mixers key parameters. www.allen-heath.com

PreSonus SL-Dante-SPK A card for the companys StudioLive AI-series (Active

Integration) active loudspeakers that includes an Ethercon

connection for Dante audio networking and remote control

via the free SL Room Control application.

It allows users to create a networked

audio system with any Dante-enabled

mixer using a standard 1 GB Ethernet

switch. Users can also connect

non-Dante mixers, such as

a first-generation PreSonus

StudioLive, to the analog

inputs of a Dante-equipped

AI loudspeaker and then

broadcast the signal over the

Dante network using Cat-5

cable. The upgrade works with

StudioLive 312AI, 315AI, and

328AI loudspeakers, as well

as the StudioLive 18sAI sub-

woofer. www.presonus.com

Shure QLX-DA digital wireless system providing

24-bit digital audio, networked control,

and compatibility

with Shures intel-

ligent rechargeable

battery technol-

ogy. It transmits

accurate audio

with extended, flat frequency response. The systems automatic

channel scan and IR sync make finding and assigning an open

frequency fast and simple. AES-256 encryption comes standard

and can be enabled for secure wireless transmission. The

systems intelligent lithium-ion rechargeable power options can

provide up to 10 hours of continuous use and report remaining

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

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

with networking tools, including Shure Wireless Workbench 6

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

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

Plus Channels mobile app. www.shure.com

www.ProSoundWeb.com July 2014 Live Sound International 9

Products Fresh Off the Truck

Celestion CDX1-1010 A lightweight, low-profile ferrite magnet compression driver

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

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

Finite Element Analysis (FEA) techniques are used to optimize

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

designed for entry-level 2-way and

3-way loudspeakers. The CDX1-

1020 variant is supplied with

partial phase plug assembly,

for applications where the

outer phase plug and horn

form a single moulding as

part of the front baffle.

http://celestion.com

Eastern Acoustic Works (EAW) Otto The first subwoofer in the

Adaptive Performance

Series, Otto is loaded

with two 18-inch woof-

ers, with acoustic energy

exiting from four spaced

apertures in the corners

of the enclosure. It is rated

to deliver output of 131 dB

(1 meter, continuous, full-space)

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

transducer is separately powered and processed, allowing

multiple directivity patterns to be created from a single module.

It can readily be combined in arrays to provide increased

pattern control and output. EAW Resolution software gener-

ates DSP parameters to simultaneously adapt the complex 3D

wavefront surface and optimize frequency response to match

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

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

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

distribution racks flown adjacent to the flybar of each array as

well as full network redundancy. www.eaw.com

DiGiCo V685 The latest software upgrade for the companys range of digital

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

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

from 32 to 40

Flexi channels.

The upgrade also

supports Optocore

DD4MR, DD2FR,

X6R and DD32R

devices in audio

I/O. Further, any

SD5, SD8, SD9,

SD10 and SD11

running Waves 9.5

will now have 32 stereo Waves racks. Theres also support for the

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

device. For theatrical environments, the Relative Faders in cue

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

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

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

charge for an introductory period. www.digico.biz

Audio-Technica ATND971 A cardioid condenser boundary

network microphone that

transmits audio and control

data together over Dante

network protocol. An

Ethernet connection

allows the ATND971

to communicate across an

existing network of Dante-enabled devices and, with the mics

programmable user switch, control any of those devices at the

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

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

ATND971 is powered by network PoE. Its also outfitted with pro-

prietary UniGuard RFI-shielding technology and UniSteep low-cut

filter. www.audio-technica.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

:: 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

Systems). The Lake variant offers Lake Processing with analog,

AES and dual-redundant Dante networking. It is supported by

the development of new custom software to provide extensive

integration with most key system manufacturers. The Tes-

ira variant is equipped with Tesira DSP and AVB audio and

control. The D Series also includes proprietary Rational Power

Management (RPM) technology, providing flexible power allo-

cation across all channels to foster efficient and rational use of

total amplifier inventory. http://labgruppen.com

JBL Professional EON610 & EON612 @Joining the EON600 Series, the EON610 (10-inch) and EON612

