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10 oyr Reader Se/ r tie t aî i
See your professional audio products
dealer for full technical information.
We start the new year with a little more
news about digital audio. Depending on
who you read, it may be the greatest
worst -thing to happen to recording
since the invention of the microphone.
Next month, we'll have a look at the
latest in the digital domain, courtesy of
some authors with contrasting views on
the subject.
The Cumulative Index, normally seen
in December issues, will appear instead
in the January 1981 issue.
John Borwick
Bruce Bartlett
John Eargle
Ralph Hodges
Greg Silsby
Barry Blesser
Norman H. Crowhurst
N. I. Weinstock
i. liard in Current Contents: Engineering and Technology
Larry Zide
John M. Woram
On this month's cover, we see some
creative microphone placement, courtesy
of photographer Robert Wolsch.
In the foreground, an Electro -Voice
RE -20, Shure SM -59 and SM -78, and a
Beyer M -I60. The middle row shows a
Ramsa WM -8100, an Audio- Technica
ATM-1 ISM and a pair of Shure SM -81s.
In the background, a Sony C -48, a Beyer
M -500 and an Audio -Technica AT -815.
Mark B. Waldstein
Crescent Art Service
Kathy Lee
Eloise Beach
Lydia Anderson
Bob Laurie
db. the Sound Engineering Magazine (ISSN 0011.71451 is published monthly by Sagamore Publishing Company. Inc. Entire contents
1980 by Sagamore Publishing Co.. 112001d Country Road. Plainview. L.1.. N.Y. 11803. Telephone 15 161 413 6530. db is
published for those individuals and firms in professional audio-recording, broadcast, audio-visual. sound reinforcement. consultants.
video recording. film sound. etc. Application should be made on the subscription form in the rear of each issue. Subscriptions are S 11.181
per year (52100 per year outside U.S. Possessions and Mexico: S13.00 per year Canada) in U.S. funds. Single copies are SI 95 each.
Editorial. Publishing and Sales Offices: 1120 Old Country Road. Plainview. New York 11803. Control Circulation Postage Paid at Old
Saybrook. CI.
Index of
Mike Shop
R. K.
Cover IV
Morrison Illust. Mats
Neal Ferrograph
attended the Natural Stereo Recording Techniques workshop at Eau Claire.
Wisconsin and found that people were
using the M -S recording technique with
the M opened up to omni or even figure -8
modes. What are the theoretical mic
patterns which result from such matrix
applications? I find to my chagrin that
I can't figure it out.
Thomas Ammons
Glenshaw. Pa.
Series Ill and HIS precision pick -up
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Series Ill arms are true low mass
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Write to Dept 1861 SM E Limited,
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Exclusive distributors for the U.S:
Shure Brothers Incorporated,
222 Hartrey Avenue, Evanston,
Illinois 60204
andin Canada:
A. C. Simmonds and Sons Ltd,
975 Dillingham Road, Pickering,
Ontario, L1 W 3B2
In Norman Crowhurst's reply (db
June 1980) to my letters to the editor of
November 1979 and April 1980, he states
that I credit myself with the original
idea for nonlinear distortion by complementary distortion. I wish to point out
that I have never claimed such originality
and. in my papers on the subject. I refer
to earlier work by others in the field. I
was, however, probably the first to
publish explicit mathematical treatments
of the possibilities and limitations of
the method.
Although Crowhurst now agrees that
his original statement (which occasioned
this exchange of letters). "In fact. however distortion gets in. you cannot take
it out again," may be modified that one
can in fact reduce (nonlinear) distortion
Circle 24 on Reader Servire Card
Pro Audio Seattle
Shure Brothers
Standard Tape Lab
Studer Revox
Telex Turner
VIZ Manufacturing
db replies:
lrr M -S recording, the M mir is usually
eardioid and the S is a .figure -8. For
details about what happens when M and
.S are cvunbined, see our .4pplicalion
.Putes in this issue.
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Manufacturers of high fidelity components, microphones, sound systems and related circuitry.
Circle 29 on Reader Service Card
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provided one has sufficient information
about the distorting mechanism. he still
stated in his June 1980 letter that distortion could not be eliminated by complementary distortion techniques. I have
now learned. through direct correspondence with Mr. Crowhurst. that he
has not had a chance, since the beginning
of this exchange of letters to the editor
in November 1979. to look at my papers
on complementary distortion, referenced
in the November 1979 letter. The papers
do. in fact, show how it is theoretically
possible to entirely eliminate nonlinear
distortion within at least a limited input
signal amplitude range. In actual practice, of course, one can only reduce such
distortion by an arbitrary amount, and
the greater the reduction wanted the
greater the circuit complexity required.
You can, therefore, take the distortion
out to whatever degree needed.
The Mike
PO Box 366A, Elmont, NY 11003 (516)
A Division of
of North Carolina
437 -7925
Omnisound Ltd.
Circle 23 on Reader Service Card
publication is
available in
.+ ai
Please send me additional information
University Microfilms
300 North Zeeb Road
Dept. P.R.
Ann Arbor, MI 48106
18 Bedford Row
Dept. P.R.
London, WC1 R 4EJ
(415) 786 -3546
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Street _
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Digital Audio
The Realities of
Digital Technology
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We are now ready to look at a digital
audio system in terms of the actual
performance which we can expect, using
real components. Up to this point, our
discussions have mainly been about an
ideal system with perfect components.
But alas, our perfect digital system
actually produces not -very -perfect results, due to defects in the components.
To be charitable, we should call them
"limitations," since nothing is ever
perfect. But because we are examining
digitization. rather than analog technology, our inclination is to be less
forgiving about defects in the digital
domain. Hence, our criterion for "acceptable" is often much higher.
We will begin our discussion about
digital audio defects with the DAC (digital-to -audio converter), since this element is used in both encoding and decoding. But, before we begin, we need to resolve a semantic difficulty with the word
DAC. The problem is that there is a circuit component which converts a digital
word to an analog voltage, and this component is used in both the encoding(analog-to- digital) and decoding (digital -toanalog) parts of the digital audio system.
From this point forward, we will refer to
the DAC as the component; and A, D
and D ,.A as the encoding and decoding
system. Both contain the DAC component.
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In previous discussions. we demonstrated the operation of the DAC component by showing the relationship between the digital number (word) inputs
and the analog (voltage) outputs. Each
word corresponds to a specific voltage.
and a uniform DAC has a consistent
voltage difference between neighboring
digital words. With a I0 -bit DAC component having a ±I0 volt output range,
there will be 1024 quantization levels,
each separated by 20 millivolts. We will
thus expect a quantization level at -30
mV, -10 mV + IOmV, +30 mV. etc. However, in a real DAC component, the
quantization levels may not be quite
where we expect them to be; there will be
an error. For example. the quantization
level which was expected to be at + IO mV
might actually be at + 14 mV. This is a
positive error of 4 mV. The quantization
level which should have been at +30 mV
might actually be at + 18 mV, a negative
error of -12 mV.
One way of specifying the quality of a
converter is to specify the largest error
between the DAC component and an
idealized perfect DAC having the same
number of bits. Since millivolts of error
is an inconvenient metric, we typically
measure the error as a ratio of the maximum error voltage to the ideal difference
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in quantization levels. In our example,
neighboring levels are intended to be 20
mV apart. This is defined as an LSB
(least significant bit), since a change in
the lowest -level bit will change the output by this amount. If the worst-case
quantization level were in error by 4 mV,
we would say that the maximum error was
1/ 5 LSB. If the worst -case error were 30
mV, we would say that the converter was
accurate to 1.5 LSBs. In this discussion,
accuracy is the same as linearity, since an
inaccuracy produces a non -linear relationship between digital input and analog output. The absolute linearity error
is the name for this kind of inaccuracy.
Until recently, most manufacturers of
DAC components specified a worst -case
absolute non -linearity oí±0.5 LSBs. This
criterion was selected because it also implied that the DAC was monotonic.
Monotonic means that a larger digital
input always produces a larger analog
output. Let us examine this case in detail. With a ± 0.5 LSB error, a quantization level at + IO mV could be any place
between 0 mV and + 20 mV, and a quantization level at 30 mV could be any place
between 20 mV and 40 mV. Notice that
these two levels could produce the same
output of + 20 mV if the higher one had a
maximum negative error and the lower
one had a maximum positive error. lithe
tolerance on the DAC component had
been ± I LSB, then these two levels could
overlap. In other words, the higher one
could have been at + 10 mV and the
lower one at + 30 mV. Hence, monotonicity, or ± 0.5 LSB, is the same specification. In the A/ D encoder, the corresponding name for this criterion is "no
missing codes," but this discussion will
be postponed.
Currently, some manufacturers are
specifying DAC components as having
more than this error criterion. An inexpensive 16 -bit DAC might have an absolute non -linearity specification of ± 2
LSBs. Such a converter is clearly not
monotonic. Each level is very inaccurate.
However, it does have the same accuracy
as a 14 -bit DAC component specified as
±0.5 LSBs. If we had simply not used the
lower two bits of this I6-bit converter,
we would have reduced the error relative to the number of bits we are using.
Manufacturers simply provide us the
extra pins to control two extra bits. These
two bits, although inaccurate, do provide
some advantage. But again, well postpone this discussion, until later.
If you will reread the above discussion,
you will note that we referred to the error as absolute non-linearity. This includes all errors from all sources: gain
error, DC drift error, temperature error,
etc. Many of these are of no interest to
the audio engineer. They result from the
fact that DAC manufacturers come to
audio late and they are used to consider-
ing absolute errors. For the audio engineer, DC drift is almost always irrelevant. Similarly, a 0.1 dB gain change is
not that important. And lastly, most
audio equipment operates in a narrow
temperature range of about 20 to 40
degrees Celsius. The only error which is
of interest is relative nonlinearity: the
degree to which the quantization levels
deviate from a straight line. But only
sometimes is this number provided. Fortunately, Murphy is working for us. since
absolute nonlinearity is always worse
than relative nonlinearity. On the other
hand, manufacturers sometimes provide us only with "typical data," rather
than worst -case. This means that the
application engineer measured the first
dozen units and then took the average. It
says nothing about what we might expect in the current production run. You
pay your money and take your chances.
A maximum error means that each unit
is, in theory, tested to be within that
specification. To make life more interesting, there is a new phrase; "guaranteed
of design. " This means that the manufacturer intends the specification to be a
maximum value, but he does not test
each unit to insure that it works to
within that specification. This is a more conservat ive version of "typical." Whereas 50 percent of the units (or 100 percent)
might be worse than a typical specification, only 10 percent might be worse than
a guaranteed -by- design specification.
Let us consider an A/ D encoder and
D/A decoder made with an imperfect
DAC, but with all other components as
Join db, on a two -week
South American Adventure
February 22 to March 8, 1981
Editor John Woram and
aboard a luxury 1
incredible "encha
the coast of Ecuador.
Then, we flyt o
invite you to join them,
t cla
miles off
itadel of Machu Picchu.
write to
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Or phone: (516) 764 -8900
d th
ideal elements. To do this analysis, we
will use a trick. We will consider the
equivalent analog error which corresponds to the incorrect digital word. This
means that we will only consider the error
in analog terms rather than in digital
terms. The reason for this is that the
quantizer may or may not result in an
erroneous digital word. We used the same
approach when we analyzed the theoretical error produced by a perfect quantizer.
Let us consider a few quantization
levels located at - IO mV, + IO mV, +30 mV
for a perfect converter. We defined the
region between + IO and +30 mV as being
+20 mV. This results in some quantization error for an input signal such as
+12 mV. It is called +20 mV in the
digital domain (center of the quantization region) and there is a -8 mV quantization error. Now, let us introduce some
error by allowing the "true" +30 mV
quantization level to be at an actual
+40 mV ( +0.5 LSBs) and the "true"
+10 mV quantization level to bean actual
0 mV ( -0.5 LSBs). Any analog voltage
between 0 mV and +40 mV will therefore
be encoded to the digital word which is
defined as +20 mV. Notice that the
maximum possible analog error is now
±20 mV, or ±1 LSB. So, depending on
what assumption we make about the
Greg Silsby
talks about
the Sentry 100
studio monitor
Product on Studro WRBR -FM South Bend. Indiana.
In all the years I spent in broadcast
and related studio production work, my
greatest frustration was the fact that no
manufacturer of loudspeaker systems
seemed to know or care enough about
the real needs of broadcasters to design
a sensible monitor speaker system that
was also sensibly priced.
Moving to the other side of the console presented a unique opportunity to
change that and E -V was more than
willing to listen. When I first described to
Electro -Voice engineers what knew the
Sentry 100 had to be, I felt like the proverbial "kid in a candy store:' told them that
size was critical. Because working space
in the broadcast environment is often limited, the Sentry 100 had to fit in a standard
19" rack, and it had to fit from the front,
not the back. However, the mounting
hardware had to be a separate item so
that broadcasters who don't want to rack
mount it won't have to pay for the mounting.
The Sentry 100 also had to be very efficient as well as very accurate. It had to be
designed so it could be driven to sound
pressure levels a rock'n roll D.J. could be
happy with by the low output available
from a console's internal monitor
In the next breath told them the Sentry
100 had to have a tweeter that wouldn't
go up in smoke the first time someone
accidentally shifted into fast forward with
the tape heads engaged and the monitor
amp on.This meant high-frequency power
handling capability on the order of five
times that of conventional high frequency some of the world's best speaker engineers disappear back into the lab to
tweak and tune. And, I spent a lot of time
Not only did it have to have a 3 -dB -down
on airplanes heading for places like Los
point of 45 Hz, but the Sentry 100's
Angeles, Grand Rapids, Charlotte and
response had to extend to 18,000 Hz
New York City with black boxes under my
with no more than a 3 -dB variation.
arm testing our designs on the ears of
And, since it's just not practical in the real
broadcast engineers.
world for the engineer to be directly onThe year was both frustrating yet enjoyaxis of the tweeter, the Sentry 100 must
able, not just for me but for Ray Newman
have a uniform polar response. The
and the other E -V engineers who were
engineer has to be able to hear exactly
working on this project. At this year's
the same sound 30 °off -axis as he does
NAB show it all turned out to be worth it.
directly in front of the system.
Sentry 100's official rollout was
Since I still had the floor, decided to go
universally accepted, and the pair of
all out and cover the nuisance items and
Sentry 100's at the Electro -Voice booth
other minor requirements that, when
was complemented by another 20 Sentry
added together, amounted to a major im100's used by other manufacturers exprovement in functional monitor design.
hibiting their own products at the show.
I wanted the Sentry 100 equipped with a
What it all boiled down to when I first
high- frequency control that offered boost
started the project was that I knew that
as well as cut, and it had to be mounted
the Sentry 100's most important charon the front of the loudspeaker where it
acteristic had to be sonic integrity. I knew
not only could be seen but was accesthat if wasn't happy, you wouldn't be
sible with the grille on or off.
happy. I'm happy.
also didn't feel broadcasters should
have to pay for form at the expense
of function, so the walnut hi -fi cabinet was
out. The Sentry 100 had to be attractive,
but another furniture-styled cabinet with
a fancy polyester or die -cut foam grille
Market Development Manager,
wasn't the answer to the broadcast inProfessional Markets
dustry's real needs.
And for a close told E -V's engineers
that a studio had to be able to purchase
the Sentry 100 for essentially the same
money as the current best -selling monitor
That was well over a year ago. Since that
time I've spent many months listening
critically to a parade of darn good prototypes, shaking my head and watching
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distribution of DAC errors, the actual
quantization process may produce twice
the error of the theoretical converter.
As much as I bit of the converter is lost
in terms of S/N or dynamic range.
A similar process happens at the D/ A
created is the average occurrence of such
errors. To truly interpret the effect of
DAC nonlinearities,
we need to know
something about the frequency of the
errors. Is the worst-case specification
only for one level? Is there a systematic
pattern to the errors, or are they random?
Unfortunately, no manufacturer will
specify his DAC components in these
terms, and we poor designers can only
infer the pattern of errors based on our
knowledge of the manufacturing process.
It would be very nice if we could
assume that the errors were random.
Then our understanding of the effects
would be easy, since a random error
would look like noise. Moreover, when a
decoding process. If we again assume
worst case, the +20 mV quantization
level could be in the opposite direction
of the encoding A/ D. A 0 mV input
signal could be encoded as +20 mV, and
this level might be +40 mV at the decoding output. We now have a peak error
possibility of 2 LSBs. Fortunately, this
worst case does not happen for many of
the quantization levels, and the random
nature of the input means that the noise
random error has a defined peak, we
could expect the RMS value to be much
smaller. But, sorry to say, most of the
DAC component errors are not random,
but systematic. To appreciate this, let us
review the way these are built. Each bit
controls a voltage or current which is
added to the input. Let us consider the
following unit. When all switches (bits)
are off, the DAC produces -10.23 volts.
The first bit (MSB -most significant bit)
adds +10.24 if on, and nothing if off; the
next bit adds +5.12 volts if on, and
nothing if off; the third. 2.56. the fourth,
1.28. etc.
Using this example, we will examine
the state of the switches as we go from
- IO mV to + 10 mV. To achieve + 10 mV.
the MSB is on and all other bits are off,
since we must have the following sum:
-10.23 (DAC baseline) + 10.24
no other switches.
