Audio Systems Guide for Houses of Worship Comprehensive review of microphones, wireless microphone systems and mixers for church sound applications. Specific sections covering miking techniques for altar, lectern and choir.

Audio Systems Guide for Houses of Worship  Comprehensive review of microphones, wireless microphone systems and mixers for church sound applications. Specific sections covering miking techniques for altar, lectern and choir.
By Tim Vear
A Shure Educational Publication
Audio Systems Guide for
Ta b l e o f C o n t e n t s
Introduction ........................................................................ 4
Chapter 1
Sound ................................................................................. 5
Chapter 2
The Sound Source ............................................................. 7
Chapter 3
The Sound System ............................................................ 8
What is good sound? ................................................... 9
Chapter 4
Microphones: Characteristics, Selection ........................... 10
Chapter 5
Microphones: Use............................................................... 17
Interference Effects...................................................... 19
Microphone Hookup.................................................... 21
Chapter 6
Wireless Microphone Systems ........................................... 25
Other Wireless Systems .............................................. 30
Chapter 8
Typical Applications ........................................................... 36
Lectern ........................................................................ 36
Altar ............................................................................ 37
Handheld Vocal .......................................................... 38
Lavalier ....................................................................... 39
Headworn ................................................................... 40
Choir ........................................................................... 41
Congregation ............................................................... 43
Musical Instruments ................................................... 43
Non-Sanctuary Applications ....................................... 45
Glossary .............................................................................. 47
Appendix One:
The Decibel ........................................................................ 50
Appendix Two:
Potential Acoustic Gain ..................................................... 51
Appendix Three:
Stereo Microphone Techniques ......................................... 53
Conclusion ......................................................................... 54
Chapter 7
Automatic Microphone Systems and Signal Processors ... 32
Signal Processors:
Equalizers and Feedback Control ............................... 33
Bibliography ....................................................................... 55
Biography ........................................................................... 55
Shure Products Selection Chart ........................................ 56
Houses of Worship
Audio Systems Guide for
Audio systems for house of worship applications
The scope of this guide is limited primarily to the
have evolved from simple speech reinforcement
selection and application of microphones
to full concert quality multi-media systems.
for house of worship applications. Since the
They run the gamut from the most traditional
microphone is the interface between the sound
services to the most contemporary services and
source and the sound system, it is necessary to
nearly every combination in between. Recording,
include some discussion of these two subjects,
broadcast, and video production are additional
and the subject of sound itself, to properly
aspects that must often be integrated with the
understand the function of the microphone.
audio system.
In addition, certain related devices such as
wireless microphones, automatic mixers, and
Though analog sound systems are still appropriate
audio signal processors will be discussed.
in many small and medium-size applications,
Large-scale mixers, power amplifiers, and
digital technology can now be found in nearly
loudspeakers are left to other publications.
every sound system component. Digital mixers,
signal processors, and networking are standard
The objective of this guide is to provide the reader
in most medium and large sound systems.
with sufficient information to successfully choose
Though the basic transducer (microphone and
and use microphones and related equipment in a
loudspeaker) remains in the analog domain,
variety of typical house of worship applications.
even those components are now paired with
However, for design and installation of a complete
technology such as digital wireless microphone
audio system the interested reader is encouraged
system or a digital loudspeaker control system.
to consult a qualified audio professional.
However, no matter how complex the overall audio
system, an understanding of the basic principles
of sound, the key elements of sound systems, and
the primary goal of “good sound” will insure the
best results in choosing and using that system.
Audio Systems Guide for
Because good sound quality is the goal of any house
of worship sound system, it is helpful to be familiar with
some general aspects of sound: how it is produced,
transmitted, and received. In addition, it is also useful to
describe or classify sound according to its acoustic
behavior. Finally, the characteristics of “good” sound
should be understood
Sound is produced by vibrating objects. These
include musical instruments, loudspeakers, and, of
course, human vocal cords. The mechanical vibrations
of these objects move the air which is immediately
adjacent to them, alternately “pushing” and “pulling” the
air from its resting state. Each back-and-forth vibration
produces a corresponding pressure increase
(compression) and pressure decrease (rarefaction) in
the air. A complete pressure change, or cycle, occurs
when the air pressure goes from rest, to maximum, to
minimum, and back to rest again. These cyclic pressure
changes travel outward from the vibrating object, forming
a pattern called a sound wave. A sound wave is a series
of pressure changes (cycles) moving through the air.
A simple sound wave can be described by its
frequency and by its amplitude. The frequency of a sound
wave is the rate at which the pressure changes occur. It is
measured in Hertz (Hz), where 1 Hz is equal to 1 cycle-persecond. The range of frequencies audible to the human ear
extends from a low of about 20 Hz to a high of about
20,000 Hz. In practice, a sound source such as a voice
usually produces
many frequencies
simultaneously. In
any such complex
sound, the lowest
AMPLITUDE frequency is called
the fundamental
and is responsible
for the pitch of the
Schematic of Sound Wave
sound. The higher
frequencies are called harmonics and are responsible
for the timbre or tone of the sound. Harmonics allow us
to distinguish one source from another, such as a piano
from a guitar, even when they are playing the same
fundamental note. In the following chart, the solid
section of each line indicates the range of fundamental
frequencies, and the shaded section represents the
range of the highest harmonics or overtones of the
The amplitude of a sound wave refers to the
magnitude (strength) of the pressure changes and
determines the “loudness” of the sound. Amplitude is
measured in decibels (dB) of sound pressure level (SPL)
and ranges from 0 dB SPL (the threshold of hearing), to
above 120 dB SPL (the threshold of pain). The level of
conversational speech is about 70dB SPL. A change of 1
dB is about the smallest SPL difference that the human ear
can detect, while 3 dB is a generally noticeable step, and
an increase of 10 dB is perceived as a “doubling” of
loudness. (See Appendix One: The Decibel.)
Instrument Frequency Ranges
Another characteristic of a sound wave related to
frequency is wavelength. The wavelength of a sound wave
is the physical distance from the start of one cycle to the
start of the next cycle, as the wave moves through the air.
Since each cycle is the same, the distance from any point
in one cycle to the same point in the next cycle is also one
wavelength: for example, the distance from one maximum
pressure point to the next maximum pressure point.
Wavelength is related to
frequency by the speed
of sound. The speed of
sound is the velocity at
which a sound wave travels.
The speed of sound is
constant and is equal to
about 1130 feet-per-second
in air. It does not change
with frequency or wavelength,
but it is related to them
in the following way: the
frequency of a sound,
Sound Pressure Level of
Typical Sources
multiplied by its wavelength
Audio Systems Guide for
Wave Equation
always equals the speed of sound. Thus, the higher the
frequency of sound, the shorter the wavelength, and the
lower the frequency, the longer the wavelength. The
wavelength of sound is responsible for many acoustic effects.
After it is produced, sound is transmitted through a
“medium”. Air is the typical medium, but sound can also
be transmitted through solid or liquid materials. Generally,
a sound wave will move in a straight line unless it is
absorbed or reflected by physical surfaces or objects in its
path. However, the transmission of the sound wave will be
affected only if the size of the surface or object is large
compared to the wavelength of the sound. If the surface is
small (compared to the wavelength) the sound will proceed
as if the object were not there. High frequencies (short
wavelengths) can be reflected or absorbed by small
surfaces, but low frequencies (long wavelengths) can be
reflected or absorbed only by very large surfaces or objects.
For this reason it is easier to control high frequencies by
acoustic means, while low frequency control requires
massive (and expensive) techniques.
Once a sound has been produced and transmitted, it is
received by the ear and, of course, by microphones. In the
ear, the arriving pressure changes “push” and “pull” on the
eardrum. The resulting motion of the eardrum is converted
(by the inner ear) to nerve signals that are ultimately perceived
as “sound”. In a microphone, the pressure changes act on a
diaphragm. The resulting diaphragm motion is converted (by
one of several mechanisms) into electrical signals which are
sent to the sound system. For both “receivers”, the sound
picked up is a combination of all pressure changes occurring
just at the surface of the eardrum or diaphragm.
Sound can be classified by its acoustic behavior; for
example, direct sound vs. indirect sound. Direct sound
travels from the sound source to the listener in a straight
line (the shortest path). Indirect sound is reflected by one
or more surfaces before reaching the listener (a longer
path). Since sound travels at a constant speed, it takes a
longer time for the indirect sound to arrive, and it is said to
be “delayed” relative to the direct sound. There are several
kinds of indirect sound, depending on the “acoustic
space” (room acoustics).
Echo occurs when an indirect sound is delayed long
enough (by a distant reflecting surface) to be heard by the
listener as a distinct repetition of the direct sound. If indirect
sound is reflected many times from different surfaces it
becomes “diffuse” or non-directional. This is called
reverberation, and it is responsible for our auditory
perception of the size of a room. Reverberant sound is a
major component of ambient sound, which may include
other non-directional sounds, such as wind noise or
building vibrations. A certain amount of reverberant sound
is desirable to add a sense of “space” to the sound, but an
excess tends to make the sound muddy and unintelligible.
One additional form of indirect sound is known as
a standing wave. This may occur when the wavelength of
a sound is the same distance as some major dimension
of a room, such as the distance between two opposite
walls. If both surfaces are acoustically reflective, the
frequency corresponding
to that wavelength will be
Sound Path
amplified, by addition of
the incoming and outgoing
waves, resulting in a
Sound Bar
strong, stationary wave
pattern between the two
Direct vs. Indirect Sound
surfaces. This happens
primarily with low frequencies, which have long
wavelengths and are not easily absorbed.
A very important property of direct sound is that it
becomes weaker as it travels away from the sound source, at
a rate governed by the inverse-square law. For example,
when the distance increases by a factor of two (doubles), the
sound level decreases by a factor of four (the square of two).
This results in a drop of 6 dB in sound pressure level (SPL),
a substantial decrease. Likewise, when the distance to the
direct sound source is divided by two (cut in half), the sound
level increases by 6 dB. In contrast, ambient sound, such as
reverberation, has a relatively constant level. Therefore, at a
given distance from a sound source, a listener (or a
microphone) will pick up a certain proportion of direct
sound vs. ambient sound. As the
distance increases, the direct
sound level decreases while the
ambient sound level stays the
same. A properly designed sound
/ M
system should increase the
amount of direct sound reaching
70 76db
the listener without increasing the
ambient sound significantly.
Inverse Square Law
Audio Systems Guide for
The sound sources most often found in worship
facility applications are the speaking voice, the singing
voice, and a variety of musical instruments. Voices may
be male or female, loud or soft, single or multiple, close or
distant, etc. Instruments may range from a simple
acoustic guitar to a pipe organ or even to a full orchestra.
Pre-recorded accompaniment is also very common.
In addition to these desired sound sources there are
certain undesired sound sources that may be present:
building noise from air conditioning or buzzing light
fixtures, noise from the congregation, sounds from street
or air traffic, etc. Even some desired sounds may become
a problem, such as an organ that overpowers the choir.
In this context, the loudspeakers of the sound system
must also be considered a sound source. They are a
“desired” sound source for the congregation, but they can
become an undesired source for microphone pickup:
feedback (an annoying howl or ringing sound) can occur
in a sound system if microphones “hear” too much of the
The acoustics of the room are often as important as the
sound source itself. Room acoustics are a function of the
size and shape of the room, the materials covering the
interior surfaces, and even the presence of the
congregation. The acoustic nature of an area may have a
positive or a negative effect on the sound produced by
voices, instruments, and loudspeakers before it is picked
up by microphones or heard by listeners: absorbing or
diminishing some sounds while reflecting or reinforcing
other sounds. Strong reflections can contribute to
undesired sound in the form of echo, standing waves, or
excessive reverberation.
Thus, sound sources may be categorized as desired
or undesired, and the sound produced by them may be
further classified as being direct or indirect. In practice,
the soundfield or total sound in a space will always consist
of both direct and indirect sound, except in anechoic
chambers or, to some extent, outdoors, when there are no
nearby reflective surfaces.
(direct sound)
(direct sound)
Direct Sound
Sound Field
Outside Noise
(e.g. street noise)
Inside Noise
(e.g. air conditioning
Reflected Sound
Sound Sources and Sound Field
Audio Systems Guide for
A basic sound reinforcement system consists of an
input device (microphone), a control device (mixer), an
amplification device (power amplifier), and an output
device (loudspeaker). This arrangement of components is
sometimes referred to as the audio chain: each device is
linked to the next in a specific order. The primary goal of
the sound system in house of worship sound applications
is to deliver clear, intelligible speech, and, usually,
high-quality musical sound, to the entire congregation.
The overall design, and each component of it, must be
intelligently thought out, carefully installed, and properly
operated to accomplish this goal.
There are three levels of electrical signals in a sound
system: microphone level (a few thousandths of a Volt),
line level (approximately one Volt), and speaker level (ten
Volts or higher). See Appendix One: The Decibel.
Sound is picked up and converted into an electrical
signal by the microphone. This microphone level signal is
amplified to line level and possibly combined with signals
from other microphones by the mixer. The power amplifier
then boosts the line level signal to speaker level to drive
the loudspeakers, which convert the electrical signal back
into sound.
Electronic signal processors, such as equalizers, limiters
or time delays, are inserted into the audio chain, usually
between the mixer and the power amplifier, or often within
the mixer itself. They operate at line level. The general
function of these processors is to enhance the sound in some
way or to compensate for certain deficiencies in the sound
sources or in the room acoustics.
In addition to feeding loudspeakers, an output of the
system may be sent simultaneously to recording devices or
even used for broadcast. It is also possible to deliver sound
to multiple rooms, such as vestibules and cry rooms, by
using additional power amplifiers and loudspeakers.
Finally, it may be useful to consider the room
acoustics as part of the sound system: acoustics act as a
“signal processor” that affects sound both before it is
picked up by the microphone and after it is produced by
the loudspeakers. Good acoustics may enhance the
sound, while poor acoustics may degrade it, sometimes
beyond the corrective capabilities of the equipment.
In any case, the role of room acoustics in sound system
performance cannot be ignored.
What is “good” sound?
The three primary measures of sound quality are
fidelity, intelligibility, and loudness. In a house of worship the
quality of sound will depend on the quality of the sound
sources, the sound system, and the room acoustics.
Typically, our references for sound quality are high fidelity
music systems, broadcast television and radio, motion
picture theaters, concerts, plays, and everyday
conversation. To the extent that the quality of many of these
references has improved dramatically over time, our
expectations of the sound quality in worship facilities has
also increased.
Typical Sound System
Audio Systems Guide for
The fidelity of sound is primarily determined by the
overall frequency response of the sound arriving at the
listener’s ear. It must have sufficient frequency range and
uniformity to produce realistic and accurate speech and
music. All parts of the audio chain contribute to this: a
limitation in any individual component will limit the fidelity
of the entire system. Frequency range of the human voice
is about 100-12kHz, while a compact disc has a range of
20-20kHz. A telephone has a frequency range of about
300-3kHz and though this may be adequate for conversational speech, it would certainly be unacceptable for
a sound system. However, even a high fidelity source
reproduced through a high fidelity sound system may
suffer due to room acoustics that cause severe frequency
imbalances such as standing waves.
The intelligibility of sound is determined by the overall
signal-to-noise ratio and the direct-to-reverberant sound
ratio at the listener’s ear. In a house of worship, the primary
“signal” is the spoken word. The “noise” is the ambient
noise in the room as well as any electrical noise added by
the sound system. In order to understand speech with
maximum intelligibility and minimum effort, the speech
level should be at least 20dB louder than the noise at every
listener’s ear. The sound that comes from the system
loudspeakers already has a signal-to-noise ratio limited by
the speech-to-noise ratio at the microphone. To insure that
the final speech-to-noise ratio at the listener is at least
20dB, the speech-to-noise ratio at the microphone must be
at least 30dB. That is, the level of the voice picked up by
the microphone must be at least 30dB louder than the
ambient noise picked up by the microphone.
The direct-to-reverberant ratio is determined by the
directivity of the system loudspeakers and the acoustic
reverberation characteristic of the room. Reverberation
time is the length of time that a sound persists in a room
even after the sound source has stopped. A high level of
reverberant sound interferes with intelligibility by making it
difficult to distinguish the end of one word from the start of
the next. A reverberation time of 1 second or less is ideal for
speech intelligibility. However, such rooms tend to sound
somewhat lifeless for music, especially traditional choral or
orchestral music. Reverberation times of 3-4 seconds or
longer are preferred for those sources.
Reverberation can be reduced only by absorptive
acoustic treatment. If it is not possible to absorb the
reverberant sound once it is created, then it is necessary
either to increase the level of the direct sound, to decrease
the creation of reverberant sound, or a combination of the
two. Simply raising the level of the sound system will raise
the reverberation level as well. However, use of directional
loudspeakers allows the sound to be more precisely
“aimed” toward the listener and away from walls and other
reflective surfaces that contribute to reverberation. Again,
directional control is more easily achieved at high
frequencies than at low frequencies.
Finally, the loudness of the speech or music at the
furthest listener must be sufficient to achieve the required
effect: comfortable levels for speech, perhaps more
powerful levels for certain types of music. These levels
should be attainable without distortion or feedback. The
loudness is determined by the dynamic range of the
sound system, the potential acoustic gain (PAG) of the
system, and the room acoustics. The dynamic range of a
sound system is the difference in level between the noise
floor of the system and the loudest sound level that it can
produce without distortion. It is ultimately limited only by
the available amplifier power and loudspeaker efficiency.
The loudness requirement dictates the needed acoustic
gain (NAG) so that the furthest listener can hear at a level
similar to closer listeners. It is relatively easy to design a
playback – only system with adequate dynamic range
based only on NAG and component specifications.
However, a sound reinforcement system with microphones requires consideration of potential acoustic gain.
Potential Acoustic Gain (PAG) is a measure of how
much gain or amplification a sound system will provide
before feedback occurs. This turns out to be much more
difficult than designing for dynamic range because it
depends very little on the type of system components but
very much on the relative locations of microphones,
loudspeakers, talkers, and listeners. (See Appendix Two:
Potential Acoustic Gain.)
Room acoustics also play a role in loudness.
Specifically, reverberant sound adds to the level of the
overall soundfield indoors. If reverberation is moderate, the
loudness will be somewhat increased without ill effect. If
reverberation is excessive, the loudness may substantially
increase but with potential loss of fidelity and intelligibility.
Although “good” sound is qualitatively determined
by the ear of the beholder, there are quantitative design
methods and measurements that can be used to
accurately predict and evaluate performance. It is
usually possible (though often not easy) to resolve the
competing factors of acoustics, sound systems,
architecture, aesthetics and budget in order to deliver
good sound in a house of worship. However, major
deficiencies in any of these areas can seriously
compromise the final result. Readers who are
contemplating major sound system purchases, acoustic
changes, or new construction are encouraged to speak
with knowledgeable consultants and/or experienced
contractors to ensure the “best” sound.
