AUDIO SYSTEMS

1. Hearing and Sound

a. Hearing and Sound - Introduction
This section includes the following topics:
 The Science of Sound
 Human Perception of Sound
Upon completion of this section, you should be able to:
 Define the terms Frequency, Octaves, and Harmonics.
 Summarize the Inverse Square Law.

b. The Science of Sound

Hearing and Sound
Much of what you need to know as an audiovisual professional revolves around sound: how sounds are generated,
how sound moves through a medium (such as air), how you might control its propagation, how sound interacts with
the environment, and how you receive and process the sound.

To understand sound, you should understand how your hearing works. Your ears are complex organs that have
several distinct parts: outer ear, ear canal, eardrum, middle ear, and cochlea. To receive and process sound, these
parts convert sound waves into mechanical movement and then to electrical signals that your brain translates into
usable information. Damage to any of these parts can reduce your ability to perceive sounds.

Sound Waves
Vibration is a way to produce sound.
A stringed instrument, like a violin or guitar, is an excellent example of this concept. When you pluck a string on a
violin, the motion of the string displaces air molecules. As those molecules move, they push against, or “compress,”
the air molecules next to them.
As the string reverses its direction, it pulls at the molecules that surround it. Because air is an elastic medium, the
displaced molecules transmit this back-and-forth motion to the molecules surrounding them. This displacement
moves the sound as a waveform out and away from the source that generated it.
The pushing together of molecules is called compression, and the pulling apart is called rarefaction. These are
areas of high and low pressure in the elastic medium. In AV, you are usually concerned with sound moving in air,
but there may be occasions that you are interested in its movement through other elastic mediums.
Glossary
Compression
In the context of networking, compression is a process that reduces very large files to smaller, manageable sizes.
Compressed files discard unnecessary information, which reduces the size of digital files and makes them easier to
transmit and store. This process is used extensively in computer applications, such as streaming audio or visual
content over the Internet.
In the context of acoustics, compression is an increase in density and pressure in a medium, such as air, caused
by the passage of a sound wave. ref
Glossary
Rarefaction
Rarefaction is a decrease in density and pressure in a medium, such as air, caused by the passage of a sound
wave. ref

A visual example of the principle of sound waves is a rock thrown into another elastic medium, water. The rock
creates a disturbance (displacement of water molecules) in the medium. Water molecules are pushed together in
compression and pulled apart in rarefaction, creating the peaks and valleys, or ripples, we see on the surface of the
water. We also see that these waves of energy move away from the source in concentric circles as the energy is
transferred to nearby molecules.

Besides vibration, sound in air can also be generated by sudden increases in velocity (speed) or turbulence. This
helps to explain why you sometimes hear sound coming from the ducts in an air ventilation system.
Wavelength
Sound waves have a physical length in addition to having a particular loudness or intensity.
Wavelength is the physical distance between two points exactly one cycle apart in a waveform. Wavelength
measures the distance between two points that occur at the same place.
To understand wavelength, pick any point on the wave below. Now move along the wave and find the next
occurrence of that exact spot. That is a wavelength.
This wave is one simple, single wave, or sine wave.
The middle horizontal line, known as the zero reference line or reference level, represents the molecules at their
rest position.

c. Human Perception of Sound

Logarithms
A logarithm of a number is how many times the number 10 must be multiplied by itself to get a certain value. For
example, the log of 10,000 is 4 and the log of 0.0001 is -4.

Logarithmic scales make the ratio values easier to express. For example, the ratio between the threshold of
hearing (when sounds become audible) and the threshold of pain is 1 to 1,000,000. No one wants to count that
many zeros.

LO glossary element icon

Glossary
Logarithm
A logarithm is the exponent of base 10 that equals the value of a number.

Think of a standard ruler. Each unit on the ruler represents a unit of one (whether it’s an inch, millimeter, etc.)
wherever it is located on the ruler. It’s a one-to-one relationship between the units shown and the units
represented. This is a linear scale.
What if the value of each unit on that ruler represented something other than a single unit of one? What if each time
you moved to the right, each unit represented ten times as much as before? Or, each time you moved one unit to
the left it represented one-tenth as much as before? In this case, comparing adjacent units on a scale would
represent a ratio of 1:10 or 10:1. This is a logarithmic scale.
Look at the graph below. The top row represents a linear scale and the bottom row represents a logarithmic scale.
You can write 10 * 10 * 10 = 1,000 or 103 = 1,000 (10 multiplied by itself three times). This would be a logarithm
with a base of 10 and an exponent of 3 as a shortcut for an equation that uses multiplication.

Therefore, you could substitute 101, 102 and 103 at the bottom of the graph for the numbers 10, 100 and 1,000.
Humans perceive differences in sound levels logarithmically, not linearly. Because of this, a base 10 logarithmic
scale is used to measure, record and discuss sound level differences.

If you used a linear scale to describe the perceived difference in sound pressure level from the threshold of hearing
to the threshold of pain, you would need to use numbers from 1 to well over 3 million.

Not only is your perception of sound intensity logarithmic, but so are your other senses. The psychophysical law (or
Fechner-Weber law) describes this generalization in psychology. It states that the intensity of a sensation is
proportional to the logarithm of the intensity of the stimulus causing it.

Human sensations such as sight, hearing or touch vary as the logarithm of the stimulus.

Decibels
Because decibel measurements are used in many aspects of the AV industry it is important for you to understand
what decibels are and how they apply to AV.
LO glossary element icon
Glossary
Decibel
Decibel (dB) describes a base-ten logarithmic relationship of a power ratio between two numbers. A decibel uses a
logarithmic scale to describe ratios with a very large range of values that can vary over several orders of
magnitude. A decibel is also used for quantifying differences in voltage, distance and sound pressure as they relate
to power.