The - 10 mV case is rather different, since
the MSB is off and all other switches
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the DC -63 variable pattern dual- membrane condenser, the DC -96 cardioid condenser, VM Series condensers, F -69 dynamic, CL -4
electret or an XY stereo mic, your professional audio dealer
should be contacted to
give you a chance to
audition for yourself
The Swedish
Steal Has A
New Name: MILAB
the performance
and value of the
Swedish alter-
-10 mV
-10.23 (DAC baseline)
2.56 + 1.28 + 0.64
+ 0.04 + 0.02.
0.32 + 0.16
Notice that to achieve a 20 mV difference, we are comparing two very large
numbers, and one of these numbers is
itself made up of nine numbers. The
20 mV difference is actually superimposed on a 10 volt pedestal. In contrast
to this example, let us compare the case
of the +10 mV and +30 mV levels. One
case is, as before:
IO mV = -10.23 (DAC baseline) + 10.24
and the other is:
30 mV = -10.23 (DAC baseline) + 10.24
+ 0.02.
To achieve this difference, we only
If the LSB were
in error by IO percent, it would be 0.018
volts instead of 0.020 volts. The error
seen is only 2 mV. However, a percent
error in the MSB would produce an error
of 102.4 mV. In other words, a very small
percentage error in one of the bits is likely
to produce a large spacing between two
quantization levels when the levels
involve the changing of many bits. In the
first example, all the bits were changing
since the digital number went from
0111111111 to 1000000000, whereas in
the second example only one bit changed.
The digital number went from 1000000000
to 1000000001.
The first change is called a major carry
since the sum of the LSB must be carried
across all the bits. DAC errors are almost
always concentrated at major carries.
This is a far contrast from random.
The major carries, in audio terms, are
at 0, +half scale, -half scale, +quarter
scale, and - quarter scale. Except for the
first, the major carries happen for high
level signals. The carry at 0 volts is the
most important in terms of audio
it is for very small signals -and it is
the one which is likely to be largest.
need to turn on the LSB.
We're Still The Swedish Steal!
For product information
are on:
write or call your nearest MILAB distributor:
Worldwide Marketing- Creative Trade CTAB AB. Knutsgatan 6.5 -265 00 Astorp.
Sweden. Tel: 42/515 21
United States-Cara International Ltd.. P.O. Box 9339. Marina del Rey. California 90291.
Tel: (213) 821 -7898
United Kingdom- Future Film Developments. P.O. Box 4B5. London WIR. Tel: (01l 437 -1892
Australia-Werner Electronic Ind. Pry. Ltd.. P.O. Box 98, Kilkenny. 5 A. 5009. Tel: (08) 2682766
O 1980 MILAB
Studer 169 and 269.
The mixers with the master touch.
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The Studer 169/269 give you
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whose center frequency is continuously tunable from 150 to
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limiters, complete reverb -send,
foldback, and pan pots, and solo,
muting, and slating facilities.
There's a built -in electret condenser talkback mike and a pre fade monitor amp. 6 -step switches
adjust input sensitivity from
61 to +16dBu, and the floating
XLR connectors provide phantom
powering, as well. Separate line level inputs are included and the
long -throw (4 ") conductive -plastic
faders have additional switching
contacts. Built in low-end and external filters are switch -selectable,
and you have your choice of PPM
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E7R1 o)E.
But whether you pick the
or the 16/2 Model
any of the variety of
10 -in /2 -out 169
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Theory St Practice
Seeking a System
Last month, we discussed the dimensions a digital audio system must encompass, based on information theory.
Now, if we are to use that information,
we must take a closer look at how to use
it. A basic principle to keep in mind is
that, since our objective is to produce an
auditory illusion, we must consider how
the human ear functions.
A fundamental difference is that we
want to transmit, or record our audio
program on a single channel or, on a very
limited number of channels, while the
human ear has a very complex nerve
bundle to convey its received information
to the recognition center in the brain.
We have a similar situation with video.
Before the advent of color TV, a satisfactory system had been developed using
sequential scanning; a dot of variable
intensity that scans the whole picture
area, line by line, 60 times a second or. to
be more exact, 30 frames a second.
Interlacing helped to reduce the apparent flicker, while improving definition
by providing twice as many lines in a
frame. The flicker frequency is raised
from once every 30th of a second, which
is the frame rate, to once every 60th of a
second, which is how often the spot
traverses the screen.
Then came color. That posed a system
problem for which a number of solutions
were proposed and tried. One of them
was sequential scanning; to cover the
picture area three times to get the full
color picture, each time being like one
color of a three -color print. This might
work if all you look at is stationary
objects, or pictures with very slow movement in them. But fast movement means
that successive frames, in different
colors, will be displaced, resulting in
color separation when anything on the
screen moves fast.
This was one deciding factor that led
to the three -gun system that was finally
adopted, so that the whole picture is
covered in the single frame, interlaced
scan, as before. That also had the advantage of compatibility; if a black and white
4600 SMPTE Tape Controller
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on Reader Service Card
picture is transmitted, it is received as
black and white, not a succession of
color- changing pictures.
Now, how can we apply that to audio?
Well. we have all those bits of information that we discussed last month to use
in conveying the full audio information.
How are we going to organize it? Can we
use a frequency scan, which was one
possibility we discussed?
What we are talking about now is a
system that goes through the audio
frequency spectrum once every five
milliseconds (or a little faster. if we can
spare the information), from 20 hertz to
20,000 hertz. However. we encounter a
problem with that. You cannot sample a
frequency of 20 hertz. plus or minus five
cents, in five milliseconds, which would
allow you only one tenth of a cycle of the
20 hertz frequency. And, of course, the
time your scanner is on 20 hertz is only a
very tiny fraction of that.
What all this means is that we must
continuously sense all of the frequencies
being scanned. picking off the information by scanning similar to the way that a
TV camera tube works. Instead of
scanning the image on the picture tube,
you'd be scanning the outputs of a
frequency analyzer, having however
many frequencies we eventually decided
were necessary. That would at least make
it possible to get outputs from all the
frequencies and code them into the
But now we need to think about something else. We picked 5 milliseconds (or
less if possible) as a valid scanning time,
because that was the time at which two
events, repeated in quick succession,
could be audibly identified as separate
events. Without too much thought that
seems sensible. But would you say that
events separated by only, say one millisecond, would not be discernibly different from a single event? Just because you
cannot separate them, does not mean
that the presence of two instead of one
doesn't make any difference.
If you doubt this, consider taking the
same program, putting in a one millisecond time delay, and mixing the delayed signal with the undelayed signal.
Will it make any difference? You know it
will. As frequencies go in and out of
phase, up the audio spectrum, you will
get cancellations and summations every
other 500 hertz. If the interval was a bit
longer, it would sound like reverberation,
but at that short interval, it will completely alter the character of the sound.
Unless you do that, you will not get
that effect. Sequential scanning, in which
you pick up each frequency only once
during the scanning interval, will not
produce that effect. But there is a related
effect we need to consider.
Remember CBS Labs' so- called isophonic system? They showed that, if you
reproduced the initial transients of
musical sound from a string bass, for
example, over little speakers placed to
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enable you to get a stereo location on the
sound, while the real low frequencies
were reproduced from a woofer concealed under the sofa, our human hearing
convinced us that the whole sound came
from a location simulated by the little
We know that happens. But why? For
one thing, it proves that in some way our
hearing faculty pays much closer attention to transient sounds, such as those
initial transients, than it does to "follow through" sounds. If it is just a very
sophisticated frequency analyzer whose
output is fed digitally to the brain, how
can it do that?
Ask yourself what information, that
you know to be in the system (your
human hearing system), could be used
for that purpose? An obvious possibility
is the AGC that we have already discussed at some length. An initial transient
is invariably a sound that is louder than
that being received immediately preceding it. This must involve the same
nervous system that conveys the sounds
heard. A sudden extra loudness sends a
message to the brain, whether by the
nerves associated with the acoustic transformer in the middle ear, or by the more
detailed nervous system from the inner
ear, which initiates a reflex action to
"turn down the gain."
If you think about it, you realize that
this also has to be the mechanism by
which the ear can pay that special attention to transient sounds. And in turn,
that must also provide the key to the ear's
ability to separate sounds that occupy
the same frequency spectrum.
Your ear is already receiving a melody
and harmony that has been going on for
some time, when a new instrument, or
group of instruments, comes in with a
new set of frequencies and overtones. But
many of them are not new, taken individually. They are new only as a set,
recognizable as whatever they are: brass,
woodwind, percussion, or whatever.
Your hearing pays special attention to
this new sound, virtually ignoring the
sound that was already there. You can
still hear it, but it is not part of what your
hearing pays special attention to.
It would seem obvious that the change
in level that triggers your ear's AGC
system is what enables your hearing to
make this distinction. Now, that happens
fast. To the best of our knowledge (we
are open to correction), precise measurements on the speed of this response have
not been made. But, assume they have,
and suppose the reaction time is the same
five milliseconds that has been identified
as being necessary to hear repetitive
sounds as separate events. What then?
If this is the "timer" that prompts a
closer analysis of the change in sound
five milliseconds after the change in level
hits, then it will analyze the change in
content at that instant. It can do so,
because every frequency has a separate
nerve fiber along which to transmit that
information. Your digital system is not
so equipped.
This would suggest that our system
needs a device that similarly pays special
attention to such transient sounds. A
constantly -scanning device would not do
it. When a transient arrives, some of the
new elements would be "in" by this time
around, while others would not make it
until the next scan. For it to work right,
all transients would have to be timed very
precisely to coincide with the beginning
of a frequency scan. Otherwise, the
quality of the transient would be falsified.
There's another angle to think about,
that may correlate with that. In the
analyzer of the inner ear, vibration of the
fluid couples with the resonant transverse
fibers that stimulate the nerve endings.
A soft tone, or a small level of some
frequency, may stimulate only the nerve
endings directly associated with that
fiber. A louder one will stimulate not
only more nerve endings associated with
that fiber, but also some of those associated with adjoining fibers, whose
resonant frequency is above and below
the frequency doing the stimulating.
In the human ear, all of this information is assimilated to convey the intensity
of that frequency element. It is that way,
because nerve fibers convey information
only relatively slowly, and so the system
uses a lot of fibers to convey more
information than each one can by itself.
We need to translate that principle for use
by a system with only one channel, but
very short time -element bits.
Perhaps continuous scan is not the
best. Maybe we need to be able to synchronize a scan with any transient that
comes in, and perhaps fill in, between
transients, with something more like a
continuous scan. Another thing to think
about is: do we need a complete scan
from 20 hertz to 20,000 hertz, giving
information about every frequencyhowever many we decide to use -in
between, for every scan?
In video, continuous scanning seems
to be necessary (who knows, maybe some
day it won't be) for every point to keep
its correct place in the picture. Sync
signals are used to lock the scan for line
and frame, and everything stays locked.
But in video, you must have a complete
picture. The frame must be filled with
something. In audio, this is not necessarily
Maybe it would be better to devise a
scan which accurately identifies each
frequency whose intensity is enough to be
significant, and then gives information
about its intensity. Conceivably, this
could reduce the information requirements a considerable degree. During
each scan, each packet of information
would consist of a code to indicate that
the next data identifies frequency,
followed by one that would separate
frequency identification from intensity
identification. Does this give you
something to thing about?
Sound Wich Images
Scenes From a Video Notebook
This observer's been doing a lot of
reporting for the consumer press lately
on the microcomputer revolution, "as it
will have impact on both video and audio
and join them together." Join them
together? Well, these are the words one
uses in describing pop phenomena in pop
magazines-get more specific and the
editor won't understand a thing, and will
simply change it to his own even more
misleading usage. (Never! -Ed.)
Consumers are not only getting the
word, they're getting a lot of expectations. What these expectations mean for
the audio professional is a changed
industry. People are expecting to be able
to afford a videodisc or VCR in their
homes; the more affluent may find they
must have both. This means at once
greater opportunity -more programming surely will be needed -and greater
quality expectations. The know -it -all
final customer may say, "I can do that
better at home," and then do it.
If the film experience is any guideline,
syntax, of accertain standards
ceptability -will probably relax as video
Our Stereo Synthesizer
isn't just for old mono records.
Applications of the 245E are limited only by your
save tracks by recording strings, horns or drums
on a single track and spreading them in the mix
create stereo depth from synthesizers, electronic
string ensembles, and electric organ
create a stereo echo return from a mono echo
chamber or artificial reverb generator
use one channel to create phasing effects
a dramatic, highly listenable sound that's fully
mono compatible -just add the channels to get the
original mono back. (If you get bored, you can
always process old mono records into pseudo
Your Orban dealer has all the details. Write us for his
name and a brochure with the complete 245E story.
Orban Associates Inc. 645 Bryant Street, San Francisco, CA 94107 (415) 957 -1067
on Reader Service Card
Already out on the limb, I'll now go
further. Recently, I was talking with a
record producer who'd worked with
Todd Rundgren on a recent album. One
does not have to admire Mr. Rundgren's
music to see some sense in his views of the
home video medium -as this producer
understands them, in any case. Rundgren
is now involved in making video accompaniments to his (or others') music.
Although he intends to make these as not
merely accompaniments, he does not
intend to produce much in the way of
linear stories, either. Rather, in my
acquaintance's expectations, a new
genre is evolving, geared for giant screen TVs that will constantly be playing
becomes more and more available. As
smaller, lighter film equipment became
available in the 1960s -not only within
that industry but to almost any interested
party -jump cuts, uneven lighting,
uncertain screen direction, jumpy cameras and other tics common to home
movies, went big time. Expectations
changed, and the language of film changed.
Now, many people are buying their
own video decks. But how many of them
will ever purchase two decks and the
editing console necessary for smooth looking cuts? I will go out on a limb and
predict that glitches will become commonly seen, and eventually accepted as
some form of scene transition.
Only VIZ bench DMM's
tell so much for so little
AC or DC
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his work.)
Now, with a bit of a glitch and a different color balance to mark the passage of
time, we switch scenes to a hundred
degree (last August) press conference
at which Technicolor announced their
plans to market a seven-pound, quarter inch videotape recorder. The recorder is
made by Funai, of Osaka, Japan, and has
been marketed there for several months
already. By the time you read this,
Technicolor supposedly will have tens
of thousands of these things out in the
retail stores, selling for $995.00. The unit
employs quarter -inch video tape, in a
cassette just slightly larger than a standard compact audio cassette. The tape
is made by Fuji -only in half-hour
lengths at first, then in hour-long size.
The format is helical scan, with a loading
system similar to that of VHS. Even if
Technicolor does not come near their
marketing objectives, I remain very
impressed with this recorder -its quality
may not be equal to half-inch, but it's not
far from it.
Other quarter -inch formats are coming
soon from Toshiba, Kodak, and. by
1985, Sony, so they say. In competition
with all of this are portable half-inch
VCRs coming in at lower prices than
ever. Stripped down, non -portable units
were shown at the Consumer Electronics
Show by Sanyo and Sharp, bringing
those prices down to the $600 range.
Where is all this going to take,us? The
Technicolor people expect their product
to compete with Super -8 film, and not
with larger-format video. Surely, Super-8
already losing this competition.
Eventually, quarter -inch video must
also take over from half -inch in the
home, just as the cassette has taken over
from open reel in consumer audio. Someday, half-inch may be the professional
medium. Or, more likely, it will linger,
but constantly decline in importance,
while three -quarter -inch solidifies its
position. But the video "semi- pros" will
use half-inch, as the audio semi -pros
(what an awful word, says everyone who
uses it).
With another glitch we slip back to
the CES again, for two other developments of some interest to this column.
The show could almost have been subtitled, Audio is Dead, Long Live Video.
VIZ Mfg. Co., 335 E. Price St., Philadelphia, PA 19144
Over 70 test instruments in the line
Circle 31 on Reader Service Card
cross between a wall
hanging and a hi -fi. One puts on a record
while talking with friends. Occasionally,
something comes on that all will listen
to; but often, as interesting as the record
may be, it will merely be background.
Deliberately, the record does not contain
all "highs" (speaking experientially, not
in terms of frequency). And, if it told a
linear story that needed attention all
the way through, it would probably be
less successful. So will new video programming eventually turn out, my
acquaintance and I agree. (It would be
interesting to see if Mr. Rundgren agrees
with this producer's interpretation of
I modified that with "almost," but, in
retrospect, audio's descent was taken
for granted by most of those at the show,
and if video hasn't taken over everyone's
affections, all saw it out there on the
horizon. Among the variations of consumer adaptations being shown, were a
couple with professional interest.
One of those products, that hopefully
foreshadows the start of a trend that
would be very welcome, was Akai's new,
portable, VHS format, Activideo System.
The important part: the soundtrack
utilizes Dolby B, a step that ought to
have been taken a long time ago. Easier
provision for double- system recording
(the usage of a separate audiotape
recorder operating either with line -sync
or connection -free crystal sync, such as
those made by Kudelshi- Nagra, Tandberg,
Uher, and others) would be the next
logical step.