Audio Systems Guide for
The microphone is the first link in the audio chain
and is therefore critical to the overall performance of a
sound system. Improper selection of microphones may
prevent the rest of the system from functioning to its full
potential. Proper selection of microphones depends on
an understanding of basic microphone characteristics
and on a knowledge of the intended application.
To be most effective, a microphone must be matched
both to the desired sound source (voice, musical
instrument, etc.) and to the sound system (PA system,
tape recorder, etc.) with which it is used. There are five
areas of microphone characteristics that must be
considered when selecting a microphone for a particular
application. They are: 1) the operating principle of
the microphone, 2) the frequency response of the
microphone, 3) the directionality of the microphone,
4) the electrical output of the microphone, and
5) the physical design of the microphone.
1) Operating Principle: How does the microphone
change sound into an electrical signal?
The operating principle describes the type of
transducer inside the microphone. A transducer is a
device that changes energy from one form into another, in
this case, acoustic energy into electrical energy. It is the
part of the microphone that actually picks up sound and
converts it into an electrical signal. The operating principle
determines some of the basic capabilities of the
The two most common types are dynamic and
condenser. Although there are other operating principles
used in microphones (such as ribbon, crystal, carbon,
etc.) these are used primarily in communications
Dynamic Microphone
systems or are of historical interest only. They are rarely
encountered in worship facility sound applications.
Dynamic microphones employ a diaphragm/voice
coil/magnet assembly which forms a miniature sounddriven electrical generator. Sound waves strike a thin
plastic membrane (diaphragm) which vibrates in
response. A small coil of wire (voice coil) is attached to
the rear of the diaphragm and vibrates with it. The voice
coil itself is surrounded by a magnetic field created by a
small permanent magnet. It is the motion of the voice
coil in this magnetic field which generates the electrical
signal corresponding to the sound picked up by a
dynamic microphone.
Dynamic microphones have relatively simple
construction and are therefore economical and rugged.
They are not affected by extremes of temperature or
humidity and they can handle the highest sound
pressure levels without overload. However, the
frequency response and sensitivity of a dynamic
microphone is somewhat limited, particularly at very
high frequencies. In addition, they cannot be made very
small without losing sensitivity. Nevertheless, dynamic
microphones are the type most widely used in general
sound reinforcement and have many applications in
worship facility sound systems.
Condenser microphones are based on an
electrically-charged diaphragm/backplate assembly
which forms a sound-sensitive capacitor. Here, sound
waves vibrate a very thin metal or metal-coated-plastic
diaphragm. The diaphragm is mounted just in front of a
rigid “backplate” (metal or metal-coated ceramic). In
electrical terms, this assembly or element is known as a
capacitor (historically called a “condenser”), which has
the ability to store a charge or voltage. When the element
is charged, an electric field is created between the
diaphragm and the backplate, proportional to the spacing
between them. It is the variation of this spacing, due to
the motion of the diaphragm relative to the backplate, that
produces the electrical signal corresponding to the sound
picked up by a condenser microphone.
Condenser Microphone
Audio Systems Guide for
The construction of a condenser microphone must
include some provision for maintaining the electrical
charge. An “electret” condenser microphone has a
permanent charge, maintained by a special material
such as Teflon™ deposited on the backplate or on the
diaphragm. Other types are charged by means of an
external power source.
All condenser microphones contain additional
circuitry to match the electrical output of the element
to typical microphone inputs. This requires that all
condenser microphones be powered: either by batteries or
by “phantom” power (a method of supplying power to a
microphone through the microphone cable itself). There
are two potential limitations of condenser microphones
due to the additional circuitry: first, the electronics
produce a small amount of noise; second, there is a limit
to the maximum signal level that the electronics can
handle. Good designs, however, have very low noise levels
and are also capable of very wide dynamic range.
Condenser microphones are more complex than
dynamics and tend to be somewhat more costly.
However, condensers can readily be made with higher
sensitivity and can provide a smoother, more natural
sound, particularly at high frequencies. Flat frequency
response and extended frequency range are much easier
to obtain in a condenser. In addition, condenser
microphones can be made very small without significant
loss of performance.
The decision to use a condenser or dynamic
microphone depends not only on the sound source and
the signal destination but on the physical setting as well.
From a practical standpoint, if the microphone will be
used in a severe environment such as a fellowship hall or
for outdoor sound, a dynamic microphone would be a
good choice. In a more controlled environment, for
example, in a sanctuary, auditorium, or theatrical setting,
a condenser microphone might be preferred for some
sound sources, especially when the highest sound quality
is desired.
The two general types of frequency response are
flat and shaped. These terms refer to the graphical
representation of frequency response or response curve.
A microphone that provides a uniform output at every
audible frequency is represented on a frequency
response graph as an even, flat line, and is said to have
a flat response. This means that the microphone
reproduces all of the sound within its frequency range
with little or no variation from the original sound. In
addition, flat response microphones typically have an
extended frequency range; that is, they can reproduce
very high and/or very low frequencies as well. Wide-range,
flat response microphones have a natural, high-fidelity,
“uncolored” sound.
Flat Frequency Response
By contrast, a shaped microphone response will
appear on a frequency response graph as a varying line
with specific peaks and dips. This shows that the
microphone is more sensitive to certain frequencies than
to others, and often has a limited frequency range.
A shaped response is usually designed to enhance the
sound of a particular source in a particular application,
while at the same time minimizing the pickup of certain
unwanted sounds. Shaped response microphones each
have a “characteristic” sound.
2) Frequency Response: How does the microphone
The frequency response of a microphone is defined by
the range of sound (from lowest to highest frequency) that
it can reproduce, and by its variation in output within that
range. It is the frequency response that determines the
basic “sound” of the microphone.
Shaped Frequency Response
Audio Systems Guide for
The selection of a flat or shaped response
microphone involves consideration of both the sound
source and the sound destination. The frequency range of
the microphone must be wide enough to pick up the
desired range of the sound source. This range must also
be appropriate to the intended destination of the sound:
that is, wider range for high-quality sound systems or
recording/broadcast systems, narrower range for speechonly public address systems.
Within its range the microphone should respond in
such a way that the sound is reproduced either with no
change (flat response) or with changes that enhance the
sound in some desirable manner (shaped response).
Normally, microphones with flat, wide-range response are
recommended for high-quality pickup of acoustic
instruments, choral groups and orchestras, especially
when they must be placed at some distance from the
sound source. Flat response microphones are less prone
to feedback in high gain, distant pickup applications
because they do not have frequency response peaks that
might trigger feedback at any specific frequency.
The most common shaped response is for vocal use.
Typically, this consists of limiting the range to that of the
human voice and adding an upper mid-range response
rise. Such a “presence rise”, coupled with controlled lowand high-frequency response can give a sound with
improved vocal clarity. This is especially true for lapel or
lavalier microphones. The pickup of certain instruments
such as drums and guitar amplifiers may also benefit
from a shaped response microphone.
Finally, the frequency response of some microphones
is adjustable, typically by means of switches, to tailor the
Omnidirectional Microphone
microphone to different applications. Most common are
low-frequency rolloff controls, which can help prevent
“rumble”, and presence rise switches to enhance
3) Directionality: How does the microphone respond to
sound from different directions?
The directional characteristic of a microphone is
defined as the variation of its output when it is oriented at
different angles to the direction of the sound. It
determines how best to place the microphone relative to
the sound source(s) in order to enhance pickup of
desired sound and to minimize pickup of undesired
sound. The polar pattern of a microphone is the graphical
representation of its directionality. The two most common
directional types are omnidirectional and unidirectional.
A microphone that exhibits the same output
regardless of its orientation to the sound source will show
on a polar graph as a smooth circle and is said to have
an omnidirectional pattern. This indicates that the
microphone is equally sensitive to sound coming from all
directions. An omnidirectional microphone can therefore
pick up sound from a wide area, but cannot be “aimed”
to favor one sound over another.
A unidirectional microphone, on the other hand, is
most sensitive to sound coming from only one direction.
On a polar graph, this will appear as a rounded
but non-circular figure. The most common type of
unidirectional microphone is called a cardioid, because of
its heart-shaped polar pattern.
Cardioid (Unidirectional) Microphone
Audio Systems Guide for
A cardioid type is most sensitive to sound coming
from in front of the microphone (the bottom of the
“heart”). On the polar graph this is at 0 degrees, or “onaxis”. It is less sensitive to sound reaching the
microphone from the sides (“off-axis”), and the direction
of least sensitivity is toward the rear (the notch at the top
of the “heart”). For any microphone, the direction of least
sensitivity (minimum output) is called the null angle. For
a cardioid pattern, this is at 180 degrees or directly behind
the microphone.
Thus, a unidirectional microphone may be aimed at
a desired, direct sound by orienting its axis toward the
sound. It may also be aimed away from an undesired,
direct sound by orienting its null angle toward the sound.
In addition, a unidirectional microphone picks up less
ambient sound than an omnidirectional, due to its overall
lower sensitivity at the sides and rear. For example, a
cardioid picks up only one-third as much ambient sound
as an omnidirectional type.
Although the output of a unidirectional microphone is
maximum for sound arriving at an angle of 0 degrees, or
on-axis, it falls off only slightly for sound arriving from within
a certain angle off-axis. The total directional range for
usable output is called the coverage angle or pickup arc:
for a cardioid microphone this is about 130 degrees.
Two related types of unidirectional microphones are
the supercardioid and the hypercardioid. Compared to
a cardioid type, these have a progressively narrower
coverage angle: 115 degrees for a supercardioid and 105
degrees for a hypercardioid. However, unlike the cardioid,
they have some pickup directly behind the microphone.
This is indicated in their polar patterns by a rounded
Supercardioid Microphone
projection, called a lobe, toward the rear of the
microphone. The direction of least sensitivity (null angle)
for these types is about 125 degrees for the supercardioid
and 110 degrees for the hypercardioid. In general, any
directional pattern that has a narrower front coverage
angle than a cardioid will have some rear pickup and a
different null angle.
Monitor Speaker Placement For Maximum Rejection:
Cardioid and Supercardioid
The significance of these two polar patterns is their
greater rejection of ambient sound in favor of on-axis
sound: the supercardioid has the maximum ratio of
on-axis pickup to ambient pickup, while the hypercardioid
has the least overall pickup of ambient sound (only onequarter as much as an omni). These can be useful types
for certain situations, such as more distant pickup or in
higher ambient noise levels, but they must be placed
more carefully than a cardioid to get best performance.
Other types of unidirectional microphones include
“shotgun” and parabolic reflector models. The shotgun
has an extremely narrow pickup pattern and is used in
very high ambient noise situations. However, its limited
off-axis sound quality makes it unsuitable for typical
religious facility sound reinforcement. It is most often used
in broadcast and film production.
The parabolic type actually employs an omnidirectional
microphone placed at the focal point of a parabolic reflector.
Like a reflecting telescope, most of the energy (sound)
striking the reflector is concentrated at the focal point. This
effectively amplifies the sound from a distant source.
However, poor low frequency response, uneven off-axis
response, and its large size make it also unsuitable for sound
reinforcement. Again, it is used primarily in broadcast
applications such as sporting events.
One additional directional microphone is the
bidirectional type. As the name implies, it is equally
sensitive to sound from two directions: directly in front of
the microphone and directly behind it. Its polar graph
consists of a front pickup area and an identical rear lobe,
and resembles a “figure 8” pattern. Although the front
Audio Systems Guide for
coverage angle of a bidirectional microphone is only 90
degrees, it has equal rear coverage. The null angle is at
90 degrees, which is directly at the side of the
microphone. While the bidirectional microphone is not
used by itself in any typical house of worship sound
application, it is occasionally used in combination with
other types for stereo sound reproduction.
It should be noted that this discussion of directionality
assumes that the polar pattern for a microphone is uniform,
that is, the same shape at all
frequencies. In practice, this is
not always achieved. Most
microphones maintain their
“nominal” polar pattern over
only a limited range of
frequencies. This is the reason
that published polar patterns
include curves measured at
several frequencies. High-quality,
well-designed microphones are
Bidirectional Polar Pattern
distinguished by the uniformity
of their polar pattern over a
wide frequency range and by the similarity of the pattern
to the theoretical ideal.
many voices. Omnidirectional microphones do not exhibit
proximity effect. In addition, omnidirectional microphones
are less sensitive to wind noise and to handling noise.
Most quality unidirectional types have effective built-in
windscreens and shock mounts to compensate.
Proximity Effect Graph
Selecting an omnidirectional or unidirectional
microphone again depends on the sound source and the
destination of the audio signal. For recording (but not
sound reinforcement) of choral groups, orchestras, or
even the congregation, an omnidirectional microphone
may be used to pick up sound from all directions rather
than emphasizing individual voices or instruments.
However, as part of a sound reinforcement or P.A. system,
an omnidirectional microphone may be more prone to
feedback because it cannot be aimed away from the
loudspeakers. (See page 34 for more discussion of
A unidirectional model can not only help to isolate one
voice or instrument from other singers or instruments, but
can also reject background noise. In addition, a properly
placed unidirectional microphone can minimize feedback,
allowing higher sound reinforcement levels. For these
reasons, unidirectional microphones far outnumber
omnidirectional microphones in day-to-day use, in almost
all worship facility sound applications.
Directional Characteristics
There are a few operational differences between
omnidirectional and unidirectional microphones. A useful
feature of most unidirectional types is proximity effect.
This refers to the increased low-frequency response of a
unidirectional microphone when it is placed closer than
1 or 2 feet to the sound source. It becomes most
noticeable at very short distances: a substantial boost in
the bass response at less than 2 inches. In particular, for
closeup vocal use, proximity effect can add fullness and
warmth to the sound and therefore may be desirable for
4) Electrical output: How does the microphone output
match the sound system input?
The electrical output of a microphone is
characterized by its sensitivity, its impedance, and by its
configuration. The same characteristics are used to
describe microphone inputs in sound systems. This
determines the proper electrical match of a microphone
to a given sound system.
The sensitivity of a microphone is defined as its
electrical output level for a certain input sound level. The
Audio Systems Guide for
greater the sensitivity, the higher the electrical output will
be for the same sound level. In general, condenser
microphones have higher sensitivity than dynamic
microphones of comparable quality. It should be noted
that for weak or distant sound, a microphone of high
sensitivity is desirable, while loud or closeup sound can be
picked up well by lower-sensitivity microphones.
Impedance is, approximately, the output electrical
resistance of the microphone: 150-600 ohms for low
impedance (low Z), 10,000 ohms or more for high
impedance (high Z). While the majority of microphones
fall into one of these two divisions, there are some that
have switchable impedance selection. In any case, the
choice of impedance is determined by two factors: the
length of cable needed (from the microphone to the
microphone input) and the rated impedance of the
microphone input.
The maximum length of cable that may be used with
a high-impedance microphone should be limited to no
more than 20 feet. For longer cable lengths, the
high-frequency response of the microphone will be
progressively diminished. Low-impedance microphones,
on the other hand, may be used with cable lengths of
1000 feet or more with no loss of quality, and are therefore
preferable for most applications.
The output configuration of a microphone can be either
balanced or unbalanced. A balanced output carries the
signal on two conductors (plus shield). The signals on each
conductor are the same level but they are of opposite polarity
(one signal is positive when the other is negative). Most
microphone mixers have a balanced (or differential) input
which is sensitive only to the difference between the two
signals and ignores any part of the signal that is the same in
each conductor. Because of the close proximity of the two
conductors in a balanced cable, any noise or hum picked up
by the cable will be the same level and the same polarity in
each conductor. This common-mode noise will be rejected
by the balanced input, while the original balanced
microphone signal is unaffected. This greatly reduces the
potential for noise in balanced microphones and cables.
An unbalanced output signal is carried on a single
conductor (plus shield). An unbalanced input is sensitive
to any signal on that conductor. Noise or hum that is
picked up by the cable will be added to the original
microphone signal and will be amplified along with it
by the unbalanced input. For this reason, unbalanced
microphones and cables can never be recommended for
long cable runs, or in areas where electrical interference
is a problem.
The two most common microphone output types and
mixer input types are Balanced Low-Impedance and
How a Balanced Input Works
How an Unbalanced Input Works
Balanced and Unbalanced Cables and Connectors
Unbalanced High-Impedance. Since all high-quality and
even most medium-quality microphones have a balanced,
low-impedance output, this is the recommended type for
the majority of worship facility sound system applications,
especially when long cable runs are used.
5) Physical design: How does the mechanical and
operational design relate to the intended application?
Microphones for house of worship sound applications
include several typical designs: handheld, user-worn,
free-standing mounted, and boundary or surface
mounted. Each is characterized by a particular size,
shape, or mounting method that lends itself to a specific
manner of use. In addition, some microphones may be
equipped with special features, such as on-off switches,
that may be desirable for certain situations.
Handheld types are widely used for speech and
singing in many areas of worship facility sound. Since they
are usually handled, passed from person to person, or
used while moving about, they must have a very effective
internal shock mount to prevent pickup of handling noise.
In addition, they are often used very close to the mouth
and should therefore be equipped with an effective “pop”
filter or windscreen to minimize explosive breath sounds.
Size, weight and feel are important considerations for a
handheld microphone.
Audio Systems Guide for
User-worn microphones include “lapel” types that
may be attached directly to clothing, lavalier styles worn
on a lanyard around the neck, and head-worn models.
In particular, head-worn microphones have become
much more common as their size has decreased. The
proximity of the head-worn microphone to the mouth
results in much better sound quality and vastly
increased gain-before-feedback when compared to a
lapel type. Small size and unobtrusive appearance are
the critical characteristics for user-worn microphones.
Free-standing mounted microphones (mounted
away from large surfaces) come in a variety of styles
suited for different fixed settings. These range from fullsize microphones on heavy-duty stands, to miniature
types on unobtrusive goosenecks or booms, to overhead
microphones of any size. Mounted microphones are
generally selected for permanent installation, although
many handheld types may be placed in mounts and
removed as needed. Shock isolation is essential if the
stand is likely to be moved or is mounted on a vibrating
stage or hollow lectern. Windscreens are necessary for
close-up vocals or if used outdoors. Again, appearance
is often a primary factor in mounted microphones.
Boundary or surface-mounted microphones are
also used in fixed positions, but the surface to which
they are attached is essential to the operation of the
microphone. These microphones are most successfully
mounted on existing surfaces (such as altars, floors,
walls, or ceilings) to cover a certain area. They depend
to a great extent on the acoustic properties of the
mounting surface (size, composition, orientation) for
their frequency response and directionality. However,
they offer a very low profile and can minimize certain
acoustic problems due to reflected sound. Appearance
and physical environment play an important part in the
selection of boundary microphones.