The basic unit, the Bel, was named after telecommunications pioneer Alexander Graham Bell. As a unit of
measurement, the Bel was too large for practical purposes, so the deci-bel (one tenth of a Bel), or dB, is used
instead. It is a logarithmic scale used to describe ratios with a very large range of values.

The decibel is a unit of measurement used to describe a base 10 or base 20 logarithmic relationship of a power
ratio between two numbers.

Decibels are also used for quantifying differences in voltage, distance, and sound pressure as they relate to power.
When you are quantifying differences, the numbers being compared to one another must be of the same type. For
example, you could compare one voltage to another voltage. You cannot compare a voltage to a wattage.

You could also compare a number or measurement to a known reference level. The amount of increase or
decrease from the chosen reference level is what the decibel system measures. This is done with sound pressure
levels, where you compare a sound pressure level measurement to the threshold of hearing reference of 0 dBSPL.

And remember, since it's a logarithm, whether the increase is from 1 to 100 of whatever you're measuring, or 100
to 10,000, for example, the increase in both cases (in base 10) is still 20 decibels (dB).

Decibels Equations
We can state the difference in decibels for two powers or voltages or distances using the following equations:
10 times the logarithm of the ratio (of the two numbers we are comparing) describes the difference in decibels if we
are comparing two power values. For example the equation:
dB = 10 * log (P1 / P2) would give us the difference in decibels if we compared one power (P1) in watts against
another power in watts (P2).

20 times the logarithm of the ratio describes the difference in decibels if we are comparing two voltages or
distances. For example the equations:
dB = 20 * log (V1 / V2) would give us the difference in decibels if we compared one voltage (V1) against another
voltage (V2).
dB = 20 * log (D1 / D2) would give us the difference in decibels if we compared the sound pressure level at one
distance from the source (D1) as compared to the sound pressure level at some other distance from the source
(D2).
Now you know that you perceive differences in sound levels logarithmically, and that you use the decibel for a
logarithmic scale, let’s explore further how the decibel is used.

Decibel Formula for Power

The formula for calculating a decibel change for power is:
dB = 10 * log (P1 / Pr)
Where:
dB = The change in decibels
P1 = The new or measured power measurement
Pr = The original or reference power measurement
The result of this formula will be either positive or negative. If it is positive, the result is an increase, or gain. If it is
negative, the result is a decrease, or loss.

Decibel Formula for Distance

The formula for calculating decibel changes in sound pressure level over distance is:
dB = 20 * log (D1 / D2)
Where:
dB = The change in decibels
D1 = The original or reference distance
D2 = The new or measured distance
The result of this calculation will be either positive or negative. If it is positive, the result is an increase, or gain. If it
is negative, the result is a decrease, or loss.

Decibel Formula for Voltage

The formula for calculating decibel changes in voltage is:
dB = 20 * log (V1 / Vr)
Where:
dB = The change in decibels
V1 = The new or measured voltage
Vr = The original or reference voltage
The result of this calculation will be either positive or negative. If it is positive, the result is an increase, or gain. If it
is negative, the result is a decrease, or loss.

Using the Decibel

The 20log Equation can compare the difference in decibels between two sound pressure levels. It can also compare a
sound pressure level against a known sound pressure reference.

Decibels are comparisons, not units of absolute quantity. For a measured SPL to be meaningful, it has to have a
reference. It would be like saying, "it is ten degrees hotter today than it was yesterday." Without knowing yesterday’s
temperature, you cannot know today’s temperature.

In the case of sound pressure, the reference for 0 dB is 0.00002 Pascals (or 0.0002 dyne/cm2). This is the threshold
of hearing. Therefore, an 85 dB SPL measurement is 85 decibels of sound pressure, referenced to the threshold of
hearing. We can write it as 85 dBSPL, where dBSPL is dB referenced to the threshold of hearing.

Reference levels are also used for other measurements. We commonly use dBu and dBV to express voltage levels,
and dBm for electrical power levels.

For example, 0 dBu = 0.775 volts while 0 dBV = 1.0 volt, and 0 dBm = 0.001 watts. While -20 dBu is a voltage less than
0 dBu it does not mean a negative voltage, but a voltage less than 0.775 volts.

The chart below shows some of the acoustic and electrical references for the decibel.

Unit Reference Level

dBW 1 Watt

dBm 0.001 Watt

dBV 1 Volt

dBu 0.775 Volts

dB SPL 0.00002 Pascals

These are some accepted generalities in relation to human hearing:

 A one-decibel change is the smallest perceptible change noticeable. Unless listening very carefully, most
people will not discern a one-decibel change.
 A “just noticeable” change, either louder or softer, requires a three-decibel change (e.g., 85 dB SPL to 88 dB
SPL).
 A ten-decibel change is required for us to perceive either a change as twice as loud as before or one-half as
loud as before. (For example, a change from 85 dB SPL to 95 dB SPL is perceived to be twice as loud as
before).

Inverse Square Law and Sound

Sound waves move away from their source. You can observe this principle when you throw a rock into the water;
the waves of energy move away from the source of molecule disturbance in concentric circles. This occurs as the
energy is transferred to nearby molecules.

With the rock and water illustration, you can only see the energy spreading out on the horizontal plane of the water
surface. When a sound is generated, however, the energy is spread spherically in all directions.

Regardless of whether the energy is propagating horizontally or spherically, you can follow the spread of the
energy by using the inverse square law. The inverse square law states that sound energy is inversely proportional
to the square of the distance from the source. Every time the distance doubles from the source of the energy, the
energy spreads out and covers four times the area it did before.
Twenty times the logarithm of the ratio describes the difference in decibels if we are comparing two voltages or
distances.

For example, if you are 5 meters from an energy source and you double your distance from the source to 10
meters, the energy must now cover an area four times larger than it did at the 5-meter distance. Every time you
double the distance from the source, you increase the surface area of the sphere four times. Subsequently, the
energy unit per unit of area is one-quarter of what it was previously.

