-
PLUS:
THE INS AND OUTS OF DI BOXES
MICROPHONES IN LIVE RECORDING
MAKING TECHNOLOGY TRANSPARENT
July 2009 | www.prosoundweb.com | $10July 2009 |
www.prosoundweb.com | $10
THE JOURNAL FOR LIVE EVENT TECHNOLOGY PROFESSIONALS
I N T E R N A T I O N A LJuly 2014 | www.prosoundweb.com |
$10
INSTALLATION | CONCERT | THEATER | CORPORATE AV | WORSHIP | CLUB
| RECORDING
DO YOU SPEAK GEEK?The unique language of audio analysis.
-
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
PaPartiaall LiListstining:g:Adappoe AAmemex Auudidio o BaBarthha
AudioioVVisisual BaBayeyer r MeMedidia a Bazelmmana s AVVRR
BrigghthtNNororwaw y ASS DDieiez zy Ca Lttdad Digital
ConsoleRRentat lDuD shhowow EEqquipiposos AApopolo Geoeorgrge e
ReRellleses SSouundnd GGMBM PPro-Souund Media Resource Group Most
High Producctitiono sPAP Profef sis onal PProroSSouundnd and Videoe
SSpepecic alEEveventntss Audio Skynight VERER TourSoundn Victory
Tour Productionn UseSSonido
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
18
6048
-
A division of Jam Industries Ltd
For more information [email protected]
-
I N T E R N A T I O N A L
VOLUME 23 | NUMBER 7
.com
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
Live Sound International111 Speen Street, Suite 200
Framingham, MA 01701Phone: 800.375.8015www.livesoundint.com
Jeff Turner | Account Executive415.455.8301 Fax:
801.640.1731
[email protected]
Mark Shemet | Associate Publisher Online,
ProSoundWeb.com603.532.4608 | Fax: 603.532.5855
[email protected]
Manuela Rosengard | Ad Production Director508.663.1500 x226 |
[email protected]
Jason Litchfi eld | Ad Production Manager508.663.1500 x252 |
jlitchfi [email protected]
Circulation and Customer Service inquiries should be made
to:Live Sound Customer Service
EH PublishingPhone: 800-375-8015, ext 294
(Outside the U.S.: 508.663.1500 x294)Fax: 508.663.1599
[email protected] Speen Street, Suite 200
Framingham, MA 01701
EDITORIAL AND READER SERVICE RELATED EMAIL ADDRESSESCirculation
& Subscriptions | [email protected]
Loading Dock Submissions | [email protected] Wide Web
Inquiries | [email protected] Rate Information
| [email protected]
REPRINTS: Erica Halloran 508.663.1500 x265 |
[email protected]
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 ::
10 Live Sound International July 2014 www.ProSoundWeb.com
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 ::
12 Live Sound International July 2014 www.ProSoundWeb.com
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 ::
14 Live Sound International July 2014 www.ProSoundWeb.com
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
-
16 Live Sound International July 2014 www.ProSoundWeb.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
-
www.ProSoundWeb.com July 2014 Live Sound International 17
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].
CARVINAUDIO.COM 800-854-2235TCSAUDIO.COM 800-403-4484
MADE IN
USA
Covering wide open spaces effortlesslyFOH (20) TRx3210 out- lls
(24) TRx3218 subs (4) TRx3903 front- lls (8) ARX4 amp racks
Delay Towers (2) (20) TRx3210A (8) TCS5800 Subs (2) ARX4 amp
racks TOTAL SYSTEM POWER 202,000 WATTS
Good Vibes Music Festival - Lake Elsinore, CA
TCS5800 dual 18 Sub9600w peak
TRx3210A 2500w each
ARX4 15,200w each
TRx3903 front- lls
A U D I OA U D I OA U D I O
MADE IN
USA
TRx3903 front- lls
-
18 Live Sound International July 2014 www.ProSoundWeb.com
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.
-
www.ProSoundWeb.com July 2014 Live Sound International 19
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
precision hand-made Italian loudspeakers
www.eighteensound.com
precision hand-made Italian loudspeakers
-
20 Live Sound International July 2014 www.ProSoundWeb.com
:: 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 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).
-
www.eaw.com
Control Freq.
Otto is the worlds first Adaptive subwoofer, extending Adaptive
Performance to the lowest octaves of the frequency spectrum. With
just a single module, Otto can provide omni, cardioid or
hyper-cardioid patterns (or anywhere in between). In arrays, much
more complex patterns and coverage are possible.