It should be noted that almost any combination of
the other four microphone characteristics can be found
in any of the physical designs mentioned here. That is,
most of these designs are available in a choice of
operating principle, frequency response, directional
pattern, and electrical output.
Though not intrinsically related to the other four
areas of microphone specification, the physical design is
no less important in the selection process and, indeed,
is often one of the first choices dictated by the
application. In any case, the other microphone
specifications should be just as carefully chosen to
satisfy the basic acoustic and electrical requirements of
the application. Ultimately, all five characteristics must
be properly specified to yield the best results.
handheld wireless
A Selection of Microphone Designs
Audio Systems Guide for
Once a microphone is selected for a given
application, it must be used properly to get the best
possible results. Again, there are two key areas: the
interface of the microphone with the sound source, and
the interface of the microphone with the sound system.
The first area involves primarily acoustic considerations
for optimum placement of one or more microphones.
The second area involves electrical and mechanical
considerations for optimum operation of microphones.
Microphone Placement
Microphone placement is a challenge that depends
on the acoustic nature of the sound source and the
acoustic characteristics of the microphone. Although this
may appear to be a very subjective process, a description
of some of the important acoustic considerations will lead to
a few simple rules for successful microphone placement.
Recall that sounds can be categorized as desired or
undesired and that the soundfield, or total sound in a
space, is made up of both direct sound and ambient
sound. The level of direct sound decreases with distance
(the inverse-square law) while ambient sound stays at a
constant level. The critical distance is the distance (from
the sound source) at which the level of direct sound has
fallen to the level of the ambient sound. Critical distance is
determined by the loudness of he direct sound relative to
the loudness of the ambient sound. A quiet talker in a noisy
room has a short critical distance while a loud talker in a quiet
room has a longer critical distance. In practice, microphones
must be placed much closer than the critical distance to
get an acceptable ratio of direct-to-ambient sound.
Critical Distance
This brings up the concept of “reach”, or distant
pickup capability. The proportion of direct vs. ambient
sound picked up by a microphone is a function not only of
distance but of the directional pattern of the microphone as
well. For a given ratio of direct-to-ambient sound, a
unidirectional microphone may be used at a greater
distance from the direct sound source than an
omnidirectional type. This is called the distance factor, and
ranges from about 1.7 for a cardioid, to 2.0 (twice the omni
distance) for a hypercardioid. See chart on page 14.
For instance, if an omnidirectional microphone
picked up an acceptable direct-to-ambient sound ratio
at 2 feet from the sound source, then a cardioid would
have the same ratio at about 3.4 feet, although the gain
would have to be increased to achieve the same output
level. However, for a very weak source, or a very high
ambient sound level, the acceptable omni location
(again, less than the critical distance) could be as little
as 3 inches away, for example. In this case, even a
hypercardioid could only be used 6 inches away. Reach
is thus a very subjective concept and is dominated by
the actual direct vs. ambient sound level at the
microphone position rather than by the directionality of
the microphone: even an omni would have excellent
reach, if no ambient sound were present. Note that
directional microphones are not more sensitive to onaxis sound. They are just less sensitive to off-axis sound!
In the normal operation of a sound system, some of
the sound produced by the loudspeakers is picked up
by the microphone and re-enters the system. As the gain
of the system is increased, the level of the sound from
the loudspeakers at the microphone also increases.
Eventually, at some gain setting, this re-entrant sound
will be amplified to the same level as the original sound
picked up by the microphone. At this point the system
will begin to “ring” or oscillate. Higher gain will result in
the sustained “howl” or drone known as feedback.
There are many factors that affect the potential
acoustic gain (maximum gain-before-feedback) of a
sound system. By far, the most important ones are the
relative distances between the sound source and the
microphone, between the microphone and the
loudspeaker, and between the loudspeaker and the
listener. The number of “open” or active microphones
also plays a strong role. These factors are discussed in
Appendix Two: Potential Acoustic Gain.
Lesser factors are the directional characteristics of the
microphones and loudspeakers, local acoustical reflections,
room reverberation, and the overall frequency response of
the sound system. Use of directional microphones and
directional loudspeakers can reduce the amount of direct
sound picked up by the microphone from the loudspeaker
by aiming them away from each other. Of course this is
limited by the directional or “pattern” control of the devices.
Audio Systems Guide for
Acoustical and Electrical Feedback Path
In practice, loudspeakers have very little directivity at
low frequencies (where the wavelength is large compared
to the speaker size).
Acoustic reflections from objects near the microphone
can aggravate feedback problems. For example: sound
from a monitor speaker placed behind the microphone can
reflect off the performer’s face into the front of the
microphone, or a lectern surface can reflect the sound from
an overhead cluster. Placing a hand on the front of or
around the grille of a microphone can severely disrupt its
polar pattern and frequency response.
Room reverberation increases the overall sound level
throughout the room. Because it causes sound to persist
even after the source stops, ringing and feedback tend to
be more sustained. Since reverberation is not uniform
with frequency it may also increase the likelihood of
feedback at certain frequencies.
In fact, the overall frequency response of the sound
system is affected by each component in the system as
well as the room response. Feedback occurs first at the
frequency that has the highest sensitivity in the system
response curve. A peak in the response of a microphone
or loudspeaker or an unusual boost in an equalizer can
trigger feedback as the system gain is increased.
Flat response systems can generally operate with more
gain-before-feedback. Judicious use of equalizers can
improve the stability of a sound system if feedback is
occurring just at a few specific frequencies. However,
equalizers will not allow the system to exceed the inherent
limits of the PAG calculation.
This leads to the first and most important rule of
microphone placement: Place the microphone as close as
practical to the desired sound source.
It has several corollaries: place the microphone as far
as possible from loudspeakers and other undesired
sources; use directional microphones to minimize
ambient sound pickup; aim directional microphones
toward the desired sound and/or away from undesired
sound; and keep the system gain to a minimum.
Ultimately, the position chosen should be consistent
with the characteristics of both the sound source and the
microphone: larger sources, such as a choir, may require
greater distance, depending on the microphones’
directionality; extremely loud sources may require greater
distance to avoid overload of some sensitive condenser
microphones; and close vocal use requires adequate
“pop” filtering. In any case, following the above rules will
give the best pickup of the desired sound, the minimum
pickup of undesired sound, and the least likelihood of
Not enough gain-before-feedback?
Here is what you can do:
(In order of importance)
• Move microphones closer to sources
• Move loudspeakers farther from microphones
• Move loudspeakers closer to listeners
• Reduce the number of open microphones
• Use directional microphones and loudspeakers
• Eliminate acoustic reflections near microphones
• Reduce room reverberation by acoustic
• Use equalizers to reduce system gain at
feedback frequencies
There are no other solutions!
Audio Systems Guide for
Interference Effects
An important consideration in microphone use is
acoustic interference. Interference effects may occur
whenever delayed versions of the same sound are mixed
together, acoustically or electrically. With microphones,
this may happen in several ways: microphones of reverse
polarity picking up the same sound, multiple microphones
picking up the same sound from different distances, a
single microphone picking up multiple reflections of the
same sound, or any combination of these. The results are
similar in each case, and include audible peaks and dips
in frequency response, apparent changes in directionality,
and increased feedback problems.
The first situation, reverse polarity, will result in
severe loss of sound, especially low frequencies, when a
microphone with reverse polarity is placed next to
another of correct polarity and set to the same level.
Signals from the microphones are then of equal strength
but of opposite polarity. When these signals are
combined in a mixer the cancellation is nearly total.
sound picked up by the more distant microphone will be
delayed relative to the near microphone. When these
signals are combined in a mixer, peaks and notches
occur at multiple frequencies which are related to the
delay time, and hence, to the distances between the
microphones. This effect is called “comb filtering”
because the resulting frequency response curve
resembles the teeth of a comb. As the delay time
increases, comb filtering starts at lower frequencies. It is
especially noticeable at middle and high frequencies,
and creates a “hollow”, distant sound.
Multi-mic Comb Filtering
Polarity Reversal
Although there is an international standard for
microphone polarity (pin 2+, pin 3-), a reversal may be
found in an incorrectly wired microphone cable. It can
be identified by checking each microphone and cable
against a microphone and cable that are known to be
correct. In any installation, all microphones and
microphone cables must have the same polarity.
The second form of interference is the result of
multiple microphone pickup and can occur whenever
more than one microphone is used. If the microphones
are at unequal distances from the sound source, the
The solution to this problem is to use the threeto-one rule: for multiple microphones, the microphoneto-microphone distance should be at least three times
the source-to-microphone distance.
For example, when using individual microphones
on a vocal group, if a singer’s microphone is one foot
away, then the next nearest microphone should be at
least three feet away from the first. This insures that
direct sound from the singer will not be strong enough
to cause noticeable interference when picked up by
the more distant microphones. As the sourceto-microphone distance increases, the distance to
adjacent microphones must also be increased.
An implication of the three-to-one rule is the following:
avoid picking up the same sound source with more than
one microphone. Microphones should be placed and
aimed to minimize areas of overlapping coverage. This is
important for a number of sound applications: for area
pickup applications, such as choir lofts and stages, each
section or area should be covered by only one
microphone; for lectern applications, only one microphone
Audio Systems Guide for
3 to 1 Rule
should be used; when a lavalier microphone wearer speaks
into a fixed microphone, one of the microphones should be
turned down.
The third form of interference, reflection pickup, may
occur whenever there are nearby sound-reflecting surfaces.
This is often true in worship facility settings: hardwood or
stone floors, brick or glass walls, wood or plaster ceilings,
and solid lecterns and altars. Recall that reflected sound is
always delayed relative to direct sound. When the delayed,
reflected sound arrives with the direct sound at the
microphone, acoustic comb filtering is again the result.
The first solution is to increase the direct sound level
by placing the microphone as close as practical to the
sound source, so that the direct sound is much stronger
than the reflected sound. Interference effects only
become noticeable when the reflected sound is
comparable in level to the direct sound. However, close
placement may not be possible in the case of area
coverage or moving sound sources.
The second solution is to decrease the reflected sound
level. The microphone may be moved away from the
reflective surface, or re-oriented for minimum pickup of
sound from that direction. The acoustically reflective surface
may possibly be moved away, re-oriented, or treated with
some sound-absorbent material. However, this is often not
feasible, for economic or aesthetic reasons.
The third alternative is to minimize the delay. Since
the delay is due to the difference in the paths of the direct
and reflected sound, this can be accomplished by placing
the microphone close to the reflective surface, so that the
direct sound and the reflected sound have nearly the
same path. This raises the frequency at which comb
filtering starts. If the microphone can be brought very close
to the surface (within one-quarter inch), any comb filtering
will occur above the audible range.
Surface-mount or “boundary effect” microphones
are designed to effectively reduce interference from the
surface on which they are located. If they are located at
the junction of two or more surfaces, such as the corner
Reflection Pickup
of a room, they reduce interference from each adjacent
surface. In addition, a boundary microphone exhibits
increased output due to its combining of the direct and
reflected sound energy.
To minimize reflection pickup, avoid using
microphones near acoustically reflective surfaces. If this
is not possible, consider using a surface-mount
microphone on the primary reflecting surface.
In addition to interference problems, the use of multiple
microphones creates other potential difficulties. One of these
is due to the fact that as the number of active microphones in
a sound system increases, the overall system gain or volume
also increases. (See Appendix Two: Potential Acoustic Gain.)
This has the effect of increasing feedback problems. And, of
course, each active microphone is adding more ambient
noise pickup to the system.
This leads to a final general rule for microphone use:
Always use the minimum number of microphones.
If additional microphones are not needed, they may actually
degrade the sound system. If the application can be satisfied
with one microphone, use only one microphone!
Audio Systems Guide for
Microphone Hookup
The second key area of microphone use is the
interface of the microphone with the sound system. As
mentioned at the beginning of this section, this involves
primarily electrical considerations. We will develop a few
simple rules for proper interface based on the electrical
characteristics of the microphone output and the sound
system input, and on the requirements for cables and
connectors to achieve maximum reliability.
In the discussion of operating principle it was
mentioned that all condenser microphones require
power for their operation. This is provided by an internal
battery in some models, or by phantom power in others.
If a condenser is selected, care must be taken to assure
that the appropriate power source (battery or phantom) is
available. A battery-powered condenser is fine for
applications such as portable recording but phantom
power should be used for any permanent microphone
Phantom Power Schematic
Phantom power, sometimes called “simplex”, is
provided through the microphone cable itself. It is a DC
(direct current) voltage that may range from 9 to 48 volts,
depending on the microphone requirement and the
phantom power source rating. This voltage is applied
equally to the two conductors of a balanced microphone
cable, that is pin 2 and pin 3 of an XLR-type connector.
The voltage source may be either in the mixer itself or
in a separate phantom power supply connected in line
with the microphone cable. Most recent mixers have
phantom power built in, and the actual voltage will be
stated on the mixer or in the operating manual.
The voltage requirement for a phantom-powered
condenser microphone will also generally be stated on the
microphone or in the manufacturer's literature. Some
types, particularly those that are externally charged, may
require a full 48 volt supply. Electret types, which have a
permanent charge, will typically operate over the entire
range from 12 to 48 volts. Unless specifically stated
otherwise by the manufacturer, these microphones will
deliver their full performance at any voltage in this range,
and further, they will not be damaged by a full 48 volt
supply. Supplying less than the recommended voltage to
either type may result in lower dynamic range, higher
distortion, or increased noise, but this also will not
damage the microphone.
Dynamic microphones, of course, do not require
phantom power. However, many mixers have only a single
switch that supplies phantom power to all microphone
inputs, which may include some used by dynamic
microphones. The presence of phantom power has no
effect on a balanced, low-impedance dynamic
microphone. It is not possible to damage or impair the
performance of a balanced microphone correctly hooked
up to any standard phantom supply.
If a balanced microphone is incorrectly wired or if an
unbalanced, high-impedance microphone is used, there
may be a loud “pop” or other noise produced when the
microphone is plugged in or switched on. In addition, the
sound of the microphone may be distorted or reduced in
level. Even in these cases, the microphone will still not be
damaged and will work normally when the wiring is
corrected or the phantom power is turned off. If an
unbalanced microphone must be used with a phantompowered input, an isolating transformer should be
inserted. By the same token, it is also not possible to
damage any standard phantom power source by
improper microphone connection.
Good phantom power practices are:
• check that phantom voltage is sufficient for the selected
condenser microphone(s);
• turn system levels down when connecting or
disconnecting phantom-powered microphones, when
turning phantom power on or off, or when turning
certain phantom-powered microphones on or off;
• check that microphones and cables are properly wired.
Following these practices will make condenser microphone
use almost as simple as that of dynamics.
Audio Systems Guide for
Phantom Power vs. Bias Voltage
In a condenser microphone, one function of the
circuitry is to convert the very high impedance of the
condenser element to a lower impedance. For an electret
condenser (the most common type) this is done by a single
transistor. Some condenser designs, such as lavalier types
or miniature hanging types, have their electronics separate
from the microphone element. In these models, the
impedance converting transistor is built in to the
microphone element itself. The main part of the circuitry is
contained in a separate module or pack usually connected
to the element by a thin shielded cable.
The main electronics of such designs operate on
phantom power supplied through the microphone cable or
by means of a battery in the pack itself. However, the
impedance-converting transistor in the microphone
element also requires power in a form known as “bias”
voltage. This is a DC voltage, typically between 1.5 and 5
volts. It is carried on a single conductor in the miniature
connecting cable, unlike phantom power which is carried
on two conductors in the main microphone cable. In
addition, the audio signal in the miniature cable is
unbalanced while the signal in the main cable is balanced.
This distinction between phantom power and bias
voltage is important for two reasons. The first concerns the
use of wireless transmitters. Body-pack transmitters which
operate on 9 volt (or smaller) batteries cannot provide
phantom power (12-48 volts DC). This prevents their use
with phantom-powered condenser microphones. However,
the body-pack transmitter can provide bias voltage (1.5-5
volts DC). This allows a condenser microphone element
with an integrated impedance-converting transistor to be
used directly with a body-pack transmitter. Miniature
condenser lavalier types as well as other designs which
have separate electronics can be operated with wireless
systems in this way.
The second reason concerns the wired installation of
condenser microphones with separate electronic
assemblies such as miniature hanging microphones for
choir, congregation, or other “area” applications. Since the
audio signal in the cable between the microphone element
and the electronics is unbalanced, it is more susceptible to
pickup of electronic noise. This is particularly true for radio
frequency noise because the cable itself can act as an
antenna, especially for a nearby AM radio station. For this
reason it is strongly recommended to keep the length of this
part of the cable as short as possible, preferably less than
35 feet. It is a much better practice to extend the length of
the balanced cable between the electronics assembly and
the mixer input.
For the expected sound level, microphone sensitivity
should be high enough to give a sufficient signal to the
mixer input. In practice, most mixers are capable of
handling a very wide range of microphone signal levels.
Occasionally, for extremely high sound levels, an
“attenuator” may be necessary to lower the output of the
microphone. These are built into some microphones and
mixers. Otherwise, accessory attenuators are available that
may be inserted in line with the microphone cable.
It has already been mentioned that balanced, lowimpedance microphones are recommended for the
majority of worship facility sound applications. This will
allow the use of long microphone cables, and result in
the least pickup of electrical noise. In any case, the
microphone impedance should be similar to the rated
impedance of the microphone input of the mixer or other
equipment. It is not necessary or even desirable to
match impedances precisely. It is only necessary that
the actual input impedance be greater than the
microphone output impedance. In fact, the actual
impedance of a typical microphone input is normally five
to ten times higher than the actual output impedance of
the microphone. The microphone input impedance of
most mixers ranges from 1000 ohms to 3000 ohms,
which is suitable for microphones of 150 ohms to
600 ohms.
When it is necessary to match a balanced,
low-impedance microphone to an unbalanced, highimpedance input, or vice versa, transformers with the
appropriate input and output connectors are readily
available. Transformers provide an impedance matching
function and can also change the configuration from
balanced to unbalanced as needed. Ideally, transformers
should be connected so that the bulk of the cable run is
balanced, low-impedance, for maximum allowable length
and minimum noise pickup. This would normally place
the transformer at the connector of the unbalanced,
high-impedance device.
Professional (and most semi-professional) equipment
has balanced, low-impedance microphone inputs using
3-pin XLR-type connectors. Less sophisticated musical
instruments, consumer electronic products, computers
and many portable recording devices typically have
unbalanced, high-impedance microphone inputs using
1/4 inch phone jacks or 1/8 inch mini-phone jacks. A few
mixers offer both types of connectors for each input
channel. Simple adapters may be used to mate different
types of connectors if no configuration change (high/low
impedance or balanced/unbalanced signal) is necessary.
Use only high-quality connectors and adapters.
