SHURE Microphone User Manual

MICROPHONE

TECHNIQUES FOR MUSIC

REINFORCEMENT

SOUND

$10.95

3

INDEX

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

INTRODUCTION . . . . . . . . . . . . . . . . . . . . . . . . . . .4

MICROPHONE CHARACTERISTICS . . . . . . . . . . . . . .4

MUSICAL INSTRUMENT CHARACTERISTICS . . . . . . .11

ACOUSTIC CHARACTERISTICS . . . . . . . . . . . . . . . .14

MICROPHONE PLACEMENT . . . . . . . . . . . . . . . . . .22

STEREO MICROPHONE TECHNIQUES . . . . . . . . . . .32

MICROPHONE SELECTION GUIDE . . . . . . . . . . . . .34

GLOSSARY . . . . . . . . . . . . . . . . . . . . . . . . . . . . .35

4

Introduction

Microphone techniques (the selection and place­ment of microphones) have a major influence on
the audio quality of a sound reinforcement sys­tem. For reinforcement of musical instruments,
there are several main objectives of microphone
techniques: to maximize pick-up of suitable
sound from the desired instrument, to minimize
pick-up of undesired sound from instruments or
other sound sources, and to provide sufficient
gain-before-feedback. “Suitable” sound from the
desired instrument may mean either the natural
sound of the instrument or some particular
sound quality which is appropriate for the appli­cation. “Undesired” sound may mean the direct
or ambient sound from other nearby instruments
or just stage and background noise. “Sufficient”
gain-before-feedback means that the desired
instrument is reinforced at the required level
without ringing or feedback in the sound system.

Obtaining the proper balance of these factors
may involve a bit of give-and-take with each. In
this guide, Shure application and development
engineers suggest a variety of microphone tech­niques for musical instruments to achieve these
objectives. In order to provide some background
for these techniques it is useful to understand
some of the important characteristics of micro­phones, musical instruments and acoustics.

Microphone Characteristics

The most important characteristics of micro­phones for live sound applications are their oper­ating principle, frequency response and direc­tionality. Secondary characteristics are their
electrical output and actual physical design.

Operating principle

- The type of transducer
inside the microphone, that is, how the micro­phone picks up sound and converts it into an
electrical signal.

A transducer is a device that changes energy
from one form into another, in this case, acoustic
energy into electrical energy. The operating
principle determines some of the basic capabili-

ties of the microphone. The two most common
types are Dynamic and Condenser.

Dynamic microphones employ a diaphragm/
voice coil/magnet assembly which forms a
miniature sound-driven 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 can provide excellent sound quali­ty and good specifications in all areas of micro­phone performance. In particular, they can han­dle extremely high sound levels: it is almost
impossible to overload a dynamic microphone.
In addition, dynamic microphones are relatively
unaffected by extremes of temperature or humid­ity. Dynamics are the type most widely used in
general sound reinforcement.

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 metal­coated-plastic diaphragm. The diaphragm is
mounted just in front of a rigid metal or metal­coated-ceramic backplate. In electrical terms this
assembly or element is known as a capacitor (his-

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torically 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.

The construction of a condenser microphone
must include some provision for maintaining the
electrical charge or polarizing voltage. An
electret condenser microphone has a permanent
charge, maintained by a special material deposit­ed on the backplate or on the diaphragm. Non­electret types are charged (polarized) by means
of an external power source. The majority of
condenser microphones for sound reinforcement
are of the electret type.

All condensers contain additional active circuitry
to allow the electrical output of the element to be
used with 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. For this
reason, condenser microphone specifications
always include a noise figure and a maximum
sound level. Good designs, however, have very
low noise levels and are also capable of very
wide dynamic range.

PHANTOM POWER

Phantom power is a DC voltage (usually 12-48
volts) used to power the electronics of a con­denser microphone. For some (non-electret)
condensers it may also be used to provide the
polarizing voltage for the element itself. This
voltage is supplied through the microphone
cable by a mixer equipped with phantom power
or by some type of in-line external source. The
voltage is equal on Pin 2 and Pin 3 of a typical
balanced, XLR-type connector. For a 48 volt
phantom source, for example, Pin 2 is 48 VDC
and Pin 3 is 48 VDC, both with respect to Pin 1
which is ground (shield).

Because the voltage is exactly the same on Pin 2
and Pin 3, phantom power will have no effect on
balanced dynamic microphones: no current will
flow since there is no voltage difference across
the output. In fact, phantom power supplies
have current limiting which will prevent damage
to a dynamic microphone even if it is shorted or
miswired. In general, balanced dynamic micro­phones can be connected to phantom powered
mixer inputs with no problem.

Fig. 3: phantom power schematic

Condenser microphones are more complex than
dynamics and tend to be somewhat more costly .
Also, condensers may be adversely affected by
extremes of temperature and humidity which can
cause them to become noisy or fail temporarily .
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.

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TRANSIENT RESPONSE

Transient response refers to the ability of a
microphone to respond to a rapidly changing
sound wave. Agood way to understand why
dynamic and condenser mics sound different is
to understand the differences in their
transient response.

In order for a microphone to convert sound
energy into electrical energy, the sound wave
must physically move the diaphragm of the
microphone. The amount of time it takes for
this movement to occur depends on the weight
(or mass) of the diaphragm. For instance,
the diaphragm and voice coil assembly of a
dynamic microphone may weigh up to 1000
times more than the diaphragm of a condenser
microphone. It takes longer for the heavy
dynamic diaphragm to begin moving than for
the lightweight condenser diaphragm. It also
takes longer for the dynamic diaphragm to
stop moving in comparison to the condenser
diaphragm. Thus, the dynamic transient
response is not as good as the condenser
transient response. This is similar to two
vehicles in traffic: a truck and a sports car.
They may have equal power engines but the
truck weighs much more than the car. As
traffic flow changes, the sports car can
accelerate and brake very quickly, while the
semi accelerates and brakes very slowly due
to its greater weight. Both vehicles follow
the overall traffic flow but the sports car
responds better to sudden changes.

Pictured here are two studio microphones
responding to the sound impulse produced
by an electric spark: condenser mic on top,
dynamic mic on bottom. It is evident that it
takes almost twice as long for the dynamic
microphone to respond to the sound. It also
takes longer for the dynamic to stop moving
after the impulse has passed (notice the ripple
on the second half of the graph). Since con­denser microphones generally have better
transient response then dynamics, they are
better suited for instruments that have very
sharp attack or extended high frequency output

such as cymbals. It is this transient response
difference that causes condenser mics to have a
more crisp, detailed sound and dynamic mics to
have a more mellow, rounded sound.

The decision to use a condenser or dynamic
microphone depends not only on the sound
source and the sound reinforcement system
but on the physical setting as well. From a
practical standpoint, if the microphone will be
used in a severe environment such as a rock
and roll club or for outdoor sound, dynamic
types would be a good choice. In a more
controlled environment such as a concert hall
or theatrical setting, a condenser microphone
might be preferred for many sound sources,
especially when the highest sound quality is
desired.

Fr

equency response - The output level or

sensitivity of the microphone over its operating
range from lowest to highest frequency.

Virtually all microphone manufacturers list
the frequency response of their microphones
over a range, for example 50 - 15,000 Hz.
This usually corresponds with a graph that
indicates output level relative to frequency.
The graph has frequency in Hertz (Hz) on the
x-axis and relative response in decibels (dB)
on the y-axis.

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Condenser/dynamic scope photo

A microphone whose output is equal at all
frequencies has a flat frequency response.

Flat response microphones typically have an
extended frequency range. They reproduce a
variety of sound sources without changing or
coloring the original sound.

Amicrophone whose response has peaks or dips in
certain frequency areas exhibits a shaped response.

A shaped response is usually designed to enhance
a sound source in a particular application.

For instance, a microphone may have a peak in
the 2 - 8 kHz range to increase intelligibility for
live vocals. This shape is called a presencepeak
or rise. Amicrophone may also be designed to be
less sensitive to certain other frequencies. One
example is reduced low frequency response (low
end roll-off) to minimize unwanted “boominess”
or stage rumble.

THE DECIBEL

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:

dB = 20 x log(V1/V2)

where 20 is a constant, V1 is one voltage, V2 is
the other voltage, and log is logarithm base 10.

Examples:

What is the relationship in decibels
between 100 volts and 1 volt?

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 0.001 volt and 1 volt?

dB = 20 x log(0.001/1)
dB = 20 x log(0.001)
dB = 20 x (-3) (the log of .001 is -3)
dB = -60

That is, 0.001 volt is 60dB less that 1 volt.

Similarly:

if one voltage is equal to the other they
are 0dB different

if one voltage is twice the other they are
6dB different

if one voltage is ten times the other they
are 20dB different

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Flat frequency response

Shaped frequency response

8

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 decibel value such
as: dBV (reference to 1 volt which is 0dBV) or
dB SPL (reference to 0.0002 microbar which is
0dB Sound Pressure Level)

Decibel scale

for dBV or dB SPL

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. A6dB SPL
change is quite noticeable and finally, a 10dB
SPL change is perceived as “twice as loud.”