Actually, we were informed (and asked
not to breathe details on this, so we
proceed to speak in generalities) that a
video noise reduction system is coming
closer to being accepted by the manufacturers of video decks. Ray Dolby has
been interested in an analogous video
noise reduction system since his first
patents. Such a system would be welcome
indeed for home video, and in fact should
be even less obtrusive than audio's
Dolby B. But any more said would be
Copies of db
Copies of all issues of db -The
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The other promised development of
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with true high fidelity, that is often
present in the large formats. For that
matter, it's one of the few ways to get any
high fidelity sound along with a video
show, excluding systems custom -made or
those that are much more expensive. It
does not include its own speakers, but
hooks up to external monitors. In all
regards, including price, the unit hardly
belonged at the CES at all, being perfectly suited for the professional who sets
up systems (particularly impermanent
systems) in industry.
Such a product should remain handy,
as all of the video studio's gear becomes
so portable that the studio itself will be
all but obsolete. With the uses of video
growing, and location work becoming
the rule, the recording engineer will most
likely be expected to be able to work with
both audio and video. Yet, though the
uses of the medium should keep growing,
all signs point to its job glamour keeping
the field as competitive as ever.
Enough of prognostications, for now.
By next time we should have assimilated
information from yet another show,
the Fotokina in Cologne, Germany, and
film will again take the stage.
ns!ara WI,.
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The GEM -7 is an eight section full
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The M97HE-AH is a precision integrated cartridge-headshell with a universal four -pin bayonet headshell connector for instant installation in many
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Circle 51 on Reader Service Card
Jensen Tools Inc. has developed a new
kit for repairing printed circuit boards,
with the emphasis on small component
removal and replacement. Designated
the JTK -47, the kit contains a 35 -watt
soldering iron, fork and hook soldering
aids, a multi -position work holder and
quick- adjust vise for holding circuit
boards, DIP extractor, miniature chain
nose plier and more. The tools are
furnished in a 19 x 7 x 7 -in. heavygauge steel tool box with a tray and
extra room for additional tools.
Mfr: Jensen Tools Inc.
Circle 52 on Reader Service Card
rack The Gemini Easy Rider
mount. stereo /dual mono compressor/
limiter, has recently been introduced by
Audio and Design Ltd. Optimum attack
time is calculated by a control which
responds to program characteristics.
Slower settings can be used safely, since
the unit will adjust its attack automatically to handle unforeseen peaks. Dynamic attack change, relative to level,
can range from 500 microseconds to five
milliseconds. Release (recovery) time
can also be programmed or set between
15 milliseconds and four seconds for
specific signal shaping. The unit offers
33 dB make-up gain, with a 25 dB control
range from limiting onset to a maximum
clip level of +18 dBm, with preset output
control user -calibrated between -10 dBm
and +12 dBm referenced to limit threshold. A 20- segment LED bar graph, which
reads gain reduction over the 20 dB scale,
is set into the unit's laminated plastic
front panel. Input /output /earth connections are via 12 -way tag strip with independent "side-chain" access facilitated
by a three -pole jack socket.
Mfr: Audio and Design Ltd.
Price: 8875.00
Circle 54 on Reader Service Card
Each TR -50 has a built -in crystal
controlled FM transmitter, super hetrodyne receiver, standard nine volt
battery supply and seven -inch receiving
antenna. A limiting circuit prevents
receiver overload, while a sensitive
squelch circuit drives it into "quieting"
during times of no transmission. Each
operator hears his own side -tone as an
indication that transmission is taking
place. The wireless intercom headphones
have five channels available for operation.
Mfr: R- Columbia Products Co.. Inc.
Price: S275.00
Circle 55 on Reader Service Card
The 22-4 and 22 -2 are two new re-
corder /reproducers in the Tascam
Creative Series. The 22-4 is a compact
4 -track 15 ips multichannel recorder with
sync that features: function and output
select; headphone monitor select; pitch
control; optional dbx interface and
optional remote pause controls. The
22-2, a compact 15 ips half -track recorder, features expanded scale VU
meters, independent monitor and record
ready controls, detachable head housing
and optional remote pause control.
Both units are three -motor three -head
transports with precision moulded reel
tables and spring- loaded reel holders.
A new audio processing device designated the SG -200 dual signal gate, was
recently unveiled. The SG -200 contains
two totally independent gates sharing a
common power supply in one, 134 -inch
high, rack mount chassis. Operating
controls for each gate include attack
time, release time, range of attenuation,
threshold, a totally silent in/ out switch,
and an internal /external gating command control.
Mfr: Srmetrix Professional
Audio Products
Price: $399.
Circle 58 on Reader Service Card
Mfr: TEAC Corporation
Price: 22 -4: $1,425.00. 11 -2: $750.00
Circle 56 on Reader Service Card
MS-14 (Ix3)
The Bulgin Soundex Audio Multi meter is a multi -purpose instrument
suitable for line testing and listening,
peak program metering, amplification
of microphone signals, calibration of
peak program monitors, bench testing
and other audio functions. The instrument combines a switched gain amplifier
with 400v peak instrumentation input
and a full spec Peak Program Meter
capable of audio program level measurements down to -72 dB with 0.1 dB accuracy at center scale, as far as -50 dB.
Amplifier input is fully protected to
400v, isolated and balanced to prevent
grounding when connected to a jack
field. The 50 ohm impedance output
has sufficient power to drive headphones.
Gain settings are achieved by eight pushbuttons on the front panel. Four other
buttons provide On /Off, battery test,
600 ohm termination and access to a
front panel variable gain potentiometer.
Mfr: H. R. Kirkland Company
Circle 57 on Reader Service Card
MS- 15 (4x3)
MS- 14.
WIN Pase
pro...ppwi Sat.
000) 638-3457
rwx 910-397-6995
l., v.,...
89101 u.S.A.
Circle 27 on Reader Service Card
4icidio T
Incorporating today's technology in a
live performance unit, the ARP8 features: monitor and effects submix bus;
built -in analog delay to provide the
desired echo effect; two 7 -band graphic
equalizers (one stereo for program left
and right; one mono for the monitor
submix) for balance among all the instruments in use; 3 bands of equalizers on
each of the eight channels; 3 VU meters
(for program left, right and submix),
headphone, cue and talkback features;
auxiliary inputs, direct bus inputs,
stacking inputs, and effects send and
Mfr: ARP Instruments, Inc.
Circle 59 on Reader Service Card
for professionals
Ampex, 3M. All grades.
On reels or hubs.
CASSETTES, C- 10 -C -90
With Agfa, TDK tape.
All widths, sizes.
Shipped from Stock!
Ask for our recording supplies catalog.
1233 Rand Rd.
312/298 -5300
Des Plaines, IL 60016
Circle 34 on Reader Service Card
book is for the AUDIO
recordist, engineer or
designer. Offered at
$45.00 you may order
direct from publisher.
This is induction loop equipment of labora-
tory quality for primary standardization of
tape recorders and tapes. Send for detailed information, prices and formats.
819 Coventry Road
Kensington, CA 94707
A three -piece disco speaker system
consisting of a low- frequency cabinet, a
midrange / high -frequency cabinet and an
electronic crossover/ equalizer is now
available from Electro- Voice. The
HFI2 -3 speaker system features an EVM
12L woofer, E -v's VMR vented midrange speaker and an ST350A tweeter.
The LFI18 low-frequency speaker system is intended to be floor mounted.
where it will produce undisturbed bass
down to 28 Hz. Protecting the woofer
from damaging subsonic information
generated by record surface irregularities is one of the main features of the
XEQ -lA crossover /equalizer. This is
accomplished by an integral high -pass
filter. Crossover frequencies are determined by plug-in modules, with the recommended crossover of this system being 125 Hz. The built -in switchable
Thiele equalizer extends the response of
the LFI 18 to 28 Hz. One XEQ-IA is required for each stereo channel.
Circle 61 on Reader Service Card
Circle 35 on Reader Service Card
Mfr: Electro -Voice
The Model 3350 is a three range, 21
frequency, reciprocal 12 dB boost or
attenuate equalizer. The high and low
range equalization curves may independently be selected as either peaking or
shelving. A 50 Hz to 15 KHz band -pass
filter may be inserted exclusive of all
other equalizer settings and an In-Out
switch with LED status indicator silently
switches the equalizer networks in or out
of the circuit. The three frequency ranges
are overlapping and are controlled by
the outer knobs of concentric switches,
the inner knobs of which set the amount
of boost or cut in steps of 2, 4, 6, 9 and
12 dB. The Model 3350 is a panel mounting unit 51/4-in. high by I 1 -in. wide and
53/4-in. behind the panel, requiring a
bipolar 15 volt dc supply.
Mfr: Modular Audio Products
Circle 60 on Reader Service Card
20,000 copies in print
Fourth big printing of the
definitive manual of recording technology!
'John Woram has filled a gaping hole in the audio literature. 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 much in demand
that we've had
So widely read
to go into a fourth printing of this all- encompassing guide to
every important aspect of recording technology. An indispensable guide with something in it for everybody
to learn. it is the audio industry's first complete
handbook on the subject. It is a clear, practical,
and often witty approach to understanding what
8 clearly- defined sections
18 information -packed chapters
The Basics
Il. Transducers: Microphones
and Loudspeakers
,phone JcaQn
ophone Technique
Noise and Noise Reduction
Noise and Noise Reduction
Studio Noise Reduction Systems
VI. Recording Consoles
The Modern Recording
Studio Console
VII. Recording Techniques
III. Signal Processing Devices The Recording Session
cho and Reverberation
The Mixdown Session
:ompressors. Limiters and
Table of Logarithms
Power. Voltage. Ratios and
ranging and Phasing
IV. Magnetic Recording
Frequency Period
ape and Tape Recorder
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
every student ... for every audio enthusiast.
1120 Old Country Road, Plainview. N.Y. 11803
copies of THE- -: CORDING
STUDIO HANDBOOK $37 50 On'15 -day approva
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WORLD, most
not all- microphones are first-order." This little tidbit of news
is from our microphone issue ofJuly, 1978, and
it refers to the types of polar patterns that are
readily available in recording studio microphones.
Theoretically, the polar response of any microphone
may be plotted from its "polar equation." For those who
worry about such things, the general equation is, A +
As the values of A and B are varied, the
microphone's pattern varies, from omni (A = 1, B = 0),
through the various cardioid patterns, to bi- directional
(A = 0, B = 1). In effect, the polar pattern potentiometer
on some dual- diaphragm microphones varies the
equation, to produce the desired response.
With a little knob-twiddling, we can set both A and B
equal to 0.5, giving us the familiar cardioid pattern. As we
continue turning, the cardioid pattern gets progressively
narrower. At the same time, a rear lobe appears. and
becomes progressively larger, until finally we arrive at a
figure -8 pattern.
Now, let's just remove that rear lobe entirely. and we
have a very narrow angle uni- directional mike: just the
thing for miking an acoustic guitar surrounded by drums,
bass and everything else. The mathematicians will tell us
that to do this, all we need is a "higher- order" equation.
For example. a sixth -order cardioid has extremely small
rear lobes, and is down about 6 dB at only 25 degrees
Hold on to your check books: here in the real world,
not all- studio microphones are "first-order,"
and you don't get your -6 dB until 70 -to-90 degrees. As
usual, we find that practice has a way to go before
catching up with theory, and those higher-order mikes
are still off in the future somewhere.
How far off? Some twenty years ago, the late Ben
Bauer wrote, "One feels we should be due for a
'breakthrough' in transducer technology. "He speculated
about a "... 'zoom' microphone, in which the directional
pattern can be adjusted to conform... to the optical angle
of a television camera." "A Century of Microphones"
(Proceedings of the IRE, May 1962).
At the recent AES convention, engineers from JVCthe Victor Company of Japan-presented a paper
entitled, "Zoom Microphone" (AES Preprint 1713, by
Ishigaki, Yamamoto. Totsuka & Miyaji). The paper's
conclusion: "The zoom microphone... synchronized
with video cameras, can provide a good integration of
sound and picture."
In other words, practice is catching up with theory, and
the second -order concept, which was actually introduced
by Dr. Harry Olson in the mid-40s. may eventually find
its way into the real world. By the way, JVC's zoom
microphone is made up of three spaced uni- directional
elements, two of which are used to create the second -order
characteristic. The microphone's polar pattern potentiometers are linked to the camera's lens, so that as the lens
zooms, so does the microphone! (Just as Bauer predicted
in 1962.)
Of course, there's a bit more to all of this than simply
combining the outputs of two first-order microphones.
Furthermore. the off-axis frequency response is still not
as good as we would expect to see in a studio -quality
microphone, although no doubt it's fine for its intended
purpose. Of course, sooner or later we shall see (and hear)
second -order microphones that are up to pro- studio
standards. In the meantime, it pays to remember that
when any two microphone outputs are combined, all
sorts of unexpected patterns may result.
Last month's "mike quiz "(buried in our "Coming Next
Month" paragraph) is now over. Of all the lures (for
complimentary subscriptions) that we've planted in the
pages of db, this one drew the least response, and the
most wrong answers, too! (The right answer is to be
found in this month's Application Note.) Perhaps this
confirms what many of us suspect: microphones are often
evaluated solely in terms of their "sound" (on guitar.
snare drum. or whatever). Just as often, all the other
variables are either misunderstood or ignored.
If this is so, we have an excellent excuse for a series of
articles on microphones, which is just what you'll find in
this month's db. We still have no word about our
definitive kazoo mike, but after scanning these pages,
perhaps you'll come up with your own definitives. which
are better than ours anyway.
This month, our features range from theory. to
practice, to anecdote. Bruce Bartlett reviews some of
those little distractions that often get in the way of good
microphone sounds. John Eargle offers some thoughts
on miking pianos and other strings. And Ralph Hodges
brings us some of engineer Fred Catero's views on mikes
and miking (and everything else, for that matter).
We also offer another db Application Note, and a
directory of microphone manufacturers. And, since this
is a transducer issue, that means loudspeakers too. We
asked John Borwick to tell us about a two -mic, thirty -six
speaker live performance recording that recently tested
the nerves of the BBC and KEF Electronics, Ltd. Also,
Greg Silsby gives us an overview-from microphone to
loudspeaker -of what it took to provide audio at the
Republican National Convention.
For still more on the subject of microphones, keep an
eye out for our early-1981 announcement of a new book
written by John Eargle. The subject? Microphones, of
course! No complimentary subscriptions if you spot the
announcement, but there will be a special pre-publication
offer to entice the early-birds in the audience. Keep
Scene from Europe:
A Classical
Speaker Situation
The concert on the last night of the 34th Annual International
Festival in Edinburgh, Scotland, presented the sound and
television engineers with a unique problem.
concert in Edinburgh's Usher Hall
was given by the London Symphony Orchestra,
conducted by Claudio Abbado. The work to be
performed in the second half was the massive
Berlioz Ïe Denim in which the LSO would be joined by the
Edinburgh Festival Chorus, the Scottish National Orchestra
Junior Chorus, Philip Langridge (tenor) and Gillian Weir
(organ). Actually, Gillian Weir did not exactly quite join the
others-she played the recently- restored "Father Willis" organ
in St. Mary's Cathedral, a mile away. The Usher Hall organ was
in need of renovation and Claudio Abbado suggested the
John Berwick
British rorrespondem.
relaying of a distant organ. After thinking about Notre Dame
in Paris and Paisley Abbey, it was decided that Gillian Weir
would play the magnificent organ in St. Mary's Cathedral and
the sounds would be relayed to the Usher Hall and reproduced
as if the organ were really there in the Hall itself. This "place
shifting" is quite common in the recording industry --we
think of the recording of the Saint -Saens Third Symphony
(CBS 2530 619) which brings together the Chicago Symphony
Orchestra conducted by Daniel Barenboim in Chicago and
Gaston Litaize playing the organ in Chartres Cathedral
in France.
But the Edinburgh experiment was much more complicated
and risky. The organ sounds had not only to be picked up
accurately and transmitted to the concert hall, they had then to
be reproduced at full volume and with a realistic spatial effect.
There was also the need for complete and confident
synchronism between the organist and the conductor with his
500 singers and musicians. Alan Bunting, the BBC's audio
manager in Scotland. was in charge of all the technical
arrangements for the programme and communications circuits. He also had to prepare everything for BBC Television,
who were telerecording the Berlioz work live on Saturday for
transmission by BBC I and several other countries on the
following evening. He called in Raymond Cooke of KEF
Electronics, and they devised an ambitious scheme using 36 of
the KEF Model 105.2 loudspeakers, each pair powered by a
separate Quad 405 power amplifier, 3,600 watts in total! (The
model 105 is seen in Made in Britain in ourJune issue, and for a
look at KEF Laboratories. see If We Can Hear h. We Can
Measure It in the July db -Ed.)
Twenty of the speakers were placed in a straight line at the
rear of the platform, along the base of the existing organ pipes,
ten on each side of the organ console. A further group of sixteen
speakers was situated centrally at the back of the Upper Circle
to produce the antiphonal effects called for by Berlioz (though
these were used at reduced power in the final balance). The BBC
engineers used a simple crossed -cardioid pair of microphones to
pick up the organ in the cathedral, placed at a rather close three
metres. since the sound emerging from the speakers would have
the added reverberation of the Usher Hall. The stereo signal
from the microphones was modulated on to a radio
transmitter/ receiver system. This was also used to carry the
two -way closed circuit TV signals, with cameras and monitor
Gillian Weir could see the
conductor, and hear the orchestra and singers on a small
loudspeaker placed alongside the console. Similarly, Claudio
Abbado had a TV monitor on his rostrum which allowed him to
see Miss Weir. He also had a microphone to talk to her during
rehearsal, and I noticed that the assistant leader of the orchestra
used this to ask Miss Weir to play an A to which the orchestra
could tune before the leader and conductor made their
entrances on to the platform.