Audio Systems Guide for
In-Line Transformers
Optimum microphone performance depends on the
associated connectors and cables. In addition to quality
connectors of the types described above, it is equally
important to use high-quality cables. Beyond the basic
specification of balanced (two conductors plus shield) or
unbalanced (one conductor plus shield), there are several
other factors that go into the construction of good cables.
The conductors: carry the actual audio signal (and
phantom voltage for condensers), usually stranded wire.
They should be of sufficient size (gauge) to carry the signal
and provide adequate strength and flexibility; use stranded
conductors for most applications, solid conductors only for
stationary connections.
The shield: protects the conductors from electrical
noise, may be braided or spiral wrapped wire, or metal
foil. It should provide good electrical coverage and be
flexible enough for the intended use: braid or spiral for
movable use, foil only for fixed use such as in conduit.
The outer jacket: protects the shield and conductors
from physical damage, may be rubber or plastic. It should
be flexible, durable, and abrasion resistant. Depending on
the location it may need to be chemical or fire resistant.
Different color jackets are available and can be used to
identify certain microphone channels or cables.
A large percentage of microphone problems are
actually due to defective or improper microphone cables.
Microphone cables should be handled and maintained
carefully for long life: position them away from AC lines
and other sources of electrical interference to prevent
hum; allow them to lie flat when in use to avoid snagging;
use additional cable(s) if necessary to avoid stretching;
do not tie knots in cables; coil loosely and store them
when not in use; periodically check cables visually and
with a cable tester.
Individual, pre-assembled microphone cables are
readily found in a wide variety of styles and quality.
In addition, multiple cable assemblies, called “snakes”,
are available for carrying many microphone signals from
one location to another, such as from the sanctuary to the
sound booth. The use of only high-quality cables and
their proper maintenance are absolute necessities in
any successful worship facility sound application.
Finally, the use of microphones for particular
applications may be facilitated by microphone accessories.
These are mechanical and electrical hardware items that
are often used in mounting and connecting microphones.
Mechanical accessories include various kinds of
acoustic devices such as windscreens and directionality
modifiers. Windscreens, usually made of special foam or
cloth, should be used whenever microphones are used
outdoors or subjected to any air currents or rapid motion.
“Pop” filters are employed when the microphone is used
close to the mouth, such as on lecterns or for handheld
vocals. These minimize noise caused by explosive
consonants such as “p”, “b”, “t”, or “d”. Although such
filters are usually supplied with microphones designed for
these applications, additional protection may be needed
in some cases. Use only high-quality screens and filters
to avoid degrading the sound of the microphone.
There are directional or “polar” modifiers available
for certain microphones that can change the pickup
pattern form cardioid to supercardioid, for example, or
from omnidirectional to semi-directional in the case of
some boundary microphones. Consult the manufacturer
for proper use of these accessories.
Audio Systems Guide for
Mounting accessories are of great importance in
many worship facility sound applications. Stands, booms,
and goosenecks should be sturdy enough to support the
microphone in the intended location and to accommodate
the desired range of motion. Overhead hardware, to allow
microphones to be suspended above a choir for example,
must often include a provision for preventing motion of
the microphone due to air currents or temperature
effects. Stand adapters or “clips” may be designed for
either permanent attachment or quick-release. “Shock
mounts” are used to isolate the microphone from
vibrations transmitted through the stand or the mounting
surface, such as a lectern.
Electrical accessories such as transformers and
phantom power supplies have already been described.
In addition, there are a variety of signal processors which
may be used directly in line with a microphone. These
can range from simple low- or high-frequency filters to
complete preamp/equalizer/limiter units, though most of
these functions are normally provided by the mixer and
subsequent elements of the audio chain.
Creative use of these accessories can allow
microphones to be placed almost anywhere, with good
acoustic results and with acceptable aesthetic appearance.
swivel adapter
accessory base
stereo mount
drum clamp
shock mount
high-pass filter
Microphone Accessories
desk stand
Audio Systems Guide for
A wireless microphone is actually a system consisting of
a microphone, a radio transmitter and a radio receiver. The
function of the microphone is unchanged and the function of
the transmitter and receiver combination is merely to replace
the microphone cable with a radio link. Although this objective
is simple, its accomplishment is not. However, with some
knowledge of the components and characteristics of wireless
microphone systems, and a clear idea of the intended
application, the selection and use of wireless microphones
can be made relatively straightforward.
The selection process for the microphone part of a
wireless system is exactly the same as for wired
microphones: the microphone must be matched to the
desired sound source and to the sound system. In this
case, the sound system consists not only of the devices
that make up the rest of the audio chain but the input to
the radio transmitter as well. Acoustically, wireless and
wired microphones behave identically: proper microphone
choice and placement is still necessary to get the best
sound and to avoid problems such as feedback.
Available microphone choices for wireless include
dynamic or condenser types, with flat or shaped frequency
response, omni- or unidirectional polar patterns, and a
variety of physical designs: lavalier, handheld, headworn,
etc. Almost any type of microphone may be used as part of
a wireless system, the notable exception being phantompower-only condensers. The choice depends on the
specific application.
For further discussion on
wireless microphone systems, see...
Radio System Diagram
1) The Microphone: How does sound enter the wireless
Shure’s Guide to the Selection
and Operation of Wireless
Microphone Systems
To download a PDF, go to...
Wireless System Components: Headset, Handheld Transmitter, Body-Pack Transmitter, Diversity Receiver and Lavalier Microphone
Audio Systems Guide for
2) The Transmitter: How does the microphone signal
become a radio signal?
The transmitter uses the audio signal from the
microphone to vary the frequency of a radio signal which
is broadcast to the receiver. The principle is called
“frequency modulation” or FM and is identical to that
used by commercial FM radio stations. Electrically, the
transmitter input must be compatible with the
microphone output both in level and impedance. The
transmitter input may also supply power for some
condenser microphone elements. The transmitter itself is
always battery-powered.
Physically, the transmitter takes one of two forms.
The first is a small box, called a “body-pack” or “beltpack”, that can be clipped to a belt or otherwise attached
to the user. The microphone connects to the body-pack
by means of a small cable. Some models have a
detachable cable that allows the transmitter to be used
with a variety of inputs. This form is most often used with
lavalier microphones but can also be connected to
electric musical instruments, head-worn microphones,
and even handheld types with appropriate cables. All
transmitters have a power on-off switch and many have a
mute switch to silence the microphone without turning off
the radio signal itself.
The second form is a transmitter that is built into the
cylindrical body of the microphone itself. This is used
almost exclusively for handheld vocal microphones and
results in a package only slightly larger than a
conventional wired microphone.
dropout: a temporary interruption of the radio signal. The
audible effect may range from a slight “swishing” noise to
a complete loss of sound.
Such dropouts may be experienced even at relatively
short distances by a mechanism called multipath
interference. Part of the signal from the transmitter (which
radiates in all directions) travels directly to the receiver,
but some of the signal is reflected to the receiver by metal
objects or other structures. When the “paths” of the direct
signal and of the reflected signal(s) are sufficiently
different, they will interfere with each other when they
combine at the receiver antenna. If the interference is
great enough, partial or complete cancellation of the
signal occurs, resulting in a dropout. It is similar to an
extremely severe “ghost” in television reception, and the
cure is the same: move the receiver antenna relative to
the transmitter. This is not usually practical since it is the
receiver antenna that is in a fixed location, while the
wireless microphone location is constantly changing.
This introduces the concept behind the second
wireless receiver configuration, called a diversity system.
A diversity receiver utilizes two separate antennas and
(usually) two separate radio circuits. When the two
antennas are separated by even a short distance, the
chance of a simultaneous interruption at both antenna
positions is extremely low. The key to the system is
additional “intelligent” diversity circuitry which
continuously monitors the received signals and takes
action according to the type of diversity employed.
3) The Receiver: How is the radio signal turned back
into an audio signal?
The receiver picks up the radio signal broadcast by the
transmitter and extracts or “demodulates” the audio signal
from it. Again the principle is the same as that of an ordinary
FM radio. The output of the receiver is electrically identical
to a microphone output and can be connected to any typical
microphone input in a sound system. Some receivers have
additional amplified outputs for headphones or auxiliary
connections to sound systems. Although most receivers
operate on ordinary AC power, battery-powered types are
available for portable use.
Wireless receivers are also designed in two different
configurations. The first is called non-diversity and consists
of a single antenna and a single radio circuit. An ordinary
FM radio is an example of a simple, non-diversity receiver.
Non-diversity receivers work well in many applications
but are subject to a phenomenon known as multipath
Receiver Illustration: Non-Diversity vs. Diversity
A simple, effective diversity technique is antenna
switching. It employs two antennas with one radio section.
The diversity circuitry switches antennas when it senses a
problem at the audio output. This type of system cannot
anticipate the result of switching and may thus switch
unnecessarily at times.
A more effective antenna switching technique called
predictive diversity evaluates the radio signal over time to
more accurately predict when a dropout is about to occur.
This avoids unnecessary switching and gives a more
consistent signal.
Audio Systems Guide for
Many diversity receivers are of the receiver switching
type. These utilize two antennas and two radio sections.
The diversity circuitry selects the better of the two received
signals (but only one) by means of an electronic switch.
If the switching is done quickly and quietly enough, the
result is nearly dropout-free performance, with minimal
audible side effects. Switching occurs only when it will
improve the signal.
The fourth diversity design is known as the receiver
combining type. This method takes advantage of the fact
that both of the received signals are usable much of the
time: in this case, using the signals from both antennas
yields better reception than using only one signal (as in the
switching type). The combining diversity circuitry adds the
signals in proportion to their relative strength. When both
are strong, the contribution from each signal is equal.
If one signal becomes weaker, its contribution is similarly
reduced. Finally, if a complete dropout occurs for one
signal, the receiver uses only the good signal. Since the
combining technique acts as a continuous balance control
rather than as a switch, it further reduces any audible
effects of diversity action. Again, it acts only when the signal
can be improved.
Historically, diversity receivers have always been
used for critical applications even though their cost was
somewhat higher. Today, the cost of wireless systems in
general and diversity systems in particular has decreased
to the point that diversity receivers are used in the majority
of higher-performance applications.
Since radio signals become weaker over
greater distances, a dropout can also occur
when the transmitter is very far from the
receiver antenna. Or even at shorter distances
when the radio signal from the transmitter
is blocked by obstacles such as walls,
equipment, or bodies.
An additional refinement in nearly all
recent wireless systems is some form of noise
reduction, or “companding”, in order to
decrease the inherent noise and increase the
limited dynamic range of radio transmission.
The word companding refers to the two steps
of the process: the signal is encoded
(compressed) in the transmitter before it is
broadcast and then decoded (expanded) in the
receiver in a complementary fashion. Although
the principle of companding is similar in all
wireless systems, significant differences
between models make it undesirable to mix
transmitters of one brand or series with receivers of
another brand or series.
Other aspects of wireless microphone systems that
must be considered in selection and use are operating
frequencies, antennas, and radio interference. All three
are especially important when planning the use of
multiple wireless systems in the same location.
Every wireless microphone system transmits and
receives on a particular radio frequency, called the
operating frequency. These frequencies may be
grouped into four bands: low-band VHF (49-72 MHz),
high-band VHF (169-216 MHz), low-band UHF
(450-806 MHz) and high-band UHF (806-952 MHz).
VHF stands for “Very High Frequency”, UHF stands for
“Ultra High Frequency”, and MHz stands for
“MegaHertz” or millions of cycles-per-second. Use of
these bands is regulated by the FCC (Federal
Communication Commission) and certain frequencies
within each band have been designated for use by
wireless microphones as well as by other devices. It
should be noted that while manufacturers must be
licensed by the FCC to sell wireless equipment, it is the
responsibility of the purchaser to observe FCC regulations
regarding their actual use.
Low-band VHF, particularly 49 MHz, is shared not
only by wireless microphones but by cordless telephones,
walkie-talkies, and radio controlled toys. For this reason, it
is almost never recommended for serious applications,
even though systems in this range are very inexpensive.
Frequency Band Illustration
Audio Systems Guide for
The high-band VHF range has been traditionally
used for a variety of applications and wireless systems of
various performance levels are still available in that range.
However, due to changes in the television broadcast band
and the continuing development of newer technologies,
the UHF band has become the primary choice for most
wireless applications. In particular, for operations
requiring 10 or more simultaneous systems UHF is the
only choice because of the greater spectrum available.
Finally, the cost of UHF systems is now on par with VHF.
Selection of operating frequency for a single wireless
system simply involves choosing a locally unused frequency.
Although some fixed frequency systems are still available,
tunable or “frequency-agile” systems are the norm for most
wireless equipment. To simplify operation even further, many
wireless receivers can now automatically scan for open
frequencies and set themselves accordingly.
Due to the nature of radio reception, it is not possible
for a single receiver to clearly pick up multiple
transmitters on the same frequency. Therefore, each
transmitter must be on a separate frequency and have a
corresponding receiver on that frequency. An additional
complication is that simultaneously operating systems,
even though they may be on different frequencies, may
still interfere with each other if those frequencies are not
carefully chosen. The rules for frequency coordination are
complex enough that computer programs are used to
calculate compatible sets of frequencies. Fortunately,
most frequency-agile wireless equipment is already
programmed with a compatible set of frequencies to allow
easy coordination of multiple systems in multiple
locations. However, it may still be desirable to consult the
equipment manufacturer for very complex setups.
Antenna selection and placement are very important
aspects of wireless system operation. There are a few
general rules about antennas to keep in mind.
First, maintain line-of-sight between the transmitter
and receiver antennas if possible. Avoid metal or other
dense materials between the two. This is particularly
important for UHF.
Second, keep the distance from transmitter to receiver as
short as practical. It is much better to have the receiver
near the transmitter and run the received audio
signal through a long cable than to transmit over long
distances or to use long antenna cables. The typical
signal strength of wireless systems is only 10 to 50 mw.
However, it is recommended to maintain a minimum
distance of at least 10 feet between a transmitter and its
receiver to avoid receiver overload.
Third, use the proper receiver antenna: a “1/4-wave”
antenna can be used if it is mounted directly to the
receiver. If the antenna is to be located at a distance
from the receiver, which will be necessary if the receiver
is mounted inside a metal enclosure or at a great
distance from the transmitter, a “1/2-wave” or other
high “gain” (sensitivity) antenna should be used.
Fourth, elevate receiver antennas and keep away from
large metal objects. This applies to receiver and
to transmitter antennas: do not coil or fold up trailing
wire antennas (or microphone cable antennas) on
body-pack transmitters. For diversity receivers, it is
recommended to angle the antennas apart by 45
degrees from vertical.
Fifth, use the proper antenna cable for remotely locating
receiver antennas: the correct impedance (usually 50
ohms) and the minimum length necessary (use low-loss
cable for longer cable runs).
Sixth, mount multiple antennas properly: at least 1/4
wavelength apart (about 17 inches for high-band VHF
or 4 inches for UHF systems). Use an amplified
antenna distribution system (sometimes called an
“active” antenna splitter) to minimize the number of
antennas and to reduce interference problems with
multiple receivers. This allows one antenna (or one pair,
for a diversity system) to be used with multiple receivers.
For further discussion on proper
antenna placement, see...
Shure’s Guide to Effective
Antenna Set-Up for
Wireless Systems
To download a PDF, go to...
The last aspect of the use of wireless microphone
systems, and perhaps the least predictable, is radio
interference. We’ve discussed potential interference from
other wireless systems operating on the same or nearby
frequencies, but what about other possible sources of
interference? The primary interfering sources are broadcast
television stations, both analog and DTV. For VHF this
includes TV channels 7-13 and for UHF this includes TV
channels 14-69. It is best to avoid using frequencies within
Audio Systems Guide for
Second, choose the microphone type. The application will
usually determine which microphone type is required:
a lavalier or clip-on type attached to clothing, or a headworn type, both for hands-free use; a handheld type for a
vocalist or when the microphone must be passed around
to different users; a connecting cable when an electrical
musical instrument or other non-microphone source is
used. Most handheld types are unidirectional, headworn
types may be omnidirectional or unidirectional, while
lavaliers are usually omnidirectional. Unidirectional
lavaliers are available for use when feedback or high
ambient noise is a problem.
Antenna Distribution
the bands of locally active TV channels (within 40-50
miles). High band VHF systems and UHF systems are
generally not subject to interference from radio stations,
amateur radio, pagers, or cellular telephones operated at a
distance. However, it is strongly advised to avoid using any
of these devices within a few feet of wireless microphone
receiver antennas.
Other local sources of interference may include the
following: any type of digital device such as computers,
digital signal processors, DAT or CD or DVD players;
electronic musical instruments such as organs or
synthesizers; neon or fluorescent light fixtures; large motors
and generators, etc. Any electrical device that uses high
voltage or high current is a potential source of radio
frequency interference. Again, keeping any of these local
sources at least a few feet away from wireless microphone
receivers will minimize the likelihood of problems.
The selection of a wireless microphone system
includes several steps, some of which are similar to wired
microphone selection. It should be remembered that while
wireless microphones cannot ultimately be as consistent
and reliable as wired microphones, the performance of
present systems can be very good, allowing excellent
results to be obtained. Following these steps will help
select the best wireless system(s) for your application.
First, define the application. In a worship facility system
this may be a wireless lavalier microphone for the
minister, a wireless handheld microphone for a singer,
or even a wireless pickup for a musical instrument.
Other applications could be meeting rooms or fellowship
halls and various indoor or outdoor events.
Third, choose the transmitter type. Again, the
application will specify the choice. All but the handheld
type will use some kind of body-pack transmitter. Some
body-pack transmitters, especially those with a multi-use
input connector, use a separate antenna wire while
others use the permanently attached microphone cable
as the antenna. A mute or audio on-off switch is desirable
to avoid turning off the transmitter power when the
microphone is not needed. Handheld types may have
external or internal antennas. Transmitter batteries may
be one of several types and their relative availability
should be considered. Also, power consumption of
transmitters varies, so be aware of expected battery life.
Fourth, choose the receiver type. The basic choice here
is diversity vs. non-diversity. For reasons mentioned in
the receiver section above, diversity receivers are
recommended for all but the most budget-conscious
applications. Non-diversity types will work well in many
situations, but the extra insurance (and usually extra
features) of the diversity receiver are worth the somewhat
higher cost. Other features of the receiver such as
headphone outputs, balanced outputs, different indicators,
and potential for battery power may be desirable.
Fifth, determine the number of systems to be used. This
should take into account future additions to the system:
choosing a system that can only accommodate a few
frequencies may someday be a limitation. It should also
take into account existing wireless systems with which
the new equipment must work.
Sixth, consult the manufacturer or a knowledgeable
professional about frequency selection to integrate
the planned number of systems. This must be done
for any multiple system installation and should be
done for even single systems to avoid potential
interference problems.
Audio Systems Guide for
Once the wireless system(s) choice is made, and the
equipment is correctly installed, proper use is necessary
for satisfactory performance.