The choice of flat or shaped response micro­phones again depends on the sound source, the
sound system and the environment. Flat
response microphones are usually desirable to
reproduce instruments such as acoustic guitars or
pianos, especially with high quality sound sys­tems. They are also common in stereo miking
and distant pickup applications where the micro­phone is more than a few feet from the sound
source: the absence of response peaks mini­mizes feedback and contributes to a more natural
sound. On the other hand, shaped response micro­phones are preferred for closeup vocal use and for
certain instruments such as drums and guitar ampli­fiers which may benefit from response enhance­ments for presence or punch. They are also useful
for reducing pickup of unwanted sound and noise
outside the frequency range of an instrument.

Dir

ectionality - Amicrophone’s sensitivity to

sound relative to the direction or angle from
which the sound arrives.

There are a number of different directional
patterns found in microphone design. These
are typically plotted in a polar pattern to
graphically display the directionality of the
microphone. The polar pattern shows the
variation in sensitivity 360 degrees around the
microphone, assuming that the microphone is
in the center and that 0 degrees represents the
front of the microphone.

The three basic directional types of micro­phones are omnidirectional, unidirectional,
and bidirectional.

The omnidirectional microphone has equal
output or sensitivity at all angles. Its coverage
angle is a full 360 degrees. An omnidirectional
microphone will pick up the maximum amount
of ambient sound. In live sound situations an
omni should be placed very close to the sound
source to pick up a useable balance between
direct sound and ambient sound. In addition,
an omni cannot be aimed away from undesired
sources such as PA speakers which may cause
feedback.

Omnidirectional
The unidirectional microphone is most sensitive

to sound arriving from one particular direction
and is less sensitive at other directions. The
most common type is a cardioid (heart-shaped)
response. This has the most sensitivity at
0 degrees (on-axis) and is least sensitive at 180

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100=1
101=10
102=100
103=1000
104=10,000
105=100,000
106=1,000,000

0
20
40
60
80
100
120

1. Compare

2. Compress 3. scale (x 20)

b a

b/a

degrees (off-axis). The effective coverage or
pickup angle of a cardioid is about 130 degrees,
that is up to about 65 degrees off axis at the
front of the microphone. In addition, the cardioid
mic picks up only about one-third as much
ambient sound as an omni. Unidirectional
microphones isolate the desired on-axis sound
from both unwanted off-axis sound and from
ambient noise.

Cardioid

For example, the use of a cardioid microphone
for a guitar amplifier which is near the drum set
is one way to reduce bleed-through of drums
into the reinforced guitar sound.

Unidirectional microphones have several
variations on the cardioid pattern. Two of these
are the supercardioid and hypercardioid.

Both patterns offer narrower front pickup angles
than the cardioid (115 degrees for the supercar­dioid and 105 degrees for the hypercardioid) and
also greater rejection of ambient sound. While
the cardioid is least sensitive at the rear (180
degrees off-axis) the least sensitive direction is
at 126 degrees off-axis for the supercardioid and
110 degrees for the hypercardioid. When placed
properly they can provide more focused pickup
and less ambient noise than the cardioid pattern,
but they have some pickup directly at the rear,
called a rear lobe. The rejection at the rear is

-12 dB for the supercardioid and only -6 dB for
the hypercardioid. Agood cardioid type has at
least 15-20 dB of rear rejection.
Supercardioid

The bidirectional microphone has maximum
sensitivity at both 0 degrees (front) and at 180
degrees (back). It has the least amount of out­put at 90 degree angles (sides). The coverage
or pickup angle is only about 90 degrees at
both the front and the rear. It has the same
amount of ambient pickup as the cardioid.
This mic could be used for picking up two
opposing sound sources, such as a vocal duet.
Though rarely found in sound reinforcement
they are used in certain stereo techniques,
such as M-S (mid-side).

Microphone Polar Patterns Compared

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Cardioid

10

In sound reinforcement, microphones must often
be located in positions where they may pick up
unintended instrument or other sounds. Some
examples are: individual drum mics picking up
adjacent drums, vocal mics picking up overall
stage noise, and vocal mics picking up monitor
speakers. In each case there is a desired sound
source and one or more undesired sound sources.
Choosing the appropriate directional pattern can
help to maximize the desired sound and mini­mize the undesired sound.

Although the direction for maximum pickup is
usually obvious (on-axis) the direction for least
pickup varies with microphone type. In particu­lar, the cardioid is least sensitive at the rear (180
degrees off-axis) while the supercardioid and
hypercardioid types actually have some rear
pickup. They are least sensitive at 125 degrees
off-axis and 110 degrees off axis respectively.

For example, when using floor monitors with
vocal mics, the monitor should be aimed directly
at the rear axis of a cardioid microphone for
maximum gain-before-feedback. When using a
supercardioid, however, the monitor should be
positioned somewhat off to the side (55 degrees
off the rear axis) for best results. Likewise,
when using supercardioid or hypercardioid types
on drum kits be aware of the rear pickup of these
mics and angle them accordingly to avoid pick­up of other drums or cymbals.

Monitor speaker placement for

maximum rejection:

cardioid and supercardioid

Other directional related microphone characteristics:

Ambient sound rejection - Since unidirectional
microphones are less sensitive to off-axis sound
than omnidirectional types they pick up less
overall ambient or stage sound. Unidirectional
mics should be used to control ambient noise
pickup to get a cleaner mix.

Distance factor - Because directional micro­phones pick up less ambient sound than omni­directional types they may be used at some­what greater distances from a sound source and
still achieve the same balance between the
direct sound and background or ambient
sound. An omni should be placed closer to
the sound source than a uni—about half the
distance—to pick up the same balance between
direct sound and ambient sound.

Off-axis coloration - Change in a microphone’s
frequency response that usually gets progressive­ly more noticeable as the arrival angle of sound
increases. High frequencies tend to be lost first,
often resulting in “muddy” off-axis sound.

Proximity effect - With unidirectional micro­phones, bass response increases as the mic is
moved closer (within 2 feet) to the sound source.
With close-up unidirectional microphones (less
than 1 foot), be aware of proximity effect and
roll off the bass until you obtain a more natural
sound. You can (1) roll off low frequencies on
the mixer, or (2) use a microphone designed to
minimize proximity effect, or (3) use a micro­phone with a bass rolloff switch, or (4) use an
omnidirectional microphone (which does not
exhibit proximity effect).

Proximity effect graph

Unidirectional microphones can not only help

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USING DIRECTIONAL PATTERNS TO

REJECT UNWANTED SOURCES

to isolate one voice or instrument from other
singers or instruments, but can also minimize
feedback, allowing higher gain. For these
reasons, unidirectional microphones are
preferred over omnidirectional microphones in
almost all sound reinforcement applications.

The electrical output of a microphone is
usually specified by level, impedance and wiring
configuration. Output level or sensitivity is the
level of the electrical signal from the micro­phone for a given input sound level. In general,
condenser microphones have higher sensitivity
than dynamic types. For weak or distant sounds
a high sensitivity microphone is desirable while
loud or close-up sounds can be picked up well
by lower-sensitivity models.

The output impedance of a microphone is rough­ly equal to the electrical resistance of its output:
150-600 ohms for low impedance (low-Z) and
10,000 ohms or more for high impedance.(high­Z). The practical concern is that low impedance
microphones can be used with cable lengths of
1000 feet or more with no loss of quality while
high impedance types exhibit noticeable high
frequency loss with cable lengths greater than
about 20 feet.

Finally , the wiring configuration of a microphone
may be balanced or unbalanced. Abalanced
output carries the signal on two conductors (plus
shield). The signals on each conductor are the
same level but opposite polarity (one signal is
positive when the other is negative). Abalanced
microphone input amplifies only the difference
between the two signals and rejects any part of the
signal which is the same in each conductor. Any
electrical noise or hum picked up by a balanced
(two-conductor) cable tends to be identical in the
two conductors and is therefore rejected by the
balanced input while the equal but opposite
polarity original signals are amplified. On the
other hand, an unbalanced microphone output
carries its signal on a single conductor (plus
shield) and an unbalanced microphone input
amplifies any signal on that conductor. Such a
combination will be unable to reject any electrical
noise which has been picked up by the cable.
Balanced, low-impedance microphones are

therefore recommended for nearly all sound
reinforcement applications.

The physical design of a microphone is its
mechanical and operational design. Types used
in sound reinforcement include: handheld, head­worn, lavaliere, overhead, stand-mounted, instru­ment-mounted and surface-mounted designs.
Most of these are available in a choice of operat­ing principle, frequency response, directional
pattern and electrical output. Often the physical
design is the first choice made for an application.
Understanding and choosing the other character­istics can assist in producing the maximum qual­ity microphone signal and delivering it to the
sound system with the highest fidelity.

Musical Instrument Characteristics

Some background information on characteris­tics of musical instruments may be helpful.
Instruments and other sound sources are char­acterized by their frequency output, by their
directional output and by their dynamic range.

Frequency output - the span of fundamental
and harmonic frequencies produced by an
instrument, and the balance or relative level of
those frequencies.
Musical instruments have overall frequency

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12

ranges as found in the chart below. The dark
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 instrument.
The fundamental frequency establishes the
basic pitch of a note played by an instrument
while the harmonics produce the timbre or
characteristic tone.

Instrument frequency ranges

It is this timbre that distinguishes the sound
of one instrument from another. In this man­ner, we can tell whether a piano or a trumpet
just played that C note. The following graphs
show the levels of the fundamental and
harmonics associated with a trumpet and an
oboe each playing the same note.