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

power amplification. Custom JBL high- and low-frequency

transducers deliver high sound pressure levels with low

distortion throughout the frequency range. JBL examined the

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

ent points, employing proprietary measurement techniques,

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

cation), providing power and DSP platforms designed to interact

intelligently and adapt to prevailing conditions, protecting driv-

ers, and significantly enhancing performance of loudspeaker

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

20 kW peak output into 4 ohms and continuous power available

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

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

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

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

manufacturers limiters and soft-knee compressors. It can route

audio from analog, AES or network sources with automatic fall-

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

It also includes GPIO and remote control covered by Ether-

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

Windows and Mac platforms is available and operable via an

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

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

Aviom A360 Display An iOS application that provides a visual display of mix infor-

mation on A360 personal mixers, allowing performers to view

volume levels, stereo placement information, tone and reverb

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

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

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

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

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

required is a D800 or D800-Dante

A-Net Distributor,

which communi-

cates with the iOS

device through a

connected WiFi

router.

www.aviom.com

:: Loading Dock ::


Grund Audio Design GA-LC9 & GA-LC9P 2Joining the companys Gala Series, the GA-LC9

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

column loudspeakers are designed for both

portable and fixed installation applications.

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

ducers. Frequency response is rated at 140

Hz 20 kHz, while system coverage provides

120-degree vertical and 10-degree horizon-

tal dispersion. The passive GA-LC9 is rated

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

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

A single LC9P can power an LC9. Enclosures

include 2 x 2 flypoints as well as pole mount

adapters on the top and bottom, compatible

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

tured in the USA, enclosures are made of

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

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

(passive/powered). www.grundaudio.com

Electro-Voice X1 & X2 The first models in the companys next generation of X-Line line

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

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

Hydra device that

emulates the acoustic

behavior of a double

line of four 3-inch point

sources, fostering opti-

mized mid-band cou-

pling of the array while

maintaining efficiency and power.

The HF section of the X1 incorporates two new ND2R

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

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

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

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

quency Hydra constant energy planar wave generators, also on

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

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

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

www.solidstatelogic.com/live

SSL Live

[email protected]

:: Loading Dock ::


Mackie SRM750 & SRM2850 @Two new models joining the SRM Series, each with 1,600

watts of onboard power paired with custom transducers

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

SRM full-range loudspeakers, the SRM750 incorporates

proprietary HD Audio Processing, which includes patented

acoustic correction algorithms for high-definition output

plus system optimization tools like application-specific

loudspeaker modes and an accurate feedback destroyer.

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

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

designed for high-output LF performance, making it suitable

for applications such as stacked rigs at festivals, clubs and

other live applications. www.mackie.com

Crown Audio XLC2800 & XLC2500 @Two power amplifiers (both have 2 channels), designed for

install applications, incorporating proprietary DriveCore technol-

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

using stereo, parallel or bridged mono outputs. The XLC2800

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

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

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

DriveCore IC chip combines the amplifier driver stage into the

power output stage along with additional audio-signal functions,

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

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

connectors. XLC Series amplifiers also have rear-panel volume

controls for each channel. www.crownaudio.com


FOCUS ON THE KNOBS?

thing looks like a nail. In other words, if we know what all those knobs (and but-tons) do, does it mean were compelled to twist the knobs and push the buttons? In many cases Im afraid its true, and yet, we can miss something in the process.

PRACTICE MAKES PERFECTAs an amateur photographer growing up in the days of film and mechani-cal cameras, I always found it useful to practice with the equipment empty before putting real fi lm at risk. In those days, every exposure cost money, and frankly, I didnt 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 transpar-ent 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

AT ONE TIME OR ANOTHER, all of us who have sat behind a mixing con-sole 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 dont 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 con-cept of if youre a hammer, then every-

together, is obviously important as long as the end result is kept in mind. The audience probably wont know if you used an actual LA-2A leveler or a plug-in equivalent on the vocals. But they know when they cant hear the words or if the bass is overwhelming the mix.

Adopting new technology into a sys-tem should not be about trying to fi nd ways to use it so we get our moneys worth. Instead, it s about having the new stuff integrate so seamlessly that we almost forget its there, except for whatever benefi ts it brings to the table in terms of better sound, smoother work ow, or faster set-up time.

VISUALIZE (AURALIZE?) Another photography analogy: Ansel Adams espoused the idea of visualiz-ing the result you wished to have when viewing a scene, to imagine how you would want it to appear in a photo-graphic print. Then, using the technol-ogy 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 repro-duced with photographic equipment.