The normal air of expectancy was made more electric as the
musicians and the 2,500- strong audience asked each other
"What happens if the organ link -up goes wrong ? ". The Te
Dew); opens with an interchange of loud chords between the
orchestra and organ, and these are negotiated with no trouble at
all. One of the main worries of the engineers had been to get a
high- enough acoustic power level without overloading the
amplifiers or speakers. The orchestra and organ chords both
peaked to 90 dB on a sound level meter in the Grand Circle. The
readings thereafter ranged from 37 dB(A) in quiet solos to 115
dB peaks when the five cymbals joined in. In fact, the organ
seemed almost too loud at first to my ears, but there was no
suggestion of clipping or distortion of any kind. During quiet
passages from the organ, such as the extended introduction to
the Judex crederis (which Berlioz described as "without doubt
the most grandiose piece I have ever created "), the inevitable
noises from the close microphone/ organ balance did reveal
themselves slightly. Using more distant microphones would
have reduced this effect, but at the expense of picking up too
much reverberant sound. not to mention the ambient noise,
compounded of remote traffic, wind etc. As it was, I could
detect the longer reverberation and just a suggestion of that
indefinable cathedral ambience at each organ entry. Circuit
screens in both locations. Thus.
Figure 1. The scene in the Usher Hall, during a
performance of Berlioz' Te Deum at the final concert
of the Edinburgh International Festival 1980. KEF
loudspeakers Model 105.2 are installed in two stacks of
10 at each side of the organ casing behind the choir.
2. Laurie Fincham and Colin Munro of KEF
Electronics setting up the loudspeakers in the Usher Hall.
Figure 3. The BBC mixing desk at St. Mary's Cathedral,
noise in the radio links was also a contributing factor as it was
only 53 dB below maximum signal level, and the organ dynamic
range exceeded this.
The performance was voted a huge success by critics and
audience alike. Will it ever be repeated this way? I doubt it.
spoke to Claudio Abbado and Gillian Weir afterwards and they
both praised the technical and musical competence of the
engineers concerned. However, they were looking forward to
their next performance of the work a few days later in
Belgium -where orchestra, choir and organ would all be in the
same auditorium.
Such collaborations between musicians and engineers often
arise in avant garde music of course (as well as in pop and rock
concerts). recently attended a concert of music by Karlheinz
Stockhausen in which one work for live musicians and 4 -track
tape involved the composer himself seated at a Neve mixing
console perched on top of the seats half-way back in the stalls
with huge speakers in all four corners of the Royal Festival
Hall. On another occasion, I remember my tonmeister students
at the University of Surrey being disappointed with the quality
hired Steinway piano, which stood alongside our own
Steinway in the University Hall for a concert recording of the
Sonata for Two Pianos and Percussion by Bartok. Immediately
after the concert, the students recorded it all over again -this
time with our other excellent Steinway located in our music
studio underneath the Concert Hall. They put TV cameras and
monitors in both locations, with foldback headphones
everywhere. And they proved their point. Comparing the
recording made live at the actual recital with the new one- in
which careful microphone balance had somehow matched the
hall and studio acoustics, with judicious panning of the two
pianos to quarter left and right- showed the superiority of our
own instruments. And anyway, it was fun to do.
Figure 4. Two Calrec 1050C cardioid mies were used in a
cross-cardioid configuration to mic the organ at St. Mary's.
Technical Data
Organ microphones
Two Calrec 1050C cardioids, as angled coincident
pair. 10 feet from pipes and 15 feet above floor.
Radio link
Main and standby picture transmissions at 7.125
gigaHertz with stereo organ signals on sub -carriers.
Mono reserve channel on 141 MHz VHF.
Total of 14 audio circuits including two -way voice
Circuit noise -53 dB.
Local control
Two BBC 6 -way mixing desks for stereo signal to front
speakers and mono (left plus right) to rear speakers.
Glen Sound distribution unit, 600 -ohm balanced lines
peaking to
+ 10
KEF designed balanced -to- unbalanced converter/
attenuator reducing level to 500 mV.
18 Quad 405 stereo power amplifiers, one channel per
speaker, delivering up to 28 volts rms (100 watts):
flat down to 30 Hz, -2 dB at 16 Hz.
Total power 3,600 watts.
Total weight 162 kg (356.4 lb).
KEF Model 105.2 speakers, sensitivity
dB/ watt
Two bands of 10 speakers on each side of organ casing
behind choir.
speakers at back centre of Upper Circle, grouped
6 + 6 + 4 on 22 -inch risers.
Total weight 1,296 kg (2,851.2 lb).
Hall acoustics
Volume 565,000 cubic feet (compared with 2.500 -6.000
cu. ft. for an average living- room).
Reverberation time 1.65 s (with audience).
Noise level less than 30 dB(A).
Measured music levels during performance
115 dB (linear).
Why The MTR -90
When you're buying studio time, it
can save you a bundle. If you're
selling the time to your clients it
can make you a bundle.
It's the MTR -90. The 16/24 channel
professional recorder that is the state
of the analog art. It's the new machine
that outpaces the big names. And, you
know who they are.
Here's why we're so confident:
Superior Tape Handling
Especially critical for wide-width
tape, the Otani Optimal Tape
Guidance system was the
industry's first three motor,
pinchrollerless two -inch tape
transport; a system so superior to
conventional pinchroller designs that it
is also utilized on a competitive
machine costing twice as much
money. The MTR -90 treats the
important ferric oxide tape coating like
a precious metal. Compared to
conventional designs, the sonic
"shine" and brilliance of a master
recording stays on the tape. Smooth,
even tape packs in all operating
modes are the rule, not the exception.
Award- winning recording engineer Phil Seretti, owner of the one-hundredth MTR -90 and
producer Janja Vujovich
Advanced Audio and Control Circuitry
Easily accessible single card
electronics not only save money,
but reduce the complexity and
problems of interconnection failures.
Active mixing of audio and bias
minimizes ringing and set -up
difficulties. There's a transformerless
playback amp for optimum transient
response and dynamic range (greater
than 71dB, 24- track: 30 ips,
unweighted 30 Hz -18 kHz @ 1040
nWb /m). High slew-rate components
in critical signal stages give you better
aural results: Distortion -less than 0.5%
at t kHz(250 nWb /m), Output: +28dBm.
Punch -ins and outs are totally
transparent and effortless due to an
integral digital timing section on each
audio card that precisely ramps the
erase and bias currents to yield
"gapless" performance, Transport
logic is digitally controlled for reliability
and ease of servicing. There's a
master crystal controlled reference
clock for capstan, counter, record
timing, bias and erase signals. Easy,
rear -panel access to time -base
functions facilitate SMPTE interface.
Easier To Maintain
The V.U. meter panel is hinged to
give you wide open access.
Remove the side panels, open
the electronics bay doors and there is
nothing you can't get to: power supply,
master bias level, playback
equalizations, motor drive amps,
capstan and reel motors. The MTR -90
is designed for the real world of studios
where routine maintenance and
care shouldn't have to be a headache.
Single card audio electronics contains record, reproduce, synchronous and timing circuits
rIt' 26
,nt Reader .tirrrrri' C-ard
Outruns The Herd.
The Extra Thought
he MTR -90 makes sessions go
smoother because we also
designed -in such features as a
VSO that offers ±20 °% speed variation
with 0.1 resolution: a precision.
continuously variable edit and cue
control: and whenever you go into
start /stop or fast wind. a circuit
automatically mutes playback level-
to expect. Competing not
only on features and ad-
vanced technology, but
also with the ruggedness
and essential service
back -up so crucial to a pro-
fessional products total
acceptance. After all, we
do know that you make
your living on our products.
And we take that very seriously
because we make
our living from you.
If you're moving up to
a better. larger format machine, or moving the old
one to the side. you need
to get acquainted with the
Easier maintenance with convenient access
to all internal components
feature that designed to reduce ear
fatigue and save your monitors. Every
MTR -90 also comes with the
industry's most advanced remote
session controller. And. when you
need the benefits of a sophisticated
ten position memory locator. just
plug -in to the back of the MTR -90 and
place the optional locator atop the
companion session controller's
convenient roll- around pedestal.
If you've gotten the impression
that its advanced... good. If you also
have any questions about how it will
hold up to professional requirements,
then let us assure you that we are
prepared to stake our 16 year
reputation for unparalleled reliability
on the MTR -90. We've earned our
place among audio professionals by
competing with the best you've come
Workhorse. You ll find
out for yourself why it
outruns the herd. Just
contact any of these
fully committed dealers.
Then arrange for a demonstration at
your studios. The MTR -90 will give you
every reason to consider that if you
buy something else. you just might be
buying something less ... for more.
Watch for Otari's MTR -10 Series.
They're the companion 14" and 12"
mastering machines that join the
MTR -90 professional recorder.
MTR -90. The New
f-7 f -T1
Remote session controller and memory
locator for complete man /machine interface
EVERYTHING AUDIO 12131995.4175
EXPRESS SOUND (7141645 -8501
FLANNER'S PRO AUDIO (414)259-9665
MARTIN AUDIO 212(541 -5900
(617(254 -2110
SOUND GENESIS (415)285 -8900
VALLEY PEOPLE(615)383 -4737
WESTBROOK AUDIO (214)699 -1203
Otan Corporation
1559 Industrial Road
San Carlos. CA 94070
(415) 592 -8311
BSR (Canada. LTD
PO Box 7003
Station B
Rexdale. Ontario M9V 4B3
Hum, Pop, Thump
and Other
Microphone Noises
Presenting some techniques to aid in the reducing of unintended
inputs and microphone noises.
of a microphone is to pick up
sound and convert it to an analogous electrical waveform. Unfortunately, a microphone is also sensitive
to unintended inputs, such as pop, hum, and handling
noise. Wherever there is concern for high -quality audio,
reducing pickup of these unwanted signals is important.
To help the user do this, this article will discuss the nature and
measurement of microphone-related noises, and will describe
several methods to minimize them.
In almost any room where a microphone is used, an
alternating current exists in the electrical wiring inside the walls.
floor and ceiling. The current oscillating through the wire
conductors creates an oscillating magnetic field in the room. If a
dynamic microphone is used in this room, the conductors of the
microphone voice coil cut the magnetic lines of force from the
oscillating field. An AC voltage is magnetically induced in the
voice coil and, after amplification, this signal is heard as a low
tone or buzz called "hum." Magnetic hum also can be induced
in impedance- matching transformers and response-shaping
networks with inductors inside the microphone. FIGURE
shows a schematic diagram of magnetic hum coupling.
Although AC power is commonly thought of as a 50 or 60 Hz
sine wave, it is often rich in harmonics due to transformer core
saturation. The higher the frequency of the oscillating magnetic
field, the greater the rate of change of flux, and so the greater the
induced voltage is. Thus, the high- frequency harmonics are
picked up more readily than the 60 Hz component of the hum.
Figure 1. Magnetic hum coupling between a power
line and a microphone.
High- frequency components are also much more audible. The
result may be a "buzzy" sound in the hum pickup.
Other sources of magnetic hum besides power wiring are
transformer radiation, SCR dimmers, and fluorescent lights.
SCR dimmers operate by clipping the power -line voltage and.
consequently, generate strong harmonics. Fluorescent light
ballasts are reactive and radiate powerful hum harmonics
A magnetic hum field and a microphone's hum pickup are
both directional. Thus, a magnetic hum field can be detected by
aiming a dynamic microphone in different directions while
monitoring its output and noting a change in the hum level.
The voltage in the power wiring can also be electrostatically
coupled to the microphone wiring; that is, a power wire and the
microphone wire act as two plates of a capacitor. As before, the
AC voltage induced in the microphone is heard as hum. FIGURE
2 is a schematic diagram of electrostatic hum coupling. With
electrostatic hum fields, the high -frequency components are
transmitted more easily through the capacitive reactance
between the power wires and microphone wires.
The higher the microphone impedance. the greater the
electrostatically -induced hum voltage. Thus. high- impedance
microphones are more susceptible to electrostatic hum pickup
than otherwise identical low- impedance microphones. (For
more on hum pickup, see Shielding and Grounding Revisited in
the October 1980 db -Ed.)
Hum pickup may be a problem whenever a microphone is
used at some distance from a low-level sound source, since
much amplification is required to obtain an adequate signal
level. Any hum voltage induced in the microphone is also
greatly amplified, resulting in a signal with audible hum.
Unfortunately, several hum -reducing techniques are available.
Figure 2. Electrostatic hum coupling between
line and a microphone.
Bruce Bartlett is a senior development engineer with
Shure Bros.
Figure 3. Synthesized pop disturbance, approaching
and then striking the pop filter of a microphone.
Unlike sound waves, pop disturbances travel through
the air by the mass movement of the air molecules
themselves. A typical propagation velocity of the air mass
is 15 feet -per- second. However, air particles within
the turbulent mass may attain peak velocities of 60 to 90
feet -per- second. By comparison, a 100 Hz sound have
with a sound pressure level of 100 dB attains a peak
particle velocity of only 0.03 feet -per- second.
Wind is also a moving mass of turbulent air and affects
the microphone much like pop, except over a broader
frequency range, and over a greater area.
I. Magnetic shielding utilizes a magnetically- conductive
shield or case around the microphone cartridge, wiring
and other internal devices. The shield offers a highly
permeable path for the magnetic lines of force, conducting
them around and away from the hum -sensitive microphone
components. Grounding of this shield is unnecessary for
magnetic hum reduction.
2. Electrostatic shielding employs a grounded electrically conductive case or screen around the microphone cartridge,
wiring and cable conductors. The shield offers a lowresistance path to ground for electrostatically -induced
microphone is moved farther away, the intensity of the
disturbance diminishes, because the pop loses energy with
distance. Closer than three inches, the pop signal diminishes in
intensity and loses low- frequency components because less
turbulence is encountered and the disturbance acts over a
comparatively smaller area.
Twisted-pair wiring inside the microphone and its cable
enables the two conductors to occupy nearly the same
position in space on the average. As a result, nearly equal
magnetic hum voltages are induced in both conductors.
These voltages cancel out at the load resistance when the
balanced connection is used.
4. With balanced lines, nearly equal hum voltages are induced
in each side of the line. Consequently, there is little
differential hum voltage between the two amplifier input
terminals to amplify.
A humbucking coil is often wired in series close to the
dynamic microphone cartridge where it is exposed to nearly
the same magnetic field as the voice coil. Since it is connected
in opposite polarity to the voice coil, the humbucking
coil generates an induced voltage approximately equal
and opposite in polarity to that induced in the voice coil,
thus cancelling hum by about 20 dB. Design effort must
toward achieving adequate cancellation at all
frequencies. Humbucking construction of the microphone
transformer helps it reject hum as well.
be directed
Another source of noise associated with microphones is
"pop." When a person says words emphasizing "p," "b," or "t"
sounds, a turbulent puff of air is forced from the mouth. If a
microphone is placed within a few inches of the mouth, the puff
strikes the microphone diaphragm and violently vibrates it,
creating an electrical signal. Or, the puff may strike a
surrounding grille structure and generate acoustical noise that
is sensed by the microphone cartridge. The resultant explosive
breath noise signal, when reproduced by a loudspeaker, sounds
like a thump or little explosion called a "pop." FIGURE 3 shows a
synthesized pop disturbance made visible by smoke.
In general, microphones have been observed to be most
sensitive to pop at about three inches from the mouth. As the
To minimize the sensitivity of a microphone to pop and wind
disturbances, the disturbance must be weakened as much as
possible before reaching the diaphragm by some sort of filtering
structure or barrier. The filtering structure itself must generate
very little acoustical noise when a pop hits it. A fine cloth or an
open -cell foam- plastic screen in front of the cartridge
commonly meets these requirements. These materials have a
high resistance to high particle velocities (such as encountered
in pop or wind disturbances), but have a low resistance to low
particle velocities (such as encountered in sound waves). This
non -linear filtering action reduces the pot and wind
disturbances reaching the diaphrahm while pass.
The bigger the pop filter or windscreen, the better it works,
but there are aesthetic and practical size limitations. Air -space
size, foam thickness, and foam porosity are critical for optimum
results. The user can further improve pop rejection by placing a
foam screen over the existing microphone grille structure.
keeping an air space, if possible, between the foam screen and
the microphone cartridge. A typical example of a built -in pop
filter is a ball- shaped grille such as shown in FIGURE 3. An
external windscreen is shown in FIGURE 4.
An external windscreen.
Pop and wind noise can also be minimized by the use of omniis
approximately 15 dB less sensitive to pop and wind noise than a
similar -sized uni- directional microphone. The net force of low frequency sound waves acting on the diaphragm of a unidirectional microphone is about 15 dB less than that of an omni-
0dB -.001V
directional microphones. This type of microphone
directional microphone. To compensate for this lower
operational force, the diaphragm damping of a uni- directional
microphone is made fairly low to increase low frequency
electrical output. Unfortunately, this also increases pop and
wind sensitivity. Thus, an omni picks up much less pop than a
uni. Note also that bi- directional ribbon microphones are
extremely sensitive to pop and wind and, therefore, require very
effective windscreens.
