SRS Training Introduction 2010

Microphone Techniques for Live Sound Reinforcement

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
capabilities 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.

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Condenser microphones are based on an electrically-charged diaphragm/backplate assembly which forms a

sound-sensitive capacitor. Here, sound waves vibrate a very thin metal or metal-coated-plastic diaphragm. The
diaphragm is mounted just in front of a rigid metal or metal-coated-ceramic backplate. In electrical terms this
assembly or element is known as a capacitor (historically called a “condenser”), which has the ability to store a
charge or voltage. When the element is charged, an electric field is created between the diaphragm and the
backplate, proportional to the spacing between them. It is the variation of this spacing, due to the motion of the
diaphragm relative to the backplate, that produces the electrical signal corresponding to the sound picked up by
a condenser microphone.

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. Non-electret 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.

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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ADDITIONAL READING:

Ribbon microphones employ a transduction method that is similar to that of dynamics. Figure 10-3 illustrates
the construction of a typical ribbon element. A very light, thin, corrugated metal ribbon, Figure 10-3 (a), is
stretched within the air gap of a powerful magnet (b). The ribbon is clamped at the ends, but is free to move
throughout its length. When sound strikes the ribbon, the ribbon vibrates in response. As is the case with the
dynamic coil element, the moving ribbon cuts the magnetic lines of force in the air gap, and a voltage is thereby
induced in the ribbon. The voltage is very small and the ribbon impedance very low, so all ribbon microphones
incorporate a built-in transformer. The transformer serves the dual functions of boosting the signal voltage and
isolating the ribbon impedance from the load presented by the input to which the microphone is connected.

Early ribbon microphones were extremely fragile. The ribbon could be damaged simply by blowing or coughing
into the microphone! Not many microphone manufacturers now make ribbon units, but those that are available
are much more rugged than older units. All but a few modern ribbon mics remain more fragile than dynamic or
condenser units, so they are used primarily in recording (a couple of notable exceptions are used for
reinforcement).

Ribbon microphones usually have excellent sonic characteristics, with great warmth and gentle high-frequency
response. They also have excellent transient response and very low self-noise. For these reasons, some ribbon
mics are prized as vocal microphones, and are also very effective with acoustic instruments.

The carbon type is among the earliest microphone elements ever developed. Figure 10-4 illustrates the
construction of a typical carbon element. A small cup, Figure 10-4 (a), is packed with pulverized carbon and
enclosed at one end by a brass disk called a button (b), which is coupled to a circular metal diaphragm Cc). The
button and a back plate at the rear of the cylinder form the connection terminals. A battery (d) provides an
activating voltage across the carbon. When sound strikes the diaphragm, the carbon granules in the button
vibrate, becoming alternately more and less dense as the diaphragm moves. The electrical resistance of the
carbon thereby fluctuates, and converts the battery voltage into a corresponding fluctuating current that is an
electrical representation of the sound. The current is stepped up by a transformer (e), which also serves to
isolate the low impedance of the element from that of the input to which it is connected, and to block the battery
DC from the input.

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Carbon microphones are not known for excellent sonic characteristics, but they are quite inexpensive, and
rugged. For this reason, they are still widely used in utility sound applications. (The standard telephone mic
element has long been a carbon type, although dynamic mics are used in many newer phones.) Carbon
microphones can lose some efficiency and become noisy if the granules in the button become compacted,
but simply tapping the element against a hard surface usually cures the problem.

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 polariziing voltage for the element tself.
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 souorce, 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.

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Transient Response
Transient response refers to the ability of a microphone to respond to a rapidly changing sound wave. A good
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.

Frequency response – The output level or sensitivity of the microphone over its operating range from lowest to
highest frequency.

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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.

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.

A microphone 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 presence peak or rise. A microphone 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.

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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

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.

Directionality – A microphone’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.

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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 cardioids (heart-shaped) response. This has the most
sensitivity at 0 degrees (on-axis) and is least sensitive at 180 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.

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. A good cardioid type has at least 15-20 dB of rear
rejection. Supercardioid

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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

Using Directional Patterns to Reject Unwanted Sources

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.

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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).

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Unidirectional microphones can not only help 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 referred
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.(high-Z). 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. A balanced 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). A balanced 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.

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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 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.

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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.

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. A microphone 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. A microphone placed at
such distance and direction tends to pick up a natural or well-balanced 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.

Dynamic range - the range of volume of an 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 difference in

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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.

Instrument Loudspeakers
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.
A cabinet 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. A unidirectional 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

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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!

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.

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 – A sound wave will typically bend around obstacles in its path which are smaller than its
wavelength. Because a low frequency 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.

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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 nondirectional. Reverberation
is an example of nondirectional 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 microphone 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.

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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
frequencies may be absorbed by surface materials such as carpeting or other acoustic treatments. Low

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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.

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. A guitar 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

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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.

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 11/2 times as far from that surface as it is from
its intended sound source. Again, this insures minimum audibility of interference effects.

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.

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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. A mic 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.

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

Where: Df = distance from sound source to furthest listener

Dn = 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 inverse-square law, which says that the sound level decreases by 6dB
each time the distance to the source doubles. For example, the sound level (without a sound system) at the first
row of the audience (10 feet from the stage) might be a comfortable 85dB. At the last row of the audience (80
feet from the stage) the level will be 18dB less or 67dB. In this case the sound system needs to provide 18dB of
gain so that the last row can hear at the same level as the first row. The limitation in real-world sound systems is
not how loud the system can get with a recorded sound source but rather how loud it can get with a microphone
as its input. The maximum loudness is ultimately limited by acoustic feedback.

The amount of gain-before-feedback that a sound reinforcement system can provide may be estimated
mathematically. This Potential Acoustic Gain involves the distances between sound system components, the
number of open mics, and other variables. The system will be sufficient if the calculated Potential Acoustic Gain

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(PAG) is equal to or greater than the Needed Acoustic Gain (NAG). Below is an illustration showing the key
distances.

The simplified PAG equation is:

PAG = 20 (log D1 - log D2 + log D0 - log Ds) -10 log NOM -6

Where: PAG = Potential Acoustic Gain (in dB)
Ds = distance from sound source to microphone
D0 = distance from sound source to listener
D1 = distance from microphone to loudspeaker
D2 = 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 increases gain-before feed back 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

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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 microphone. 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:

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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
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 the microphone signal may be overloading your mixer’s
input. To correct this situation, use an in-line attenuator, 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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