Instrument spectra comparison

The number of harmonics along with the relative

level of the harmonics is noticeably different
between these two instruments and provides
each instrument with its own unique sound.

A microphone which responds evenly to the full
range of an instrument will reproduce the most
natural sound from an instrument. Amicrophone
which responds unevenly or to less than the full
range will alter the sound of the instrument,
though this effect may be desirable in some cases.

Directional output - the three-dimensional pat­tern of sound waves radiated by an instrument.

A musical instrument radiates a different tone
quality (timbre) in every direction, and each part
of the instrument produces a different timbre.
Most musical instruments are designed to sound
best at a distance, typically two or more feet
away. At this distance, the sounds of the various
parts of the instrument combine into a pleasing
composite. In addition, many instruments pro­duce this balanced sound only in a particular
direction. Amicrophone placed at such distance
and direction tends to pick up a natural or well­balanced tone quality.

On the other hand, a microphone placed close to
the instrument tends to emphasize the part of the
instrument that the microphone is near. The result­ing sound may not be representative of the instru­ment as a whole. Thus, the reinforced tonal bal­ance of an instrument is strongly affected by the
microphone position relative to the instrument.

Unfortunately, it is difficult, if not impossible, to
place a microphone at the “natural sounding”
distance from an instrument in a sound rein­forcement situation without picking up other
(undesired) sounds and/or acoustic feedback.
Close microphone placement is usually the only
practical way to achieve sufficient isolation and
gain-before-feedback. But since the sound
picked up close to a source can vary significantly
with small changes in microphone position, it is
very useful to experiment with microphone loca­tion and orientation. In some cases more than
one microphone may be required to get a good
sound from a large instrument such as a piano.

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200 500 50004000300020001000

oboe

trumpet in B

b

frequency

Another instrument with a wide range of charac­teristics is the loudspeaker. Anytime you are
placing microphones to pick up the sound of a
guitar or bass cabinet you are confronted with
the acoustic nature of loudspeakers. Each indi­vidual loudspeaker type is directional and dis­plays different frequency characteristics at differ­ent angles and distances. The sound from a loud­speaker tends to be almost omnidirectional at
low frequencies but becomes very directional at
high frequencies. Thus, the sound on-axis at the
center of a speaker usually has the most “bite” or
high-end, while the sound produced off-axis or
at the edge of the speaker is more “mellow” or
bassy. Acabinet with multiple loudspeakers has
an even more complex output, especially if it has
different speakers for bass and treble. As with
most acoustic instruments the desired sound only
develops at some distance from the speaker.

Sound reinforcement situations typically require
a close-mic approach. Aunidirectional dynamic
microphone is a good first choice here: it can
handle the high level and provide good sound
and isolation. Keep in mind the proximity effect
when using a uni close to the speaker: some
bass boost will be likely. If the cabinet has only
one speaker a single microphone should pick up
a suitable sound with a little experimentation. If
the cabinet has multiple speakers of the same
type it is typically easiest to place the micro­phone to pick up just one speaker. Placing the
microphone between speakers can result in
strong phase effects though this may be desirable
to achieve a particular tone. However, if the
cabinet is stereo or has separate bass and treble
speakers multiple microphones may be required.

Placement of loudspeaker cabinets can also have a
significant effect on their sound. Putting cabinets
on carpets can reduce brightness, while raising
them off the floor can reduce low end. Open-back
cabinets can be miked from behind as well as from
the front. The distance from the cabinet to walls or
other objects can also vary the sound. Again,
experiment with the microphone(s) and placement
until you have the sound that you like!

instrument from its softest to its loudest level.

The dynamic range of an instrument determines
the specifications for sensitivity and maximum
input capability of the intended microphone.
Loud instruments such as drums, brass and
amplified guitars are handled well by dynamic
microphones which can withstand high sound
levels and have moderate sensitivity. Softer
instruments such as flutes and harpsichords can
benefit from the higher sensitivity of condensers.
Of course, the farther the microphone is placed
from the instrument the lower the level of sound
reaching the microphone.

In the context of a live performance, the relative
dynamic range of each instrument determines how
much sound reinforcement may be required. If all
of the instruments are fairly loud, and the venue is
of moderate size with good acoustics, no reinforce­ment may be necessary . On the other hand, if the
performance is in a very large hall or outdoors,
even amplified instruments may need to be further
reinforced. Finally , if there is a substantial dif fer­ence in dynamic range among the instruments,
such as an acoustic
guitar in a loud rock
band, the micro­phone techniques
(and the sound sys­tem) must accom­modate those differ­ences. Often, the
maximum
volume of the
overall sound
system is limited
by the maximum
gain-before-feed­back of the softest
instrument.

An understanding
of the frequency
output, directional output, and dynamic range
characteristics of musical instruments can help
significantly in choosing suitable microphones,
placing them for best pickup of the desired
sound and minimizing feedback or other unde­sired sounds.

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0 20 40 60 80 100 120

CYM BAL

SNARE DRUM

BASS DRUM

FEMALE VOICE

MALE VOICE

TRUMPET

HAR MONICA

SAX OPHONE

GUI TAR

PIA NO

VIOLIN

Intensity Level in Decibels

(at distance of 10 feet)

INSTRUMENT LOUDSPEAKERS

Dynamic range - the range of volume of an

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Acoustic Characteristics

Sound Waves

Sound moves through the air like waves in water.
Sound waves consist of pressure variations travel­ing through the air. When the sound wave travels,
it compresses air molecules together at one point.
This is called the high pressure zone or positive
component(+). After the compression, an expan­sion of molecules occurs. This is the low pressure
zone or negative component(-). This process con­tinues along the path of the sound wave until its
energy becomes too weak to hear. The sound
wave of a pure tone traveling through air would
appear as a smooth, regular variation of pressure
that could be drawn as a sine wave.

Frequency, wavelength and the speed of sound

The frequency
of a sound wave
indicates the rate
of pressure vari­ations or cycles.
One cycle is a
change from
high pressure to
low pressure
and back to high pressure. The number of cycles
per second is called Hertz, abbreviated “Hz.”
So, a 1,000 Hz tone has 1,000 cycles per second.

The wavelength of a sound is the physical distance
from the start of one cycle to the start of the next
cycle. Wavelength is related to frequency by the
speed of sound. The speed of sound in air is about
1130 feet per second or 344 meters/second. The
speed of sound is constant no matter what the fre­quency . The wavelength of a sound wave of any
frequency can be determined by these relationships:

Loudness

The fluctuation of
air pressure created
by sound is a change
above and below
normal atmospheric
pressure. This is
what the human ear
responds to. The
varying amount of
pressure of the air
molecules compress­ing and expanding is
related to the appar­ent loudness at the
human ear. The
greater the pressure change, the louder the
sound. Under ideal conditions the human ear
can sense a pressure change as small as 0.0002
microbars (1 microbar = 1/1,000,000 atmospher­ic pressure). The threshold of pain is about 200
microbars, one million times greater! Obviously
the human ear responds to a wide range of
amplitude of sound. This amplitude range is
more commonly measured in decibels Sound
Pressure Level (dB SPL), relative to 0.0002
microbars (0 dB SPL). 0

dB SPL is the threshold

of hearing L

p

and 120 dB SPL is the threshold of
pain. 1dB is about the smallest change in SPL
that can be heard. A3dB change is generally
noticeable while a 6dB change is very notice­able. A10dB SPL increase is perceived to be
twice as loud!

Sound Propagation

There are four basic ways in which sound can
be altered by its environment as it travels or
propagates: reflection, absorption, diffraction
and refraction.

The Wave Equation: c = f • l

speed of sound = frequency • wavelength

or

wavelength =

speed of sound

frequency

for a 500Hz sound wave:

wavelength =

1,130 feet per second

500Hz

wavelength = 4.4 feet

Approximate wavelengths of common

frequencies:

100 Hz: about 10 feet
1000 Hz: about 1 foot
10,000 Hz: about 1 inch

Ambient sounds

DISTANCE

WAVELENGTH

PRESSURE

▲

▲

+

0
_

1 CYCLE

▲

▲

▲

▲

1

/2CYCLE

▲

▲

AMPLITUDE

Schematic of sound wave

140

130

120

110

100

90

80

70

60

50

40

30

20

10

0

1. Reflection - A sound wave can be reflected

by a surface or other object if the object is physi­cally as large or larger than the wavelength of
the sound. Because low frequency sounds have
long wavelengths they can only be reflected by
large objects. Higher frequencies can be reflect­ed by smaller objects and surfaces as well as
large. The reflected sound will have a different
frequency characteristic than the direct sound if
all frequencies are not reflected equally.

Reflection is also the source of echo, reverb, and
standing waves:

Echo occurs when a reflected sound is delayed
long enough (by a distant reflective surface) to
be heard by the listener as a distinct repetition of
the direct sound.

Reverberation consists of many reflections of a
sound, maintaining the sound in a reflective space
for a time even after the direct sound has stopped.

Standing waves in a room occur for certain fre­quencies related to the distance between parallel
walls. The original sound and the reflected sound
will begin to reinforce each other when the dis­tance between two opposite walls is equal to a
multiple of half the wavelength of the sound. This
happens primarily at low frequencies due to their
longer wavelengths and relatively high energy.