Pop disturbances are emitted from the mouth within a
narrow conical angle. The user can take advantage of this fact
by placing the microphone out -of-the-way of the air puff at the
corner, above, or below the mouth. Note, however, that "t"
sounds are directed downward. The reader can experience these
effects by saying "p" and "t" sounds at his hand at various
The spectral amplitude of pop signals is strongest at low
frequencies (usually around 100 Hz) and decreases as frequency
increases. Thus, some pop reduction may be attained by rolling
off low frequencies on a mixing console when "p " sounds occur
in the vocalist's signal.
Microphones are often used in situations where they may
touch sources of mechanical vibration or shock. Some such
situations are: handling noises during hand -held use; stand
noises associated with floor, boom, or desk -mounted
placement; clothing noises due to lavalier use, and cable noises
occurring with all applications. Much of this shock excitation
covers a wide frequency spectrum.
How do mechanical vibrations cause an output from a
microphone? Consider a moving -coil microphone. It consists of
a diaphragm with an attached voice coil that is suspended in a
narrow circular gap in a magnetic structure (FIGURE 5). A
strong magnetic field exists in the gap. Relative motion between
the voice coil and the magnetic gap induces a voltage in the
voice coil, producing an electrical output from the microphone.
This relative motion can be caused in two ways: by sound waves
vibrating the diaphragm and voice coil with respect to the
magnetic structure, or by mechanical excitation vibrating the
magnetic structure with respect to the diaphragm and voice coil.
Due to the inertia of the diaphragm/ coil assembly, it tends to
remain motionless when the supporting magnetic structure is
suddenly vibrated by a shock. Consequently, there is relative
motion between the voice coil and magnetic structure, which
produces an electrical signal heard as a thump or scraping handle sound. Microphones are most sensitive to shock in the
direction of normal (axial) diaphragm motion.
Sectional view of
Vibration sensitivity of microphone cartridges.
A condenser or ribbon microphone has a diaphragm of much
lower moving mass than that of a moving coil microphone.
Thus, the condenser or ribbon microphone is inherently less
sensitive to structure -borne noise than a moving coil
microphone of similar size and directional characteristics.
One goal in designing a microphone is to reduce the
undesired output caused by mechanical vibration; that is, to
minimize the diaphragm motion due to a giver. mechanical
In a dynamic microphone, the diaphragm response to shock
is greatest at its resonance frequency (typically around 150 Hz
in a uni- directional microphone, or 500 to 1,000 Hz in an omnidirectional unit). The response to shock at resonance can be
reduced by increasing the diaphragm damping resistance. This
damping is partly provided by the internal friction of the
diaphragm itself and mainly by acoustical damping material
placed behind the diaphragm. There are limits to the amount of
damping that can be used, since damping also can affect
frequency response and directional characteristics.
As stated before, the diaphragm resonance of an omnidirectional microphone is much more damped than that of a
uni- directional microphone. Consequently, omni -directional
microphones are approximately 15 dB less sensitive to
mechanical vibration than uni-directional microphones of
comparable frequency response and transducing principle (see
A popular technique of vibration reduction uses a high compliance internal shock mount. In this method, an elastic
mounting is used between the cartridge and microphone case to
Figure 7. Graphical demonstration of the influence
of a shock mount on the vibrational sensitivity of a
dynamic microphone cartridge.
dynamic microphone.
i I
provides effective shock isolation in a small package. FIGURE 9
shows its performance. It has maximum compliance in the axial
direction where it is needed most.
Microphones and their associated preamplifiers produce
minute noise or "hiss" -like signals. The major source of noise
associated with dynamic microphones is the thermal noise of
the resistance part of the microphone impedance. The
8. A
high- compliance shock mount.
isolate the cartridge from vibration. The mass of the
microphone cartridge and the compliance of the elastic
suspension form a mechanical system with a certain resonance
frequency. Mechanical excitation at this resonance frequency
results in large motion of the microphone cartridge. The
resulting output from the cartridge will be greater in this case
than if there were no shock mount at all. However, at
frequencies above resonance, the excitation transmitted to the
cartridge becomes less as frequency increases, because the mass
reactance of the system increases with frequency. Thus, at some
frequency above resonance, the shock -mounted cartridge is
producing less output voltage than it would be if it were not
shock -mounted. In short, the shock -mount acts like a partially
damped low -pass filter to mechanical vibrations transmitted
from the microphone case to the cartridge (FIGURE 7).
By making the shock mount sufficiently compliant, the
shock -mount resonance frequency can be placed below the
lowest acoustical frequency of importance for the microphone.
There are limits to making the shock mount too loose or
springy: the cartridge may "bottom out" when it vibrates,
especially since microphone size usually is limited. Maximum
compliance should be in the axial direction because that is the
direction in which the cartridge is most sensitive to shock.
For a further improvement in vibration isolation, the entire
microphone can be suspended in a compliant shock mount
placed on a microphone stand. "Rubber- band" type shock
mounts can provide sufficient compliance, but are generally
large and obtrusive. A device such as the one shown in FIGURE 8
Figure 9. Comparison of microphone performance
with various types of shock mounting. (A) Rigid mount.
(B) "Rubber band" mount. (C) High- compliance
isolation mount.
subjective noise level of these microphone types decreases as the
microphone sensitivity increases for a given impedance level.
Professional -quality condenser microphones generally have
a lower subjective noise level than dynamics. With condenser
microphones. noise is produced mainly by the impedanceconversion circuitry within the microphone itself. Some
microphones are quieter than others, depending on the active to- stray- capacitance ratio, the cartridge output level and the
input resistance of the impedance- conversion circuitry.
The following is a summary of recommendations for
reducing undesirable microphone noises:
For minimum hum:
Use condenser microphones with adequate shielding or
dynamic microphones with humbucking coils.
For microphones without humbucking coils, use those with
metal cases that provide adequate shielding.
Avoid SCR dimmers and fluorescent lights.
Orient the microphone for minimum hum pickup.
Use balanced lines.
Use low- impedance microphones.
For minimum pop:
Choose omni -directional over directional microphones,
if feedback and background sounds are not severe.
Use microphones with built -in filters.
Place a foam windscreen over the microphone grille, keeping
an air space between the foam screen and the microphone
Use the microphone closer or further than three inches
from the mouth.
Place the microphone out of the path of pop travel.
For minimum pickup of mechanical vibration:
Choose omni -directional over directional microphones, if
feedback and background sounds are not severe.
Generally, choose condenser microphones over ribbons, and
ribbons over moving -coil microphones. Note, however, that
some shock- mounted moving coil microphones are comparable
in vibration isolation to condenser and ribbon microphones,
depending on the effectiveness of the shock mount design.
Place microphones in shock mounts on stands.
For minimum self- noise:
Choose high -sensitivity dynamic condenser microphones
designed for low -noise performance.
AT 0.2
The author wishes to thank R. Anderson, D. Arnold,
W. Bevan, T. Locke, G. Plice and R. Schulein for their con-
tributions to this article.
I. L. J. Anderson. "Sensitivity of Microphones to Stray Magnetic
Fields." Transactions of the IRE-PGA. pp. 1-6, January February
2. Robert B. Schulein, Charles E. Seeler. and
I3K 23K
William R. Beven,
"Design of a Studio -Quality. Condenser Microphone Using Electret
Technology." Journal of the Audio Engineering So(ielI. Vol. 26.
No. 12. pp. 947 -957. December 1978.
3. Terry R. Locke, "An Effective Mechanopneumatic Shock Mount
for a Dynamic Microphone." Journal of the Audio Engineering
Society. Vol. 26. No. 9. pp. 623 -628, September 1978.
4. Gerald W. Plice. "Microphone Accessory Shock Mount for
Stand or Boom Use." Journal of the Audio Engineering Society,
Vol. 19. No. 2. pp. 131 -137. February 1971.
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few suggestions for miking pianos and other instruments
with strings, excerpted from the author's forthcoming
microphone textbook.
HE MERE MENTION Of Studio microphone techniques
evokes a wide range of responses from recording and
broadcast engineers. Many have evolved their own
unique techniques and problem -solving procedures
through years of "cut-and -try" experiments, while others have
spent their early years assisting and studying more experienced
Here, we'll offer some "textbook" solutions to common
studio problems, with much of the advice coming as a direct
application of microphone basic theory. There is always room
for other approaches, and the enterprising engineer will keep a
good eye and ear out for what others are doing, and for the
many techniques which are being developed in studios
No other instrument poses quite as many pickup problems as
does the piano. The sound radiation pattern is complex, and the
spectral content may vary considerably with even the slightest
re- positioning of the microphone(s). For example, consider the
brief musical passage shown in FIGURE I. Several single microphone placements are also shown, along with the one third octave peak -hold spectrum for each position.
The studio trying to make do with anything less than a six foot grand is making a mistake. Better yet are the seven -foot
grands -and of course, a nine -foot grand is pure luxury! The
bass registers of smaller instruments tend to produce a wooden
sound, no matter how well -regulated they may be.
Conventional wisdom states that pianos improve with age.
However, this is not true. A well -built instrument reaches its
prime in a comparatively short time, after which it begins a slow
John M. Eargle is vice- president
at JBL. Northridge. California.
of Product
period of deterioration. The fact that many newer pianos are
not built as well as their predecessors may suggest that the
instrument will probably "mature" over the years. But this
never happens. A well -regulated new instrument will exhibit
even voicing, and just as important, mechanical noises due to
the action and dampers will be minimal.
Some suggested microphone positions are shown in FIGURE 2.
The stereo placements are intended for pickup of the solo
instrument, and will produce a broad spectrum of sound. A
single microphone will generally produce the best mono pickup
if it can be located at some distance from the instrument.
Alternatively, the stereo pairs may be combined for mono. In
doing this, note carefully any apparent cancellations in the
spectrum. An evenly -played chromatic spectrum through the
mid -range will usually identify this.
It should never be necessary to place a microphone under the
instrument. Not only is the sound extremely dull, but the
thumping noise of the damper action may become obvious.
When the piano is to be picked up in a crowded studio, where
the sound level from other instruments is quite high, the
techniques shown in FIGURE 3 may be the only ones that will
work. A microphone placed in any one of the holes in the cast
frame will produce a distinctly unnatural sound, but one which
may be quite workable
even preferable
pop or rock
recording. Omni -directional microphones will be best. If the
sound is too "mid-rangy," try adding another microphone at
one end of the holes toward the tail of the instrument. If more
high end is desired, a microphone may be added above the metal
frame close to the top 11/2 octaves of strings. With microphone
one doing most of the work, microphones two and three should
be carefully mixed in, to provide the desired spectral correction.
This is, of course, a mono pickup, with the outputs of all three
microphones combined.
If even more isolation from loud instruments is needed, a
heavy blanket may be draped over the piano lid, with the lid at
"half- mast."
.- - -
15C cm
Figure 1. Variation of one -third octave peak -hold spectra
for various microphone positions in piano pickup.
(A) Top and front views, showing four microphone placements. (B) The brief musical passage produces different
spectra at each microphone position.
The techniques described above should result in a sound well isolated from anything else in the studio. It will be quite dead,
and accordingly may benefit from a slight touch of artificial
reverberation. In -line equalizers, especially if they are capable
of band- reject action, may help to tone down the brightness of
the sound somewhat.
Another important observation of this kind of pickup is the
effect of normal dynamic range changes in the playing of the
instrument. The piano is capable of an extremely wide dynamic
range, and loud passages will always be brighter than softer
ones. They may. in fact, become too bright for such close
pickup. In that case, the performer should be requested to
contain the dynamics within some allowable range determined
experimentally. Any additional dynamic control can always be
provided by the engineer and producer in the control room.
Never wait until the time of the session setup to determine the
pickup of the piano. It is, after all, one instrument that is always
in the studio, or will have been brought in well ahead of the
session. Thus, there should be ample time for experimenting.
Do not be afraid to tell a pianist to alter playing style, as may be
required by the needs of the session. Some players are noisy
pedalers. and close microphone placement may pick up too
much thumping noise from the dampers. A slight change in
technique on the part of the player will usually fix this.
Figure 2. Some suggested microphone placements
for normal stereo pickups.
A\. .
I or classical piano, the
normal studio environment will not
satisfactory. There are only a handful of really good studios
around the world suited for this purpose. The best recording
venues are medium -sized live recital halls, or old- fashioned
ballrooms. What is most desireable is a room whose
reverberation characteristics emphasize the mid- and high frequency portions of the spectrum. Many halls which are
excellent for orchestral recording may simply be too big for
piano recording.
A fairly high level of reverberation may be perfectly
satisfactory, if it is not too long. Like speech. the piano needs to
be well- articulated, and a too-long reverberation time will blur
musical detail. It should not exceed about 1.5 seconds. Some
details of microphone setup are shown in FIGURE 4.
Many piano recordings suffer from poor focus or imaging in
stereo playback. A coincident or quasi- coincident approach
will cure this. If a spaced -apart approach is favored. there
should always be a center microphone to stabilize the image.
Establishing the fore -aft position of microphones is critical.
It is very instructive to begin close in and move the microphones
outward in small steps. perhaps no more than about 30
centimeters at a time. There is often a fairly- narrow range.
depending on the room and the instrument, where all musical
details seem to fall into place. It is well worth taking the time to
find the choice location for the microphone. Especially in
classical recording, one cannot stress the importance of a fine
instrument enough. Nothing less than a nine -foot instrument
should he used, and it should he in rune.
co'IVC.E:.T o.VRI
Two other keyboard instruments are notorious for their
noisy actions: the harpsichord and the celesta. These
instruments are encountered less and less in pop recording.
since their specific roles have been pretty much filled by various
electronic keyboard instruments.
The harpsichord has the noisiest action of just about any
instrument. When a note is released, the plectrum contacts the
still -vibrating string before the damper can mute it. In the mid
and upper portions of the keyboard, this may not be a problem.
but the lower keyboard suffers from various "clunks" and
"thunks" of the action. At close quarters, one is aware all the
more of the action noise, as well as a high -end "sizzle." Clearly,
the harpsi hord is an instrument meant for a specific
environment: it works best in intimate halls with reverberation
times up to say, two seconds. Such environments tend to downplay the raucous high end considerably.
The engineer who is faced with recording a harpsichord as
part of a typical pop session has few options. The instrument
produces only a moderate level, and is inherently of limited
dynamic range. A single omni -directional microphone located
close to the raised lid, about one -third down the instrument,
will pick up a good balance with minimum action noise. As a
rule, its use in pop recording will be relegated to passages where
it is exposed and not forced to compete with louder musical
3. A
Recording the piano in
concert hall.
Bowed instruments possess a fairly -wide dynamic range, but
their overall output capabilities are limited. This. of course. is
why they are used in large groups. In most pop recording. the
string choir, consisting of violins, violas. cellos and string
basses, is used for adding warmth. or for "sweetening" the
musical texture in the form of a counter melody or other secondary musical role. Because of this. they are often Dyer-dubbed
at a later date. when the studio is free for them alone. A single
stringed instrument may be picked up monophonically if that
will satisfy the musical requirements. In this case. a single
microphone located about or 1.5 meters above the sounding
board will suffice. However. where an ensemble is involved.
other techniques will be required.
Even though the session budget may demand that you make
an attempt. it is difficult to make a handful of strings sound
like a large group. Microphones should not be too close. since
that will invariably produce a harsh sound. A natural stereo
spread to the string ensemble sound is essential. and multi mono pickups are usually disastrous. While coincident pickup
will produce an integral. well -spaced sound. the particular goals
of pop recording will probably be met best through the use of
spaced -apart microphones, with sufficient acoustical leakage
between them to add the required warmth. In fact, the spaced apart approach will make the group seem somewhat larger
than it actually is.
closeup mono pickup of the piano.
Tn. .IEw
Allow sufficient room for the string players. Normally. they'
are used in pairs. with each pair requiring about 1.5 meters on a
side. Microphones should be located overhead. at a distance
of about three meters. The number of microphones will depend on the size of the group. A typical group may be made up
of six first violins. six second violins. four violas. four cellos
and one or two basses. A group of this size may require up
to six or seven microphones. each panned across the stereo
array as required. An additional microphone may he needed
for the string basses to provide a firm bass line under the
control of the mixer and producer.
Artificial reverberation is usually a necessity in recording a
small group of strings in even a large studio. While some studios
can be adjusted to be fairly reflective acoustically, they are
usually too small to produce a significant reverberant field so
ÿa, , \
O \\
- -°10vÓ O,,i
Figure 5. Signal processing scheme for studio pickup
of a medium -sized string ensemble. (A) The floor
plan. (B) The signal flow diagram.
essential to good string sound. Today's better stereo reverberation generators, both digital and analog. are equal to the task.
A typical floor plan and signal flow diagram for recording a
string section is seen in Figure 5. While this multi- microphone
approach may worry some "purist" engineers, it does provide a
high degree of flexibility in string recording. Individual
microphone inputs are panned to their respective positions in
the stereo array. In Figure 5B, the AT units represent time delay modules. Those feeding off the stereo mix would be set in
the 30-to -50 millisecond range, where their effect would be to
simulate early reflections in a concert hall. The delay module
in series with the reverberation generator would be set in the
40 -to-70 millisecond range, in order to simulate the normal
delay in a concert hall before the onset of reverberation.
