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THE SOUND ENGINEERING MAGAZINE
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ELI
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£.6IJ60dlOZl1V
SEPTEMBER 1981
$1.95
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The Model 6590
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Two totally independent plug -in chanovabk from the front panel,
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cm-Align is a Trademark of E.M. Long Assoc Oaklanc. CA
referenced Trademarks are property of. or licensed by
Un led Recording Electronics Industries, a URC Compan
.
AU
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From One Pro To Another
United Recording Electronics Industries
8460 San Fernardo Road. Sun Valley. California 9'352 1213) 767 -1000 Telex- 65 -1389 UREI SNVv
Worldwide. Gotham Export Corporation, New York. Canada. Gould Marketing. Montreal H471E5
See your professional audio products
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i.. El 1-
Publisher
Larry Zide
Editor
John M. Woram
Associate Editor
Mark B. Waldstein
Advertising Production
THE SOUND ENGINEERING MAGAZINE
Layout
Kathy Lee
&
SEPTEMBER 1981 VOLUME 15, NO. 9
Classified Advertising
Carol Vitelli
Book Sales
Lydia Calogrides
Circulation Manager
DEPARTMENTS
Eloise Beach
Art Director
CALENDAR
EDITORIAL
CLASSIFIED
Bob Laurie
6
25
62
Graphics
K&S Graphics
Typography
Spartan Phototype Co.
DIGITAL AUDIO
Barry Blesser
8
THEORY AND PRACTICE
sales offices
Roy McDonald Associates, Inc.
Dallas, Texas 75237
First Continental Bank Bldg.
5801 Marvin D. Love Freeway
Suite 303
(214) 941 -4461
Denver, Colorado Area
Englewood, Colorado 80112
14 Inverness Dr. East
Bldg. 1- Penthouse
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Houston, Texas 77036
6901 Corporate Drive
Suite 210
(713) 988 -5005
Los Angeles Area
Glendale, California 91204
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Portland Area
Hillsboro, Oregon 97123
510 South First
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(503) 640-2011
San Francisco Area
Emeryville, California 94608
265 Baybridge Office Plaza
5801 Christie Avenue
(415) 653 -2122
Sagamore Publishing Co., Inc.
New York
Plainview, NY 11803
1120 Old Country Rd
(516) 433-6530
Norman H. Crowhurst
16
SOUND WITH IMAGES
Len Feldman
22
NEW PRODUCTS AND SERVICES
55
PEOPLE, PLACES, HAPPENINGS
64
FEATURES
BROADCAST AUDIO -THE INCREASED NEED
FOR AUDIO PROCESSING
George Berger and Robert L. Deitsch
26
BROADCAST AUDIO PRODUCTION CONSOLES
Thomas Mintner
32
DYNAMIC PROCESSING -PART lI
Nigel Branwell
42
A COMPUTERIZED CONSOLE FOR
RADIO BROADCASTING
Vladimir Nikanorov
48
db APPLICATION NOTE:
AN ARTIFICIAL MICROPHONE
A
Mel Sprinkle
52
ABOUT THE COVER
Seated at the controls of the WABC
Production Control Room in New York
City is Debra lacovelli, staff engineer,
WABC /WPLJ Radio.
db. the tinund t ngmeenng Magaiuc 11titi\ 0011,145i r, published monthly h. tiagamore Publishing Company. Inc. I non: contents ram right , 1981 bS tiagamore I'ubinhmg(lr. 112001d I domo Road. Plains re..
't 1811. lekphune 15161433 6 510
db r. pufihslted for those indi rdual, und 'tons in prore.aunal audio-recording. broadcast. audio -usual. sound ramlineemenl. consultants sided recording. him sound ere Spphcanon should he made on the subscription lui ni in the rear u. each issue Subset 1pIuns arc S15 00 per sear 1528.00 per rear outside .ti. Possessions. 516181 per sear l- anadar in I S. ronds Single copes are SI 95
each I.dmtnal. Publishing and Sale. (blue, 1120 Old uunbs Road. Plains res. Sea or). 11190 Cuntrolkd eueular um postage
paid at Plains es. \1 118t1í and an additional marling orrice.
1
I
l
I.\
1
Calendar
No you
ear it...
Index of
Advertisers
Ampex
Atlas Sound
Audio -Technica
Cover Ill
45
11
Bose
SEPTEMBER
The Society of Broadcast Engineers 9th Annual Central New
York Regional Convention. Syracuse Hilton Inn. For more information contact: Convention Chairman Hugh Cleland, WCNY TV/
FM. Liverpool, NY 13088. Tel:
-15
Mc
(315) 457-0040.
del 105
OCTOBER
Notiir
don't.
The Garner Audio Tape Degaussers are truly the most
flexible and thorough audio
tape erasers available. With them,
you can erase cassettes, reels,
and cartridges in just four
seconds. And, we guarantee
that your erasure wil meet the
Natural Stereo Techniques for
Recording Music Workshop. University of Wisconsin at Eau Claire.
For more information contact:
Burton Spangler, Audio Coordinator, Media Development Center, UW -Eau Claire, WI 54701.
7 -9
Tel: (715) 836-2651.
The 11th Conference of the Western Educational Society for Tele-
13 -15
communications. Harrah's, Reno,
Nevada. For more information
contact: Dr. Donald Price, Media
Production Services, California
State University, Los Angeles,
most stringent recording
standards. It really is a case of
"Now you hear it...
Now you don't."
Cetec Gauss
Cetec Vega
Crown
dbx
Electro -Voice
Fitzco Sound
Garner
J BL
Magnefax
Milab
Orban
Otari
Panasonic
Polyline
Pro Audio Systems
Rauland -Borg
Sescom
Shure
Sony
Sound Technology
Soundcraftsmen
Standard Tape Lab
Studer Revox
Telex
Telex Turner
Tentel
TOA
Tweed Audio
UREI
White Instruments
Yale Radio Electronics
9
59
8
39.41
10
17
61
6
33 -36
14
Cover
23
24
IV
30, 31
20
55
56
60
7
10. 60
13
22
46
29
21
19
47
I5
12
Cover II
18
16
CA 90032. Tel: (213) 224 -3396.
123rd SMPTE Technical
25 -30 The
Conference Exhibit. Century
Plaza Hotel. Los Angeles. For
more info contact: SMPTE, 862
Scarsdale Ave., Scarsdale, NY
10583. Tel: (914) 472-6606
NOVEMBER
12 -15
GARNER INDUSTRIES, INC.
4200 No. 48th St.. Lincoln. Nebraska 68504
Phone (402) 464 -5911
There's more
hear...
to write
7040.
25 -27
Garner Industries for
more information:
Just
NAME
TITLE /POSITION
Billboard Magazine's 3rd Annual
International Video Entertainment /Music Conference. Beverly
Hills Hilton, Los Angeles. For
more information contact: Billboard Magazine's Conference Bureau, 9000 Sunset Blvd.. Los Angeles, CA 90069. Tel: (213) 273-
Prosound '81 Professional Sound
Equipment Exhibition. West Centre Hotel, London. For more information contact: Batiste Exhibitions & Promotions, Pembroke
House. Campsbourne Road. London N8. Tel: 01 -340 3291.
COMPANY
ADDRESS
CITY
L
STATE
-- - - - --
ZIP
-.II
Circle 27 on Reader Service Card
www.americanradiohistory.com
Coming
Next
Month
In October, the British are coming!
And Paul Revere will not be riding out
to warn anyone! A sixteen -page editorial
supplement will cover various aspects of
what is happening in pro audio on the
other side of the pond. This supplement
appears in addition to all our regular features. Articles on Architectural Acoustics is our subject for the month. Michael
Rettinger is just one of the featured authors. And there will be our new regular
roster of columnists -Barry Blesser,
Norman H. Crowhurst, and Leonard
Feldman. All this and more coming in
October in db -The Sound Engineering
Magazine.
Twice again: Shure sets
the standard for the industry!
Introducing two new microphone mixers
Ten years ago -with the introduction of the
M67 and M68 -Shure set the standards of
the industry for compact, portable micro-
M267
For Professional Broadcasting
Both TV and Radio -in the studio and for
remote broadcast applications.
phone mixers.. Shure is now introducing two
new mixers with features and improvements
that will make them the new industry standards.
e
M268
For Public Address and Paging
hotels, schools, churches, community
centers, f-ospitals. etc.
In
For the Serious Tape Recording
For Professional Recording
Enthusiast
For Professional Sound
As an Add -On Mixer for
Reinforcement
Expanding Current Equipment
With all these new features:
For more complex public address
systems.
With all these new features:
Switchable, fast -attack limiter
LED peak indicator
All inputs switchable for mic or line
Simplex power
Greater headphone power
Built -in battery supply
Lower noise
Reduced distortion
...and all of the famous M67 original featires
Lower noise
Dramatic reduct on in distortion
Mix bus
Automatic muting circuit
Simplex power
...and all of the famous M68 original features.
.
Both new models inc,ude the same ruggedness
and reliability that have made the M67 and M68
the top -selling mixers in the industry.
For complete information on the M267and M268 send in for a detailed product brochure (ask for AL669).
The Sound of the Professionals'
I--IURE
Shure Brothers, Inc., 222 Hartrey Avenue, Evanston, IL 60204
In Canada: A. C. Simmonds & Sons Lim ted
Manufacturers of high fidelity components, microphones, sound systems and related circuitry.
(lrele
36 on Reader Service Card
BARRY BLESSER
Digital Audio
Delta Modulation:
The Poor Man's Converters
A delta modulation A /-D and D/A
system is just about the cheapest way to
digitize audio since the parts cost can be
as low as 70 cents. If such a system could
be made to be high quality, digital audio
would be cheap, easy, and close to "apple
pie." Before showing you its defects, let's
consider the way that it works. FIGURE I
shows a simplified delta modulation
system. The comparator compares the
audio input with an internal approximation to it, called al(t). We also note
that the decoder has an identical circuit
which generates a2(t).
Since at(t) and a2(t) are both created
from the digital output bit from the flipflop, they will be identical if the circuit
component values are identical. For the
moment, assume this to be the case. The
actual conversion is performed by the
-bit comparator which determines if the
approximation at(i) is larger or smaller
than the actual audio input. If the audio
input is larger, the comparator output
will be an H (high = 5 volts). When the
clock comes along, this information is
transferred to the flip- flop's output. This
is the I -bit digital signal which represents the sign of the difference between
the input and the approximation. When
the difference is positive, the flip- flop's
output is negative, causing the integrator's output to ramp in the positive
direction (the integrator inverts). This
brings the approximation closer to the
input.
If the audio input is below the approximation a1(1), then the comparator output
would be an L. The flip -flop output is
now positive, and the integrator ramps
downwards. We see that this is a kind of
feedback loop in which the digital word
tries to keep the error (i.e., the difference
between the approximation and the
input) as small as possible. Unlike
analog negative feedback, the error
cannot be driven to 0 since the integrator's values are quantized. Since
the +V (or -V) will be applied to the
integrator for a full clock cycle, the
integrator will be more a fixed step
downwards (or upwards). For example,
with a MHz clock, the integration will
last for I psec. In FIGURE I, consider that
V = I volt, C = 10 nF, and R = 10 kohms.
During a I µsec interval, the integrator
can move exactly 10 millivolts upwards
or downward. We say that the change in
the integrator is quantized in steps of
I
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millivolts, e.g. the effective LSB size.
The error is also the "noise" in the
output signal, since the two approximations a1(i) and a2(t) are, in principle, the
same. For the remaining part of the discussion, we will treat these two signals
10
equivalently.
www.americanradiohistory.com
shows the
Figure 1. Simplified delta modulation
encoder and decoder.
I bit /sample
digital output
ENCODER
I
Circle 25 on Reader Service Card
FIGURE 2
response of at(t) to a ramp -like audio
input. In the region where the input is
constant, the output oscillates about that
value. The integrator ramps upwards
for one cycle and determines that it is
too high; it then ramps downwards and
determines that it is too low. This kind
of behavior is called hunting, or limit cycle oscillation, and is based on the step
size. It is directly analogous to the LSB
quantization noise.
In the ramp part of the input, the
approximation attempts to track the
audio input signal, but the fastest it
can go is still too slow. Even with the
integrator stepping upwards every clock
cycle, the error becomes larger and
larger. When the audio input levels off,
the approximation eventually does catch
up. This kind of error is called slew limit
or rate limit. This is the maximum signal
limit, and is a function of the size of the
encoder. Notice that the large signal limit
is a rate or slope limit and not an amplitude limit. Assuming that the integrator
Digital
input
THE
SUPER BOSE
FALL
SPECIAL.
Now is the time to invest in
a Super-Bose System. Because if you
purchase your System before October 31, we'll send you a coupon
worth S150 towards the purchase of any other products at your
authcrized Bose Professional Products dealer.
The Super -Bose System consists of
two stacked pairs of Bose 802 Loudspeakers with a matched 802 -E
Active Equalizer. Together. they give
you twice the projection ano four times
the bass of a single pair of 802s. And
you get lifelike clarity, exceptional rug-
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Don't miss this chance to get the
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[Bose
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The Mountain
Framingham, Massachusetts 01701
Please send me a copy of the Bose
Professional Products Catalog and
a
complete dealer list.
Name
Address
State__
City
Zip_
LTelephone
J
Better sound through research.
Thi; offer applies only to complete Super -Bose Systems (four 802 speakers with 802 -E Equalizer)
puochased at participating authorized Bose Professional Products dealers between September and
1
October 31. 1981 All sales subject to verification Void where profubited
Covered by patent rights issued and /or pending 802 speaker design
Corporation. t© Copyright 1987 Bose Corporation
ts a
trademark of Bose
MICS THAT
DON'T
MEET
Audio input
a (t)
INJDUUT
STANDARDS BUT
SET THEM.
fhe two mics shown here are obviously as different as a transistor is to
a transformer. One is the smallest type we make, the other the largest. One
goes on camera, the other off. Yet the Sony ECM -50PS lavalier mic and the
Sony C -76 shotgun (both with conventional battery and phantom power
source operation) have one thing in common. Sound engineers throughout
the world consider them the standard of excellence.
So before you buy your next lavalier or shotgun mic, we ask you:
"Would you rather have the microphones made by a follower, or the microphones made by a leader?"
Pnd<..nmaI Audio
SONY
C 1981 Sons Corporation
of America.
ECM -
9 West 57th Street. New York.
NY
10019. Sony is a
registered trademark of the Sons Corporation.
i
SOPS
C-76
Circle
15
on Reader Service Card
rid of unwanted noise from
carts and transmission systems.
With dbx Type II Noise
Reduction, you get a full
40 dB increase in dyprovnamic range. The
new dbx Model 140
ides wo channels of encoding and two channels
of decoding- usable separately or
simultaneously. Provision forJensen output transformers. Active balanced inputs and other good stuff.
See your dbx Pro dealer, or write for complete technical information.
CLEANGet
U P YOUR
ACT
$59
'Manufacturer's suggested retail price.
Model 140 Type
11
Noise Reduction System
approximation
Figure 2. Example of slope overload at
delta modulation decoder. Approximation
is output for given inputs.
has a large voltage range, the approximation can get to any large value if one only
waits long enough. This means that the
maximum level sine wave at frequency!
will be twice the maximum level at
frequency 2j. Each octave increase in
signal frequency produces a 6 dB reduction in maximum amplitude. Of course,
one could increase the step size to get
larger signals, but this would also increase the LSB-type noise. A more effective alternative is to increase the sampling
frequency. The a(t) slope can be doubled
by doubling the number of steps per
second without increasing the LSB noise.
To appreciate the kind of numbers
involved, let us compare a delta modulation system to a straight PCM system
with the integrator step defined as an
LSB. A 12-bit converter has 4000 steps
between its minimum and maximum
values. What sampling frequency would
we need to follow a 10 kHz sine wave with
the same 4000 steps? In any sine wave,
the maximum slope occurs at the zero
crossing, and a little mathematics will
show that this slope is equivalent to 4000
steps in about 25 psec. This translates to a
delta modulation clock frequency of
160 MHz. If the comparison had been
made with a 16-bit PCM converter, we
would have come up with a number of
about 2.5 GHz. Clearly, these numbers
are extremely high and not easy to
implement. Moreover, the data rate in
terms of bits -per- second is much higher
than the PCM equivalent.
BIT INEFFICIENCY
The bit inefficiency of delta modulation can be appreciated from the following kinds of comparisons. To double the
dynamic range of a normal PCM system,
we need to add one more bit to the A/D
converter since this gives us a 6dB higher
signal level. At a 50 kHz sampling frequency, an increase of bit is 50,000 bitsper-second increase. With delta modulation I need to double the clock frequency.
Hence, if I were at a clock at 100 MHz, I
would need to go to 200 MHz, for an
increase of 100 M bits -per- second.
1
dbx, Incorporated, Professional Products Division,
71 Chapel St, Newton, Mass. 02195 U.S.A.
Tel. (617) 964 -3210, Telex: 92 -2522. Distributed in
Canada by BSR (Canada) Ltd., Rexdale, Ontario.
Circle
12
dbx
on Reader Service Card
www.americanradiohistory.com
Actually, the dynamic range is proportional to
rather than to! The reason
is that a doubling of the frequency also
doubles the frequency over which the
quantization energy is distributed. Thus,
f \'
A Sub -microscopic Coating
A quick "spitz," a gentle buffing, Lnd
you've dry-c dated the record with
an anti -stat /lubricant rust microns
thick. You can't see if even with a good
microscope, you can't hear it, but you
can feel the surface get slick and
stay slick, and you'll notice no static
cling of dus: or record sleeves.
Record cleaning is actually imprcved
(Lifesaver is unaffected by any
common wet or dry methods). And
even years later, when you dig thro ugh
your mu;ic library the Lifesavertreated records will still
be protected, sti_I
sound great.
Your records need protection. From
minor scratches, dust, dirt, and just too
much handling, no matter what your
program format.
Introdu_ing Lifesaver with DiscProTecT" Formula. A one -step record
preservative, anti -stat and dry lubricant.
One treatment lasts 5) plays or more
(Len Feldman` prove] Lifesaver lasts
100 plays, but we'll be conservative).
Records Sound Better Than New
From the very first audition, a Life Saver- treated record will have less
distortion than even a new untreated
record. And 50 or even 101) plays
later, it will still outperform that new
record, with no build -up of record
noise!
».
Save Records,
Save Money
Fewer Repeats, Repeats,
Get the dust, the
static. and the mir.or
scratches out of your music
library today Preserve the
investment in your music and effects
production library Start using Lifesaver from Audio -Technica today.
You'll sound better tomorrow!
AUDIO-TECHNICA U.S., INC.,
1221 Commerce Drive,
Stow, OH 44224. Dept. 91BÁ.
Repeats...
With most Lifesaver treatments,
minor scratches and nicks play right
througi withoat repeats (no guarantee,
but the odds are with. us). So if you're
into golden oldies, o- vintage jazz,
or classical music, you don't have to
cross your fingers, or do those abrupt
fades in the middle of the best part.
Nor will you need extra copies of
the records on your "heavy playlist."