2. Absorption- Some materials absorb sound

rather than reflect it. Again, the efficiency of
absorption is dependent on the wavelength.
Thin absorbers like carpet and acoustic ceiling
tiles can affect high frequencies only, while thick
absorbers such as drapes, padded furniture and
specially designed bass traps are required to
attenuate low frequencies. Reverberation in a
room can be controlled by adding absorption:
the more absorption the less reverberation.
Clothed humans absorb mid and high frequen­cies well, so the presence or absence of an audi­ence has a significant effect on the sound in an
otherwise reverberant venue.

3. Diffraction - Asound wave will typically

bend around obstacles in its path which are
smaller than its wavelength. Because a low fre-

quency sound wave is much longer than a high
frequency wave, low frequencies will bend
around objects that high frequencies cannot.
The effect is that high frequencies tend to have a
higher directivity and are more easily blocked
while low frequencies are essentially omnidirec­tional. In sound reinforcement, it is difficult to
get good directional control at low frequencies
for both microphones and loudspeakers.

4. Refraction - The bending of a sound wave

as it passes through some change in the density
of the environment. This effect is primarily
noticeable outdoors at large distances from loud­speakers due to atmospheric effects such as wind
or temperature gradients. The sound will appear
to bend in a certain direction due to these effects.

Direct vs. Ambient Sound

A very important property of direct sound is that it
becomes weaker as it travels away from the sound
source. The amount of change is controlled by
the inverse-square law which states that the level
change is inversely proportional to the square of
the distance change. When the distance from a
sound source doubles, the sound level decreases
by 6dB. This is a noticeable decrease. For exam­ple, if the sound from a guitar amplifier is 100 dB
SPL at 1 ft. from the cabinet it will be 94 dB at 2
ft., 88 dB at 4 ft., 82 dB at 8 ft., etc. Conversely ,
when the distance is cut in half the sound level
increases by 6dB: It will be 106 dB at 6 inches
and 112 dB at 3 inches!

On the other hand, the ambient sound in a room
is at nearly the same level throughout the room.
This is because the ambient sound has been
reflected many times within the room until it is
essentially non-directional. Reverberation is an
example of non-directional sound.

For this reason the ambient sound of the room
will become increasingly apparent as a micro­phone is placed further away from the direct
sound source. In every room, there is a distance
(measured from the sound source) where the
direct sound and the reflected (or reverberant)
sound become equal in intensity . In acoustics,
this is known as the Critical Distance. If a micro-

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16

phone is placed at the Critical Distance or farther,
the sound quality picked up may be very poor.
This sound is often described as “echoey”, rever­berant, or “bottom of the barrel”. The reflected
sound overlaps and blurs the direct sound.

Critical distance may be estimated by listening
to a sound source at a very short distance, then
moving away until the sound level no longer
decreases but seems to be constant. That dis­tance is critical distance.

A unidirectional microphone should be positioned
no farther than 50% of the Critical Distance, e.g.
if the Critical Distance is 10 feet, a unidirectional
mic may be placed up to 5 feet from the sound
source. Highly reverberant rooms may require
very close microphone placement. The amount of
direct sound relative to ambient sound is con­trolled primarily by the distance of the micro­phone to the sound source and to a lesser degree
by the directional pattern of the mic.

Phase relationships and interference effects

The phase of a
single frequency sound
wave is always described
relative to the starting
point of the wave
or 0 degrees. The
pressure change is also

zero at this point. The
peak of the high pressure zone is at 90 degrees,
the pressure change falls to zero again at 180
degrees, the peak of the low pressure zone is at
270 degrees, and the pressure change rises to zero
at 360 degrees for the start of the next cycle.

Two identical sound waves starting at the same
point in time are called “in-phase” and will sum
together creating a single wave with double the
amplitude but otherwise identical to the original
waves. Two identical sound waves with one
wave’s starting point occurring at the 180 degree
point of the other wave are said to be “out of
phase” and the two waves will cancel each other
completely. When two sound waves of the same
single frequency but different starting points are
combined the resulting wave is said to have

“phase shift” or an
apparent starting
point somewhere
between the origi­nal starting points.
This new wave
will have the same
frequency as the
original waves but
will have
increased or
decreased ampli­tude depending on
the degree of
phase difference.
Phase shift, in this
case, indicates that
the 0 degree
points of two
identical waves
are not the same.

Most soundwaves are not a single frequency but
are made up of many frequencies. When identical
multiple-frequency soundwaves combine there
are three possibilities for the resulting wave: a
doubling of amplitude at all frequencies if the
waves are in phase, a complete cancellation at all
frequencies if the waves are 180 degrees out of
phase, or partial cancellation and partial reinforce­ment at various frequencies if the waves have
intermediate phase relationship. The results may
be heard as interference effects.

The first case is the basis for the increased sensi­tivity of boundary or surface-mount micro­phones. When a microphone element is placed
very close to an acoustically reflective surface
both the incident and reflected sound waves are in
phase at the microphone. This results in a 6dB
increase (doubling) in sensitivity , compared to the
same microphone in free space. This occurs for
reflected frequencies whose wavelength is greater
than the distance from the microphone to the sur­face: if the distance is less than one-quarter inch
this will be the case for frequencies up to at least
18 kHz. However, this 6dB increase will not
occur for frequencies that are not reflected, that is,
frequencies that are either absorbed by the surface
or that diffract around the surface. High frequen-

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Phase relationships

Sound pressure wave

+

+1

0

-1

+1

0

-1

a

=

+2

0

-2

+

+1

0

-1

+1

0

-1

b

=

0

+

+1

0

-1

+1

0

-1

c

=

“phase shifts”

“in-phase”

”180

0

out

of phase”

+2

+1

0

-1

-2

▲

▲

one cycle or one period

90

000

18002700360

0

cies may be absorbed by surface materials such as
carpeting or other acoustic treatments. Low fre­quencies will diffract around the surface if their
wavelength is much greater than the dimensions
of the surface: the boundary must be at least 5 ft.
square to reflect frequencies down to 100 Hz.

The second case occurs when two closely spaced
microphones are wired out of phase, that is, with
reverse polarity . This usually only happens by
accident, due to miswired microphones or cables
but the effect is also used as the basis for certain
noise-canceling microphones. In this technique,
two identical microphones are placed very close
to each other (sometimes within the same hous­ing) and wired with opposite polarity . Sound
waves from distant sources which arrive equally
at the two microphones are effectively canceled
when the outputs are mixed. However, sound
from a source which is much closer to one ele­ment than to other will be heard. Such close-talk
microphones, which must literally have the lips of

the talker touching
the grille, are used
in high-noise
environments
such as aircraft
and industrial pag­ing but rarely with
musical instru­ments due to their
limited frequency
response.

Polarity reversal

It is the last case which is most likely in musical
sound reinforcement, and the audible result is a
degraded frequency response called “comb filter­ing.” The pattern of peaks and dips resembles the
teeth of a comb and the depth and location of
these notches depend on the degree of phase shift.

With microphones this effect can occur in two
ways. The first is when two (or more) mics
pick up the same sound source at different dis­tances. Because it takes longer for the sound to
arrive at the more distant microphone there is
effectively a phase difference between the sig­nals from the mics when they are combined

(electrically) in the mixer. The resulting comb
filtering depends on the sound arrival time
difference between the microphones: a large
time difference (long distance) causes comb
filtering to begin at low frequencies, while a
small time difference (short distance) moves
the comb filtering to higher frequencies.

The second way
for this effect to
occur is when a
single micro­phone picks up a
direct sound and
also a delayed
version of the
same sound.
The delay may
be due to an

acoustic reflec­tion of the original sound or to multiple sources
of the original sound. Aguitar cabinet with
more than one speaker or multiple loudspeaker
cabinets for a single instrument would be
examples. The delayed sound travels a longer
distance (longer time) to the mic and thus has a
phase difference relative to the direct sound.
When these sounds combine (acoustically) at the
microphone, comb filtering results. This time
the effect of the comb filtering depends on the
distance between the microphone and the source
of the reflection or the distance between the
multiple sources.

Reflection comb filtering

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Multi-mic comb filtering

18

The 3-to-1 Rule

When it is necessary to use multiple micro­phones or to use microphones near reflective
surfaces the resulting interference effects
may be minimized by using the 3-to-1 rule.
For multiple microphones the rule states
that the distance between microphones
should be at least three times the distance
from each microphone to its intended sound
source. The sound picked up by the more
distant microphone is then at least 12dB less
than the sound picked up by the closer one.
This insures that the audible effects of comb
filtering are reduced by at least that much.
For reflective surfaces, the microphone
should be at least 1

1

/2times as far from
that surface as it is from its intended sound
source. Again, this insures minimum
audibility of interference effects.

3-to-1 rule

Strictly speaking, the 3-to-1 rule is based on
the behavior of omnidirectional microphones.
It can be relaxed slightly if unidirectional
microphones are used and they are aimed
appropriately, but should still be regarded as a
basic rule of thumb for worst case situations.