First, Adams suggested exposing the fi lm in order to ensure that there were details in the shadows (above the noise oor). Then he gave pointers as to how the fi lm 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, pro-viding the viewer with the impression desired by the photographer in the orig-inal 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 stum-

Making technology transparent in the quest of art.

by Karl Winkler

OUTLOOK


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 loud-speaker 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, lets 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 were 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 were 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 cre-ate 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 doz-ens of songs, it was very eye-opening because we realized how each different

producer and engineer had exploited the available technology to achieve certain results, thereby enhancing the musical experience. Once in a while wed also notice the bad examples where some aspects of the recording or mixing tech-niques got in the way of the results, and even ruined the recording.

One final thought: its 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, thats what its 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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GOING DIRECT

ance characteristics to connect with the sound reinforcement system properly without adding excessive noise and/or altering frequency response.

PRIMARY APPLICATIONSDI boxes provide three basic functions. First, they convert unbalanced signals from sources such as instrument pick-ups and electronic instruments into balanced signals that can travel longer distances without induced interference or signal degradation. Second, they help with impedance matching, espe-cially from high-impedance sources like passive guitar pickups being fed into low-impedance mic inputs on the mixing console. Third, while perform-ing the above electronic functions, DIs act as an interface to change from one connector type to another, typically from 1/4-inch to XLR.

As an added benefit, the audio trans-former within a passive DI will break a ground loop. Some active DIs also place a transformer within the input-

GIVEN THE WIDE VARIETY of audio sources that are connected to the microphone and line inputs of the mix console, the availability of high-quality 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 imped-

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.

PASSIVE & ACTIVEDI boxes are made in both passive and active formats, and each has its primary uses and advantages. Basically, a passive

design requires no exter-nal power to function, and its internal audio-quality transformer per-forms the conversion functions. An active DI 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 theyre labeled or not, can usually be done by noting the absence or presence of a battery compartment, an on/off switch, and an LED. (Note, however, that there are active units that do not have all of these elements.)

A basic rule of thumb is to use a passive DI with an active source, and an active DI with a passive source. A standard electric guitar and many basses are passive sources, as are most acous-tic guitars with under-saddle pickups and other instruments with piezo pick-ups. Keyboards, active pickup systems, effects and other electronic devices, as well as audio sources with a battery or an AC plug, are active sources.

Mick Conley, mix engineer for country musician Marty Stuart, says that when connecting a bass with pas-sive pickups, he usually uses an active DI to help with the signal level and clarity, while a bass with active pick-ups will have a passive DI, since theres no need for the extra output, and a pas-sive DI can help keep the tone from

The ins and outs of DI boxes.

by Gary Parks

CLEARPATH

Andy Heller and Gary Wood, co-owners of Audio Production Group (San Carlos, CA), with their Countryman Type 10S and Type 85S stereo direct boxes.


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

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 pri-marily for keys and various processors. He adds that keyboards and rhythm boxes, as well as the ever encroaching array of vocal effects processors, also benefi t greatly when used with a DI.

Nick Malgieri, AV manager and audio engineer at Stanfords Bing Con-cert Hall, states, Generally speaking, I prefer active DIs. I fi nd the higher out-put and high-presence tone to be bet-ter for most applications. Passives work as well and are often preferred by rock n roll engineers who prefer a softer, rounded tone, although I fi nd them to sound a bit fl at and the low output can drive up noise fl oor.

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 volt-age as theyre 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 Country-man Type 10 active DI works with both phantom and battery power, and it also

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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 main-tain 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 attenuat-ing the instruments 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 micro-

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 channels 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 DISTANCEAn 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 ana-log 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 sig-nal to the performers on-stage ampli-fier, so that the guitar or bass amp sees the instruments pickups. Thus the DI can also function as a signal splitter.

Along with filling their functions of signal balancing, correcting imped-ance mismatches, and breaking ground loops, DIs are audio devices that are inserted directly in the path between the transducer capturing the audio source and the mixing and amplifica-tion components of the sound rein-forcement 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-

The Klark Teknik DN100 foregoes battery operation.

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

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ity is an important differentiator between a passable and a high-performance pro-fessional unit. This choice can have audi-ble results. Companies such as Lundahl and Jensen specialize in manufacturing transformers with excellent audio charac-teristics 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 fl atness across the audio spectrum.

ROAD READYBecause DI boxes are distributed around the stage in unprotected loca-tions, theyre usually ruggedly built devices, often weighing a pound or two, though a few more diminutive (yet still rugged) units can also be had. Connec-tors and switches are typically recessed

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

offering some protection to switches and attached cable connectors.

Internal durability is also a factor. The quality of the switches, connec-tors, 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 expen-sive DI is inconsequential compared to the price of a good instrument or mixing console, investing in quality is wise.