Quick Tip: Inverse Square Law

The inverse square law does not just apply to sound. It applies to other things such as light, gravity and the electric
field.
Sound pressure is reduced by 6 dB every time the distance from the source is doubled. Conversely, the sound
pressure increases by 6 dB when the distance from the source is reduced by one-half. This is often referred to as
the “6 dB per doubling” rule.

In practice, the “6 dB per doubling” rule occurs only in a “near-field” or “free field” environment. These terms
describe the space around a sound source, but they do not include the energy that is reflected back by boundaries
such as walls, ceilings, and floors.

d. Hearing and Sound Section Test

2. Acoustics
a. Acoustics - Introduction
This section includes the following topics:
 Acoustics
Upon completion of this section, you should be able to:
 Define acoustics and identify two acoustical factors that affect audio systems.

b. Acoustics
Acoustics
Acoustics is a branch of science that is focused on the qualities and characteristics of sound waves. As you can
see from this definition, the study of acoustics covers many topics. It is more than simply hanging some fabrics on a
wall to solve an “acoustical problem.”
The study of acoustics includes:
 Production
o How sound is generated
 Propogation
o How sound energy moves through air and other media (like concrete, steel or water)
o How dimensions and shapes affect the way sound behaves in an environment
 Control
o How sound energy can be prevented from leaving or entering a space through partitions or
vibrations
 Interaction
o What happens to sound energy when it encounters a boundary (materials)
 Your perception of sound, as processed by your ear and brain
Sound can be generated by vibrations (for example, vibrations from a stringed instrument). Many things besides
strings and loudspeakers can vibrate and generate sound. Sounds can be generated by the mechanical motion of a
loudspeaker, the structural elements of a building or a multitude of other ways.

Glossary
Acoustics
Acoustics is the science and technology of sound in all its aspects. Acoustics covers sound production, propagation
and control, its interaction with materials, its reception by the ear, and its effects on the hearer. Ref
Sound Energy
Sound energy moves out and away from the source in all directions. Unless the sound is generated in a completely
free space, the energy will encounter a boundary or surface. If you are outdoors, the only likely boundaries may be
the ground or nearby buildings. There are many boundaries and surfaces when you are indoors. Besides the walls,
ceilings and floors that you may be thinking of, furniture and people also affect what happens with the sound
energy in the environment.
So, what happens to the sound energy produced by either a sound reinforcement system or other source of
generated sound? The Law of Conservation of Energy tells us that the total energy is neither increased nor
decreased in any process.
Glossary
Law of Conservation of Energy
The Law of Conservation of Energy says that the total energy cannot be created or destroyed. Energy can be
transformed from one form to another, and transferred from one body to another, but the total amount of energy
remains constant.
When the sound energy encounters a surface or room boundary, these three things occur:
 Reflected – As the sound energy moves away from the source, some of it will be reflected off various
surfaces back into the room. The reflections either can be in a specular (direct) fashion or diffused
(scattered). Either way, the energy remains in the space.
 Absorbed – This is sound energy that is absorbed either in the air (not much of an issue except in
extremely large spaces) or absorbed by the materials in the space (sound energy converted into heat).
 Transmitted – This sound energy actually passes from one space into another through a partition or other
barrier.
Reflected Sound Energy
Direct, or "specular" reflections bounce directly off a surface like light bouncing off of a mirror. Like light, the
incoming angle—the angle of incidence—will equal the outgoing angle—the angle of reflection.
A specular reflection is mirror-like; most of the energy is reflected back in a single direction. Just as you see with
light, the direction of the angle of the reflected energy is determined by the incoming angle as the angle of
incidence is equal to the angle of reflection.

Glossary
Specular Reflection
A specular reflection is mirror-like; most of the energy is reflected back in a single direction.
A type of hard reflection is a flutter echo. A flutter echo is a series of reflections that continue to bounce back and
forth between parallel hard surfaces, such as large walls, ceilings, windows, and floors. Flutter echoes are
distracting and should be avoided.
This can be accomplished by large flat surfaces and/or using surface treatments.

Glossary
Diffusion
Diffusion is the scattering or random redistribution of a sound wave from a surface. It occurs when surfaces are at
least as long as the sound wavelengths, but not more than four times as long.
When dealing with varying wavelengths, the transition from specular to diffuse is not immediate. Some wavelengths
will react one way or another and some will react in between. The behavior will be determined by the size, material,
and mass of the boundaries of the space.

In every situation involving boundaries or surfaces, you will always have the direct sound – the sound that arrives at
the listener position in a direct, straight line from the source to the listener, and the reflected sound – the sound that
takes any indirect path from the source to the listener.
Reflected energy arrives later in time than direct sound. This should be obvious – the shortest path between two
points is a straight (direct) line. Taking any other pathway requires traveling a longer distance and requires more
time. Thus, reflected sound will always arrive later in time than the direct sound. Reflections can be either “early” or
“late.” A “late” reflection is called an echo.

Glossary
Echo
An echo is a reflected version of sound energy acoustically, or duplicated version of a signal electronically, that
arrives to the listener with sufficient delay and separation from the original signal to allow the delayed signal to be
perceived distinctly and later in time from the original signal.

Reverberation
When a room has many hard reflective surfaces, the combined energy level of the reflections can remain quite
high. As the energy reflects off of more and more surfaces around the room, the listener will begin to receive
reflected energy from all directions. When the energy level remains high and the number of reflections becomes
quite dense in relation to one another, it is called reverberation. Reverberation is simply numerous, persistent
reflections.
Glossary
Reverberation
Reverberation is the combination of many acoustic reflections that are dense enough in time so as to not be
audible as reflections but instead to act as a statistical sonic decay “tail” to sounds in the room.
True reverberation is a phenomenon seen in larger rooms with many hard reflective surfaces. While a typical
conference room may have troublesome reflections, it will not exhibit true reverberation.

