Advision - American Radio History

Advision - American Radio History
September/October 1991
serving: recording, broadcast and sound contracting fields
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Street, Meridian, MS 39301, 601.483 -5365. Telex: 504115, Fax:484-4275
ii U.S.A.
Vol. 25, No. 5
serving: recording, broadcast and sound contracting fields
Cover Story: Advision
Small Group Setup for Live Recording by Malcolm Chisholm
Recording a live performance poses special problems.
by Randy Savicky
The U.K' s top mobile recording facility.
Audio for the National Victory Celebration
For once, our peripatetic soundman remained in the U.S.
Ed Learned
A Theater Sound System by Barry Luz
A performance complex in St Louis gets new sound equipment.
see page 21
Creating a Church Recording Studio by Hal Swinhart
Sophisticated recording equiment for a church.
Audio for the Church by Brent Harshbarger
Why is church audio so poor ?.
Using Drums in the Church by Joe Ciccarello
Do drums belong in the church?
How to Use EQ
by Bruce
and Jenny Barlett
Improving the sound quality of what goes on tape.
Hot Tips: Designing Vocals: Part III
Getting good vocal sound onto tape.
page 38
John Barilla
Advision's main room. Apparently out of the mainstream of England's London music scene,
Advision has been able to build a
world-renown studio and mobile
operation from their locaction in
Brighton. Proof, that if you do it
right, people will beat a path to your
door! See Randy Savicky's article
beginning on page 10.
Tek Text: 3 -D Audio: Wave of the Future?
by Jim Paul
New Products
The ELAR Booksource
Buyer's Guide: Signal Processing I: Crossovers, Delays,
Equalizers, Multi-Effects Processors, Reverbs
Hotline: A Broadcast Audio Question and Answer to
Randy Hoffner
People, Places, Happenings
The 4th Digital Radio Station
Seminar on Sept. 13 will offer pres-
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See ARK at AES Booth 3909
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entations on equipment and software for digital audio storage and
distribution. Discussions during
the afternoon will focus on Digital
Audio Broadcasting with presentations by each of the DAB system proponents. An additional materials
fee is charged for this seminar. A
panel with representatives of each
DAB system will close the day's
For more information on this and
other seminars during the convention, please call NAB Science &
Technology at (202) 429 -5346.
For registration information,
please call (800) 342 -2460.
Roland J. Zavada, longtime
standards advocate, will deliver the
Keynote Address for the 133rd
SMPTE Technical Conference
and Equipment Exhibit which
will be held from Oct. 26 to Oct. 29
at the Los Angeles Convention Cen-
ter. Zavada's speech, tentatively titled "Managing the Moving Image,"
will cover the Society's 75 -year history from an engineering point of
view. He will present the Keynote
Address during the opening session
on Saturday morning. Other featured speakers of the conference are
Gregory Peck, who will speak at
Among the many highlights of
the technology program at the Na-
tional Association of Broadcaster's Radio 1991 Convention
is a day -long seminar on digital radio. The convention is scheduled for
Sept. 11 -14 at San Francisco's
Moscone Convention Center.
the Honors and Awards Luncheon
on Oct. 26, and Burton (Bud)
Stone, of Deluxe Laboratories, who
will speak at the Fellows Luncheon
on Oct. 27.
The Sony Professional Audio
Training Group will offer a seminar on the CD Mastering Format
Oct. 28 and an applications seminar on the Sony APR -24/APR-
Sound. Technology.
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Circle 15 on Reader Service Card
Larry Zide
Associate Publisher
Elaine Zide
Senior Editor
John Barilla
Assistant Editor
Caryn Shinske
Contributing Editors
Bruce Bartlett
Brian Battles
Drew Daniels
Robyn Gately
Len Feldman
Shelley Herman
Brent Harshbarger
Randy Hoffner
Jim Paul
Graphics & Layout
Karen Cohn
db, The Sound Engineering Magazine(ISSN 0011 -7145)
is published bi- monthly by Sagarr re Publishing Company Inc. Entire contents copyright 1991 by Sagarnore
Publishing Company Inc., 203 Commack Road, Suite
1010, Commack, NY 11725. Telephone: (516)586 -6530.
db Magazine is published for individuals and firms in
professional audio recording, broadcast audio-visual,
sound reinforcement -contracting, consultants, video recording, film sound, etc. Application for subscription should
be made on the subscription form in the rear of each
issue. Subscriptions are $15.00 per year ($28.00 per
year outside U.S. Possessions, $16.00 per year in Can ada)and payable in U.S funds. Single copies are $3.50
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Trademarked names are editorially used throughout this
issue. Rather than place a trademark symbol next to
each occurance, we state that these names are used
only in an editorial fashion and to the benefit of the trademark owner, and that there is no intention of trademark
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Circle 18 on Reader Service Card
The Editor
One of the more difficult tasks
assigned the less -experienced
and equipment-endowed engineer is the recording or reproducing of the human voice
singing or speaking. Toward
that end we have been reading
John Barilla's excellent series
in each issue. His Designing
Vocals series has so far not
only hit the mark but given us
some insight into what we can
do with our existing mics and
sound shaping electronics.
We've managed to develop a
decent living out of the local tal-
ent pool that rotate through
our clubs. Of course, much of
the work is recording live in
those clubs with a minimum of
mics and onto a Fostex R8 with
a Tascam 12/8 console in a portable case. Often, we are also
asked to augment the house
sound. Toward that end the
wagon also has several speaker
systems, and a well -travelled
Crown DC -30OA.
The bottom line is how much
we have learned from db
Magazine articles, and particularly Mr. Barilla's Designing Vocals. Will there be
learn more about the Sennheiser excellence in design and engineering, please call or write.
6 Vista Drive, PO Box
Old Lyme, Cf 06371
# 203 -434 -9190
Manufacturing Plant:
D -3002
# 203-434 -1759
Wedemork, Germany
Circle 20 on Reader Service Card
Jerry E. Cohn
LIVE Music, Inc
Anaheim, CA
From the Editor:
Jerry, we though you'd never
get to the last question, but you
did. John's column is in this issue and covers more on Designing Vocals. What's more,
he has told us that he has more
to say. It will be in the November /December issue.
We're making the laws of physics
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If you think it's easy to develop full performance from
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It was no small task to make so little do so much.
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(800) 992 -5013 Fax (508) 2.34-8251
Circle 21 on Reader Service Card
(508) 234 -6158
Some of you are seeing db Magazine for the first time, having picked up your copy
at the AES Convention. Hopefully, you like what you are reading and will join our
growing family of professional audio subscribers. Whether you have entered this industry for only a few days or for many years, we have something for you. Welcome.
We do not do record reviews, leaving that chore to those consumer music magazines that are out there. So, why am I now writing one such review? That's because,
while the recording in question does contain music, it is its technical engineering
qualities that make it important to note.
The recording in question is from Delos International, Inc. and it is called Engineer's Choice. The engineer is John Eargle who has had an over -25 -year career as a
recording or chief engineer for RCA and Mercury Records, and is now Director of Recording for Delos.
The CD album contains twenty -two cuts of excerpts from Delos releases, each
picked by Eargle to illustrate a microphone and/or orchestral pickup technique.
All the selections are of classical works, but range from oversized orchestra to solo
piano, vocal, or chamber selections. Each is an example of recording/microphone
placement at their very best.
Here's a quote from the extensive liner notes that Eargle wrote: "...I have selected
22 examples covering musical styles from classical to modern and ensembles ranging from solo instruments to orchestra. I will explain in detail how and why each microphone setup was used. There are nine examples which feature the Seattle Symphony Orchestra directed by Gerard Schwartz. This should come as no surprise,
inasmuch as that ensemble records more CDs today than any other U.S. orchestra!
The Seattle Opera House has become a laboratory, so to speak, and has given me
the opportunity to experiment and refine orchestral recording techniques."
The CD is a textbook of classical recording technique and therefore belongs in the
tool chest of every present or future soundman. I strongly recommend it. L.Z.
Meet db Magazine at the
Convention at Booth #1722
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Circle 22 on Reader Service Card
Advision: The ..s Top
o e Recording
More Than The U.K's Top Mobile Recording Facility Advision: A Powerful Three -Way CombinationRemote Truck, Recording Studio & Post Facility.
United Kingdom recording
studio marketplace has undergone some profound funda-
mental changes in the past year.
Continuing pressures on studio
rates, the growing number of project or private studios, high interest rates and new business tax
rates in parts of the country have
made it even harder for U.K studios to compete. Some studios have
been forced to leave the heart of the
U.K recording in-
Hopkins, Advision is comprised of
several individual facilities, offering a highly effective and symbiotic
combination Europe's top mobile
recording facility, /a newly opened
state -of-the -art recording studio,
and a top -flight post -production fa-
Having recently relocated from
the West End of London to Brighton, the seaside resort on England's
south coast, Advision has always
charted a unique course through-
mammoth rent increases and a new
business tax rate.
Faced with these
pressures, a number of studios have
changed hands or
gone under, and
even such London
recording institutions as Air Studios have made
the decision to re-
United Kingdom to offer eighttrack recording facilities.
Until its recent move, Advision
was located for 21 years in the
heart of London's West End where
it had grown into an expansive,
multi- faceted facility -three recording studios offering both analog and digital 48-track recording, a
digital editing suite for compact
disc production, a programming
suite, video /film post -production
facilities, and the Advision Mobile.
The Advision Mobile continues to be
widely used across
Europe for live concert recording and
work. It has been
used to record
many major music
festivals, including,
most recently, the
Nelson Mandela
Freedom Concert at
worth '90, The
Prince's Trust, and
Paul McCartney's
latest "Unplugged"
locate to new faalbum.
Other notable live
In this era of turc moil and in- Figure 1. The Advision Mobile is used throughout Europe for live con- events handled by
the Advision Mobile
IS creased specialicert work, television sound work, and many music festivals.
include the Comzation in the U.K
monwealth Games
out its nearly forty year history.
Q studio market, Adand the Donnington Monsters Of
o vision Ltd. remains one of the leadRock '90 Festival in addition to
16 ers. Today, led by Director Doug
work for Anglia Television, GranAdvision was formed in 1954 as a
ada Television, London Weekend
film production company
Television, Tyne Tees Television
sound studio. By the early 1960s, it
and Yorkshire Television.
had dropped all film work and was
Randolph P Savicky is president
only. In
A short list of music clients infact, Advision was the first studio in
N port, NY, a public relations and
Bon Jovi, Bruce Spring magcludes
the United
marketing company specializing
Clapton, Prince, Placido
Domingo and U2.
became the first studio in the
and broadcast industries.
In the right hands they're
incredible tools.
We'd like to introduce you to the
Crown PZM*-30F and 30R, two extremely versatile and indispensable
microphones for the recording studio.
Like other professional tools, it
takes some experience and skill to
discover everything they're capable of
doing. But, when used properly, they
do what no other microphones can do.
Unlike conventional mics, the
pressure -zone design of the 30 series
uses a miniature condenser mic
capsule to receive direct and reflected
sound simultaneously. Direct sound
from the source and reflected sound
from a wall, floor or other boundary
combine in -phase to produce a wide,
smooth frequency response free from
phase interference. The result is
increased sensitivity, superior clarity
and "reach" with little or no off-axis
coloration. All you receive is clear,
natural sound.
Because of their unique PZM
design, the 30 series excels when used
to record sound near hard reflective
surfaces such as pianos or instruments
surrounded by reflective baffles.
They're also an excellent choice for
recording room ambience or, when
used in pairs, for recording near-coincident or spaced -pair stereo.
With some experimentation, you'll
find the 30R and 30F open a whole
range of possibilities and solutions not
available with any other microphone.
We think you'll find them an important
part of your recording "toolkit."
Like all Crown professional mics
the 30 series carries a full three year
unconditional warranty
against malfunction
with a lifetime warranty on the acoustic
system. **
For more
information on our
complete line of
PZMs or a free
Call for
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copy of the Bound- your
the Boundary Microphone
Application Guide
ary Microphone
Application Guide see your Crown
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*The PZM -30F features a flat frequency response, the 30R
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** See your Crown dealer for complete warranty details.
P.O. Box 1000
Elkhart, IN
46515 -1000
© 1990 Crown Internatonal, Inc. PZM and Crown are
registered trademarks of Crown International, Inc.
Circle 24 on Reader Service Card
"We feel that the Advision Mobile
is one ofthe best equipped and most
versatile mobile production facilities operating in Europe," said Ultan Henry, Advision's technical engineer. `There are lots of inputs for
live gigs-which is one of those
many areas where you can never
have enough -and the truck is outfitted with an extensive collection
of top -flight, tried and true studio
"We are seeing a continued and
even stronger demand for the best
audio quality possible at live
events," Henry continued. "Concert
audiences are more demanding for
studio quality -like sound, and the
home viewer and listener is even
more demanding. They know how
good their home stereo system can
sound, so when they're watching an
event like the Mandela concert on
television, they expect sound quality that will rival their home systems."
Fully air-conditioned and featuring closed circuit TV monitoring,
the Advision Mobile is built around
a 62 -input Helios console with full
equalization, four auxiliary sends,
split 24 -track monitoring with aux
sends, mix-minus, solos, cuts, and
VU and PPM metering. There are
in -line monitoring facilities for 48track recording, and it is possible to
record from the monitor independent of solos or cuts.
Sony PCM -3324 24 -track digital
tape recorders are available for 24track or 48 -track digital recording,
and there are two Otari MX80 24-
track analog tape machines and
two Studer stereo tape machines.
On hand are 24 channels of Dolby A
noise reduction, with Dolby SR
(Spectral Recording) available as
audio problems on another remote
"We also feel that this ability to
continue working on a project once
the live event is over is particularly
attractive to our clients," Henry
added. "In this way, we are able to
insure the audio integrity of the
work we produce throughout the
whole project."
Advision's new studio facility,
which opened in October 1990, is
housed in a converted church close
to Brighton's many shops and leisure facilities. A fully residential
recording studio, only one hour
from London, Advision offers one of
Europe's finest recording rooms,
technical expertise, the latest in
digital technology, and first class
accommodations and catering.
"We wanted to open a residential
facility," Henry said, "but we didn't
want the typical English residential facility, a mansion in the country with horses. Instead, we
wanted something unique and different.
"We found that in Brighton," he
said. "Brighton offers the sun and
the sea, as well as clubs, pubs and
nightlife. It's the perfect place to
work during the day and then go
out at night."
The studio itself, which is naturally lit by high arched windows, is
approximately 36 feet by 33 feet by
30 feet. There is a dead/soft booth
that is approximately 10 feet by 10
The equipment includes a Solid
State Logic SL 6048 E 48 -input console with G Series Studio Com-
puter and Total Recall, two Sony
PCM -3324 24 -track digital tape
machines, Otari MTR-90 24 -track
analog tape machine, Otari MX80
Figure 2. The studio is housed in a converted church in Brighton and
is naturally lit by high arched windows.
feet, a live/hard booth that is approximately 10 feet by 15 feet, and
an isolation area that is approximately 10 feet by 10 feet. Afloating
riser for drum kits, etc., measures
approximately 12 feet by 15 feet.
"The church's natural acoustics
give the studio a wonderful sound
quality that would take a great deal
of exacting planning and careful
construction to recreate in any
other space," Henry said. The control room is a spacious 15 feet by 27
feet with an annex for meetings,
coffee, and such. Both are also lit
with natural daylight.
The decision to scale down the
studio's scope from its London size
also resulted in an even better recording facility with a wealth of top
studio gear, Henry explained.
"With three studios, you can be
sure we had a lot of equipment," he
said. "What we decided to do was
keep all the best equipment and
consolidate it into one superb studio. This also allowed us to expand
our range of outboard gear and our
selection of mics, offering a much
wider and deeper inventory than
any multi -room facility could possibly dedicate to one room."
The Advision Mobile also includes an extensive selection of microphones and a well- rounded collection of outboard gear.
"Much ofthe work is brought back
from the Advision Mobile to the studio for mixing," Henry noted. "We
feel that we have the unique ability
to know how to capture the live
sound correctly so that any potential problems are eliminated while
we are recording a concert or event.
Our house engineer, Pete Craigie,
is available with the mobile to oversee the project from recording to
mixing. We can't try to blame any
don. We're also looking to increase
our share of album projects. In fact,
Jason Bonham is coming in soon for
six weeks to work on his new LP."
To house clients of this caliber,
Advision offers a bright and sunny
rooftop apartment. The sitting/dining room has a view of the sea
through a rose window and glass
doors that open onto two sun terraces. The apartment is completely
self-contained with a fully -fitted
kitchen and three double bedrooms, including one with its own
Figure 3. The decision to scale down the studio's size resulted in an
even better facility. Here, the new SSL is being moved in.
24 -track analog tape machines,
and Studer half-in. and quarter -in.
tape machines with center -track
time code.
Noise reduction includes a Dolby
XP24 rack and four Dolby 361 with
A type noise reduction, and digital
mastering is offered via Audio + Design ProDAT, Sony PCM -701 and
Sony C9 Betamax.
"For music -to- picture work, we
use TimeLine's Lynx Synchronization system," Henry said. `The
Lynx modules are easily configured
for various machine setups and
have proven quick and reliable to
"Let me add, however, that we
also offer the friendliest atmosphere in the industry," he continued. "Great equipment is only part
of the secret of a studio's success. If
you don't have the right people and
the right attitude, you won't have a
successful studio."
Some of Advision's studio clients
include David Bowie, Dire Straits,
Elton John, George Michael, Paul
McCartney, Peter Gabriel, Pink
Floyd and ZZ Top.
"We're also getting a lot of new clients," Henry noted, "including a lot
more dance music than we attracted when we were back in Lon-
Figure 4. Upon completion, the control room is 15 feet by 27 feet with
an annex for meetings. Both are lit with natural daylight.
private bath.
The basement of the studios includes a main recreation and
lounge area, kitchen and dining
room. The basement also includes
another two large double bedrooms, including one with its own
private bath.
Advision's third facility is a post production suite based around a
tapeless recorder /editor/digital mixer. Henry
explained that the Opus system
was originally installed in the Advision Mobile approximately one
year ago.
"Many broadcast companies were
very enthusiastic about the idea of
a system such as this being available for hire on an as and when
needed basis," Henry said. "However, despite the enthusiasm, most
British video work is still based in
London's West End where parking
is a problem.
"Our Opus has now been installed
within a video facility in this area
known as Wiseman," he continued.
"The system is heavily booked, with
many clients being converted to the
benefits of hard disk editing."
Several television series for
Channel 4 are now post -produced
on the Opus system, including `The
Crystal Maze" and "Star Test,"
Henry noted.
As Advision enters into its fifth
decade in the studio business, it
seems perfectly situated and
streamlined to continue as one of
the leading facilities in the years
Advision Mobile-Equipment
62 -input Helios console
Sony PCM -3324 24 -track digital
Dynamite gates
Scamprack with parametric EQ
and gates
UREI graphic EQs
tape recorders
Otani MX80 24 -track analog tape
Studer stereo tape machines
Dolby -Anoise reduction (24 chan-
The Studio- Equipment
Solid State Logic SL 6048 E 48input console with G Series Studio
Monitoring: Altec/Crown, Yamaha NS10,Auratones
Multi -standard time -code generators and readers
Color closed circuit video monitoring
250 meters of 24 mic and communications cable
4 x 24 way custom -built splitter
Comprehensive range of cables,
stands, direct boxes, etc.
Mies: AKG, Beyer, Crown, Electro-Voice, Neumann, Shure and
Outboard equipment:
UREI 1176 limiters/compressors
Drawmer compressor
Drawmer dual gates
Audio + Design Compex limiter
Kepex Gain Brains
Yamaha SPX90
Ursa Major 8X32 reverb
Lexicon PCM60 reverb
Ursa Major Stargate reverb
AMS RMX16 reverbs
Quantec QRS
Lexicon PCM60
Yamaha Rev7
Yamaha SPX90
Lexicon 480L
Kong SDD3000 digital delay
Lexicon Super Prime Time
Drawmer DL201 dual gates
Computer and Total Recall
Sony PCM-3324 24 -track digital
tape machines
Otani MTR-90 24 -track analog
tape machine
Otani MX80 24 -track analog tape
Studer half-inch or quarter-inch
tape machines with center-track
time code
Denon DRM24HX cassette decks
Audio + Design ProDAT
Sony PCM -701
Sony C9 Betamax
Sony 5850 PAL U -matic
TimeLine Lynx Synchronization
Dolby XP24 rack
Dolby 361 with A type noise reduction
Monitors: Dynaudio Jade 2, 3 k
watt monitors, Yamaha NS10 Auratones
UREI 1178 dual limiter
UREI 1176 limiters
dbx 160 dual limiter
dbx 160X dual limiters
Compex limiter
Maglink time -code reader
Pultec valve EQ
dbx 902 de- essers
Bel Flanger
Eventide SP2016
Marshall Tape Eliminator
Tempo Check Metronome
Audio + Design Panscan
Ursa Major MSP126
Drawmer DL231 expander/compressor
Audio + Design Xpress limiter
Marshall Time Modulator
Bosendorfer grand piano outfitted with MIDI
Outboard equipment:
0, NI
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Creating a Church Recording
In 1970, sermons at the First Church of the Nazarene in Bethany, OK, were recorded primarily for the
pastor's benefit so he could analyze the outline, delivery and continuity of his sermons. The church did
not have a permanent tape recorder, so an AKAI 770X was carried up and down two flights ofstairs every
Sunday for four years.
there became a demand
for cassette copies of the
sermons as well as copies of special services and big musical services since music is of major
importance to the church's worship. As a result, Sunday services
were recorded and duplicated on a
small scale.
In 1974, there was a major overhaul of the sound system. The
church later invested in a highspeed duplicator to produce more
tapes in a shorter amount of time.
Some congregants were serving as
missionaries in foreign countries
and requested tapes of the services.
In 1978, the church decided to
move the recording operation to another location within the church. A
room had originally been set aside
for members to bring their tape recorders to record any service. After
gathering ideas and putting them
together along with a few pieces of
equipment, we had a recording
We installed two organ mic lines
and two audience mic lines and terminated them at the studio. There
were also some great times with
those Shure M68 mixers and an EVlTapco 12- channel mixer. Around
1982, we took a giant leap forward
and purchased a Tascam M15
mixer! It came with only eight input modules, and we knew we
would add more. With our tape
ministry doing as well as it was, we
went on a "module-a- month" plan,
filling out the board to 24 channels
to mix and 8 outputs. Cabinets
were added to the studio, and the
table top was cut down to insert the
1. The view of the studio. The sanctuary is off to the left through
the window.
mixer and have its controls at
working desk height (see Figure 1).
At this point, we decided that in
order to hear all the sound that we
should be hearing, we should have
some excellent studio monitoring
speakers. Some old floor monitor
boxes were worked over, and some
15 in. 2-way speakers were installed. A Tascam 80 -8 8- channel
recorder with a dbx encode/decoder
was the church's next big purchase.