%VES
YOUR RECD
rackability,
r°
icioeff
uces static and 0
`Extends useful
i
/audio-technica
Here's
SPECIAL INTRODUCTORY OFFER
To encourage you to prove the effectiveness of Lifesaver
treatment, we'll send you a full -sized $12.95 bottle to treat
60 sides for just $5.00. Just send this ad (or a copy) and
your letterhead. Just one sample to a station, please.
And we'll tell you where you can get low -cost refills, plus
see and hear other fine Audio -Technica products, including
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products, and the best cartridges and tone arms in the
business' Send today.
Harmonic d,stortan of an
Mn-ad au during fret playing
an excerpt from the Len Feldman report in Audio Magazine.
We'll send you the full story with your order.
Harmonic drstnrtqn 01 an
unhealed disc atol too pfayings
Harmonic dlstoMon of an denlleal
Orsc first playing after LIFESAVER
lien menu Distortion is
erradately rp.Ked
Circle 18 on Reader Service Card
Harmonic distortion e a
LIFESAVER treated du after IDO
playings Distorter romains lower
than a new untreated disc
the amount of noise in the audio band
will decrease by 3 dB when the total
bandwidth is doubled.
With a normal PCM system, doubling
the number of bits doubles the dynamic
range in dB, e.g. a 16 bits /sample PCM
has 96 dB of range compared to 48 dB for
an 8 bits /sample PCM. A delta modulation system gives an additive increase of
9 dB for a doubling of the bit rate.
Just as there is no free lunch, we do get
something for our increase in delta
modulation frequency: we increase the
system's ability to handle high frequency
signals above the audio region. A 200
MHz delta modulation system can
encode small signals at 50 kHz, 200 kHz,
etc. This is a useless feature for audio.
The extra bit inefficiency is actually
it 100 millivolts. This would allow the
LSB to be small when the audio was
carrying information about high frequencies, but there happen to be no
signals in this region. We ignore things
like tape recorder bias signals and stereo
subcarriers.
small, yet it could follow a fast signal
without slew limiting. This just leaves us
with the question of how to control the
step size.
The digital bit stream contains all the
information we need. If we see a bit
stream that looks like l0l 1111 111 11010,
then we can conclude that the series of Is
means that the integrator can't catch up
to the fast-moving audio input signal.
Similarly, a series of 00000000 would
indicate that the signal was moving
downwards too fast. In contrast, a series
of 10101010101 means that the approximation is hunting for the correct value,
but cannot get to it because the step size
is too large.
An adaptive algorithm might thus be
expressed as the following:
A) when there are 3 successive Is or three
successive Os, double the step size.
If the 4th bit is still in the same direction, double the step size again.
B) when there are 3 alternating bits,
halve the step size. If the 4th bit is still
alternating, halve the step size again.
An exact analysis of such an algorithm is
very complex but we can see in principle
that such a system could handle both
large and small signals. It is directly
analogous to floating-point PCM which
also changes its LSB size depending on
the signal amplitude. Unlike the floating
point system, the adaptive delta modulation can determine its step directly from
the actual transmitted data. One does
not need an additional path for the gain
ADAPTIVE DELTA MODULATION
The defects just described make a
classical delta modulation system impossible for audio. There is, however, an
interesting variation which is usable for
audio. Consider the idea of making the
integrator's step size variable. For small
signals we could have it be millivolt,
whereas for large signals we could make
1
How a Tweed Audio custom - designed
console helped Grampian Television
to make music and news!
For years, Grampian Television
(Aberdeen, Scotland) had the problem of limited production flexibility. There were two possible solutions. The first was to modify a
stock console (too expensive). The
smarter alternative was a console
from Tweed, custom -designed for
newscasts and music programming, and built after in -depth
consultation.
The module -based design includes 24 input channels with
stereo or mono program out, and
several cleanfeed outputs (international sound, mix-minus etc.) for
linkup with other T.V. stations. A
comprehensive system is built-in
for communications between
TWEED
studios, mobile remotes and
camera crews, and in spite of the re-
stricted dimensions, the console
even has room for adding facilities
if ever needed.
So, while the picture above looks
like just another hardware photo, it
is really a demonstration of how
Tweed Audio solves
I
client's
film -production or recording
studio, please contact us. We will be
happy to supply anything from
general specifications for existing
consoles to "clean- sheet" designs to
solve unique problems.
audio
Los Angeles, Vancouver, Great Britian
Pinnaclehill Ind. Est.
Kelso, Rox burg hsh re
Scotland
Telephone: (05732) 2983
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Circle 37 on Reader Service Card
1
information.
FIGURE 3 shows the behavior. We
assume that the audio signal had been
large with an a(t) step size of 16 units.
The audio becomes small and the integrator oscillates about the audio producing the sequence, 010 (circled); at this
point, the adaptive algorithm says,
reduce the step for the next point to 8
units. The next point is a I which is still
alternating, e.g. 0101, so the step is
reduced to 4. The process continues until
the step size reaches its maximum, which
is I unit in our example. At some time
later, the audio begins to increase and
there is a III series (also circled) which
forces the next step size to increase to
2
units. Following this point, the
sequence is 011011011 which is neither
an increase nor a decrease. There are
neither three-bits -the -same nor three bits-alternating.
The technical implementation is
slightly complex, but a digital engineer
can come up with many circuits which
will change the step size. An example is
shown in FIGURE 4 for a system with
8 possible steps sizes each of which is
twice the size of the previous one. A
common implementation is to use an
8 bit D/A converter as the step size
generator. The converter is not used in
the usual way but as a source of 8 precision voltages. The MSB provides a
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CJ
t
0
I
-a-
Il
S
4
0
I
_
I
0
I
0
O
1
1
0
1
0
1
1
I
I
Audio
I
Figure 3. Delta modulation decoder outputs with adaptive step -size strategy.
Audio represents the input, a(t) repreBents the output.
Figure 4. Block diagram of encoded algorithm for adaptive step -size.
Bits
Step size
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Shi11
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(,ninlel
register
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source of IO volts, the next bit 5 volts,
the next 2.5, etc. A 3-bit counter represents which bit is to be used. The 3 -bit
words (e.g. 000, 001, 010, 011, etc.) feed
a 3 -to -8 decoder such that the 0th line is
active for 000, the 1st line for 001, the
2nd line for 010, the 3rd for 011, etc.
The up; down counter can thus go to a
step size which is twice the size by stepping upwards one count, or it can go to a
step size which is half by stepping down.
The encoded data from the delta
modulator
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is passed
into
a 3 -bit
shift
register so that three sequential bits can
be examined at one time. The three bits
feed digital detectors. When the "000"
detector senses the correc pattern, it
increments the counter; when the "010"
detector finds its pattern, it decrements
the counter. This circuitry must be
implemented at both the encoder and
decoder since the step sizes must be the
same.
Systems like these have been built and
the performance is quite good in corn -
parison to a floating point PCM. Dynamic ranges of 100 dB are possible
because this is only determined by the
step size. Notice that with the use of 8
steps, the dynamic range improvement
from step changes is 48 dB. However,
when the design is complete with all of
the optimizations, the hardware becomes
quite complex and is no longer a bargain
compared to straight PCM of comparable quality.
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NORMAN
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H
CROWHURST
Theory & Practice
Complexity of Sound Waves
Acoustic waves that propagate sound
are generally referred to as longitudinal.
which means that movement of the air
W
particles takes place back -and -forth
along the same direction in which the
wave propagates. This is to distinguish
them from electromagnetic waves, in
which the oscillations of the magnetic
field take place along directions mutually
at right angles (transverse) to the direction of propagation.
Both those terms of reference are
basic. They are only fully realized when
the waves are also what are called "plane
waves." This means that the wave we are
analyzing is at considerable distance.
relative to its wavelength. from the
source. so that it is no longer an expanding wave. It is the property of both kinds
of wave at such distances from their
respective kinds of source that enables
them to radiate their respective kinds of
energy so efficiently.
With electromagnetic waves, the
traveling electric wave maintains the
traveling magnetic wave, and vice versa,
each mutually at right angles to the
direction in which both of them are
traveling, or propagating. With acoustic
Circle 26 on Reader Service Card
www.americanradiohistory.com
waves, alternations of pressure above
and below mean atmospheric pressure
are accompanied by momentary particle
movements, forward and backward,
along the direction of propagation. The
pressure changes and particle movements
are mutually self- sustaining.
This means that momentary particle
movement forward conveys the increase
of pressure forward. while momentary
particle movement backward conveys
the reduction of pressure forward. At the
same time, pressure higher than mean
atmospheric causes forward particle
movement. while pressure lower than the
mean causes backward particle movement.
The whole wavefront
is
maintained
because the waves we are considering
are plane waves. This means that all
points in the same phase of propagation
are in a plane at right angles to the
direction of propagation. If adjacent
points in the plane all have the same
instantaneous pressure, there will be no
lateral particle movement, only movement to or from other planes. where the
pressure is higher or lower; particle
movement occurs because of pressure
difference.
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Pressure is a function of the elasticity,
or compressibility, of the medium
through which the sound travels, generally air. But if this were its only
property, acoustic waves would be no
more complex than the pressure changes
in a bicycle pump, and waves would not
propagate. The other property of the
medium is its density, or inertia, of the
particles that make it up.
Because of elasticity, pressure differences cause particle movements from the
higher to the lower pressure. Because of
inertia, particles in movement convey
changes in pressure from point to point.
Pressure, as from a bicycle pump. causes
air particles to flow from higher to
lower pressure. And air particles in
movement, such as in a wind, cause an
increase in pressure when some obstruction slows them down.
Those effects can be observed separately. but acoustic waves occur when
they both happen together, each mutually
supporting, or causing, the other. And
this happens at a characteristic velocity,
which in air is a little over 1000 feet
per second.
The only way to generate a plane wave
would be by using some form of plane.
theoretically infinitely large, and vibrating the whole surface back and forth.
That is not very practical. Sound waves
must come from sources of finite size,
often much smaller than the wavelength
of the sounds the source propagates. So
we have to consider kinds of source other
than this theoretical infinite vibrating
plane.
0111,
Brit* Art, Taming!
In our October issue, we will be featuring a special sixteen -page supplement on the current state of British made audio. Among the articles included will be "The
-
British Audio Industry Today," "The British Sound
Truth or Legend," "Britain's Recording Equipment and
Its Reputation Abroad," "Digital Recording in Britain,"
"Creating a Sound Studio in Residential London," and
(of course) many more. This special and exclusive -to-db
supplement has been produced by the British Trade
Council, and we are sure you will find it of extreme interest. See also our Coming Next Month feature on
page 5 for all the other articles that will be in October.
www.americanradiohistory.com
In acoustic theory, the next kind of
source theorized is the so- called spherical
source, which can be visualized as a
small balloon, in which the air is rapidly
compressed and rarefied. This causes
the surface of the balloon to rapidly
become larger and the smaller.
With the plane wave, pressure changes
and particle movements occur in phase.
With a spherical wave near to its source,
which means where the distance from
the theoretical point source (which a
small balloon only approximates) must
be small compared to the wavelength
of the sound being radiated, particle
movement must be much greater than
pressure changes, just to get the air
necessary to cause those pressurechanges,
"in and out of the point source. This
increased particle movement, close to the
source, is not in phase with the pressure
changes associated with it, but gets to be
almost in quadrature. Only the energy
component is in phase.
So far we are thinking of a spherical,
or point source, acting in an infinite
medium. As you move out into the
medium, a section of the wave more
closely approximates a plane wave, with
the result that pressure and velocity get
more closely into phase. But now. if you
obstruct the wave, within the range that
may be considered as "close to a point
or spherical source.- the higher velocities
encountered there will cause pressure
changes due to the presence of the
obstruction (just as a fence or wall
creates pressures when a wind strikes it).
And like a fence or wall, some of the air is
diverted over the wall, instead of building
up pressure forever.
The final basic kind of acoustical
source is a dipole, or a pair of point or
spherical sources. This can have variations, as we shall see. You can think of a
dipole as a pair of small balloons, one of
which expands as the other contracts.
and vice versa. Another way to think of
it is as a tiny piston that vibrates back
and forth in open air, creating pressure
on one side and rarefaction on the other
at one moment, reversing it the next.
Now perhaps we can consider these
basics in relation to some types of
transducer, with which audio people
associate them. A lot of small loudspeaker units, mounted in a plane and
connected in phase, will radiate a close
approximation to a plane wave, at least
directly "in front" of it. But around the
edges, because the radiating "plane"
comes to an end, the condition we
described earlier as belonging to a plane
wave will no longer apply. Particle
movement will "escape" from the edges
of the pseudo plane wave into the
surrounding medium which is not being
moved in the same way. But the central
part of the plane wave will maintain its
direction better. This is the kind of
system attributed to Bose.
If. instead of a plane, the units are
arranged in a line, the end units will
have the same kind of velocity "spill
out,- but the middle units will spill out
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4'1
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Quality Procucts for The Audio Professional
Enclose your old
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of the line, not lengthwise.
all the middle units will have
the same sideways spill out, the wave
radiated will approximate cylindrical,
with the axis of the cylinder coinciding
with the line of units.
Now think about what happens with a
loudspeaker unit in which both back and
front are open like a small vibrating
plane: not a small piston, which has small
dimensions compared to any of the wavelengths it radiates, and not big enough
to radiate plane waves from back and
front, out of phase. What happens here
is that air rushes back and forth around
the edges in particle movement that is in
quadrature with the pressure that
accompanies it (a sort of squashing
action, like the water in some washingmachine agitators).
This becomes a type of transverse
wave. The sound wave moves away from
the unit, but the particle movement
remains edge -on to the direction of
propagation, and also in quadrature with
it. Close to the edge, particle movement is
considerable, but it falls off rapidly with
at the sides
And
as
distance.
The energy component of the radiated
wave can best be thought of when the
distance from the source is very large
compared to wavelength. Although it
now approximates a plane wave, it is
really a small part of an expanding
Circle 44 on Reader Service'Card
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www.americanradiohistory.com
p,ri
sphere. So when the diameter of the
sphere doubles, its surface area quadruples, and the energy in a small piece of
it, say one square foot, will be reduced
to one fourth. This is what is meant by
the "inverse square law." Energy intensity is inversely proportional to the
square of the distance from the source.
Such waves are always close to longitudinal, and pressure and velocity are
always close to in- phase. Quadrature
components of velocity are always much
bigger than these in -phase components,
where the quadrature components occur.
And because they reduce much more
rapidly than according to the inverse
square law (involving inverse cube
terms), they disappear as distance from
source increases, leaving only the longitudinal component.
In a position edge -on to a loudspeaker
unit with open edges, and at some
distance away, the transverse component has virtually disappeared and
there is very little longitudinal component anyway. So theoretically, there
will be zero sound in this direction, at
any distance from the source. The fact
that you can hear it is due to reflected
sound, either from obstructions close
to the unit, from other parts of the
room, or both.
This can be verified by careful measurements in an anechoic chamber. But
now let us turn to think about how we
hear. If you think about the outer and
middle ear structure, you will realize that
the human ear is an almost perfect
pressure transducer. It knows nothing
of sound particle velocity, directly. A
compression travels the short distance
from the side of your face to the tympanic
diaphragm, which it depresses in order
to compress the air in the middle ear
(long-term averaged by the Eustachian
tube) and the movement of the diaphragm thus caused is transformed by
the 3 -bone structure for transmission to
the inner ear, and nervous transduction.
Sound from directly in front, behind,
overhead, or any direction such that it
reaches both ears absolutely simultaneously, will send messages to the
brain via both ears simultaneously.
Sound from one side, to go to another
extreme case, will produce two or
maybe even three effects that the brain
can observe. In this position, the particle
movement is such that the head constitutes a more definite obstruction than
it does in the other set of positions.
In the equidistant set of positions, the
sound wave flows over both ears, so that
each of them can "sample" the pressure
fluctuations as it goes by. When sound
comes from one side, the particle velocity
comes up against the near side of your
head and the obstruction effect translates
that into sound pressure, more than
would be encountered in the other directions of approach. But on the far side of
your head, a much reduced pressure is
generated.
i here is
also a difference between
the times at which the waves arrive at the
two ears. When it comes from one side, it
arrives at the nearer ear sooner. Other
things happen to the sound wave, at each
ear. dependent on the shape of your face
and especially those external appendages
that we call "ears.' but which contribute
nothing directly to the transduction.
However, they do moditc the sound wale
due to its particle movement. and thus
pros ide more clues as to direction than
you could get if your head merely had
plain holes in it!
So far we have talked about radiating
sound. and about how we hear the
simplest firms of arriving sound. But
when we get into stereo, and stereo
reproduction. there are more interactions that happen. the system we use
to radiate the "stereo" can now produce
more complex patterns of particle movement that can be used to create illusions
for you, using these properties of hearing
in more complex ways. I hat is what we
want to get into nest.
All Wireless
Microphones Are
Not Created Equal
This One is a
Telex
This publication
is available
in microform.
Wireless mics aren't new, and sometimes it seems as if all
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The Telex wireless sounds as good as a hard wired mic. offers
plenty of options and is economically priced. If you're interested in a wireless system that is more than equal -write us
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Ann Arbor. Mi. 48106
U.S.A
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IU
LEN FELDMAN
Sound wies Images
.
Stereo TV
Japanese TV broadcasters have been
transmitting multi -channel audio along
with their video signals for the better
part of three years now. Note that I called
their audio transmission multi -channel.
rather than stereo. That's because the
transmission system is used not only to
transmit stereo audio (which at the
moment is limited almost entirely to
videocasts of concerts and other musical
events), but also as a means of transmitting bilingual audio information. In
fact, if you were to pick up the equivalent
of a TV Guide Magazine in Japan, you'd
probably find more of these bilingual
What's the Holdup?
in Tokyo or Osaka. But that is no surprise, since there is not a great deal of
video fare (either in Japan or here) that
lends itself to stereo audio. Back in the
a local stereo FM station. Evidently.
the argument that people won't enjoy
over
watching a small picture while listening
to a big sound doesn't hold water.
Well, rather than wait for us, the
Japanese. in their by -now familiar way.
went right ahead and developed a
perfectly workable system for two channel audio transmission and managed to effectively squeeze it compatibly
into the same NTSC video format that
we use in this country. (I hey adopted the
NTSC system at a time when they didn't
know better and were still taking technological cues from us, I think.) Interestingly, in the case of stereo FM. the
Japanese also took their cue from us and
early 1960s, our own Federal Communications Commission, when asked to consider the possibility of stereo audio for
TV. sagely decreed that no one would
want to watch a tiny video screen and
hear sounds coming from "way out to the
left or right." That was before the days
of large screen projection TV. of course.
I know a great many people, though, who
even with their "small screen" TV sets
enjoyed simulcasts of concerts in which
the video was broadcast over Public
Televsion, and the audio transmitted
transmissions than actual stereophonic
audio transmissions on a typical evening
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(714) 556 -6193
Circle 29 on Reader Service Card
www.americanradiohistory.com
CANADA:
E. S.
Gould, Montreal, Quebec H4T
1E5
_
ended up with the same AM- suppressed
carrier system that we use. I suspect they
learned from this experience and went
ahead on their own when it came to
deciding upon a multi-channel audio
system for TV.