Absorption
The materials used in the construction and finish of a venue (walls, ceiling, floor, furniture, curtains, windows,
seating, etc.), as well as people themselves, all play a part in the amount of sound energy absorption available.
One type of absorber you may be familiar with is the porous absorber. As the displaced air molecules pass through
a porous absorber, the friction between the molecules and the material of the absorber slows the molecules down.
While there may still be some reflected sound back into the room, a much greater portion of the sound has
otherwise been absorbed.

Typical porous absorbers include carpets, acoustic tiles, acoustical foams, curtains, upholstered furniture, people,
and their clothing. The effectiveness of different types of absorbers, such as porous or resonant absorbers, is
frequency (wavelength) dependent.

Porous absorbers are primarily effective at middle and high frequencies. Fittingly, this range is where the ear is
most sensitive, and where noise control is most needed in many environments.

Transmission
When sound is absorbed, it doesn't just go away. It may be converted to heat, or it may be transmitted through the
absorbent material to an adjoining space. Some materials transmit more noise than others. You can exert control
over the propagation of noise from one adjacency by selecting the barrier or partition between them.

Glossary
Transmission
Transmission is the passing of sound energy through partitions or structure borne vibrations. A partition's ability to
transmit sound energy will vary with frequency.
Ambient Noise
Ambient noise is any sound other than the desired signal. While an electronic sound system has inherent noise in
the electronic components, rooms also have noise associated with them. Anything heard in a room other than the
desired signal from the sound reinforcement system would be considered noise. Unwanted background noise
found within a room can come from equipment fans, office machines, the HVAC (heating, ventilation, and air
conditioning) system, or noise from the people in the room. Noise can intrude from outside the room as well,
through partitions or windows. Outside sources can include vehicular traffic, adjoining corridors, and structure
borne vibrations.
Since excessive noise levels interfere with the message being communicated, ideally, background noise-level limits
will be specified by an acoustician, audiovisual consultant, or designer appropriate to the type of room and its
designed purpose. In other words, the criteria and limits for background noise levels for a gymnasium will be much
different than those of a conference room. The HVAC system, partitions, and any necessary acoustical treatment
will be designed and applied so that the background noise level criteria is not exceeded.
A room’s acoustical properties (e.g., reflections and types, amount of transmission allowed) and background noise
levels are very significant contributors to a sound system’s overall effectiveness.
Glossary
Ambient Noise
Ambient noise is the sound that is extraneous to the intended, desired, or intentional background noise.

c. Acoustics Section Test

3. Capturing Sound
a. Capturing Sound - Introduction
This section includes the following topics:
 Microphone Types
 Microphone Specifications
 Microphone Signal Transport
Upon completion of this section, you should be able to:
 Identify the characteristics of different types of microphones.
 Identify the characteristics of different types of microphones.
 Understand a general description of how microphones transport signals.

b. Microphone Types
Dynamic Microphone
Understanding a microphone’s construction and its intended usage, as well as its directional, sensitivity, frequency
response and impedance characteristics will help you select the appropriate microphone for each usage.
There are two common types of microphones:
1. Dynamic microphones, which do not require phantom power
2. Condenser microphones, which require phantom power or batteries
If the microphone has a power button, it's a condenser microphone
Verify that the power button is not mistaken for a mute button
In a dynamic microphone, a coil of wire is attached to a diaphragm and placed in a permanent magnetic field.
Sound pressure waves cause the diaphragm to move back and forth, thus moving the coil of wire attached to it.
As the diaphragm and coil assembly moves, it cuts across the magnetic lines of flux of the magnetic field, inducing
a voltage onto the coil of wire.

The voltage induced into the coil is proportional to the sound pressure and produces an electrical audio signal. The
strength of this signal is very small and is called a mic level signal. Typical voltages vary from a few millivolts to
several hundred millivolts, depending on the construction.

Dynamic microphones are economical, durable, and will handle high sound pressure levels. Unlike condenser
microphones, they also do not require a source of power (often called Phantom power).

Condenser Microphone
In the study of electricity, you will find that if you have two oppositely charged (polarized) conductors separated by
an insulator, an electric field exists between the two conductors. The amount of potential charge (voltage) that is
stored between the conductors will change depending upon the distance between the conductors, the surface area
of the conductors, and the dielectric strength of the insulating material between the two conductors. An electronic
component that uses this principle is called a capacitor.
Glossary
Condenser Microphone
A condenser microphone transduces sound into electricity using electrostatic principles.

Condenser microphones have a conductive diaphragm and a conductive backplate. Air is used as the insulator to
separate the diaphragm and backplate. Most of the condenser type microphones used in an AV system are of the
Electret type. Electret mics have a permanent polarizing voltage.

In a condenser microphone, sound waves cause the diaphragm to move back and forth, subsequently changing the
distance between the diaphragm and backplate. As the distance changes, the amount of charge, or capacitance,
stored between the diaphragm and backplate changes. This change in capacitance produces an electrical audio
signal at the output of the capsule.

The strength of the signal from a condenser microphone is not as strong as the microphone level signal we see
from the typical dynamic microphone. To increase the signal, a condenser microphone includes a preamplifier. This
preamplifier amplifies the signal in the condenser microphone to a microphone level signal. Often the built-in
preamplifier receives it's operating voltage from the mixer which supplies what is often called "phantom power.”
This voltage varies between about 15V to 48V.
The diaphragm used in a condenser microphone is small, requiring less mass than other microphone types.
Because of this, the condenser microphone tends to be more sensitive than other microphone types and responds
better to higher frequencies with a wider overall frequency response.

Phantom Power
Phantom power is the remote power required to power a condenser microphone. It typically ranges from 12 to 48
volts DC (VDC).
Glossary
Phantom Power
Phantom power is a means of supplying direct current (DC) to a condenser microphone. The amount of voltage
depends upon the microphone.