After that, there was a need to keep
the recorders from being over-recorded and distorting the tapes, so
a UREI 1176n 2-channel limiter
was added to the sound system. It
keeps levels up to the desired setting without distorting the recordings. The church next decided to
purchase a reverb unit to aid in
more quality recordings. Purchasing a unit turned out to be almost
the hardest decision we faced because of the vast number of units
available. We selected the Yamaha
REV-7, which is used almost exclusively for all music in our mix. The
Aphex Aural Exciter also adds a
nice touch to the vocal solos and
groups and their place in the mix
(see Figure 2).
Currently, the studio consists of a
24- channel mix board, 12- channel
side mixer, digital reverb, 2 -channel limiter, audio processor, auto reverse stereo cassette deck and an
8-track/8- channel recorder. The
studio monitoring system consists
of a 2- channel equalizer, 2- channel
power amplifier, and two studio
quality monitor speakers (see Figure 3).
This setup is unique in several aspects. The first is mixing all audio
activity as if it were a live album.
Next is the fact that there are two
mix locations, one for house sound
2 OF;Dhti MIES
Figure 2. The control-room wiringpatching system.
Pdtchr 1DIGITfil.
T 0 /FROM. Ffi
and one for recording. The third aspect is that in most cases, one microphone feeds two mix board inputs without Y transformers.
Fourth is the installation of patch
lines between the two boards and
assignment of wireless mies and
trax feed. The fifth aspect is the installation of audience mies and organ mies which are only available
to the recording board.
The mix board was chosen because of its flexibility; since it is a
studio board, not a PA board, it has
specifications which match most
studio boards. There are eight main
outputs which are normalled to the
8 -track inputs. One of the two stereo outputs is normalled to the auto
reverse cassette, one mono output
is normalled to the reverb input,
and the other is patchable to feed
the hard -of-hearing transmitter
and the foyer speaker system. As a
result of the board's flexibility, it is
possible to route all of the above signals to their desired destination in
one take. If the output is used to
feed television or radio, then the
and there are thirty preset and
sixty user- modified settings. The
front panel is the user end of a
small computer. The wired remote
control has become an often used
item, even with limited control.
A 2- channel limiter was the next
item on the never -ending list of
"wants" for the studio. The limiter,
which was chosen for its accuracy
and reliability, has an extremely
low noise figure
major concern
in the studio. It is a very useful tool,
especially for controlling an input
with dynamics to spare.
The newest unit in the studio is
one that re- creates and enhances
harmonics. It keeps a solo voice
above an orchestra and choir.
The studio monitoring system is
made up of a 2- channel equalizer,
stereo power amplifier and 2 -way
speakers. The system is equalized
for the flattest possible sound, considering the room acoustics.
The control room is set up on the
live- end -dead -end (LEDE) arrangement. A thick carpet covers
the rear wall to eliminate bounce,
mix is basically the same as a recorded mix.
An 8- channel/8 -track recorder
was basically chosen because of its
adaptability to the board and its
price at the time. It records eight
channels at one time, which puts it
in the multi -track category. The 8track has all of the features its big
brothers have, except punch -out.
One big feature is the built -in dbx
encode /decode adapter, while another nice addition is the remote
controller which sits on the board.
The auto reverse cassette deck
was selected for its automatic features. The unit records in either direction, a plus, since there is no
time in the middle of a sermon to
turn a tape over. This particular
deck also has dbx encode/decode
built in. The remote control unit is
also a very handy addition.
The digital reverb is a fairly new
addition to the studio and was selected because of its versatility. It is
a stereo unit with mono capabilities
Let's face it.
As an electronic musician, the new board you'd really like to
buy has keys, not faders. In fact, you probably wouldn't be
buying a new board at all if you didn't need more inputs.
That's whyTascam's M -600 Input Console is the board for you.
It gives you top panel access to as many as 64 stereo or 128
mono inputs. That's capacity.
Equally accessible is the M -600's surprisingly low price. You
can put one in your studio for less than $10,000, thanks to its
modular design. Starting with 16 channels, the M-600 expands
to a full
32 as you need them.
But despite its attractive price, the M -600 has that great
'sound of boards costing twice as much. And when you experience its intuitive feel, the way everything is where you need
it when you need it, you'll know this is the board you've been
looking for.
Write for our free Configuration and Installation Guide. Then
get your hands on the powerful M -600 Input Console at your
Tascam dealer.
It may be the last board you ever have to buy.
©1989 TEAC Amerza,
Inc., 7733 Telegraph Road,
Montebello, CA 9064Q, 213!726 -0303.
Circle 23 on Reader Service Card
and a window in front of the mixer
isolates the studio sound from the
sanctuary sound.
An extensive patch bay system
rounds out the studio's versatility.
All needed inputs and outputs are
grouped as to mic and line signal
levels, and some are normalled. In
the middle of the bay separating
mic levels and line levels is a special
panel with impedance matching
transformers plus switching to access the patch lines to the PA board
on the sanctuary floor (see Figure
3). When the PA operator is asked
to play a soundtrack, it can be sent
through one or two patch lines if a
stereo track is used. The normalled
outputs through the patch bay are
as follows: the 8 -track outputs are
normalled to Aux A, Aux B, and
mixdown inputs to the board; the 8track inputs are normalled to their
respective group master outputs,
the cassette recorder inputs are
normalled to the Mon B output; and
the cassette outputs are normalled
to Tape 1 at the control room selector. The audio processor is also inserted into group masters seven
and eight.
There was a big debate as to the
addition of mic Y transformers due
to a phantom power problem, the
loading down of the output of the
mic, and the expense of thirty -five
transformers. After weighing all
the facts and a trial hookup, a decision was made simply to Y the mic
jack to both mixers. A test was
made for loss ofsignal and degradation of quality to the two mixer inputs. lb our surprise, there was
very little audible distortion and
only about one decibel ofsignal loss.
After several years of dual use and
several different types of mics, we
felt we made the right decision in
not using a Y transformer.
An early decision was made to recreate as much of a live feel as possible, since the philosophy of the
studio is to re- create the feel of being tenth -row center. Every service
is a live album and therefore requires a lot of attention. There are
two boundary -type mics, hung
from the ceiling near the first row,
to audibly separate the orchestra
from the audience. We also added
two hanging mics above the organ
Figure 3.A close up of the studio
equpment rack
and Tascam 80-8 recorder.
Tascam 15 24 x 8 mixer
E- V/Tapco C12 Catalina 12 x 4 mix-
Tascam 80 -8 recorder/reproducer
with DX-8
Tascam 33405 4- channel recorder/reproducer
Sony TC -755 2- channel recorder
Revox A77 half-track recorder /reproducer
Tascam V95RX cassette recorder
Uni -Sync 50 2- channel power amplifier
Yamaha REV 7 digital reverb
UREI 1176n dual limiter
Aphex 103 Aural Exciter
The studio is centered around the
Tascam M -15 mixing console. AuxA
controls the output to the Williams
transmitter and the foyer/nursery
amplifier. Aux B sets the input level
to the reverb. The green knobs,
Auxes 3 and 4, set the input to Aux
B mix which is set to about the 12
o'clock position. The reverb return
is first routed through echo receive
7 and 8 controls and assigned
switches. To rough mix the 8- track,
we select Tape A, 1 through 8, and
adjust Mon A, 1 through 8. Mon A
on Echo Rev (Receive) is selected for
reverb. To set up for a mixdown, input select on modules 1 through 8 is
located and set to the unmarked,
square box position. Next, the trim
is set to 11 o'clock and Mon B is selected at the Control Room module
and set to the desired listening
level. Since the cassette recorder
input is normalled to Mon B output,
a mix can be performed.
Cassette play to Control Room
Line 1 in
Monitor B output to Cassette line
Aux B send to Reverb Left/mono in
Reverb Left to Echo Receive 7
Reverb Right to Echo Receive 8
Aphex Left to Bus Access 7
Aphex Right to Bus Access 8
Aux 1 through 8 out to 8 -track inputs 1 through 8 respectively
8 -track out to Tape A, 1 through 8,
Patch line 1 (wireless 1) to M-15
line in 11
Figure 4. The Sanctuary. This view is from the sound -reinforcement
console position
Patch line 2 (wireless
2) to M -15
line in 12
Patch line 3 (track R) to M -15 line
in 13
Patch line 4 (track L) to M -15 line
in 14
The patch is set up in a waterfall
arrangement; the top two rows are
outputs and the bottom rows are inputs of like devices. In other words,
mic outputs connect to mic inputs
and line outputs patch to line inputs. The mics are all balanced
(three wire connections), and the
line in/outs are unbalanced (shield
and signal connections). Do not
patch between mic and line in or
out. There are signal level differences (1,000 times) and phantom
voltage ( +40 volts DC) on the mic
lines. Also, do not patch one end of a
patch cable to a mic outlet and
touch the other end to your mouth.
If a patch cord does not appear to
work, exchange it with another until it does work, then mark the one
that doesn't work. When finished
with the project you are working
on, disconnect all patch cords and
return all controls to the setting
you found them (or as near as you
We use Sony ECM -44 clip mics for
first and second violins. The players, at our instructions, clip them
below the bridge pointed towards
the played strings. Due to the proximity of the strings, we have to pull
off the extreme high end EQ. We
use Sony ECM -30 clip mics on violas and cellos with the same EQ
considerations. Sony EMC -33s are
used on flutes in a peeking -overthe- music -stand boom arrangement. A Neumann KM -84 on a
short stand is placed at the music
stand to cover the clarinets. On
trombones, a middle height stand
is used under the music stand with
a Neumann KM -84 mic. Rather
than a tall boom stand on tuba, we
use a Sony ECM-30 clip mic on the
bell of the upright Yamaha.
Sometimes the french horns are
mic'd with a Neumann KM -84 on a
short stand to the right of the
player. The trumpets are the biggest problem in that their play levels change many dB during the
same song. We have tried many
ways to solve this dilemma and
have settled on two Neumann mics
at music stand height. We also use
two Neumann mics on the grand piano -one on the high end and one
on the low strings. AKurzweil electronic piano is used direct through
Shure A -95 impedance- matching
transformers. When we have guitar -bass, we take a DI out from it.
We have had a harp from time to
time, and have found a way of
mic'ing it that works best for uswe use a Sony ECM -30 clip mic and
wrap it one time around the main
post, about 18 in. from the floor, and
point the mic toward the sounding
board. We clip the mic to its own
lead wire and then route the wire in
front and then to a mic outlet. This
close- mic'ing technique is employed for a number of reasons.
First, we have to satisfy both PA
and recording requirements. Second, it is a live recording situation
and we have to make the best of it.
If we place a string -group mic in the
air, our mix will suffer because
there are stronger sounds than
strings. We have used a brass group mic with fairly good results,
except that if it is in the air for
height, it tends to be obtrusive to
worship. Third, the technique with
minor EQ adjustments comes close
to group mic'ing, except we have to
pay a little more attention to the
mix. When we do a mixdown of the
music and want to get the most
from it, we first have to plan our
musical strategy; we plan and mix
each section as if it were an overdub
session. With proper mic'ing techniques and equalization, that plan -
ning pays off.
Occasionally we hear a brass
player cleaning out his spit valve or
a string player plucking his strings
to check his tuning, or an occasional
wrong note. Ahh, such are the
things to look forward to in a live
situation. Then we have to decide
(in the mixdown) whether to pull
that part out, or bring that group in
and redub, or try again later. For
the choir area, we use eight to
twelve KM -84s depending on the
occasion. These mics are placed
mainly to satisfy PArather than recording.
The art of doing a live recording
has been superseded by multi tracking and studio sessions. The
tapes produced by our churches
should represent our best, and with
proper planning, they can be a reality. We have been pleasantly surprised by a request from a local radio station for our music for their
Sunday programming. When we
did a television program, we did our
own audio mix because we felt it
put our best foot forward. Our television producers were free to do
their camera magic and did not
worry about the audio. It's a lot of
work, but very rewarding when we
get comments like, "Get a
sounds better than live."
The AES 91st Convention
DATES: 1991 Oct. 4 through Oct, 8 LOCATION: New York Hilton, N.Y.C.
Plan to attend the most productive
Professional Audio Event of 1991...
the 91 st AES Convention in New York
Audio Engineering Society, Inc.
60 E. 42nd Street, New York, NY 10165 USA
(212) 661 -8528 OR (E00) 5417299 ...except N.Y.
Fax: (212) 682 0477
Telex: 62098UW
Audio for the National
Victory Celebration
On June 8, 1991, Washington, D.C. witnessed its largest military parade since World War 1, part of an
elaborate tribute and "welcome home" for the military veterans of Operations Desert Shield and Desert
hardware, from tanks and
guns to a "fly -by" by every
type of helicopter and aircraft used in the Gulf theater of operations, were included. To enhance the public's enjoyment of
this extraordinary event, an extensive sound reinforcement system
was required. The challenge presented by an event of this magnitude was substantial; I was pleased
that RCI Sound Systems, audio
contractor for the parade and
"thank -you" picnic, subcontracted
my engineering services for the
Sound reinforcement needs for
the celebration were twofold: an extensive speech -only reinforcement
system was required to cover the
The Ellipse stage and audience area.
to insure quality sound reinforcement over a large area. The parade
system posed a whole different set
of challenges; it proved that in our
haste to modernize, we shouldn't
neglect the value of our older gear.
Hargrove, Inc., of Washington
D.C., the overall event coordinator,
Figure 2. RCI Sound Systems engineers Lorne Greene (L) and Craig
Jensen (R) adjust the picnic grounds system. Delay systems and wireless transmitters are visible in the background.
entire parade route, and the picnic
grounds needed a large concert
sound system, capable of handling
the varied audio needs of different
military bands and covering a
crowd of 25,000 military personnel
and their families.
Rick Shepard, RCISS managing
partner, planned to use the company's standard concert rig (see db
Magazine, March/April 1990),
augmented by large delay towers,
to cover the Ellipse picnic ground's
stage and audience area (see Figure 1). The delay stacks each had
their own AC generator (60 amps,
single phase 120 volts), and were
fed from the central mix point via a
Yamaha DD1 -3 into dedicated
wireless links; there would be no
cable runs to the distant stacks, insuring both a clean "look" and
safety. A Yamaha PM -3000 40channel console, with a full complement of signal processing equipment, controlled the house system
(matrix outputs fed mains left and
right, delay towers and a press
mult) and also provided four monitor mixes for stage foldback (see
Figures 2 and 3). RCISS used some
of today's most modern equipment
Figure 3. A view of picnic grounds stage with the main house stacks
flown from scaffolding bays on either side.
envisioned sound reinforcement as
an essential element of the parade.
To an average viewer, the parade
would look like troops marching
down Constitution Avenue with
odd military hardware accompanying them. Narration would greatly
enhance the experience for all concerned; not only could an announcer identify the unit and/or
equipment rolling by, he could also
document their roles and subsequent successes in the Gulf operations. These announcements
would both inform and honor,
which was the whole idea behind
the parade in the first place.
The reality of this sound- for-narration idea was sobering: cover two
miles of parade route with even
speech reinforcement, paying particular attention to the President's
reviewing stand and VIP seating
areas. With such a long route, it
was obvious that a single narrator
wouldn't work; the route had to be
split up into "zones," so each area
could get a description of what was
in view at that particular moment.
That meant separate systems for
each zone. Whatever equipment
was used had to generate enough
level to compete with both military
brass bands and tank engines;
sound had to be clearly audible on
both sides of the street, with coverage for an estimated audience of
one million. Lastly, we had to be
careful that the sound from one
zone didn't spill into another, resulting in a confusing cacophony of
narration. The system had to be
tamperproof and weatherproof in
the middle of a major city with no
security to watch it at night. No cable runs could touch the ground in
non -secured areas.
The only place to put a parade
paging system on a city street, for
both coverage and security reasons, is in the air. RCISS decided to
use street lamp poles as the means
Figure 4. A close -up of the Unistat fastening system for the D.C. lampposts.
to elevate speakers to a height of
14 -16 feet. Shepard had three im-
mediate concerns: "permits, permits and permits. We had to get
permission from the DC government to actually hang our equipment from the lamp posts and
string the wire between them. We
started pursuing these permits a
good two weeks before work was
scheduled to begin. I've learned
that any delay with city government can be fatal to your schedule.
Any streets crossed by cables re-
quired a minimum clearance
height of twenty feet; crossing Constitution Avenue required an extra
five feet of clearance for parade
Figure 5. A fourhorn cluster located on the corner of 7th and
Constitution at
the beginning of
Zone #1.
floats and military hardware. We
figured we'd need about a week to
get the system up and working, so
we also had downtown parking to
consider." Nicely understated.
Most of the work on the parade
route took place between May 31
and June 7; the normal business of
Washington, D.C. did not stop
while we were out on the street
stringing wire and hanging equipment, so the job became a creative
game of dodging cars and searching
for parking spots. Constitution
Avenue is a major artery during
rush hour, with parking restricted
from 4 p.m. to 6:30 p.m., and rush
hour stops for no one. Even with
permits, motorists and police took a
dim view of our rolling scaffolds
and scissor lift blocking precious
traffic lanes. Despite these obstacles, the work was completed on
The speaker system used for the
majority of the parade route was
comprised of outdoor paging horns.
For the most part, these were Electro -Voice 848A re- entrant fiberglass horns with either 1828 or
1829 drivers. Each horn incorporated a 70 volt transformer, tapped
for either 15 or 30 watts, depending
on the driver. Not exactly cuttingedge technology, but perfect for this
application. "These paging horns
are the most efficient means of conveying speech to a large area," said
Shepard. "Their pattern is easy to
control, and they have a relatively
full bandwidth for speech. In areas
where I could only put horns on one
side of the street, I knew they had
the `throw' to reach the opposite
side with sufficient spl. They are
weather- resistant, reducing the
likelihood of problems from inclement weather. The only exposed surface is the horn itself; that and the
fiberglass construction offer some
protection from vandalism, an important consideration when you
have to leave them up on a city
street overnight with no security."
The fastening system was based
on two pieces of B -109 Unistrut (1
1/4 in. x 11/4 in.), bolted across the
street lamp fixture with 1/2 in. x 9
in. galvanized bolts (see Figure 4).
At selected street corners, additional Unistrut was attached to
form a vertical boom; this provided N
The zonal layout.
the extra height needed to attain
the twenty feet required for cross street cable runs. Up to four horns
could be attached to the basic strut
with grade 5 bolts and aimed in any
direction (see Figure 5). Each horn
was equipped with a lamp safety
cable for additional protection.
Over one hundred of these horns
were used in zones 1, 2, 4 and 5, and
parts of 3N and 3S, forming the
largest outdoor paging system seen
in Washington, D.C. since the famous Dr. Martin Luther King, Jr. "I
Have A Dream" speech in the
1960s. By carefully aiming the
horns and leaving about two hundred feet uncovered between zones,
we minimized zonal interaction
(see Figure 6).
The reviewing stand area required special treatment (see Figure 7). Bleachers surrounding the
President's specially -built reviewing stand were designated as VIP
seating. A higher standard of fidelity was required here, yet weather
resistance was still a parameter.
The reviewing stand was a secured
area (Presidential security is
tight), so RCISS elected to use E -V
Musicaster II speakers for bleacher
coverage. AMusicaster consists of a
12 in. woofer and T-35 tweeter in a
weatherproof bass -reflex enclosure; twenty of these were used in
N the area, mounted via the same
Unistrut system. Musicasters will
system. Four JBL Control 5 and
two JBL Control 12 speakers were
flown from the reviewing stand roof
(see Figure 10): the Control 5s covered the upper rear area (closer to
the roof) and the Control 12s covered the lower frontal area (further
from the roof). Incorporated into
this lower area was a special cubicle for President Bush, constructed
of bulletproof glass. Inside this cu-
not throw sound a long distance, so
cabinets were placed on each side of
the street and focused to cover the
bleachers immediately below them
(see Figure 8).
The President's reviewing stand
(see Figure 9) had its own special
Figure 7. The reviewing stand area. The M callouts are Musicaster II
speaker systems.
\OMS o
biele were two TOA SM -60 speakers (each containing two 4 in
speakers) that had their own vol-
could adjust their own levels accordingly.
ume controls, so the occupants
Each zone had its own complete
set of mixing, electronics and amplifiers. Most mixers were six channel JBL, Shure or Yamaha rackFigure 8. Musicasters in the
VIP bleacher area.
9. Workmen putting
the final touches on the Presidential reviewing stand. Note
the bulletproof glass enclosure for President Bush.
Figure 10. A JBL Control 5
suspended from the reviewing
stands roof.
mount mixers. There was a press
mult in each zone; most of these
had twenty-four outputs. Each location had a 1 octave graphic
(Yamaha or UREI) for overall system EQ. Crown MT-1200 and BGW
750 amplifiers, strapped in mono,
provided system power.
Most zones used a single amp to
power all speakers, with the exception of zone 3, which had the largest
variations in equipment. The two
mix points in zone 3 (see Figure 11)
had numerous amps and graphics,
with independent sends: paging
horns south, paging horns north,
Musicasters south, Musicasters
north, reviewing stand, President's
cubicle. Zone 6 was another area of
special concern. Due to the proximity of the Vietnam Veterans Memorial, spl was restricted. Asmall Virginia -based sound company, Onyx,
was contracted by RCISS to put
four E -V Delta-Max cabinets on
poles in this area, firing directly
into the Bacon Drive bleachers
from the same side of the street.
This pinpoint, high -fidelity reinforcement helped contain the
sound to the bleacher area, preserving the reverent ambience at
the Vietnam Veterans Memorial.
Each narrator/announcer was positioned on a twelve -foot high platform, adjacent to the press stand in
each zone. We took great care during setup to avoid focusing any
horns or speakers in the direction of
the announcer's platform, hoping
to preserve the best gain- beforefeedback we could.
Mic'ing the narrators posed another challenge -we hoped to avoid
hand-held or stand -mounted mics
that could cause inconsistency in
level. For instance, if the narrator
turned to view what was coming
next, the mic would not move along
with him. The distance between
mouth and mic would change, so we
would experience a level drop. We
solved this problem by giving each
narrator a Beyer DT-108 single muff headset. The headset mounted mic stayed positioned
tight to the mouth; if the narrator
turned, so did the mic. A special
push -to -talk switch, located on the
table in front of the narrator, was
incorporated into the mic line, so
the narrator could control when the
mic was on or off; we didn't have to
worry about open mics. The ear-
Figure 11. RCISS
managing partner Rick Shepard
operates Mix
Point 3S during
system testing on
June 7th.
piece was fed from a foldback send
on the mixer; the narrator monitored his voice through the headset
with excellent gain-before -feedback, so he wasn't confused by the
sound of the PA bouncing off buildings, down streets, etc. A Shure
SM -58 hard -wired hand -held inic
was positioned on each table as a
June 8 proved to be a long day. We
arrived on site at 5:45 a.m., fifteen
minutes before all the streets were
officially closed. Our generators
Figure 12. Just a part of the huge crowd that saluted our troops during
the parade.
had been placed the day before, and
were now surrounded by security
fencing. Most zones had a 40 amp,
120V generator; zones 3N and 3S
had 60 amp generators. Each electronics/amps package was dropped
off at its respective location,
quickly attached to the pre -run,
pre -tested wires, and checked. Our
two days of pre- testing paid off
everything worked when it had to.