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 prod-uct manager for Clear-Com, handling RF planning software sales with EDX Wireless, and managing loudspeaker and wireless products at Electro-Voice.

Some makers offer DI boxes with single and dual channels, as seen here with the Countryman Type 10 and 10S (S stands for stereo).

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t 9:30 pm on a Tuesday in December, I had five teams total-ing 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 sys-

48 HOURSIN LAS VEGAS

Upgrading the PA for Blue Man Group.by Marcus Ross

Atems 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 produc-tion finished, the company started look-ing at other opportunities to recreate this experience. The Las Vegas production was a natural fit.

Excitement & ImpactIn September, 2011, when I joined Blue Man, my goal was to bring consistency

:: Blue Man Group::


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

to our productions around the world. In this quest Ive experimented with every-thing 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 numer-ous 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 Hol-lywood Bowl for more than 17,000.

With similar content across all pro-ductions, 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 sys-tem, my focus is on SPL, bandwidth, coverage, polar stability and any physi-cal limitations that might exist.

The musical composition for Blue Man shows is very dynamic and con-tains a significant amount of band-width. 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 Ive worked with. This is instantly noticeable with Blue Man instruments, ranging from the acousti-cal 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.

Knowing In AdvanceThe mix volume for the Las Vegas pro-duction of Blue Man Group spans from 80 dB to 105 dB (A-weighted). To

insure suffi cient headroom, the speci-fi cation 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 specifi cation 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 dif-ferential of fi ve times from front to back, this venue is very well suited to a line source array. This is just over two dou-blings 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::

The combination of coverage and size limitations really dictated that a modular line source array would be the right solution. By segmenting the low-frequency component from the main element, it allows for a reduction in ele-ment 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 fi t. It was more than capable of delivering the dynamic range and response the show requires, and being smaller than the previous sys-tem, 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, keep-ing refl ections off the architecture and allowing for a larger stereo fi eld.

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

almost perfectly meet the coverage goals. This approach reduced the over-all 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 fi lls.

With the limited time for the tran-sition, 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 fi rst 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 diffi cult to fi nd a decent

compromise in a theater space when the seating sections are on multiple planes.

By using the main arrays to repro-duce the full range it becomes possible to have a uniform tonal balance across the entire audience as opposed to a sig-nifi cant buildup of LF in the fi rst few rows, due to the proximity of the seats to the loudspeakers. To achieve these two goals, lines of six SB18i subwoof-ers are fl own 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, were able to produce more than 95 percent of the shows musical con-tent from the arrays and subs.

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

Within SOUNDVISION mod-

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.

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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eling, I was able to ensure that loud-speaker 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 loud-speakers 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 under-

balcony 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 sys-tem. The result is very homogeneous

coverage across the entire venue with the SPL difference well within the target.

For enhanced effects purposes, four SB28 dual-18-inch subwoofers are posi-tioned 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 repro-duce down to 25 Hz, they provide the infrasonic portion of the show, really focusing on the 25-50 Hz region.

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 pur-poses supply very high output and are also passive, reducing the need for addi-tional 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

:: Blue Man Group::

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the center cluster while avoiding comb fi ltering between the two arrays.

On the electronics side, all ampli-fi cation and DSP was simplifi ed 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 pro-viding plenty of EQ for room-specifi c issues. In the end only a couple of fi lters were needed in any part of the system.

Go TimeStarting at 10 pm on Tuesday, we removed the previous loudspeakers, amplifi ers, DSP and cable, and working with our integrator, Clearwing Produc-tions (Phoenix, Milwaukee), the new system elements were all installed and tested by 8 am. Following a rest period and fortifi ed 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 fi nished before break.

As noted, the measured results of the system were almost spot-on with the predictions from SOUNDVI-SION. Within three hours, the system response was unifi ed and the adjust-ments 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 amplifi er/controller channels, tuned the PA and gotten some sleep! From the fi rst 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 nearfi eld monitors in a studio, not a PA at 70 feet away in a 1,500-seat theater. Hav-ing made many adjustments to shows over the past several years for Blue Man

Group, this has been the single biggest leap in performance the show has seen.

Ill close with the reaction of senior music director Byron Estep: The L-Acoustics system handles high vol-ume and density perfectly, without los-ing clarity in the transients or changing the tonal character of the mix as the musical dynamics change. Our per-formers 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 work-ing on shows and listening to different systems in different rooms, I can say without hesitation that L-Acoustics loudspeakers sound the best and pro-vide the most musical listening experi-ence for our audience.