Positive voltage is supplied equally to the two signal conductors of a balanced circuit. This is accomplished through
standard microphone cable because standard microphone cable contains two signal conductors plus a shield.
Since the voltage is applied equally on both signal conductors, applying phantom power will not cause damage to
dynamic microphones.
Phantom power is most often available from an audio mixer. It may be switched on or off at each individual
microphone input, or from a single button on the audio mixer that makes phantom power available on all the
microphone inputs at once.
If phantom power is not available from the audio mixer, separate phantom power supplies may be used.
Electret Microphones
An electret microphone is a type of condenser microphone.
The electret microphone gets its name from the prepolarized material, electret, applied to the microphone’s
diaphragm or backplate. This provides a permanent, fixed charge for one side of the capacitor configuration.
This permanent charge eliminates the need for the higher voltage required for powering the typical condenser
microphone. This allows the electret microphone to be powered using small batteries as well as the normal
phantom power.
Electrets are small, lending themselves to a wide variety of uses and quality levels.
Glossary
Electret Microphone
An electret microphone is a type of condenser microphone. It has prepolarized material, called "electret," which is
applied to the microphone’s diaphragm or backplate. This provides a permanent, fixed charge for one side of the
capacitor configuration. This permanent charge eliminates the need for the higher voltage required for powering the
typical condenser microphone, so it can be powered using small batteries and normal phantom power.

Microphone Physical Design and Placement

Whether dynamic, condenser, electret or otherwise, microphones come in an assortment of configurations to meet
a wide variety of uses.
Handheld microphones are used mainly for speech or singing. Since it is constantly moved about, a handheld
microphone includes internal shock mounting to reduce handling noise.
Glossary
Handheld Microphone
A handheld microphone can be held in the hand or mounted on a lectern or stand for hands-free operation.

Boundary microphones are mounted directly against a hard surface, such as a conference table, wall or ceiling.
The acoustically reflective properties of the mounting surface affect the microphone's performance.
Mounting a microphone on the ceiling typically yields the poorest performance because the sound source is much
farther away from the intended source and much closer to other noise sources such as ceiling mounted projectors,
HVAC diffusers and other devices.

Glossary
Boundary Microphone
A boundary microphone relies on reflected sound from a surrounding surface. A boundary microphone sits directly
on a table or surface. They are also called "surface mount microphones."
Gooseneck microphones are used most often on lecterns and conference tables. The microphone is attached to a
flexible or bendable stem, which comes in varying lengths.

Shock mounts are available to isolate the microphone from table or lectern vibrations.

Shotgun microphones are named for their physical shape, as well as their long and narrow polar pattern.

Most often used in film, television, and field production work, a shotgun microphone can be attached to a long
boom pole called a "fishpole," or studio boom to be used by a boom operator, or fitted to the top of a camera.

Glossary
Shotgun Microphone
A shotgun microphone is a long, cylindrical, highly sensitive, unidirectional microphone used to pick up sound from
a great distance.

Lavalier and head microphones are worn by the user.

A lavalier (also called a lav or lapel microphone) is attached directly to clothing, such as a necktie or lapel.
A head microphone (also called a headmic), is a microphone that is attached to a small, thin boom and fitted
around the ear. Since size, appearance, and color matter for these microphones, lavaliers and headmics are most
often an electret microphone. Often worn by a presenter, they are commonly used in television and theater
productions.

Glossary
Lavalier Microphone
Lavalier is a small microphone designed to be worn either around the neck or clipped to apparel.

c. Microphone Specifications

Microphone Polar Patterns

One of the characteristics to look for when selecting a microphone for a particular use is its pickup pattern. The
pickup pattern describes the microphone’s directional capabilities, or in other words, the microphone’s ability to pick
up the desired sound in a certain direction while rejecting unwanted sounds from other directions.

Pickup patterns are defined by the directions from which the microphone is optimally sensitive. These pickup
patterns help you determine which microphone type you should use for a given purpose.

There will be occasions when you want a microphone to pick up sound from all directions (like an interview) and
there will be occasions that you do not want a microphone to pick up sounds from sources surrounding it (like
people talking or someone rustling papers). The pickup pattern is also known as the polar pattern or microphone’s
directionality.

Four common polar patterns (omni, cardioid, supercardioid, bi-directional)

The different types of pickup patterns include:
 Omnidirectional – Sound pickup is uniform in all directions.
 Cardioid (Unidirectional) – Pickup is from the front of the microphone only (one direction) in a cardioid
pattern. It rejects sounds coming from the side but the most rejection is at the rear of the microphone. The
term cardioid refers to the heart shaped polar plot.
 Supercardioid – Provides better directionality than the hypercardioid. It has less rear pickup than the
hypercardioid.
 Hypercardioid – Is a variant of the cardioid. More directional than the regular cardioid because it rejects
more sound from the side. The trade off is that some sound will be picked up directly at the rear of the
microphone.
 Bi-directional – Pickup is equal in opposite directions with little or no pickup from the sides. This is
sometimes also referred to as a “figure-eight” pattern, referring to the shape of its polar plot.
Microphone Sensitivity
Microphones are characterized by their sensitivity specification, which compares their electrical output signal level
against a reference input level. Sensitivity defines how efficiently a microphone transduces or converts acoustic
energy into electrical energy.

If you expose two different types of microphones to an identical sound input level, a more sensitive microphone
provides a higher electrical output than a less sensitive microphone. Condenser microphones are usually more
sensitive than dynamic microphones.

Does this mean that lower sensitivity microphones equates to lesser quality? Not at all. Microphones are designed
and chosen for specific uses. A professional singer for example, using the microphone up close can produce a very
high sound pressure level. In contrast, a presenter speaking behind a lectern and a foot or two away from the
microphone would benefit from a microphone with higher sensitivity.