The security sweep of the President's reviewing area took twoand -a -half hours, at which point a
huge crowd had already gathered.
The parade came off without incident; a crowd estimated at 750,000
gave the troops a "Welcome Home"
they'll never forget (see Figure 12).
We did have to deal with one lastminute add -on: the Air Force decided to run their own set of wires to
each zone. These were fed from a
central announce location, so an Air
Force announcer could narrate the
elaborate aircraft fly -by to all zones
simultaneously. These wires were
hastily run during the evening of
June 7, and were just completed
when we arrived at the site.
checked these lines, incorporating
them smoothly into our equipment.
Unfortunately, someone forgot to
tell the Air Force about the need for
extra clearance when crossing Constitution Avenue with wires.
The last float in the parade, representing a "thank you" from the
troops to all those Americans who
had sent letters to "Any Servicemen," measured twenty- two -anda -half feet in height. As it rolled
down the street, it neatly severed
the Air Force feed wires, so when
the fly -by occurred, we had no
sound source! Such is the penalty
for overlooking small details; I'm
sure glad it didn't happen to us.
As the parade ended, the picnic
began. The troops and their families were bussed to the Ellipse picnic grounds for free food and entertainment by the Army Blues (a
twenty -piece big band), the Air
Force's High Flight (a Manhattan
Transfer-ish vocal group) and the
Navy's Country Current (a country
music group).
While the picnic sound crew was
hard at work, the parade crews
packed up all their electronics and
returned them to RCISS offices in
Rockville, MD by mid -afternoon.
Everyone enjoyed a truly spectacular fireworks show later that
night. After a well-deserved day off
Sunday, we were back at it on Monday, taking everything down, a job
which took two days. It also made
me appreciate just how long 14,000
feet of wire is!
The Earlham College Monster Language Lab was used by students in 1962 -1963 at Earth am College in Indiana. The one -of-akind Language lab, designed by Cmwn Engineer Wayne Blakesley (right), was used to record and play backfor students in foreign language classes. Max Scholfielcl; the head ofengineering at Crown during the early 1960s, is pictured at left.
Small Group Setup
for Live Recording
Life in studios is getting awfully complicated these days. New this, digital that, advanced and integrated
the other, all very complex, all very glitzy, and most about half-understood.
straints, the modern engineer
wouldn't get any work done at
all; he'd be too busy reading
books on the new equipment. It
would be nice if somewhere in the
midst of all the complexities there
was something simple and dumb
that just worked.
There is.
Before getting to all those wonderful toys, with their hundreds of
knobs and infinite capabilities,
there's the matter of putting out
microphones and picking up sound
from the instruments. That's the
raw material. If it sounds good, you
can do tricks with it and maybe
make it sound very good. If it
sounds lousy, you can do tricks with
it and make it sound lousy. The only
way to make a silk purse from a
sow's ear is to start with a silk sow.
Short of going out and playing
everything himself, the engineer
can't upgrade the sow. A bad band
will be bad under any conditions,
and a great recording only reveals
the horrible details of just how bad
it really is.
On the other hand, the engineer
can damn near kill the sow. It's possible to condemn a good band to bad
performance with an inappropriate
setup, and it happens, because musicians are peculiarly vulnerable to
Amusician who spends his entire
life practicing by himself can learn
to play very well. By himself. Musicians who play in groups learn to do
that by practicing in groups, as
often as not on stage during public
The performance environment is
where musicians develop their ensemble skills, and it's the one in
which they are accustomed to play ing at their best. That environment
can be fairly described as too many
players cramped into too little
space at one end of a large, very
noisy room. Since that's where your
clients learn to play at professional
levels, a good case could be made for
duplicating it in a studio. Make 'em
feel at home. It won't work. For one
thing, putting almost anything
close to a wall screws up isolation,
and for another, audiences are
pretty much out unless you're faking a remote. Lots of fun, by the
way, but only when it's intentional.
Just because you can't have it all
doesn't mean you can't have some of
it, though, and the basic bandstand
setup works very well in a studio.
This brings us to the fundamental
purpose of setting up for a session,
which is not, repeat not, to make
the engineer's job easy, nor to correct for rotten studio acoustics, nor
even to arrange things for pretty
Happy and comfortable in this
context defines a situation in which
musicians can play as well as possible, and know it. Given that, they'll
likely perform to the max. They'll
sound terrific, and with a little luck
so will you. If not, they probably
won't, which means you can't, no
matter what you do in the control
room. Aband that goes into a studio
knowing it's good but comes out
sounding like a bunch of amateurs
is not a great ad for the business.
Worse, you can't defend yourself.
It's a fact that nobody can tell the
difference between rotten playing
and a rotten mix. Nobody includes
the mixer, who has repeatedly got
egg on face by blaming the band for
miserable sound before getting up
off his dead butt and listening in
the studio.
Since we all know that the engineer is automatically to blame for
everything that goes wrong in a
studio, and musikers are not famous for admitting to crummy
playing, it's in your interest to arrange things so musicians can play
to the best of their abilities.
Besides, it's your job.
Engineers are hired to get good
sound. If you do, you're a hero. If
you don't you're a bum, and nobody
much cares how you did it either
It's commonly said that eighty
percent of a mixer's work is the
setup, and it's true, but setup is not
just a matter ofgetting appropriate
isolation or using the best possible
Great sound comes from well-recorded great performances, and the
studio setup is critical to a group's
ability to perform, as opposed to
merely playing some notes in the
same place at the same time.
Alast argument for good setups is
that they are competitive. A mixer
who consistently makes poor setups will turn out consistently
shabby sound from both first rate
and not so great bands, and one
who uses competent setups will
drive that turkey into the ground
like a tent peg.
And so, finally, on to the what,
why, and how.
In the beginning, there were
There still are, and we start with
them. Mic'ing drums is its own subject, but since that's been covered in
a previous article (see db Magazine, November/December 1990),
we'll pass on except to put the
drums in the middle of the room
with a low three -sided absorptive
screen around the back and sides of
the kit. Proper screen height is just
over three feet.
The screen is not there to kill the
drum sound in the room. Nothing
but a closed room will do that, and a
booth puts the drummer out of contact with the rest of the musicians.
There's always headphones, but if
they were any good, we wouldn't
spend multiple thousands of dollars building acoustically- correct
control rooms for speakers. Besides, when's the last time you saw
people wearing headphones on
Headphones are occasionally
useful, but they're nowhere as good
as speakers, and they don't even
compare with live sound for cueing.
Avoid them. If you need cueing
leave the studio speakers on (which
works better than you might
think), or do what the experts do:
use stage monitors.
Since a decent setup mostly
eliminates the need for artificial
cueing anyway, back to the drums.
The drum screen is used partly to
give the drum mics something absorptive to work into and partly to
prevent the drummer's hearing
three sets of reflections from the
kit. With the screen in place there's
only one, from the control room wall
and glass. He (she ?) can play the
other way, but the screen makes
things a little more comfortable.
Drummers like it.
With the drums centered and
screened, the piano goes beside
them with the hinge side toward
the kit. Acouple of ribbon mics work
best on piano, but if you don't have
them, use what you've got. Don't
use more than two mics; you'll
never get them in phase. Positions
are shown in plan on Figure 1. The
high end mic is usually 18 inches off
the strings, and the low end mic is
at about six.
If you don't have ribbons, or the
studio acoustics are a problem, a six
by eight foot screen will get you respectable isolation on the piano.
Bass and guitar are usually electric, in which case the amps can be
placed at the front edges of the
drum screen with their sides toward the kit. Put the amps up on
chairs or the equivalent. The musicians can hear them better that
way, and it improves isolation quite
a lot.
Absorbtive flats and mics in a typical small setup.
Electric instruments are typically picked up with direct boxes,
but ifmics are needed, the speakers
can be close mic'd at their edges, or
the amps can be put in three -sided
4 x 4 absorber screens similar to the
drum unit to get some isolation
with the mics a few feet off.
Given acoustic instruments, the 4
X 4 absorbers afford surprisingly
good isolation with the instru-
ments in about the same positions
while preserving a tight setup.
Don't forget to give the bass a resonator. A four foot square of wooden
flooring (even plywood will do) on 2
X 4s will amplify and improve the
bass sound a great deal. Basses and
celli are made to work on wood
floors, and without the floor, you're
working with half the instrument.
Mics are dealer's choice here, but
if you've got a condenser on a bass,
look out for very low frequency energy. You may find the bass is badly
out of balance on a small speaker
because of excessive signal below
60 Hz. Some 700 Hz boost will bring
it up on the little speakers, and a 60
Hz cutoff will bring it down on the
big ones. The cutoff helps a lot on
poorly damped electric basses, by
the way, and they're getting real
common of late.
Drums are directional. Except for
cymbals, there is surprisingly little
sound back of a kit, and even that
can be killed by making the back of
the drum screen higher than the asides. Coming up to five feet will do cn
it. Looks funny, but it works well. 43
The quiet area back of the drum m
kit makes vocal placement obvious. F
Close to the piano eliminates some m
cueing problems and generally Ó
works out nicely.
When a Hammond organ is on the 6
menu, it can be placed in the other- m
wise useless spot just back of the o
drum screen. That keeps the key- 4°
boards close, and both can cue off
the vocalist's live sound.
Figure 2. Typical mic pickup patterns are shown in this layout.
A Hammond needs no screening,
but a vocal does. Three -piece six footers at the usual four -wide do
very well. Four inch casters bring
the final height up to about six -feetfive, and with the vocal mic pointed
down as in film practice, that's
enough. Down mic'ing vocals from
eye level has a bunch of other advantages anyway, so nothing's lost.
Vocal backups are best worked as
close to the lead as possible, especially for gospel and the like. Avery
useful trick for that situation uses
either a bi- directional ribbon or two
cardioid whatevers at right angles
to the lead mic, and about three feet
away. With a ribbon the lead -background isolation is absolute, and
it's only two steps from one mic to
the other. A ribbon mic is excellent
for gospel, where you have very
tight arrangements and inter changing lead vocalists. Depending
as always on studio acoustics, extra
screens may prove useful on either
side of the background mic(s).
Horn bands are a special problem, mostly because when brass instruments play loud they tend to go
sharp, but the reeds go flat. That
doesn't create a problem for big
bands as the sections are separated
enough so they don't mix, but with
a only a few instruments you get
studio intermodulation distortion,
and it sounds terrible.
That's not to say horn bands can't
play in tune. They can and do, but
they need to hear each other better
than most, which makes stringing
a short section out in a line a really
bad idea. The obvious solution is to
seat them facing each other at
ninety degrees to the drum kit, and
the logical pickup is bi- directional
mics in the middle. You can run into
ego problems here as a good many
section players tend to think of
themselves as soloists in concert
rather than members of a group.
Actually musicians are both, but
they sometimes feel very strongly
about being individually mic'd. If
that happens, use individual mics
with the same seating pattern. It's
not optimal, but when clients insist
on telling you how to do your job, do
it their way and let 'em find out why
you like yours.
The overall result of all the above
is a very tight setup in which no instrument is playing into another's
mic, the players can hear each
other without cue systems, and in
which they can pass a note without
standing up. It's about as close as
you can get to the normal performance environment. The communications are excellent, the feeling of
"groupness is strong, and when the
band hits a tutti, you can hear the
room modes WHOMP! in response.
That's the sound of power. Very satisfying stuff.
As to performance, you know
you've got a session when the musicians remark that they're having a
good time, and the setup system detailed here is designed to produce
that feeling. Additionally, it's dead
reliable. It works every time without anything much in the way ofadjustments after the session starts.
Best of all, it's probably somewhat familiar. A great many live
session engineers set up along
these general lines as we all face
the same problems and so find similar solutions.
put it another way, this setup
is neither a revelation nor a revolution. It is, however, a well -integrated system that has been
worked out and used over a course
of nearly thirty years of live session
work without ever having needed
to be revised on session.
There are a bunch of ideas here. If
you see one you like, try it. It works
for me, and it might work for you. ar
Why Is Church Audio So Poor?
Many years ago, I set out on a
mission to analyze why church
audio as a whole sounded poor, and
how to improve its overall quality.
My findings were not all positive;
many of the problems were or are
not going to be easily solved, but
they can be improved by education.
I have found that poor sound
quality is mainly due to a lack of
education, either by the operator or
the person who installed the sound
system. Church sound systems are
complicated because they require
two different approaches in design -one for speech and one for
music. If a system is designed for
speech, the music minister may be
unhappy and want to replace it
with a system designed for music.
However, a music system may
make speech difficult to understand or cause extra feedback prob-
Audio is not as mysterious as people may lead you to believe. People
who act as if audio is a mysterious
art form usually don't understand
the principles themselves. In Don
& Carolyn Davis' book Sound System Engineering, they describe the
differences between art and science
in working with sound systems in
stating, `The science is in the step by -step, logical procedure that one
follows when faced with a completely unknown problem. The art
is in recognizing the cause of the
problem when it is first discovered.
This comes with experience. When
art fails, science takes over." Audio
is a science that deals with acoustics, electronics and electro- acoustics (the transformation from
acoustic to electrical and back). A
popular belief is that if you know
electronics, you can be a great
audio engineer. Knowledge of electronics is a great start, but you will
have problems with making microphones work in the same room with
a speaker, or making speech intelli-
gible in a very reverberant room, if
you are not familiar with audio.
'lb improve the performance of
church sound systems, you have to
increase your knowledge of acoustics, electronics and electro- acoustics. There are several committees
that set standards and provide
training for the sound and communication industry, such as the Audio
Engineering Society (AES), the Institute of Electrical and Electronic
Engineers (IEEE), and the National Sound and Communications
Association (NSCA), but these
standards are more or less unwritten (a true standard is one specified
by a group or committee that governs the industry). However,
churches have specialized needs as
well, and need a group to set up and
provide training and standards for
technical ministries. Such a group
would provide training and certification for different levels of knowledge and experience. This group as
a whole would improve relations
between manufacturers and con-
tractors, thereby meeting the
needs of church technical people.
For this segment, I have discussed testing and certification
with many people including manufacturers, church members, contractors and audio consultants.
From these conversations came the
following test. This test is for you to
see how many questions you can
answer correctly, and to see which
areas of your knowledge need improvement. The other reason for
the test is to show you a sample
exam that would be given for certification.
There should be multiple levels of
certification, such as Operator,
Senior Operator, Technician, Senior Technician and Master Technician. Part of the certification would
require a specified time period of
experience, preferably under someone with certification one level
higher than the level trying to be
obtained. lb get certification of Operator, for example, you would have
to complete course material, have
so many hours of hands -on operation under a Senior Operator, and
complete a test before certification
is awarded.
The following would be task -required for each level of certification.
Equipment set-up and tear down
Working knowledge of mixing
console and processing equipment
Mixing procedures
A determined amount of operat-
ing experience
Satisfactory test score
Senior Operator
A determined amount of operating experience
Working knowledge of all audio
Basic electronics, i.e. DC /AC
Basic understanding of electroacoustics
Satisfactory test score
Complete all requirements for
Operator and Senior Operator
Detailed knowledge of audio systems and their interconnection
Working knowledge of electroacoustics
Basic understanding of acoustics
Good trouble -shooting skills
Working knowledge of basic
audio measurement (RTA)
Satisfactory test score
Senior Technician
Complete all of the above
A determined time of experience
as a technician
Detailed knowledge of electroacoustics
Advanced level of system and
component trouble- shooting
Working understanding of acoustics
Experience using RTA
Working knowledge of RT60 and
advance measurement
Satisfactory test score
Master Technician
Complete all of the above
Detailed knowledge of acoustics,
electro-acoustics and electronics
with an understanding of psychoacoustics
Be able to design and write specifications of a working sound system
Detailed knowledge of the five parameters of sound system design
Experience in advanced audio
Experience with an Audio CAD
Satisfactory test score
Please take the following test,
ponder the concepts, and write in
care of db Magazine-we would
like to hear your comments.
1. What is the purpose of an equalizer?
2. What is the procedure for setting up
the mixer's gain structure?
3. Every time you double the amount
of microphones that are turned on, you
lose /gain
decibels before it
feeds back.
4. Every time you double the power,
for example, when you go from 100
watts to 200 watts, you have a level
change of
5. How many decibels does it take to
perceive double the volume?
6. If you have a mic line and a quarter
inch line from a keyboard, which will
have a stronger level?
7. What device would you use to
match a line, such as from a keyboard,
to a mic level?
8. What is the difference between a
balanced line and an unbalanced line?
9. What is the ratio for the proper spacing between mics?
10. When wiring a mic, the hot pin is always on pin number?
11. The shield is on what pin of an
12. An echo send or Effect auxiliary is
usually pre or post fader?
13. A monitor send or Monitor aux is
usually pre- or post -fader?
14. On a mixing console, the high and
the low controls are what type of EQ filter?
15.On a mixing console, the mids use
what type of EQ filter?
16. An Omni -type mic has a 180 degree pick up pattern. (T or F)
17. A cardioid mic pattern is a good
mic to use in a high decibel environment because
18. Which type of mic has more gain
before feedback?
19.1f have a level of 100 dB with one
mic on, what will my level be with eight
mics on?
20. What is the international standard
on the use of polarity on XLR connectors?
21. What is a good way to locate the
primary source of hum in a system?
22. When installing a reverb unit in a
church sound system, should it be
placed at the input of the main power
amp or in a loop of the mixer?
23. What is constant Q in equalizers,
and what is its advantage?
24. Where does hum come from?
25. What is the result when two channels are wired in opposite polarity?
26. Define RT60.
27. What is a preferred listening
28. What's the difference between a
speaker's sensitivity and its efficiency?
29. Which produces less handling
noise -an omni or cardioid mic, and
30. What is RFI, and what is one way
to prevent it?
31. What is input overload, and what
device is used to prevent it?
32. Do constant -voltage amplifiers put
out a constant voltage, and under what
circumstances can a direct -coupled
amplifier drive a 70 volt line?
33. What is critical distance and how
many values for it can exist in a room
with a multiple -horn speaker cluster?
34. What is the standard rule-ofthumb for providing adequate damping
to a woofer as it relates to cabling?
35. What advantage does bi- amplification have over passive crossover networks?
36. What is the primary reason for using High Q devices in a speaker cluster?
37. When people in the first third of the
audience from the stage complain that
they can't understand what is being
said, this is usually caused by what?
Select one of the following:
Speaker dispersion
Crowd noise
38. In a fan -tailed hall, the reverberation tends to congregate in what part of
the hall?
The front
The back
The middle
Evenly distributed
39. In a "shoe box" hall with raised
seating, the reverberation tends to congregate in what part of the hall?
The front
The back
The middle
Evenly distributed
40. The condition of scattered sound
is known as?
41. The condition that describes the
bouncing of sound is known as?
42. The scattering of sound is known
43. The attenuation of sound (in the
acoustic environment) is known as?
44. The bending of sound is known
45. The bending of sound by density
variations in the conducting is known
46. Room modes are known as?
47. Carpet on the floor of a hall
the RT60 of the
but not for the
Choose from:
48. Define Echo.
49. If a small room (1000 cu. ft.) has a
RT60 of one second, and a large room
(10,000 cu. ft.) has the same floor, wall
and ceiling materials, the RT60 will be
Choose from:
10 seconds
5 seconds
3 seconds
50. The proper way to set up a mix is to
turn your gain all the way up with the
channel slider on 0 dB (or at unity gain),
and then slide your masters up just before you get feedback. (T or F)
51. Phase and polarity mean the
same thing. (T or F)
52. When using a compressor, usually
an ideal compression ratio is 2:1 for vocal work. (T or F)
53. What is a corn pander?
54. A noise gate is used in what situ-
55. What are the five parameters for a
good sound system design?
56. What is noise reduction, and how
does it work?
(A very general answer is fine.)
57. Should you use noise reduction
with desolvertones or timing codes?
58. What does the abbreviation MIDI
59. What is MIDI?
thank some of the industry's leaders for taking time out
of their busy schedules to contribute to the compilation of this test:
I would like to
Murray, Pro Sound
manager of TOA Electronics
Art Noxon, president of Acoustic
Sciences Corporation
Monty Ross, engineer, Rane
Larry King
shall, King
of Klepper, Mar-
Using Drums in the Church
Psalm 150:4,6-Praise Him with the drums...Praise Him with the cymbals, yes, loud resounding cymbals.
Church Drums
gospel and contemporary
Christian music has not
only brought change to the
entertainment and music industries, but to the church as well. This
change has introduced new rhythmic and upbeat music into the
church, thus emphasizing the use
of drums.
The fear, though, of many music
directors and pastors is that the addition of drums will not only raise
the overall volume ofthe music, but
overpower the other instruments.
This fear can easily be dispelled
by using the proper techniques of
tuning, baffling and mic'ing drums.
By observing the following guidelines, the drums can become a dy-
namic addition to the church service.
Imagine using a $1,000 mic on a
$100 guitar, or a cheap set ofstrings
on a Stradivarius violin. The results would obviously be disappointing. What might not be obvious, though, is that poor sounding
drums will not be helped by expensive mics or an elaborate sound system. The equation still holds true
garbage in, garbage out. So the
place to begin is to properly tune
the drums. The best person for the
job is a drummer, but if he or she is
inexperienced in tuning, you would
want to follow these basic guidelines.
The kick drum pillow and mic in position.
Each drum should be tuned to a
specific pitch, similar to a timpani.
Personal taste will dictate what
pitches will be used. Begin with the
tom -toms. If there are three, pitch
them low, medium and high, and if
just two, low and high. With drum
key in hand, begin by tightening
the first lug so that the drum head
reaches the desired pitch. Continue
diagonally across the head and
tighten the second lug so that the
head matches the pitch of the first.
Proceed in the same manner until
all eight lugs are tightened and the
head is completely tuned. If a bottom head is being used on the same
drum, tune it similarly only at a
slightly lower pitch. The kick and
snare drum should be tuned in the
same manner. It is common practice to remove or cut a hole in the
front head of the kick drum and insert a pillow or rug inside (see Figure 1). This is done to reduce the
resonance of the drum and produce
a deep punchy sound. It also facilitates an opening for a mic.
If after tuning the drum still
sounds flat or dead, it is possible
the head may need changing.
Check for dents. The more there
are, the less the head will resonate.
Replace with either a plastic head
or an oil -filled head. Another
method of tuning that I have used
with much success is match tuning.