Having heard the system in Japan on
several occasions, I can tell you that it
works extremely well. And, of course. it
generated a whole new group of products, such as stereo TV tuner -only units
(which can be added to existing color TV
sets), add -on amplifier; speaker units
and, last but not least, new top quality
color TV sets which have the multichannel decoding and playback facilities
built right in along with left- and right channel speakers. It is a real joy to be able
to switch from Japanese-dubbed dialogue, heard with an American made
film, to the original English sound track.
And, of course, this added capability
means that there is no need for sub -titles.
which I often find so difficult to read.
especially on a TV screen, when contrasts obliterate :he titles.
The Japanese multi -channel TV audio
system employs an all -FM system. That
is. both main -channel and sub -channel
are frequency modulated. Sum -anddifference techniques are used when
stereo is being broadcast, while for bilingual applications, the main language
is transmitted over the main channel
while the alternate language is transmitted via the sub -channel. Simple -and
it works.
better than the others, the FCC should
certainly arrive at that same conclusion
and make it the official standard for the
U.S., regardless of what is done in
other countries.
Both of the competing systems proposed by U.S. companies utilize an
AM subcarrier, for reasons we won't go
into just now. But as anyone who is
experienced with stereo FM knows, any
multi -channel system using sum -anddifference techniques (L +R on the main
channel; L -R on the sub -channel) and
an AM subcarrier is going to end up
being noisier (at weak signal locations,
tion of multi- channel audio on TV, and
so now, the Electronic Industries Assoociation (EIA, a trade association, one of
whose groups embraces consumer electronics manufacturers) is hard at work,
with several committees attempting to
develop a report for the FCC which will
enable it to promulgate transmission
standards for multi -channel audio on TV.
By now, you may be wondering why
such committee work is needed if the
Japanese system seems to work so well
in Japan. After all, audio and video
electromagnetic radiation works about
the same in Asiatic longitudes as it does
in the Western hemisphere. The reason
for the deliberations is that more than
one system of multi -channel transmission is being evaluated by the committees. In fact, there are three. The
system now on- the-air in Japan is being
sponsored as an official entry in this
country by the EIAJ (Electronic Industries Association of Japan) a trade association that is the equivalent of our own
EIA. But two other systems also proposed are those developed by Zenith
Radio Corporation and Telesonics Corporation. To be sure, having competitive systems is certainly in our American
tradition, and if one of the systems is
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In this countr}, the FCC has had
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www.americanradiohistory.com
out in the suburbs) than a system which
uses an FM subcarrier, all other things
being equal.
So now, many people have become
concerned about reduced audio coverage.
In all fairness, it should be pointed out
that even with an all -FM system, some
increase in noise would be inevitable (you
hardly ever get something for nothing.
and cramming two channels where only
one used to be has got to mean some
tradeoffs). With that in mind, it was
proposed that perhaps the answer was to
do a bit of electronic noise reductionbuild the noise reduction right in. as a
standard, right from the beginning. So,
the call went out for various noise
reduction folks to come forward as
proponents. And before long, there were
three of these: Dolby. dbx, and CBS,
with a variation on their new CX noise
reduction; companding system. dbx had
to modify their familiar linear companding system because one of the most
important criteria for any system to be
approved by the FCC would be that it
had to be "compatible." In other words,
owners of present TV sets must be able to
receive their TV audio information
without having to pay for anything extra
-and that audio should sound about like
it always did. Well, if you've ever heard
un- decoded dbx -encoded material you
know that 2:1 compression is not very
-
pleasant to listen to unless you decode
it (1:2). So, dbx dutifully modified their
system to make it compatible. The CBS
CX system was already compatible
(that's one of its claims to fame -you
don't have to buy a decoder to listen to
CBS -CX discs, but you won't get the
noise reduction benefits unless you do).
As for Dolby. all they had to do was
decide whether they wanted to offer
Dolby B (with its IO dB of noise reduction at high audio frequencies) or their
newer Dolby C (with as much as 20 dB of
noise reduction). They chose Dolby C.
So, here's where matters stand. We
now have three basic transmission
systems proposed for multi -channel TV
and each of those systems may be augmented or accompanied by one of three
proposed noise reduction systems. You
could have the EIAJ system with the
Dolby noise reduction system, or perhaps the Zenith system with the CBS
CX companding system or...well, I'm
sure you get the idea.
It should be pretty clear from all of this
that it's going to be some time before
the FCC gives the go -ahead fora particular system. h is even conceivable that
they may not choose one of these nine
permutations at all, but might say to the
industry (broadcasters and consumer
hardware makers alike) that they are
free to experiment with any combination
they choose. This is the so-called "Let
The Marketplace Decide" approach
which has been mentioned from time to
time in connection with the still- awaited
stereo AM decision and the long postponed (and almost forgotten) quadraphonic question.
I don't know whom to pity more while
we wait -American TV broadcasters, or
Japanese hardware manufacturers. The
broadcasters would like nothing more
than to have the extra capability of stereo
and / or bilingual audio. As for the
Japanese manufacturers, right now they
have to make two basic kinds of models:
twin channel versions of video products
for their own domestic consumption, and
(from their point of view) somewhat
archaic mono models for export to the
rest of the world and to the U.S. in
particular. And remember, everything
we've talked about applies to VCRs as
well as to basic TV sets. I must confess
that I was somewhat amused a few
months ago when one maker of VCRs
introduced into the U.S. market a video
cassette recorder that "actually had
stereo recording and playback cap bility." That must have seemed like quite
an advance to the uninitiated here. To
those of us who were aware of what goes
on in Japan, it was nothing more than a
clever way to save money by not having
to have two basic types of VCRs.
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Circle 23 on Reader Service Card
www.americanradiohistory.com
Editorial
Broadcast Audio
& PAPALLAC10. August 1981. On the offchance that some of our readers may not instantly
recognize our dateline cities. the former is a small
village some twenty miles outside of Quito,
Ecuador, while the latter is a still -smaller village some
twenty miles beyond Pifo. Papallacto used to be the end
of the road: if you wanted to penetrate further into the
Ecuadorian interior. you could rent a mule here for your
PIFO
continuing journey.
Times have changed. The road now extends just a
little further into the jungle, and there's a new hydroelectric plant just outside of town. The plant supplies
electricity for a 500 kW transmitter at Pifo. That's right,
500 kW. You might say that the folks in Pifo have really
taken the NRBA's "We're for Radio" slogan a little too
seriously. But not quite: Pifo happens to be the transmitter site for the legendary HCJB Radio. Here. "La
Voz de Los Andes" broadcasts in some 14 languages to
listeners as far away as Siberia and Japan. It's quite a
story, and you'll read more about it in our March. 1982
International Audio issue.
As luck would have it, this month's broadcast audio
editorial came due just as we were making our way
through the Andes from Pifo to Papallacto. (Actually.
it was due before you left New York- Publ.) In this
little corner of the world, its difficult to comprehend
just how close we are to New York. In space. we're about
as far from home as it we had gone off to Los Angeles.
Our closeness in time is quite another matter. The dam
and the transmitter it serves are twentieth century -all
else here is not. It's a difficult place from which to write
a 1981 -style editorial. The pressures of deadline-meeting.
and other clock -related matters. don't seem very important -maybe it's the altitude.
As we left home, there were rumors that our fast moving FCC was about to leap into action on the subject
of FM quad broadcasting. Now that most of us have
forgotten what that is. presumably it's safe for them to
make a decision. Since they've now had some ten years
to think it over, let's hope their actions will be a little
more carefully thought out than the recent AM stereo
comedy of errors.
While we're waiting for the learned Commissioners to
lead us down the path to higher technology (or failing
that, at least to stop blocking traffic), it's refreshing to
note the increasing interest by everyone else in higher
quality broadcast audio. The signal path within the
broadcast studio used to be one of the weaker lines in the
audio chain. But lately. the line is getting stronger all
the time. As many of this month's feature stories illustrate, prominent manufacturers of recording studio
consoles have responded to the demand with speciallydesigned boards to meet requirements of the qualityconscious broadcaster.
It's no longer enough to be the loudest station on the
dial, and broadcasters are discovering that a sizeable
segment of the audience will tune in to quality when and
if it's available. It has been discovered that most listeners
can manage to adjust the volume control to get the loudness they desire. For those stations that seek to spare
the listener this inconvenience, by broadcasting loudness
and little else, some of those listeners are responding by
adjusting the station selector knob instead.
Of course, the loudness race has by no means been
called off entirely, nor is this likely to happen in the
immediate -or even distant -future. But the days when
broadcast signal processing required only a brute -force
compressor are fast drawing to a close. Broadcast audio
is now getting the kind of serious attention it deserves.
Here at db (that is. at our Papallacto branch office),
we're glad to do our little bit by presenting this broadcast
audio issue.
www.americanradiohistory.com
GEORGE BERGER and ROBERT L. DEITSCH
Broadcast AudioThe Increased Need for
Audio Processing
Here, authors Berger and Deitsch provide us with a behindthe- scenes look at the problems (and the solutions) faced
by engineers in the broadcast field.
will deal with the basic concepts which
underlie the design of facilities for the ABC radio
broadcast studios in New York. Since 1975, when a
slow but major reconstruction was begun, we have
argued with our friends about what we believe are the fundamental and philosophically different aspects of broadcasting
which distinguish it from the rest of the audio field. There are
to this day several misconceptions about what broadcast audio
is and ought to be. which still disturb audio professionals
and amateurs alike.
is there any reason why that which is broadcast should be
identical to the original source? Well, there are two parts
necessary to the answer. The first, and we grant some nitpicking, is that none of the records and tapes which are our
primary sources are unprocessed, and secondly, shouldn't the
broadcaster be granted latitude in the same sense in which
recording engineers use processing to compensate for the
limitations of their media or to create a uniqueness of sound for
THIS ARTICLE
co
George Berger is chief engineer WPLJ /NY & project
manager AM stereo A BC radio division. Robert L.
Deitsch is the senior design engineer ABC radio. NY.
group? We suggest that to answer no is intellectually dishonest, and thus we have ceased to feel any necessity to defend
our position. We must make it understood, but the proof comes
from that understanding a priori.
Our somewhat dogmatic philosophy about our studios is that
an on -air control room should be as specifically tailored to
the format as possible, and that a production facility must be so
general and versatile that it never limits the creativity of either
the producer or technician. Technical equipment should be
of the highest possible quality. All of our installations since
1975 have been centered around custom -designed mixing desks
from Rupert Neve, all reel -to-reel tape decks have been Studer
A -80 mastering recorders.
It would be unfair for the reader familiar with the two
ABC -owned radio stations in New York to be saying, "What a
waste of good facilities: by the time 1 hear the product it sounds
very different. " The AM station is WABC which, over the years,
has been the premier clear channel station in the U.S. ABC's
FM station in New York is WPLJ, which is the top -rated album
rocker in New York. The two stations provide some excellent
examples of the use of audio processing within our medium.
a
EASE VERSUS QUALITY
One of the problems broadcasters have faced since the early
sixties has been the conflict between the desirability of using
tape cartridges for ease of production and the degradation of
quality, especially in stereo, which seemed to be inherent in
using them. The two major problems were phasing and degradation of signal -to -noise ratio. As early as 1973 we were
experimenting with a matrix encoding system to minimize
phase instability. In 1975 we decided to use dbx® noise reduction as well.
There has been much debate over the relative merits of dbx
and Dolby noise reduction. Because so much of the source
www.americanradiohistory.com
(dbx 165)
Cumpresors Limiters
Figure 1. WPLJ's "better mousetrap" consists of an
Ashly SC -77 Crossover and six dbx model 165 "Over Easy"
Compressor /Limiters. (A) Block diagram; (B) rack mounted equipment.
material we use comes to us already compressed, and also
because dbx offers a far greater amount of noise reduction to
precede the additional compression which we use, dbx was to
us a far preferable choice. Both of our stations now use this
combined processing system. This type of processing should
be transparent; the listener, either in the studio or at home,
should not be able to detect its presence.
The type of processing which causes controversy is compression. limiting, and clipping, as well as any equalization
or reverb which may be added. It should be realized that the
"sound" of a radio station is an important programming
consideration. Psychoacoustic analysis is one of the elements
which enters into such decisions, but the broadcaster must also
consider competitive loudness and dynamic range.
Many listeners are in environments with high ambient noise
levels, such as automobiles; a station can ignore this part of its
potential audience only at its peril. On the other hand, if
listeners find the signal unenjoyable because too much
processing is added to "fix" this type of problem, audience
may also be lost. Consequently, the final decisions, within the
limits imposed on all broadcasters by the FCC rules and
regulations, must be made among what are still a scientifically
ill- defined set of possibilities. Having made those tough choices,
the broadcaster must enter into the area of receiver technology
in order to determine how to best accomplish those aims.
It is reasonable to look at the FM broadcast chain, including
the home receiver, as offering the potential of true reproduction. But it is by no means true that all FM receivers can
reproduce what is in the broadcast signal with true high fidelity.
Thus, even with FM, the broadcaster must attempt the clearest
possible definition of the audience. With AM, the waters
become much more murky. AM transmission is for all practical
purposes capable of equal fidelity! It is, of course, subject to
atmospheric and electrical interference, but the broadcast
signal is high fidelity. THD and IMD figures of less than one
percent are achievable. Signal to noise ratio approaches 60 dB.
Today, most AM receivers begin to roll off frequency response
at I kHz and are down more than 6 dB by 5 kHz.
In defense of receiver manufacturers (but certainly not to
agree with them), it is, of course. more expensive to build wide
response high fidelity AM receivers. Most firms claim that there
is no mass market for such equipment. We as broadcasters
counter that this is a matter of consumer education; there was
no immediate large -scale rush to FM equipment either.
So, we the broadcasters are left attempting to compensate
for the varying quality of receivers, both AM and FM, that are
in the hands of the public. Both of our New York stations play
"popular" music. The problem faced by the technical staff to
meet the desires of the WABC programming department is to
provides a sound which makes WABC -AM sound as good on
the radio as an FM station. This means that we must compensate
for the "poorer" AM circuitry in AM FM combination radios
and receivers.
Through experimentation, we have found that no single
equalization curve can accomplish our purpose. The single
most obvious objection to a simple equalization attempt comes
from the announcers and disk jockeys. They feel that they don't
sound like themselves. They are correct. We have found that a
multiband compressor/ limiter system has been the answer.
Many of the ideas for use of equipment start at our visits to
Audio Engineering Society conventions. For example, we were
extremely impressed with the "Over Easy®" compression
curve used in some dbx equipment. Many years of experience
have taught us that the use of any single band compression
device does not yield the desired audio density without destroying the clarity of tone at the lower and higher ends of the
frequency spectrum. Such single-banded attempts typically
yield muddy drums and cymbals.
www.americanradiohistory.com
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Figure 2. Custom -designed Neve boards. (A) Studio 8C;
(B) Studio 8X.
For many years. WPLJ-FM has used Dorrough discriminate
audio processors. But all of our attempts to use them and other
readily available tri-band processors on AM had yielded, what
were to us, unsatisfactory results. We knew that it was relatively
easy to make music sound good, but found that the quality of
voices was changed too much. Finally, we bit the bullet and
decided to attempt to build a better mousetrap out of commercially available components.
Figure
3. The
BUILDING THE PROVERBIAL MOUSETRAP
"e found that an Ash!) crossuver with sellable depth and
12 dB/ octave crossover curves was best. Steeper curves didn't
sound as good to us. We wanted to assemble a tri -band
processor with compressor and limiter in each band to use
in our air chain at the studio, plus a single processor for final
equipment rack in Studio 8C.
ar.ci.t
00
N
gain compensation at our transmitter. A separate compressor
and limiter is used in each frequency band to achieve a
uniformly loud sound without pumping.
The dbx 165 offers a number of advantages beyond the fact
that we liked its sound. It can be coupled for stereo, which
probably is coming to AM. It turns out to be extremely important that it has user access to the limiter sidechain. It also
offers externally accessible automatic or variable attack and
release times, infinitely variable compression ratio, and
adjustable level threshold.
The crossover is adjusted to split the frequency spectrum
optimally to allow the compressor, limiter pairs to be set for
thumping lows. clear midrange. and crisp highs. For most
formats, we feel that two crossover points are sufficient. For
disco. one might want to add an extra band for very low frequencies. We feel that different formats will be helped by
adjustment of both crossover points and depth. Delay and
equalization are inserted into the side chain to eliminate
phase- related interactions of the recombined audio. Small
amounts of delay added to the side chain signal can achieve
some remarkable results. Equalization can, for example. aid in
de- essing; this is important before clipping when attempting
to maximize loudness in the baseband. A truly flat AM signal
yields a muddy sounding AM signal because of receiver design.
We have achieved a psychoacoustically pleasing sound which
simulates wideband reception on most AM receivers.
The program director of WPLJ wants a compressed sound.
This is achieved using a Dorrough 310C discriminate audio
processor with modified recovery times, Moseley limiters,
and composite clipping. When composite clipping is used, the
broadcaster must adjust it so that he can pass an FCC proof of- performance with it in the circuit as it is for everyday
transmission. WPLJ is the loudest FM station in New York.
We have designed our technical facilities to allow for rapid
modification of air sound. We are proud of the quality of our
plant and of our ability to provide the sound required by our
programmers.
www.americanradiohistory.com
Portrait of a Quiet Family
Meet the members of our S;ader
console family. The compact,
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269. And the full size 369 studio
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traits...
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Studer consoles are designed for
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each master module rias a 'main limiter. As far S /N, well it's
no surprise that the :69 has been
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On the 169 /269. power supply
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Long -Term Relialbil.ity
Studer console* are built for the
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The Sound
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tor monitor sends. Pan pot
controls allow panning to the
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level controls permit 16 x 6
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sources. Portable operation
is a snap with easy access
connectors.
And the WR -8716 features
plastic ccnductive faders
for greater reliability and
smooth, low -noise operation;
external power supply for
light weight, and switchable
48V DC phantom power
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moxular construction, input
moules, power supo Les, and
fadas as the WR -8716 plus
may important recording
advantages. Lice direc: cut put for 4, 8, or 16 track recording and peak-reading
LED meters that let ya moni:cr any 4 out of 24 signals
with c ear, quick response.
You'll command a variable
frequency EQ section with
3 frequency settings or the
n gh and low frequercies
p,LE cortinuously variable
midrange. Siereoecho send
replaces the separate --ono
controls you' finc on pompetitive bcarik. And you get
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monitor controls-cm for
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for cont-ol room moritorsa special feature for any
mixer in this cbss. And there
are other riportant features
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balanced rric inputs with
new high -speed IC's, 16
switchable post-fader s`do
controls ari XLR-type mic
connectors.
Ramsa cffers a full line of
specialty mixers includingg
the more compact WR -3210
recording mixerand'WF-.30
sound reinforcement -n xer.
Sc don't hold dcwn your
professional sound, ca
(201) 348 -7470. because
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PROFESSIONAL AUDC DIVISION
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THOMAS MINTNER
Broadcast Audio
Production Consoles
The Studer series 900 console is one way to integrate the
various requirements of multitrack music recording and advanced
broadcast production into one console system.
A
higher quality production
for broadcast audio increases, choices involving
multitrack recorder/ console systems become
more difficult. In order to illustrate some of the
problems which designers and users must resolve, a brief
review of console evolution and a description of a new console
IIII I.MPHASIS ON COMPLEX.
system are given.