For the singer, a dynamic microphone may be the best choice, as it will typically handle the higher sound pressure
levels produced by the singer without distortion while still providing more than adequate electrical output. The
presenter produces a much lower sound pressure level because his microphone is farther away than the singer’s
microphone. The presenter would certainly benefit from using a more sensitive microphone.

An example of a sensitivity specification may look like this:

-54.5 dBV/Pa
1 Pa = 94 dB SPL

“Pa” refers to the Pascal and it is a unit of pressure. 1 Pascal is the equivalent of 94 dB SPL. In this example, if you
were to put 94 dB SPL into the microphone, you would realize a -54.5 dBV electrical output signal.

Although most manufacturers use 94 dB SPL as the reference input level, you may also find 74 dB SPL (0.1 Pa)
used as the input reference level. Using a different input reference level would obviously produce a different output
level.
Microphone Frequency Response
The frequency response specification is an important measure of a microphone's performance. This defines the
microphone’s electrical output level over the audible frequency spectrum, which in turn helps to determine how an
individual microphone “sounds.”
A microphone’s frequency response gives the range of frequencies, from lowest to highest, that the microphone
can transduce. The microphone's directional and frequency response characteristics are represented graphically
through a polar plot.
The polar plot shows the directional and frequency response characteristics on a two-dimensional graph of
electrical output vs. frequency.

With directional microphones, the overall frequency response will be best directly into the front of the microphone.
As you move off-axis from the front of the microphone, not only will the sound be reduced but the frequency
response will change as well.
Microphone Impedance
For a microphone to be of any use, you must plug it into something. How do you know if the microphone is
compatible with the device you are plugging it into?
Another microphone specification that must be considered is its output impedance. Impedance is the opposition to
the flow of electrons in an AC circuit. Your audio signals are AC circuits.
Glossary
Impedance
Impedance is the total opposition to current flow in an AC circuit. Like a DC circuit, an AC circuit contains
resistance but it also includes forces that oppose changes in current (inductive reactance) and voltage (capacitive
reactance). Impedance takes into account all three of these factors. Impedance is frequency dependent, measured
in ohms and is symbolized using the letter "Z."
Back in the early days of the telephone and vacuum tubes, it was necessary to match an output impedance with an
input impedance for maximum power transfer. Modern audio systems use a maximum voltage transfer instead. To
accomplish this, a device’s output impedance should be one-tenth or less of the input impedance the device is
being plugged into.
For example, a professional microphone’s output impedance specification should be 200 ohms or less. A
component to be used with a professional microphone will have an input impedance of 2,000 ohms or more.
Microphones can fall into two categories based upon output impedance:
 Low impedance – 200 ohms or less (some as high as 600 ohms)
 High impedance – More than 25,000 ohms
 Professional microphones are low impedance microphones. Low impedance microphones are less
susceptible to noise and allow for much longer cable runs than high impedance microphones.

d. Microphone Signal Transport

Wireless Microphones
Sometimes called “radio microphones,” wireless microphones use RF (radio frequency) transmission in place of a
microphone cable. Some wireless systems use IR (infrared) transmission.

For a handheld microphone, a standard microphone casing is often integrated onto the top of a transmitter and the
microphone casing and transmitter are finished as one unit. At other times, a small “plug-on” style transmitter is
attached to the bottom of a regular handheld microphone.
Glossary
Radio Frequency (RF)
Radio frequency (RF) is the portion of the electromagnetic spectrum that is suitable for radio communications.
Generally, this is considered to be from 10 kHz up to 300 MHz. This range extends to 300 GHz if the microwave
portion of the spectrum is included.

A wireless receiver (top), and handheld microphone.

For hands free applications, a lavalier or headmic microphone is plugged into a body pack style transmitter. The
body pack transmitter is then clipped onto a belt or placed in a pocket or pouch. Either way, at the other end of the
RF or IR transmission is a receiver tuned to the transmitter’s specific frequency.

Modern RF wireless microphones allow you to change frequencies in order to avoid interference from outside
sources as well as interference from other wireless microphones that may be in use. This is called frequency
coordination, and it will be specific to your geographical area. Your wireless microphone manufacturer can provide
help in coordinating compatible frequencies for your area.

Microphone Cables and Connectors

Microphones and wireless microphone receivers are connected to audio mixers with a cable and a connector.
Professional microphone cables utilize shielded twisted pair cable.

Shielded twisted pair contains:

Two small gauge insulated copper wires (conductors) twisted together

An aluminum foil or a copper braided shield that covers the twisted conductors
The twisted pair and shield are covered with a protective jacket (rubber or plastic)
Typically, the ends of the shielded twisted pair cable are finished (terminated) with an XLR connector.

“XLR” is now a generic term describing the common audio connector. Although they are available in 3- to 7-pin
configurations, the 3-pin XLR is used for almost all microphone cable applications.

XLR male (left), XLR female (right)

When multiple twisted pair cables are needed, snake cables are used. Snake cables combine anywhere from 2 to
58 shielded twisted pairs under a single protective jacket.

An audio snake cable with 8 connections.

e. Capturing Sound Section Test

4. Audio Signal Levels

a. Audio Signal Levels - Introduction
This section includes the following topics:
Audio Signal Levels
Upon completion of this section, you should be able to:
Identify the process for signal transfer from microphone to loudspeaker in order to ensure compatibility and
make adjustments for sound quality.

b. Audio Signal Levels

Audio Signal Levels
A microphone, regardless of the type, produces a signal level called mic level. Mic level is a very low level signal,
with only a few millivolts (abbreviated as mV to express one thousandth of a volt).
Glossary
Mic Level
Mic level is a very low level signal. It is only a few millivolts of electrical energy.

Since mic level operates at a few millivolts, it is more prone to interference. The microphone preamplifier amplifies
the mic level signal to line level for routing and processing.