Match tuning is using a sampled
drum from a drum machine as a
standard and tuning the real
drums to it. There are many inexpensive drum machines on the
market (under $300) that produce
excellent sampled drum sound
with incredible authenticity. Two w
people, each equipped with a wireless headset, are needed to tune the
drums in this way. First, mic the
real drums and connect the drum
machine to your sound system.
Have one person in the audience
monitoring the sound while the
other person strikes the real drum
by the corresponding
drum sample. Simply tune the
drum by matching it to the electronic sampled drum. Mic placement will play a major role in the final sound and can only be arrived
at through trial and error. Let your
ears be the final judge.
After the drums are tuned, the
next things to check for are any rattles, vibrations and ringing. This
may sound trivial, but remember
that a small squeak won't sound
small after it's amplified through a
sound system. Play each drum individually and check for any ringing
from other drums. Inevitably, the
snare drum will cause you the most
frustrations. When any drum in the
set is hit loud enough, especially
the bass drum, the snare will vibrate. To minimize this, simply
tighten the snare. If it persists, a
small piece of duct tape placed at
the end of the snare wire should
dampen the rattle.
Probably no other
instrument has the
combination of
mic'ing possibilities
and techniques as do
the drums.
Now that the drum set is tuned
up and free of rattles, rings and vibrations, the next question is
whether you need to decrease the
overall loudness of the drum set by
using baffles. The answer lies in the
acoustical properties of the church.
A church whose interior is constructed of mainly hard, nonporous
surfaces (especially in the area
where the drums are located) will
require baffling. Baffles (or gobos)
are walls or partitions constructed
of soft, porous material which absorb sound. In the church setting
described above, the drum set
should be enclosed by baffles (see
Figure 2). To maintain the drummer's vision, glass or plexi -glass
continues from the top of the baffle
to the top of the highest cymbal.
If the interior of your church is
filled with padded pews, carpeted
floors and other porous materials,
baffling may not be necessary. Most
churches, however, fall into a middle category, a combination of both
hard reflective surfaces and soft absorbent surfaces. One such church
is the Christian Victory Center in
Hempstead, New York. The church
meets in a theater where the playing area is a mixture of hard and
soft surfaces. A spacious stage
made of wood is surrounded by tall
brick walls and rows of thick, heavy
curtains. The combined surfaces
make necessary only a partial baffling of the drums (see Figure 3).
The baffle is 31/2 -feet high and
made of plexi -glass and masonite;
both highly reflective surfaces. The
direct sound of the drums gets reflected back by these materials into
the highly absorbent curtains
which are directly behind and
above the drums. The shorter baffle
allows for a more `live' feel between
the drummer and the rest of the
musicians. Previously, the drums
were totally enclosed. The side and
rear baffles were made ofhighly absorbent material. The front wall
was plexi -glass and the ceiling was
The advantages were extremely
low sound levels on stage and total
control over the volume through
the sound system. This type of baf-
fling produced a true studio drum
sound. However, the disadvantages were considerable. First, the
drummer was acoustically isolated
from the rest of the band. Aregular
floor monitor or hot spot was prohibitive because it would feed back
into the mics, requiring a special
headphone system. Ventilation
was also a problem. Fan noise
would inevitably be picked up by
the five mics inside the booth. Finally, this partitioned box was not
aesthetically pleasing to the eye.
The present baffling set up, therefore, was constructed to utilize the
best of both worlds. Remember, the
Figure 2. In a church setting, baffle- enclose the drums.
When using duct tape on any of
the drum heads or cymbals, remember that too much tape will
eliminate overtones, causing the
drums to sound dead and/or the
cymbals to lose their brightness.
Gauze pads or tissue paper can also
be used to reduce vibration and
ringing. Simply fold a piece of tissue paper to the size of a matchbook. Move this pad to the location
on the drum head where the unwanted noise is occurring, then secure it with duct tape. The ringing
in cymbals can be reduced with
masking or duct tape applied on the
underside of the cymbal in radial
on tuning, the front head should be
removed along with enough dampening material to reduce any unwanted resonance. The mic can
either be placed on top of a pillow
resting against the head at a ninety
degree angle or on a mic stand
Figure 3. A partial plexi -glass baffle.
main question when designing
your own baffling system is "What
are the acoustical properties of my
church ?"
Probably no other instrument
has the combination of mic'ing possibilities and techniques as do the
drums. Each technique is based on
either the style of music, acoustical
environment or desired sound. To
achieve a contemporary drum
sound however, the most common
way of mic'ing is the close mic'ing
technique. Starting with the snare
drum, most audio engineers prefer
the sound of either a moving coil or
capacitor mic for snare. The Shure
SM57 or Sennheiser 421 both reproduce the sound of the snare very
well. Place the mic at a thirty to
forty degree angle, horizontal to the
drum head and about one inch in
from the rim and one inch above the
head. Make minor moves from that
point to get the sound you like. If
you have the luxury of using two
mics, place one under the snare in a
similar fashion as above. This mic
will pick up more of the snare's
There are two common methods
for mic'ing the high -hat. The first is
mic'ing it separately using a condenser mic. A good choice is the
AKG 451. Place the mic approximately three inches above the high-
hat thirty degrees down from horizontal, pointing away from the
snare and toward the drummer's
left elbow.
If your church doesn't
own much
equipment, or your
mixing console has
limited inputs, I
would suggest using
just three mics -one
for the bass drum and
two overhead.
The second method of mic'ing the
high -hat is using one mic for both it
and the snare drum. The mic of
choice would then either be a Shure
SM57 or 58. Place the mic between
the snare and high -hat and adjust
according to the desired sound. Be
careful not to place the mic near the
edge of the high -hat; the two cymbals coming together produce a
rush of air which will be picked up
by the mic.
The kick or bass drum is the foundation of the drum set and along
with the bass guitar, supports the
rest of the musical elements. It is
therefore important to mic it properly. As stated earlier in the section
pointing directly at the head.
Either of these methods will produce a fuller sound with maximum
bass response. If the mic is pointed
slightly away from the head toward
the side of the drum, more of its
overtones will be emphasized.
Since the bass drum produces high
levels of sound pressure, a moving
coil or dynamic mic should be used.
Some good choices are the Shure
SM57, the Neumann U67 and the
Sennheiser 421.
Tom -toms can be mic'd one of two
ways, like the snare from above or
from underneath. Place the mic at a
thirty degree angle and about two
to three inches from the drum head.
Mic'ing it any closer than two
inches will give the drum a nasal
quality. When mic'ing from underneath, remove the bottom head.
This mic'ing technique will produce
a flatter and more percussive
sound. My favorite mic for toms is
the Shure SM58. Some other good
choices are the Sennheiser 421 or
the AKG 414.
Finally, to cover the cymbals, two
overhead mics are used. Place
them about 15 in. above the cymbals and three to four feet apart.
Any high quality condenser mic
should be used because it gives the
cymbals an open, airy sound. My
choice is either the Shure SM81 or
the AKG C 1000S.
If your church doesn't own much
equipment, or your mixing console
has limited inputs, I would suggest
using just three mics -one for the
bass drum and two overhead. The
only adjustment would be to lower
the overhead mics in order to pick
up the toms, snare and high -hat.
In closing, remember that the
presence of a drum set in some
churches is still taboo. Even if your
drummer played with feathers, it's
still a psychological barrier to some
My suggestion would be to purchase a decent quality drum machine and start using it in services.
After the faithful become used to
hearing it, the transition to the real
a: w
thing won't be as traumatic.
3 -D
Audio:Wave of the
Many people connected with the audio industry could not help but hear the recent clamor about the new
three -dimensional audio systems on the market.
These systems claim that
while using only two speakers, they can locate sound
far out beyond the edges of
the speakers and even place
sounds behind the listener's head
in true 3 -D sound. While these
claims are extremely easy to make,
delivering on them is not so easily
accomplished. This article provides
a simplified explanation of how the
brain perceives sounds in three dimensions, and then takes a quick
look at products from several
manufacturers which claim to provide 3 -D or enhanced experience in
sound. Please note6 that the information provided here is gleaned
from the results of many long hours
of research done by various manu-
facturers, universities and private
audio research labs over long periods of time and some of this research has included very complex
mathematical models of sound and
hearing done on large mainframe
computers. A tip of the hat to researchers at Hughes Aircraft,
Archer Communications, Pete
Meyers Productions, Roland Corporation, Bedini Audio, and the
many others who continue to contribute to this fascinatingand everexpanding body of knowledge.
The human brain is remarkable
in its ability to perceive sonic directionality. There are three basic elements in sound perceived by the
M ears and decoded by the brain to in-
terpret exactly where a sound is
coming from. These three elements
are: a) differences in amplitude, b)
difference in phase or arrival time
and c) differences in certain frequencies induced by the shape of
the outer ear. It is the combination
of these three elements which allow
humans to pinpoint the location of
a sound in space with uncanny accuracy.
The record buying public's
fascination with the idea of
3 -D sound was evidenced by
its embracing the ill-fated
and short -lived concept of
The first clue, difference in amplitude, is caused by the fact that the
head creates a sonic shadow to
sound waves. Consequently, a
sound coming from more than several degrees off axis from the nose
will be perceived as louder by the
ear on the side of the arriving
sound. This loudness difference
will help tell the brain the general
direction of the sound, whether off
to the right, left, or directly in front
or back. It should be noted here that
until the advent of these "3 -D" processes in recording and mixdown,
this was the primary method used
to create a stereo spectrum in a conventional stereo recording via the
pan control on the recording console during mixdown.
Make the sound louder in the left
speaker, and the brain will perceive
the sound as coming from that general direction. However, the lack of
extremely clear imaging and directionality in conventional stereo
demonstrates that this method
alone is not sufficient to provide the
necessary information to the brain
for accurate localization. It is this
very deficiency in standard stereo
which has led to the discovery and
exploitation of the other localization clues.
The second clue, difference in
phase and arrival time, is caused by
the fact that a sound arriving off
axis will arrive at one ear first and
then arrive at the other ear several
milliseconds later, thus creating an
out of phase condition with the
sound the first ear hears due to the
difference in arrival time. The
brain takes this phase information
and decodes it, adds it to any amplitude difference present, and is able
to localize sound quite well with
just these two clues. An analogy
might be an artist drawing a 3 -D
object on two-dimensional paper
using perspective and shading. A
good rendering can provide a great
deal of depth and detail, but is still
not a complete 3-D picture. Difference in phase is used in a majority
of 3 -D audio systems and does manage to provide fairly accurate localization clues, but to get the whole
picture, there is a third aural clue
which completes the aural image
and transforms this 2-D "sketch"
into full 3 -D sound.
The third aural clue, change in
frequency, has to do with the actual
shape of the pinna or outer ear. Depending on from which direction a
sound comes, the shape ofthe outer
ear actually attenuates certain frequencies, mostly in the upper register, while allowing other frequencies to pass unchanged. When
combined with the other two localization clues, it is this slight aural
shaping of the harmonic spectrum
which provides the brain with the
third and final clue as to the location of a sound in space. This creates for the listener a sense of depth
and space and directionality so accurate, that people with closed eyes
are able to point directly at the
point of origin of a sound with great
The record buying public's fascination with the idea of 3 -D sound
was evidenced by its embracing the
ill -fated and short -lived concept of
quad. Four-channel stereo's promises of "being there" and "true 3 -D
sound" in theory were wonderful,
but their implementation proved
impractical, expensive and the listener's enjoyment of the material
was extremely dependent on his location within the sound field.
Undaunted by the catastrophic
failure of quad, several companies
began research programs to study
just how the ears and brain processed sound and how this might be
recreated by two speakers. Although albums and soundtracks
using a two speaker 3-D process are
just now entering the market, the
research on the subject began back
in the early eighties. Pioneers like
Pete Meyers, John Bedini and even
Hughes Aircraft foresaw a time
when 3 -D audio from two speakers
would become a reality. They studied the process, learned how it
works and began to devise ways to
implement the process in recorded
It is well and good to understand
how the ears and brain work together to provide localization in
real life situations. It is, however,
another matter entirely to some-
how recreate and/or process sounds
captured by a recording medium,
EQed, mixed, sliced, diced and
chopped, played back through two
speakers, and yet still provide the
listener with enough of the aural
clues to locate the recorded sounds
in 3 -D space.
As it has turned out, the leaders
in the field went down two different
pathways. One path led to the development of methods to expand
the stereo image beyond the edges
of the speakers and provide a much
tighter stereo image with regard to
placement of individual instru-
ments within the stereo spectrum.
The other pathway led to the creation of a true, 3 -D audio experience
which would allow the mixing engineer to place sounds far to the listener's right and left as well as
above, below and even behind the
listener's head. The companies
manufacturing these devices are
split neatly into two categories
based upon whether they try to "expand" the sound or to "three- dimensionalize" it.
One would hope that the
advent of 3 -D audio does not
turn out to be just another
"emperor's new clothes'
phenomenon, but will turn
out to be a truly viable
technology which will
enhance the listener's
enjoyment of the music.
The first category is the "sound
widening" group. These companies
market devices which enhance the
width of the stereo field and
sharpen stereo imaging. Representative of this category are products such as B.AS.E. from Gamma
Electronics and Sound Retrieval
System from Hughes Aircraft.
These manufacturers do not claim
behind the head 3 -D sound, but
only to restore a greatly enhanced
sense of ambience, space and imaging to any musical source, live or
prerecorded. These devices may be
used as playback devices or during
tracking to capture the effect directly to a multi -track master.
The second category is the "3 -D"
group which claims to provide
sound above, below and even behind the listener. Examples of this
group are Roland Sound Space
from Roland Corporation and the
much hyped Q-Sound from Archer
Communications. While the Roland product seems to be slowly
gaining favor from various studios
and producers, Q -Sound burst onto
the scene with a roar. Beginning
with a super -hyped Super Bowl
commercial in 1990 (which was reportedly underwhelming), to the
recent release of albums by Sting
and Madonna using the Q -Sound
process, Archer Communications
has by far gained the most public
In fulfilling its commitment to
keep its readers informed, db
Magazine will in future articles
report on the progress of this new
technology, now in its infancy. It
must be noted, however, that not all
the attention and press these 3 -D
products have received has been
positive. There are complaints
about the so- called "sweet spot,"
which is a very small location between the speakers where the 3 -D
effect may be heard, concern about
mixes sounding overly bright, and
the British Broadcasting Company
reportedly is considering banning
all 3 -D encoded material from
broadcast because ofproblems with
mono compatibility. Some of these
complaints are troublingly reminiscent of statements detractors
made about quad during its brief
rise and fall. One would hope that
the advent of 3 -D audio does not
turn out to be just another "emperor's new clothes" phenomenon,
but will turn out to be a truly viable
technology which will enhance the
listener's enjoyment of the music.
After all, isn't that the reason for
doing all this in the first place?
Compact discs and audiophile LPs
sound pretty good just as they are
and while it would be nice to hear
the cellos behind the right ear and
the lead guitar behind the left ear,
consumers can hope that manufacturers will never lose sight of that
old and time -honored saying, "If it
ain't broke, don't fix it!"
A Theater Sound System
The MUNY (Municipal Theatre Association of St. Louis, MO), a twelve -thousand -seat outdoor amphitheater built in 1917, provides its patrons with grand theatrical productions in a natural outdoor setting.
not-for -profit organization
in 1919, The MUNY has
been producing large -scale
Broadway style shows, operettas
and ballets in this outdoor musical
theater which is the largest in the
country (see Figure 1). Over forty four million people have enjoyed
these incredible productions. The
stage itself includes a forty -eight
foot revolving platform which allows elaborate scene changes in a
little over one minute!
Equally impressive is the sound
system for this summer's series of
shows. The Riverfront 7£mes said
about the first show of the season
(It's Delightful, It's Delovely, It's
Cole Porter), "Kaye Ballard is a delightful storyteller when Porter
supplies the words. And you could
understand those words, thanks to
Figure 1. The main
entrance to the
MUNI' amphitheater.
the best sound I've
heard at the MUNY."
To put together the
sound system for this
MUNY called upon the
Otts Munderloh, a
year, The
services of
sound designer whose experience
Figure 2. The speaker system is Apogee 3X3-Ils flown from cable above
the audience.
includes Dreamgirls, Jerome Robbins' Broadway, Sophisticated Ladies, Little Shop of Horrors and
many other productions. "Four
Apogee 3X3 -IIs are flown thirtyfive feet high on a cable that runs
the width of the facility over the
stage apron in a distributed system
approach to cover two -thirds of the
seating area," says Munderloh (see
Figure 2).
"There is a single 3X3 -II mounted
on the pylons that supports the cable for additional coverage at the
extreme sides of the seating area.
We're also using three AE -5s on
either side of the stage on the pro scenium arch to cover the first third
of the seating area in addition to
the six AE -2s used for front audience fill (see Figure 3). There is a
single AE -3 at the edge of the seating area on either side for extra fill
Barry Luz is Manager of Marketing and Technical Training at
Apogee Sound in Petaluma, California.
mounted on the pylons that support the steel cable where the 3X3 IIs are flown. The sound quality of
the system is exceptional, "he said.
The schedule for the actors and
production staff is grueling. This
season there is one show each week
with performances every day and
seven shows in all. Ashow will open
on Monday evening while the next
show is being rehearsed throughout the week. Saturday mornings,
from midnight to 6 a.m., runthroughs are done for the next
week's show to focus lights. At 6
a.m. everyone goes home for a short
rest with the rehearsal on Sunday
at 3 p.m. to hear the cast sing with
the orchestra in a rehearsal space.
Then, at 7 p.m. Sunday, the last
performance of the previous show
is held. The sound crew comes in
Monday at 10 a.m. and final set
changes for the next show are completed. The dress rehearsal of the
new show runs from 1 p.m. to 6 p.m.
with the cast and orchestra on the
main stage. The new show opens 8
p.m. Monday.
There are several
mixing consoles in
use, including a
Yamaha PM -3000
with forty inputs
from which
Munderloh mixes the
show and a Yamaha
PM-2000 submixed
into the PM -3000 for
more orchestra
Figure 3. This view is from the stage and shows the expansive outdoor
stage and turn on the appropriate
microphone. Now, in the nineties,
audiences expect every person in
the cast to be heard as if you were
standing next to them. With twenty
wireless units, the rehearsal time
constraints are very difficult. This
year we're using the Sennheiser
wireless UHF units with the computer- controlled remote system.
It's been working very well for us
and because it's an outdoor venue,
anything that isn't on a microphone
can't be heard. I'm also using pit
singers on mics. We're also using a
large orchestra by Broadway
spent finding out that the wireless
mic is hidden under the costume or
the transmitter antennae is bent
instead of the `luxury' of working on
sound cues, equalization and sound
levels," said Munderloh.
"In the old days, when you had
five -foot mics, a six hour rehearsal
was enough to write down where
the actors were positioned on the
two people."
There are several mixing consoles in use, including a Yamaha
PM-3000 with forty inputs from
which Munderloh mixes the show,
and a Yamaha PM -2000 submixed
into the PM: -3000 for more orchestra inputs (see Figure 4). A Soundcraft 200B is also used to submix
six reed instruments with its output being sent to the PM -3000.
Some of the signal processing
equipment includes a Lexicon
Figure 4. Apogee AE -2s mounted along the front edge of the orchestra
pit for front fill.
"Most of this rehearsal time is
standards, approximately thirty -
.. .
Figure 5. The mix position and its
view of The MUNY stage.
PCM-70 delay, and a SPX-1000
digital reverb and several dbx 900
mainframes with compressors/limiters for all wireless units on insert
points of the console.
Another integral part of the
sound system is the use of Apogee
Sound's proprietary equalization
technique CORREQT, (Computer
EQualization Technique). This
technique involves the use of very
narrow -band frequency analysis
and equalization as Ken DeLoria,
president of Apogee Sound and developer of the CORREQT technique, explains. "Many people will
tune a sound system using very sophisticated test instruments without ever really listening to how live
speech and music sounds through
the system. The goal of the sound
system is to convey whatever signal
source it is given in the clearest,
cleanest, most accurate fashion. To
do that consistently, the CORREQT engineer has to work with
the venue's acoustics. His task is to
minimize the detrimental acoustical anomalies of the facility. Listening to how speech and music
sounds through the system, in addition to extensive instrumentation -based analysis, are extremely
important and form the basis of our
technique," DeLoria said.
All of this technology
and effort is put forth
to achieve one goal: to
provide to the
patrons of The MUNY
the best show
One of the advanced features of
CORREQT is the ability to use music and speech, as well as test signals, for the source which stimulates the system. This allows
real -time equalization during the
show with the audience present.
After DeLoria's initial equalization and setup of the system, a second EQ specialist, Alexander YuillThorton II (Thorny), was brought
in to attend the first live show and
further refine the system's equalization. This served to accurately
"fit" the EQ curves to the venue's
acoustical characteristics with a
live audience in place.
All of this technology and effort is
put forth to achieve one goal: to provide to the patrons of The MUNY
the best show possible. If the reviews from the St. Louis Post Dispatch are any indication, the following reviewer's statement holds
`The MUNY's new sound system
is well -nigh perfect. I spent the second act prowling the vast seating
area and I heard every word without a trace of crackles, buzzes or
static (see Figure 5). The system
has excellent balance and both the
orchestra and the performers
sounded splendid, from my regular
seats down front all the way to the
top, only a couple of rows from the
free seats. I checked the center and
both sides, practically from top to
bottom, and never missed a thing?
American Heart
How To Use EQ
Even if you can record a performance accurately, you might not
like how it sounds. This is where
the equalization (EQ) process
comes in. EQ lets you improve on
reality: it makes an instrument
sound warmer or less harsh, removes noises and adds punch.
EQ adjusts the bass, treble and
midrange of a sound by turning up
or down certain frequency ranges.
7b do this, it operates on the spectrum of the sound source -its fundamental and harmonic frequencies. The spectrum helps give the
instrument its distinctive tone
quality or timbre.
If some of these frequencies
change in level, the tone quality
changes. An equalizer raises orlowers the level ofa particular range of
frequencies (a frequency band),
and so controls the tone quality.
That is, it alters the frequency response. For example, a boost (a
level increase) in the range centered at 10 kHz makes percussion
sound bright and crisp. A cut at the
same frequency dulls the sound.
often incorrectly called "parametric," which also allows control of
bandwidth. You won't find a true
parametric equalizer in home -studio equipment, however.
The parametric equalizer allows
continuous adjustment of frequency, boost or cut, and bandwidth -the range of frequencies affected. Figure 4 shows how a
parametric equalizer varies the
bandwidth of the boosted portion of
the spectrum.
Agraphic equalizer (see Figure 5)
is usually external to the mixing
console. This type has a row of slide
potentiometers dividing the audible spectrum into 5 to 31 bands.
Let's review the types of equalizers from simple to complex. The
most basic type is a bass and treble
control (often labeled LF EQ and
HF EQ). Its effect on frequency response is shown in Figure 1. Typically, this type of EQ provides up to
15 dB of boost or cut at 100 Hz (for
the low-frequency EQ knob) and at
10 kHz (for the high- frequency EQ
Bass and treble EQ.