The string bass may require attention. It is not unusual for
studio orchestras to have only a single string bass, and in order
to pick up sufficient fundamental sound, a single microphone
placed on a low stand in front of the instrument will be needed.
Its output will have to be panned and balanced into the
stereo array carefully.
For jazz recording, the string bass takes on an entirely different role, and it needs to be highlighted as a virtuoso solo
instrument. It is almost always played pizzicato, or plucked, in
jazz work, and this demands that finger noises and rattling of
the strings on the finger board be picked up accurately, rather
than suppressed. A single microphone at a height of about one
meter. aimed at the front sounding board, at a distance of
about 0.5 meters, will insure a good pickup. Be sure to thin out
the bottom end a bit if the sound seems too muddy.
Another original from audio & design recording
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Distribution Amplifier
L.E.D. Quad Display Column
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Getting Down
to Two Tracks
Fast and Other Stories
A veteran engineer revives the live
mix... and lives to
tell about it.
TO ENGINEER Fred Catero, whose story
this actually is, I'd like to dedicate this article to all those
eager -to- please session people who have been really and -truly "fixed" in the mix: to all those engineers who
have played multi-track musical chairs with a philistine
producer: to all those producers who have faced the crack of
dawn with the realization that "It's garbage. What I've got
here is garbage," and to all those music lovers who bought
the album because they heard the group the night before, but
now can't discover any similarity whatever.
Who, after all, speaks up for these people? Who offers
consolation other than the perfunctory, "It'll be better next
time?" No one, really. If that is not the actual reason why the
album Black Pearl got made, it is at least the message many of
its makers hope and trust it will convey.
Black Pearl is an album that no one lost a night's sleep over,
because as soon as the band stopped playing, it was done. There
were no subsequent mix -downs, no sweetening sessions, no
after -the-fact overdubs, and no 4:00 AM erraticisms and coffee
nerves. There couldn't be. because the recording was made on
two tracks, and what can you do with a two -track tape, except
keep it or throw it out? All of which seems to have suited the
people involved with Black Pearl just fine. Everything was
nailed down solid, and everybody went home happy.
In the beginning, producer (and prominent jazz critic)
Conrad Silvert and featured soloist Herbie Hancock undertook
to do an LP for Victor Records of Japan with the I9 -piece
Tokyo Union Orchestra under T. Takahashi. In short order,
Richie Cole and Slide Hampton were also invited in, along with
their arrangements. The recording sessions. to take place at San
Francisco's Automatt, were scheduled to correspond with a
visit by the orchestra to the U.S.
It was Japan Victor's plan to turn the recording into
something of a technical tour -de- force. in order to increase its
commercial chances. Word came that they intended to send a
PCM recorder along with the orchestra - and, presumably, to
stamp "digital" on the record jacket, to arouse a little
excitement. Neither Silvert nor recording engineer Fred Catero
could see anything wrong with that, and there things stood
until Victor announced that they would not send the recorder
after all, but would oversee arrangements to rent one in the
States. That was fine too. A short time later they sent notice that
the rental idea was out, that the Automatt should proceed to
record the tape conventionally, and that perhaps it would be
converted to digital back in Japan. Something about this
scheme seemed to border on the illogical, to say the least.
Consultations were held to find some other technical hook to
hang the project on. The U.S. team came up with the notion of
doing it two track, start to finish: live mix, in other words. How
was that for revolutionary?
Well, at first it was a little too revolutionary for Japan. Could
such a thing be done? Catero was cooly confident. He pointed
out that there was a time when virtually all records were made
that way: that he had in fact participated in that time and could
Ralph Hodges is a freelance writer !editor living in
the San Francisco hay area.
Figure 1. Engineer Fred Catero prepares microphone
placement before recording the 19 -piece Tokyo Union
Orchestra at The Automatt in a "live" session that
was mixed and recorded direct to a 2 -track Dolbyized
tape recorder. Recorded in this fashion to capture
the interaction and spontaneity of the musicians, the
LP features Herbie Hancock, Richie Cole and Slide
Hampton with the orchestra performing songs and
arrangements by Hancock and Hampton.
still hold
at the end, it was the two -track tape that went gaily forth into
the world. Nobody ever mentioned the twenty -four -track backup, and Catero hopes nobody ever will. From here on, it is his
story, so I'll let him tell it in something like his own words,
complete with outbreaks of spleen, finger -wagging and control room philosophy. Wish you had been there to hear it in person.
I like to use (this is Catero speaking) RCA ribbons on
trumpets when I'm close miking, because of the mellow quality
they impart to really hot brass. When condensers are worked
close, such saw -tooth stuff comes out that the sound almost
seems to splatter at times. However, condensers are fine for
trombones and saxes: Neumann 87s, and I also like the Sony
C22s, especially on cymbals and drums. The Shure S M-56 is a
great workhorse that I use quite a bit on drums and electric
guitars, principally because of its tremendous headroom and
the sort of soft self-limiting action characteristics of its
diaphragm construction. But, in general, I don't play favorites.
If I need frequency response, almost any good condenser will
make me happy. For the rest, a handful of dynamics and not
even expensive ones; $150 or so is okay- should do the job.
For Black Pearl. we miked the orchestra in pairs, if not in
sections. I prefer this for the sake of the better blend. The only
people who got individual mikes were those being featured, or
who had to be ridden up and down, or who needed some delay
and rcverb. As a rule, I try to pick up the electronic instruments
directly. With jazz, the dynamic swings can have you down in
the grass at one moment and right through the roof the next. A
a job. Japan was still dubious. The inscrutable Catero
was reassuring; of course, there would be no problem in backing
up the session with a twenty -four tracker running simultaneously. That clinched it.
As everyone must know by now, jazz has a large and wildly
enthusiastic following in Japan. Without waxing overly
sociological, I would suggest that it is a valued counterbalance
certain devotion to formal orderliness that characterizes
Japanese lives and affairs. Comparatively, jazz is spontaneous,
chance to relax and, every
unstructured, let -it -all-hang -out
now and again, to get riotously and even sloppily drunk. So
naturally, when you invite a Japanese jazz orchestra to come
play in your studio, its members turn up scrubbed, sober.
utterly serious, studious and studied, and on time. Catero
couldn't believe it when, at ten minutes to the session, he
glanced out the control -room window to see everyone in place
and tuning up, with the obvious intention of starting on the dot.
He thought his watch had stopped and that he had lost at least a
half hour somewhere along the way. This was not the way
professional session musicians are supposed to behave.
It was a foretaste of things to come. Did the orchestra know
the charts? It did. Was it prepared to play, and play well, even for
whole minutes at a stretch? Of course. Was it intending to tie
things up in union regulations and grave deliberations over
coffee breaks and who pays for the new saxophone reeds? Never
entered their minds.
When I spoke with Catero some weeks after the session, he
was still high from the experience. It had been three and a half
days of near- hitchless wonder and mutual congratulations, and
Figure 2. In The Automatt's Studio A control room are
(left to right) Tokyo Union Orchestra leader Tatsuya
Takahashi, session producer Conrad Silvert, Herbie
Hancock, and Fred Catero, along with members
of the Tokyo Union Orchestra.
direct feed makes life a lot easier under these conditions, even if
the amp doesn't hum and the speaker doesn't break up. I never
have any trouble getting a guitar or bass to sound the way l want
through processing on the spot or after. If need be. I'll lay down
the track via a direct feed during the session, and then play the
track back through speakers later. But for this session. I broke
my own rule (which is what rules are for, if you can see a better,
more logical way) and miked the guitars. It just seemed to
provide a more live, more realistic sound in the context of what
we were doing. But the bass and the Fender R hodes were direct.
The Rhodes is a case in point. It has stereo outputs, and with a
direct feed the separation tends to be much more distinct than
with an acoustic pickup.
Hancock's piano was miked very close: mikes on the high and
Figure 3. Herbie Hancock, who wrote and arranged the
material and is a feature performer on Side A of the
album, conducts the orchestra.
low strings, and a third underneath, with its phase reversed.
wanted to be able to keep everything the man did. so I went all out for minimum leakage from the rest of the band. In general.
Ranted leakage- depended on it, in fact. Studio A is a large
space (1.500 square feet) of the sort I tend to call an old fashioned room: hardwood floors, hard walls with absorption
here and there, and so on. M usic sounds good in it. and I can use
this fact, make it a real contribution. just as I can use some inter mike leakage to create a blend and a sense of spatial continuity.
Hence everything was left out in the open: no gobos, no screens
of any kind, no isolated drum room. The soloists worked the
middle of the room, with the orchestra facing them along the
front and rear walls. Here, we did have a bit of a problem with
the horns. They were up front, facing the drums way in the back.
and they couldn't hear them too well, nor could they deal with
the slight but significant time delay. As sharp as they were. they
couldn't anticipate the beat in a way crucial to some of the
syncopated stuff. Finally. I set up a speaker beside them and fed
it just the snare and the kick, and that was enough.
I use two processors faithfully and usually only two:
delay and something for reverberation
plate or a chamber of
some kind. You need both -that initial single delay
representing the first reflection and then the drawn -out
reverberant tail
you're going to locate an instrument in a mix
and give it a "somewhere," which is what I aim to do. When and
how much I process depends on where the instrument is
supposed to be. If it's up front, it shouldn't need anything. If I
want it in a corner in back, it gets an early initial delay and quite
a bit of reverb. if in mid -room. maybe just a touch of reverb and
a later initial delay, because there should be no reflecting
surfaces nearby.
Earlier, we were talking about the sense of depth in a
recording, which is something I completely believe in. First
reflection and subsequent reverb are key cues for this sense. Of
course, there are others: general character of the reverb, spectral
No way could the sound of Black Pearl have been gotten in the
semi -anechoic sound-sink that passes for a studio today. Those
places eat musical sound. devour its character. and few people
seem to understand how much needs to be put hack into make it
whole again. The engineer adds a little echo and says that's it.
But there's really much more.
Naturally, I have used and use -stereo and binaural miking
all the time; it's uniquely valuable for the realistic depiction of
an experience. like a night in a jazz club or at the opera.
Howe\er, for the ideal representation of music in a plausible
space for a commercial recording. I find it completely
satisfactory only when I can locate one or two places where
can stand. in relation to the performance, and hear everything
exactly as I want to hear it. This is rare. With multi- miking
you're not so constrained, and you have a chance to patch up
those little anomalies you will hear even in the presence of live
music. I definitely do not favor the wholesaleglamourization of
music or the deliberate distorting of a musical experience- unless the composer's intent is surrealism, of course. But
appreciate the opportunity to make slight improvements here
and there, and to eliminate the niggling things that really
needn't be there.
Finally, multi -miking is expected these days. No longer is a
producer willing to interrupt a good take to move the trumpet
closer to the mike, and perhaps away from the spot where he can
hear the other players he depends on for cues. For better or
worse, it's now solely the engineer's business to solve these
problems of balance, and to create a balance that would
perhaps. be impossible in real life, because you just couldn't
position the musicians in that particular was and expect them to
play properly.
Tatsuya Takahashi of the Tokyo Union Orchestra.
So few engineers are capable of doing a live mix these days.
Or even if they could do one, they think they can't because they
characteristic of the entire sound, and even -but not always- loudness. But the first two are of paramount importance. Of
course. you can achieve them without signal processing per se.
If you want to make someone recede. turn down his mike and
turn up the ones he's leaking into. It accomplishes the sane
basic thing.
I love good acoustic chambers. The problem is that many are
not good. and sound tubby or hollow. Another problem is that
most have a fixed characteristic. I can't alter the initial delay
unless I physically walk into the chamber and move the speaker
or the mike, and who has time for that? So I depend mostly on
the black boxes. Which reminds me; I have a new toy, an EMT
with FRAI's on the plate. Their pickup seems more linear to
me. and better on the highs, especially when I'm hitting the plate
hard. By now, l'ye had enough experience with it to say I'm
"knocked out." (For more on FRAPs. see The FRAP PointSource Alicrophotte in the December. 1979 db -Ed.)
Incidentally, Black Pearl was not engineered for any
remarkble sense of depth. This is not because it was a real -time
mix, but because the people didn't want that. They preferred a
refined studio with a plausible ambiance, but everything right
there, up Iront, and alive without going to excess, of course.
The drums, for example, are more forward and certainly wider
than you'd have perceived them had you been in the studio. This
is deliberate: call it a bit of poetic license. if you will. It's a slight
enhancement of reality. but it doesn't overthrow credibility.
There is no stereo miking to speak of in Black Pearl: instead.
there is leakage. I'm appalled by all the multi -track mono I hear.
with a guy here, here, here, and there, and dead, empty
nothingness in between. with no sense of environment, even if
it's supprc.+ed to he a contrived environment. Used properly.
leakage is powerful. It keeps the instruments pinpointed, but it
also tills the areas in between and makes the space glow and live.
Learning how to control leakage is a matter of learning how to
use your mikes (which we'll get to in a moment) and the room.
The Crown Real Time Analyzer (RTA -2) can help
you stay ahead of the competition. It will show you
what's wrong with frequency response in studios,
control rooms, circuits or equipment. You'll know
exactly where to start to improve signal quality.
The RTA -2 is yours free,
for thirty days.
It's easy to use, rugged and self- contained. 60dB
dynamic range. Five -inch CRT. Complete with highquality microphone. If it doesn't help, send it back.
No obligation.
Call Dennis Badke at 219/294 -5571 for the details.
1718 W. Mishawaka Rd.,
Elkhart, IN 46517.
Innovation. High technology American. That's Crown.
Circle 25 on Reader Service Card
Members of the Tokyo Union Orchestra.
never have, and that's just as fatal. The only mix they know is
the one in which they sit around experimenting, fooling with the
bass and drums for a while, trying to add a guitar (maybe it
works, maybe it doesn't), perhaps starting all over again,
listening to the kick alone for a while, ad infinitum. Where does
this lead except to becoming a slave not to the music, which
would be fine. but to the technology, which isn't designed for a
slave and shouldn't have one? And what effect does all this have
on the musicians? Could they go on the road with that mess of
mistakes they bring to the studio to record? Certainly not; but
they've learned it needn't be any better for the recording session
because they know that someone is going to sit down and
expend hours and agony fixing it. But is this music they're
making? I say no, and I say that when you hear it you can tell it's
The glory of Black Pearl is that I didn't have to spend any
time making the musicians look good when they weren't, and so
I could get on with my work. And because I could get on with
my work, it was done before it had a chance to get in the way of
their music. There is nothing special about a live mix, in my
opinion. If the band can play the music, and if it sounds at least
acceptable in the studio, then that is a natural way of recording
it. Of course, the engineer has to focus his full attention on the
project at every moment, and he has to have a reasonable feeling
for what to do in a tight spot -which can sometimes mean, do
nothing and go with it. He cannot let himself get carried away to
the point where he doesn't know where he is and what he's
supposed to be doing, and he shouldn't let himself get blown off,
and out, by 115 dB levels. (Some people don't realize you can't
balance at such levels; the ear acts as a natural limiter and
everything sounds balanced.) But with the right attitude and a
modicum of skills on the part of everyone, a live mix should
glide through takes, playbacks, and approvals like a zephyr
through a willow grove. And then everyone can go home and
get a good night's sleep.
about was where to put the mikes.
I see engineers miking amps by putting the mike an inch away
from the center of the speaker. This is fine, if you realize what
you're going to get, which is the sound of an area of speaker
cone about as big as a silver dollar. Meantime, there are things
happening 10, 20, even 50 feet out from the speaker that you
don't have a chance of getting. Same thing with a tom -tom. You
can space a mike an inch away from it and you'll get a "sound,"
but it won't be the sound of that tom -tom. because there is a lot
more to the instrument than a square inch or two of drum head.
I don't hesitate to use close touch -up mikes on a drum set when
they seem called for. but I start with an overhead pair to get
an idea of what the whole thing sounds like. A good trick I
learned many years ago: If you want to record something but
don't know where to put the mike, block one ear and stick the
other where you think you might like to put that mike. You'll
know soon enough whether you want that mike there or not.
I am similarly pragmatic about the environments in which I
set up microphones. I have miked a guitar in a bathroom for the
(to me) excellent reason that everyone loved the way it sounded
in there, including myself. It's all a matter of what it takes. Don't
ever forget the easy way out.
Ultimately, I think too many people who use microphones do
not bother to learn about them and what they do. They know
that a mike has X frequency response and Y dynamic range and
whatever distortion. But they can invariably find another
microphone that has the identical specs and costs $400 less.
Why? I've discovered that, like everything else in this business.
some mikes stick to specifications very rigidly. As for others. the
specs mean that if you're lucky enough to buy a good one, this is
what you get.
What I've always done is to take advantage of the things the
better microphones can do. The pickup pattern is very
important. I've gone up to drummers between takes and moved
a mike a tiny bit, perhaps %4 -inch, and they've looked at me like I
had gone crazy.
"What are you doing? What did you just come out and do?"
I've never been one to get hung up on the microphone -the
mike without which I refuse to do a session, the mike that is the
only one that can get the snare right. The issue has no meaning
for me. A mike's a tool, not a totem or a voodoo doll. It can be
used successfully with understanding of its specific nature, and
perhaps unsuccessfully without.