To better explain broadcast production console requirements, some further definition may be in order. For purposes
of this article,
we will define broadcast production as: support
audio production for radio and television, including station
spot and advertising production, multitrack and simple stereo
program recording for later broadcast or post -production and
also such related audio tasks as video sweetening, film work,
etc. Broadcast production should not be confused with the
recent term "production equipment," coined by manufacturers
to refer to equipment formerly called "semi- professional,"
although certainly some equipment of this type is in use today
for broadcast production. We will also exclude from discussion
the "air console' used for normal broadcast, usually of a
considerably different (often simpler) pattern than the produc-
tion console.
In radio's earlier days, the production console was often
the station's previous air console, now relegated to the back
room with a similar collection of recorders. This arrangement
was adequate for bread -and -butter spot production and similar
activities.
In the television area, sophisticated audio production was
formerly quite rare, being limited mainly to networks or
network owned-and -operated affiliates where a large audio
console for production activity was usually custom designed.
As recording studio activity flourished in the late 1960s,
equipment manufacturers brought multitrack facilities in one
form or another to more and more users. Eventually, there was
a demand for better audio in broadcast. The industry responded
quickly; today in scanning the pages of broadcast journals,
it is easy to see that audio signal processing and control is a
subject easily holding its own among the other major broadcast
concerns such as ENG and distribution by satellite. Indeed,
the new distribution technologies, including analog and
rNi
Thomas Mintner is the manager of Broadcast Products
with Studer Revox America, Inc.
digital audio channels of very high quality, have effectively
removed the old excuses for letting broadcast audio quality
remain at lower levels, and thus serve as a further impetus
for more sophisticated audio production.
While large broadcast operations may be able to afford
custom -designed hybrid consoles with multitrack capabilities
combined with necessary broadcast features, many other
users have been forced to outfit themselves with small general
purpose recording consoles or with the one basic architecture
of the studio music recording console where little thought is
given to combination use. Some manufacturers make both
types of consoles, others only one. The first type is the conventional production or small recording console, with one to
four main mixing buses, for use with 2- or 4-track recorders.
This type of console is often supplied to broadcasters as -is, or
modified with a number of useful items such as stereo line
level input modules, multi -input selector switches, broadcast
type cue features and fader start switches or other remote
control facilities for tape recorders.
Generally, the basic philosophy is that inputs (typically
12 to 16 in this size console) will be simply mixed live to stereo
(or possibly to a 2- or 4 -track recorder), with monitoring in
mono, or stereo derived from the main mixing buses or tape
output.
At the time when the recording industry was first expanded
to 8 and 16 track operation, this type of console could still be
used in studios, with the relatively minor additions of an
independent multitrack metering bridge and some type of
monitor arrangement for the multitrack recorder. This might
be a simple external 8 -by -2 or I6 -by -2 matrix mixer which
could then be selected on the monitor panel. This was early
multitrack production with the following characteristics:
the number of mixing buses was much smaller than the number
of recorder tracks, a simple multitrack monitor allowed for
multitrack listening during tracking, there were no direct
outputs from input channels, and much patching was required.
Gradually, the console described above pretty much disappeared, as the increasing demands of music recording led
to the evolutionary development of a "Mark Il" console,
which in general had the following characteristics: the number
of mixing buses is equal or near to the number of recorder
tracks, there is full metering for the buses and multitrack,
internally switchable, and there is a well -developed multitrack
monitor section, either in -line or as separate side monitor
Circle 22 on Reader Service Card
www.americanradiohistory.com
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Introducing the
JBL Bi-Radial
Studio Monitors.
No one has to tell you how important
flat frequency response is in a studio monitor. But if you judge a monitor's performance by its on -axis response curve,
you're only getting part of the story.
Most conventional monitors tend to
narrow their dispersion as frequency
increases. So while their on -axis response
may be flat, their off-axis response can
roll off dramatically, literally locking
you into the on- axis "sweet spot." Even
worse, drastic changes in the horn's
directivity contribute significantly to
horn colorations.
Polar response ofa typical two -way coaxial
studio monitor:
At JBL, we've been investigating the
relationship between on and off axis
frequency response for several years.
The result is a new generation of studio
monitors that provide flat response over
an exceptionally wide range of horizontal and vertical angles. The sweet
spot and its traditional restrictions are
essentially eliminated.
Polar response ofa 4430 studio monitor.
TYPICAL
HORIZONTAL
JBL 4430
HORIZONTAL
TYPICAL
VERTICAL
JBL 4430
VERTICAL
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The Bi- Radial Horn
The key to this improved performance
lies in the unique geometry of the
monitors' Bi- Radial horn? Developed
with the aid of the latest computer
design and analysis techniques, the horn
provides constant coverage from its crossover point of 1000 Hz to beyond 16 kHz.
The Bi- Radial compound flare configuration maintains precise control of the
horn's wide 100° x 100° coverage angle.
Since this angle is identical to the coverage angle of the low frequency driver at
crossover, the transition from driver to
driver appears seamless and the monitors
present a fully coherent sound source.
And the BiRadial horn's
performance
advantages aren't
limited to just
beamwidth control. The horn's
rapid flare rate,
for instance, dramatically reduces
second harmonic
distortion and its
shallow depth
allows for optimal
acoustic alignment
of the drivers. This
alignment lets the
Acoustic alignment
monitors fall well
ofdrivers 14430)
below the Blauert
and Laws criteria
for minimum audible time delay discrepancies.
The practical benefits of the Bi- Radial
horn design include flat frequency
response and remarkably stable stereo
imaging that remain valid over a wide
range of listening positions. The design
also allows considerable latitude in
control room mounting. Finally, the flat
on and off axis frequency response of the
horn means that less high frequency
equalization will be required to match
typical house curves.
But while the Bi- Radial horn offers
outstanding performance, it's only part
of the new monitors' total package.
Extended Response in a
Two -Way Design
Coupled to the horn is a new compression driver that combines high
reliability and power capacity with
extended bandwidth and smooth, peak free response. The driver features an
aluminum diaphragm with a unique
three-dimensional, diamond -pattern
surround) Both stronger and more
flexible than conventional designs, this
surround provides outstanding high
frequency response, uniform diaphragm
control, and maximum unit -to -unit
performance consistency.
To ensure smooth response to the
lowest octaves, controlled midband
sensitivity. extremely low distortion, and
tight transient response, the Bi- Radial
monitors also incorporate the latest in
low frequency technology. The loudspeakers' magnetic structures feature
JBL's unique Symmetrical Field Geometry (SFG) design to reduce second
harmonic distortion to inconsequential
levels. Additionally, the speakers utilize
exceptionally long voice coils and carefully engineered suspension elements
for maximum excursion linearity, and
coritplete freedom from dynamic instabilities for tight, controlled transient
41111%
7:110
Symmetrical magmatic Feld 71J111,
SFG deren great.y rehires -:istortion.
response.
JR /..i diamond suspension diaphragm
combines per/rrannce
wih rthability.
Blending the Elements
The Dividing Network
Challenge
-
Tailored to the acoustical characteristics of the Bi- Radial monitors' high and
low frequency drivers, the dividing
network provides the smoothest possible
response over the widest bandwidth
while restricting any anomalies to an
extremely narrow band. During the
network's development, JBL engineers
paid considerable attention to on -axis,
off -axis, and total power response. As a
result, the electrical characteristics of the
network are optimized for flat response
www.americanradiohistory.com
over the monitors' full coverage angle.
The network also provides equalization of the compression driver for flat
power response output. This equalization is in two stages with separate adjustments for midrange and high frequencies.
Judge For Yourself
Of course, the only way to really judge
studio monitor is to listen for yourself.
So before you invest in new monitors,
ask your local JBL professional products
dealer for a Bi- Radial monitor demonstration. And consider all the angles.
a
L Patent applied for.
Specifications
4430
Frequency response
3 dB)
35
16,000 Hz
-
4435
30 - 16,000
Hz
(±
Power Capacity
(Continuous Program)
300 W
375 W
Sensitivity
93 dB
96 dB
(1W,lm)
Nominal Impedance
Dispersion Angle
(- 6 dB)
100° x 100°
Crossover Frequency
1
Network Controls
UBL
Available .n Canada
Ohms
8
kHz
Ohms
8
100° x 100°
1
kHz
Mid Frequency Level
High Frequency Level
Switchable Bi- Amplification
James B. Lansing Sound, Inc.
8500 Balboa Boulevard,
Professional
Products
Division
Quebec.
P.O. Box 2200
ridge, California 91329 U.S.A.
4430/35
8-81 Printed in U.S.A.
www.americanradiohistory.com
JBL /serrnin
Figure 1. The Studer series 900 broadcast production
console.
modules. A direct output is available from each input module,
and sophisticated internal console switching for meters and
monitors is provided.
These specifications evolved in the studio for maximum ease
in the three major areas of multitrack music work: original
tracking, sync overdubbing, and, of course, mixdown.
Because many users of today's multitrack studios entered the
field after the introduction of this type of console, there sometimes is considerable confusion concerning the actual shape
of a modern console. There is confusion concerning such
things as the relationship between the number of mixing
buses on a console and the number of outputs (direct) from that
console, or between the number of outputs and the maximum
number of tape monitor/ meter channels available. Because it
seems that the maximum numbers of these specifications are
continually increasing (more buses, more channels of monitor
for dual machine sessions, etc.) in the music recording studios,
it is important to remember that other production users may
not at this time require all of these maximum numbers.
Unfortunately it is a fact that console designs regardless of
basic size have, within a series, tended to be either of one type
above or the other, and as of now these two types (general
purpose and multitrack) have diverged so far as to have left
a considerable hole between them. Hence, for broadcast
production use, many consoles are either too simple or too
music/ recording specialized. The broadcast production user
must often either adapt an unsuited music recording console
to the need or sacrifice features for certain practical considerations.
LOCATION OF CONTROLS
The typical multitrack console has evolved into a rather deep
(front -to -back) size due to several factors, including more
elaborate equalization facilities, the space needed for in -line
monitor controls on the input modules, more bus assignment
buttons, and the need to retain a low profile for good aural
and visual "sight lines" in the control room. Controls are then
placed at increasingly distant spots from the operator based on
their frequency of use in music recording session practice.
Indeed, some controls used only in broadcast may be completely absent. Unfortunately, the frequency of use may be
considerably different in a production room where the use
of the multitrack recorder is only one of the possible normal
procedures.
IN -LINE MONITORING
The in -line monitor section mentioned above is a rather
clever solution to a problem which arose early on in recording.
As the number of input channels and tape channels began to
increase greatly, consoles with separate multitrack monitor
sections became increasingly long. This was more or less of a
problem depending upon the division of responsibilities
between engineer, producer, and assistant engineer, but sheer
physical size was definitely becoming an operating problem,
wheeled chairs notwithstanding! The addition of a monitor
section physically folded back onto the input modules provided
a greatly- reduced console length. However, since the introduction of the in -line monitor, other changes such as the use
of the dual-channel module (two monitor channels in the width
of one input channel) and the emerging demand for independent equalization on the monitor section have made the
so-called side monitor again popular in many applications.
In particular, the in -line monitor design has had a special
impact on the broadcast production user, since the inclusion
of this special section forces the production user into the
particular range of a manufacturer's consoles which are
designed primarily for music recording studios. Often, many
of the desirable features the broadcast user may want and
need are available from the maker on the "broadcast" models
but not on the multitrack models from which he must select
--- Circle 22 on Reader Service Card
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Co)
20,000 copies in print
ofis'
Fourth big printing of the
definitive manual of recording technology!
'John Woram has filled a gaping hole in the audio litera
ture. This is a very fine book
I recommend it highly.
-High Fidelity. And the Journal of the Audio Engineering
Society said, A very useful guide for anyone seriously
concerned with the magnetic recording of sound.So widely read ... so much in demand ... that we've had
to go into a fourth printing of this all- encompassing guide to
every important aspect of recording technology. An indis-
pensable guide with something in it for everybody
it is the audio industry's first complete
handbook on the subject. It is a clear. practical,
and often witty approach to understanding what
makes a recording studio work. In covering all
aspects, Woram, editor of db Magazine, has provided an excellent basics section. as well as more
in -depth explanations of common situations and
problems encountered by the professional engineer.
It's a "must" for every working professional ... for
for every audio enthusiast.
every student
to learn.
8 clearly- defined sections
18 information -packed chapters
I.
The Basics
I
he Decibel
Sound
II. Transducers: Microphones
and Loudspeakers
Microphone Design
Microphone Technique
Loudspeakers
Ill. Signal
Processing Devices
Echo and Reverberation
Equalizers
Compressors. Limiters and
Expanders
Flanging and Phasing
IV.
Magnetic Recording
Tape and Tape Recorder
Fundamentals
Magnetic Recording Tape
The Tape Recorder
Noise and Noise Reduction
Noise and Noise Reduction
Principles
Studio Noise Reduction Systems
V.
VI.
Recording Consoles
The Modern Recording
Studio Console
VII. Recording Techniques
The Recording Session
The Mixdown Session
VIII. Appendices
Table of Logarithms
Power. Voltage. Ratios and
Decibels
Frequency. Period and
Wavelength of Sound
Conversion Factors
NAB Standard
Bibliography
Glossary
r
SAGAMORE PUBLISHING COMPANY, INC.
1120 Old Country Road. Plainview, N.Y. 11803
Yes! Please send
copies of THE RECORDING
STUDIO HANDBOOK. $37.50 On'15 -day approval
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Address
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(In N.Y.S. add appropriate sales tax)
Please charge my
Use the coupon to order your
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for his multitrack production room. Or worse yet, the manufacturer may really only make multitrack studio consoles
which are then changed in minor ways for broadcast production
use.
INPUT MODULES
While the broadcast production user may require a variety of
input modules, including stereo input modules, these are
rarely offered on the typical multitrack music console.
The overall ergonometric design of the typical multitrack
music console is for a very limited range of functions. The
distance and relationships between operating controls, the
relative positioning of metering and the overall physical areas
of the controls are seldom optimized for broadcast production
use. However, many other features, such as the basic low
profile, the extensive equalization, metering and the auxiliary
outputs are well suited to production.
It might be perfectly adequate to plan a given broadcast
multitrack production room around an 8 bus, 24 track monitor
console with direct outputs available. However, this type of
console can be difficult to locate in the music recording console
field these days except on a custom basis. Customers would be
urged to "buy the full 24" when they might not need it.
Similarly, it can be difficult to secure a top quality 8 bus,
8 track monitor console for production without ending up with
a 16 or 24 track frame and a lot of blank panels.
It is thus proposed that although modern broadcast production studios may require 8. 16, or 24 tracks of recorded audio,
the associated console designs are not necessarily best derived
from recording studio consoles with the same number of
tracks. The increasing sophistication of techniques points
towards recording -type consoles, but with significant differences in the mechanics and audio architecture of the design.
One attempt to resolve this problem of making available
features of both conventional and multitrack based consoles
for broadcast production is to be found in the new Studer 900
console series described below.
The emphasis in the design of the consoles is on providing a
mixing system that can be configured for both types of applications discussed above, utilizing standard modules based
either on a conventional or multitrack monitor type format.
An additional emphasis is placed on making a system that
fulfills custom type requirements in recording and broadcast
without requiring custom metalwork and frame design. Some
basic considerations and specifications of the system are
given below.
Mechanical Considerations -The basic low profile of the
console allows for either multitrack or production control
room use. The frame itself is modular, allowing for many
variations in size within the same series. This means that
designs from 12 inputs and 2 outputs to those with 50 inputs,
24 buses, and 24 track monitor can be achieved economically
within the same system. In a multi -console facility, consoles
of widely different applications would use the same basic
spares. In addition, the middle configuration consoles so
important in broadcast production (4-16 buses, with or without
multitrack monitor) are an integral possibility within the
system, not just "odd" sizes stuck between two distinct console
lines.
For uses in both very large recording versions and remote
vans, the compact basic module width of 40 mm retains
manageable size. In addition, master output modules and
monitor modules are dual density, allowing for very compact
but still highly accessible units. Various combinations of
metering are available including very compact VU or peak
reading bargraph types also mounted in the dual- density
arrangement. In multitrack versions. the track assignment
buttons share the overbridge with vertical bargraph metering.
Track metering is thus placed conveniently over the associated
monitor/ output sections and the console depth is reduced by
the positioning of the assignment buttons.
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are from contributors to recent issues of PZMemo. a newsletter of PZM ideas published by Crown.
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co
(u
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on Reader Service Card
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Figure
2. A
mono mic line input module.
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3.
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The VCA fader/limiter module.
Electronic Considerations -Any new generation of consoles
must be aimed at performance compatibility with digital
recorders and transmission systems. In addition to newly
designed minimum path signal electronics and such options as
transformerless microphone inputs, other aspects include
transformerless tape line returns and a newly designed VCA
system (described further below) for optimizing noise considerations.
Because of the wide variety of application of the series, the
module selection is particularly diverse.
Input Modules -The basic Mono Input Unit has four inputs.
These include MIC, which is actually a variable level input for
microphone to line level signals from -70 to +20 dBu. The
input is floating and balanced with switchable phantom powering and may optionally be transformerless. The next input is
LINE, which is a transformer coupled line input with a variable
trim. The third input is TAPE, which is the primary signal
input for mixdown application and is transformerless and
balanced. The final input is GEN, which is an auxiliary tone
insert input to the individual channel.
The equalization available is quite comprehensive but
compact due to the use of switch/ pot combinations and push pot combinations for many functions. The two sections include
an HP /LP filter section with switched bypass for each filter,
which has a 12 dB/ octave slope and variable frequency. The
second section is a four -band semi -parametric equalizer with
shelving or peaking selection on the extreme frequency bands,
and a choice of Q factor on the inner frequency bands. Of
course, boost/ cut amount and frequency are continously
variable.
The module has the provision for stereo or quad panning
and a direct feed switch. Auxiliary output feeds from this
module include three mono auxiliary outputs, each switchable
pre- or post- fader, and an additional stereo auxiliary output.
On some console versions, an additional four auxiliary outputs
will also be available on the upper switching modules. Other
features include a silent mute and a user-assignable switch
with LED indicator. An internal preamp overload indicator is
also included. The solo buttons commonly found for channel
solo are mounted external to this module.
There are several Stereo Input Modules available with
features similar to the basic Mono unit. These include a Stereo
Microphone Input unit, with equalization, a Stereo High
Level Input unit and a Stereo High Level Input unit without
equalization.
Switching Modules -On console versions with up to four
buses, the input module contains all the necessary assignment
switches. On larger versions, this switching is continued up
vertically on switching modules in the overbridge section.
Even in multitrack uses, this format helps to easily identify
buses 1-4, which may be used in special mixdown options.
Fader Modules- Because these units match the various other
modules for functions and because of the availability of the
VCA system, there are a number of different fader modules,
including those which contain master output line amplifiers,
part of the very compact design of the console system. The
basic single knob fader may be mono or stereo control. Below
each fader are two pushbuttons to control the PFL /channel
monitor solo and APL (after -pan listen) /channel positional
solo functions. In addition, there is a separate monitor section
solo function on multitrack monitor versions.
The dual fader unit corresponds to the stereo fader, but has a
separate fader control for each channel. Another version is the
master fader which contains faders, line amplifiers and an
optional VCA -based limiter.