Figure 2. Multiple frequency EQ.
6K B00ST
You have more control over tone
quality with a multiple -frequency
equalizer: you can boost or cut several frequency bands (see Figure 2).
Sweepable EQ is even more flexible; the exact frequency range
needing adjustment can be "tuned
in." (see Figure 3) Sweepable EQ is
When the controls are adjusted,
their positions graphically indicate
the resulting frequency response.
Usually, a graphic equalizer is used
for monitor- speaker EQ.
So far we've classified equalizers
according to the frequency bands
they control, but they can also be
classified by the shape of their frequency response. Apeaking equalizer (see Figure 6) creates a response in the shape of a hill or peak
when set for a boost. With a shelving equalizer, the shape of the frequency response resembles a shelf,
as in Figure 7.
Afilter causes a roll -off at the frequency extremes. It sharply rejects
(attenuates) frequencies above or
below a certain frequency. Figure 8
shows three types of filters: lowpass, highpass and bandpass.
For example, a 10 kHz lowpass
filter (high-cut filter) removes frequencies above 10 kHz. Its response is down 3 dB at 10 kHz and
more above that. This reduces hiss type noise without affecting tone
quality as much as a gradual treble
roll -off would. A 100 Hz highpass
filter (low -cut filter) attenuates frequencies below 100 Hz. Its response is down 3 dB at 100 Hz and
more below that. This removes low pitched noises such as room rumble, microphone handling noise and
mic breath pops. Finally, a 1 kHz
bandpass filter attenuates frequencies above and below a frequency band centered at 1 kHz.
The crossover filter in most monitor speakers consists of lowpass,
highpass and bandpass filters that
send the lows to the woofer, mids to
the midrange, and highs to the
Afilter is named for the steepness
of its roll -off: 6 dB per octave (first order), 12 dB/octave (second-or-
der), 18 dB /octave (3rd order) and
so on.
If your mixer has bass and treble
controls, their frequencies are preset (usually at 100 Hz and 10 kHz).
Set the EQ knob at 0 to have no effect ("flat" setting). Turn it clock-
Figure 3. Sweepable EQ.
Parametric EQ.
Before using EQ, try to get the desired tone quality by changing the
mic or its placement. This gives a
more natural effect than EQ.
Should you apply EQ during recording or mixdown? If you mix
more than one instrument to the
same track, you can't EQ them independently during mixdown unless their frequency ranges are far
apart. Suppose a recorded track
contains lead guitar and vocals. If
you add a midrange boost to the
guitar, you'll also hear it on the vocals. The only way around this is to
EQ the lead guitar independently
when it's recorded.
Hz, full at 100 Hz, honky at 600 Hz,
presence at 2 -3 kHz, sizzly and
raspy above 6 kHz.
Drums: Full at 100 Hz, wooly at
250-600 Hz, trashy at 1 -3 kHz, attack at 5 kHz, sizzly and crisp at 10
Kick drum: Full and powerful
below 60 Hz, papery at 300 -800 Hz
(cut at 400 -600 Hz for better tone),
click or attack at 2 -6 kHz.
Sax: Warm at 500 Hz, harsh at 3
kHz, key noise above 10 kHz.
Acoustic guitar: Full or thumpy
at 80 Hz, presence at 5 kHz, pick
noise above 10 kHz.
Voice: Full at 100-150 Hz
(males), full at 200 -250 Hz (females), honky or nasal at 500 Hz -1
kHz, presence at 5 kHz, sibilance
( "s" sounds) above 6 kHz.
wise for a boost, counterclockwise
for a cut. If your mixer has multiple- frequency EQ or sweepable EQ,
one knob sets the frequency range
while another sets the amount of
boost or cut.
Table 1 shows the fundamentals
and harmonics of musical instruments and voices. For any particular instrument, turn up the lower
end of the fundamentals for
warmth and fullness. Turn down
the fundamentals if the tone is too
bassy or tubby. Turn up the harmonics for presence and definition;
turn down the harmonics if the tone
is too harsh or sizzly. Avoid excessive boost because it can distort the
signal. Try cutting the lows instead
of boosting the highs.
Here are some suggested frequencies to tweak for specific instruments. If you want the effects
described below, apply boost. If you
don't, apply cut.
Bass: Full and deep at 60 Hz,
growl at 600 Hz, presence at 2.5
kHz, string noise at 3 kHz and up.
Electric guitar: Thumpy at
You can set an equalizer by ear as
well as by knowing the frequency
ranges of an instrument. One way
Figure 5. A typical graphic
is to tune the equalizer to the approximate frequency range you
need to work on (you'll soon know
where by experience). Then apply
full boost or cut so the effect is easily audible. Finally, fine -tune the
frequency and amount of boost or
cut until the tonal balance is the
way you like it.
For example, if a close -mic'd vocal
sounds unnaturally bassy, reach
for the low- frequency EQ (say, 100
Hz) and turn it down, adjusting the
cut for the desired tonal balance.
Ifyou hear a coloration in the tone
quality of an instrument, set a
sweepable equalizer for extreme
boost. Then sweep the frequencies
until you find the frequency range
matching the coloration. Cut that
range by the amount that sounds
right. For example, a piano mic'd
with the lid closed might have a
tubby coloration -say, excessive
output around 300 Hz. Set your
low- frequency EQ for boost, and
vary the center frequency until the
tubbiness is exaggerated. Then cut
at that frequency until the piano
sounds natural.
6. A peaking EQ.
Shelving EQ.
Figure 8.Bandpass EQ.
If a track contains bass and cymbals, you can EQ the low end of the
bass without affecting the cymbals
because the bass produces mostly
low frequencies, while the cymbals
produce mostly high frequencies.
If you assign each instrument to
its own track, the usual practice is
to record flat (without EQ) and then
track during
mixdown. I record with EQ, even
when multi -tracking. Why? The
monitor mixer in my board has no
EQ, so when I play back the multitrack recording through the monitor mixer, it doesn't sound right unless the tracks are already
equalized. Later, when I start to
mix down, the tracks already sound
good and need little tweaking.
If you're using a bass cut or treble
boost, you can get a better signal to -noise ratio by applying this EQ
during recording, rather than during mixdown. But if the EQ used is
a treble cut, applying it during
mixdown will reduce tape hiss.
tom gives a fuller sound, or cutting
around 250 Hz on a bass guitar aids
clarity. The frequency response and
placement of each mic affects tone
quality as well.
Special production effects.
Extreme EQ reduces fidelity, but it
can also make interesting sound effects. Sharply rolling off the lows
and highs on a voice, for instance,
gives it a "telephone" sound. A 1
kHz bandpass filter does the same
thing. An extreme boost at 5 kHz
can accent the impact of a snare
Reducing noise and leakage.
You can reduce unwanted low-fre-
quency sounds -bass leakage, airconditioner rumble, mic stand
thumps -by turning down low frequencies below the range of the instrument you're recording. For exUSES OF EQ
ample, a fiddle's lowest frequency is
Here are some applications
about 200 Hz, so you'd set the
equalizer's frequency range to 40 or
where EQ comes in handy.
Improving tone quality. This 60 Hz and apply cut. This roll -off
is the main use of EQ. It can make
won't change the fiddle's tone quality, because the roll -off is below the
an instrument sound better torange offrequencies that the fiddle
nally. For example, you might use a
high-frequency roll -off on a singer
to reduce sibilance, or on a direct Similarly, a kick drum has little
recorded electric guitar to take the
or no output above 5 kHz, so you
"edge" off the sound. As another excan filter out highs above 5 kHz on
ample, boosting 100 Hz on a floor
the kick drum to reduce cymbal
TABLE 1. Frequency ranges of musical instruments and voices.
261 -2349
261 -1568
165 -1568
62 -587
165 -988
87 -880
73 -587
49 -587
100 -200
30 -147
300 -587
196 -3136
131 -1175
65 -698
41 -294
41 -300
82 -988
82 -1319
82 -1319
28 -4186
247 -1175
175 -698
131 -494
87 -392
French horn
Snare drum
Kick drum
Acou. Bass
Elec. Bass
Acou. Guit.
Elec. Guit.
Elec. Guit.
Bass singer
Approx. range
3 -8 kHz
2 -12 kHz
2 -10 kHz
1 -7 kHz
1 -7.5 kHz
1 -6 kHz
1 -7.5 kHz
1 -4 kHz
1 -20 kHz
1 -6 kHz
1 -15 kHz
4-15 kHz
2 -8.5 kHz
1 -6.5 kHz
1 -5 kHz
1 -7 kHz
1 -15 kHz
1 -3.5 kHz (through amp)
1 -15 kHz (direct)
5 -8 kHz
2 -12 kHz
2 -12 kHz
1 -12 kHz
1 -12 kHz
leakage. If this filteringis done during mixdown, it will also reduce
tape hiss.
Filtering out frequencies below
100 Hz on most instruments reduces air -conditioning rumble and
muddy bass.
Remixing mono tracks. EQ
can actually change the mix between instruments within a single
track. To illustrate this, suppose
you have an old mono jazz recording of bass, drums and sax. Let's
say you want to remove everything
but the sax solo, and overdub new
bass and drum tracks with a contemporary, bright sound.
Here's how. Filter out the lows
and highs on the original recording
to remove bass and cymbals. You're
left with the midrange, which is
mainly sax. Copy this sax recording
to one track on your multi -track recorder. Now overdub bass and
drums on other tracks in sync with
the sax solo.
This trick was used in the motion
picture Bird to produce a high-fidelity sound track from Charlie
Parker's original recordings. A contemporary studio bass player and
drummer played along with
Parker's original sax solos, which
were gleaned from Parker's records
by filtering out the original bass
and drums.
Compensating for the
Fletcher- Munson effect. As dis-
covered by Fletcher and Munson,
the ear is less sensitive to bass and
treble at low volumes than at high
volumes. So, when you record a
very loud instrument and play it
back at a lower level, it might lack
bass and treble. To restore these,
you may need to boost the lows
(around 100 Hz) and the highs
(around 4 kHz) when recording
loud rock groups. The louder the
group, the more boost is needed. As
an alternative, use cardioid mics
with proximity effect (for bass
boost) and a presence peak (for
treble boost).
Making a pleasing blend.
When several instruments are
heard together, they sometimes
"crowd" or overlap each other in the
frequency spectrum. That is, it may
be difficult to distinguish the instruments by tonal differences. By
equalizing various instruments at
different frequencies, you can
make their timbres distinct, which
results in a more pleasing blend.
This procedure also evens out the
contribution of each frequency
band to the total spectrum, yielding
a mix that is tonally well -balanced.
If you have two instruments that
sound alike, such as lead guitar and
rhythm guitar, you can make them
more distinct by equalizing them
differently. You might make the
lead guitar edgy by boosting 2-3
kHz, and make the rhythm guitar
mellow by cutting 2 -3 kHz.
The same philosophy applies to
bass guitar and kick drum. Since
they occupy about the same low frequency range, they tend to mask
or cover each other. To make them
distinct, either fatten the bass and
thin out the kick a little, or vice
Compensating for response
deficiencies. The mics, tape re-
corder, monitor speakers and the
mixing board itself may not have a
flat frequency response. EQ can
partly compensate for these deficiencies. If a mic has a gradual
high-frequency roll -off, for exam-
ple, a high- frequency boost on the
console may help restore a flat response. On the other hand, if a mic
"dies" above a certain frequency, no
amount of boost can help it. Some
directional mics have proximity effect-a bass boost when used up
close. A bass roll-off on the console
can compensate for this boost.
Many purists shun the use of EQ,
complaining of excessive phase
shift or ringing caused by the equalizer. Instead, they use carefully
placed, high quality mics to achieve
a natural tonal balance without
EQ. The resulting sound is said to
be less strained and more natural.
Compensating for mic
placement. Often, you must place
a mic very close to an instrument to
reject background sounds and leakage. Unfortunately, a close -placed
mic emphasizes the part of the instrument the mic is near; the tone
quality picked up may not be the
same as that of the instrument as a
whole. EQ can partly compensate
for this effect.
For example, an acoustic guitar
picked up with a mic next to the
sound hole sounds bassy because
the sound hole radiates strong low
frequencies, but a complementary
low- frequency roll -off on your
mixer can restore the natural tonal
balance. This use of EQ can save
the day by fixing poorly -recorded
tracks in live concert recordings.
During a concert, the stage monitors might be blaring into your recording/P.A. mics, so you're forced
to mic close in order to reject monitor leakage and feedback. This
close placement, or the monitor
leakage itself, can give the recording an unnatural tone quality. In
this case, EQ is the only way to get
usable tracks.
You may be fortunate enough to
use optimally-placed mics in a
great acoustic environment. In that
case, EQ is not wanted nor needed,
but your recording will sound better with EQ than without it.
In the May /June issue we omitted two illustrations from the Bartletts'column. They are here included. A
photo of a cassette multi -track recorder was also omitted and is not reproduced herein.
These components are found in the cas-
Figure 2.
sette well.
head azimuth angle.
Designing Vocals: Part III
Parts I and II of this series on
designing vocals have focused on
the important foundational principles for capturing an inspired performance on tape -and doing it
with all the necessary technical
savvy. These are extremely important steps in the process, for a
poorly recorded track or a spotty
performance will undoubtedly
come back to haunt you in the mix.
In the case of flawed tracks, it's
amazing what a creative and technically competent engineer can do
to redeem the song, but it's really
very wise to make sure your vocal
tracks are solid, both technically
and performance -wise before going
into a mix. It's like the parable of
the house built on solid rock versus
the house built on sand: which one
do you suppose could survive a
storm with its torrential rains and
blustering winds?
Mixing can be compared to a
storm in that it starts out in utter
chaos -all the tracks are separate,
competing entities, needing to be
placed in proper relationship to
each other. Establishing order from
chaos is a rigorous task whose outcome can be frustrating or satisfying depending on certain factors.
For example, the more well- defined
the individual tracks are, the easier
it will be to find their proper niche.
With vocals, it is best to have dynamics somewhat restricted during the recording process, as it can
be rather nightmarish to try placing a vocal in the mix when it is unpredictably loud or soft. When applied during a mix, a compressor or
limiter might have to work so hard
to compensate for this that the
track would end up sounding flat
and lifeless. So the importance of
starting a mix with good tracks
should not be underestimated.
Still, the mix itself is probably the
most critical stage in the recording
process; it is (proverbially speaking) "where the rubber meets the
road," for it is at this stage that all
relationships between instruments, vocals and the sonic environment are permanently defined.
The product of this mixing session
is what ultimately reaches the ears
of the listener, so an engineer needs
to exercise the greatest degree of
discernment or else even well -recorded tracks will fail to shine.
Good tracks, particularly vocal
tracks, can easily get bogged down
in a bad mix. If they are dull or
harsh, too far under or over the music, or are buried beneath a wave of
excessive effects, your brilliant engineering and production will all be
in vain. It is therefore extremely
important that we carry a high
level of technical and aesthetic
awareness through the final stages
of production. With this in mind,
let's take a look at some of the critical factors in mixing vocals.
Vocal tracks will (in most cases)
have been subjected to some form of
dynamics control -limiting, compression or both-while they were
being recorded, but at mix time it is
frequently necessary to further
control the transient dynamics.
(When I use the term transient dynamics, I am referring to the shortterm peaks that make a track difficult to record and also to place in
the mix.) Smooth sounding vocals
are definitely a mark of professional- sounding recording. High end studios have very sophisticated
(and expensive) devices which apply both compression and limiting,
and assure that the output of the
track remains at a steady level- irrespective of the input. If you, like
most of us, cannot afford the bigbucks devices, you can still achieve
a sophisticated vocal sound by recompressing and/or limiting the
track on mixdown.
Whether you'll utilize compression or limiting depends on lots of
factors. You'll need to experiment in
both modes to see which works
best. The safest route in most cases
though, is to limit the vocal on recording, so as to preserve the nuances of the performance without
overloading the tape; limiting (with
the threshold properly set) just lops
off the large dynamic peaks in the
performance without really affecting the moderate and lower level
signals. Then later, upon mixdown,
if you really want a much tighter
uni- dynamical sound, you can apply some heavy compression to iron
out all the wrinkles, or some lighter
compression to just tighten it up a
There are many differences between types of compressors/limiters. Some respond to peaks, and
some (RMS sensitive devices) respond to an average energy profile,
rather than every little blip that
passes by. Each type has its
strengths and weaknesses and
each vocal performance is a unique
event, so while you can easily develop some rules of thumb, don't allow yourself to lapse into formula.
It's usually best to (as quickly as
possible) try several different settings of limiting or compression
and see which renders the desired
effect. Controlling short-term dynamics is one ofthe first things that
should be done before you attempt
to place the vocal track within the
mix. When this has been accomplished, you can confidently move
on to adjusting EQ and creating a
vocal ambience.
If there is but one axiom you
should remember with regard to 7
equalization of vocals, it is this: .ó
"Don't overdo it!" Many people 2
equate an excellent vocal sound
with a certain shimmering bril- v,
liance in the high end. This, I am
sure, started out as an attempt to
duplicate the frequency response of
expensive studio microphones
which can accurately capture the
high -end harmonics in a voice. But
as time rolled on, people became entranced by the magic of high frequencies, no longer to compensate
for deficiencies, but for the sheer
thrill of coming up with the hottest
possible sound. With this quest, of
course, came problems -lots of
them. When excessively laden with
highs, a vocal can quickly become
shrill and robbed of its power, and
words containing strong s sounds
(sibilants) become altered into sh
Some of this can be treated after
the fact by running the vocal
through a de -esser (a fast limiter
which is maximally sensitive to a
frequency responsible for the sibilance). Although the de -esser can
momentarily suppress peaks in the
fundamental range of sibilance and
fool the ear into thinking it's not
there, this device cannot totally
undo the mutation of the sound.
Some vocalists are more prone to
"messy esses" than others because
of the anatomy of their mouth, but
in most cases, otherwise normal vocalists are made to sound like hissing dragons due to the excesses of
an engineer who EQ'd the voice at
some dangerous frequency. Often,
the problem of sibilance is not
picked up until it is too late -when
the master is being prepared for duplication. So if there is a moral in
this message, it is this: unless you
have a real good reason to do so and
you're absolutely sure that it won't
cause excessive sibilance, avoid
positive EQ in the range of 6 k to 9
k. Negative EQ might frequently be
used to diminish sibilance, but
positive EQ should probably be
avoided like the plague!
The next logical question is, how
do we get sparkle into the vocal and
do it safely? If your mixing console
has a high frequency shelf (usually
affecting every frequency above 10
k), a small boost here (2 -4 dB) will
usually do quite nicely. Other consoles have peaking equalizers (affecting a select narrow band) in the
high frequency range. In this case,
a couple of options are open to you:
a modest boost in the 10 k range
will give you a strong dose of very
powerful highs, but be conservative
here because 10 k is close to 9 k; it is
just outside the range of sibilance
and if the cue (bandwidth) of the
equalizer is wide enough to overlap, you could possibly be augmenting the sibilance, so monitor carefully. Another option preferred by
many engineers is to give a liberal
dose of positive EQ in a portion of
the range most distant from the
sibilance, say 15 k, 16 k or higher.
The effect here is to boost the
weaker high harmonics of the
voice. Since the ear is not so sensitive to frequencies up that high, 6
dB or more can be added without
any deleterious results.
While we're on the subject of EQ,
it's worth noting a few more key frequencies that can really help you
place a vocal in the mix. For example, 5 k is usually considered some
sort of a magic number when trying
to get vocals to cut through in a mix.
It is a powerful upper midrange frequency that can really alter the apparent "presence" of a track. In
other words, adding 5 k will cause
the track to proceed (move forward
towards the listener), and subtracting 5 k will cause the track to recede
(move away from the listener). This
is, of course, only an illusion; it is a
based on the fact that human ears
are maximally sensitive in the region of 5 kHz. Any track that is exciting the eardrum at that frequency will seem like it's closer.
Excesses in this area are to be
avoided lest the voice becomes
brassy and hornlike.
Other areas of great power can
also be found. There are narrow frequency ranges called "formants"
which differ for male and female vocalists, and also between individuals. They are tied in to the anatomy
of the throat and head, and resonate at a characteristic fixed frequency irrespective of the pitch of
the note being sung. (For a more detailed discussion on formants and
EQ, see my article, The Art Of
Equalization: Part 2 in the
May /June 1990 edition of db
Magazine.) You really have to fool
around with a good sweepable
equalizer to home in on a formant,
but once you hit it, you will find that
small amounts of boost in this area
will increase the apparent power of
the track appreciably. Statistically,
there are considered to be two formant areas worthy of your investigation: the low formant, which is
roughly around 500 Hz for men and
1 k for women, and the high for mant, which is centered around 2.8
k for men and around 3.2 k for
women. If you use these figures as
starting points and carefully sweep
the area below and above, you can
usually find some sort of a "hot"
area which will prove useful in
shaping your vocal sound.
One quick word on background
vocals. Getting background vocals
to blend with a lead vocal is sometimes a little difficult when both
lead and background vocals are full
bandwidth tracks. In other words,
when both leads and backgrounds
have been EQ'd to sound magnificent -with sizzling highs, a tight,
powerful midrange and a warm low
is possible that their timbres will compete with each other,
thereby diminishing the overall
clarity of the mix. In that case,
something has got to give, and in
most cases, it ought to be the background vocals. Usually, by shaping
the backgrounds in some sort of
complimentary way, things will fit
a lot more easily. One of my favorite
techniques is radically rolling off
both the highs and lows on the
backgrounds, leavingthe midrange
intact. In this way, the backgrounds retain much of their power
and fullness, but draw less attention to themselves because they are
not full range tracks. Under alternate circumstances, the exact opposite technique can also work quite
well: rolling off lots of midrange
power, but leaving the high end sizzle and low end woof. The point is to
differentiate the lead and the background vocal, and this is usually accomplished by weakening the backgrounds so that they don't
dominate such a large region in the
frequency spectrum.
One major factor remains to be
explored: creating a vocal ambience
with echo, early reflections, reverb,
chorus, doubling, delay and so on.
We will do so in depth when Designing Vocals continues in the
next issue of db Magazine.
This console has over ten thousand different console configurations, and can be called up with all
parameters stored to hard disk and
instantly recalled in less than 30
mS. It features MIDI, SMPTE and
EBU interfaces along with remote
control of external tape machines.
There are three input connectors
per channel that select or recall
forty -eight out of one hundred and
forty -four sources at any time. Each
output channel, either group or matrix, is fitted with two outputs giving a maximum of ninety -six output
points and up to forty -eight outputs
simultaneously. A 32 -way matrix of
inputs /outputs along with sixteen
output buses are available so the
console can be configured for Front
of House, Monitor or Recording.
There are also sixteen stereo auxiliary returns with eight VCAcontrol-
led DCAgroups.