Good practice with any wireless system is to check out
the system ahead of performance time, with all other
systems and devices on. This will reveal potential problems
that were not apparent in a wireless-system-only test.
Receivers are equipped with a “squelch” circuit; this
sets the basic sensitivity of the receiver to avoid picking up
interfering signals, or background radio noise, when the
transmitter is turned off or if a dropout occurs. Though
most are automatic, a few are adjustable and should be
adjusted according to the manufacturer’s instructions.
Once the system is on, use the “mute” or “mic” switch
to turn off the audio if necessary. Do not turn off the
transmitter until after the event is over and/or the receiver
is turned off. This will avoid an “open” receiver, which can
pick up other radio signals that may be present. Some
wireless systems are equipped with special squelch circuits
that do allow transmitters to be turned off without any noise
or interference problems. However, it is still recommended
to mute unused receiver channels in the sound system.
Finally, always use fresh batteries of the correct type
in the transmitter! Most manufacturers recommend only
alkaline or lithium type batteries for adequate operation.
Avoid rechargeable batteries: their actual voltage is usually
less than stated, and they may not operate satisfactorily in
a wireless transmitter. In addition, the actual operating
time of a rechargeable battery is usually much less than
an alkaline type.
A Note on DTV:
As of June 12, 2009 the United States has completed
the transition from analog television broadcast to digital
television broadcast (DTV). All full power analog TV stations
have given way to full power digital TV stations. While an analog
television signal consisted of just three discrete frequencies
in a 6 MHz band, a DTV signal occupies the entire 6 MHz
band. To avoid interference and comply with FCC regulations,
it is always recommended to avoid using frequencies that are
occupied by a local TV channel.
In addition to the replacement of analog television
broadcast with DTV broadcast, the overall television broadcast
band has been reduced from TV channels 2-69 to TV
channels 2-51. The range from 698 MHz to 806 MHz
(formerly TV channels 52-69) has been reallocated by the
FCC for use by various telecommunications services and
by public safety services. Newer wireless systems avoid this
“700 MHz” band but any older system that operates in this
band can no longer be legally used in the US.
Other wireless systems:
Two other wireless systems that may be found in
worship applications are assistive listening systems and
in-ear monitor systems.
Assistive Listening Systems are generally used to
provide improved sound to individuals with hearing
impairments. They may also be used to provide
simultaneous translation of the service into other
languages. They consist of a single transmitter and as
many receivers as are required by members of the
congregation. The transmitter is about the same size as a
typical wireless microphone receiver with an attached
antenna and is AC-powered. It is usually located in the
sanctuary where it can broadcast throughout the room.
The receivers are small, battery operated packs with an
attached earpiece or, in some models, a coil that can work
with the user’s hearing aid. These systems are FM radio
types and operated in the 72 Mhz band or 216 Mhz band
which are reserved specifically for them. No license is
required. The sound quality of Assistive Listening Systems
is usually optimized for speech intelligibility and is typically
monophonic. The source is usually a feed of the overall
mix from the main sound system.
Example of an Assistive Listening System
An alternative technology uses infrared transmitters
and receivers. Again, a single transmitter is used with
multiple receivers. The transmitter is a panel covered with
multiple IR (infrared) emitters approximately one foot
square and is also AC-powered. It is usually placed at an
elevated location at the front of the sanctuary where
listeners facing forward can see it. The receivers are
sometimes a small clip-on pack with an IR sensor at the
top or occasionally a headset with an attached IR sensor.
Since these are not radio systems, there is no concern for
frequency, licensing, or radio interference. The only
operating concern is to avoid strong, direct sunlight on the
receiver IR sensors.
Audio Systems Guide for
Assistive Listening Systems are a reliable and
relatively inexpensive technology, widely used in worship
facilities, theaters, and schools. In fact, the Americans
with Disabilities Act (ADA) requires their use in many public
facilities. Receivers are usually made available to people at
the worship facility for each use, but they are affordable
enough that many individuals purchase their own
receivers. Since the transmitters are fairly standardized,
they can often be used at many different locations.
Another wireless technology that has applications in
some worship facilities is the personal monitor system.
These systems are used to provide monitoring or foldback
directly to the ears of a performer. The system parts are
essentially the same as an Assistive Listening System: an
AC-powered FM transmitter, a battery-powered body-pack
receiver and earpieces. However, in-ear monitor systems
are engineered to provide full-range, high-fidelity, stereo
sound to listeners with normal hearing. Most of them
operate in the UHF band which allows multiple system
use and freedom from most radio interference. In addition,
the ear-pieces are designed to seal out ambient sound to
provide greater control of the foldback mix and a fair
degree of hearing protection.
By replacing traditional monitor loudspeaker
systems, in-ear monitors also eliminate many of the
problems associated with these systems. These problems
include monitor feedback, hearing damage from loud
stage sound, and monitor “splash” or interference with
the main sound system. In addition to these acoustic
benefits, the bulk and expense of monitor speaker boxes,
power amplifiers, and cables are also removed.
The source(s) for personal monitor systems is usually
a combination of auxiliary mix outputs and/or direct
channel outputs depending on the requirements of the
listener. It is possible to customize a different mix for
individual performers if each has his or her own
transmitter/receiver. These systems are easily integrated
with conventional mixers or dedicated monitor consoles.
Historically, personal monitor systems have been very
expensive and were used only by major touring
companies. More recently, these systems have become
comparably priced to conventional monitor systems and
their use is becoming more widespread.
For further discussion on selection and operation
of personal monitor systems, see...
Shure’s Guide to the Selection
and Operation of Personal
Monitor Systems
To download a PDF, go to...
Example of a Personal Stereo Monitor System, including Transmitter, Earphones, and Body-pack Receiver
Audio Systems Guide for
The reasons for using an automatic microphone
system relate to the behavior of multiple microphone
systems. Each time the number of open or active
microphones increases, the system gain or volume also
increases. The effect of this is greater potential for
feedback as more microphones are added, just as if the
master volume control were being turned up. In addition,
unwanted background noise increases with the number
of open microphones. Here, the effect is a loss of
intelligibility as the background noise level rises closer to
the level of the desired sound. (See Appendix Two:
Potential Acoustic Gain)
Examples of Automatic Microphone Mixers (shown front and back)
The solution is to activate microphones only when
they are addressed and to keep them attenuated or
turned down when not being addressed. In addition,
when more than one microphone is addressed at a time,
the system volume must be reduced appropriately to
prevent feedback and insure minimum noise pickup.
An automatic microphone system is comprised of a
special mixer and an associated group of microphones.
The function of an automatic microphone system is
twofold: to automatically activate microphones as needed
and to automatically adjust the system volume in a
corresponding manner. In some systems, ordinary
microphones are used and all of the control is provided by
the mixer. In others, special microphones are integrated
with the mixer to provide enhanced control.
There are several techniques used to accomplish
channel activation or “gating” in an automatic microphone
system. In most systems, a microphone is gated on when
the sound that it picks up is louder than some “threshold”
or reference level. When the sound level falls below the
threshold, the microphone is gated off. This threshold may
be fixed, adjustable, or even automatically adjustable. In any
case, the threshold should be set so that the microphone is
not activated by background noise but will be activated by
normal sound levels.
Traditional threshold systems distinguish between
background noise and the desired sound only by level.
However, if background noise becomes sufficiently loud,
it may activate microphones unless the threshold is
adjusted to a higher level. Subsequently, if the
background noise decreases, normal sounds may fail to
gate the microphones on unless the threshold is lowered
as well. Threshold adjustment is critical to automatic
microphone systems of this type.
Some recent automatic mixers incorporate noise
adaptive threshold circuitry. These have the ability to
distinguish steady signals such as background noise from
rapidly changing signals like speech. They can
automatically and continuously adjust individual channel
thresholds as ambient noise conditions change. In
addition, some designs can recognize that the same signal
is being picked up by more than one microphone. In that
case, only the channel with the strongest signal is activated.
This prevents both microphones from being activated
when a talker is in between two microphones for example.
Certain other automatic systems, with integrated
microphones, can actually sense the location of the sound
source relative to the ambient noise and activate
microphones only when the sound comes from the
desired direction. These “directional gating” systems do
not require any threshold adjustments.
There is another circuit within every automatic mixer
that continuously senses the number of open
microphones (NOM) and adjusts the gain of the mixer
accordingly. With a properly functioning automatic
system, if each individual microphone is adjusted to a
level below the feedback point, then any combination of
microphones will also be below the feedback point.
Many automatic microphone mixers have additional
control circuitry, often in the form of logic connections.
These are electrical terminals that can be used for a
variety of functions, including: microphone status
indicators, mute switches, loudspeaker attenuation, and
the selection of “priority” channels. Some automatic
Audio Systems Guide for
mixers have an adjustable “off attenuation” control:
instead of gating the microphone completely off, it can be
“attenuated” or turned down by some finite amount, to
make the gating effect less noticeable in certain
applications. Another control included on some units is an
adjustable “hold time”: when the desired sound stops, the
channel is held on for a short time to avoid gating the
microphone off between words or short pauses. In
addition, a function which locks on the last microphone
activated insures that at least one microphone is on, even
if no one is speaking. Finally, most automatic mixing
systems are able to be expanded by adding individual
channels and/or by linking multiple mixers together to
control large numbers of microphones simultaneously.
An automatic microphone system should be
considered whenever multiple microphones (four or
more) are being used, particularly if the sound system is
intended to run hands-free, that is, without a live operator.
This is often the case not only in the worship facility itself
but in fellowship halls, conference rooms, and auditorium
systems. Microphones should be selected and placed
according to the normal guidelines (integrated systems
require a microphone choice from the selection available
for those systems). It is recommended that the
manufacturer or a qualified installed sound professional
be consulted on the details of a particular automatic
microphone system.
Signal Processors:
Equalizers and Feedback Control
Signal processors fall into three main categories
based on which property of the audio signal they affect:
equalizers affect frequency response, dynamics
controllers affect amplitude, and delays affect time
properties such as phase. Each of these can be useful in
the operation of microphones but equalizers are of
particular interest because of their potential use in
feedback control.
Feedback is a very frequency-dependent phenomenon.
Because it occurs first at peaks in the overall sound
system frequency response, equalization of the response
may significantly affect the onset of feedback. System
response peaks may be due to many factors including
system components, transducer location, or room
acoustics. In principle, the system response must be
reduced at those frequencies which trigger feedback.
The goal is to allow the system to operate at higher gain
without ringing or feedback.
Low Cut: -6dB/octave below 125 Hz
High Cut: -6dB/octave above 2 kHz
Low Cut and High Cut Filters
Equalizers are frequency-dependent filters that fall
into several categories based on the characteristics of the
filters and their adjustment. Hi-cut and lo-cut (or,
alternately, lo-pass and hi-pass) filters progressively
attenuate or reduce all frequencies above (or below) a
certain cutoff frequency. That is, the attenuation increases
with frequency further above (or below) the cutoff
frequency. The cutoff frequency may be adjustable:
down to 5000Hz for hi-cut and up to 500Hz for lo-cut.
The “slope” or rate of attenuation may also be adjustable
from a minimum of 6dB/octave to as steep as
24dB/octave. Hi-cut and lo-cut filters are used to reduce
the bandwidth or frequency range of the signal to remove
unwanted high frequency or low frequency sounds such
as hiss or rumble.
Shelving equalizers allow low frequencies (or high
frequencies) to be cut or to be boosted. The cut or boost
is not progressive: it is the same at all frequencies below
(or above) the filter frequency. The response curve looks
somewhat like a shelf above or below the filter
frequency. The amount of cut or boost is adjustable
typically up to ± 15dB. The filter frequency is usually
fixed: about 250Hz and below for low frequencies, about
8000Hz and above for high frequencies. Shelving
equalizers are used for general response shaping at low
and high frequencies. They are the type of filter used as
“bass” or “treble” tone controls.
Low Shelf: -10dB below 125 Hz
High Shelf: -10dB above 2 kHz
Shelving Equalizers
Audio Systems Guide for
Bandpass equalizers allow frequencies within a
certain band or range to be cut or boosted. They are
classified according to their bandwidth and/or according to
the number of filters employed. Bandwidth is usually given
as a fraction of an octave (an octave represents a doubling
of frequency such as 400Hz-800Hz or 4000Hz-8000Hz).
For example, a midrange tone control is a single bandpass
filter with a 1 octave bandwidth designed to affect the
frequency range between a bass control and a treble
control, typically 500Hz-1000Hz. Again, the range of cut
or boost is typically up to ± 15dB. Bandpass filters have
fixed frequency and fixed bandwidth.
Sets of multiple bandpass filters are used for more
precise overall response shaping. When vertical slide
controls are used for adjustment these are called graphic
equalizers because the shape of the resulting response
curve is visually approximated by the control positions.
Graphic equalizers also have fixed frequency and fixed
bandwidth. Typical variations are 1 octave (8-10 bands),
1/2 octave (12-15 bands), and 1/3 octave (27-31 bands).
The narrower the bandwidth, the more filters are available
and the more precise the adjustment capability.
Low: +6dB @ 100Hz, 1/3 octave
Middle: -12dB @ 1kHz, 1/3 octave
High: +6dB @ 10k kHz, 1/3 octave
Graphic Equalizers
A set of bandpass filters whose frequency and
bandwidth can also be adjusted is called a parametric
equalizer because all of its “parameters” are adjustable.
Parametric equalizers can be “tuned” to any desired
frequency, adjusted to a suitable bandwidth, and boost or
cut as needed. They typically have a frequency range of
20-20,000Hz, a bandwidth range of 1/10 to 2 octaves,
and cut or boost of ± 15dB. Most parametric equalizers
have at least 3-5 independent filters, though some
midrange controls on mixing consoles are actually a
parametric filter. Parametric equalizers can provide very
precise frequency response shaping.
A special type of parametric filter is the “notch” filter.
It has both variable frequency and bandwidth but is used in
a “cut only” mode, typically down to -18dB. In addition, the
Low: 1/40 octave, -18dB @ 30Hz
Middle: 1/3 octave, -18dB @ 30Hz
High: 1 octave, -18dB @ 30 kHz
Parametric Equalizers
bandwidth of some notch filters can be as narrow as 1/40
octave. Notch filters are the most useful filters for feedback
control because they allow precise attenuation at any
frequency with minimal effect on adjacent frequencies. A
number of notch filters can be activated with very little
audible effect on overall sound quality.
Other equalizer types, even 1/3 octave graphics, have
a very noticeable effect on sound quality due to the
relatively large bandwidth of their filters, especially when
adjacent filters are used to reduce an “in between”
frequency. Similarly, use of hi-cut, lo-cut, or shelving
equalizers for feedback control can result in severe loss of
sound quality and is warranted only if the feedback is at
an extremely high or low frequency.
The use of an equalizer for feedback control is of
course limited to the degree that the feedback is due to
inequalities in system components or room acoustics. It
cannot compensate for badly-located microphones
and/or loudspeakers and certainly will not eliminate all
possibility of feedback. Poorly-designed systems or
unreasonable operating conditions can’t be fixed by even
the most powerful equalizer. Nevertheless, judicious use
of equalization can improve the feedback stability of a
well-designed system and may even allow a marginal
system to operate adequately.
The traditional approach to “ringing out” or
equalizing a sound system for feedback problems is to
gradually bring up the gain of the system until ringing or
feedback begins, identify the offending frequency, and
insert an appropriate filter until the feedback stops. The
process is repeated until either the desired gain is
reached (hopefully) or all the filters are used up. The most
difficult steps are: identifying the feedback frequency and
inserting the appropriate filter. Even very experienced
Audio Systems Guide for
sound engineers often have to rely on special equipment
to pinpoint the feedback frequency. In addition, the use
of parametric filters or notch filters is not very intuitive.
Recently, products called feedback controllers have
become available which “automatically” identify and
reduce feedback. They employ complex algorithms
(mathematical modeling techniques) to identify sustained
single frequency sounds and to deploy a notch filter of the
correct frequency and attenuation. These devices
typically have 5-10 filters that can be automatically set.
The filters are narrow enough (1/10 octave) that their
effect is not noticeable beyond the reduction of feedback.
A bypass switch is usually provided to compare the
equalized and un-equalized sound after the filters have
been set.
Some feedback controllers have other functions
built-in. These may include other types of equalizers such
as graphic or parametric, or other types of processors
altogether such as limiters and time delays. Certain
models offer computer interfaces for programming,
external control, and monitoring.
Though none of these devices can anticipate
feedback, they can still respond to the onset of feedback
or ringing more quickly and accurately than most human
operators. However, feedback controllers do not equalize
the system for good sound, merely for least feedback. It is
still up to the system designer and operator to insure the
desired sound quality.
Within the limitations mentioned earlier, such
automatic feedback controllers can be quite useful. They
can be used on the main sound system, the monitor
system, or even inserted on an individual channel. If the
sound system is normally controlled by an operator, they
can assist in the ringing out process. The operator merely
continues to slowly turn up the system level until the
major feedback frequencies have been identified and
“notched” out. Alternately, the device can be left active to
take care of feedback that may occur during unattended
system operation. However, these devices cannot
distinguish between sustained musical notes and
feedback. That is, a sustained note on a keyboard
or guitar may be interpreted as feedback and a
corresponding filter will be inserted at that frequency.
For this reason, it is recommended that for musical
performances these devices should be “locked” after
the initial ringing out. When used properly, feedback
controllers can improve gain-before-feedback by up to
6-10dB. Remember that more substantial improvements
can often be made just by repositioning microphones
or loudspeakers.
Example of a PC-controlled Feedback Reducer and Equalizer.
(Shure DFR22 shown on bottom of rack.)
For further discussion on
audio signal processors, see...
Shure’s Guide to the Selection
and Operation of Audio
Signal Processors
To download a PDF, go to...
Audio Systems Guide for
In order to select a microphone for a specific
application, it is first necessary to know the important
characteristics of the sound source(s) and of the sound
system. Once these are defined, a look at the five areas of
microphone specifications previously discussed will lead
to an appropriate match. Finally, correct placement and
proper use will insure best performance. In this section,
we will present recommendations for some of the most
common worship facility sound applications. The sound
system in the following examples is assumed to be of high
quality, with balanced low-impedance microphone inputs
and available phantom power.
Lectern Application
The desired sound source for a lectern microphone
is typically a speaking voice, though one may occasionally
be used for singing. Undesired sound sources that may
be present are nearby loudspeakers (possibly a central
cluster overhead), and ambient sound (possibly
ventilation or traffic noise, and reflected sound).
The basic performance requirements for a lectern
microphone can be met by either dynamic or condenser
types, so the choice of operating principle is often
determined by other factors, such as appearance.
In particular, the desire for an unobtrusive microphone is
better satisfied by a condenser design, which can
maintain high performance even in very small sizes.
Dynamic types are somewhat larger, but they do not
require phantom power.