VCA System -A newly-designed VCA system is available in
the console series. This new VCA is designed and built by
Studer and has at least two interesting aspects. I) Excellent
long -term temperature stability makes periodic adjustment
unnecessary and accordingly, there is no trim pot for this
re- alignment; 2) The VCA operates in either class A or
www.americanradiohistory.com
class AB mode, depending upon an applied control voltage.
In the design of the VCA fader modules, these possibilities
have been utilized to provide the following action: depending
upon the applied input level, the optimum operating point
between class A and AB is selected. The mutual dependence
of noise performance -versus- distortion is shifted in the
direction of large dynamic range if a very low level signal is
applied, and in the direction of minimum large signal distortion
in the case of a high level signal.
The VCA fader system is available for external control by
any DC control automation system, and additionally provides
the following features: Up to ten DC control groups can be
created. If no group is selected, but both the MASTER and
LINK pushbuttons are selected, there will also be created a
separate Stereo Group independent of the other groups, with
the channel to the right. Muting control of channels is available.
The built -in limiter /compressor also utilizes the VCA as the
control element. The ratio can be adjusted from 2:1 to 20:1.
It increases continuously from I:1 to the selected value in a
soft transfer zone. The release control allows adjustment of
the recovery time which is also program controlled depending
upon the initial level.
Monitor Modules -Tape Monitor modules, when specified in a
system, are dual density (two channels /40 mm) units with
normal monitor functions plus three -band equalization,
four auxiliary feeds, and provision for reversing the functions
between the rotary monitor level control and the linear fader of
the master output. A monitor solo function is also included.
The monitor section can be group source switched between
console output or tape output.
-A
console master Remix mode is provided on
Remix Mode
multitrack versions which will switch channel inputs to tape
and switch mixdown and multitrack metering. ln addition,
mixdown can be achieved either through the channel section
or through the monitor section. On non -multitrack monitor
versions, the Remix function still exists to switch channel
inputs to tape.
Studio Monitor & Talkback Unit -This module contains
switching and talk back control for studio monitoring from
many sources, as well as bus amplifiers for solo functions
within the console. Extended talkback facilities can be supplied
for remote van applications or others where needed.
Control Room Monitor Unit -Up to 15 monitor sources may
be selected. Various housekeeping functions, such as monitor
level, phase reverse, dim, headphone jacks, etc. are included in
this module.
Auxiliary Master Unit -Each group of four auxiliary outputs
has an output master unit including high -pass filter equalization and mute and solo functions.
Other Modules -As part of the system for dual use, other
modules include Direct Input to Auxiliary and Direct Input to
Monitor modules, special Switching Modules, including
master clear feed versions, Input Selector Modules for multiple
source selection beyond that already available, and a set of
Remote Control & Timer Modules.
SUMMARY
Even though current multitrack console design practice has
resulted in an audio and physical console architecture that is
overly specific to a single use, it is possible to integrate the
various requirements of multitrack music recording and
advanced broadcast production in one console system which
can then fulfill requirements in both of these overlapping
areas.
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The MX -4 includes signal -present and
overload indicators for each band. At a
glance it tells you how well it's working.
Precision stepped rotary switches select
18 dB/octave Butterworth filters. Easy to set. Easier
to reset.
XLR balanced, or phone -jack unbalanced, input/
output Extra XLR input connector for "daisy- chaining:' BAR-in polarity switches on XLR outputs.
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www.americanradiohistory.com
NIGEL BRANWELL
Dy nam Ic Processing
Part II
Author Branwell continues his discussion on dynamic processing,
focusing in on the concept of multiband processing.
PART I of Dynamic Processing, multi -band processing
was touched on briefly as a method of achieving greater
gain reduction than broadband processing will allow due
to modulation effects.
The concept of multi -band processing is not new; as far
back as the early seventies, Audio and Design introduced its
first band -splitting systems and other systems may, indeed,
have preceded these.
MULTI -BAND IN PRODUCTION
Conventional compressors and limiters attenuate equally
throughout the audio frequency range as the gain reduction is
triggered by a signal at any frequency exceeding the preestablished threshold. Hence, high level, low frequency signals
(e.g., organ, drums, timpani, or bass) are frequently impossible
to compress or limit without also modulating high frequency
program material or ambience. This can be noticeably degrading.
Nigel Branwell is vice president
Recording Inc.
of Audio +
Design
The ear detects gain change by its modulation of other signal
content, particularly ambience. Normally, using close- microphone techniques and compressing individual instruments
(or groups of instruments) robs the ear of an external reference
so that considerable compression can be achieved without being
apparent. However, there are many times when such a technique is not possible: for example, problems arise when
transferring the finished recording from one medium to
another (e.g., tape -to-disc, cassette, optical film track, video
tape, and broadcast transmission).
In the case of disc, difficulties are often found in the low frequency region, giving rise to tracking problems. For high
speed duplication of cassettes, it is usually the sibilant area
that will spoil an otherwise good copy. In optical track, video
tape and broadcast transmission, it can be a combination of
the above plus the restrictions imposed by the limited dynamic
range of the new medium compared to the wider dynamic
range of the original. In the case of broadcasting at least,
there is often the requirement to maximize apparent loudness
and impact. Bear in mind that at this stage, a final balanced
program is being processed -usually the result of many hours
of hard work by producer, engineer, and artists resulting in
what they ideally hope the listener will want to hear. It is the
production engineer's job to see that it gets over to the final
medium with the minimum change or losses.
Very often, in attempting the transfer, the choice is to either
lower the modulation level on the new medium or insert fixed
www.americanradiohistory.com
Limit threshold
+8dBm
rsetting up (correctly adjusted)
i
Unity gain in return loop
(Below limit threshold)
-
Figure 1. Here, the high -pass filter has been used to
select the low -frequency content (shaded) which is routed
to the limiter. After limiting, it is re- combined to provide
a flat response at the main output.
flat response at main E500 output.
Figure 2. The output response, with the limit threshold set
at 0 dBm.
gives shelf -boost
Limit threshold
OdBm
Figure
3.
The effect of gain in the send /return loop.
This publication is available in microform.
University Microfilms
international
Please send additional information
for
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Street
300 North Zeeb Road
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London W1N 7RA
England
W
attenuation (equalization) in the troublesome areas -which
permanently degrades the signal. The ideal answer when
minimal modification of the original is desired is Dynamic
Equalization; this is the selective and momentary attenuation
of frequencies determined by program level at those frequencies
only. Note that this is nor the same as having a frequency selective "side -chain" (or control voltage) in a limiter, where
the whole program is still modulated and is thus only really
useful on individual tracks.
The problem portion of the signal must be split off from the
rest of the program and then added back at a later stage after
processing. Equalization is then present only above certain
levels (using a compressor or limiter) or below, using an
expander (or gate). At all "normal" levels the system response
is quite flat; the program is therefore momentarily modified to
suit the new medium, as necessary.
While the advantages of dynamic equalization have been
appreciated for some time, the circuitry required to split the
bandwidth in such a way that it can be accurately reassembled
is complex -especially when coupled with the need for versatile
selection facilities. Taking Audio & Design's Transdynamic
Tri -band processor as an example, the unit first splits the
incoming signal into its hi pass band pass / lo filter combination (6 dB or 12 dB per octave) as defined by the turnover
points selected in the crossover and provides a line level send
to, and return from, any external audio processor for each of
the three frequency bands. If the send is directly linked to the
return (unity gain), the signal appearing at the main output
will be identical to that at the input and any action taken in
the filter sections has no effect. Should the loop be broken,
the unit then acts as a static equalizer, providing high- and low pass and band -pass filters. Providing some fixed amount of
Figure 4. The E500 Band Processor integrated into an
operational system. The output from the master tape
is fed into the main input, and the main output is connected
to the copy tape machine. The external send/return
connections are routed to an external compressor/
limiter, etc.
RETURN
INPUT
ES00-RS
Figure 5. The Transdynamic Tri -band Processing System.
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attenuation in the loop (e.g., IO dB) will result in a shelf-type
equalization (with 10 dB shelf). These are passive uses of the
device. Its real benefits can be realized by inserting a level
control device in the external loop (limiter, compressor,
expander) of any or all three bands. Equalization will now be
dependent on program level and the amount of attenuation
in the selected region only will be determined by the program
level relative to the threshold setting of the external processor.
In the case of the Transdynamic, the crossover filters are
variable between 75 Hz and
kHz on the Low -Pass -to-Mid
section, and between 1.1 kHz and 15 kHz on the Mid -toHigh -Pass section. With a compressor-limiter- expander on
all three bands, significantly higher level can be achieved
while retaining a flat dynamic response or, if desired, introducing dynamic equalization to suit the following medium
(more on this when discussing broadcast applications).
Note that the above applications requiring the compressor limiter- expander as an external processor are, of course,
solutions to level control problems of one type or another.
It is equally valid to use a multi -band system to define a group
of instruments or voices for external processing via time delay
(phase, flange, ADT etc.) or other effects equipment. It is even
possible to add echo equalization and double tracking post
production and thus add selectively to a master tape.
1
MULTI -BAND IN BROADCASTING
Here, the same principles of multi -band processing
as
dis-
cussed above are used. The goals however may be quite differ-
ent. Broadcast stations, particularly radio in the United States
(and perhaps soon in Britain and Europe), tend to concentrate
on particular types of program material (format) and seek to
engineer a station "sound" that will help put them ahead in
terms of the competition (i.e., audience rating). Often this has
turned into a question of who can be the loudest station in the
coverage area for a given format which has in turn led to
"loudness wars" between competing stations; fortunately,
increasing evidence supporting the theory that loudness per se
does not lead to increased listeners is mounting.
Broadcasting is about communication (as is the entire audio
industry); to communicate, it is necessary to remain in touch
with the listener. Audio processing is all about communication;
it seeks to sensure that the listener is A), attracted to the
station as he flips across the dial, and B), that the signal is
maintained at the highest quality while remaining at all
times intelligible to the listener. This means having a sufficiently
high average level in order to stay above the ambient noise in
the listener's environment with low level signal still audible
at typical listening levels. It could be that the same program
is transmitted on both AM and FM, being processed differently
to meet the requirements of the two transmission mediums
(given the current poor quality of AM receivers), and the
restricted listening dynamic range available while travelling
and the wider dynamic range in the listener's home.
M ulti-band processing, in this case with processors operating
on each of two or more frequency bands, will increase energy
levels without the modulation effects of a broadband processor,
distributing the frequency spectrum dynamically according to
the format, the transmission medium, and probably the
program director too! As before, the process of band -splitting,
the quality and flexibility of the band compressors, and that
of the final peak limiter will determine the overall sound
quality that can be achieved.
A three -band system is more than adequate for purposes of
obviating modulation effects; high compression can be
achieved for AM (more so if the external processors have auto
release networks for AGC action) and overall gain in excess
of 16 dB can be successfully processed on 6 dB/ octave crossovers without modulation problems. The use of 12 dB/ octave
PROVEN PERFORMERS
FOR STAGE AND STUDIO
SB-36W
The preference for
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range: 48 " -72"
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The ideal for music
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Height range:
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Height Range:
26 " -66"
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SB -100W
The quality standard of the broadcast industry.
Optimum positioning -range and
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microphone -follower
-by single knob
operator -control.
Precision balanced, oversized
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Boom length: 110"
-Height range:
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e
ATLAS SOUND, DIVISION OF AMERICAN TRADING AND PRODUCTION CORPORATION
10 POMEROY ROAD, PARSIPPANY, NJ 07054 (201) 887-7800
A
N
Circle 45 on Reader Service Card
www.americanradiohistory.com
DIR -MIXTM
PROCESS
t
m presso'
6dB
Input
Figure 6. The DIR -MIX'" process for wide range dynamic
processing using a Transdynamic Processor and three
F760X Compox- Limiters.
ST
PRECISION
O
MAGNETIC
TEST TAPES
STANDARD TAPE LABORATORY, INC.
EDEN LANDING
ROAD
5
RAY SAN.
RNIA 94545
loudness), it can be easily overdone; a degree of increase in
low level high frequency content relative to high -level mid/
low- frequency content can be subjectively attractive. Many
kinds of dynamic response can be shaped; by increasing gain
on high- and low- frequency sections and applying greater
compression so that the high level output remains flat, low
level signal will increasingly be lifted as signal level falls,
giving a dynamic Fletcher- Munson loudness response.
By selecting a limit or tight compression slope for the mid frequency section and a soft ratio on low and high frequencies,
presence signals can be maintained at high level until full
recovery, while LF and HF is reducing. This gives a dynamic
presence peak at mid amplitudes that progressively flattens
at high and low levels. When applied to the 3 kHz frequencies,
this can be useful where bandwidth is restricted to 5 kHz or
I
kHz (AM). For subtle operation on wide dynamic program,
soft ratios (1.5:1 or 2:1) will be found the best; using tighter
slopes (5:1 or 10:1) will maximize loudness.
Using one external unit to provide selective HF or LF
limiting; or two for selective HF and LF control prior to
master limiting. excessive HF or LF (or both) signals that
would normally trigger limiting are momentarily attenuated
while the master limiter, if operating at all, responds to the
mid -band signal.
With this approach, the signal retains its full dynamics with
only occasional and momentary modification, which is totally
inaudible except on A/ B comparison under hi -fi monitoring
conditions; yet the transmitter output can effectively be
increased by as much as 6 dB.
The selection of de-emphasis in the master limiter will act as a
peak indicator of excessive HF content which can be dynamically controlled in the external compressor. Thus, selective HF
compression can be adjusted to give dynamic de-emphasis
given the selection of a comparable turnover frequency to
that of the transmitter pre -emphasis, on the MF crossover.
For wide dynamic formats, discussed in Part 1, the direct
signal can be paralleled with the multiband processed signal
1
Frl
2640
filters will allow even greater density but are of more use
where overall bandwidth is severely restricted (e.g., 5 kHz in
the UK on the AM band). Other than modulation effects, the
main problem with any form of compression is that source
noise is increased by the gain set under quiescent conditions
in the compressors; in any well- designed compressor, source
noise will always exceed system noise (even from digital tapes)
assuming IO dB overall increase in average level. Expanders
can help here by operating below wanted signal to attenuate
each band progressively as level falls. Given sensible crossover points in the filters, expanders will work well on orchestral
music and are ideal for use in AM applications where a substantial amount of compression is often used. This is a form of
dynamic equalization -as the high frequency content falls, or is
absent, the effect will be that of a 6 d13 octave (or 12 d B/ octave)
shelving filter acting on low amplitudes only.
The gain change in multi -band processing for broadcast use
should be identical throughout the input range in each band to
retain a flat response dynamically. In practice, the gain change
or compression will vary from moment -to- moment in each
band dependent on the program energy levels at the time. If a
sensibly flat response is required, ratio, threshold, release, or
crossover point of the filters can be adjusted to spread the
compression over comparable ranges in each band section.
Gain change in each band is then closely matched to retain a flat
dynamic response. Note too that fired parameter multi -band
systems will, more often than not, act as dynamic equalizers
to the degree that gain change varies between bands (the more
bands, the worse the problem). This can lead to a total corruption of internal balance, a flattening of the sound and worst
of all, listener fatigue. Some systems also resort to band
coupling to reduce distortions of dynamic response. Here, the
result will be increased susceptibility to modulation problems,
thus defeating the original purpose of band -splitting.
While subtle variations in dynamic response can lead to an
enhancement effect on the signal (apart from any increase in
14151
ro
a
Circle 31 on Reader Service Card
www.americanradiohistory.com
Figure 7. The Audio & Design Transdynamic Processor
and the model E500 Band Processor.
producing the effects of voltage addition. Mixing the compressors in at low level (some 25 dB down with reference to
peak signals -which would mean 25 dB -plus compression at
high level), the compressors predominate at low level. adding
gain, while as level increases, the direct signal increasingly
takes over and predominates at peak levels. Added equally at
low level, the gain will increase 6 dB returning to unity at
peak levels. FIGURE 5 shows the effect of adding a compressori
expander combination slope to the direct signal level. The
advantages of this system outweigh any disadvantages substantially -with the compressors added 25 dB down on the
main signal, any noise or distortion is a further 25 dB down
on normal. Any ratio selected is further softened (2:I becomes
1.3:1 approx.) and compression is spread quite smoothy
throughout the input range so that internal dynamic relationships are more accurately retained. This type of system works
very well for wide dynamic program retaining the crystal clarity
of the upper dynamics without the lower level compression
being noticeable.
As an experiment. this system was tried on a sizeable
orchestra. Having been recorded flat. the tapes were then
replayed using the system described above. Not one musician
realized that his/ her instrument had been modified in any way.
It works and is a system that should be of interest to classical
and jazz stations as well as an operational balance aid when
recording a difficult wide dynamic program.
The advent of digital mastering is an interesting prospect
for use of a quality multi -band processing system. The major
problem, other than modulation effects, is the way source
noise is increased with compression. With a digital recording
capable of accommodating the widest input dynamics, the
ambience of the studio will always mask any electronic noise.
Subsequent processing through a multi -band system of suitable
quality will only increase ambience rather than electronic noise.
Audio & Design produce a number of systems employing
multi -band techniques. The latest is the Transdynamictri -band
system, some features of which have already been described.
Other features include a high -quality wide -band master limiter
and optional clipper following recombining of the three
frequency bands, selectable 0, 25, 50 or 75µs pre -emphasis in
the control circuit rather than the audio path, adjustable
assymetry for AM applications, optional bandwidth contouring filters at the main inputs and outputs. LED bargraph
metering indicates gain reduction in the master limiter- clipping
action (if switched in) and return levels from the external
processors. A pulsed pink noise generator facilitates initial
The E500 Band Processor's most pertinent features are a
Selective Band Processing section on each of the two channels
comprising two fourth order filters, one High Pass and one
Low Pass -both continuously variable over a frequency
range of 100 Hz to 10 kHz. Each filter can be cancelled to
allow operation individually or together when a mid -band
frequency component is selected for processing.
HOW DO YOU
MEASURE TAPE
TENSION?
If something SOUNDS FISHY it
may be your fish scale approach to
measuring tension.
Tentel Tape Tension
is designed to diagnose problems in your
The
Gage
magnetic
tape
equipment.
Virtually all recorder manufacturers use and recommend the TENTELOMETER
for use with their equipment.
TENTELOMETER
measures
tape tension while your
transport is in operation. so you can "see" flow your transport is
handling your tape. high tension causing premature head and tape
wear. low tension causing loss of high frequencies. or oscillations
causing wow and flutter. Send for the Tentel 'Tape Tips Guide
The
The T2 -H20
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sells for S245
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complete
1506 DELL AVENUE
IENTEL
CAMPBELL. CALIF 95008
1408) 379 -1881
1- 800-538 -6894
calibration of the system.
Circle 33 on Reader Service Card
www.americanradiohistory.com
v
VLADIMIR NIKANOROV
A Computerized Console
For Radio Broadcasting
Automatrix is a new broadcasting console which uses modern
technology to produce professional-quality audio, combining the
features of manual and automated operation.
ONVENTIONAL "AIR - CONSOLES for radio broadcasting
are a product of long established broadcast studio
procedures. Generally there are two types of studio
operations in which a typical broadcast air console
plays the main role. One is a manual operation, another is an
automated operation.