Prices: Memory Frame 32
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The DMR8 combines an 8 -track
digital recorder, digital mixer, locator and mixing automation, allow-
ing production facilities, artists,
producers and engineers to accomplish all phases of digital audio production in one desktop -sized unit.
The DMR8 has 20-bit sound quality
comparable to or better than that of
compact discs. It is a totally integrated system, permitting the user
to accomplish the entire process in
the digital domain from recording to
mixdown without having to think
about repatching, digital format
conversions, and the multiple A/D,
Manufacturer: Infoscene
Technologie Inc.
Circle 60 on Reader Service Card
D/A conversions. Of equal importance are the DMR8's automation
features. Console setups and real
time mixes are stored to either a
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The DMR8 can digitally memorize,
then instantly recall all static con-
trol settings -including panning,
EQ, track assignments, effects and
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Motorized faders indicate channel
level changes relative to time code.
Manufacturer: Yamaha
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Price: base, $34,000.00
Circle 61 on Reader Service Card
The RM220 loudspeaker is oper-
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posts power handling figures of 200
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Drivers incorporated in the loudspeaker's 3 -way proprietary Wave front Coherent design start at the
home for the IntelliSense circuitry
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Prices: the RM220 is $1,400.00
and the 220 System Controller is
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ranges from 80 Hz to 15,000 Hz. Its
impedance is 240 ohms (balanced),
with 1000 ohms being recommended for the minimum load impedance. Other specifications include a signal -to -noise ratio of 66 dB
at 94 dB SPL, a maximum SPL of
120 dB SPL (which produces three
percent THD), and a power sensitivity of -56 dB re 1 mW/Pa. Accessories included are a foam windscreen, mic swivel mount, two mic
cables and a carrying pouch.
Manufacturer: Crown
The CM -230 tridundant con-
Price: $795.00
denser microphone's capsules work
in conjunction with an interface
equipped with three transformer isolated outputs. Designed for use
with either a stand or a standard
gooseneck (a permanently mounted
female 5/8 in. -27 threaded Atlas
flange is provided to accommodate
the latter), the CM -230 comes with
two shielded cables with 4-pin XLRtype connectors. The typical frequency response of the CM -230
Circle 63 on Reader Service Card
The SM102 miniature hanging
condenser mic has a uniform cardioid polar pattern, low noise, high
sensitivity, and smooth, natural frequency response from 40 to 20,000
Hz. Its miniature mic head is permanently mounted to a 6 in. gooseneck, allowing the capsule to be
aimed and angled when the assembly is suspended from overhead. A
30 -foot attached cable exits from the
opposite end of the gooseneck and
connects to an included preampli-
low end with twin 8 in. ferro -fluid
cooled dual- spider woofers. Proceeding up the frequency spectrum
to the midrange, each loudspeaker
packs a single M -200 compression
driver, which employs a one -piece
Mylar diaphragm and features a 2
in. exit throat. High frequencies are
directed to another compression
driver which is equipped with a titanium diaphragm and a 1 in. exit
area. Crossover points are at 800 Hz
and 3 kHz. A220 System Controller,
a proprietary dedicated system, is
fier. Two types of preamps are available: a wall plate version and an inline version. Both contain a low -cut
switch, and the wall -plate version
will fit in a standard electrical wall
Manufacturer: Shure Brothers,
Price: from $215.00 to $260.00,
depending upon finish color and
preamplifier style
Circle 64 on Reader Service Card
38 Pine Hill Lane
Dix Hills, NY 11746
Bulk Rate
Permit # 120
Commack, NY
In one complete up -to-date book
here are all the practical as well
as theoretical aspects of sound reinforcement. The detailed chapters include information on electrical fundamentals, acoustical
fundamentals and psycho- acoustical aspects; high -, low-, and
mid-frequency systems; microphones in sound reinforcement
and system architecture; central
loudspeaker arrays, distributed systems,
speech reinforcement and paging systems; system intelligibility; high -level sound reproduction, a theater sound overview, and sections on
live music reinforcement, line arrays and sound
347pp. Hardcover
$37.50 #12 -991
CHURCHES by Curt Taipale
Tapes recorded during an actual seminar, you
will hear Curt present on these tapes not only
the basics of microphones, but how to get the
most out of your console both in a live setting
and in the studio; how to deal with feedback,
how to recognize phase cancellations caused by
poor speaker placement and much, much more.
Helpful diagrams are enclosed where appropriate. This is your chance to learn from Curt's
mistakes and his triumphs. His accomplishments and his failures are freely shared in an
encouraging manner.
Four cassette tapes, nearly five hours!
$35.00 #14 -991
HANDBOOK by John M. Woram and Alan P.
This new edition has been acIti n(w
cepted as the long -awaited replacement to the original book
published in 1976. The new edition is not "old wine in a new bottle." The revision has been done
by Professor Kefauver. He is the
Coordinator of the Recording Arts
and Sciences Department and Director of Recording at the prestigious Peabody Conservatory of Music. The book
is used by most of the recording schools and
universities here and abroad. This book contains all the basics for the recording studio engineer, as well as more advanced information
covering MIDI, Automated Consoles, SMPTE
Time Code and Digital Audio. This book has
been and remains the "bible" of the audio inRccundìny
Geared primarily for the aspiring professional,
this book provides a comprehesive discussion of recording, engiIntroduction to
Prolcsciotwl Recording
neering and production techTechniques
niques. Special coverage of
microphones, microphone techniques, sampling, sequencing,
and MIDI is also included.
416pp. Paper
$29.95 #8 -991
by Harvey Rachlin
This book and cassette meets the
_ songwriter on the level that he
an idea for
a song, it travels through inspiration and creativity, writing lyrics,
making a demo, understanding
MIDI and how to pitch songs to
the industry. Each lesson is to be
learned through reading and also
hearing the lesson, and is taught
by experts such as John Barilla.
96pp. and 2 cassettes, Paper
$24.95 #9 -991
This second edition of a very popular book has
an additional 40 pages and covers all basic aspects of sound reinforcement. The new topics
include MIDI, synchronication, and an Appendix on Logarithms.
$34.95 #17 -791
525pp. Hardcover
$45.95 #13 -991
LIVE SOUND! by David Scheirman
This excellent video is targeted at first-time users and musicians new to the field of sound reinforcement. However, the video contains insider tips and sophisticated approaches to
using the equipment.
The video covers:
Equipment selection
Loudspeaker placement and setup
Mic selection and placement
Monitor systems
Mixer Position
How to Soundcheck
System Assembly & Cables
Power Amps
Running the Mixer
This is a must have video!
75 minutes
$39.95 #16 -791
Like the first edition, this comprehensive text provides readers
with useful information for the
day-to -day work of designing
sound systems. This updated version contains in -depth coverage
that carefully examines acoustic
gain, clarity of sound, and required electrical power.
688pp. Hardcover
$49.94 #2 -991
Sound System
This book shows how to set up, maintain, and
operate sound and lighting equipment for the
performance of amplified music or any kind of
touring production. An excellent reference
and/or guide to procedure, the book provides
descriptions of all the components that make
up a system, explanations of how they all work
together, and photographs and illustrations
that show specific equipment and proper stage
178pp. Hardcover
$27.95 #4 -991
This brand -new second edition
has been updated to include the
latest in MIDI, cinema sound,
tranformers and compact discs.
Readers learn the new developments in audio electronics, circuits, and equipment. There is
also an in-depth examination of
disc, magnetic, and digital recording and playback.
$99.95 #5-991
Michael Talbot -Smith
This is an introduction to the
technical aspects of sound in radio
and television. It examines in detail the main items in the broadcast chain: studio acoustics, microphones, loudspeakers, mixing
consoles, recording and replay
(analog and digital), and the principles of stereo. It offers a easy
technical treatment of audio principles and broadcast hardare.
224pp. Hardcover
$42.95 #7 -991
-- r
Leaving the higher levels of theory to other digital audio texts,
this handbook emphasizes princiAUCNCS
OPerC"" ples for the studio and those aspects of digital audio appropriate
for day -to -day sound engineering
operations. It describes the sampling process, error correction, editing systems and different recording options. This book is
producers and engineers in the
written to
studio get the best possible results from the
high quality standard equipment in use today.
256pp. Hardcover
$39.95 #6 -991
GronC, Rufn58Y
by Bruce Bartlett
This book is extremely timely for
sound engineers and video or
audio producers. Also, as Digital
Audio Tape (DAT) production becomes less costly to use in the
field, all electronic media will be
trying to achieve the highest
level sound production possible.
This book tells how to position
the correct microphones in the
proper locations in order to record optimal
quality stereo sound. The many illustrations
and clear organization easily explain the theory
behind stereo mic'ing methods, and describe
specific techniques, including comparative
evaluations. In addition, it offers suggestions
on session procedures and stereo troubleshooting as well as recent developments in binaural
and transaural stereo and stereo boundary arrays.
192pp. Paper
$24.95 #3 -991
This book is written to assist in
the design, purchase, and operation of a sound system. It procuU
vides the basic information on
sound systems that is most
needed by ministers, members of
Boards of Trustees and worship
and music committees, interested
members of congregations, and
even employees of musical instrument dealers that sell sound systems. To be of greatest value to all, it is written
to be both nondenominational and "non -brandname."
183pp. Paper
$24.95 #1 -991
setting levels, working with the console, doing
a recording, overdubs and mixdown. This is a
very important video for all persons planning
on setting up a studio.
Approximately one hour long
This handbook covers the entire
range of sound recording and emphasizes the technology of the
field as well as the aesthetic asHA NOOK
pects of actual recording. It feaFRINÉpNN
tures a sequence of six chapters
covering the basic tools of the
trade, from the acoustical recording environment through the re-
$34.95 #18 -791
producing environment. Eargle
details the actual production decisions which are made in classical and popular
recording. Also covered are the physical and operational principles of tape, disc, and digital recording, as well as the economic and physical
aspects of the low -cost studio.
405pp. Hardcover
$66.95 #11 -991
Using a "hands on" approach, this video gives
you an overview of the basics in setting up a
personal studio. Among items covered are:
hooking up instruments, connecting equipment,
516 586-6530 -tor Customer Service
516 586- 6810
by Jim Mandell
Here is a comprehensive survey
on the state of recording studio
business and management in the
nineties that includes startup and
equipment cost comparisons from
low budget to world class operations;
equipment purchasing
strategies; rate- setting factors;
actual examples of pre- session
contracts; how different studios
handle billing, credit applications, payment
guarantees, conflicts and collections; how to
write publicity releases that will get into print;
what to avoid in advertising;
336pp. Paper
$29.95 #10 -991
38 Pine Hill Lane, Dix Hills NY 11746
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db BUYR'5
Crossovers, Delays, Equalizers,
Multi- Effects Processors and
On the pages that follow, we present this issue's Buyer's Guide on Crossovers, Delays, Equalizers, Multi -Effects
Processors and Reverbs The information contained is supplied by the respective manufacturers. Further, if a manufacturer
that you seek is not listed, the chances are strong that, as many times as we tried, we could not get information from them.
Model 1631A is a two -way electronic crossover using plug -in modules to select crossover frequency and configure specific equalization to
provide flat power response for various horn /driver combinations. The high -pass output has a level control and the low -pass output has a
delay adjustment of 0 to 25 ms.
Dimensions: 1.75 in. X 19 in. X 4.875 in.
Weight: 4.74 lbs.
Price: $660.00
The 1632A Electronic Dividing Network is a dual channel two -way or single channel three -way active crossover, 24 dB /octave, selectable
from 50 Hz to 10 kHz; elect. balanced in /out with xfmr in /out optional; 30/60 Hz HP inputs, hard limiters on all 4 outputs; sub -modules to
customize response.
Dimensions: 1.75 in. X 19 in. X 9.75 in.
Weight: 8 lbs.
Price: $1,150.00
The 15594A Low Pass Crossover /Equalizer Module is a plug -in module for the 9400 series power amplifier; has 18 dB /octave roll -off
pre-programmed at 125 Hz, 500 Hz, 800 Hz, 1250 Hz; customer programmable for other frequencies; programmable 12 dB HP roll -off with
pre -sets at 16 Hz or 32 Hz.
Dimensions: 1.6 in. X 2 in.
Weight: 1.6 oz.
Price: $96.00
The 15595A High Pass Crossover /Equalizer Module is a plug -in module for the 9400 series power amplifier; 18 dB /octave roll -off
pre-programmed at 125, 315, 500, 800, 1250 Hz; customer programmable for other frequencies; sub -modules available to customize
frequency response to horn /driver.
Dimensions: 1.6 in. X 2 in.
Weight: 1.6 oz.
Price: $96.00
XR22E is stereo two-way 12 dB/octave.
Price: $379.99
XR77E is stereo three -way, mono four/five way 12 dB /octave.
XR -4000B is a 24 dB/octave, XLR- equipped stereo four-way crossover.
Price: $769.00
-See our ad on page 2
The EC -1 is a low noise Linkwitz -Riley 24 dB Phase correct electronic crossover. It is ideally suited for all studio installation and live sound
Dimensions: 1.75 in. X 19 in. X 6 in.
Weight: 2.5 kg
Price: $299.00
BRYSTON/Brystonvermont Ltd.
The 10PBX is a 2 -way stereo 3 -way mono crossover with 12 switchable turnover points, 3 switchable slopes of 6, 12 +18 dB per octave;
balanced input/output; high frequency gain or cut control and mute switches; S/N ratio of -90 dB; distortion of 0.005 percent; 20 k ohm input
impedance and output impedance of 100 ohms.
Dimensions: 1.75 in. X 19 in. X 10 in.
Weight: 12 lbs.
Price: $1,295.00
The 10PBX LR is a 2 -way stereo, 3 -way mono Linkwitz -Riley slopes with fixed cross -over points; high frequency gain or cut control; S/N ratio
of -90 dB; distortion of 0.05 percent; 20 k ohm input impedance and output impedance of 100 ohms.
Dimensions: 1.75 in. X 19 in. X 10 in.
Weight: 12 lbs.
Price: $1,350.00
The Audio Logic X 34 Stereo 3 -way, Mono 4 -way Crossover features 24 dB per octave Linkwitz -Riley filter topology; continuously variable,
extended range, crossover points; independent level; mute and polarity controls on each output; and 15 Hz, 4th order Butterworth high -pass
filters on each input.
Dimensions: 1.75 in. X 19 in. X 6.5 in.
Weight: 3 lbs.
Price: $475.00
ELECTRO-VOICE, INC. -See our ad on page 3
Model XEQ -3 /Electronic Crossover features 3 -way configurations; allows low- frequency signal delay for source alignment; low- frequency
boost for extended bass; step -down operation of TL bass system. Has simple, easy to install modules for compression- driver high-frequency
Dimensions: 1.73 in. X 19 in. X 7.28 in.
Price: $820.00
The Model TX -324 stereo 2- way /mono 3 -way crossover features 24 dB /octave rolloff slopes. Field Select allows optimizing filters for
long -throw (Butterworth) or near field (Cauer); hard limiters on each output with adjustable threshold provide speaker protection; includes
on /off transient muting; ground lift switch; in /out level controls; limit threshold indicators. Optional balanced configuration.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 7 lbs.
Price: $419.00
Model TX-424 stereo 3- way /mono 4 or 5 -way crossover has features similar to the Model TX -324.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 9 lbs.
Price: $549.00
Model TX -524 stereo 4 -way crossover has features similar to the TX -324.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 9 lbs.
Price: $679.00
The Model TX-3A is a 12 dB/octave tunable crossover that may be used for either stereo 2 -way or mono 3 -way applications. Includes
calibrated input/output level controls, power indicator and ground lift switch. Optional balanced configuration.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 7 lbs.
Price: $319.00
The ECU -2 is a stereo electronic crossover unit capable of stereo bi- amping as well as stereo tri -amping. Crossover points are continuously
variable from 70 Hz to 11 kHz. It has 12 dB /octave Butterworth filters; summed mono output for subwoofer operation; individual phase
switches on mid and high bands.
Price: $295.00
The WS -SP2A Subwoofer Processor (crossover) is networked for use with Ramsa loudspeakers. Includes 6th order alignment network for
Ramsa subwoofers; has frequencies of 50 Hz, 80 Hz and 120 Hz; A and B (left and right) inputs; XLR; +4 dB balanced; A and B outputs;
phone jack; +4 dB unbalanced; VLF is sum of left and right passed through crossover filter network; mono has phone jack, +4 dB
Dimensions: 1.75 in. X 19 in. X 7.875 in.
Weight: 6 lbs.
Price: $275.00
The V4X is a variable 4 -way electronic crossover; low, mid, high and very high level controls; switchable high EQ; balanced outputs; high and
low pass filters; at 40 Hz and 20 kHz, calibrated System Gain Control; balanced XLR and 1/4 in. input jacks; transformer- balanced XLR and
1/4 in. output jacks for all four bandpass outputs.
Dimensions: 1.75 in. X 19 in. X 9 in.
Weight: 8 lbs.
Price: $399.99
The PC4 -XL is a totally programmable, all digital four -way (mono) crossover; three -way mono with 4th output as additional LF out; MF out or
HF out; two -way mono or stereo; 48 kHz sample rate; 24 -bit internal processing; 64 times oversampled A -D; 70 to 650 ms of pre -delay time;
up to 10 ms of delay on each output for driver alignment; two balanced inputs, four balanced outputs; selectable filter type.
Price: $799.99
The AC 22 and AC 23 State Variable Time Correcting crossovers feature 24 dB /octave Linkwitz -Riley filter performance via 41- detent
frequency selector controls; built -in variable time delay for phase correction; automatic internal configuration switching; mute switches and
input/output level controls with 6 dB gain each.
Dimensions: 1.75 in. X 19 in. X 5.25 in.
Prices: $389.00 and $499.00
The FAC 24 Flex Series Crossover features 24 dB/octave Linkwitz -Riley performance; 24- position digital frequency selector switch for plug -in
card accuracy; electronic phase alignment; built-in adjustable CD -horn EQ; mono sub -bass input; and fully balanced ins /outs in half rack
Dimensions: 8.5 in. X 1.75 in. X 8 in.
Weight: 4 lbs.
Price: $339.00
The FAC 28 Flex Series Crossover is identical to the FAC 24 except that it features 48 dB/octave slopes to minimize the crossover region
and associated problems.
Price: $449.00
The 524E multi -mode crossover has four configurable bands; precision cards that set frequencies and slopes; limiter attack/release times; HF
horn EQ in /out; flat response from 20 Hz to 50 kHz; 0.01 percent distortion; threshold, gain, mute, phase reverse and phase adjust controls.
Dimensions: 1.75 in. X 19 in. X 9 in.
Weight: 10 lbs.
Price: $1,095.00
The DSP 5000 has a digital crossover, delay and parametric equalization all in one rack space; 19 bit, user configurable single channel in, 4
out; remote control capability via PA -422; MIDI or contact closures.
Dimensions: 1.75 in. X 19 in. X 12 in.
Weight: 9 lbs.
Price: $3,400.00
EVENTIDE, INC. -See our ad on page 5
The PD860 precision delay features stereo; 20 kHz frequency response; delay adjustable from 15 milliseconds to 5.24438 seconds;
adjustable in microsecond increments.
Dimensions: 1.75 in. X 19 in. X 12.5 in.
Weight: 8 lbs.
Price: $2,695.00
The BD980 broadcast delay has stereo; 20 kHz frequency response; 10 seconds maximum delay; dump; wait & exit; ramp to zero functions.
Dimensions: 1.75 in. X 19 in. X 12.5 in.
Weight: 15 lbs.
Price: $5,495.00
The BD955 broadcast delay features mono; 15 kHz frequency response; variable delay; dump and catch -up functions.
Dimensions: 1.75 in. X 19 in. X 12.5 in.
Weight: 10 lbs.
Prices: $3,360.00 (3.2 seconds maximum) or $4,300.00 (6.4 seconds maximum)
The BD941 broadcast delay features mono; 20 kHz frequency response; fixed delay; delete and bypass functions.
Dimensions: 1.75 in. X 19 in. X 9.4 in.
Weight: 5.5 lbs.
Prices: $1,795.00 (6 seconds) or $2,195.00 (12 seconds)
The BD942 broadcast delay features stereo; 20 kHz frequency response; fixed delay; delete and bypass functions.
Dimensions: 1.75 in. X 19 in. X 9.4 in.
Weight: 5.6 lbs.
Prices: $1,995.00 (3 seconds) or $2,395.00 (6 seconds)
The DN716 is a one in, 3 out 16 bit digital delay line with less than 90 dB dynamic range, 20 Hz -20 kHz, unweighted. Delay times from 0 -1.3
seconds in minimum increments of 20us; input level indicator and level control; non -volatile memory; electronically balanced input;
unbalanced outputs; transformer balancing optional.
Dimensions: 1.75 in. X 19 in. X 11.75 in.
Weight: 5.5 lbs.
Price: $1,625.00
The DN726 has two in, two out stereo 16 bit digital delay; 100 percent stereo tracking; control functions lock out and non -volatile memory;
dynamic range of less than 90 dB; 20 Hz -20 kHz unweighted; electronically balanced inputs; unbalanced outputs; transformer balancing
Dimensions: 1.75 in. X 19 in. X 11.75 in.
Weight: 5.5 lbs.
Price: $3,500.00
The DN726V is very similar to the DN726, but will display in either milliseconds or fields, and is switchable between PAL and NTSC
standards (internally). Also has a (4) GPI control function to automatically follow delay introduced by other devices. For use in video
Dimensions: 1.75 in. X 19 in. X 11.75 in.
Weight: 5.5 lbs.
Price: $3,900.00
The DN775 is a stereo disc -cutting delay, switchable to select 33 or 45 RPM; 100 percent stereo tracking; less than 90 dB dynamic range;
20 Hz -25 kHz; unweighted, electronically balanced inputs; transformer balanced outputs (standard); frequency response of 20 Hz -25 kHz +
dB, any level, any delay.
Dimensions: 12.75 in. X 19 in. X 11.75 in.
Weight: 5.5 lbs.
Price: $3,900.00
The LXP -15 combines range of reverb, pitch shifting and delay effect with fast editing of presets, MIDI control in a single rack -space package
and user interface. Offers 128 preset effects with up to five pages of parameters per effect, and the ability to store 128 of your own effects
and five external analog inputs for foot switches or pedals.
Dimensions: 1.75 in. X 19 in. X 13.9 in.
Weight: 12 lbs.
Price: $1,050.00
The WZ -9375 has 2 inputs with 2 outputs, alternately, 1 input with 4 outputs; up to 654 msec @ 100 kHz sampling rate; 10 microseconds to
1 millisecond of delay time steps; 50 kHz or
100 kHz sampling rate; frequency response of 20 Hz to 20 kHz, +0.5, -2 dB at 100 kHz
sampling rate; dynamic range of more than 90 dB; less than 20 micro-seconds of group delay; less than 0.03 percent at 100 kHz sampling
rate T.H.D.