In addition to spectacular coverage, Otto also provides users
with the ability to easily balance their goals including active
rejection in non-audience areas, consistency and SPL. Otto is the
first subwoofer that allows users to define the coverage they want
and then provides them with the best solution to achieve it.
Like control? Youll like Otto.
For more information about Otto and Adaptive Performance, visit
EAW.com.
-
22 Live Sound International July 2014 www.ProSoundWeb.com
:: Clear Path ::
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).
-
Uniting the audience
Todays gig. Tomorrows gig.
Arena, festival, or concert hall.
Its all about consistent audience coverage, front to back,
whatever
the acoustic challenges.
A shared experience they will never forget.
Thats the Martin Audio Experience.
Unite your audience at www.martin-audio.com
-
24 Live Sound International July 2014 www.ProSoundWeb.com
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::
-
www.ProSoundWeb.com July 2014 Live Sound International 25
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.
-
26 Live Sound International July 2014 www.ProSoundWeb.com
:: 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.
-
A superior HF solutionfor your entire cabinet family
www.celestion.comFind out more
A comprehensive range of Celestion CDX 1" exit compression
drivers provides system builders withoptimised high frequency
performance for projects of every scale and budget.
Find out more about Celestion professional loudspeakers and
compression drivers at celestion.com.
Celestion CDX Series Compression Drivers
-
28 Live Sound International July 2014 www.ProSoundWeb.com
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::
One of the two KARAi and SB18 flown array sets.
Scan here to learn more.
lectrosonics.com or 1-800-821-1121
In Canada, call 877-753-2876
Post-ApocalypseTheater Sound
Made in the USA by a Bunch of Fanatics.
Only two things will survive...
-
www.ProSoundWeb.com July 2014 Live Sound International 29
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.
-
30 Live Sound International July 2014 www.ProSoundWeb.com
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.
-
www.ProSoundWeb.com July 2014 Live Sound International 31
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
where it most rejects the other instruments and amps, is
where
it ends up.
Contact us today to learn more!
Tel: 712-322-3900
PROFESSIONAL CASES & RACKS
www.grundorf.com
The Grundorf Combo Case not only protects your gearit helps you
work more efficiently.
HHighly customizable, these cases can accommodate mixers,
computers, power amps, signal processors, video gear, and more.
AAvailable in three stylescarpet finish, ATA, or our Studio
Series corporate/install versionthey get you in, set up, and
operational in no time flat.
BBuild yours with key locks, shelves, drawersyou can just about
name it!
Grundorf Combo CasesWill Protect Your Gear
and Streamline Workflow
Cable feed access - easy access to all compartmentsCustom sizing
- for larger mixers, computer monitors, connectors, etc.Multiple
add-on options including: Rear Rack Rail, Table Legs, Cable Hatch,
and Pull-out Trays
-
32 Live Sound International July 2014 www.ProSoundWeb.com
:: 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.
DR
PE
TER
JON
ES
.CO
M
-
www.riedel.net
The Future Is Networked.
A-Link-Interface Cardfor MediorNet
up to 384 channels per card
realize exible audio router setups
save signi cant costs in cabling and maintenance of the
infrastructure
unparalleled exibility and signal quality
MEDIORNET MODULAR
Fiber signal transport for 3G/HD/SD-SDI video, audio, data &
intercom
Supports any combination of network topologies Integrated CWDM
multiplexing Uncompressed real-time signal distribution and routing
Supports 3rd party router control Integrated signal processing and
conversion System architecture provides full redundancy
including auto re-route Future-proof hardware platform
See us at IBC Stand 10.A31
-
34 Live Sound International July 2014 www.ProSoundWeb.com
:: 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
loudspeakers | compression drivers | crossovers | horn flares
enclosure design software | loudspeaker protection
studioEvery Eminence product is designed to meet the most
important
standards we know YOURS.
-
36 Live Sound International July 2014 www.ProSoundWeb.com
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
-
www.ProSoundWeb.com July 2014 Live Sound International 37
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.
-
38 Live Sound International July 2014 www.ProSoundWeb.com
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
-
www.ProSoundWeb.com July 2014 Live Sound International 39
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.
-
40 Live Sound International July 2014 www.ProSoundWeb.com
:: 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