QUALITY CAPTURE

a plethora of models tough enough to handle the rigors of the live realm while delivering the desired audio quality.

Theres also a wide range of types. Large and small diaphragm. Dynamic, condenser and ribbon designs. Cardioid, supercardioid, hypercardioid, and fig-ure 8 patterns. Vocal mics, except when theyre used on instruments. Instru-ment mics, except when theyre used on vocals. Drum mics, except when theyre used on other instruments... If youre 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 STRATEGYBefore 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 record-ing 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

WHEN DONE WELL, a live record-ing captures the energy and personal-ity of the performance, along with the ambiance and (if desired) audience response. There are many different ways to record a live show, but regard-less of the approach, a good recording starts with the right microphones, cor-rectly placed.

By right Im referring to mics that fit the particular application, tak-ing factors such as pickup pattern and SPL handling into account. Mics tend to be categorized as live and studio. Yet while its true that certain models are too delicate for live use, and other certain models lack the sonic character-istics sought in the studio, today theres

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 dont need to be fancy or expen-sive. For example, one trick I use is to set the boom of a mic stand to a T shape and then drape a packing blan-ket from my truck over the T. Viola! Its 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 close-mike instruments as much as possible

Microphone choice and application for live recording.

by Craig Leerman

INFOCUS

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


in order to only pick up the intended sound. In the live world, we tend to like cranking the gain up until its 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 pick-ing up unwanted sound (and noise remember, this is live).

Overheads on drums tend to pick up a lot of sound we dont want, so bring them in as tight to the kit/cymbals as possible. On a loud stage I tend elimi-nate the overheads altogether and just close-mike the cymbals from under-neath. This technique can also work well for straight-up live sound.

Keep stage rumble to a minimum. In addition to rolling off very low fre-quencies with EQ or high-pass filter-ing, I also make sure mic stands are in good shape and have rubber feet for isolation from the stage. If stage vibra-tions 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 musi-cians 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 instru-ments 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

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 con-denser but the focus is the narrowest

pattern I can get away with, depending on the singers 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 mics pickup pat-tern. For singers who hold the mic away from their face or down by their belt, I outfit them with a wireless headworn

The place where each mic sounds the best to my old ears, and

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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 vocalists 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 isnt a hole (com-mon 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 con-

denser is my go to mic, but a dynamic can also work well if it needs to be posi-tioned where theres 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 supercar-dioid 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 record-ing. 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 isnt in the

way of the musician. For bongos, its 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. Theyre 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 page-turning noises. If the piano is full size, I also opt for a larger diaphragm mic over the low strings. A single bound-ary 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.

Electric Guitar/Bass Amps. For guitar amps, a cardioid dynamic placed off-axis of the speaker (or one of the speakers) is a quick way to get a good sound, but a newer ribbon mic marketed for guitar amps is a great choice if avail-able. For bass amp, a large-diaphragm

Getting close to capture snare and toms.

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 players way.

Banjo. A small-diaphragm con-denser 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 bot-tom rotor joined by two small-dia-phragm condensers for the top horn

one at each side of the cabinet about 6 inches from the spinning horn cap-tures the unique sound of this instru-ment. 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, Ive 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 fi rst 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 cap-ture the audience, I place a few shot-gun-type mics at the stage wings on stands, pointed at the crowd and posi-tioned higher than the fi rst few rows (or thats all theyll pick up). Ive also suspended cardioids over an audience with good results.

Live recording doesnt require doz-ens of different microphone models. Just take stock of what you have avail-able, select the best ones for each appli-cation, be patient and diligent with positioning, and youll be good to go.

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

A ball-style mic for harmonica. This beauty is the Shure 520DX, a.k.a., green bullet.

Live or in the studio

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PREPARED TO MANAGESteps to a successful pre-production process.

by Danny Abelson

Dave Natale kicking back at Right Coast Recording, his recording studio in Pennsylvannia.

neers. A mutual respect here will go a long way to your success.

THE PROCESSDuring early rehearsals with a new act, I always go out of my way to spend time onstage. Its 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 Im interested in what their instru-ments sound like.

Also, I always try to get a separate room with some isolation to mix in. I recommend using some big loud-speakers/monitors because if the band decides to come in and listen, youll need something that sounds impres-sive. If they listen to mixes through nearfields on the console, they wont get the full effect of what youre trying to do live. Were not making recordings here, were trying to a) learn the mate-rial, and b) demonstrate what it should sound like live.