Dimensions: 3.5 in. X 19 in. X 13.75 in.
Weight: 19.5 lbs.
Price: $4,500.00
Features digital companding PCM system equivalent to a 16 -bit A/D /A converting system; dynamic range more than 100 dB; frequency
response ranges from 10 Hz to 17 kHz with delay time from 0 to 1,500 ms; can store up to 8 different programmable memories.
Dimensions: 1.75 in. X 19 in. X 11.75 in.
Weight: 11 lbs.
Price: $1,095.00
The SSD550 surround and ambience delay unit features two channels of delay; 5 to 50 ms; may be switched to sequential for 10 to 100 ms..
variable mix of original and delayed signals available; passive surround decoder for film; S/N 90 dB; response 10 to 8000 Hz.
Dimensions: 3.5 in. X 19 in. X 9 in.
Price: $975.00
The DSP 5000 has a delay, parametric equalization and digital crossover all in one rack space; 19 bit, user configurable single channel in, 4
out; remote control capability via PA -422, MIDI or contact closures.
Dimensions: 1.75 in. X 19 in. X 12 in.
Weight: 9 lbs.
Price: $3,400.00
The MEQ -230 Precision Equalizer has dual 30 band, 1/3 octave EQ in single 19 in. rack space; interface provided by means of 1/4 -in. and
RCA jacks; center frequencies range from 25 Hz to 20 kHz and are set to ANSI /ISO standards; each band provides 12 dB cut/boost; in /out
Dimensions: 1.75 in. X 19 in. X 4 in.
Weight: 2.5 lbs.
Price: $249.00
The 8558B Programmable Microaudio Equalizer offers eight memories; only one rack space; no front panel controls; 28
12 dB of cut/boost; fixed HP /LP filters; elect. balanced in /out; xfmr in /out optional barrier strip only.
Dimensions: 1.75 in. X 19 in. X 7 in.
Weight: 5.9 lbs.
Price: $1,320.00
octave filters with
The 1750A Cut -Only 1/3 Octave Mono Equalizer has 28 constant -Q filters from 31.5 Hz to 16 kHz; 15 dB of attenuation per filter; 20 dB of
broadband gain; variable HP /LP filters; elect. balanced in /out with optional xfmr, XLR and barrier strip.
Dimensions: 3.5 in. X 19 in. X 9.75 in.
Weight: 10.7 lbs.
Price: $1,200.00
The 1753A Boost-Cut 1/3 Octave Mono Equalizer has 28 constant -Q filters from 31.5 Hz to 16 kHz; 12 dB cut/boost per filter; 20 dB
broadband gain; variable HP /LP filters; elect. balanced in/out with optional xfmr, XLR and barrier strip.
Dimensions: 3.5 in. X 19 in. X 9.75 in.
Weight: 10.7 lbs.
Price: $1,200.00
-See our ad on Cover III
The HD 31, Model 350 is an active balanced 1/3 octave 31 band equalizer featuring constant Q filters; 60mm sliders, switchable 15 and 7.5
dB level scale; switchable subsonic and ultrasonic filters; hard bypass at no power and S/N of 115 dB.
Dimensions: 3 in. X 19 in. X 6.25 in.
Weight: 8 lbs.
Price: $425.00
The HD 15, Model 340 is an active balanced 2/3 octave 15 band equalizer with constant Q filters; 60mm sliders; optional XLR connections;
switchable subsonic and ultrasonic filters; hard bypass at no power and S/N of 115 dB.
Dimensions: 3 in. X 19 in. X 6.25 in.
Weight: 8 lbs.
Price: $425.00
-See our ad on page 2
The EQ 30 and 60 are ultra low noise constant Q 1/3 octave graphic equalizers featuring balanced XLR and jack inputs /outputs and
switchable ±6 dB or 15 dB of cut/boost.
Dimensions: The EQ 30 is 3.5 in. X 19 in. X 10 in.
The EQ 60 is 5.25 in. X'9 in. X 10 in.
Weight: The EQ 30 weighs 3 kg
The EQ 60 weighs 4.5 kg
Prices: The EQ 30 is $899.00
The EQ 60 is $1,349.00
The Multi Q is a six channel/band fully variable parametric EQ Featuring ARX Auto Patch, the Multi Q allows the user to select any number
of channels without the need for patch cables.
Dimensions: 1.75 in. X 19 in. X 6 in.
Weight: 2.5 kg
Price: $688.00
GQ -215 is a stereo 15 -band
octave graphic equalizer.
Price: $549.99
GQ -131 is mono with 31 bands of
graphic eq. There is a stereo version, model GQ -231.
Price: GQ- 131 -$599.99, GQ- 231 -$1,099.99.
The 905 three -band parametric equalizer features instant before /after comparisons available by switch bypass; symmetrical peak/dip; and
switchable notch mode on each band.
Dimensions: 5.25 in. X 1.5 in. X 9.5 in.
Weight: 0.75 lbs.
Price: $499.00
The 1531X graphic equalizer has selectable 15 band stereo (2/3 octave) or 31 band mono (1.3 octave) equalizer on ISO centers; constant -Q
and symmetrical peak/dip curves with selectable 7.5 or 15 boost or cut; and switchable HP filtering at 20 Hz, 60 Hz or 120 Hz.
Price: $419.00
The DigiTech MEQ 28 Mono 28 -band MIDI Programmable Graphic EQ is a two space, rack-mount mono graphic EQ that is fully MIDI
controllable and programmable with 99 user-definable programs. It features 28 bands of 12 dB cut/boost equalization.
Dimensions: 3.5 in. X 19 in. X 8.5 in.
Weight: 7 lbs.
Price: $569.95
The DOD 830 Stereo 15 band per channel, 2/3 Octave Graphic EQ is a two rack space EQ featuring 20 Hz to 20 kHz equalization; 12
cut/boost; low cut filter; 90 dB S /N; THD 0.006 percent and 5 percent frequency tolerance.
Dimensions: 3.5 in. X 19 in. X 8.5 in.
Weight: 7 lbs.
Price: $319.95
ELECTRO- VOICE, INC. -See our ad on page 3
Model 2710 1/3 octave graphic EQ features 27 -band, 1/3- octave equalizer; constant range variable -Q filters; minimal interference between
adjacent filters; user-selectable high- and low -pass filters; built -in pink -noise generator for noise masking; system equalization and other
Dimensions: 3.5 in. X 19 in. X 10.25 in.
Weight: 11.5 lbs.
Price: $1,130.00
Model GQ -31 is a 31 -band single rack space graphic equalizer. Design results in extremely low noise, even with large amounts
of boost or
cut. Features include ±12 dB of equalization; gain control; LED indicators for overload; EQ in, and power, as well as Loc Cut
button and
ground lift switch. Optional balanced configuration.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 6 lbs.
Price: $369.00
The Model GQ -15 stereo graphic equalizer is the same as model GQ -31, except it has two channels, each with 15 bands spaced
at 2/3
octave intervals. Single rack unit height.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 6 lbs.
Price: $379.00
The Model GQ -62 stereo 31 -band graphic equalizer is the same as model GQ -31, except it has two complete 31 -band channels in one
double -height rack chassis.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 10 lbs.
Price: $699.00
The Model PQ -4 parametric equalizer has constant -Q equalization curves; peak/shelf switches on top and bottom bands; extra wide range
bandwidth and EQ adjustment. Includes input level control; EQ in button, as well as overload; EQ status and power indicators; high and low
level inputs /outputs; and footswitch jack, allowing use as a preamp. Balanced configuration is optional.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 6 lbs.
Price: $379.00
The DN410 is a dual (5) band/Single (10) band parametric equalizer with 100 percent frequency overlap on all bands; +15/ -25 dB boost/cut;
1/42 to 2 octave bandwidth; separate variable high/low pass filters (each channel); separate EQ in/out switch
on all bands plus overall noise
less than -94 dBm; 20 Hz -20 kHz, unweighted.
Dimensions: 3.5 in. X 19 in. X 9.25 in.
Weight: 10 lbs.
Price: $1,195.00
The DN405 is the same as above, but with single (5) band only.
Dimensions: 1.75 in. X 19 in. X 9.25 in.
Weight: 7.7 lbs.
Price: $775.00
The DN360 is a dual channel 30 band 1/3 octave graphic equalizer with switchable 12 dB /6 dB scale on faders; switchable high pass filters;
electronically balanced inputs, unbalanced outputs; transformer balancing optional; noise less than 90 dBm; 20 Hz -20 kHz unweighted.
Dimensions: 5.25 in. X 19 in. X 8 in.
Weight: 10 lbs.
Price: $1,795.00
The DN300 is a single channel 30 band 1/3 octave equalizer with continuously variable high and low pass filters; switchable 12 dB /6 dB fader
scale; noise less than 90 dBm; 20 Hz-20 kHz unweighted; electronically balanced input; unbalanced output; transformer balancing optional.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 7.7 lbs.
Price: $1,150.00
The DN301 is a single channel 30 band 1/3 octave Cut only graphic equalizer with continuously variable high and low pass filters; switchable
12 dB/6 dB fader scale; electronically balanced input; unbalanced output; transformer balancing optional; noise less than 94 dBm; 20 Hz-20
kHz unweighted.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 7.7 lbs.
Price: $1,150.00
The DN332 is a dual 16 band
2/3 octave graphic equalizer with +12 dB boost/cut; switchable high pass filters; electronically balanced inputs;
unbalanced outputs; transformer balancing optional; noise less than -90 dB;, 20 Hz -20 kHz unweighted.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 7.7 lbs.
Price: $1,095.00
The PEQ is a dual -channel, 4 -band parametric equalizer with selectable peak/dip or shelving response on upper or lower bands, overall
hard -wire bypass and individual bypass on middle 2 bands. Bandwidth variable from 0.15 to 2 octaves.
Price: $595.00
The PEQ -1 is a single-channel version of the PEQ -2. Utilizes a single -rack space.
Price: $349.00
The Model 642B dual channel /stereo is a fully parametric equalizer with 4 bands per channel, switchable to 8 channels mono; each band
with separate bypass, Q, frequency and fine tuning control; high pass and low pass filters per channel; minimum 40 dB notch per channel.
Dimensions: 3.5 in. X 19 in. X 11.25 in.
Price: $1,200.00
The models 672A/674A mono /stereo 8 -band graphic parametric equalizers have long throw faders controlling boost and cut for each band;
high pass and low pass filters with separate outputs for use as 2 -way crossover.
Dimensions: 3.5 in. X 19 in. X 5.25 in.
Prices: $725.00 for the 672A
$1,525.00 for the 674A
Model 764A features programmable, digitally -controlled parametric equalizer version of the 642B; controls up to 99 channels of masters and
slaves; stores up to 99 presets; has four bands, dual channel, with high and low pass filters; and programmable input attenuator.
Dimensions: 3.5 in. X 19 in X 9.625 in.
Price: starting at $1,900.00, depending on configuration
The DEQ -1 High Resolution programmable 1/3 octave equalizer has 29 1/3 octave filters adjustable in 1/2 octave spacing; 8 presets with
security; balanced inputsioutputs; PA -422.
Dimensions: 1.72 in. X 19 in. X 13.5 in.
Weight: 13 lbs.
Price: $1,060.00
The DEQ -II High Resolution Programmable 1/3 octave Equalizer has 29 1/3 octave filters adjustable in 1/2 dB steps; high/low pass filters
selectable on 1/6 octave spacing; large LCD display and front panel cortrols make programming simple; 8 presets with security; balanced
inputs/outputs; PA -422.
Dimensions: 3 in. X 19 in. X 13.5 in.
Weight: 15 lbs.
Price: $1,400.00
-See our ad on Cover II
The AEQ 2800 is an automatic equalizer with up to 12 complete EQ memories; automatic EQ curve fit; 28 -band EQ on 3rd octave centers;
user friendly; ±12 dB in 1 dB steps; 40 X 2 character liquid crystal display; 128 complete EQ program memories.
Price: $499.99
The PME 4 is a 4 -band parametric equalizer with control over 11 octaves via state -variable filters; four bands with calibrated adjustment; 18
dB boost/cut; 1/2 to 2 full octave range.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 6 lbs.
Price: $229.99
The Autograph is programmable with automatic EQs with up to 128 user -selectable program memories; complete with real -time analysis EQ
capability; ±12 dB in 1 dB steps; ±6 dB in 0.5 dB steps; 8 settings; MIDI-controllable sliders; rack mountable.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 7 lbs.
Price: $549.99
The PME 8 is a stereo version of the PME 4.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 10 lbs.
Price: $349.99
The PME 4000 is a parametric control over 11 octaves; top and bottom bands switchable (peak to shelving); +4 balanced in and out; 4
parametric bands.
Price: $349.99
The EQ 215 has two 2/3 octave graphic equalizers; ±6 or ±12 dB ranges; level control; EQ bypass; +24 dBv input and output capability.
Price: $399.99
The EQ 31 has 31 bands of graphic EQ; 150 centers; ±6 or ±12 dB ranges; level control; low and high cut titers; +24 dBv input and output.
Price: $379.99
The ME 30 and ME 15 MicroGraphic Equalizers feature constant -Q 1/3 and stereo 2/3 octave performance in single rack space packaging;
with switchable ±6/12 dB boost/cut; input level; hard -wire bypass and 20 mm center-detent sliders.
Dimensions: 1.75 in. X 19 in. X 5.25 in.
Weight: 5 lbs.
Prices: $359.00 and $369.00
The GE 27 and GE 14 Graphic Equalizers feature constant -Q 1/3 and 2/3 octave performance in two rack space packaging; with 45 mm
center- detent sliders; level control; hard wire bypass; low noise and low distortion circuitry.
Dimensions: 3.5 in. X 19 in. X 8.5 in.
Weight: 9 lbs.
Prices: $499.00 for GE 27 and $529.00 for GE 14
The SP 15 Studio Parametric Equalizer /Notch Filter provides 5 bands, each with 4- octave sweep; bandwidth from 1.5 to 0.03 octave; +12/ -15
dB boost/cut; individual bypass; overall bypass and gain control; and fully balanced input/output. Noise and distortion specifications exceed
16 -bit digital performance.
Dimensions: 1.75 in. X 19 in. X 5.25 in.
Weight: 5 lbs.
Price: $599.00
The FPE 13 Flex Series Parametric Equalizer features three fully parametric full -range bands in a single channel half -rack format. Vertically
or horizontally mountable, the unit provides fully balanced three -pin and 1/4 in. input/output; exclusive I/O patch point; overall gain and
bypass; and bandwidth range from 0.03 to 2 octaves and 10 Hz -20 kHz frequency range for each band.
Dimensions: 8.5 in. X 1.75 in. X 8 in.
Weight: 4 lbs.
Price: $289.00
The FME 15 Flex Series MicroGraphic Equalizer is a single channel 2/3- octave Interpolating Constant -Q graphic equalizer with dual boost/cut
range switch; input/output level controls; exclusive Patch I/O jack; and fully balanced three -pin, terminal strip and 1/4 in. input and output
Dimensions: 8.5 in. X 1.75 in. X 8 in.
Weight: 4 lbs.
Price: $289.00
The MPE SERIES Programmable Equalizers feature the MPE 28 1/3 octave and MPE 14 Dual 2/3 octave equalizers with 128 memory
locations plus a software package that enables curve weighting (adding 2 curves together); real time program changes; remote control;
copying; data- dumping, full MIDI mapping and other functions.
Dimensions: 1.75 in. X 19 in. X 8.5 in.
Weight: 6 lbs.
Prices: $749.00 for the MPE 28 and $799.00 for the MPE 14.
The FBX Feedback Exterminator is a microprocessor- controlled, parametric, filtering device which automatically seeks out and eliminates
feedback in sound systems and continuously updates the filters as necessary.
Dimensions: Single space rack mount
Price: $550.00
The PRO-EQ 22 C -MOS 0.1 dB Differential /Comparator Octave Equalizer is a two- channel device with 10 octave -wide bands of adjustment
for each channel featuring C -MOS Digital Switching; Differential/Comparator 0.1 dB True Unity Gain controls; LED True Unity Gain indicators;
EQ defeat totally bypasses equalizer; Pre/post EQ processor loops.
Dimensions: 3.5 in. X 19 in. X 11 in.
Weight: 15 lbs.
Price: $349.00
The PRO -EQ 44 is a C -MOS 0.1 dB Differential/Comparator Third Octave featuring C -MOS digital switching; two independent channels of
EQ; 1/3 octave 40 Hz/1 kHz; alternate 1/3 octave 1 kHz /16 kHz; exclusive differential /comparator unity-gain circuits; balancing LEDs for instant
adjustment to unity gain; pre -post EQ loops and EQ defeat switch.
Dimensions: 3.5 in. X 19 in. X 11 in.
Weight: 15 lbs.
Price: $549.00
The EQP -200A is a dual program equalizer utilizing tube gain make -up stages with 990, balanced output. All units are hand -crafted and
burned in for ten days or more.
Dimensions: 3.5 in. X 19 in. X 10.5 in.
Weight: 19 lbs.
Price: $2,100.00
The EQF -100 Full Range Vacuum Tube Equalizer is a full- range, single channel, four band equalizer with Hi /Lo pass filter section; musically
selected center frequencies; with bands one and four peaking of shelving selectable; 990, balanced output.
Dimensions: 3.5 in. X 19 in. X 10.5 in.
Weight: 21 lbs.
Price: $2,200.00
The SX201 parametric EQ has three overlapping bands; +15 dB boost; -30 dB cut; 0.05 octave to 3.3 octaves bandwidth; 119 dB S/N ratio;
20 Hz to 20 kHz response ( +0, -1 dB).
Dimensions: 1.75 in. X 8 in. X 5.5 in.
Weight: 5 lbs.
Price: $259.00
The Model 4700/4700 -2 is a digitally -controlled
octave equalizer; has one or two channel; controllable from the front panel with password
protection or software control via RS -232 or PA -422 interface with Pilot 447 software provided.
Dimensions: 1.75 in. X 19 in. X 12 in.
Weight: 9 lbs.
Prices: $875.00 mono/$1,375.00 dual
Model 4710 is a digitally -controlled 1/6 octave 55 band equalizer in one rack space. Controllable from the front panel with password
protection; has 10 memory locations and 10 separate preset locations in non -volatile storage.
Dimensions: 1.75 in. X 19 in. X 12 in.
Weight: 9 lbs.
Price: $1,550.00
The Model 4650/4660 is a 60 mm slider controlled 1/3 octave filters 31.5 Hz -16 kHz; ±12 dB, 10 dB gain; variable high/low pass on 4660;
XLR and 1/4 jack connectors; input/output transformer available (4622).
Dimensions: 3.5 in. X 19 in. X 5 in.
Weight: 7 lbs.
Prices: 4650 is $699.00/4660 is $750.00
Model 4675 is a 60mm slider controlled stereo 2/3 octave; filters 40 Hz -16 kHz ±12 dB range, 10 dB gain; variable high pass, fixed low pass;
XLR connections; servo -balanced differential input/output circuit.
Dimensions: 3.5 in. X 19 in. X 5 in.
Weight: 7 lbs.
Price: $795.00
Model 4400 has L -C active 1/3 octave filters 31.5 Hz -16 kHz; ±10 dB range; variable high/low pass; 3 outputs and crossover socket for
optional bi -amp /tri -amp operation; input/output transformers available; noise -90 dBu worst case.
Dimensions: 3.5 in. X 19 in. X 8 in.
Weight: 15 lbs.
Price: $1,050.00
The Model 4500 is R -C active 1/3 octave filters 31.5 Hz -16 kHz ±10 dB range; variable high/low pass; 3 outputs and crossover socket for
optional bi- amp/tri -amp operation; input/output transformers available; noise -80 dBu worst case.
Dimensions: 3.5 in. X 19 in. X 5 in.
Weight: 7 lbs.
Price: $790.00
Model 4100A has L -C active stereo octave band 31.5 Hz -16 kHz ±10 dB range; variable high pass; fixed low pass; bi -amp available;
input/output isolation transformer available; noise -92 dBu worst case; L.A. approved.
Dimensions: 3.5 in .X 19 in. X 5 in.
Weight: 11 lbs.
Price: $975.00
The DSP 5000 has 12 bands of parametric equalization, digital crossover and delay all in one rack space; 19 bit, user configurable single
channel in, 4 out; remote control capability via PA -422, MIDI or contact closures.
Dimensions: 1.75 in. X 19 in. X 12 in.
Weight: 9 lbs.
Price: $3,400.00
The SGX -2000, Model 500, is for guitar. Tri- channel programmable tube and solid state preamp with stereo digital effects; full 20 kHz
bandwidth; 24 bit processing; seven band equalizer.
Dimensions: 3 in. X 19 in X 9 in.
Weight: 15 lbs.
Price: $829.00
The SGX NightBass, Model 490, is for bass guitar. Tri- channel programmable tube and solid state preamp with stereo digital effects, full 20
kHz bandwidth; 24 bit processing; seven band equalizer and selectable crossover.
Dimensions: 3 in. X 19 in X 9 in.
Weight: 15 lbs.
Price: $839.00
The Power Plant, Model 410, is a dual channel guitar preamp. Channels are switchable between clean and dirty with their own separate EQ
effects loop; separate guitar, line and power amp and headphone outputs.
Dimensions: 1.5 in. X 19 in X 10 in.
Weight: 11 lbs.
Price: $329.00
The DigiTech DSP 256XL Digital Effects Processor features 21 different studio -quality effects, up to 4 simultaneously, and is built tough
enough for road use.
Dimensions: 1.75 in. X 19 in. X 8.5 in.
Weight: 5.5 lbs.
Price: $439.95
The DigiTech DSP 16 Effects Processor contains 128 MIDI changeable programs utilizing 16 different reverb and delay effects; a 3 -band EQ
provides tailoring of the sound.
Dimensions: 1.75 in. X 19 in. X 8.5 in.
Weight: 4.5 lbs.
Price: $299.95
The ECC is a digital delay system with microplate reverb. Delay and reverb may be used simultaneously or independently; delay range is
ms to s; effects include doubling, chorus, flange, plate reverb with delay, acoustic chamber and tremolo.
Dimensions: 1.75 in. X 19 in. X 7.5 in.
Price: $995.00
The Ultraverb II is a digital multi- effects processor with 15 kHz bandwidth; 256 internal programs; 128 user editable effects; real time MIDI
control; each preset transfer step reversible up to last keystroke; full MIDI access.
Dimensions: 1.75 in. X 19 in. X 6.5 in.
Weight: 6 lbs.
Price: $349.99
All effects of the AddVerb II, except reverbs and specials, may be modified and stored at any of 199 program presets; full MIDI control
capability; 50 reverb presets; 40 programmable delay/echo and modulated presets; 10 combinations; presets may be mapped to any of 128
MIDI program numbers.
Dimensions: 1.75 in. X 19 in. X 8 in.