Glossary
Preamplifier
A microphone preamplifier, also known as a mic pre, amplifies a microphone level signal to line level for routing and
processing.
Line level is where all signal routing and processing is performed. Line level in a professional audio system is about
1.23 volts.
Professional line level is 1.23 V (+4 dBu), while consumer line level 0.316 V (-10 dBV). Use of an RCA or phono
connector is often an indicator of a consumer level signal.

Glossary
Line Level
Line level is the strength of an audio signal. Line level is used for all routing and processing between components.

Once you have routed and processed the signal, it is sent to the power amplifier for final signal amplification up to
loudspeaker level. The loudspeaker takes that amplified electrical signal and transduces the electrical energy into
acoustical energy.
Audio signal types and levels:
 Mic level – A few millivolts (mV)
 Professional line level – About 1 volt
 Loudspeaker level – A few volts up to about 100 volts

Signal Level Compatibility

As you begin to connect your system, you need to make sure the components of your system are compatible with
one another.
For example, while you could plug a microphone directly into the input of a power amplifier, you would not get much
sound level. The mic level signal isn’t strong enough by itself.
You also would not want to connect the output of a power amplifier into a device expecting either a mic or line level
– you would almost certainly damage components.
What about plugging your microphone directly into the back of a powered loudspeaker? Some companies
manufacture “powered loudspeakers” that are all-in-one devices meant to simplify setup and provide for easy
portability. In the case of a powered loudspeaker, all of the signal requirements listed above are built in to the
loudspeaker. If the powered loudspeaker has microphone inputs, it has a microphone preamplifier and any internal
processing will be done at line level. It will also have the power amplifier built in to power the loudspeaker.
Read the operating manual for the device to verify correct signal levels.
Line Mic Input on a Powered Loudspeaker

Signal Level Adjustments

When working with signal levels, you may need to make slight changes in order to provide a more adequate signal
or to avoid the signal distortion that results from too much signal.
Adjustments you make to signal levels are called gain adjustments. A gain control is the term used to describe the
general ability to adjust the signal levels.

If the signal level is increased, it is called gain and it refers to the amount of amplification applied to the signal. If the
signal level is decreased, it is called attenuation.

If neither gain nor attenuation are applied, the signal has unity gain. Unity gain means that the signal is passing
through the gain control without any changes to the signal level. The signal you put into the system will be the
same signal you will get out of the system.

Glossary
Gain
Gain refers to the electronic amplification of a signal.

Glossary
Attenuate
To attenuate is to reduce the amplitude (strength) of a signal or current.

Glossary
Unity Gain
Unity gain is derived from the number 1. Unity gain refers to no change in gain.

c. Audio Signal Levels Section Test

5. Audio Components
a. Audio Components – Introduction
This section includes the following topics:
 Audio Components

Upon completion of this section, you should be able to:

 Describe the function of Audio Mixers, Equalizers, Delays and Power Amplifiers and describe how each audio
component supports the audio system

b. Audio Components
Audio Mixers
In its most basic form, an audio system has a sound source at one end and a destination for that sound on the
other. In almost all situations, there is more than one source. Audio technicians deal with multiple and varied
sources of sound. The sources could be several vocalists with instruments at a concert, playback devices such as
CD, DVD or MP3, multiple panelists at a conference or several actors in a theatre performance. All of these signals
come together in the audio mixer.
All audio mixers serve the same purpose – to combine, control, and route audio signals from a number of inputs to
a number of outputs. Usually, the number of inputs will be larger than the number of outputs.
Audio mixers are often identified by the number of available inputs and outputs. For example, an 8x2 mixer would
have eight inputs and two outputs.
Each incoming mic or line level signal goes into its own channel. Many mixers provide individual channel
equalization adjustments as well as multiple signal routing capabilities called main or auxiliary busses.
A larger audio mixer is often called a mixing console, console or a mixing desk. Regardless of the size and
complexity, any mixer that accepts mic level inputs will have microphone preamplifiers (preamps). Once the mic
level is amplified to line level by the preamp, it can be sent through to the rest of the mixer.
Between the inputs and outputs, the typical audio mixer provides multiple gain stages for making adjustments.
These adjustments allow the mixing console operator to balance or blend the audio sources together for the most
realistic sound appropriate for the listening audience.
Some audio mixers will turn microphone channels either on and off automatically like an on/off switch. These are
called gated automatic mixers. Others will turn up microphone channels being used and turn down microphone
channels that are not being used, like a volume knob being turned up or down. These are called gain sharing
automatic mixers.
Glossary
Gain Sharing Automatic Mixers
A gain sharing automatic mixer is an audio mixer that automatically turns up microphone channels that are in use,
and turns down microphone channels that are not being used.

Glossary
Gated Automatic Mixer
A gated automatic mixer is an audio mixer that turns microphone channels either up or all the way down
automatically, like an on/off switch.
The channels set in an automatic mixer for automatic mixing are to be used for speech-only situations. Other sound
sources, such as music, would not be set to be mixed automatically. Music and other uses still require live
personnel for operation.
Compressors
A compressor controls the dynamic range of a signal by reducing the part of the signal that exceeds the user-
adjustable threshold. When the signal exceeds the threshold, the overall amplitude is reduced by a user-defined
ratio, thus reducing the overall dynamic range.
Compressors are a type of digital signal processors that compensate for loud peaks in a signal level. All signal
levels below a specified threshold will pass through the compressor unchanged, and all signals above the threshold
will be compressed. In other words, compressors keep loud signals from being too loud. This reduces the variation
between the highest and lowest signal levels, resulting in a compressed (smaller) dynamic range. Compressors are
useful when reinforcing energetic presenters, who may occasionally raise their voice for emphasis.
Glossary
Compressor Threshold
The compressor threshold sets the point at which the automatic volume reduction kicks in. When the input goes
above the threshold, an audio compressor automatically reduces the volume to keep the signal from getting too
loud.