I prefer large full-range boxes like (Clair) S4s because they fit comfort-ably through doorways and are easy to stack; however, it may be simpler to use a few of the loudspeakers youll actually be using on tour. In the past Ive 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 youve ever lis-tened to high-powered loudspeakers at close range (at a professional level), you

WERE CONTINUING our discus-sions with veteran independent touring engineer Dave Natale, this time focus-ing on pre-production. Daves prepared for band rehearsals, production rehears-als, and tours countless times, with pre-production 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.

TALK TO EVERYONE When planning for band rehearsals, if its an act Ive not mixed before, I start by talking with the production manager, who will usually have a copy of the stage and mic info from the previous tour. This is usually an excellent source of information. Next I usually talk with the backline crew. These are folks you

spend a lot of time with, and theyre 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 prefer-ences covered.

Always provide a copy of all docu-mentation you generate mic chart, console files if youre mixing digital, etc. to the production manager. There are circumstances where an engineer may get sick or self-implode, and hav-ing a complete set of documentation in the production office can be helpful in maintaining continuity.

Once your research is complete, its 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 previ-

ous engineer. I have rather sim-

ple tastes in mics, so typically its out with the Neumann U87s and in with models

from Shure, Sennheiser, and Electro-Voice. Dedicating the necessary time at this stage is essential to insuring you have the gear you need when arriving at rehearsals.

One other important note: you must fit in with the backline crew. They were there before you, and will probably be there after youre gone. They can make life easy or miserable, so make friends and keep them. They have an even more direct path to the artists than most engi-

PERSPECTIVE


understand why I do this. Its simply not any fun listening to small loud-speakers. Theres 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 cli-ent, the principals came in to have a listen after the fi rst night of rehears-als. I was a bit nervous but had con-fi dence 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 confi dence 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 audio-wise, ever. No suggestions to turn this up or down. Nothing. They trust me,

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

SURVIVE THE FIRST SHOWSometimes just getting through the first show takes some calm nerves. Ive had tours where after four weeks of band rehearsals, the fi rst 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 after-noon of the fi rst 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-profi le act, with all of the media in the known universe on hand and celeb-rities galore crawling all over the front of house platform. Not an ideal circum-stance for a fi rst 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.

Presenting a decent mix on big

loudspeakers really helped to earn

their confi dence and alleviated any

questions.


TECHTOPIC

The unique language of audio analysis. by Pat Brown

Measurement Glossary

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 fi eld, and even spill into seem-ingly non-technical fi elds 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 fi nd that the rigid defi nitions may not even seem to apply to audio, since they were not devel-oped 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 defi nitions 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.

As one engaged in both web-based and in-person training, I have the benefi t of observing fi rst hand how those new to the fi eld 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 mean-ingful data from their measurement platform.

These are ordered in a way logical to learning measure-ment 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 effi cient 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 2: The FFT and Inverse FFT.

Figure 3: The Impulse Response.Figure 1: The Time and Frequency Domains.

Time Domain

FrequencyDomain

FFT

iFFT


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.

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 signifi cant (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 specifi c 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 refl ections on the transfer function, effectively allowing anechoic loudspeaker measurements to be made indoors.

Symmetrical Time Window Symmetrical time windows are used when there is signifi cant 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 fi eld arrival (Figure 8).

Asymmetrical Time Window A time window that is not tapered (rectangular) at the leading edge, but tapered at the trail-ing 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 fi nd 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 refl ections to be excluded from the high frequency response, but allows the room into the measurement for computation of the low fre-quency response. This appears to emulate the way that humans perceive sound. In effect, at low frequencies (long wavelengths) the contribution of the room refl ections become part of the woofers 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

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 fi lters (frequency response ripples caused by the phase interaction of multiple sound arrivals), leaving the part that can be meaningfully improved using an electronic equalizer.


:: Tech Topic ::

Finite Impulse Response (FIR) An impulse response of fi xed time length. An example is a wave fi le of the IR of a room or loudspeaker, but it could also be a high or low pass fi lter used to form a crossover network.

Infi nite Impulse Response (IIR) An impulse response that is generated on-the-fl y with feedback of previous sample values. In theory, such an IR would never decay to zero. Analog fi lters are IIR, as are some digital fi lter types. In general, IIRs have lower latency than FIRs.

Frequency Response Magnitude A measure of the rela-tive or absolute level of the signal vs. frequency. There is infor-mation 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 major-ity 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 fi eld 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

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 fl ight 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

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