Weight: 6 lbs.
Price: $319.00
The Multifex contains four 16 -bit, digital, multi- effects modules in one 19 in. rack mount package. Each module delivers user-adjustable echo;
pre-delay; early reflections; room size; tonal color; reverb time; left and right stereo channel delay; left and right stereo echo feedback; chorus
rate; depth; delay time; feedback and multi- effects algorithm facilities.
Dimensions: 1.75 in. X 19 in. X 9.125 in.
Weight: 7 lbs.
Price: $1,099.99
ProFex is a programmable MIDI controlled multi-effects preamp featuring digital stereo multi -effects processor; switchable for line level input
of instrument level; independent effect blocks can be combined in series or parallel in any order to form multi -effect chains; each effect block
has independent mix and level control; programmable noise gate in all programs; 128 presets mapped to 128 programs for front panel, MIDI
or footswitch access.
Price: $799.99
The DSR 1000 is a MIDI capable digital stereo reverb/multi -effect processor; six powerful multi-effect algorithms; all effects re- mappable; full
suite of echo /chorus /reverb facilities; 16 -bit processing.
Price: $349.00
The QFX is a 4- channel digital multi-effects processor with full MIDI implementation; 16 -bit processing; stereo /mono; I.U. 19 in. rack
package; re- mappable effects positioning; up to 2.75 seconds of digital delay available.
Price: $1,099.00
The Quadraverb Plus features 1.5 seconds of delay memory for sampling; independently adjustable multi-tap delays; programmable panning;
new ring moduator and resonator configuration along with the 20 K bandwidth reverb; delay; chorus; flanging; parametric EQ; leslie simulator;
and comprehensive onboard digital effects mixing system of the original Quadraverb.
Price: $499.00
The Microverb Ill, a 16 -bit stereo digital reverb and effects processor, has 256 preset programs: 112 reverbs; 32 gated /reverse reverbs; 80
delays; and 32 multi-tap and effects programs. The 19 in. rack mountable unit features 15 kHz bandwidth and two bands of EQ (100 Hz and
4 kHz) for fine tuning of programs.
Price: $249.00
The Midiverb Ill is a digital stereo multi- effects unit capable of generating four effects at a time: delay; reverb; and chorus or flange. Features
200 memory locations, with 100 reserved for factory presets. Real -time MIDI control.
Price: $399.00
The Multiverb ALPHA, Model 470, is a 24 bit full 20 kHz digital signal processor capable of combining seven effects at once. Has
programmable seven band EQ; reverb; two octave of Pitch Transposing; 20 delay types including sampling.
Dimensions: 1.5 in. X 19 in. X 9.25 in.
Weight: 11 lbs.
Price: $499.00
The Multiverb LT, Model 420, is a studio digital effects signal processor with instant access to 192 pre -programmed presets of up to three
effects at once. Effects include reverb; delay; chorus; flanging; gated and reverse reverb and panning.
Dimensions: 1.5 in. X 19 in. X 9.25 in.
Weight: 10 lbs.
Price: $299.00
EVENTIDE, INC. -See our ad on page 5
The H3000KS Kitchen Sink Ultra- Harmonizer has all SE and B features plus Vai Presets and HS322 Internal Sampler Board (23.71 seconds
mono/11.35 seconds stereo sampling, pitch shifting and time compression /expansion, more).
Dimensions: 3.5 in. X 19 in. X 13.5 in.
Weight: 13 lbs.
Price: $4,590.00
The H3000SE Studio Enhanced Ultra- Harmonizer has 19 algorithms, including vocoder; dense room; multishift; band delay; string modeller;
phaser, stutter and patch factory; 200 presets; function generator (programmable parameter modulation); soft functions (user -definable Soft
Dimensions: 3.5 in. X 19 in. X 13.5 in.
Weight: 13 lbs.
Price: $2,995.00
The H3000B Broadcast/Post Ultra- Harmonizer has 14 algorithms, including TimeSqueeze (stereo time compression /expansion with machine
control); stutter and patch factory (white noise generator, filters, pitch shifters, delay lines and more); 80 presets; function generator, soft
Dimensions: 3.5 in. X 19 in. X 13.5 in.
Weight: 13 lbs.
Price: $2,995.00
The H3000S Studio Ultra- Harmonizer has 11 algorithms including diatonic shift; dual shift; layered shift; stereo shift reverse shift; swept
combs; reverb factory; ultra -tap; dual digiplex; long digiplex; 48 Steve Vai presets; 58 factory presets.
Dimensions: 3.5 in. X 19 in. X 13.5 in.
Weight: 13 lbs.
Price: $2,495.00
The DN780 offers full control over several parameters including predelay time; level and pattern of reflections; low and high frequency decay
times; and room size. Supplied with remote controller; has 50 non -volatile user memories; 32 bit VLSI circuitry.
Dimensions: 3.5 in. X 19 in. X 12.25 in.
Weight: 16.5 lbs.
Price: $2,865.00
The 300 Digital Effects System is designed for the small professional studio. Features include two stereo inputs /outputs (balanced XLR) and
digital inputs /outputs in the AES/EBU and SPDIF formats. The 300 features 50 event effects recall via SMPTE time code; full MIDI
implementation; and 96 dB signal -to -noise ratio; and reverb; ambiance; stereo pitch shifting and mastering type algorithms.
Dimensions: 3.5 in. X 19 in. X 13.6 in.
Weight: 18.9 lbs.
Price: $4,795.00
The RCC reverb control center is a complete microplate reverb system for use with or
2 additional stereo sources; and output for a tape recorder, plus 3 -band equalization.
Dimensions: 1.75 in. X 19 in. X 7.5 in.
without a mixing board.
has 2 mic inputs; inputs for
Weight: 7 lbs.
Price: $695.00
The Univerb Il has 128 stereo 16 -bit effects; bandwidth of 20 Hz to 12 kHz; VLSI technology; remote bypass capability; stereo and mono to
stereo capability; single rack space chassis.
Dimensions: 1.75 in. X 19 in. X 8.125 in.
Weight: 5 lbs.
Price: $249.99
The R -880 digital reverb has four independent DSPs; reverb; ncn- linear reverb; early reflections; chorus; delay; EQ; compression; flat
frequency response; 90 dB dynamic range; analog, AES/EBU digital I/O connections; accommodates 48 kHz, 44.1 kHz signals.
Dimensions: 3.56 in. X 19.18 in. X 16.56 in.
Weight: 22 lbs.
Price: $3,995.00
The GC -8 is a graphic controller remote control unit for the R -880 featuring large, 256 X 64 dot LCD; five rotary knobs and numeric keypad
for easy programming; memory card slot for storing and loading programs.
Dimensions: 2 in. X 13.125 in. X 6.94
Weight: 2 lbs., 10 oz.
Price: $850.00
Alesis Studio Electronics
3630 Holdrege Avenue
Los Angeles, CA 90016
Electro-Voice, Inc.
600 Cecil Street
Peavey Electronics
Buchanan, MI 49107
711 A Street
Ashly Audio, Inc.
Eventide, Inc.
100 Fernwood Avenue
Rochester, NY 14621
One Alsan Way
Lisle Ferry, NJ 07643
Altec Lansing Corporation
10500 West Reno Avenue
P.O. Box 26105
Oklahoma City, OK 73126
an Sound, Inc.
30 ich Street
G enbrae, CA 94904
Applied Research and
215 Tremont Street
Rochester, NY 14608
ARX Systems
28271 Bond Way
Silverado, CA 92676
979 Franklin Lane
Maple Glen, PA 19002
dbx Professional Products,
a division of AKG
Acoustics, Inc.
1525 Alvarado Street
k- Teknik Electronics,
20 Sea Lane
Fa mingdale, NY 11735
icon, Inc.
10 Beaver
W tham, MA 02154
10802 47th Avenue West
Everett, WA 98204
Roland Pro Audio/Video
7200 Dominion Circle
Los Angeles, CA 90040
Sabine Musical
Manufacturing Company,
4637 Northwest 6th Street
Gainesville, FL 32609
Sound Concepts Inc.
Post Office Box 135
Brookline, MA 02146
Or tan,
a division of AKG
Ac rustics, Inc.
15' 5 Alvarado Street
2200 South Ritchey
Santa Ana, CA 92705
Sa Leandro, CA 94577
P.O. Box 1678
Los Gatos, CA 95031
Ox ¡poor
Parkway Office Circle
ingham, AL 35244
DOD Electronics
Pa iasonic Pro Audio
Sy tems
5639 South Riley Lane
Salt Lake City, UT 84107
Rane Corporation
L7 Sound
79 0 LT Parkway
Lii Lonia, GA 30058
San Leandro, CA 94577
Meridian, MS 39301
Katella Avenue
Cy ress, CA 90630
65! )
Summit Audio, Inc.
Symetrix, Inc.
4211 24th Avenue West
Seattle, WA 98199
White Instruments
1514 Ed Bluestein Boulevard
Austin, TX 78721
LINE ........................
A Broadcast Audio Question and Answer to Randy Hoffner
Dear Mr. Hoffner:
First of all, I would like to thank
you for the enlightening articles
you have in db Magazine under
the Broadcast Audio column, especially "Multichannel Sound
Around the World" which ran in the
November/December 1989 issue.
Since I am at a radio and television
broadcasting station here in Singapore, the articles are particularly
relevant to me and my colleagues.
However, here is a query that I
would be much obliged if you could
shed some light on. We had viewers
comment that our station is not
"punchier" or "brighter" sounding,
and that it has a narrower stereo
spread than our neighboring stations. In house, we have two
schools of thought to tackle this
problem. One group says that including an exciter and/or compressor like the Aphex Aural exciter
and the Compellor between the
studio and transmitter link is the
solution. The other group says that
there should be no processors in the
above link (at the most, a limiter to
prevent an overload), but individual sources/programs (sound balancers or operators in the various
areas) should do their own "brightening," compatible level control
and imaging. This means a flat unenhanced broadcast chain that is
compatible for various types of pro-
grams. What is the conventional or
modern broadcast practice?
Another question related to this
problem developed. Since we
adopted the NICAM 728 stereo system, we have had a mixture of both
mono and stereo programs. However, a marked loudness difference
between stereo and mono programs was apparent for those
viewers having mono sets (non NICAM receivers) and was more than
accountable due to "center channel
buildup" when stereo was summed
into mono and in the reverse, a general drop in loudness when stereo
was heard on a mono set. It is very
noticeable when commercials of
both varieties have to be aired alternately in between breaks.
I hope you can share some views
and provide some guidance in solving these problems (which I hope
have been presented clearly).
Jibby Jacob
The Reply From Randy
Dear Mr. Jacob:
Thank you for the kind words
about my db Magazine articles.
Of course, compliments are always
appreciated, but it is nice to hear
that my writing is found useful by
those in the business. Let me now
address your two questions. Please
remember that a lot of what is said
below falls into the category of personal opinion.
Your first question raises a number of issues, some of which are perennial topics of discussion among
radio and television audio engineers. The issue of "punch" or
"brightness" in the on -air sound is
distinct from stereo spread or separation. It can be dissected into two
components. You have raised the
question of which type of broadcast
chain is preferable: a "flat," unenhanced chain or one that incorporates processing. In the abstract, of
course, that depends upon the philosophy of the individual broadcaster. In practice, there is advantage to including processing in the
broadcast chain. Abig advantage in
my opinion is that the broadcast
station is much better able to maintain a consistent on -air sound. If all
sweetening and processing is done
by the individual balancers or operators, every program will have a
different "sound," depending on the
taste of the individual sound balancer. In addition, of course, it will
be much more difficult to assure
consistent audio levels from program to program, and from commercial to commercial. It is also
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true that you would be hard pressed to find a television or radio
broadcast station in the United
States that does not do some processing in the broadcast chain.
You have not said as much, but if
your studio -to- transmitter link is a
microwave system, or any system
using FM subcarriers to carry the
audio signals, it requires at the
very least a protection limiter, one
that is sensitive to the pre- emphasis curve the system employs, to
prevent overdeviation of the FM
transmission channels and consequent high frequency distortion.
Even if you are not using FM sub carriers in your studio-to-transmitter links, and even though you are
transmitting stereo digitally with
the NICAM system, your transmitted monophonic signal (the one
that most of your viewers are hearing) is itself carried on an FM
transmission channel.
While United States television
aural transmitters and studio -totransmitter links use a 75 microsecond pre- emphasis network, producing a boost of 16dB at 15 kHz,
your equipment may use 50 microsecond pre- emphasis, but this only
reduces the severity of the problems that pre- emphasis can generate and does not eliminate them.
Without a frequency- sensitive limiter in front of FM subcarrier
stages to protect them from overdeviation, the overall audio level to
their inputs must be reduced so
dramatically that the resultant
signal -to -noise ratio will be compromised.
The use of a compressor in conjunction with the broadcast limiter
will serve two functions: it will increase the "punch" of your on -air
sound and raise the average audio
level over the inherent noise floor
of your studio-transmitter link and
your monophonic aural transmitter. The choice of which compressor
to use is, of course, a matter of
taste. You have mentioned the
Compellor, which is one of a number of high quality devices available on the market.
If you wish to add brightness to
your on -air sound, you would be
better advised to use one of the very
good split -band audio processors
available and increase the high end balance by judicious adjust-
ment ofits control, keeping in mind
what effect pre -emphasis will have
on the high-frequency content of
your audio program material. To
put it in a nutshell, I recommend
the use of a high -quality broadcast
compressor/limiter combination,
set up for your pre -emphasis curve,
to be placed ahead of the studio -totransmitter link in your broadcast
audio chain.
There should be nothing in your
broadcast audio chain that would
compromise the stereo separation,
which leaves stereo "spread" a
function of the audio balancer.
Some United States television and
FM radio stereo encoders, because
they operate in the sum- and -difference domain, have circuitry that
increases stereo spread by manipulating the amplitude of the stereo
difference signal, but NICAM does
not work this way, so that option is
not readily available to you.
In response to your second question, the issue of the relative loudness difference between a mono
signal and the mono sum of a stereo
signal arose when we were contemplating the conversion of the NBC
Television Network to stereo operation. This within itself is not really a big problem, as I hope to
show. There may, however, be other
problems with other causes that
generate the phenomena you are
Let us assume that we have two
audio paths, which we will call A
and B. IfA andB have identical signals (from the same source) at
equal level of 0 dB on them, these
two signals are perfectly correlated. This would correspond to a
split mono signal. Avery simple example is a sine wave from a single
oscillator connected to both paths.
If these two signals are summed,
the resultant signal has an amplitude twice that of either of its components. Stated differently, ifA and
B each have an in -phase (correlated) 0 dB signal on them, the sum
of A and B is a signal of +6 dB. If A
and B have totally uncorrelated
signals corresponding to maximum
stereo separation of amplitude 0
dB on them, their sum will be +3
dB. For typical stereo television
program material, the degree of
correlation between channels varies from complete correlation to
tal lack of correlation. The worstcase difference between two-channel mono and separated stereo is 3
dB, and the typical difference is in
fact around 1 -2 dB.
When we converted the NBC
Network to stereo, the transition
was made this way. On a particular
day, we started feeding two -channel audio, either stereo or two channel mono, on two audio sub carriers of our satellite system. By
the time that day arrived, all of our
mono affiliates (at that time, over
150 stations) had to have put in
place a summing network to combine the two audio signals, or else
they would be airing only one channel of any stereo program. We had
no significant problems with mono
versus stereo levels, and that has
maintained from 1985 until today.
You have not mentioned how
your mono audio signal gets fed to
its transmitter. At least some NICAM encoders furnish a mono sum
audio output for feeding the mono
aural transmitter. If this is the way
our mono transmitter is fed, and if
the level of that signal into your
aural transmitter is set correctly,
no substantial difference should be
noted between stereo and two channel mono. Feeding the mono
transmitter in this way can complicate things, however, because of
the differing limiter requirements
of NICAM and FM audio transmission. Other ways of feeding the
mono transmitter are possible, and
some could cause level problems if
not properly set up. For instance, if
there is some scheme whereby a
switch is made between single channel mono and stereo sum, the
relative levels of those two sources
could be disparate. It is also possible that if extreme separation is
present in stereo segments, this
could result in noticeable loudness
differences between mono and stereo material.
I hope that my views on the questions you have asked will give you
some ideas on how to go about solyingyour audio problems. I would be
happy to continue this dialogue if
you find it useful. I wish you good
fortune in your work, and again, I
thank you for your kind words.
Randy Hoffner
NBC -New York
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Construction is complete for the
University of Utah's Dolores
Dore' Eccles Broadcast Center.
Facility tours of the complex will
include visits to the 3,500 sq. ft.
television studio for public television station KUED which encompasses two floors, the Audio Post
Production Suite, the open plan
VTR/Rack Room, control rooms for
radio and television, production
rooms and suites for radio, television and future Student Broadcast
activities, and administrative
spaces located on both levels.
Unique to the second floor is a
viewing window located in the corridor and the full -length window located in the multi -media room both
providing unobstructed views into
the KUED studio.
Sennheiser Electronic Corporation was awarded the "ARTIST Stage Design Prize" at the recent MUSIKMESSE Fair in
Frankfurt, Germany, for their
SEMS Microphone stand. Special
emphasis was placed on the ergonomic and operation features of the
mic when it was designed.
After being destroyed by fire in
July, 1990, the Music Annex's
Studio C is back in operation. Studio C features a new 56-input
Soundcraft 3200 console, with
Discmix II automation, a Studer
A827 24 -track recorder, Otani
MX15 2- track, UREI 813C time
align monitors and a full complement of outboard gear, including a
Lexicon 480L and an Eventide
HD3000 SE. The Music Annex's recording studios are located in
Menlo Park, CA, _Also adding
equipment to its studios is Ron
Rose Productions in Southfield,
MI. AudioFile II Plus, which is the
latest generation of hard disc based digital audio recording and
editing equipment, has been installed in Ron Rose Productions'
Miami Vice studio. The equipment's internal processing unit is a
transputer -the same space -age
technology used in F-15 fighter
planes -which processes incredible amounts of data in split seconds. Ron Rose Productions is the
first company in the metro Detroit
area to have this new, updated version...The first multi -console order
for Korea has been delivered to
Seoul Broadcasting Systems in
South Korea. Eight Harrison PRO790s, which range from twelve to
twenty -four input channels, have
been installed in SBS's new radio
facilities in Seoul...Nutmeg Recording in New York City has installed Solid State Logic's Screen Sound digital audio- for -video
editing/mixing system for work on
its post production projects. The
four -studio facility features two 24track video interlock rooms, a third
MIDI room and a fourth room designed around the ScreenSound
system...A Versadyne 1500 Series
high -speed tape duplication system has been delivered to Preci-
sion Sound Corporation of
Burnaby, British Columbia, Canada. The new system will be used
primarily for music duplication.
Other Precision Sound purchases
include an Otani MTR -12 mastering deck, Dolby encoders, Versadyne SR-150 slave reader, VerPT-250
totalizer, and an assortment of test
A new AutoCAD- compatible
acoustic design program entitled
CART (Computerized Acoustic
Ray Tracing) has been developed
by John Storyk in a collaborative
effort with Walters-Storyk inhouse CAD consultant Marcolm
Young. CART is an automated
process which calculates and
graphically displays acoustic ray
behavior -specifically, how sound
rays of varying frequencies bounce,
reflect and re- radiate against the
interior surfaces of any given
space. The CART program, in conjunction with the TEF System -20
Sound Lab audio analyzer, has contributed to design programs for
Storyk projects including Studio 9
at Howard Schwartz Recording in
New York City, and new facilities
for JSM Music and Sound Shop.
Gene Nyland, vice president of
Operations at Ampex Recording
Media Corporation's Opelika,
AL, manufacturing facility, retired
Aug. 30 with twenty -seven years of
service to Ampex...Ronald Remschel has been appointed marketing manager, Professional Audio
Products, Sony Business and
Professional Group...As part of
the reorganization of its research
and development staff, RenkusHeinz Inc. has appointed Frank
E. Ostrander as chief engineer...John Bolstetter has been
named a vice president of Mark IV
Audio, Inc., and Al Watson has
recently been named vice president
of engineering at Electro- Voice,
Inc., one of seven companies in the
Mark IV Audio Group...Charles
Meyer has been promoted to vice
president of engineering at NVi-
sion, Inc.
He will assume full responsibility
for all research and development
for NVision's line of digital
audio/video distribution and trans-
mission equipment.
-21 -18 -15 -12
+3 +6 +9 +12 +15 +18 +2: +2-
ms 4 sec.
-50dBm -'20dBm
The MDC 2001 is a 'evolutionary new product
that can improve performance in any audio
application in which it is used.
essential in digital mastering. And because the
MDC is fully stereo, all the left/right balance
stays intact.
In live sound, the compressor can smoothly and
transparently control the musical dynamics. The
gate and expancer can totally eliminate stage
noise, gate off microphones for higher gain
before feedback and eliminate drum and amp
bleed. The de -esser will knock out excessive
sibilance protecting your drivers and balancing
the material. The all new exciter circuit brings out
the upper harmonics for sparkling clarity, even at
high volumes. The final peak limiter can be set to
the maximum input voltage of the amps to
prevent your system from ever being overdriven.
MJC can smooth out the
dynamics of voices and instruments, eliminate
background noise for much cleaner recordings
and remove harsh "s" sounds before they
saturate the tape. It can also add back in all the
natural harmonic brilliance that is lost for
recordings with crystal clarity. The peak limiter
will prevent the ±ape from overloading and is
churches, theaters and fixed installations, the
MDC is the one piece of gear that will provide
level protection, noise gating, de- essing, signal
enhancement and dynamic control in full stereo
in a single rack space. All functions are independently controllable and may be taken out of the
chain if desired. Ducking, keying, and linking
jacks are provided, and the audio performance of
each individual processing circuit is world class.
In the studio, the
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Switchable detector loop
Stereo gate key input
Full stereo processing
Threshold activation LEDs for all functions
Parasweep integrated controls simplify
Actual output monitoring limiting circuit
Unsurpassed audio specifications
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COMP ^Fs,c1x
Dynamics sensitive gating control
Unequaled audio performance
Individual isolated processing circuits
21 LED gain control metering
Switciable input and output metering
Balanced XLR inputs /outputs
Tip /ring /sleeve ba anced inputs /outputs
Limiter link /voltage. control jack
Stereo auto -detect circuit
"I've been sold on Beta's superiority since I first tried them.
I use them on vocals, drums, amps, and brass because their
sensitivity and resistance to feedback make them the
perfect fit for the groups I work with. And the Beta 58 Wireless
is the first system I've found that gives my artists the freedom
of a radio mic without sacrificing sound quality."
Paul Dalen, Sound Engineer or David Sanborn and Lisa Stansfield.
Shure Beta Microphones.
Buy Them On Word Of Mouth Alone.
Before you select a microphone, listen to the leading pros who use the Shure Beta Series on stage. They'll tell you about
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