In the manual operation, the console is basically a single output summing amplifier. Remote start- and -stop controls are
often a part of the console, as seen in FIGURE L Turntables,
open reel playback machines, cartridge playback machines, and
microphones feed the individual inputs of the board. The
console operation opens and closes input faders and activates
the sources by the remote control system on the board.
In a typical automated operation, an air console is basically
used as a microphone mixer to control levels and as an audio
monitor. The console provides live audio source to the automation system, which in turn performs as a summing amplifier and
as a programmable switcher, alternating the audio sources in
the desired sequence (FIGURE 2).
v
Vladimir Nikanorov is technical director
Broadcast Consultants. Tenafly. NJ.
of Bonneville
Both systems have their advantages and shortcomings. The
manual operation requires constant and intense attention.
Routinely, records are spun, tape decks are fired up, and
announcements are made. usually by the same person who is
also expected to pay attention to levels on VU meters and
to read the modulation monitors. The quality of the show is
greatly dependent on this operator, whose skill and individuality really makes this type of operation an art. Some advantages of the manual operation are flexibility in selecting audio
sources, spontaneity, and sometimes the involvement of the
announcer's personality.
However, when the operation is partially automated, the
intensity of the manual real -time show can be diluted over an
extended period of time. This may remove the "art" and
individual aspect (along with a great number of errors) from
the operation.
The fully- automated broadcast operation usually uses
recorded announcements, and no provision is made for
handling live audio or for playing gramaphone records.
Typically, the tape reproducers become part of the automation
system, which consists of huge I9 -in. racks full of equipment.
Open reel and cartridge players are hardwired to the system
switcher whose function is to connect one source or another to
a program amplifier. The output of the amplifier feeds the
air board. The switcher normally is controlled by a microprocessor, which in turn is fed with computer data.
Selecting an audio source becomes a slow and premeditated
process which usually requires a separate operator and, in
many cases, makes the manual operation appear more desirable
www.americanradiohistory.com
because
of its flexibility.
Despite the fact that automated broadcasting relaxes the
on-air studio operation, it is potentially hazardous because
of the system's structural design. For example, if a snitcher
or program amplifier fails, then the station is off-the-air
regardless of the state of the audio sources.
THE AUTOMATRIX AIR CONSOLE
A desire to achice the flexibility and cost- effectiveness of
manual operation, together with the "error free" advantages
of automated operation was the design philosophy behind
the Automatrix computer- assisted air console.
The heart of the console is the Audio Main Frame Unit
(AMU), which -as in a regular air console -is nothing but a
complex summing amplifier with a single audio program
output. The audio output is fed to outside circuits: radio
broadcast processing equipment, exciter and transmitter.
Typically, 16 stereo sources can be connected to 16 individual
inputs of the AMU. The number can be extended up to 32
sources.
The Machine Control Unit (MCU) provides automatic
start -stop functions for the audio sources. The Board Module
Group (BMG) contains monitors, VU and PPM meters, and
a manual remote control switchboard for the 16 sources.
So far, the block diagram is quite similar to a typical
manually- operated air console. However, similarities of the
console structure with conventional designs start and end
with the fact that it is made to deal with audio sources. Unlike
most of the air consoles, it functions automatically, taking
commands from the System Control Unit (SCU), the next
block on the block diagram.
The SCU feeds 8 -bit data to the D1 A converters on each
individual audio input card. The D/ As supply variable DC
voltage to the VCAs (Voltage Controlled Amplifiers) on the
audio input cards, which in turn change the audio levels.
The levels can be brought to a predetermined value or completely turned off.
The next and the last block is the Programming Computer
Group (PCG). The PCG computer controls the SCU functions
over a desired period of time.
AUDIO MAIN FRAME UNIT (AMU)
the AMU contains 16 or 32 stereo cards. A block diagram of
a typical audio input card is on FIGURE 3.
The input is active balanced and is based on an instrumentation amplifier feeding two identical VCAs; one for program
and one for cue outputs. Audio level information for the
VCAs is carried by an 8 -bit data bus from the SCU to the
D/ A converters. The D; As are calibrated with a reference
generator which makes possible precise level adjustments.
The levels can be pre -programmed or changed by the BMG
during the course of the program if necessary.
Outputs of the input cards are fed into the line amplifier
which is terminated by a summing program amplifier on the
program output card.
Start -stop of audio sources is done automatically in different
ways, depending on operational requirements. An end -of-event
circuit (EOE) senses the 25 Hz tone routinely used to stop -start
open reel players or a 150 Hz tone for cart players. The circuit
can be tuned either way.
Detection of the tone will cause the comparator to send a
I -bit digital signal to an addressable buffer circuit and on to
the SCU via the data bus. Then, depending on what it was told
to do previously by the computer, the SCU will react and
activate the next desirable event.
Another way of controlling events is based on the absence
of audio. In addition to feeding the audio program to the
external broadcast chain. the left and right summing amplifier
outputs are summed and rectified. The resultant DC is converted into a digital word which is constantly read and compared to a reference standard. If a no- signal condition is
detected, the SCU will activate a new event. An optional
alarm will indicate silence.
SYSTEM CONTROL UNIT (SCU)
A block diagram of the SCU is seen in FIGURE 4. The SCU
is the intelligence (limited however) inside the console. It
contains 4K EPROM (Erasable Programmable Read Only
Memory), 1K RAM (Random Access Memory), the microprocessor, control, address and data buffer drivers and standard RS232 communication interfaces which are linked to the
main Progamming Computer Group (PCG) and to the
console's Board Module Group (BMG).
The heart of the SCU is the Microprocessor Unit (MPU).
It manages the following operations:
I. Operations within the AMU, for controlling the VCA
2.
setting.
Operations within the MCU: machine starts and stops, level
sensing and silence sensing.
3. Runs program which scans EOE (End -of- Event) buffers.
4. Reads A/ D converters, starts A, D silence-sensing con-
verters and checks status of RS232 interface registers.
Depending on the system operation software, which operates
the EPROM (Erasable Programmable Read Only Memory),
the MPU starts and stops sources selectively, sets up the
levels, makes sure that audio of a source is actually at the
system output and keeps in RAM (Random Access Memory)
what should be done for the next events.
THE BOARD MODULE GROUP
A block diagram of the BMG is shown in FIGURE 5. This
section of the system is designed specifically to provide the
extra flexibility required for automated broadcast operations.
The BMG consists of the following modules: a microprocessor -based console control module which can act
independently or in conjunction with the SCU through the
RS232 interface and a special functions module. This module
and the MPU of the group act in the same manner as the SCU
when it senses the levels and sets the sources on and off. The
special functions module can be used for live cross fades or
predetermined fades of any duration, which makes it possible
for live cross fades or for example to play gramaphone records
either in the traditional manual fashion or in automated mode.
One of the most important functions of the console control
module is to convert the automated operation into manual.
This is done by a single manual /automated push button. At
the moment of the conversion, 16 pairs of push buttons on the
front panel become start /stop remote controls for the audio
sources. The Board Module Group also contains microphone
modules, VU and PPM meters, earphone feeds, manual cue
amplifier, network access module and telephone (patch)
module.
THE PROGRAMMING COMPUTER GROUP (PCG)
A computer in the system is used as an extended intelligence
linked to the MPU of the SCU. The computer, which is interfaced with the MPU by standard communication interface
RS232, constantly updates the system and typically has a
capacity of remembering about 10,000 events.
Standard operational software programs are written for the
Apple II -Plus computer and is used with pre- edited events
sequencing and record keeping. Alternate or extended
computer systems can also be used, in which case the software
and the interface should be modified.
BROADCAST PROGRAMMING
Conventional broadcast programs are based on an audio
source schedule. This will include level, fade and cross-fade
information, cues for live announcements, and assists for
semi -live functions such as gramaphone record playing.
The programming is compiled by the main computer keyboard and CRT, located on the right side of the console.
The on-air announcer has instant access to the keyboard which
is in fact a part of the air console. Thiseliminates the additional
operator normally required in conventionally- structured
automated operations.
www.americanradiohistory.com
m
AUTOMATED SYSTEM
Ren
urn tiblr.
I
Open reel
playback
machine
ntr
Iene
controls
.t.
r
Cartridge
Open reel
playback
machine
Program
amplifier
Program
-7 output
Cartridge
0-
Program
Swil her
Microphones
output
r
Swit her
o
control
Microphones
o
Figure
1.
Block diagram of
a
2.
Figure
3. The
typical manual operation.
The programming can be done either in English language
(i.e., BASIC) or in code. The program, which controls the
system in real time, is stored in the floppy disc memory and
is transferred in increments to the console's MPU, which in
turn will transmit the events instructions to the SCU drivers.
While transferring, the CRT can be used for a number of
different purposes. For example. it can provide indications
for up to 12 events in advance with a current event countdown
or it can display an analog clock with local time (the analog
clock is a standard feature of the Apple I1 -plus computer).
The disc's memory is also used to keep records of real time
events which are then transferred to hard copy. The printouts are used for the FCC logs. billing, etc.
or 15011,
tone sensor
AMX console).
Comparator
To SCU
Instrumentation
atop
Left
input
DA
RF
Left
program
output
filter
Left cue
output
From voltage
reference generator
N/
Neither AMX console resembles a recording desk or a
conventional broadcast console. They look more like space
age office desks with lots of controls on the left. a computer
keyboard and CRT on the right and a meter panel at the rear
center. The desk provides a working space for the announcer
operator who, by the nature of the procedure, has to deal with
a lot of paper work.
As mentioned before, the console is a modular structure.
The modular philosophy was carried all the way through.
Each group of modules constitute a bigger module, practically an independent box. This allows the separation of audio
and digital circuits, makes possible relatively easy expansion,
simplifies service and is cost effective.
The Automatrix console is marketed by Aphex Systems. Ltd.
The basic price of the model AMX 16 is about $15.000,
computer excluded.
Great attention is given to the station's environmental
circumstances. For example, both "high" and "low" ends of
the balanced audio circuits pass through RF filters before
entering or leaving the Audio Main Frame Unit (AMU) box.
The AMU is contained in one or two (depending on the
model) standard I9 -in. chassis which are mounted on the
lower back of the deck.
The input and output connections are on a barrier strip
terminal (about the only thing which is done conventionally
in the
Audio Mainframe Unit (AMU).
25
PHYSICAL DESIGN, QUALITY OF
AUDIO PERFORMANCE OPERATION
o
rn
Block diagram of a typical automated operation.
Figure
1)/A
From
dala
buffer
driver
f
D/A
Right
input
SAME AS LEFT
The inputs and outputs are active balanced, bridging;
however, external input- output transformer barriers can be
furnished if desired. Input- output clipping occurs at +27 dBV,
as referenced to 0.775 VRMS. For a typical broadcast input
level of +8 dBm, the head room of the board is + 19 dBm.
Signal -to-noise ratio is -80 dB, referenced to the nominal
signal level of +8 dBm. THD at the nominal level is 0.01 percent.
Frequency response is flat within 0.1 dB from 30 Hz to 30 kHz.
A number of the boxes containing primarily digital control
circuits are mounted in a separate I9 -in. rack. Among them
are the System Control Unit (SCU). the Machine Control Unit
(MCU) and the power supply. Actually, the Digital -Analog
www.americanradiohistory.com
To R\1G
--F RS232
interface
RS232
interface
Module (DAM) and the Board Module Group (BMG) are the
only modules that are plugged into the console mother board.
To P('(;
CONSOLE OPERATOR
I
LPROM
4K
RAM
IK
Clock
ontfol
data
Address
data
driver
driver
(1
V
W
W
To MCU
To
Data
hotter
driver
Í. v
V
To AMU
addressable
8-bit buffer
Figure
4.
The System Control Unit (SCU).
Figure 5. The Board Module Group (BMG).
Cue
VU
speaker
1'1'\1
earphone
From
program
or cue
From
From
MCU
Buffer.
SCU
RS232
tute face
Outside
feeds
Special
ProEram
manual
Cue
manual
functions
Mike
preamps
r'
From
AMU
When an announcer/ operator sits at the console, he or she
confronted with just one push button. This button switches
between manual and automated modes.
One of the drawbacks of regular automation systems, and
therefore of broadcast formats which are run automated, is
the difficulty of inserting anything within the preprogrammed
flow.
This is especially true in the case of instant live announcements. Short of bypassing the system all together, the announcements must be programmed ahead of time -which of
course takes away the spontaneity and value of timeliness.
The AMX completely overcomes this type of drawback.
When a need arises to go live or insert a new piece of programming into a regular format, the announcer simply pushes
the manual/ automated button.
The current event is halted by its dedicated remote stop
button and the new program is started. The microprocessor of
the Board Module Group remembers where the main program
was stopped. It is activated by the manual automated push
button and then switches all the source function controls to
manual mode. However, the end -of- events circuits (EOE)
remain activated. A switchboard of 16 or 32 pairs of LED illuminated push buttons becomes the manual remote control
board for the audio sources. If a source is activated, then the
LED on its dedicated start button is illuminated. Previouslyprogrammed operation will resume (if not re- programmed)
right from the place where it was left off at the return of the
button to the automated position.
The simplicity and quick switchover from automated to
manual and back in combination, with the ability to set
different levels with automatic fade and cross fade, allows
the operator to do neat things with insertions and spots. For
example, the console can be used for precise real time editing
and processing of commercials on the spot.
Actually, the potentials of the console may suggest an entirely
new style of radio broadcast production routine. A provision
is made in the design for the production to be done while on
the air. Cue circuits which duplicate the program chain can be
used for production, while the main chain runs the regular
material. Spots are known for bad recording qualities, and
especially for discrepancy in levels. In the new style of production, the commercial can be auditioned, and the levels logged
and changed when the spot is programmed at the computer.
The possibility of real-time preprogrammed edits or live
changes of levels and instantaneous change from and to
automated mode also opens new possibilities for formats
based on playing gramaphone records
Normally the DJ is involved in many mechanical operations
which consume his or her time "behind the scenes" and often
gets in the way of the quality of the announcements. Auto matrix can take over most of these time-consuming operations.
It will take care of starts and stops of the turntables and
tape machines, give a cue and open the announcer mike, make
the right choice at the right time (about the next source) and
will allow the announcer to prerecord and properly insert
commercials and portions of program which are difficult to do
live. The disc jockey will be left with only the function of
announcing, cueing the records and entering instructions in the
computer.
is
CONCLUSION
From
AMU
A computer assisted console with excellent audio quality is
not something unknown. The recording industry has used
automated consoles for multitrack mixdowns for years.
In the radio broadcasting industry, many automation
systems are nothing else more than glorified switching
mechanisms, which quite often do not satisfy the main purpose
of broadcasting, to produce high quality professional audio.
www.americanradiohistory.com
rr
441"!
Application Notes
MELVIN C. SPRINKLE
An Artificial Microphone
With the use
of an artificial microphone, obtaining Proof of
Performance measurements can be as easy as one, two, three.
IN I HE RULES & REGULATIONS governing standard broad cast stations, including those using the AM as well as
the FM system of modulation, Uncle Charlie requires
that every station make a Proof of Performance
measurement. This consists of a number of tests made on the
station's audio system and the transmitter.
Although the vast majority of stations originate their entertainment material from either disc or tape recordings, the
Proof of Performance regimen requires that the station make
an amplitude-frequency response using a microphone channel
as an input.
This requirement presents some problems for those who lack
proper equipment, but is a breeze for those with the right
apparatus.
In order to make the measurement, a sine wave signal is fed
into a microphone input -that is by far the most common
method. The input signal is measured if the station has a
sensitive AC voltmeter.
Unfortunately, however, this method is fraught with pitfalls
and one is more than likely to wind up with the wrong answer
for amplitude-frequency response.
There are several problems associated with this approach:
(I) all microphone preliminary amplifiers have a balanced input
and (2) a very minute signal is required to prevent overload and
distortion in this first amplifier: a typical value is I millivolt.
(3) The proper source impedance is not presented to the input
stage.
Lo
Mel Sprinkle is a principal in Sprinkle Associates.
Audio Acoustical Engineers
The vast majority of sine wave signal sources available on
the market are eminently unsuited for use in a Proof-ofPerformance measurement on a microphone input. In the first
place, most of them are unbalanced and the direct connection
of an unbalanced piece of equipment to a balanced circuit
without an intervening transformer is a "no -no" in good
engineering practice. It might be argued that quality. professional grade mike inputs do have an input transformer and
therefore one side may be grounded with impunity. This
euphoria is quickly dispelled by the realization that when a real
microphone is connected to the input, the system is then
balanced so that operating in an unbalanced mode is not
representative of things as they really are. To paraphrase
Walter Cronkite's famous tag line: that isn't the way it is.
The second difficulty encountered in the conventional
approach is getting the proper signal level. One audio oscillator
has a step attenuator with IO dB per step, but most of the
"garden variety" of oscillators have nothing more sophisticated
than a simple pot -and probably a linear taper to boot! With
such a control it is virtually impossible to attain a miniscule
output voltage on the order of one millivolt. The harried
engineer is between Scylla and Charybdis: on one hand is zero
signal and then, with the slightest rotation, the amplifier is
overloaded. Then there is the ever -present problem of drift,
especially with carbon controls, for after carefully finding the
correct signal level, one finds that the signal has now increased
or decreased by a hair's breadth and another round of setting
comes up. Never forget that the inexorable decibel laws operate
with minute signals in exactly the same manner as with larger
ones. Thus, going from one millivolt to two (as by control drift)
represents a 6 dB change in level, and this error is built into the
measurement!
www.americanradiohistory.com
0
ITEM
I
DESCRIPTION
Dual binding posts.
" spacing,
Red -Black
QTY
with mounting
I
hase
-
Microphone connector,
3
Transformer, 1,00 00012, one winding split and strapped
for 15052
4
Aluminum muu box enclosure
Resistor, fixed carbon. 0.5watt.'7552, ±5'Resistor, fixed. carbon, 0.5 watt, 16052. ±5'.
Resistor. fixed. carbon. 0.5 watt. 13052, 5'-Resistor, fixed. carbon. 0.5 watt, 20052. ' 5';
Resistor. fixed. carbon. 0.5 watt. 11052. '5
Resistor. fixed, carbon, 0.5 watt. I5052, '5
Resistor, fixed, carbon. 0.5 watt, 3652. ,_5''
5
h
7
8
9
10
I
I
3
pin male
I
I
I
1
Figure 1. The three parts of the artificial microphone: e
transformer, a resistive attenuator and a build -out network
which restores the 150 ohm balanced impedance.