To match the desired sound source (the voice), the
microphone must have a frequency response that covers
the vocal range (approximately 100Hz to 15kHz). Within
that range the response can be flat, if the sound system
and the room acoustics are very good; but often a shaped
response, with some presence rise, will improve
intelligibility. Above 15kHz and below 100 Hz, the
response should roll off smoothly, to avoid pickup of noise
and other sounds outside of the vocal range, and to
control proximity effect.
The choice of microphone directionality that will
maximize pickup of the voice, and minimize undesired
sounds, is unidirectional. This type will also reduce the
likelihood of feedback since it can be aimed toward the
talker and away from loudspeakers. Depending on how
much the person speaking may move about, or on how
close the microphone can be placed, a particular type
may be chosen: a cardioid for moderately broad, close-up
coverage; a supercardioid or a hypercardioid for
progressively narrower or slightly more distant coverage.
The electrical characteristics of the microphone
are primarily determined by the sound system: in this
case a balanced low-impedance type would match the
inputs on the mixer. Of course, this would be the
desired choice in almost all systems due to the
inherent benefits of lower noise and longer cable
capability. Sufficient sensitivity for lectern use can be
achieved by either condenser or full-size dynamic
types, since the sound source is fairly strong and
picked up from only a slight distance.
The physical design of a lectern microphone must
blend performance with actual use. The most effective
approach is a gooseneck-mounted type, which places the
microphone close to the sound source and away from
both the reflective surface of the lectern and noise from
the handling of materials on it. Another approach is the
use of a boundary microphone on the lectern surface, but
this method is limited by lectern design and by the
potential for noise pickup. As mentioned above, the
desired physical design may also suggest the operating
principle: the most effective small gooseneck or boundary
styles are condensers.
Audio Systems Guide for
The ideal placement of a lectern microphone is 8 to
16 inches away from the mouth, and aimed toward the
mouth. This will guarantee good pickup of the voice and
maximum rejection of unwanted sources. Locate the
microphone a few inches off-center and below the mouth
level. This will greatly reduce breath noise that occurs
directly in front of the mouth but will still provide good
coverage throughout the pickup angle of the microphone.
If possible, adjust the sound system to provide stable
operation with the lectern microphone at a nominal
distance of 12 inches. This will provide relatively less
change in level with changes in distance than if the
microphone is placed much closer, due to the inversesquare law. For example, with a nominal distance of 12
inches a change of ±6 inches results in a -3.5dB to +6dB
level change. For a nominal distance of only 6 inches, the
same distance change results in a -6dB to greater than
+18dB level change, a much larger variation. The
difference in potential acoustic gain between the two
nominal positions is 6dB.
For proper operation, the microphone must be
connected to the sound system with quality cables and
connectors. The correct phantom power should be
applied if a condenser microphone is used. Use a shock
mount to control mechanical noise from the lectern itself.
Some microphones are equipped with low-cut or low-end
roll-off filters, which may further reduce low-frequency
mechanical and acoustic noise. Goosenecks should be
quiet when flexed. It is strongly recommended that a pop
filter be placed on the microphone to control explosive
breath sounds, especially when using miniature
condenser types.
Good techniques for lectern
microphone usage include:
• Do adjust the microphone position for proper
• Do maintain a fairly constant distance
(8-16 inches).
• Don’t blow on microphone, or touch
microphone or mount, in use.
• Don’t make excess noise with materials
on lectern.
• Do speak in a clear and well-modulated
Altar Application
The desired sound source for an altar application is
a speaking (or sometimes singing) voice. Undesired
sounds may include direct sounds, such as choir, organ,
or loudspeakers and ambient noise sources, such as
building noise or the congregation itself.
A boundary microphone is the physical design best
suited to this application. Its use will minimize interference
effects due to reflections from the altar surface and will also
result in increased microphone sensitivity. A condenser
type is the most effective for this configuration, due to its
high performance and small size.
The frequency response should be optimized for the
vocal range and will benefit from a slight presence rise.
A unidirectional (typically cardioid) pattern will give the
broadest coverage with good rejection of feedback and
noise. A condenser microphone will provide the highest
sensitivity. Finally, the microphone should have a
balanced low-impedance output.
Good techniques for altar
microphone usage include:
• Do observe proper microphone placement.
• Do speak within coverage area of microphone.
• Don’t make excess noise with materials on altar.
• Do project the voice, due to greater microphone
Audio Systems Guide for
The microphone should be placed flat on the altar
at a distance of 2 to 3 feet and aimed towards the
normal position of the person speaking. It should be
located or aimed away from other objects and from
any local noise such as page turning. Unless there is
more than one distinct position to be covered, and
unless these positions do not violate the 3-to-1 rule,
use only one altar microphone.
The microphone should be connected and powered
(if a condenser) in the proper fashion. If the altar itself is
a source of noise or vibration, isolate the microphone from
it with a thin foam pad. A low-frequency filter may be a
desirable or even necessary feature. A pop filter is not
normally required. Do not cover the microphone with
heavy altar linens.
Handheld Vocal
The desired sound source for a handheld
microphone is a singing or speaking voice. Undesired
sounds may include other singers, musical instruments,
and various ambient sounds. In addition to the normal
loudspeakers, the sound system may also have nearby
“monitor” speakers aimed toward the singer.
Suitable microphone performance for this application
can be provided by dynamics or condensers. Due to
frequent handling and the potential for rough treatment,
dynamic microphones are most often used, though
durable condensers are available for high-performance
applications. The preferred frequency response is
shaped: vocal range, with presence rise for intelligibility
and low-frequency roll-off for control of proximity effect
and handling noise. These
microphones should always
be unidirectional: a cardioid
pattern is most common,
while supercardioid and
hypercardioid types may
be used in difficult noise
or feedback situations.
Balanced, low-impedance
output configuration is
standard, while adequate
sensitivity may be achieved
with dynamic or condenser
Handheld Vocal Application
types. Finally, the physical design is optimized for
comfortable handheld use, and generally includes an
integral windscreen/pop filter and an internal shock
mount. An on-off switch may be desirable in some
Positioning a handheld microphone at a distance
of 4 to 12 inches from the mouth (and aimed towards
it) will give good pickup of the voice. In addition,
locating the microphone slightly off-center, but angled
inward, will reduce breath noise.
With high levels of sound from adjacent musical
instruments or other singers, it may be necessary to
hold the microphone closer to the mouth. If the
distance is very short, especially less than 4 inches,
proximity effect will greatly increase the low-frequency
response. Though this may be desirable for many
voices, a low-frequency roll-off may be needed to
avoid a boomy sound. Additional pop filtering may also
be required for very close use.
Use of rugged, flexible cables with reliable
connectors is an absolute necessity with handheld
microphones. A stand or holder should also be provided
if it is desirable to use the microphone hands-free.
Good techniques for handheld
microphone usage include:
• Do hold microphone at proper distance for
balanced sound.
• Do aim microphone toward mouth and away
from other sound sources.
• Do use low frequency roll-off to control
proximity effect.
• Do use pop filter to control breath noise.
• Don’t create noise by excessive handling.
• Do control dynamics with voice rather than
moving microphone.
Audio Systems Guide for
The desired sound source for a lavalier microphone
is a speaking (or occasionally singing) voice. Undesired
sources include other speaking voices, clothing or
movement noise, ambient sound, and loudspeakers.
Balanced low-impedance output is preferred as
usual. Adequate sensitivity can be achieved by both
dynamic and condenser types, due to the relatively close
placement of the microphone. However, a condenser is
generally preferred. The physical design is optimized for
body-worn use. This may be done by means of a clip, a
pin, or a neck cord. Small size is very desirable. For a
condenser, the necessary electronics are often housed in
a separate small pack, also capable of being worn or
placed in a pocket. Some condensers incorporate the
electronics directly into the microphone connector.
Provision must also be made for attaching or routing the
cable to allow mobility for the user.
Placement of lavalier microphones should be as close
to the mouth as is practical, usually just below the neckline
on a lapel, a tie, or a lanyard, or at the neckline in the case
of robes or other vestments. Omnidirectional types may be
oriented in any convenient way, but a unidirectional type
must be aimed in the direction of the mouth.
Avoid placing the microphone underneath layers of
clothing or in a location where clothing or other objects may
touch or rub against it. This is especially critical with
unidirectional types. Locate and attach the cable to
minimize pull on the microphone and to allow walking
without stepping or tripping on it. A wireless lavalier system
eliminates this problem and provides complete freedom of
movement. Again, use only high-quality cables and
connectors, and provide phantom power if required.
Lavalier Application
A condenser lavalier microphone will give excellent
performance in a very small package, though a dynamic
may be used if phantom power is not available or if the size
is not critical. Lavalier microphones have a specially
shaped frequency response to compensate for off-axis
placement (loss of high frequencies), and sometimes for
chest “resonance” (boost of middle frequencies). The
most common polar pattern is omnidirectional, though
unidirectional types may be used to control excessive
ambient noise or severe feedback problems. However,
unidirectional types have inherently greater sensitivity to
breath and handling noise. In particular, the consonants
“d”, “t”, and “k” create strong downward breath blasts
that can result in severe “popping” of unidirectional
lavalier microphones. Placing the microphone slightly off
to the side (but still aimed up at the mouth) can greatly
reduce this effect.
Good techniques for lavalier
microphone usage include:
• Do observe proper placement and orientation.
• Do use pop filter if needed, especially with
• Don’t breathe on or touch microphone or cable.
• Don’t turn head away from microphone.
• Do mute lavalier when using lectern or altar
• Do speak in a clear and distinct voice.
Audio Systems Guide for
Again, the desired sound source for a headworn
microphone is a speaking or singing voice. Undesired
sources include other voices, instruments, ambient sound
and sound system loudspeakers.
Most headworn microphones are of the condenser
type because of their small size and superior sound
quality. A dynamic type can be used for speech-only
applications or if larger size is not an issue. For either type,
the frequency response is shaped for closeup vocal with
some presence rise. An omnidirectional polar pattern
is suitable for most applications, especially if the
microphone does not reach all the way in front of the
mouth. A unidirectional pickup is preferred in very high
ambient noise applications or to control feedback from
high volume monitor speakers. For proper operation,
unidirectional types should be positioned in front of or
directly at the side of the mouth and aimed at the mouth.
A windscreen is a necessity for a unidirectional headworn
Balanced low-impedance output is preferred for
hardwired setups but headworn types are often used in
wireless applications. In that case, the impedance and
wiring are made suitable for the wireless system.
For condenser types, the bodypack transmitter provides
the necessary bias voltage for the microphone element.
There are many different headworn mounting designs.
Most have a headband or wireframe that goes behind the
head, while a few are small enough that they merely clip
over the ear. In all cases, the microphone element is at the
end of a miniature “boom” or flexible arm that allows
positioning close to the mouth. Again, an omnidirectional
element can be positioned slightly behind or at the side of
the mouth while the unidirectional type should be at the
side or in front and aimed toward the mouth.
The main advantages of the headworn microphone
over the lavalier are greatly improved gain before
feedback and a more consistent sound level. The
increase in gain before feedback can be as much as 1520 dB. This is completely due to the much shorter
microphone-to-mouth distance compared to lavalier
placement. The headworn can nearly rival a handheld
type in this regard. In addition, the sound level is more
consistent than with the lavalier because the headworn
microphone is always at the same distance to the mouth
no matter which way the user may turn his head.
Headworn Application
Good techniques for headworn
microphone usage include:
• Do observe proper placement and orientation.
• Do adjust for secure and comfortable fit.
• Don’t allow microphone element to touch face.
• Do use pop filter as needed, especially for
• Do adjust vocal “dynamics” to compensate for
fixed mouth-to-microphone distance.
Audio Systems Guide for
Choir Application
The desired sound source is a group of singing
voices. Undesired sound sources may include the organ
or other musical instruments, loudspeakers, and various
ambient noise.
A condenser is the type of microphone most often
used for choir application. They are generally more
capable of flat, wide-range frequency response. The most
appropriate directional type is a unidirectional, usually a
cardioid. A supercardioid or a hypercardioid microphone
may be used for slightly greater reach or for more ambient
sound rejection. Balanced low-impedance output is used
exclusively, and the sensitivity of a condenser microphone
is desirable because of the greater distance between the
sound source and the microphone.
The physical design of the microphone for choir
pickup should lend itself to some form of overhead
mounting. It may be supported by its own cable or by
some other fixture, such as a stereo microphone mount.
Finally, it may be a full-size microphone or a miniature
type for unobtrusive placement. Application of choir
microphones falls into the category known as area
coverage. Rather than one microphone per sound
source, the object is to pick up multiple sound sources (or
a large sound source) with one (or more) microphone(s).
Obviously, this introduces the possibility of interference
effects unless certain basic principles (such as the
“3-to-1 rule”) are followed, as discussed to the right.
For one microphone picking up a typical choir, the
suggested placement is a few feet in front of, and a few
feet above, the heads of the first row. It should be
centered in front of the choir and aimed at the last row.
In this configuration, a cardioid microphone can cover
up to 15-20 voices, arranged in a rectangular or
wedge-shaped section.
For larger or unusually shaped choirs, it may be
necessary to use more than one microphone. Since
the pickup angle of a microphone is a function of its
directionality (approximately 130 degrees for a
cardioid), broader coverage requires more distant
placement. As choir size increases, it will eventually
violate the cardinal rule: place the microphone as
close as practical to the sound source.
In order to determine the placement of multiple
microphones for choir pickup, remember the following
rules: observe the 3-to-1 rule; avoid picking up the same
sound source with more than one microphone; and
finally, use the minimum number of microphones.
For multiple microphones, the objective is to divide
the choir into sections that can each be covered by a
single microphone. If the choir has any existing physical
divisions (aisles or boxes), use these to define basic
sections. If the choir is grouped according to vocal
range (soprano, alto, tenor, bass), these may serve
as sections.
Audio Systems Guide for
Good techniques for choir
microphone usage include:
• Do place the microphones properly.
• Do use the minimum number of microphones.
• Do turn down unused microphones.
• Do let the choir naturally “mix” itself.
• Don’t “over-amplify” the choir.
• Don’t sing “at” the microphone.
Microphone Positions - Side View
If the choir is a single, large entity, and it becomes
necessary to choose sections based solely on the coverage
of the individual microphones, use the following spacing: one
microphone for each lateral section of approximately 8 to 12
feet. If the choir is unusually deep (more than 5 or 6 rows),
it may be divided into two vertical sections of several rows
each, with aiming angles adjusted accordingly. In any case,
it is better to use too few microphones than too many.
It is very important to locate choir microphones as far
away from loudspeakers as possible. Be aware of the rear
pickup of supercardioid and hypercardioid types when aiming
microphones. Try to avoid pickup of organ pipes or speakers
in the choir loft. And, of course, keep microphones away from
other noise sources such as air ducts.
Once overhead microphones are positioned, and the
cables have been allowed to stretch out, they should
be secured, if necessary, to prevent turning or other
movement by air currents or temperature changes. Fine
thread or fishing line will accomplish this with minimum
visual impact. Use only the highest-quality cables and
connectors, particularly if miniature types are specified.
0.6 - 1m
(2 - 3 ft)
• Do sing in a natural voice.
The use of choir microphones is governed, to some
extent, by the intended destination of the sound. In general,
high-level sound reinforcement of a choir within the main
body of the worship facility is not recommended. In fact, it
is not possible in most cases, unless the choir itself is isolated
from the main body of the worship facility. Use of area
pickup microphones in the same acoustic space as area
coverage loudspeakers results in severe limitations on
gain-before-feedback. The best that can be done in this
circumstance is low-level reinforcement in the immediate
area, and, possibly, medium-level reinforcement to distant
areas, such as under balconies or in foyers. Destinations
such as isolated listening areas, recording equipment, or
broadcast audiences can receive higher levels because
feedback is not a factor in these locations.
Many older worship facilities are very reverberant spaces,
which provide natural, acoustic reinforcement for the choir,
though sometimes at the expense of speech intelligibility.
Some modern architecture has been designed to provide a
less reverberant space, both for greater speech intelligibility
and to accommodate modern
forms of music. This results
in a greater reliance on electronic reinforcement. However,
it is still not practical (and
probably not aesthetically
advisable) to make a choir of
20 sound like a choir of 200.
The sound system (and the
microphones) can provide
some useful enhancement,
2.5 - 3.5m
but a large acoustically dead
(8 - 12 ft)
worship facility simply requires
a large live choir.
Choir Microphone Positions - Top View
Audio Systems Guide for
The desired sound source for a congregation
microphone is a group of speaking or singing voices.
Undesired sources are usually the sound system
loudspeakers and various ambient sounds.
Condensers are the choice for highest-quality sound at
a distance. A flat, vocal-range frequency response is
usually desirable, with a unidirectional polar pattern to
minimize pickup of unwanted sound. The electrical output
should be balanced low-impedance, and the physical
design should accommodate overhead mounting, by cable
or other fixture. The microphone may be either full-size or
miniature, depending on visual requirements.
Since this application of microphones is another
example of area coverage, the placement should be in front
of, above, and aimed toward the faces of the congregation.
Though similar in concept to the choir example, fewer and
more distant microphones may be used to pick up the
overall ambience of the congregation.
A particular method that is sometimes suggested for
overhead placement is a ceiling-mounted microphone,
usually a boundary microphone. This position should be
used with caution, for two reasons: first, it often places the
microphone too far from the desired sound source,
especially in the case of a high ceiling. Second, the
ceiling, in buildings of modern construction, is often an
extremely noisy location, due to air handling noise,
lighting fixtures, and building vibration. Remember that a
microphone does not “reach out” and “capture” sound:
it can only respond to the sound in its immediate vicinity.
If this local soundfield is louder than the distant sound
from below, there is no hope of picking up a usable sound
with a ceiling-mounted microphone.
Congregation area microphones are used exclusively
for recording, broadcast, and other isolated destinations.
It is never intended to be mixed into the sound system for
local reinforcement. If it is desired to reinforce an
individual member of the congregation, it can only be
done successfully with an individual microphone in the
congregation: a stand-mounted type that the member
can approach or a handheld type (wired or wireless) that
can be passed to the member.
Good techniques for congregation
microphones include:
• All the techniques for “Choir Microphone
Usage” (see previous page).
• Do use only at a level sufficient to add
ambience to a recording.
• Don’t mix area microphones into the sound
reinforcement system.
Musical Instruments
A tremendous variety of musical instruments is used
in current worship facility services. In fact, almost any
instrument that exists may be used: from classical
symphonic instruments, to modern electronic
instruments, to historical and ethnic instruments of any
description. Presented here will be techniques for three
musical instruments that are widely used today: the
acoustic guitar, the piano, and the organ. Use of
microphones with many other instruments is discussed
in Shure’s Guides to Microphone Techniques. See back
cover for more details on ordering these publications.