Mostly for my own amusement, I went to the last Berkeley
Jazz Festival and miked Santana from the audience. For mikes
1 used what was handiest, which happened to be the ones built
into a JVC binaural headset. A few days later, one of the group's
managers stopped by. I played him the tape and he was stunned.
"Why can't my remotes get that sound? ", he wanted to know.
Probably because the recording team is always thinking about
the truck, the gear, the schedule, and all the rest. All I thought
just moved it a little bit."
"What's that gonna do?"
"Look, this microphone cost $600, and there's a reason. Just
one degree off axis gives me better separation between this tom
and the other, and I'm using that." I know some engineers who
figure that as long as the mike is somewhere in the
neighborhood of the device it's supposed to pick up, it'll do it,
with no consideration of pattern, of dynamic range, of
anything. But it's more important to know what a microphone
will actually do, than to know that a Neumann is good on this, a
Sennheiser is good on that. or a dynamic is good on drums. Yes.
there's nothing more important to this business than a mike, but
at today's state of technology. and at today's prices, it's hard to
find a microphone that won't do the job, if you know how to
give it a chance.
Application Notes
\(K recording
sessions. the outputss
microphones always manage to get
combined either on- session. or later. during the
mixdown. the combinations are for the obvious
purpose of getting a better balance than would be possible
from a single microphone.
What may not be so obvious are some of the polar patterns
that may result when the outputs of closely-spaced
microphones are combined. With a little planning, a two -mic
combination may yield just the right pattern. However, with a
little lack of planning. the combination may produce some
unexplained results.
To point out just how combining outputs may produce
unexpected results. take a look at the well -known but often
misunderstood M -S microphone system. M -S stands for
Middle -Side. in which one uni- directional microphone points
straight ahead at the middle of the group while the other
microphone is bi- directional: placed as close to the M mic as
possible. but with one of its dead sides aimed at the middle of
the group. In other words. neither live end of the microphone is
pointing in the direction of the music an arrangement that
may not seem too promising to those not familiar with M -S
fo understand M -S, it is important to remember that the
output polarity from an audio signal reaching the rear lobe of
any microphone will be reversed. as compared to the same
signal reaching the microphone's front lobe. On super- and
hyper -cardioid studio microphones, this polarity reversal is
probably of no practical consequence. However. when the rear
lobe reaches the same site as the front lobe -as on a fully bidirectional microphone then the polarity reversal must
definitely he taken into account. In fact, good set -up practice
suggests that the rear lobe of a bi- directional mic be kept at a
safe distance from the front lobe of uni- (or other bi -)
directional microphones. to prevent unwanted cancellation
of at least a
But what about wanted cancellation effects? When
side -
oriented, bi- directional pattern is intentionally combined with a
middle -oriented uni -. one of two resultant patterns may occur.
Assuming the front of the bi- is aimed to the right side. that lobe
will reinforce the right -hand side of the M microphone. while
the rear lobe will subtract from the left -hand side of the M mic.
The result is a uni- directional pickup pattern, oriented about
midway between the M and the S: in other words, a cardioid
pattern, angled toward the right -hand side of the group.
However, if the polarity of the S microphone is reversed
before combining it with the M microphone, the opposite
orientation occurs: we now have a cardioid pattern angled at the
left -hand side of the group.
Of course, with a suitable matrix network, we can achieve
left- and right- oriented cardioid outputs at the same time. Why
then not simply use two suitably- positioned cardioid
microphones in the first place, and never mind the complexities
of the M -S matrix?
The M -S technique provides a measure of control that may
not be possible with two cardioids. With the M microphone
providing a single mono pickup, the output of the S mic may be
adjusted to provide just the right amount of stereo imaging. If
desired, the M output can also function as a separate mono
feed, with no phase cancellation problems, simply because this
mono output is not derived from more than one microphone in
the first place.
Later on, if a mono mix is required from the left and right
M -S derived outputs. there's still no possibility of phase
cancellation. since the two S components simply cancel each
other out, leaving just the original M output.
Of course. if the M and S outputs are not matrixed before
recording on tape. then the two -channel tape may be used later
on for either mono or stereo playback: mono. by playing back
the M track only, and stereo by combining the M and S tracks
during playback. The technique can also be useful if the M -S
pickup is part of a multi -track recording say. a string overdub.
With the M track panned center. the S output can be adjusted to provide the right amount of stereo spread to suit
as before
the mixdown. Therefore. you're not locked into the stereo
image that sounded right during the overdub.
Since the M -S technique gives us two cardioid patterns which
are derived from one uni- (M) and one bi- (S) directional
microphone. it follows that an M -S pickup can also be derived
from two cardioid microphones. With the microphones angled
at + and 45 degrees, combining their outputs provides a single
forward- oriented (M, that is) cardioid output. Reversing the
polarity of one of the microphones before combining them will
give a side- oriented (S) bi- directional pattern. For M -S
applications. this may be more trouble than its worth. However.
it serves as a reminder of what may happen if an unintentional
polarity reversal sneaks into any combination of microphone
For example. two cardioid microphones may be angled over
the top of the drum set. with each microphone feeding a
separate track. When the microphones (that is. the tracks) are
combined later on, an unintentional polarity reversal may
demolish the drum sound. especially if anything important
happened to be located dead -center. On the other hand. the
reversal may be just what's needed to attenuate some leakage
from other nearby instruments. In either case. the polarity
reversal makes a figure -8 microphone from two cardioids. Just
make sure it isn't happening unintentionally, later on.
For the mathematically -inclined, the formulas for various
polar patterns are given here:
A = 1.00 + 0.00 cos a omni- directional
.5 + .5 cos 0 uni- (cardioid)
.37 + .63 cos 0 uni- (super-cardioid)
.25 + .75 cos o uni- (hyper-cardioid)
+ 0.0 + 1.00 cos d bi- (figure -8)
For example, what's the relative output of a hyper -cardioid
microphone, at an angle of 145 degrees?
A = .25 + .75 cos(I45°)
_ .25 + .75( -.819)
From the five polar pattern formulas given above, note that
the output of an omni -directional microphone is -as
expected- always the same (1). and therefore, the output level
is always 0 dB down. Also, the output of the bi- directional
microphone is simply, cos0, which is why it is sometimes
referred to as a "cosine microphone."
Keep in mind that these formulas describe theoretically -ideal
microphones. In real -time life, there's no such thing as an angle
at which a microphone is totally dead. About the closest thing
to it is the side of a good bi- directional microphone- -a side
worth remembering when trying to keep leakage at a minimum.
Just make sure it doesn't happen by accident!
AKG Acoustics
77 Seileck Street
1145 65th Street
Stamford. Connecticut 06902
Oakland, California 94608
(203) 348-2121
(415) 652-2411
Audio -Technica U.S., Inc.
c/o Gotham Audio Corp.
741 Washington Street
New York, New York 10014
33 Shiawassee Avenue
Fairlawn. Ohio 44313
(216) 836-0246
Beyer. Dynamic, Inc.
5 -05 Burns Avenue
Hicksville, New York 11801
(516) 935 -8000
(212) 741-7411
Bruel & Kjaer Instruments. Inc.
185 Forest Street
Marlborough, Massachusetts 01752
(201) 348-7000
(617) 481-7000
Calrec Audio
c ¡o Edcor (see listing below)
2913 Governors Drive
Huntsville, Alabama 35805
(205) 533 -9232
Cetec Vega
P.O. Box 5348
El Monte, California 91731
(213) 442 -0782
PZM Microphones
Manufactured by Crown (see listing above)
Schoeps Microphones
c/o Posthorn Recordings
Crown International
1718 W. Mishawaka Road
Elkhart, Indiana 46517
142 W. 26th Street
(219) 294 -5571
16782 Hale Avenue
Irvine, California 92714
(714) 556 -2740 & (800) 854 -0259
Electro-Voice, Inc.
600 Cecil Street
Buchanan, Michigan 49107
(616) 695 -6831
HME, Inc.
Fairmount Avenue
San Diego. California 92120
(714) 280-6050
JVC (US JVC Corp.)
41 Slater Drive
Elmwood Park, New Jersey 07407
(201) 794-3900
Cara International Ltd.
P.O Box 9339
Marina Del Ray, California 90291
(213) 821 -7898
c o
Panasonic Professional Audio Division
50 Meadowlands Parkway
Secaucus, New Jersey 07094
Piezo Co., Ltd.
c/o Hutco, Inc.
New York, New York 10001
Sennheiser Electronic Corp.
10 West 37th Street
New York, New York 10018
(212) 239 -0190
Shure Brothers, Inc.
Hartrey Avenue
Evanston, Illinois 60204
(312) 866 -2200
Sony Industries
9 West 57th Street
New York, New York 10019
(212) 371 -5800
Swintek Enterprises, Inc.
1180 Aster Avenue, Unit J
Sunnyvale, California 94086
(408) 249 -5646
Telex Communications, Inc.
9600 Aldrich Avenue South
Minneapolis, Minnesota 55420
(612) 884 -4051
Sound Reinforcement and
Broadcast Audio at the
Republican National
Presenting the trials and tribulations of providing audio
.for the three ring circus known as the political convention.
HAS BEEN SAID that political conventions are spaced
so that broadcasters and sound reinforce-
four years apart
ment personnel can get a chance to catch their breath
before the next one begins. Last summer -the four -year
cycle completed - it was time for the Republicans to come to
With the Republicans came the media: the stars and the
super -stars of newscasting. They were there, of course, to see to
it that we saw and heard all the events and all the speeches. They
were also there to interrupt those speeches (which they knew we
would not find interesting), in order to inform us what the
speaker was saying, or more often, what he was not saying.
Along with the media came a virtual city of engineers and
technicians, with sufficient equipment to assemble the complete
radio and television studios which would be their home -awayfrom-home throughout the convention.
The responsibility for providing audio to broadcasters, and
sound reinforcement in the Joe Louis Arena, was in the hands
of two teams. With producer Jack Kelly in charge, CBS
provided broadcast audio and video to the other broadcasters.
Video control was centered outside, in a trailer, from which
Jack exercised control. Because of the magnitude of the
production, only a small amount of audio gear was out in the
trailer. Most of the equipment was in the control center, a
windowless room directly beneath the speaker's podium.
Figure 1. The presidential podium at the Republican
National Convention, miked with Electro -Voice RE18s.
Greg Si/shy is market development manager.
professional markets, at Electro- Voice. Inc.. Buchanan,
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2. The
pool audio system.
Referred to as the "inside pool," this was capably controlled by
operations engineer Jim Hargreaves, of CBS Radio and
Television. Washington, DC. Jim's crew was made up of CBS
technicians from Washington. Chicago, St. Louis and Los
Angeles. In addition, he had the assistance of audio services
technician Tim Kerr from Filmways, the company from whom
the audio -pool equipment had been leased. I was on hand to
assist the pool with microphone usage.
Sound reinforcement for the house was provided through the
combined efforts of two consulting firms: Coffeen, Anderson
and Associates, and KLA Laboratories, Inc.. a Dearborn -based
The technical requirements for broadcast audio were spelled
out by CBS and ABC, and then put out for bidding. The
equipment package was leased from Filmways, and later on
ABC used it to handle the Democratic convention in New York
I asked Jim Hargreaves if he had a specific design philosophy
regarding the system. "There was complete redundancy.
Obviously, each of the components -from a systems
standpoint -had to meet broadcast standards, which are
somewhat critical, and they did. But along with that was the
redundancy, in the event anything failed. The microphone
switcher for the floor (you're dealing with 56 delegations) was
completely redundant. It was a 60 -by-2 switcher. Should
anything in either one of the banks fail, the other one was there
as a backup. All of the microphone pre-amps were redundant; it
was a double -entry system. And in the distribution to the
customers -that is, to the broadcasters -there was also the
availability of redundant signals.
For miking the delegates, Hargreaves wanted microphones
with low handling noise, absence of proximity effects, wide
pick -up pattern, and durability. According to Hargreaves, these
mics (Electro-Voice D056 omnis) were the only components in
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floor delegate microphone switcher system.
the system which were not doubled. Each microphone was
secured to its custom stand by an EV 342 security mount, to
avoid easy "rip -offs." They had been modified by removing
their on -off switches. The treatment they received confirmed
our worst fears, and convinced us we had made the proper
selection of both microphone and mounting method.
On the podium, there was 100 percent left- and -right
redundancy, using RE18 Variable -D® cardioid microphones.
An active microphone was mounted on each side of the podium
to supply the broadcast feeds. Star iby mikes were mounted
alongside the actives, to provide a backup system. Fortunately,
the alternate mics were never required. An additional R E 18 was
mounted in front of the podium, to supply the house feed.
The podium itself had a very "drummy" sound. Because of
this, each microphone was mounted in an EV 313A shock
mount. The combination of the shock mount and the mic's own
internal shock mount resulted in a staggered- resonance system
which eliminated all mechanically -induced noise.
The orchestra used a combination of EV RE20, REIO, RE15
and DS35 mies. Guest performers appearing during the
convention. including Glen Campbell, Tanya Tucker, Wayne
Newton, Donnie and Marie Osmond, and others, used REI8s.
Obviously. no one was going to settle for less than broadcast
standards, but shooting for reliability was certainly a key
design, installation and operational factor. The audio pool had
its own 15 KW standby generator that was checked -out, cable
run, then never used.
The sound reinforcement people were fed both output banks
of the switcher (redundancy again). They also had a split before
the podium -mie pre -amps. If they wanted it. they could choose
an orchestra mix or. do their own. Sub -mixes from the audio
pool available to the PA also included things like orchestra minus- entertainers, and entertainers- minus-orchestra.
In addition to microphones, the audio pool had both audio
and video tape sound to mix and distribute. Mixing was
accomplished on a Heider -modified Auditronics board which
Filmways Audio Services, Inc. supplied the equipment
Filmways' general manager John Phelan
supplied us with the following description:
seen here, and
Left -hand rack (Rack 2 of FIGURE 2):
The two top slots are BGW 100 amps for monitoring
and checking signal flow. The test panel immediately
below the amps has a dbx meter in the center, flanked
by two VU meters. Each meter may be switched to
either the console monitor (normal position) or to an
input jack on the patch bay, to test any of the DA lines.
The input of the monitor amps is switched at the same
Center rack (Rack i of FIGURE 3):
The top portion behind the dark plexiglass is the
microphone preamp section. These are Sphere preamps
on plug -in cards. Next is the patch bay for this rack,
where all preamp ins and outs, and relay ins and outs
show up. So do some tie lines and the line amp ins and
outs. Below the patch bay are some more preamp cards.
All switching relays are located in the back of the rack.
The bottom three sections are the Sphere power supplies
for the switching relays and the preamps.
Right -hand rack:
This is the utility rack. The top section has storage
slots for the NAB cartridges that will be used. Next is a set
of three Broadcast Electronics (another Filmways
company) tape machines: one record / play and two
playback -only. The setup is duplicated in the next
section. Next, there are two RTS power supplies for
the PL system, and the two empty spaces on the bottom
are for the Auditronics console power supplies.
General comments:
All inputs and outputs are through the back in a
normal situation. However, the patchbay offers ins and
outs as needed. The system was built in 23 days, from
design start to the point seen in the photo. The switcher
is a 60 -by -2 for the delegate mics. The distribution
amplifier is 44- by -176, strapped as shown in the drawings. In a modified version, this setup was used for the
Baltimore and Cleveland debates. Filmways president
Ken Fause designed most of the system to the needs of
the ABC and CBS networks.
Below, a detail drawing of the distribution amplifier
system, showing full schematic of a single channel.
(Photo by Bill Isenberg and line drawings courtesy of
Filmways Audio Services, Inc.)
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was fed sub -mixes from two smaller mixers. a Neve ( for vocals)
and a Yamaha (for orchestra). These sub -mixes were also
available to the house PA. Delegate microphones were assigned
to the main board via the special 60 -by -2 switcher with preamps. According to Hargreaves, "Each individual switch point
was a relay card. so that any given switch point that failed would
pop out to allow another card to be plugged in without
disturbing the system."
Joe Louis Arena was redecorated with a new speaker cluster
just for the convention. I asked Bob Coffeen. "Why did the
room need any other sound reinforcement, other than what was
already there?"
He answered. "That's the easiest question of all to answer."
The main reason was that the loudspeakers in the existing
cluster are directed only into the upper fixed -seating areas.
"There was absolutely no high -frequency energy directed to the
floor, where 2500 delegates sat."
There was a second reason. "The house cluster is in the center
of the room and mounted at a height of about 70 feet.
Consequently, even if loudspeakers in that cluster had been
oriented down toward the floor, the directional realism would
have been horrendous; that is. you would have thought
everything was coming from over your head while you were
looking at the podium."
"Additionally, the echo created by the long delay from that
cluster -with sound directed back to the person speaking
would have been very confusing, and would have made it very
difficult to speak. It turns out it's easier to speak with a cluster
400 feet away from you than at 80 feet. (That's my opinion.)
Once it's 400 feet away, the echo is distinct from what you are
saying. It doesn't over -lap very much. And consequently, you
can ignore it easier. But when it is about 80 feet away, or
something like that. it overlaps with what you are saying
enough to confuse you."