MICROPHONE PHASE EFFECTS

One effect often heard in sound reinforcement
occurs when two microphones are placed in close
proximity to the same sound source, such as a drum
kit or instrument amplifier. Many times this is due to
the phase relationship of the sounds arriving at the
microphones. If two microphones are picking up the
same sound source from different locations, some
phase cancellation or summing may be occurring.
Phase cancellation happens when two microphones
are receiving the same soundwave but with opposite
pressure zones (that is,180 degrees out of phase).
This is usually not desired. Amic with a different
polar pattern may reduce the pickup of unwanted
sound and reduce the effect or physical isolation can
be used. With a drum kit, physical isolation of the
individual drums is not possible. In this situation the
choice of microphones may be more dependent on
the off-axis rejection characteristic of the mic.

Another possibility is phase reversal. If there is
cancellation occurring, a 180 degree phase flip will
create phase summing of the same frequencies. A
common approach to the snare drum is to place one
mic on the top head and one on the bottom head.
Because the mics are picking up relatively similar
sound sources at different points in the sound wave,
you may experience some phase cancellations.
Inverting the phase of one mic will sum any frequen­cies being canceled. This may sometimes achieve a
“fatter“ snare drum sound. This effect will change
dependent on mic locations. The phase inversion can
be done with an in-line phase reverse adapter or by a
phase invert switch found on many mixers inputs.

Potential Acoustic Gain vs. Needed
Acoustic Gain

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.

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To calculate NAG: NAG = 20 x log (Df/Dn)

Where: D

f

= distance from sound source to

furthest listener

D

n

= distance from sound source to

nearest listener

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 inverse­square 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). Below is
an illustration showing the key distances.

PAG

The simplified PAG equation is:

PAG = 20 (log D

1

- log D2+ log D0- log Ds)

-10 log NOM -6

Where: PAG = Potential Acoustic Gain (in dB)

D

s

= distance from sound source to

microphone

D

0

= distance from sound source to

listener

D

1

= distance from microphone to

loudspeaker

D

2

= distance from loudspeaker to

listener

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.

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-before-feedback by 6dB while
moving it to 4 ft. away will decrease it by 12dB.
On the other hand, moving it to 6 in. away

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D

0

D

s

D

2

D

1

20

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.

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.

Understanding principles of basic acoustics can
help to create an awareness of potential influ­ences on reinforced sound and to provide some
insight into controlling them. When effects of
this sort are encountered and are undesirable, it
may be possible to adjust the sound source, use a
microphone with a different directional charac­teristic, reposition the microphone or use fewer
microphones, or possibly use acoustic treatment
to improve the situation. Keep in mind that in
most cases, acoustic problems can best be solved
acoustically, not strictly by electronic devices.

General Rules

Microphone technique is largely a matter of per­sonal taste—whatever method sounds right

for
the particular instrument, musician, and song
is right

. There is no one ideal microphone to use
on any particular instrument. There is also no
one ideal way to place a microphone. Choose
and place the microphone to get the sound you
want. We recommend experimenting with a
variety of microphones and positions until you

create your desired sound. However, the desired
sound can often be achieved more quickly and
consistently by understanding basic microphone
characteristics, sound-radiation properties of
musical instruments, and acoustic fundamentals
as presented above.

Here are some suggestions to follow when mik­ing musical instruments for sound reinforcement.

• Try to get the sound source (instrument, voice,
or amplifier) to sound good acoustically
(“live”) before miking it.

• Use a microphone with a frequency response
that is limited to the frequency range of the
instrument, if possible, or filter out frequencies
below the lowest fundamental frequency of the
instrument.

• To determine a good starting microphone posi­tion, try closing one ear with your finger.
Listen to the sound source with the other ear
and move around until you find a spot that
sounds good. Put the microphone there.
However, this may not be practical (or healthy)
for extremely close placement near loud
sources.

• The closer a microphone is to a sound source,
the louder the sound source is compared to
reverberation and ambient noise. Also, the
Potential Acoustic Gain is increased—that is,
the system can produce more level before feed­back occurs. Each time the distance between
the microphone and sound source is halved, the
sound pressure level at the microphone (and
hence the system) will increase by 6 dB.

(Inverse Square Law)

• Place the microphone only as close as neces­sary. Too close a placement can color the
sound source’s tone quality (timbre), by pick­ing up only one part of the instrument. Be
aware of Proximity Effect with unidirectional
microphones and use bass rolloff if necessary.

• Use as few microphones as are necessary to get
a good sound. To do that, you can often pick
up two or more sound sources with one micro-

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phone. Remember: every time the number of
microphones doubles, the Potential Acoustic
Gain of the sound system decreases by 3 dB.
This means that the volume level of the system
must be turned down for every extra mic added
in order to prevent feedback. In addition, the
amount of noise picked up increases as does
the likelihood of interference effects such as
comb-filtering.

• When multiple microphones are used, the dis­tance between microphones should be at least
three times the distance from each microphone
to its intended sound source. This will help
eliminate phase cancellation. For example, if
two microphones are each placed one foot
from their sound sources, the distance between
the microphones should be at least three feet.

(3 to 1 Rule)

• To reduce feedback and pickup of unwanted
sounds:

1) place microphone as close as practical
to desired sound source

2) place microphone as far as practical
from unwanted sound sources such as
loudspeakers and other instruments

3) aim unidirectional microphone toward
desired sound source (on-axis)

4) aim unidirectional microphone away
from undesired sound source (180
degrees off-axis for cardioid, 126
degrees off-axis for supercardioid)

5) use minimum number of microphones

• To reduce handling noise and stand thumps:

1) use an accessory shock mount (such
as the Shure A55M)

2) use an omnidirectional microphone

3) use a unidirectional microphone with
a specially designed internal shock
mount

• To reduce “pop” (explosive breath sounds
occurring with the letters “p,” “b,” and “t”):

1) mic either closer or farther than 3
inches from the mouth (because the
3-inch distance is worst)

2) place the microphone out of the path
of pop travel (to the side, above, or
below the mouth)

3) use an omnidirectional microphone

4) use a microphone with a pop filter.
This pop filter can be a ball-type grille
or an external foam windscreen

• If the sound from your loudspeakers is distort­ed even though you did not exceed a normal
mixer level, the microphone signal may be
overloading your mixer’s input. To correct this
situation, use an in-line attenuator (such as the
Shure A15AS), or use the input attenuator on
your mixer to reduce the signal level from the
microphone.

Seasoned sound engineers have developed
favorite microphone techniques through years of
experience. If you lack this experience, the sug­gestions listed on the following pages should
help you find a good starting point. These sug­gestions are not the only possibilities; other
microphones and positions may work as well or
better for your intended application.
Remember—Experiment and Listen!

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Lead vocal:

Handheld or on stand, microphone
windscreen touching lips or just a
few inches away

Backup vocals:

One microphone per singer.
Handheld near chin or stand-mounted.
T ouching lips or a few inches away

Choral groups:

1 to 3 feet above and 2 to 4 feet in
front of the first row of the choir,
aimed toward the middle row(s) of
the choir, approximately 1 micro­phone per 15-20 people

Miniature microphone clipped
outside of sound hole

Miniature microphone clipped
inside sound hole

Acoustic guitar:

8 inches from sound hole

3 inches from sound hole

4 to 8 inches from bridge

6 inches above the side, over the
bridge, and even with the front
soundboard

miniature microphone clipped
outside of sound hole

miniature microphone clipped
inside sound hole

Bassy, robust
(unless an omni
is used)

Bassy, robust
(unless an omni is
used)

Full range,
good blend,
semi-distant

Natural,
well-balanced

Bassy, less
string noise

Bassy

Very bassy, boomy,
muddy , full

Woody, warm,
mellow. Midbasy,
lacks detail

Natural, well­balanced,
slightly bright

Natural, well­balanced

Bassy, less
string noise

Minimizes feedback and leakage.
Roll off bass if desired for more
natural sound.

Minimizes feedback and leakage. Allows
engineer control of voice balances. Roll
off bass if necessary for more natural
sound when using cardioids.

Use flat-response unidirectional micro­phones, Use minimum number of
microphones needed to avoid overlap­ping pickup areas.

Good isolation. Allows freedom of
movement.

Reduces feedback.

Good starting placement when leakage
or feedback is a problem. Roll off bass
for a more natural sound (more for a
uni than an omni).

Very good isolation. Bass rolloff
needed for a natural sound.

Reduces pick and string noise.

Less pickup of ambience and leakage
than 3 feet from sound hole.

Good isolation. Allows freedom of
movement.

Reduces feedback.

Microphone Placement Tonal Balance Comments

VOCALS • STRINGS

23

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Banjo:

3 inches from center of head

3 inches from edge of head

Miniature microphone clipped to
tailpiece aiming at bridge

Violin (fiddle):

A few inches from side

Cello:

1 foot from bridge

Miniature microphone attached to
strings between bridge and tailpiece

6 inches to 1 foot out front, just
above bridge

A few inches from f-hole

Wrap microphone in foam padding
(except for grille) and put behind
bridge or between tailpiece and body

Harp:

Aiming toward player at part of
soundboard, about 2 feet away

Tape miniature microphone to
soundboard

Bassy, thumpy

Bright

Natural

Natural

Well-defined

Bright

Well-defined

Full

Full, “tight”

Natural

Somewhat
constricted

Rejects feedback and leakage.
Roll off bass for natural sound.

Rejects feedback and leakage.