Piano and Guitar Application
In each of these examples, the desired sound source
is the musical instrument itself. Possible undesired sound
sources include other nearby instruments, singers,
loudspeakers, and the usual ambient noise sources.
Audio Systems Guide for
Since accurate, wide-range reproduction of musical
instruments is the goal, the use of condenser
microphones is often preferred, although certain
instruments, such as drums, can be well suited to
dynamic types. The frequency response is usually flat and
wide-range, especially for organ or piano. Unidirectional
designs are preferable, to minimize pickup of undesired
sound. Again, balanced low-impedance models are the
best choice. Because close microphone placement is
used, dynamic and condenser types have suitable
sensitivity for general sound reinforcement. However,
condensers are recommended for highest-quality sound.
The physical design, though, can vary widely in
instrument applications, depending on the desired
placement and use.
Good techniques for an acoustic
instrument microphone usage include:
• Do experiment with placement for best sound.
• Do maintain a constant distance.
• Do use a shock mount if stage noise is present.
• Don’t position microphone where it may be
struck by instrument.
• Don’t allow voice to be picked up by instrument
The acoustic guitar is
a relatively small sound
source that can normally
be picked up quite well by
only one microphone.
Since most of the sound
comes from the sound
hole and the top of
the guitar, a microphone
positioned in front of the
guitar can get an excellent
Guitar Application
overall sound. This sound
will vary, however, as a function of the microphone distance
from the sound hole and its orientation to the top of the
guitar. The sound will be louder and “bassier” the closer to
the sound hole; softer and thinner farther away. Proximity
effect will also increase the bass response at close distances.
A full-size microphone can be positioned on a stand to
give the desired sound. An alternate approach is to mount a
miniature microphone directly on (or in) the guitar by means
of a clip or holder. This keeps the microphone at a constant
distance, and allows freedom of movement for the performer,
especially if used with a wireless transmitter. In either case,
care must be taken to position the microphone to avoid
interfering with the player.
The piano is a
relatively large acoustic
source whose sound
soundboard, the strings,
and reflections from
the lid and other body
parts. Although the
piano is normally heard
at a distance, it is not
feasible to use a distant
microphone on a piano
Piano Application
for sound reinforcement, due to gain-before-feedback limitations. Placing the
microphone close to or inside of the piano is the normal
procedure. The resulting sound is not entirely natural, but
careful microphone placement can yield very good results.
Depending on placement, several microphone
physical designs may be used. A conventional, full-size
microphone can be positioned close to or inside of the
piano (with the lid open) using a stand and boom.
A position over the treble strings will yield a bright sound
while a position over the middle or low strings will
correspond to a bassier sound. A sharper attack is heard
near the hammers, while a softer sound is heard farther
away. For greater isolation from other sounds and to reduce
feedback, a boundary microphone is sometimes attached
to the underside of the lid, which is then partially or
completely closed.
Since very close microphone placement may not pick
up the full sound of a large source, it is sometimes desirable
to use two (or more) microphones, especially for stereo reproduction. In this case, microphone placement becomes
more subjective due to the possibility of interference effects.
A good starting point is one microphone over the treble
strings and a second over the bass strings. This will often
produce a more balanced sound, and does allow a greater
range of control. However, some experimentation will be
necessary to get the best sound from a specific instrument
in a specific room.
Audio Systems Guide for
Good techniques for piano
microphone usage include:
• Do experiment with placement for best sound.
• Do adjust lid for best sound and/or isolation.
• Do use shock mounts if vibration is a problem.
• Do listen for interference effects with multiple
• Don’t allow voice to be picked up by
instrument microphone.
to place microphones to pick up the organ sound only.
This will require that a microphone be placed close enough
to each of the main locations of pipes or tone cabinets so
that the microphone hears primarily the local organ sound,
rather than the ambient or room sound.
This may involve several microphones, depending
on the number and location of sound sources.
Individual placement should be done according to the
guidelines given earlier with respect to choir pickup,
although it may be possible to mount microphones on
stands in organ galleries as well as overhead in front of
exposed ranks. In any case, some experimentation
with microphone positioning, and careful mixing of
microphone signals will be necessary to get a full,
balanced sound.
Non-Sanctuary Applications
The organ is potentially the largest sound source in
some worship facility applications. However, pipe organs
and large electronic organs are not normally reinforced by
sound systems, but rather are picked up for recording or
broadcast purposes. Since the organ is also the widest
range instrument, the careful placement of high-quality
microphones is essential for best results.
A large organ produces sound from many ranks of
pipes, or, for an electronic type, from a number of tone
cabinets. Since it is not possible to use microphones on
individual pipes or loudspeakers, some type of area
coverage must be employed. Often, the groups of pipes or
tone cabinets are widely separated, sometimes even
located on opposite sides of the worship facility, as is the
case with antiphonal ranks. This will require a decision on
the goal of the sound.
If the goal is to reproduce the sound as heard by a
listener in the house of worship, one or two (for stereo)
microphones can be positioned in the body of the worship
facility, over the congregation, and aimed toward the main
organ ranks. This will pick up a representative organ sound,
with a high proportion of room sound (ambient sound), as
well as the sound from the choir and from the sound
system itself. If the room has reasonably good acoustics,
and if the level of the organ is well-balanced with both the
choir and the sound system, this is the simplest and most
effective way to simulate being in the religious facility.
In some arrangements, the choir microphones themselves
will pick up a suitable organ sound.
On the other hand, if the goal is to reproduce a concert
organ performance that does not rely heavily on the room
acoustics, or if it is desired to control the level of the organ
independently of the choir and other sounds, it is necessary
Today, the life of worship facilities extends far
beyond the sanctuary, in the form of classes, meetings,
plays, social events, and fund-raising activities, both
indoors and out. Even the weekly service may not always
be held in the same location. Sound systems can play
an important role in all of these situations. While it is not
possible to detail microphone techniques for every
application, a few examples will show how to use some
of the ideas already presented.
Though most classrooms are not large enough to
require the use of a sound system, it is sometimes
necessary to record a class, or to hold a very large class
in an auditorium. In these cases, it is suggested that the
teacher wear a wireless lavalier microphone to allow
freedom of movement and to maintain consistent sound
quality. If it is desired to pick up the responses of students,
it is possible to use area microphones in a recording
application, but not with a sound system. A better
technique is to present questions at a fixed stand
microphone, or to pass a wired or wireless handheld
microphone to the student.
Meetings and conferences often involve a large
number of microphones in the same room. Use
unidirectional types, dynamic or condenser, and locate
them as close to the participants as practical. Observe the
3-to-1 rule and use as few microphones as necessary.
Usually, one microphone can cover two people. Boundary
types are very useful on tables if tabletop noise is low,
otherwise conventional types on short stands or
goosenecks should be used. Turn up microphones only
as needed. Due to the potential for feedback, noise and
interference from multiple microphones, it is suggested
that an automatic microphone mixing system be
Audio Systems Guide for
Microphone use for plays and other theatrical events
involves both individual and area coverage. Professional
productions usually employ wireless microphones for all
the principle actors. This requires a complete system
(microphone, transmitter, receiver) for each person, and
the frequencies must be selected so that all systems will
work together without interference. While it is possible to
purchase or rent a large number of wireless systems, it is
often more economical to combine just a few wireless
systems with area microphones for the rest of the
players. Use unidirectional boundary microphones for
“downstage” (front) pickup and unidirectional overhead
microphones for “upstage” (rear) pickup. Always use a
center microphone, because most action occurs at center
stage. Use flanking microphones to cover side areas but
observe the 3-to-1 rule and avoid overlapping coverage.
Turn up microphones only as needed.
Social events, such as dances or carnivals, generally
require only PA (public address) coverage. Use
unidirectional, handheld or stand mounted microphones.
A dynamic type is an excellent choice, because of its
rugged design. The microphone should be equipped with
an on-off switch if it is not possible to turn down the
microphone channel on the sound system. In any case,
turn up the microphone(s) only as needed.
A typical fund-raising activity is the bingo game.
Again, only PA coverage is needed. A unidirectional,
dynamic microphone mounted on a stand works very well
in this application. Alternatively, the caller may choose a
lavalier or headworn type to permit freedom of movement.
A convenient addition is a handheld wireless microphone
for the person who verifies the cards in the audience.
For further discussion on audio systems
for theater performances, see...
Shure’s Audio Systems Guide
for Theater Performances
To download a PDF, go to...
Outdoor use of microphones is, in some ways, less
difficult than indoor. Sound outdoors is not reflected by
walls and ceilings, so reverberation is not present. Without
reflected sound, the potential for feedback is also reduced.
However, the elements of nature must be considered:
wind, sun, and rain. Because of these factors, dynamic
types are most often used, especially in the likelihood of
rain. In any case, adequate windscreens are a must.
Microphone principles are the same outdoors, so
unidirectional patterns are still preferred. Finally, because
of frequent long cable runs outdoors, balanced lowimpedance models are always recommended.
Example of Unidirectional Boundary Microphones Being Used to
Provide Area Coverage for an On-Stage Application.
Reference Information
3-to-1 Rule
When using multiple microphones, the distance between
microphones should be at least 3 times the distance from
each microphone to its intended sound source.
The dissipation of sound energy by losses due to sound
absorbent materials.
Active Circuitry
Electrical circuitry which requires power to operate,
such as transistors and vacuum tubes.
Room acoustics or natural reverberation.
The strength or level of sound pressure or voltage.
Audio Chain
The series of interconnected audio equipment used
for recording or sound reinforcement.
The solid conductive disk that forms the fixed half
of a condenser element.
A circuit that carries information by means of two equal
but opposite polarity signals, on two conductors.
Bidirectional Microphone
A microphone that picks up equally from two opposite
directions. The angle of best rejection is 90 degrees
from the front (or rear) of the microphone, that is,
directly at the sides.
Boundary/Surface Microphone
A microphone designed to be mounted on an
acoustically reflective surface.
Cardioid Microphone
A unidirectional microphone with moderately wide
front pickup (131 degrees). Angle of best rejection
is 180 degrees from the front of the microphone,
that is, directly at the rear.
Cartridge (Transducer)
The element in a microphone that converts acoustical
energy (sound) into electrical energy (the signal).
Clipping Level
The maximum electrical output signal level (dBV or
dBu) that the microphone can produce before the
output becomes distorted.
Close Pickup
Microphone placement within 2 feet of a sound source.
Comb Filtering
An interference effect in which the frequency response
exhibits regular deep notches.
Condenser Microphone
A microphone that generates an electrical signal when
sound waves vary the spacing between two charged
surfaces: the diaphragm and the backplate.
Audio Systems Guide for
Critical Distance
In acoustics, the distance from a sound source in a
room at which the direct sound level is equal to the
reverberant sound level.
Charge flowing in an electrical circuit. Analogous to the
amount of a fluid flowing in a pipe.
Decibel (dB)
A number used to express relative output sensitivity.
It is a logarithmic ratio.
The thin membrane in a microphone which moves in
response to sound waves.
The bending of sound waves around an object which is
physically smaller than the wavelength of the sound.
Direct Sound
Sound which travels by a straight path from a sound
source to a microphone or listener.
Distance Factor
The equivalent operating distance of a directional
microphone compared to an omnidirectional
microphone to achieve the same ratio of direct to
reverberant sound.
Distant Pickup
Microphone placement farther than 2 feet from the
sound source.
Dynamic Microphone
A microphone that generates an electrical signal when
sound waves cause a conductor to vibrate in a magnetic
field. In a moving-coil microphone, the conductor is a
coil of wire attached to the diaphragm.
Dynamic Range
The range of amplitude of a sound source.
Also, the range of sound level that a microphone
can successfully pick up.
Reflection of sound that is delayed long enough
(more than about 50 msec.) to be heard as a distinct
repetition of the original sound.
A material (such as Teflon) that can retain
a permanent electric charge.
Equalization or tone control to shape frequency
response in some desired way.
In a PA system consisting of a microphone, amplifier,
and loudspeaker, feedback is the ringing or howling
sound caused by amplified sound from the loudspeaker
entering the microphone and being re-amplified.
Flat Response
A frequency response that is uniform and equal at all
Audio Systems Guide for
The rate of repetition of a cyclic phenomenon such
as a sound wave.
Frequency Response Tailoring Switch
A switch on a microphone that affects the tone
quality reproduced by the microphone by means
of an equalization circuit. (Similar to a bass or treble
control on a hi-fi receiver.)
Frequency Response
A graph showing how a microphone responds to
various sound frequencies. It is a plot of electrical
output (in decibels) vs. frequency (in Hertz).
The lowest frequency component of a complex
waveform such as musical note. It establishes the
basic pitch of the note.
Amplification of sound level or voltage.
The amount of gain that can be achieved in a sound
system before feedback or ringing occurs.
Movable panels used to reduce reflected sound in
the recording environment.
Frequency components above the fundamental
of a complex waveform. They are generally multiples
of the fundamental which establish the timbre or
tone of the note.
A unidirectional microphone with tighter front pickup
(105 degrees) than a supercardioid, but with more rear
pickup. Angle of best rejection is about 110 degrees
from the front of the microphone.
In an electrical circuit, opposition to the flow of
alternating current, measured in ohms. A highimpedance microphone has an impedance of 10,000
ohms or more. A low-impedance microphone has an
impedance of 50 to 600 ohms.
Destructive combining of sound waves or electrical
signals due to phase differences.
Inverse Square Law
States that direct sound levels increase (or decrease)
by an amount proportional to the square of the
change in distance.
Freedom from leakage; the ability to reject
unwanted sounds.
Pickup of an instrument by a microphone intended to
pick up another instrument. Creative leakage is
artistically favorable leakage that adds a “loose” or “live”
feel to a recording.
Reference Information
Maximum Sound Pressure Level
The maximum acoustic input signal level (dB SPL) that
the microphone can accept before clipping occurs.
Needed Acoustic Gain is the amount of gain that a
sound system must provide for a distant listener to
hear as if he or she was close to the unamplified
sound source.
Unwanted electrical or acoustic energy.
Noise Cancelling
A microphone that rejects ambient or distant sound.
Number of open microphones in a sound system.
Decreases gain-before-feedback by 3dB every time
NOM doubles.
Omnidirectional Microphone
A microphone that picks up sound equally well from
all directions.
Output Noise (Self-Noise)
The amount of residual noise (dB SPL) generated by
the electronics of a condenser microphone.
Exceeding the signal level capability of a microphone
or electrical circuit.
Potential Acoustic Gain is the calculated gain that a
sound system can achieve at or just below the point
of feedback.
Phantom Power
A method of providing power to the electronics of a
condenser microphone through the microphone cable.
The “time” relationship between cycles of different waves.
Pickup Angle/Coverage Angle
The effective arc of coverage of a microphone, usually
taken to be within the 3dB down points in its directional
The fundamental or basic frequency of a musical note.
Polar Pattern (Directional Pattern, Polar Response)
A graph showing how the sensitivity of a microphone
varies with the angle of the sound source, at a particular
frequency. Examples of polar patterns are unidirectional
and omnidirectional.
The charge or voltage on a condenser microphone
Pop Filter
An acoustically transparent shield around a microphone
cartridge that reduces popping sounds. Often a ballshaped grille, foam cover or fabric barrier.
Reference Information
A thump of explosive breath sound produced when a
puff of air from the mouth strikes the microphone
diaphragm. Occurs most often with “p” and “b” sounds
(forward) and “d”, “t”, and “k” sounds (downward).
Presence Peak
An increase in microphone output in the“presence”
frequency range of 2,000 Hz to 10,000 Hz. A presence
peak increases clarity, articulation, apparent closeness,
and “punch.”
Proximity Effect
The increase in bass occurring with most unidirectional
microphones when they are placed close to an
instrument or vocalist (within 1 foot). Does not occur
with omnidirectional microphones.
Rear Lobe
A region of pickup at the rear of a supercardioid
or hypercardioid microphone polar pattern. A
bidirectional microphone has a rear lobe equal to
its front pickup.
The bouncing of sound waves back from an object
or surface which is physically larger than the wavelength
of the sound.
The bending of sound waves by a change in the density
of the transmission medium, such as temperature
gradients in air due to wind.
The opposition to the flow of current in an electrical circuit.
It is analogous to the friction of fluid flowing in a pipe.
The reflection of a sound a sufficient number of times
that it becomes non-directional and persists for some
time after the source has stopped. The amount of
reverberation depends on the relative amount of sound
reflection and absorption in the room.
A gradual decrease in response below or above some
specified frequency.
A rating given in dBV to express how “hot” the
microphone is by exposing the microphone to a specified
sound field level (typically either 94 dB SPL or 74 dB
SPL). This specification can be confusing because
manufacturers designate the sound level different ways.
Here is an easy reference guide: 94 dB SPL = 1 Pascal =
10 microbars. To compare a microphone that has been
measured at 74 dB SPL with one that has been
measured at 94 dB SPL, simply add 20 to the dBV
rating. E.g. -40 dBV/Pa = -60 dBV/microbar.
Shaped Response
A frequency response that exhibits significant variation
from flat within its range. It is usually designed to
enhance the sound for a particular application.
Audio Systems Guide for
Signal to Noise Ratio
The amount of signal (dBV) above the noise floor when
a specified sound pressure level is applied to the
microphone (usually 94 dB SPL).
Sound Chain
The series of interconnected audio equipment used for
recording or sound reinforcement.
Sound Reinforcement
Amplification of live sound sources.
Speed of Sound
The speed of sound waves, about 1130 feet per second
in air.
Sound Pressure Level is the loudness of sound relative
to a reference level of 0.0002 microbars.
Standing Wave
A stationary sound wave that is reinforced by reflection
between two parallel surfaces that are spaced a
wavelength apart.
Supercardioid Microphone
A unidirectional microphone with tighter front pickup
angle (115 degrees) than a cardioid, but with some rear
pickup. Angle of best rejection is 126 degrees from the
front of the microphone, that is, 54 degrees from the rear.
3-to-1 Rule
(See top of page 47.)
The characteristic tone of a voice or instrument; a
function of harmonics.
A device that converts one form of energy to another.
A microphone transducer (cartridge) converts acoustical
energy (sound) into electrical energy (the audio signal).
Transient Response
The ability of a device to respond to a rapidly changing
A circuit that carries information by means of one signal
on a single conductor.
Unidirectional Microphone
A microphone that is most sensitive to sound coming
from a single direction-in front of the microphone.
Cardioid, supercardioid, and hypercardioid microphones
are examples of unidirectional microphones.
Voice Coil
Small coil of wire attached to the diaphragm of a
dynamic microphone.
The potential difference in an electric circuit. Analogous
to the pressure on fluid flowing in a pipe.
The physical distance between the start and end of one
cycle of a soundwave.