The new cluster was located about 52 feet above the Floor, or
about 42 feet above the podium, and slightly in front of it. The
reason for the location was easy: "That's where we could put it.
It was alright in that location, but it was a little bit too far into
the room. That had to be done to miss lighting and other things.
We used the existing cluster to cover the upper fixed -seating
areas. We let the new cluster overlap into the lower fixed- seating
and let the upper cluster (existing) cover the middle and upper
Coffeen's cluster was made of Electro-Voice components.
There were eight TL -606D bass boxes. one TL -606A bass box
(pointed down) and a total of twelve high -frequency horns.
consisting of 6040As and 9040As with DH 1012 drivers. These
were powered by SAE P -50 amplifiers, as was the upper cluster,
portions of which were delayed with Lexicon time delays to
coordinate it with the new cluster. Both Electro -Voice and
Crown crossovers were employed. Each cluster was equalized
with a White 4001 equalizer plus White 3900 series plug-in units
for narrow band equalization.
A ring of small broadcast studios had been set up around the
upper perimeter of the arena and these were also fed the house
P.A. mix. Again, a Lexicon unit was used to delay the audio
which was delivered through existing 8 -inch speakers. Each
studio opened up onto a balcony overlooking the arena floor.
thus requiring the time delay. Each 8 -inch speaker also had a
control for turning it down or off.
Electro-Voice FM 12 -3 stage monitors, powered by SAE P -50
power amplifiers, provided some very high monitor levels to
both the vocal entertainment and orchestra locations. For some
reason, these locations were completely across the arena floor
from each other, which meant a lot of monitor or some very
upset musicians.
Systems of this size don't just fall into place at the last minute.
Eight long days were spent in installing the reinforcement
system. This was preceded by two to three weeks investment by
KLA in fabricating the cluster and other system parts. Room
equalization was done at night. The crew worked three nights,
from about four in the afternoon until about seven-thirty in the
4. Wayne Newton and friend (an EV PL80)
during rehearsal for the convention.
The broadcast audio pool started coming together a full
sixteen days prior to air time. Hargreaves was there five days
before that. to start laying out the on- the -site ground work.
But what the audio crews will long remember may not be so
much the long hours and the intense work as the interesting
experiences that come out of working a national political
First there was Detroit! The people of Detroit did everything
right. They were the perfect hosts and thousands of convention
goers will never lose that image of the "Motor City."
For the CBS crew, there was the "Ice Palace," as the cramped
control room at times could have been best described. Jim
Hargreaves had arrived during a Detroit heat wave to find that
the control room had no air conditioning. The temporary
system he then had installed was certainly up to the task. In fact,
we may have been the only people in Detroit wearing jackets in
July. Hargreaves explained. "We discovered we were in the
visitors' dressing room in a hocky arena. The visitors. we
learned to our dismay. are traditionally given a hard time. They
can have heat in the summer. but not air conditioning. All the
ducts were there, and the arena was completely air- conditioned.
but not in the visitors' dressing room. and that's where we were.
So the tradition of hockey caught up to us."
Then there were the bomb "sweeps," when the security
officers would follow their sniffing dogs all over the arena.
Security, of course. was tight. We had to have three sets of
credentials. According to Bob Coffeen. he spent a total of about
twelve hours "just trying to get credentials."
We kept on running into different kinds of security guards:
Secret Service, plain- clothes Detroit policemen. and who
knows what else. Unfortunately. each seemed to respond only
to a certain pass, and even with a chain around your neck laden
with impressive plastic and paper (plus a Secret Service -issued
gold lapel pin and your Junior G -Man secret decoder ring). there
were times when you could not get to where you had to he.
Hargreaves commented. "We doubled the size of the crew and
hoped enough would get through to get on the air."
The fact that both the sound reinforcement and the broadcast
audio systems operated so effectively to overcome the many
potential obstacles is a credit to the designers, installers and
operators. There was the job of reinforcing the voices from the
podium, at times over unbelievable ambient noise levels; there
were the problems of dealing with inexperienced microphone
users; then, there was the RF in the arena. It was almost as if
everybody there had a walkie- talkie on their belt or a microwave transmitter on their back -pack, but there was never a
problem with it.
And where was E. Murphy. with all of those, potential
problems surrounding us? I think he just became bored by all
the redundancy. He met up with a crew of veterans who saw him
coming and wouldn't let him in -or maybe, he didn't have the
right pass.
Professional Systems, 8535 Fairhaven.
San Antonio, TX 78229. 512 -690 -8888.
Console with automation interface, producers desk, quad mix -down and many
other custom features, available with or
Closing date is the fifteenth of the second month preceding the date of issue.
Send copies to: Classified Ad Dept.
1120 Old Country Road. Plainview. New York 11803
without automation -$25,000 without
automation. Altec 604E Speakers -$225.
Contact Chris Bishop (215) 561 -3660.
Minimum order accepted $10.00.
ORBAN. All products in stock FOR IMMEDIATE DELIVERY. UAR Professional
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TX 78229. 512 -690 -8888.
Rates: 50e+ a word.
Boxed Ads: $25.00 per column inch.
db Box Number: $1.00 per issue.
Frequency Discounts: 3 times,10%; 6 times,20 %; 12 times,33%.
AMPEX AG- 350 -2: consoled, $1300.00;
unmounted, $1100.00. Crown 800 transports -quad heads. $500.00; with electronics, $650.00. Magnecord 1028 -2
$225.00; no electronics $75.00. (215)
338 -1682.
FOR SALE: EVENTIDE 910 harmonizer
(all option), Eventide 1745 DDL; as
packaged. RPM Sound Studios, 12 E. 12th
St., N.Y.C., NY 10003, call (212) 242 -2100.
RECORDING SECRETS MOST ENGINEERS WON'T TELL, $7.95 Tunetonics, P.O. Box 55, Edgewater, N.J. 07020.
SCULLY, NEW and used: FOR IMMEDIATE DELIVERY. UAR Professional Systems, 8535 Fairhaven, San Antonio, TX
78229.512- 690 -8888.
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.
model 636 with light meters, Parametric
Equalizer option, 36 x 36 in and out
4 EXRA Wild VCA faders. Option for 2 -24
track tape meter housing and 2 custom
Lexicon Prime Time: FOR IMMEDIATE
DELIVERY. UAR Professional Systems,
8535 Fairhaven, San Antonio, TX 78229.
512 -690 -8888.
Crown, Otari, Altec, Sound Workshop,
AKG, MXR-Pro, dbx & more. Sales, design, service and demonstration showroom. Autograph Pro Audio, 601 E. Black Iidge Dr., Tucson, AZ, (602) 882 -9016.
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,
made producers tables -$75,000.00.
Criteria Recording Studios, 1755 N E
149 St., Miami, FL 33181. (305) 947 -5611.
-INCH TAPE duplicating system. Six
Crown 800 transports. New 4 -channel
heads. Solid state. Mint. $4,200.00. (215)
Denver, Colo. 80218.
Interested in semi-pro
multitrack recording?
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Orban, JBL, Beyer, AKG, Technics,
FREE consultation on TEAC Tascam,
Ampex, Sennheiser, Eventide, Studio
Master, UREI, BGW, EV, Lexicon, ADA,
ano many mnr
Waste or ca
MXR & more. Paul Kadali s Inc., Baton
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ins; 15 minutes George Washington
Bridge. Professional Audio Video Corporation, 384 Grand Street, Paterson,
New Jersey 07505. (201) 523 -3333.
AKG. E.V. Sennheiser Shure. Neuman;
FOR IMMEDIATE DELIVERY most models UAR Professional Systems, 8535
Fairhaven, San Antonio, TX 78229. 512690 -8888.
will beat any legitimate
advertised price or price
quote on any 3M PRO audio
product, or we will give you
$50 CASH! Call now. Dealers excluded.
AMPEX, OTARI, SCULLY -In stock, all
major professional lines, top dollar trade -
662e HORNW000
HOUSTON. TX. 77071
REMANUFACTURED ORIGINAL equipment capstan motors for all Ampex and
Scully direct drive recorders, priced at
$225., available for immediate delivery
from VIF International, PO Box 1555,
Mtn. View, CA 94042, phone (408) 739-9740.
THE PA BIBLE from Electro- Voice, a professional guide addressing sound reinforcement and public address applications,
specifications from the club, church,
school level up through auditoriums, outside stadiums, road system situations.
To receive your copy of this highly
regarded tool, including all existing
supplements, and to be put on the distribution list for future additions, send $2.00
to Electro- Voice, Box No. 122, 600 Cecil
Street, Buchanan, Michigan 49107.
$12,500.00. Contact Jim Herrera or Tom
Harney at Hun Sound Inc., San Rafael, CA
(415) 454 -2911.
PRO -SOUND equipment. Specializing in
phone and mail orders. Free discount
catalog. Write or call Sonlx Co.. Dept. D,
P.O. Box 58, Indian Head, MD 20640
(301) 753-6432.
DOLBY M16 cat. 22 & cat. 44 cards wanted.
Reasonable! Jay Sound, 4300 Watertown
Rd., Maple Plain, MN. 55359-Ph. (612)
475 -3151.
TEST RECORD for equalizing stereo
systems. Helps you sell equalizers and
installation services. Pink noise in 1.3octave bands, type OR- 2011 -1 (a) $3800.
Used with various B & K Sound Level
Meters Bruel & Kjaer Instruments, Inc.,
185 Forest Street, Marlborough, Mass.,
LEXICON 224 Digital Reverberation. 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 Spring,
MD 20910. Write for complete product list.
SCULLY 100 -16, 16 and 8 track heads,
lb tracks DBX, excellent condition,
Eventide Flanger, ADS Graphic EQ,
$12,000. (215) 687 -6474.
BX20 AND BX10 AKG reverberation systems. FOR IMMEDIATE DELIVERY. UAR
Professional Systems, 8535 Fairhaven,
San Antonio, TX 78229. 512- 690 -8888.
NEW YORK CITY recording
studio needs additional maintenance
men. Dept. 111, db Magazine, 1120 Old
Country Road, Plainview, New York 11803.
coast mixing console manufacturer seeks
position as service or marketing manager
of reputable pro audio manufacturing
firm. For resume write to Dept. 120, db
Magazine, 1120 Old Country Road,
Plainview, NY 11803.
IVIE 10E REAL TIME Analyzer and 20B
Noise Generator $600 /obo, like new
condition. Call Mike (213) 570 -0938.
(Los Angeles, CA)
WANTED: POSITION as a 1st or 2nd
recording engineer, or position with a
good professional sound company or
professional audio equipment companyinstallations. Experience in studio recording, live sound reinforcement, video
work and audio installations. Good, hard,
efficient worker. Eddie -(205) 263 -6353.
condition, roll -about wood cabinet, remote control. 4K$ or best offer. (212)
757 -3650. Marc.
AMPEX SPARE PARTS; technical support;
updating kits, for discontinued professional audio models; available from
VIF International, Box 1555, Mountain
View, Ca. 94042. (408) 739 -9740.
items. UAR Professional Systems, 8535
Fairhaven, San Antonio, TX 78229. 512690 -8888.
MAGNETIC HEAD relapping -24 hour
service. Replacement heads for professional recorders. IEM, 350 N. Eric Drive,
Palatine, IL 60067. (312) 358 -4622.
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.
AKG BX20, EMT 140, MicMix
B. Orban 111b, Tapco 4400,
Sound Workshop 242A. New: Ecoplate
original, Ecoplate II, MicMix XL210,
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models Westrex, HAECO, Grampian.
Modifications done on Westrex. Quick
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for sale. Send for free brochure: International Cutterhead Repair, 194 Kings Ct.,
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UAR Professional Systems, 8535 Fairhaven, San Antonio, TX 78229. 512 -6908888.
The gooseneck
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Shop for pro audio from N.Y.'s leader,
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to one) cassette duplication with half
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masters. Tobÿ s Tunes, Inc., 2325 Girard
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Littlite- I:
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Fast emergency service. Speaker
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Universal Recording Corporation
has recently purchased a 3M 32-track
digital recorder with 4-track mix down.
According to Universal President Murray
Allen, the digital recorder will be available
for use in any of Universal Recording's
studios by the end of 1980.
Glen E. Meyer, formerly marketing
manager of commercial products at
Electro -Voice Inc., has joined the marketing group at Ivie Electronics in Orem,
Utah. In his new post, Meyer will be
responsible for marketing the new !vie
5000 modular sound reinforcement
system. In addition to his marketing
responsibilities, he will be working closely
with consultants and sound contractors
in providing design and applications
assistance for the use of Ivie products.
Dave Kelsey, president of The Film ways Audio Group, has announced the
appointment of Linda Feldman to handle
marketing for the group. Prior to this
position, Ms. Feldman was a communications journalist and marketing consultant. She will be based in the firm's
Hollywood, CA offices.
Dimension Five Studios, a professional audio company headquartered
in Womelsdorf. PA. recently unveiled
plans for greatly expanded coverage and
service for Philadelphia's broadcasting,
commercial sound and recording studios
and announced the appointment of a new
general manager and relocation of its
Philadelphia offices as major steps
towards accomplishing this coverage.
David Meyer, formerly a northeast
regional field sales rep for the Bose
Corporation's Professional Products
Division. was appointed Philadelphia
area general manager for Dimension
Five effective immediately. His primary
responsibility will be to develop the
broadcast and commercial market for
sound sales, as well as continuing to
service the Philadelphia area professional
and musician sound market. Dimension
Five will move from its present location
in Philadelphia to new, larger offices in
Bala Cynwyd.
Alexander M. Poniatoff, whose
boyhood fascination with a locomotive eventually led to two major
technological breakthroughs as the
founder of Ampex Corporation,
died October 24. at the age of 88.
He founded Ampex in 1944 and
served as president until 1955, when
he was elected chairman of the
board. He resigned as board chairman in 1970 and was named chairman emeritus. Although not active
in recent years in the management
or administration of Ampex, Poniatoff maintained an office at corporate headquarters in Redwood
City, California, participating in
several foundations undertaking
research in health and preventive
In 1947, Ampex was down to
eight employees in a post-World
War II recession, when they introduced the first practical magnetic
audio recorder in the United States.
That development was followed by
the introduction of the first practical
videotape recorder in 1956, an
invention that revolutionized the
television broadcast industry and
gave Ampex a worldwide reputation for technical innovation.
Ampex has since grown into close
to a 1/2- billion dollar corporation
with worldwide operations and over
12.000 employees. The company's
name comes from Poniatoffs ini-
tials, together with EX for excellence.
Poniatoff was born in Kazan,
Russia, on March 25. 1892. During
an interview when he was 84,
Poniatoff recalled that he saw his
first horseless vehicle, a locomotive,
when he was seven. "l decided right
then that I would build these
locomotives." he told the interviewer.
He attended the University of
Kazan, the Imperial College in
Moscow, and the Technical College,
Karlsruhe. Germany, obtaining
degrees in mechanical and electrical
He was a pilot in the Imperial
Russian Navy during World War I,
and then in the White Russian
Forces that were defeated during
the revolution. He escaped to
Shanghai, China, in 1920 and
worked as an assistant engineer for
the Shanghai Power Company until
1927, when he immigrated to the
U.S. He became an American citizen
in 1932.
He lived in Atherton, California.
with his wife Hazel. In addition to
his wife, he is survived by a niece.
Mrs. Peter (Anna) Kashkadanmoff.
of San Francisco.
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AMPEX MM-1200.
Select your multitrack recorder as carefully as the other
facets of your studio. Select the Ampex MM -1200.
Because its the one multitrack recorder that can go
through every change your studio goes through on its
way to greatness. And still be as economical and easy
to operate as the first great day you got it.
Growth without growing pains. With the MM -1200,
you'll seldom be faced with a situation you can't solve.
Because the MM -1200 comes prewired to make upgrading from 8 to 16, or 16 to 24 -track operation simple and
swift. And if adding channels won't solve your problem,
the MM- 1200's versatility will. Mastering, live sound rein-
forcement, double system sound, video sweetening
or film and TV production /post production are all jobs
that the MM -1200 can handle. Built -in single point
search -to -cue. elevated record level capability, 16" reel
capacity and fast start times also help you grow.
Performance you can depend on. The MM -1200 has
proven itself under some of the most adverse conditions.
The massive, stable top plate comes aligned and stays
aligned ... through repeated sessions in the comforts of
the studio, or on remote locations.
Ampex keeps your options open. The list of optional
accessories for the MM -1200 is the longest in the busi-
on Reader Service Card
ness. You can add multi -point search -to-cue and store
20 cue locations. This time -saving tape handling acces-
sory provides tape time readout, cue point readout,
"on-the -fly" cueing and more. Other accessories include
the PURC" Record Insert Controller, Search-ToCue Remote Control, and MSO -100 Synchronizer for jobs
that require more than 24 tracks. Contact your Ampex
sales representative for complete details.
Ampex Corporation. Audio-Video Systems Division
401 Broadway, Redwood City, CA 94063 415/367-2011
we had known in
advance the economic and
technical needs of 1980...
we had chosen to develop
a recording console that would
precisely meet those needs...
we couldn't have done better
than this...
now morQthcn ever
Di Harrison
P.O. Box 22964, Nashville, Tennessee 372C2
(615) 834 -1184, Telex 555133
Circle 12 on Reader Service Card
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