Rejects feedback and leakage. Allows
freedom of movement.

Well-balanced sound.

Well-balanced sound, but little isolation.

Minimizes feedback and leakage.
Allows freedom of movement.

Natural sound.

Roll off bass if sound is too boomy.

Minimizes feedback and leakage.

See “Stereo Microphone Techniques”
section for other possibilities.

Minimizes feedback and leakage.

Microphone Placement Tonal Balance Comments

STRINGS

General string instruments (mandolin, dobro and dulcimer):

Acoustic bass (upright bass, string bass, bass violin):

24

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Grand piano:

12 inches above middle strings, 8
inches horizontally from hammers
with lid off or at full stick

8 inches above treble strings, as
above

Aiming into sound holes

6 inches over middle strings,
8 inches from hammers, with lid on
short stick

Next to the underside of raised lid,
centered on lid

Underneath the piano, aiming up at
the soundboard

Surface-mount microphone mounted
on underside of lid over lower treble
strings, horizontally close to ham­mers for brighter sound, further from
hammers for more mellow sound

T wo surface-mount microphones
positioned on the closed lid, under the
edge at its keyboard edge, approxi­mately 2/3 of the distance from mid­dle Ato each end of the keyboard

Surface-mount microphone placed
vertically on the inside of the frame,
or rim, of the piano, at or near the
apex of the piano’s curved wall

Natural,
well-balanced

Natural, well­balanced,
slightly bright

Thin, dull, hard,
constricted

Muddy, boomy,
dull, lacks attack

Bassy, full

Bassy, dull, full

Bright, well­balanced

Bright, well­balanced, strong
attack

Full, natural

Less pickup of ambience and
leakage. Move microphone(s) far­ther from hammers to reduce attack
and mechanical noises. Good coinci­dent-stereo placement. See “Stereo
Microphone T echniques” section.

Place one microphone over bass
strings and one over treble strings for
stereo. Phase cancellations may occur
if the recording is heard in mono.

Very good isolation. Sometimes
sounds good for rock music. Boost
mid-bass and treble for more natural
sound.

Improves isolation. Bass rolloff and
some treble boost required for more
natural sound.

Unobtrusive placement.

Unobtrusive placement.

Excellent isolation. Experiment
with lid height and microphone
placement on piano lid for desired
sounds.

Excellent isolation. Moving “low”
mic away from keyboard six inches
provides truer reproduction of the
bass strings while reducing damper
noise. By splaying these two mics
outward slightly, the overlap in the
middle registers can be minimized.

Excellent isolation. Minimizes
hammer and damper noise. Best if
used in conjunction with two sur­face-mount microphones mounted
to closed lid, as above.

Microphone Placement Tonal Balance Comments

STRINGS

25

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Upright piano:

Just over open top, above treble
strings

Just over open top, above bass
strings

Inside top near the bass and
treble stings

8 inches from bass side of
soundboard

8 inches from treble side of
soundboard

1 foot from center of soundboard on
hard floor or one-foot-square plate
on carpeted floor, aiming at piano.
Soundboard should face into room

Aiming at hammers from front, sever­al inches away (remove front panel)

1 to 2 feet from bell. Acouple of
instruments can play into one
microphone

Miniature microphone mounted
on bell

Natural (but
lacks deep bass),
picks up ham­mer attack

Slightly full or
tubby, picks up
hammer attack

Natural, picks up
hammer attack

Full, slightly
tubby, no
hammer attack

Thin, constricted,
no hammer attack

Natural, good
presence

Bright, picks up
hammer attack

On-axis to bell
sounds bright; to
one side sounds
natural or mellow

Bright

Good placement when only one
microphone is used.

Mike bass and treble strings for
stereo.

Minimizes feedback and leakage.
Use two microphones for stereo.

Use this placement with the
following placement for stereo.

Use this placement with the
preceding placement for stereo.

Minimize pickup of floor vibrations
by mounting microphone in low­profile shock-mounted microphone
stand.

Mike bass and treble strings for
stereo.

Close miking sounds “tight” and
minimizes feedback and leakage.
More distant placement gives fuller,
more dramatic sound.

Maximum isolation.

Microphone Placement Tonal Balance Comments

Brass (trumpet, cornet, trombone, tuba):

The sound from these instruments is very directional. Placing the mic off axis with the bell
of the instrument will result in less pickup of high frequencies.

STRINGS • WIND INSTRUMENTS

26

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

French horn:

Microphone aiming toward bell

Afew inches from and aiming into bell

A few inches from sound holes

A few inches above bell and aiming
at sound holes

Miniature microphone mounted on bell

Afew inches from area between
mouthpiece and first set of finger holes

A few inches behind player’s head,
aiming at finger holes

Woodwinds (Oboe, bassoon, etc):

About 1 foot from sound holes

A few inches from bell

Natural

Bright

Warm, full

Natural

Bright, punchy

Natural, breathy

Natural

Natural

Bright

Watch out for extreme fluctuations
on VU meter.

Minimizes feedback and leakage.

Picks up fingering noise.

Good recording technique.

Maximum isolation, up-front sound.

Pop filter or windscreen may be
required on microphone.

Reduces breath noise.

Provides well-balanced sound.

Minimizes feedback and leakage.

Microphone Placement Tonal Balance Comments

Flute:

The sound energy from a flute is projected both by the embouchure and by the first open
fingerhole. For good pickup, place the mic as close as possible to the instrument. However,
if the mic is too close to the mouth, breath noise will be apparent. Use a windscreen on the
mic to overcome this difficulty.

Saxophone:

With the saxophone, the sound is fairly well distributed between the finger holes and the bell.
Miking close to the finger holes will result in key noise. The soprano sax must be considered
separately because its bell does not curve upward. This means that, unlike all other saxo­phones, placing a microphone toward the middle of the instrument will not pick-up the sound
from the key holes and the bell simultaneously. The saxophone has sound characteristics sim­ilar to the human voice. Thus, a shaped response microphone designed for voice works well.

WIND INSTRUMENTS

27

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Harmonica:

Very close to instrument

Accordion:

Miniature microphone mounted
internally

4 inches from grille cloth at center
of speaker cone

1 inch from grille cloth at center of
speaker cone

Off-center with respect to speaker
cone

3 feet from center of speaker cone

Miniature microphone draped over
amp in front of speaker

Microphone placed behind open
back cabinet

Bass guitar amplifier/speaker:

Mike speaker as described in
Electric Guitar Amplifier section

Mike speaker as described in
Electric Guitar Amplifier section

Full, bright

Emphasized
midrange

Natural, well­balanced

Bassy

Dull or mellow

Thin, reduced
bass

Emphasized
midrange

Depends on
position

Depends on
placement

Depends on
brand of piano

Minimizes feedback and leakage.
Microphone may be cupped in hands.

Minimizes feedback and leakage.
Allows freedom of movement.

Small microphone desk stand may be
used if loudspeaker is close to floor.

Minimizes feedback and leakage.

Microphone closer to edge of
speaker cone results in duller sound.
Reduces amplifier hiss noise.

Picks up more room ambience and
leakage.

Easy setup, minimizes leakage.

Can be combined with mic in front
of cabinet, but be careful of phase
cancellation.

Improve clarity by cutting
frequencies around 250 Hz and
boosting around 1,500 Hz.

Roll off bass for clarity, roll off
highs to reduce hiss.

Microphone Placement Tonal Balance Comments

Electric guitar amplifier/speaker:

The electric guitar has sound characteristics similar to the human voice. Thus, a shaped
response microphone designed for voice works well.

WIND • ELECTRIC INSTRUMENTS

Electric keyboard amplifier/speakers:

28

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Leslie organ speaker:

Aim one microphone into top
louvers 3 inches to 1 foot away

Mike top louvers and bottom bass
speaker 3 inches to 1 foot away

Mike top louvers with two micro­phones, one close to each side. Pan
to left and right. Mike bottom bass
speaker 3 inches to 1 foot away and
pan its signal to center

Natural, lacks
deep bass

Natural, well­balanced

Natural,
well-balanced

Good one-mike pickup.

Excellent overall sound.

Stereo effect.

Microphone Placement Tonal Balance Comments

Drum kit:

In most sound reinforcement systems, the drum set is miked with each drum having its own
mic. Using microphones with tight polar patterns on toms helps to isolate the sound from each
drum. It is possible to share one mic with two toms, but then, a microphone with a wider
polar pattern should be used. The snare requires a mic that can handle very high SPL, so a
dynamic mic is usually chosen. To avoid picking up the hi-hat in the snare mic, aim the null
of the snare mic towards the hi-hat. The brilliance and high frequencies of cymbals are picked
up best by a flat response condenser mic.

DRUM KIT

Front View T op View

1. Overhead-Cymbals:

One microphone over center of drum
set, about 1 foot above drummer’s
head (Position A); or use two spaced
or crossed microphones for stereo
(Positions Aor B). See “Stereo
Microphone T echniques” section

Natural; sounds
like drummer
hears set

Picks up ambience and leakage. For
cymbal pickup only, roll off low fre­quencies. Boost at 10,000 Hz for
added sizzle. To reduce excessive
cymbal ringing, apply masking tape
in radial strips from bell to rim.