IMPEDANCE MATCHING
In recent years the principle of impedance matching has
fallen into disrepute and this is, perhaps. unfortunate. Be
that as it may, there is an area associated with Proof of
Performance measurement in which impedance matching is
important. All well designed (in this author's opinion!) microphone amplifiers are equipped with an input transformer with
a primary impedance of 150 ohms. When a transformer
designer assigns numerical values to the impedances of the
windings, it does not mean that these are the actual impedances
of the windings. Rather, they are the values of source and load
impedance between which the transformer is designed, and
intended, to work. Thus, for example. a microphone input
transformer with a 150 -ohm primary and a 3.000 -ohm
secondary impedance rating, means that the transformer
should be fed from a 150 -ohm Thevenin source and, for the
purpose of measurement, terminated with a 3,000 -ohm
resistor. Since the output impedance (Thevenin source
impedance) of many audio oscillators is 600 ohms, it im=
mediately follows that this is four times the correct impedance
from which a I50-ohm microphone transformer should be
fed. Feeding a transformer from a source impedance greater
than the design value has two deleterious effects. As is well
known, the transformers low frequency response begins to fall
off when the primary inductive reactance equals the source
impedance. Thus for a higher source resistance, the 3 dB
rolloff begins at a higher frequency. The second effect is,
perhaps, not so well known. When a transformer is connected
to a source of AC, a current is drawn by the transformer which
sets up the magnetic flux in the core. This current is called
the "exciting current" and is non -linear. If the signal source
is a sine wave and the transformer is fed through a source
resistance. the distorted exciting current will produce a voltage
drop across the source resistance which adds vectorially to the
sine wave signal, producing a distorted voltage which is
impressed on the transformer primary and thus appears on
the secondary. To avoid this problem. a transformer should
always be fed from its rated source impedance. never from a
higher impedance. such as feeding a 150 -ohm primary from a
600 -ohm (four times) source. Exciting current distortion is
well known to transformer designers and references to it may
be found in transformer texts such as "Magnetic Circuits &
transformers- by the EE Staff of Massachusetts Institute of
Technology, John Wiley. 1943. pages 160 -173.
THE ARTIFICIAL MICROPHONE
The artificial microphone is a device which electrically
simulates a real microphone in signal level. balanced circuitry
and source impedance. It makes possible accurate and repeatable measurements in microphone circuits including amplitude frequency response. distortion and residual noise.
As shown in FI( ;RI 1. the device is. like Caesars Gaul.
divided into three parts. These are (I) a transformer; (2) a
resistive attenuator; (3) a build -out network which restores the
150 ohm balanced impedance.
The transformer is used in the high -level portion of the
circuit for two reasons: (I) with a relatively high level of
signal, the requirements of shielding are much less severe;
(2) the performance of many transformers is degraded with
very low values of input signal of I millivolt or less. The
transformer used is a repeat coil with unity turns ratio and with
one winding split. This winding has the sections connected in
parallel rather than in series with a consequent impedance of
150 ohms. The transformer must be of the highest quality that
www.americanradiohistory.com
rn
w
the budget can afford since it is the sole element which determines the amplitude -frequency response of the artificial
microphone.
The second section consists of two lattice attenuators
connected in series. The lattice network is an excellent attenuator for balanced circuits. Two sections are used -one of
24 dB and the other of 16 dB -for a total loss of 40 dB, or a
voltage attentuation of 100 times. The resistor values have been
slightly modified to fit the 5 percent preferred values which are
more readily available than the "funny" values turned up by
pad formulas. Two sections are used because, if one attempts to
design a single section of 40 dB loss or more, the values come
out so close to one another that resistors must be individually
measured on a bridge. It's the old story of the difference
between two large quantitites. Two sections are much better.
The third section has two series resistors which restore the
150 ohm source resistance needed. It will be recalled that the
impedance of a terminated pad is half of the termination.
The two resistors immediately following the transformer are
used to provide a 150 ohm source resistance for the pad, since
the signal at the input terminals is held constant during a
frequency run and therefore is a zero impedance point. These
resistors also provide a6 d B voltage attenuation at no extra cost.
The overall device has an attenuation, including the transformer, of 52 dB or a voltage attenuation of 398.11 times or a
ratio of 0.00251. Thus, if 0.5 volt is applied at the input terminals, the open circuit output voltage will be 0.00126 volt.
This theoretical value will be subject to some variation due
to the tolerance variation in the resistors. A unit constructed
with this circuit measured almost exactly one millivolt out for
0.5 volt in. Since one millivolt behind 150 ohms represents an
available power of -57.8 dBm, the signal from the artificial
microphone simulates the real thing. The circuit is passive and
linear so that signals at the output of other power levels are
readily obtainable by varying the input signal. The output
will vary in
a linear manner, dB for dB.
The author's unit was constructed in an aluminum "Mini Box" with the transformer inside. The pads assemble very
readily on a turret type insulating board. The input connector
is a pair of binding posts with % -inch spacing. The output is a
standard three -pin microphone.
Since there is a resistive termination at the output and
some 52 dB attenuation toward the input, a system's residual
noise level may be measured by sirhply removing the signal
source and terminating the input with a double banana plug
with a 620 ohm resistor between the two plugs.
A few tips on checking a completed unit may be in order.
The amplitude -frequency response of the transformer may be
measured easily if one has a voltage that is constant with
frequency behind a 600 ohm source. Such is the case with most
audio oscillators from the major manufacturers. If one is not
sure of the output impedance it may be readily measured.
Simply measure the open circuit (no load) voltage from the
oscillator. Then connect a 620 ohm resistor across the generator's output and remeasure the voltage. If the output
(source) impedance is 600 ohms, the voltage will be halved
(6 dB drop). The response is measured with a voltmeter
connected across the transformer's secondary. Don't worry
about balance here; the series resistors are further along
in the circuit.
To measure the output voltage (open circuit), one must
consider balance. The sensitive electronic voltmeters capable
of measuring one millivolt are invariably unbalanced. Thus
we use a 1:1 transformer or repeat coil ahead of the voltmeter.
Needless to say, a 'scope is mighty handy when checking
these minute voltages. It will quickly tell whether one is
measuring a 1,000 Hz tone or some stray hum that sneaked in
when you weren't listening! Electronic voltmeters have an
output connection using the meter amplifier and these are a
convenient place for a 'scope connection.
Most people think heart disease
happens only in the elderly.
It happens in children as well.
Things like rheumatic heart disease
and congenital heart defects. Each
year, nearly one million Americans
of all ages die of heart disease and
stroke. And 20,000 of them die from
childhood heart diseases.
The American Heart Association
is fighting to reduce early death
and disability from heart disease
and stroke with research, professional and public education, and
community service programs.
But more needs to be done.
You can help us save young lives
by sending your dollars today to
your local Heart Association, listed
in your telephone directory.
Put your money where
your Heart is.
American
Heart
Association
WERE FIGHTING FOR YOUR
www.americanradiohistory.com
LIFE
New Products & Servkes
REMOTE CONTROLLED
AUDIO ATTENUATOR
FSR. Inc. has recently introduced a
new remote audio attenuator specifically
designed for high quality sound systems.
Available in single (LDR -I) or dual
(LDR -2) channel, the units feature a
12 dB pad bypass already mounted on
pc board: no audio distortion and no
control induced noose. The audio attenuators operate off 24VDC power.
Mfr: ['SR. Inc.
Circle 46 un Read.r Service Card
TAPE TRANSPORT BRAKING SYSTEM
Pentagon Industries. Inc. has developed a new concept for reel to reel
tape transport braking system that is
particularly applicable to high speed
duplication of audio tapes. The compound electro- mechanical brake system.
currently employed in the 1100 Series
Duplicating System, provides smooth
handling for all standard tapes and has
enabled open reel duplication at speeds
of 120 riches per second. It does not
require adjustments due to wear or
temperature and humidity changes. Four
inch reel masters in connection with
"Nagra SN" recordings of .150 mil wide
times one quarter mil thick master tapes
are duplicated without any danger of
stretching, as in the case of conventional
brakes.
Mfr: Pentagon Industries. Inc.
Circle 48 on Reader Service Card
ANALOG BARGRAPH
I. ON LINE
Z. ON BUDGET
3. ON TIME
The APM Series 20 is a new solid -state
analog bargraph indicator featuring a
3 -in., 20 element bargraph. The APM 20
meters provide a low cost alternative to
mechanical meters and are available in a
number of standard voltmeter and
ammeter ranges. Options include offset
span, differential input, reduced response
time and either single or dual setpoint
controls. Meters meet ANSI 39.1 shock,
temperature and humidity requirements.
Display brightness is controllable with
an external potentiometer.
Mir: Bowmar /ALL Inc.
Price: $60.00 and down
Circle 47 on Reader Service Card
Three good reasons
to skip the big guys
and buy from
Pro Audio Systems.
for orders call:
I-800-426-6600
Alaska & Hawaii)
(including
Featuring MCI, Sound Workshop, Tascam & other fine audio equipment.
11057 8th Avenue N.E.
Seattle, Washington 98125
Circle 32 on Reader Service Card
(206) 367 -6800
RMS ANALOG VOLTMETER
PURE PRO
audio design
introduces the remarkable
3535 LOG -8 Microphone Mixer
..---..
with
Voice-Entree-option
Permits the activation and simultaneous use of multiple microphones
without feedback and does the job
automatically -no operator, no
manual adjustments required ...
ASK FOR DESCRIPTIVE FOLDER
The IET Model MV -800 Analog
Voltmeter is a true RMS instrument and
will therefore accurately present the
effective power of waveforms which
depart from a true sine wave. Fifteen
ranges are provided from 30µV full
scale to 300V, and from -90 dB full scale
to +50 dB in decades. A front panel
bandwidth switch permits a choice of
wide band measurements, standard first
audio bandpass (20-20 kHz) or external
filter. A group of standard or custom
plug-in filter modules are available
from IET. Internal rechargeable nicad
batteries allow the Model M V-800 to be
totally isolated from the power line and
external grounds. The Model MV -800
can be used as an oscilloscope preamplifier through the use of its front panel
BNC output jack. A chart recorder.
DV M. X -Y plotter may be connected
to the rear BNC connector for permanent
records of readings or under -over limits
observations. This terminal provides a
linear DC output of 0-1 volt proportional
to meter deflection.
Mrr: IET
Lah.c. Inc.
Price: $495.00
Circle 49 on Reader Service Card
LOOK TO RAULAND FOR EVERYTHING
IN PROFESSIONAL SOUND
SYNTHESIZER
The Realistic MG -I Synthesizer h
is a musical synthesizer with professional features designed for club or
concert performance, composing and
rehearsing. The MG -1 features a 2Y2octave full- chromatic keyboad and a
versatile control panel. divided into
related sections and color-coded for easy
use. Two independent three-octave tone
sources feature variable waveshapes;
Tone Source One offers a 2:1 sync
selector, 3- position octave selector and
2- position (square/ triangular) wave shape selector. Tone Source Two offers a
detune control for dissonant or full interval offsetting of its pitch, 3- position
octave selector and 2- position waveshape
selector. The modulation section can
create vibrato and tremolo effects,
pulsating notes, random tone sequences
and glide effects. It controls the tone
sources. filter, keyboard glide and
modulation rate, and includes a 3 -position waveshape selector and 2- position
auto-contour trigger. The 4-control filter
adjusts the tonal characteristics (timbre)
of the synthesizers output by controlling
cutoff frequency, peak emphasis and
contoured cutoff. A master volume
control adjusts the output of the synthesizer. which is available both at a headphone jack and at phonotype output
jacks that permit direct connection for
most stereo and sound systems.
tlfr: Tandy Corporation' Radio Shack
c ircle 50 on Reader Service Card
Moog
Spectrum- Master Equalization (Pat-
ented): Most complete line available.
Includes 1/3- and 1- octave models,
test set, and tunable notch filter.
Spectrum- Master In -Wall Amplifiers:
Available in 35 -watt, 60 -watt, 100 -watt
outputs. Each with built -in 1-octave
equalizer and D.R.E.
Spectrum- Master Amplifiers: Incomparable Model DX (5 -year warranty) and
TAX amplifiers, from 70 to 250 watts
RMS to meet any professional audio
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Spectrum Series Speakers: Professional
quality two -way speaker systems embodying superior components in advanced system designs.
WRITE FOR TECHNICAL BULLETINS
The Quality Name in
Professional Audio
RAULAND -BORG CORPORATION
3535
W.
Addison St.. Dept. N., Chicago. Ill. 60618
CD
Circle 20 on Reader Service Card
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PRO MIXING CONSOLE
Special binders
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SILENT PLUGS AND PHONE PLUGS
Switchcraft Silent Plugs feature a
circuit closing device that stops hum,
squeals and pops when plug is removed
from the jack. Also featured are new
solder lug terminals and mylar tube insert
for greater electrical protection, and onepiece tip rod assembly to insure plug
quality. Silent Plugs utilize cables up to
.25 -in. maximum diameter and can
accommodate parallel or shielded cable.
The phone plugs also employ new solder
lug terminals for easy connection, with
one piece tip rod standard on all plugs.
Four different models are available.
Mfr: Switchcraft. Inc.
Circle 52 on Reader Service Card
Just $7.95 each, available in
North America only. (Payable
in U.S. currency drawn on U.S.
banks.)
r
The Series 1000 professional mixing
console is made for small sound reinforcement jobs, submixers, keyboard
mixers, as well as for theatrical, church
and broadcasting applications. They are
available in 8, 12 and 16 channel versions
(the 8 channel is available in a I9 -in.
rack -mounted version). The Series 1000
has balanced transformerless inputs and
outputs, and accepts high and low
impedance inputs. Input attenuation
switches allow precise and repeatable
selection of input gain from +50 dB to
- 10 dB in 10 dB steps. Each channel has a
LED preamp level monitor which indicates critical headroom level, and has a
responsive 3 -band EQ. Other standard
features include: output patch points for
inserting EQ's, limiters, 10- segment LED
bar graph displays, headphone monitoring, and work lamp socket. A sum switch
converts the stereo sends into subgroups. The built -in reverb system is
patchable for external effects, using the
"Audy effects loop" concept.
Mfr: Audi' Instruments, Inc.
Circle 5I on Reader Service Card
J
nounced by Electronic Specialists are
the newest additions to their patented
isolator filter/ suppressor line of interference control products. Available on
all isolator models, the remote AC power
control switch can be mounted with the
audio equipment for convenience. "Total
Remote Switched Isolator capacity is
1875 watts max, with each socket capable
of handling a I KW load.
Mfr: Electronic Specialists. Inc.
Price: $79.95 and up
Circle 53 on Reader Service Card
www.americanradiohistory.com
FLOOR MONITOR
The TA -I2 is a compact floor monitor
suited for applications ranging from rock
'n roll to speech or acoustical instrument
reinforcement. The system utilizes an
E -I2 Bag End'" I2 -in. loudspeaker and
an ST350B Electro -Voice tweeter in a
specially designed crossover network
with solid state tweeter protection. The
system employs the Time Alignment
Technique" of E. M. Long Associates.
making it the first Time Align performance monitor of its kind.
Mfr: Modular Sound Systems Inc.
Circle 54 on Reader Service Card
Your
performance
insurance
policy.
ANALOG -TO- DIGITAL CONVERTER
The PCM75 A, D converter is designed for PCM Audio applications and
is compatible with EIAJ STC -007 specifications. The internal I6 -bit digital -toanalog converter is available for the
designer to utilize in the playback mode.
The conversion time can be reduced from
15 µsec to 8 µsec with some increase
in distortion. Distortion is specified
on the data sheet to assure performance
in critical audio applications. The
PCM75 is complete with internal reference and clock.
Mfr: Burr-Brown Research Corporation
Circle 55 on Reader Service Card
PATCH CORD
The Model 4812 Minigrabber patch
cord utilizes 22 AWG, two conductor
shielded microphone cable, and standard
PJ-05 I R phone plug. The gold plated.
beryllium copper Minigrabber provides
maximum sensitivity.
Mir: ITT Pomona Electronics
Price: $19.50
Circle 56 on Reader Service Card
Tim should not have to worry about your e¢tipment while performing. You need to know trat
the beat will go on. lour reputation depends
MI it.
SIGNAL MULTIPLEXER
The 1360 Programmable Signal Multiplexer is a microprocessor-based. IEEE488 compatible. system instrument that
can be used to multiplex electrical
signals. 7 he 1360 includes two separate
chassis: the 1360P Programmable Switch
Controller and the 13605 Switch Matrix.
Up to four 13605 Switch units can be
operated by the same 1360P Controller.
For each Switch Matrix, the user can
choose to multiplex one output with
thirty -two inputs, two ganged outputs
with sixteen inputs or four ganged outputs with eight inputs.
Mfr: /ekrrontx, Inc.
Price: 1360P-$2.500.00.
13615- $1.500.00
Circle 57 on Reader Service Card
Gauss knows that. We build cost effective ht adspeakers that are more reliable.
Our test report is proof that every louispeater
has been through the "Gauss School of rays
Knocks" before it reaches the stage.
put them
through a tougher test than you ever will in every
mode of operation so that you can rest assured
It
that the show will not stop.
Our testing is your best and most cos: effective
performance insurance policy.
'giber naprfatioet worth it.
t
(L
!C Gauss
c
9130 Gleoalm Blvd.
Sun Valle.. CA 1352
It' /6 on Reader Service Card
ROTARY SWITCH
AUDIO SPECTRUM ANALYZER
STEREO
HEADPHONE BOX
The Digital Sona -Graph (Model 7800)
audio spectrograph which combines
the features of a conventional spectrum
analyzer. a 3 -D display. a high resolution
grey scale printer and an oscillograph.
The signal is recorded in a 64K word by
IO bit memory. As much as 2.56 seconds
of signals at 8 kHz (up to 41 seconds at
500 Hz) can be analyzed using several
display modes and filter bandwidths.
Frequency changes over time can be
displayed and analyzed. As little as 6ms
of any portion of the time domain signal
can be displayed and printed.
Mfr.- Kay Elenterrics Corp.
Price: 313.500.00
Circle 59 on Reader Service Card
is an
The Model RT -A is a printed circuit
mounting rotary switch for use in test
equipment, oscilloscopes and medical
equipment. The Model RT -A. with binary or decimal coding, features moveable contacts that are obtained by coded
discs and housed in a flat case. The contacts. BBM or M BB. are silver plated and
have 5 VA switching power. RT -A contacts are designed to be protected against
flux contamination at the moment of
soldering (in the bath or flow) and
printed circuit board cleaning. Available
options include a threaded bushing
mount, adjustable stop. screwdriver slot.
flatted shaft, solder lugs or P.C. pins.
HOLDS UP TO
SIX HEADPHONES
s
to
STEREO COMBINER
SEND FOR YOUR FREE COPY OF OUR CATALOG
5E5
CUM
SESCOM. INC.
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Mfr: ITT Schadokc, Inc.
002A9a-0990
1900) 634-3457
TWV
Price: I K Pieces, $1.39
Circle 58 on Reader Service Card
910-397-69%
99101 U.S.A.
on Reader Service Card
REVERBERATION SYSTEM
Nwraxgrato
SONY
IMPROVES THE
MID-SIDE MIC
BY ONE G.
In the world of dB's and kHz's, Sony is
an unquestioned leader.As is the case when it
comes to G's. A fact most recently proven
when Sony priced its new ECM -989 mid side mic a whopping one grand below the
nearest competitor without sacrificing
quality, In fact, the ECM-989 even allows
the capsule assembly to be remoted from its
power supply.