Audio Systems Guide for
The decibel (dB) is an expression often used in
electrical and acoustic measurements. The decibel is a
number that represents a ratio of two values of a quantity
such as voltage. It is actually a logarithmic ratio whose
main purpose is to scale a large measurement range
down to a much smaller and more useable range. The
form of the decibel relationship for voltage is:
Reference Information
Appendix One:
The Decibel
Since the decibel is a ratio of two values, there must
be an explicit or implicit reference value for any
measurement given in dB. This is usually indicated by a
suffix on the dB. Some devices are measured in dBV
(reference to 1 Volt = 0 dBV), while others may be
specified in dBu or dBm (reference to .775V =
0dBu/dBm). Here is a chart that makes conversion for
comparison easy:
dB = 20 x log(V1/V2)
where 20 is a constant, V1 is one voltage, V2 is a reference voltage, and log is logarithm base 10.
What is the relationship in decibels between 100
volts and 1 volt? (dbV)
dB = 20 x log(100/1)
dB = 20 x log(100)
dB = 20 x 2 (the log of 100 is 2)
dB = 40
That is, 100 volts is 40dB greater than 1 volt.
What is the relationship in decibels between
.0001 volt and 1 volt? (dbV)
dB = 20 x log(.001/1)
dB = 20 x log(.001)
dB = 20 x (-3) (the log of .001 is -3)
dB = -60
That is, .001 volt is 60dB less than 1 volt.
If one voltage is equal to the other, they are 0dB
If one voltage is twice the other, they are 6dB
If one voltage is ten times the other, they are
20dB different.
Conversion Chart
Audio equipment signal levels are generally broken
into 3 main categories: Mic, Line, or Speaker Level. Aux
level resides within the lower half of line level. The chart
also shows at what voltages these categories exist.
One reason that the decibel is so useful in certain
audio measurements is that this scaling function closely
approximates the behavior of human hearing sensitivity.
For example, a change of 1dB SPL is about the smallest
difference in loudness that can be perceived while a 3dB
SPL change is generally noticeable. A 6dB SPL change
is quite noticeable and finally, a 10dB SPL change is
perceived as twice as loud.
Reference Information
Audio Systems Guide for
A p p e n d i x Tw o :
Potential Acoustic Gain
Potential Acoustic Gain (PAG)
vs. Needed Acoustic Gain (NAG)
The basic purpose of a sound reinforcement system
is to deliver sufficient sound level to the audience so that
they can hear and enjoy the performance throughout
the listening area. As mentioned earlier, the amount of
reinforcement needed depends on the loudness of the
instruments or performers themselves and the size and
acoustic nature of the venue. This Needed Acoustic Gain
(NAG) is the amplification factor necessary so that the
furthest listeners can hear as if they were close enough to
hear the performers directly.
The simplified PAG equation is:
To calculate NAG: NAG = 20 x log (Df/Dn)
PAG = 20 (log D1- log D2+ log D0- log Ds)- 10 log NOM- 6
Where: Df = distance from sound source to
furthest listener
Where: PAG = Potential Acoustic Gain (in dB)
Potential Acoustic Gain
Ds = distance from sound source to microphone
Dn = distance from sound source to
nearest listener
D0 = distance from sound source to furthest
log = logarithm to base 10
Note: the sound source may be a musical instrument, a
vocalist or perhaps a loudspeaker
The equation for NAG is based on the inverse-square
law, which says that the sound level decreases by 6dB
each time the distance to the source doubles. For
example, the sound level (without a sound system) at the
first row of the audience (10 feet from the stage) might be
a comfortable 85dB. At the last row of the audience (80
feet from the stage) the level will be 18dB less or 67dB.
In this case the sound system needs to provide 18dB of
gain so that the last row can hear at the same level as the
first row. The limitation in real-world sound systems is not
how loud the system can get with a recorded sound
source but rather how loud it can get with a microphone
as its input. The maximum loudness is ultimately limited
by acoustic feedback.
The amount of gain-before-feedback that a sound
reinforcement system can provide may be estimated
mathematically. This Potential Acoustic Gain involves the
distances between sound system components, the
number of open mics, and other variables. The system
will be sufficient if the calculated Potential Acoustic Gain
(PAG) is equal to or greater than the Needed Acoustic
Gain (NAG). Following is an illustration showing the key
D1 = distance from microphone to nearest
D2 = distance from loudspeaker to furthest
NOM = the number of open microphones
-6 = a 6 dB feedback stability margin
log = logarithm to base 10
In order to make PAG as large as possible, that is, to
provide the maximum gain-before-feedback, the following
rules should be observed:
1) Place the microphone as close to the sound
source as practical.
2) Keep the microphone as far away from the
loudspeaker as practical.
3) Place the loudspeaker as close to the audience
as practical.
4) Keep the number of microphones to a minimum.
Audio Systems Guide for
System will work: PAG>NAG
Reference Information
A p p e n d i x Tw o :
Potential Acoustic Gain
The NOM term in the PAG equation reflects the fact
that gain-before-feedback decreases by 3dB every time
the number of open (active) microphones doubles. For
example, if a system has a PAG of 20dB with a single
microphone, adding a second microphone will decrease
PAG to 17dB and adding a third and fourth mic will
decrease PAG to 14dB. This is why the number of
microphones should be kept to a minimum and why
unused microphones should be turned off or attenuated.
Essentially, the gain-before-feedback of a sound system
can be evaluated strictly on the relative location of
sources, microphones, loudspeakers, and audience, as
well as the number of microphones, but without regard to
the actual type of component. Though quite simple, the
results are very useful as a best case estimate.
In particular, the logarithmic relationship means that
to make a 6dB change in the value of PAG the
corresponding distance must be doubled or halved. For
example, if a microphone is 1 ft. from an instrument,
moving it to 2 ft. away will decrease the gain-beforefeedback by 6dB while moving it to 4 ft. away will
decrease it by 12dB. On the other hand, moving it to 6 in.
away increases gain-before-feedback by 6dB while
moving it to only 3 in. away will increase it by 12dB. This
is why the single most significant factor in maximizing
gain-before-feedback is to place the microphone as close
as practical to the sound source.
System will not work: PAG<NAG
Reference Information
Appendix Three:
Stereo Microphone Techniques
Audio Systems Guide for
An exception to the minimumelements 6 to 12 inches apart, and
number-of microphones rule is
at some angle relative to each other.
stereo sound pickup: at least two
In this method, the stereo image is
microphone elements are needed
a function not only of directionality
to pick up true stereo. These may
but also of distance. The result is
be in the form of either separate
good image width and accurate
standard microphones, or a single
image localization. Since there is
stereo microphone, combining the
a finite distance between the
elements in one housing. In either
microphones, and hence, some
case, the object of stereo
delay between the sounds picked
microphone application is to add
up, there may be some noticeable
the aspects of width and depth to
Example of Stereo Pick Up Technique Using
interference effects if the signals
the reproduced sound. This results
Two Cardioid Microphones
are combined monophonically.
in a more realistic image when
Spaced techniques may use
heard through a stereo sound system. There are many unidirectional or omnidirectional microphones. They
techniques used to accomplish this goal, but they may all are placed 3 to 10 feet apart and may or may not be
be categorized as follows: coincident, near-coincident, angled relative to each other. Here, the stereo image
or spaced.
is primarily a function of the distance between the
Coincident techniques use directional microphones, microphones, and not their directionality. This
with the elements placed as close together as possible, technique results in exaggerated stereo separation
but angled apart. The stereo image is a function only of and somewhat indistinct imaging, and is primarily
the directional patterns of the microphones and the used to pick up the ambient sound of a space. Due to
relative angle between them. This generally yields a stereo the large distance between microphones, severe
effect with modest “width” but good “localization” interference effects may result when combining direct
of sound sources. Single-housing/multi-element sounds in mono.
stereo microphones are also
coincident types. They may
contain unidirectional elements,
bidirectional elements, or some
combination of the two. Some
of these, such as the M-S
(Mid-Side) design, are capable
of excellent (and sometimes
variable) stereo width.
Since there is very little
distance between coincident
microphones, there is essentially
sounds picked up by them.
This eliminates any potential
interference (comb filtering) if
the signals are combined for
a monophonic sound system.
Coincident techniques are thus
Near-coincident techniques
also use unidirectional microphones,
but they are placed with their
Stereo Microphone Techniques
Audio Systems Guide for
Though it is perhaps the smallest part of an audio system, the microphone is
one of the most important. Because it is the interface between the sound source
and the sound system, it must interact properly with each. Choosing and using
microphones successfully requires knowledge of the elements of sound, the
sound system, the microphone itself, and the specific microphone application.
Though many components of the sound system have undergone dramatic
changes, particularly with the integration of digital technology, the basic function
of those components, including the microphone, has
not changed. The selection and use of microphones for
house of worship applications continue to apply the
principles illustrated in this guide.
Audio Systems Guide for
We’ve included a reading list for those of you who would like to learn more about the technical aspects of audio. The resources
below are comprehensive, yet for the most part do not require that the reader have an extensive technical background.
Bartlett, Bruce
Introduction to Professional Recording Techniques. Howard W. Sams & Co., Indianapolis, IN
(Excellent recording reference, good microphone section)
Bore, Dr. -Ing. Gerhart
Microphones for Professional and Semi-professional Applications.
Gotham Audio Corporation (U.S. distributor), New York, NY.
(More technical, very good dynamic/condenser comparison)
Burroughs, Lou
Microphones: Design and Application. Sagamore Publishing Co., Plainview, NY.
(A classic, very readable)
Davis, Don, and Davis, Sound System Engineering. Howard W. Sams & Co., Indianapolis, IN
(Very detailed and comprehensive technical reference)
Davis, Gary D., and
Jones, Ralph
Sound Reinforcement Handbook. Hal Leonard Publishing Co., MIlwaukee, WI
(Complete and not overly technical)
Eargle, John
The Microphone Handbook. Elar Publishing Co., Plainview, NY
(Another classic, quite useful)
Eiche, Jon F., ed.
Guide to Sound Systems for Worship. Hal Leonard Publishing Co., Milwaukee, WI
(Based on Sound Reinforcement Handbook, excellent reference)
Huber, David Miles
Microphone Manual-Design and Application. Howard W. Sams & Co., Indianapolis, IN
(Thorough treatment, understandable)
Tim is a native of the south side of Chicago (Go White
Sox!). A lifelong interest in both entertainment and science has
led to the field of audio as his choice for combining these
interests in a useful way. Prior to joining Shure he worked as
an engineer for recording, radio and live sound, operated his
own recording studio and sound company, and continues
to play music professionally. He holds a BS degree in
Aeronautical and Astronautical Engineering, with a minor
in Electrical Engineering, from the University of Illinois,
Urbana-Champaign. While at the University, Tim also worked
as chief technician with both the Speech and Hearing Science
and Linguistics departments.
Since joining Shure in 1984, Tim has served in a technical
support and training capacity for multiple departments. He has
been active in product and applications education for Shure
customers, dealers, and installers, as well as company staff.
His major goal has been to increase the understanding
of quality audio by presenting technical information in a way
that is thorough but still very accessible. Tim's particular
emphasis is on the contribution of proper selection and
technique for both wired and wireless microphones.
In addition, Tim has done technical presentations for
many industry organizations (NAB, NAMM, AES, and SBE),
as well as for US government entities such as the White
House Communication Agency and the US Air Force.
Through His international assignments he has been
fortunate to be able to deliver presentations in more than
twenty countries and on all but one of the continents (still
waiting for an offer from Antarctica...).
He has provided specific applications assistance to
various performing artists including the Rolling Stones and
U2, for theme parks such as Disney and Universal Studios,
and performance groups such as Cirque du Soleil.
While at Shure, Tim has authored several educational
booklets including "Selection and Operation of Wireless
Microphone Systems" and "Audio Systems Guide for Houses
of Worship."
His articles have also appeared in Recording Engineer
Producer, Live Sound Magazine, Pro AV, Technologies for
Worship, and Church Sound Magazine.
This book is dedicated to Lottie Morgan.
Audio Systems Guide for
Reference Information
Shure Product Selection Charts
MX400 Series
Shaped, vocal
Bal. Low Imp.
Miniature gooseneck
MX300 Series
Bal. Low Imp.
Beta 87A/C
Beta 58A
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Beta 53
Beta 54
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Shaped, vocal
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Miniature Lavalier
Miniature Lavalier
Miniature Lavalier
Ultra-Miniature Lavalier
Sub-miniature Lavalier
MX200 Series
Shaped, vocal
Flat, variable
Flat, variable
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Miniature overhead
Full size
Full size
Full size
Full size
MX200 Series
Shaped, vocal
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Miniature overhead
Full size
Full size
Acoustic Guitar
Beta 57A
Shaped, inst.
Flat, variable
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp
Bal. Low Imp.
Full size, stand mount
Full size, stand mount
Full size, stand mount
Full size
Beta181 Series
Flat, variable
Flat, variable
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Full size
Full size
Small side-address
Full size
Full size
Flat, variable
Flat, variable
Flat, variable
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Full size
Full size
Full size
Full size
MX300 Series
MX200 Series
Shaped, vocal
Shaped, vocal
Bal. Low Imp.
Bal. Low Imp.
Boundary, floor mount
Miniature overhead
SM81 (pair)
Beta181 Series (pair)
KSM32 (pair)
KSM137 (pair)
Flat, variable
Flat, variable
Flat, variable
Flat, variable
M-S Stereo
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Bal. Low Imp.
Full size, stand mount
Full size, stand mount
Small side-address
Full size
Full size
Reference Information
Audio Systems Guide for
Shure Product Selection Charts
Wireless Systems
Wireless Systems
Wireless Systems
Digital Systems
Digital Systems
Handheld, Headworn, Lavalier,
Guitar/Bass, Instrument
Handheld, Headworn, Lavalier,
Guitar/Bass, Instrument
Handheld, Headworn, Lavalier, Handheld, Headworn, Lavalier,
Guitar/Bass, Instrument
Guitar/Bass, Instrument
Receiver Options
Single, Half-Rack, Dual
Single, Guitar Pedal
Single, Dual, Quad
Compatible Systems Per Band*
Up to 12
Up to 8
Up to 12
Up to 60
Up to 60
Compatible Systems Using
Multiple Bands
Up to 4 Typical; 8 Maximum
Selectable Frequencies
Group Channel Only
Group Channel Only
Tune in 25KHz
Tune in 25KHz
Tune in 25KHz
Auto Setup Features
Scan / Sync
Scan / Group Scan / Sync
Scan / Group Scan / Sync
Audio Reference Companding
Audio Summing
Dual and Quad Only
Furnished Antennas
BLX4/BLX88 = internal
BLX4R = removable 1/4 wave
GLXD4 = attached 1/4 wave
GLXD6 = Internal Monopole
Detachable 1/4 wave
Detachable 1/2 wave
Remoteable 1/2 wave
Receiver Networking
Yes (Single)
Yes (Dual and Quad)
Computer Monitoring & Control
Yes (WWB6 only)
Yes (WWB6 only)
Rack Hardware
Optional (URT2)
Half Rack: Included
Optional (URT2)
Transmitter Display
Battery: BiColor LED
G/CH: 7 Segment LED
Tri-Color LED
Backlit LCD + Multi-color LED
Backlit Multi-function LCD
Backlit Multi-function LCD
Yes; AES-256
Yes; AES-256
Batteries, Battery Life
2 AA > 14 hrs
Lithium Ion Rechargable
up to 16 hrs
2 AA > 8 hrs
2 AA > 9 hrs
(1 mW / 10mW RF power)
2 AA > 11 hrs
(1 mW / 10mW RF power)
or > 5 hrs (20 mW RF power)
Receivers >>
SB900 > 10 hrs
Handheld, Headworn, Lavalier,
Guitar/Bass, Instrument
SB900 > 12 hrs
(1 mW / 10mW RF power)
or 8 hrs (20 mW RF power)
*Actual number of compatible systems will vary depending on setup and environment
Audio Systems Guide for
Reference Information
Shure Product Selection Charts
Model >>
Transformer-balanced input
Active-balanced input
Transformer-balanced output
Active-balanced output
Low-Z mic-level input
Line level input
Aux level input
Mic level output
Line level output
Phono jack aux level output
Headphone output
Phantom power
48 V phantom power
VU meter
Peak meter
Tone oscillator
Slate mic + tone
Stereo operation
AC operation
Battery operation
Inputs x outputs
XLR & Phoenix
Rack space
1 rack
Audio specs
Dynamic range > 110 dBA
Matrix Mixer
Full matrix mixer
Front panel controls
Preset selector for 16 presets.
Controls for DFR parameters
Front panel audio metering
Mute, 20 dB, 0 dB, Clip LEDs
for each input and output
Automatic feedback reduction
Drag and drop blocks for
5-, 10-, and 16-band single
channel and stereo DFR
DFR filter removal
Auto clear
Additional processing
Drag and drop blocks for GEQ, PEQ,
cut/shelf, delay, single channel and
stereo compressors and limiters,
peak stop limiter, AGC, gate,
downward expander, ducker,
External control options
DRS-10 & serial commands
(AMX or Crestron); contact
closures and potentiometers for
preset, volume and mute.
Control pin inputs
Logic outputs
Front panel lockout with password
protected multi-level security
Shure link
1 From optional external adapter.
2 Internal modification or optional accessory.
Systems >>
with P3R
with P3RA
mono, stereo, mix mode
1/4” (6.3mm)
1/4” (6.3mm)
15 per band
mono, stereo, mix mode
1/4” (6.3mm)
1/4” (6.3mm)
15 per band
stereo, mix mode
2 XLR/TRS, line level
2 TRS, duplicate input
20 per band
stereo, mix mode
2 XLR/TRS, line level
2 TRS, duplicate input
39 per band
Listening mode
Split outputs
On board mixing
Frequency agile
Maximum compatible systems
Remote antenna options
Switchable Transmitter power
Rechargeable option
Model >>
Additional Shure Audio Institute Publications Available:
Printed or electronic versions of the following guides are available free of charge.
To obtain your complimentary copies, call one of the phone numbers listed below
or visit
• Selection and Operation of Personal Monitor Systems
• Selection and Operation of Wireless Microphone Systems
• Audio Systems Guide for Video Production
• Microphone Techniques for Live Sound Reinforcement
• Microphone Techniques for Studio Recording
Our Dedication to Quality Products
Shure offers a complete line of microphones and wireless microphone systems for everyone
from first-time users to professionals in the music industry–for nearly every possible application.
For over nine decades, the Shure name has been synonymous with quality audio.
All Shure products are designed to provide consistent, high-quality performance under the
most extreme real-life operating conditions.
United States, Canada,
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Shure Incorporated
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Niles, IL 60714-4608 USA
©2013 Shure Incorporated
Phone: +1 847-600-2000
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Fax: +1 847-600-6446
©2015 Shure Incorporated
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Phone: +49-7262-92490
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Hong Kong
Phone: +852-2893-4290
Fax: +852-2893-4055
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