29

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

2. Snare drum:

Just above top head at edge of
drum, aiming at top head. Coming
in from front of set on boom
(Position C); or miniature micro­phone mounted directly on drum

Remove front head if necessary .
Mount microphone on boom arm
inside drum a few inches from beater
head, about 1/3 of way in from edge
of head (Position D); or place sur­face-mount microphone inside drum,
on damping material, with micro­phone element facing beater head

4. Tom-toms:

One microphone between every two
tom-toms, close to top heads (Position
E); or one microphone just above
each tom-tom rim, aiming at top
head (Position F); or one microphone
inside each tom-tom with bottom
head removed; or miniature micro­phone mounted directly on drum

5. Hi-hat:

Aim microphone down towards the
cymbals, a few inches over edge
away from drummer (Position G).
Or angle snare drum microphone
slightly toward hi-hat to pick up
both snare and hi-hat

Full, smooth

Full, good
impact

Full, good
impact

Natural, bright

Tape gauze pad or handkerchief on
top head to tighten sound. Boost at
5,000 Hz for attack, if necessary.

Put pillow or blanket on bottom of
drum against beater head to tighten
beat. Use wooden beater, or loosen
head, or boost around 2,500 Hz for
more impact and punch.

Inside drum gives best isolation.
Boost at 5,000 Hz for attack, if
necessary.

Place microphone or adjust cymbal
height so that puff of air from closing
hi-hat cymbals misses mike. Roll off
bass to reduce low-frequency leak­age. To reduce hi-hate leakage into
snare-drum microphone, use small
cymbals vertically spaced 1/2” apart.

Microphone Placement Tonal Balance Comments

3. Bass drum (kick drum):

Placing a pad of paper towels where the beater hits the drum will lessen boominess. If you
get rattling or buzzing problems with the drum, put masking tape across the drum head to
damp out these nuisances. Placing the mic off center will pick up more overtones.

DRUM KIT

30

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

6. Snare, hi-hat and high tom:

Place single microphone a few inch­es from snare drum edge, next to
high tom, just above top head of tom.
Microphone comes in from front of
the set on a boom (Position H)

7. Cymbals, floor tom and high tom:

Using single microphone, place its
grille just above floor tom, aiming
up toward cymbals and one of high
tomes (Position I)

Natural

Natural

In combination with Placements 3
and 7, provides good pickup with
minimum number of microphones.
Tight sound with little leakage.

In combination with Placements 3
and 6, provides good pickup with
minimum number of microphones.
Tight sound with little leakage.

Microphone Placement Tonal Balance Comments

One microphone: Use Placement 1. Placement 6 may work if the drummer limits playing to
one side of the drum set.

Two microphones: Placements 1 and 3; or 3 and 6.

Three microphones: Placements 1, 2, and 3; or 3, 6, and 7.

Four microphones: Placements 1, 2, 3, and 4.

Five microphones: Placements 1, 2, 3, 4, and 5.

More microphones: Increase number of tom-tom microphones as needed. Use a small micro­phone mixer (such as the Shure M268) to submix multiple drum microphones into one channel.

Timbales, congas, bongos:

One microphone aiming down
between pair of drums, just above
top heads

Tambourine:

One microphone placed 6 to 12
inches from instrument

Natural

Natural

Provides full sound with good
attack.

Experiment with distance and
angles if sound is too bright.

DRUM KIT

31

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Steel Drums:

Tenor, Second Pan, Guitar

One microphone placed 4 inches
above each pan

Microphone placed underneath pan

Cello, Bass

One microphone placed 4 - 6
inches above each pan

Xylophone, marimba, vibraphone:

Two microphones aiming down
toward instrument, about 1 1/2 feet
above it, spaced 2 feet apart, or angled
135

º

apart with grilles touching

Glockenspiel:

One microphone placed 4 - 6 inches
above bars

Bright, with
plenty of attack

Natural

Natural

Bright, with lots
of attack.

Allow clearance for movement of pan.

Decent if used for tenor or second pans.
T oo boomy with lower voiced pans.

Can double up pans to a single
microphone.

Pan two microphones to left and right
for stereo. See “Stereo Microphone
Techniques” section.

For less attack, use rubber mallets
instead of metal mallets. Plastic
mallets will give a medium attack.

Microphone Placement Tonal Balance Comments

Stage area miking Tonal Balance Comments

Downstage:

Surface-mount microphones along
front of stage aimed upstage, one
microphone center stage; use stage
left and stage right mics as needed,
approximately 1 per 10-15 feet

Upstage:

Microphones suspended 8 -10 feet
above stage aimed upstage, one
microphone center stage; use stage
left and stage right mics as needed,
approximately 1 per 10-15 feet

Spot pickup:

Use wireless microphones on
principal actors; mics concealed in
set; “shotgun” microphones from
above or below

Voice range,
semi-distant

Voice range,
semi-distant

Voice range,
on mic

Use flat response, unidirectional
microphones. Use minimum num­ber of microphones needed to avoid
overlapping pickup area. Use shock
mount if needed.

Use flat response, unidirectional
microphones. Use minimum
number of microphones needed to
avoid overlapping pickup area.

Multiple wireless systems must
utilize different frequencies. Use
lavaliere or handheld microphones
as appropriate.

DRUM KIT/STAGE

32

STEREO MICROPHONE TECHNIQUES

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Stereo Microphone Techniques

These methods are recommended for pickup of
orchestras, bands, choirs, pipe organs, quartets,
soloists. They also may work for jazz ensem­bles, and are often used on overhead drums and
close-miked piano.

Use two microphones mounted on a single stand
with a stereo microphone stand adapter (such as
the Shure A27M). Or mount 2 or 3 microphones
on separate stands. Set the microphones in the
desired stereo pickup arrangement (see below).

For sound reinforcement, stereo mic techniques
are only warranted for a stereo sound system and
even then, they are generally only effective for
large individual instruments, such as piano or
miramba, or small instrument groups, such as
drum kit, string section or vocal chorus.
Relatively close placement is necessary to
achieve useable gain-before-feedback.

Coincident T echniques

Microphone diaphragms
close together and aligned
vertically; microphones
angled apart. Example:
135

0

angling (X-Y).

MS (Mid-Side)

A front-facing cardioid car­tridge and a side-facing bidi­rectional cartridge are mount­ed in a single housing. Their
outputs are combined in a
matrix circuit to yield discrete
left and right outputs.

Near-Coincident
Techniques

Microphones angled and
spaced apart 6 to 10 inches
between grilles. Examples:
110

0

angled, 7-inch spacing.

Comments

Tends to provide a narrow
stereo spread (the reproduced
ensemble does not always
spread all the way between the
pair of playback loud-speak­ers). Good imaging. Mono­compatible.

Comments

Provides good stereo spread,
excellent stereo imaging and
localization. Some types allow
adjustable stereo control.
Mono-compatible.

Comments

Tends to provide accurate
image localization.

Musical Ensemble

(T op View)

Musical Ensemble

(T op View)

Musical Ensemble

(T op View)

33

STEREO MICROPHONE TECHNIQUES

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

Spaced Techniques

Two microphones spaced
several feet apart horizontal­ly, both aiming straight ahead
toward ensemble. Example:
Microphones 3 to 10 feet
apart.

Three microphones spaced
several feet apart horizontal­ly, aiming straight ahead
toward ensemble. Center
microphone signal is split
equally to both channels.
Example: Microphones 5
feet apart.

Comments

Tends to provide exaggerated
separation unless microphone
spacing is 3 feet. However,
spacing the microphones 10
feet apart improves overall
coverage. Produces vague
imaging for off-center sound
sources. Provides a “warm”
sense of ambience.

Improved localization com­pared to two spaced micro­phones.

Musical Ensemble

(T op View)

Musical Ensemble

(T op View)

34

BETA58A™

SM58

BETA57A™

SM57
BG3.1
BG2.1
BG1.1

BETA87

®

SM87

BG5.1

WH10XLR

SM10A
SM12A

512

SM81

SM7

BETA87

®

SM87

BG5.1

SM81
SM94

BG4.1

VOCALS

PERFORMANCE

VOCAL (dynamic)

PERFORMANCE

VOCAL (condenser)

HEADWORN

VOCAL

STUDIO

VOCAL

ENSEMBLE

VOCAL

BETA56™

BETA57A™

SM57
BG3.1
BG2.1
BG6.1

BETA52™

SM7

BETA57A™

BETA56™

SM57

BETA52™

SM91A

BETA57A™

SM57

BETA57A™

BETA56™

SM57

BG3.1

SM98A

1

BETA57A™

BETA56™

SM57

BG6.1

INSTRUMENTS

GUITAR

AMPLIFIER

BASS

AMPLIFIER

KICK

DRUM

SNARE

DRUM

TOMS

RACK & FLOOR

SM81

SM94
BG4.1

SM98A

BETA56™

BETA57A™

SM57

SM81
SM94

BG4.1

SM81

BETA57A™

SM57

SM81
SM91

BG4.1

OVERHEAD

CYMBALS HIGH HAT

2

CONGA MALLET

INSTRUMENTS

2

MARIMBA & OTHER

PERCUSSION

2

PIANO

2

SM81

SM94
BG4.1
SM11

4

SM98A

4

BETA52™

SM81
SM94

BG4.1

SM98A

3

BETA56™

BETA57A™

SM57

SM81

SM98A

BG4.1

SM98A

3

SM7

BETA56™

BETA57A™

SM57

STRINGS ACOUSTIC

BASS

BRASS

INSTRUMENTS

WOODWINDS SAXAPHONE

SM81
SM94

BG4.1

BETA57A™

SM57

SM11

4

520D “Green Bullet”

SM57
SM58

BETA57A™

BETA56™

SM57

BG3.1

SM91A

SM81
SM94

BG4.1

SM81 (pair)
SM94 (pair)

BG4.1 (pair)

VP88

ACOUSTIC

GUITAR

HARMONICA LESLIE

SPEAKER

ORCHESTRA

2

LIVE CONCERT

RECORDING OR

STEREO

PICKUP/AMBIENCE

2

SM81
SM94

BG4.1

SM58S

565
BG3.1
BG2.1

BG1.1 (Hi or Lo Z)

SAMPLING KARAOKE

This guide is an aid in selecting microphones for various applications. Microphone sound quality and
appearance are subject to specific, acoustic environments, application technique and personal taste.