So, tf you want an M -S stereo mic that's
equally at home in the studio as the concert hall,
ask your Sony dealer about the new Sony ECM -9897
It'll sound just as good to you as it will
to your accountant.
t>fOr ., 0
SONY
1.719111
Sm, (
:11
n1,d10
potation of Amerna.9 Wnt )7th Street. Ne. York. NY 10014
Sm 1, a reFlxered trademark of Ihr Sm, ('orp.na0on
'Wean n0hpt.aul,eNr
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Circle 4/ on Reader Service Card
www.americanradiohistory.com
The XL -121 Reverberation System is
designed to interface with virtually any
other audio equipment. The Preamp
Gain control allows the unit to accept a
low -level musical instrument output
such as an electric guitar. as well as
higher level signals associated with sound
reinforcement consoles. In addition. it
can serve as a preamplifier to interface
directly with any power amplifier. This
feature enables the reverb to be used in
the signal chain at all times. and a
standard footswitch can be used to
switch the reverb effect in and out without affecting the direct signal. The
Output Level control also permits interfacing with all other signal processing
equipment. the front panel Output Mix
control allows blending of any desired
amount of reverberated and direct signal.
The Equalization section is intended to
allow the user to tailor the reverberant
sound characteristics. Included in the
section is a Low. Mid and High control.
all with 12 dB of boost or cut. The EQ
section only affects the sound of the
reverb. and the controls are non- interacting. A dual -colored LED serves as a
Power Overload indicator.
Mfr: M /('.1II.1. Audio Products. Inc.
Price: $450.00
Circle 60 on Reader Service Card
SOUND SYSTEM
The Entertainer is a complete, portable, powered sound system consisting
of a 10- channel I50 watt per channel
powered mixing console and two constant directivity speaker systems. The
IOOM powered mixing section of the
system has a ten -input stereo out configuration. Two channels have line -level
inputs that can be used with tape decks.
allowing pre -recorded portions of a
performance to be Introduced. The IOOM
also features fluroscan bar graph displays
for across -the -stage meter reading. The
IOOS speaker system is housed in a
ítCo
SOUND,
INC.
polyethylene cabinet. The system's
twelve -inch woofer is complemented by a
constant directivity tweeter. The IOOS
systems are rated to handle 100 watts
long -term and up to 400 watts for short term peaks.
Mir: E-VITA PC()
Price: $2.195.00
Circle 6/ on Reader Service Card
AAE "Concept One"
Automation
AKG
AUDIOARTS ENGINEERING
BGW TANNOY
REAL TIME ANALYZER
The RTA 150 real time analyzer is a
combination of four instruments in one:
real time analyzer, sound level meter.
wave analyzer and pink noise generator.
The RTA 150 has a matrix of 209 LEDs
with 36 dB of displayed dynamic range.
An over or under range arrow indicates
the switch to press to get the spectrum
back on display. In addition to 15 frequency bands, there is an extra yellow
band that continuously displays the
broad band sound pressure level (SPL)
of all frequencies. The yellow horizontals
LEDs become the flat reference line. The
digital display that corresponds to the
LEDs can be adjusted from 50 dB to
125 dB (in 5 dB increments) by pressing
the reference buttons up or down as
needed. The RTA I50 uses a scan
function that allows you to place a
bright display cursor at any desired
frequency band. The display of the
scanned frequency will appear much
brighter and the decay rate will increase
to follow signal transients more accurately. The scanned frequency will also
appear on the analog display meter to
allow that band to be read in I dB increments. A 31 -bit pseudorandom digital
number generator provides a stable,
accurate noise source. The pink noise
buttons, located on the front panel, are
adjustable from 80 to 110 dB in IO dB
increments. When the pink noise generator is switched off, the gain of the pink
HANNAY
(Cable Reels)
EVENTIDE
(VIE
LEXICON
SENNHEISER
TECHNICS
VEGA
WIREWORKS
If It Sounds Good
Its From FITZCO
204 N. Midkiff
Midland, Texas 79701
(915) E84-0861
noise automatically returns to the
80 dB gain level.
Mir: Pulsar Laboratories. Inc.
rn
Circle 63 on Reader Service Card
Circle 43 nn Reader Senvice Card
Classified
Closing date is the fifteenth of the second month preceding the date of issue.
Send copies to: Classified Ad Dept.
db THE SOUND ENGINEERING MAGAZINE
1120 Old Country Road, Plainview, New York 11803
Minimum order accepted: $25.00
Rates: $1.00 a word
Boxed Ads: $40.00 per column inch
db Box Number: $8.50 for wording "Dept.
Plus $1.50 to cover postage
Frequency Discounts:
6
times, I.5%;
ALL CLASSIFIED ADS MUST
12
XX,'
etc.
times, 30%
BE PREPAID.
FOR SALE
REELS AND BOXES 5" and 7" large and
small hubs, heavy duty white boxes.
W -M Sales, 1118 Dula Circle, Duncanville,
Texas 75116 (214) 296-2773.
AKG BX20 reverberation and C414 microphones. FOR IMMEDIATE DELIVERY.
UAR Professional Systems, 8535 Fairhaven, San Antonio, TX 78229. 512690 -8888.
TASCAM, BGW, JBL, EV, Nikko, Technics, Onkyo, Haller, Audio Research,
Conrad Johnson, dbx, etc. P. K. Audio,
4773 Convention, Baton Rouge, LA 70806.
(504) 924 -1001.
RECORDING SECRETS MOST ENGINEERS WON'T TELL, $7.95 Tune tronics, P.O. Box 55, Edgewater, N.J.
07020.
101
LEXICON, dbx, & UREI Most items for immediate delivery. UAR Professional Systems, 8535 Fairhaven, San Antonio, TX
78229. (512) 690 -8888.
FREE
CATALOG d AUDIO APPLICATIONS I
CONSOLES
KITS A WIRED
AMPLIFIERS
MIC., EO,ACN,LINE,
TAPE, DISC, POWER
OSCILLATORS
AUDIO. TAPE RIAS
POWER SUPPLIES
CV
co
OPA N1 Y
I..A BS
I
NC.
AMPEX, OTARI, SCULLY -In stock, all
major professional lines; top dollar trade ins; write or call for prices. Professional
Audio Video Corporation, 384 Grand
Street, Paterson. New Jersey 07505. (201)
523-3333.
EMT 156 PDM COMPRESSOR for sale.
Used only a few hours. Absolutely mint,
was $6,800.00 new. Will sell for $4,500.00,
barely used. Satisfaction guaranteed.
Call (813) 447-1487 at 12 PM or 6 PM or
after 11 PM at nite.
DISC CUTTING SYSTEM, Westrex 2B
head rest of system stereo, Scully Lathe 2
track Ampex, Limiter, Eqs., Altec /McIntosh Monitors, priced to sell. Sound wave, 50 W. 57 St., N.Y.C. 10019 (212)
582 -6320.
LEXICON 224 Digital Reverberation. FOR
IMMEDIATE DELIVERY UAR Professional Systems, 8535 Fairhaven. San Antonio. TX 78229, 512- 690 -8888.
THE LIBRARY...Sound effects recorded
in STEREO using Dolby throughout. Over
350 effects on ten discs. $100.00. Write
The Library, P.O. Box 18145, Denver,
Colo. 80218.
TECHNICS, BGW, EVENTIDE, AKG,
Scully, and many more IN STOCK. FOR
IMMEDIATE DELIVERY: UAR Professional Systems, 8535 Fairhaven, San
Antonio, TX 78229. (512) 690 -8888.
TASCAM MODEL
5,
like new, full war-
ranty, $1,100.00. Two Electro -Voice
RE -15's, $150.00 each, RE -20's, $300.00
each. N.A.B., Box 7, Ottawa, IL 61350.
PROFESSIONAL AUDIO EQUIPMENT
Shop for pro audio from N.Y.'s leader,
no matter where you live! Use the Harvey Pro Hot -Line. (800) 223 -2642
(except NY, AK, & HI). Expert advice,
broadest selection such as: Otani,
EXR, Ampex, Tascam and more. Write
or call for price or product info:
Harvey Professional Products Division
2 W. 45th Street
New York, NY 10036
(212) 921 -5920
MICROPHONES BY UPS. Quicker. You'll
save more with us. All popular models for
immediate delivery. UAR Professional
Systems. (512) 690 -8888.
FREE
CATALOG d AUDIO APPLICATIONS
CONSOLES
KITS & WIRED
AMPLIFIERS
MIC., FO, ACN,LINE,
TAPE, DISC, POWER
OSCILLATORS
AUDIO, TAPE BIAS
POWER SUPPLIES
P
OPAM
LABS INC.
1033 N. SYCAMORE AVE.
LOS ANGELES, CA. 9003E
(213) 934 -3566
www.americanradiohistory.com
1033 N. SYCAMORE AVE.
LOS ANGELES, CA. 90031
(213) 934-3566
=..,_ ,.,
PRO AUDIO GEAR
-
New & Used
In Stock
Our Price Can't Be Beat
AUDIO TECHNICA
HAVE YOU EVER BEEN FRUSTRATED
WITH FEEDBACK HOWL EVEN AFTER
USING EQUALIZERS? FIND OUT
ABOUT OUR FEEDBACK STABILIZERS
MOD. NO. 741 XR AND 742 XR.
JBL
CROWN
TAPCO
ORBAN
ELECTRO -VOICE
UREI
AND MUCH MUCH MORE
FANFARE AUDIO
BODE SOUND CO.
1344 ABINGTON PL.
NO. TONAWANDA, N. Y. 14120
716692 -1670
University Microfilm, Inc.
300 North Zeeb Road
Ann Arbor, Michigan 48106
STEVENSON INTERFACE MIXER, 16
inputs, expandable to 24. 1 104N mix down, 104M master, like new $3,500.00.
Call Linda Cooper (212) 738-9379 or
(212) 641 -5432 (ans. machine).
EMPLOYMENT
2101 W. CENTRAL BLVD.
ORLANDO. FLORIDA
305/843 -9383
ORBAN. All products in stock. FOR IMMEDIATE DELIVERY. UAR Professional
Systems, 8535 Fairhaven, San Antonio,
TX 78229. 512- 690 -8888.
AMPEX, OTARI & SCULLY recorders in
stock for immediate delivery: new and
rebuilt, RCI, 8550 2nd Ave., Silver Springs,
MD 20910. Write for complete product list.
BLANK AUDIO AND VIDEO CASSETTES
direct from manufacturer, below wholesale, any length cassettes: 4 different
qualities to choose from. Bulk and reel
mastertape -from -inch to 2 -inch. Cassette duplication also available. Brochure.
Andol Audio Products, Inc., Dept. db,
42 -12 14th Ave., Brooklyn, NY 11219. Toll
free 1 -800- 221 -6578, ext. 1, NY residents
/
(212) 435 -7322.
Copies of db
Copies of all issues of db -The
Sound Engineering Magazine starting with the November 1967 issue
are now available on 35 mm. microfilm. For further information or to
place your order please write directly to:
1
NEVE COMPLIMITER MODEL Stereo
2254/E with pwr supply. In mint condition.
Barely used. Satisfaction guaranteed.
Call (813) 447-1487 at 12 PM or 6 PM or
after 11 PM at nite. Ask for Jon.
ENGINEER /ENGINEER ASSISTANT
seeking full time permanent work. 4 years
audio, 2 years video, and maintenance
experience. Equipment to offer. Call
(205) 263 -6353.
SERVICES
LIKE NEW CONDITION 351 -2 Ampex-
lnovonics recorder, Soundcraftsmen
Graphic EC) SG -2205 Master Room & Ill
Echo Unit. Virtue Recording (215) 763I
2825.
I
ACOUSTIC CONSULTATION- Specializing in studios, control rooms, discos.
Qualified personnel, reasonable rates.
Acoustilog, Bruel & Kjaer, HP, Tektronix,
!vie, equipment calibrated on premises.
Reverberation timer and RTA rentals.
Acoustilog, 19 Mercer Street, New York,
NY 10013 (212) 925 -1365.
Microphones
j
Speakers
MAGNETIC HEAD relapping -24 hour
service. Replacement heads for professional recorders IEM, 350 N. Eric Drve,
Palatine, IL 60067. (312) 358 -4622.
VIF INTERNATIONAL will remanufacture
Shure Brothers,a leader in audio communications, is seeking individuals to work as technical coordinators. Responsibilities include
providing technical assistance to dealers and key users, assist in
new product presentations, and provide support to the advertising
and sales departments.
At least two years sales or marketing experience related to sound
reinforcement equipment, microphones or musical instruments
is necessary.
Excellent starting salary and company benefits Reply in complete
confidence including salary requirements to.
your Ampex or Scully (Ashland/Bodine)
direct drive capstan motor for $200.
Average turn around time -2 -3 weeks.
For details write PO Box 1555, Mtn. View,
CA 94042, or phone (408) 739 -9740.
CUTTERHEAD REPAIR SERVICE for all
models Westrex, HAECO, Grampian.
Modifications done on Westrex. Quick
turnaround. New and used cutterheads
for sale. Send for free brochure: International Cutterhead Repair, 194 Kings Ct.,
Teaneck, N.J. 07666. (201) 837 -1289.
G.L. Breyfogle
SHURE BROTHERS, INCORPORATED
222 Hartrey Avenue
Evanston, Illinois 60204
An Equal Opportunity Employer
SHURE
JBL AND GAUSS
SPEAKER WARRANTY CENTER
Fast emergency service. Speaker reconing and repair.. Compression driver
diaphragms for immediate shipment.
NEWCOME SOUND, 4684 Indianola
Avenue, Columbus, OH 43214. (614)
268 -5605.
m
ca
People/Places/Happenings
Rupert Neve Incorporated has announced the appointment of Mr. Anthony
H. Langley as vice president sales.
Langley was first employed by Neve's
parent company in England starting in
1972 in product design and production
engineering. He was appointed to
manager of Test and Quality Control at
the Kelso, Scotland factory in 1974,
which position he held until early 1976
when he was transferred to Rupert
Neve Incorporated as marketing support
engineer. Since 1977 Langley has held
the position of marketing manager for
Neves U.S. operation.
Joe Bean has joined Studer Revox
as a sales representative,
according to the firm's president. Bruno
Hochstrasser. Bean will concentrate on
developing the broadcast market for the
Studer professional lines in the Southeast.
Bean comes to Studer after serving for
31 years as a sales representative for
Audio Consultants, a Nashville studio
supply and design company.
America
The Denver-based acoustical consulting firm of David L. Adams Associates has announced a relocation of
its offices to larger quarters at 1701
Boulder Street, Denver. Colorado
Digital Recording Corporation
(NASDAQ -DRSO) announced the opening of its new Nashville facility for its
Soundstream, Inc. subsidiary. The new
facility will be equipped with Sound -
stream's advanced digital recording
equipment for serving the Midwest and
Southeast recording markets. A demonstration of Soundstream equipment was
held on July 20 in Nashville to which
the trade press, recording industry, and
the public was invited.
Bob Silk and Paul Blank, owners of
The Mike Shop, are pleased to announce
that The Mike Shop has moved to new,
larger quarters in Lynbrook, New York
in order to provide improved sales and
service to its customers. Comprising
4000 square feet. the newly renovated
building will house The Mike Shop's
offices, service department, shipping
department, and warehouse space, all on
one floor. Originally offering only
microphones and accessories, The Mike
Shop now specializes in providing a full
line of professional audio equipment and
services to recording studios, broadcast
stations, churches and independent
sound engineers. The Mike Shop's mailing address and telephone number have
not been changed.
80211. A new brochure describing the
firm's experience and qualifications
also available upon request.
is
Bose Corporation has announced
that John Strand has been appointed to
the post of applications engineer, Professional Products. Strand, a native of
Milwaukee, is a graduate of the University of Wisconsin in Electrical Engineering and Acoustics. He will be based at
Bose corporate headquarters in Framingham, Massachusetts. Strand will head
Bose 's liaison program with acoustical
consultants and sound system designers
and will provide applications assistance
to sound contractors and end users of
Bose Professional Products. The appointment of Strand, according to Bose Pro
Marketing Manager Roy Komack,
co
marks a new phase in Bose Corporation's
expansion in the engineered sound
business. This program will take its
next step with a major new product
introduction this coming winter.
The Motown Sound
is
now being
exclusively recorded and mixed on
U.R.E.1's new Series "A" Time -Aligned
studio monitoring professional loudspeakers. Installed as a standard reference system throughout the Motown/
Hitsville multi- studio complex (5). the
first pair of 811's retrofitted into the
control room soffit without extensive
modifications.
Paul Murphy, general manager of
Beyer Dynamic, Inc., announced the
appointment of Tom Bensen as national
sales manager for Consumer and Professional Products. Bensen has a varied
background in the electronics industry
Mr. Hosoda, president of Otari
Electric Co., Ltd., has announced the official opening of Otari Electric Deutschland GMBH, (West Germany). The sales
and service organization, located near
Dusseldorf, is headed up by Mr. Ken
Hirano, operations manager. Opened
in June. the new branch of the Japanese
professional audio tape machine manufacturer will handle European business
and service for all Otari products.
Tentel, manufacturer of the Tentelometer tape tension gage and similar
service tooling, has announced a move
into newer and larger quarters. Their
new address is 1506 Dell Avenue,
Campbell, California 95008. The new
phone number is (408) 379 -1881 or toll free (800) 538 -6894, continental U.S.
except California.
Bonneville Productions has installed a
Q-Lock 310 SMPTE time code synchronizer system. The system is portable
and can be used in any of Bonneville's
three specialized studios to run two or
three tape sources simultaneously in
exact synchronization. It is interfaced to
Ampex brand MM1200 twenty -four
track. 440C eight track, ATR 104 four
track audio machines and to a Sony
2860 U -matic video cassette recorder.
Hal and Vio Michael have recently
announced the completion of Spindletop
Recording Studios in Hollywood, CA.
Spindletop features an MCI 636 console,
complete with J H -50 automation. J H -24
24 -track recorders. JH -110B 1/4-in. 2track recorder, JH -110B 1 -in. 4-track
recorder, JH -110B grin. 2 -track mastering recorder, and a J H-45 SMPTE-EBU
Based Generator¡ Synchronizer, providing the capability of 48 -track audio as
well as video interlock.
21/2 years as Audio
Products division manager for Eumig.
He was technical director for TDK
before joining Eumig and was an-'
nouncer; engineer for Beck -Ross Communications Inc. (WGLI Radio) for
The appointment of Dom Notto as
general manger/ vice president of Nagra
Magnetic Recorders, Inc., was announced recently by Kudelski, S.A. of
Switzerland, parent group for NMRI.
Mr. Notto who, for 10 years, has been
vice president, sales for the company.
succeeds Jean -Jacques Broccard, who
remains general manager of NMRI of
2 years.
California.
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1
MASTERS OF THE STUDIO.
There's as much magic in the mixing board as there is
in the keyboard.
That's why, when we awarc the Ampex Golden
Reel, it goes to both the recording artist and the
recording studio. Together they provide the magic that
turns a reel of recording tape into an outstanding
creative achievement.
The Ampex Golden Reel Award honors those
achievements that were mastered on Ampex
professional recording tape. They've earned a place in
the ranks of the world's most successful recorded
albums and singles'.
Along with the Award, we also present $1,000 to a
charitable organization. Since we started the Golden
Reel Awards three years ago, there have been over 200
recipients, and more than $200,000 donated on their behalf.
Congratulations to all of them. The masters on
both sides of the microphone.
AMPEX
Ampex Corporation, Magnetic Tape Division,
cAmpen 1981
Golden Reel Wnnurs a at
=_
I
E.1
401
Broadway, Redwood City, California 94063, (415) 367 -3889.
Circle l0 on Reader Service Card
'RIAA Certi ieo Golo
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