1

With A98MK drum mount kit.

2

For single point stereo miking, use VP88 MS Stereo Microphone.

3

Bell-mounted with A98KCS clamp.

4

With RK279 mounting kit for instrument applications.

SHURE MICROPHONE SELECTION GUIDE

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.

Absorption-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.

Ambience-Room acoustics or natural
reverberation.

Amplitude-The strength or level of sound
pressure or voltage.

Audio Chain-The series of interconnected audio
equipment used for recording or PA.

Backplate-The solid conductive disk that forms
the fixed half of a condenser element.

Balanced-A circuit that carries information by
means of two equal but opposite polarity signals,
on two conductors.

Bidirectional Microphone-Amicrophone that
picks up equally from two opposite directions.
The angle of best rejection is 90 deg. from the
front (or rear) of the microphone, that is, directly
at the sides.

Boundary/Surface Microphone-Amicrophone
designed to be mounted on an acoustically reflec­tive surface.

Cardioid Microphone-Aunidirectional micro­phone with moderately wide front pickup (131
deg.). Angle of best rejection is 180 deg. 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).

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.

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.

Current-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.

Diaphragm-The thin membrane in a microphone
which moves in response to sound waves.

Diffraction-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-Amicrophone 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.

35

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

GLOSSARY

36

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

GLOSSARY

Dynamic Range-The range of amplitude of a
sound source or the range of sound level that a
microphone can successfully pick up.

Echo-Reflection of sound that is delayed long
enough (more than about 50 msec.) to be heard
as a distinct repetition of the original sound.

Electret-Amaterial (such as Teflon) that can
retain a permanent electric charge.

EQ-Equalization or tone control to shape
frequency response in some desired way .

Feedback-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 frequencies.

Frequency-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-Agraph showing how a
microphone responds to various sound frequen­cies. It is a plot of electrical output (in decibels)
vs. frequency (in Hertz).

Fundamental-The lowest frequency component
of a complex waveform such as musical note. It
establishes the basic pitch of the note.

Gain-Amplification of sound level or voltage.
Gain-Before-Feedback-The amount of gain that

can be achieved in a sound system before
feedback or ringing occurs.

Harmonic-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.

Hypercardioid-Aunidirectional microphone
with tighter front pickup (105 deg.) than a
supercardioid, but with more rear pickup. Angle
of best rejection is about 110 deg. from the front
of the microphone.

Impedance-In an electrical circuit, opposition to
the flow of alternating current, measured in ohms.
A high impedance microphone has an impedance
of 10,000 ohms or more. Alow impedance
microphone has an impedance of 50 to 600 ohms.

Interference-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.

Isolation-Freedom from leakage; ability to reject
unwanted sounds.

Leakage-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.

NAG-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.

Noise-Unwanted electrical or acoustic interference.
Noise Canceling-A microphone that rejects

ambient or distant sound.
NOM-Number of open microphones in a sound

system. Decreases gain-before-feedback by 3dB
everytime NOM doubles.

37

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

GLOSSARY

Omnidirectional Microphone-Amicrophone that
picks up sound equally well from all directions.

Overload-Exceeding the signal level capability of
a microphone or electrical circuit.

PAG-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.

Phase-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
response.

Pitch-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.

Polarization-The charge or voltage on a
condenser microphone element.

Pop Filter-An acoustically transparent shield
around a microphone cartridge that reduces
popping sounds. Often a ball-shaped grille, foam
cover or fabric barrier.

Pop-A thump of explosive breath sound produced
when a puff of air from the mouth strikes the
microphone diaphragm. Occurs most often with
“p,” “t,” and “b” sounds.

Presence Peak-An increase in microphone output
in the “presence” frequency range of 2000 Hz to
10,000 Hz. Apresence 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 ft.). Does not occur with omnidirectional
microphones.

Rear Lobe-A region of pickup at the rear of a
supercardioid or hypercardioid microphone polar
pattern. Abidirectional microphone has a rear
lobe equal to its front pickup.

Reflection-The bouncing of sound waves back
from an object or surface which is physically
larger than the wavelength of the sound.

Refraction-The bending of sound waves by a
change in the density of the transmission medium,
such as temperature gradients in air due to wind.

Resistance-The opposition to the flow of current
in an electrical circuit. It is analogous to the
friction of fluid flowing in a pipe.

Reverberation-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.

Rolloff-A gradual decrease in response below or
above some specified frequency .

Sensitivity-The electrical output that a micro­phone produces for a given sound pressure level.

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.

38

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

GLOSSARY

Sound Chain-The series of interconnected audio
equipment used for recording or PA.

Sound Reinforcement-Amplification of live
sound sources.

Speed of Sound-The speed of sound waves,
about 1130 feet per second in air.

SPL-Sound Pressure Level is the loudness of
sound relative to a reference level of 0.0002
microbars.

Standing Wave-Astationary sound wave that is
reinforced by reflection between two parallel
surfaces that are spaced a wavelength apart.

Supercardioid Microphone-Aunidirectional
microphone with tighter front pickup angle (115
deg.) than a cardioid , but with some rear pickup.
Angle of best rejection is 126 deg. from the front
of the microphone, that is, 54 deg. from the rear.

Timbre-The characteristic tone of a voice or
instrument; a function of harmonics.

Transducer-A device that converts one form of
energy to another. Amicrophone 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 input.

Unbalanced-A circuit that carries information by
means of one signal on a single conductor.

Unidirectional Microphone-Amicrophone 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.

Voltage-The potential difference in an electric
circuit. Analogous to the pressure on fluid flow­ing in a pipe.

Wavelength-The physical distance between the
start and end of one cycle of a soundwave.

RICK WALLER

Now residing in the
Chicago area, Rick grew up near Peoria, Illinois.
An interest in the technical and musical aspects
of audio has led him to pursue a career as
both engineer and musician. He received a
BS degree in Electrical Engineering from the
University of Illinois at Urbana/Champaign,
where he specialized in acoustics, audio
synthesis and radio frequency theory. Rick is
an avid keyboardist, drummer and home
theater hobbyist and has also worked as a
sound engineer and disc jockey. Currently he
is an associate in the Applications Engineering
Group at Shure Brothers. In this capacity Rick
provides technical support to domestic and
international customers, writing and conducting
seminars on wired and wireless microphones,
mixers and other audio topics.

JOHN BOUDREAU

John, a life­long Chicago native, has had extensive
experience as a musician, a recording engineer,
and a composer. His desire to better combine
the artistic and technical aspects of music led
him to a career in the audio field.

Having received a BS degree in Music
Business from Elmhurst College, John
performed and composed for both a Jazz and
a Rock band prior to joining Shure Brothers
in 1994 as an associate in the Applications
Engineering group. At Shure, John leads
many audio product training seminars and
clinics, with an eye to helping musicians and

others affiliated with the field use technology
to better fulfill their artistic interpretations.

John continues to pursue his interests as a
live and recorded sound engineer for local
bands and venues, as well as writing and
recording for his own band.

TIM VEAR

Tim is a native of Chicago
who has come to the audio field as a way of com­bining a lifelong interest in both entertainment
and science. He has worked as an engineer in live
sound, recording and broadcast, has operated his
own recording studio and sound company , and has
played music professionally since high school.

At the University of Illinois, Urbana­Champaign, Tim earned a BS in Aeronautical
and Astronautical Engineering with a minor in
Electrical Engineering. During this time he also
worked as chief technician for both the Speech
and Hearing Science and Linguistics departments.

In his tenure at Shure Brothers, Tim has
served in a technical support role for the sales and
marketing departments, providing product and
applications training for Shure customers, dealers,
installers, and company staff. He has presented
seminars for a variety of domestic and international
audiences, including the National Systems
contractors Association, the Audio Engineering
Society and the Society of Broadcast Engineers.
Tim has authored several publications for
Shure Brothers and his articles have appeared
in Recording Engineer/Producer, Live Sound
Engineering, Creator, and other publications.

39

MICROPHONE

TECHNIQUES FOR MUSIC

SOUND

REINFORCEMENT

ABOUT THE AUTHORS

Shure Brothers Incorporated

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Phone: 800-25-SHURE Fax: [phone]

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Outside of U.S. and Europe, Phone: [phone] Fax: [phone]

http://www.shure.com

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