Glossary
Compressor Ratio
The compression ratio of an audio compressor determines how much the volume reduces depending on how far
above the threshold the signal is.
The compressor ratio is the amount of actual level increase above the threshold that will yield one decibel in gain
change after the compressor. For example, a 3:1 ratio would mean that for every 3 dB the gain increases above the
threshold, the audience would only hear a 1 dB difference after the compressor. Likewise, if the level were to jump
by 9 dB, the final level would jump only 3 dB.
You can also set the attack and release times on a compressor. The attack time is how long it takes for the
compressor to react after the compressor exceeds the threshold. The release time determines when the
compressor lets go after the level settles below the threshold. Both of these functions are measured in
milliseconds.

Limiters
Limiters are similar to compressors in that they are triggered by peaks or spikes in the signal level. They are used
to limit the impact of extreme sound pressure spikes, such as dropped microphones, phantom powered
microphones being unplugged without being muted, and equipment not being turned on or off in the correct order.
They protect downstream gear by preventing severe clipping and overdriving amps and speakers. For this reason,
many amplifiers have built-in limiters to protect themselves.
Glossary
Limiter
A limiter is an audio signal processor that functions like a compressor except that signals exceeding the threshold
level are reduced at ratios of 10:1 or greater.

Expanders
Expander is an audio processor that comes in two types: a downward expander and a part of a compander.
Downward expanders increase the dynamic range by reducing, or attenuating, the level below the adjustable
threshold setting. They increase gain if the signal is very low, such as a presenter with a weak voice. This reduces
unwanted background noise, and it's especially useful in a system using multiple open microphones.
Expanders are primarily intended for recording and transmission. When used in an amplified environment,
expanders can push a system into painfully loud feedback.

Introduction to Filters
Filter
A filter removes or passes certain frequencies from a signal.

Low Pass Filter

A low pass filter is a circuit that allows signals below a specified frequency to pass unaltered while simultaneously
attenuating frequencies above the specified limit.
Low pass filters are useful for eliminating hiss in a system. If you have a source with a lot of hiss, like a low-cost
consumer level tape deck or cheap MP3 player, you can apply a low pass filter to eliminate the high frequency hiss.
High Pass Filter
A high pass filter is a circuit that allows signals above a specified frequency to pass unaltered while simultaneously
attenuating frequencies below the specified limit.

High pass filters are useful for removing low frequency noise from a system, such as rumble from an HVAC system
or the proximity effect on a microphone.
Gates
Gates mute the level of all signals below an adjustable threshold. This means that the signal levels must exceed
the threshold setting before they are allowed to pass. This can be used to turn off unused microphones
automatically. You can control when the gate activates by setting the gate's attack and hold times. Gates are found
in some automixers and are useful for noise control, such as from a noisy multimedia source.
Parametric Equalizer
Equalizers, or EQs, are frequency controls which allow you to boost (add gain) or cut (attenuate) a specific range of
frequencies. The simplest equalizer comes in the form of the bass and treble tone controls found normally on your
home stereo or surround receiver. The equalizer found on the input channel of a basic audio mixer may provide
simple high, mid and low frequency controls.

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Glossary
Equalizer
An equalizer is a piece of electronic equipment that adjusts or corrects the frequency characteristics of a signal.

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Glossary
Parametric Equalizer
The parametric equalizer allows discrete selection of a center frequency and adjustment of the width of the
frequency range that will be affected. This can allow for precise manipulation with minimal impact of adjacent
frequencies.

Parametric Equalizers allow boost and cut adjustments, and they also allow the user to select the center frequency
of the filter as well as adjust the bandwidth of the filter. A parametric equalizer allows the user to create their own
filters using individual controls of frequency centering, amplitude, and width of the frequency range (bandwidth) to
be controlled.

In short, a parametric equalizer allows you to make many, large-scale adjustments to a signal with fewer filters.
A parametric equalizer offers greater flexibility than a graphic equalizer. Not only will the parametric provide boost
or cut capability like the graphic, but it also allows center frequency and bandwidth adjustments.
Graphic Equalizers
Instead of using parametric equalizers, some audio professionals prefer to use graphic equalizers.
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Glossary
Graphic Equalizer
A graphic equalizer is an equalizer with an interface that resembles a graph comparing amplitude on the vertical
with frequency on the horizontal. Graphic equalizers normally come in 2/3 octave, or, more often, 1/3 octave filters
sets. Filters are usually set on ISO (International Organization for Standardization) defined center frequencies.
Center frequencies and bandwidth are fixed for these filters, so named as the adjustments to the sliders offer a
"graphic" representation of the frequency response. Active graphic equalizers can provide boost and cut capability.

This graphic equalizer has 31 controllable filters. This type of display gives you fine control of dozens of specific
frequencies, where you can grab any control and add precise amounts of boost or cut to that particular frequency.
In this display, the line that runs near each filter control represents the combined interaction between the filters. Do
you see the bumpy, rippled area around 315 Hz? That is an area with phase interference, due to the way the filters
around that frequency are arranged. This is a downside of using a graphic equalizer. Parametric equalizers will
avoid these phase ripples, since they use fewer, smoother filters. In addition, since they use fewer filters,
parametric equalizers in DSPs tend to use less processing power than a graphic equalizer.

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Glossary
1/3 Octave Equalizer
A 1/3 octave equalizer is a graphic equalizer that provides 30 or 31 slider adjustments corresponding to specific
fixed frequencies with fixed bandwidths, with the frequencies centered at every one-third of an octave. The
numerous adjustment points shape the overall frequency response of the system. This makes the sound system
sound more natural.
Delays
Summarize the reason for the purposeful delay of audio signals to loudspeakers.
Delays are used in sound systems for loudspeaker alignment, either to align components within a loudspeaker
enclosure or main loudspeaker array, or to align supplemental loudspeakers with the main loudspeakers.

Within a given loudspeaker enclosure, the individual components may be physically offset, causing differences in
the arrival time from those components. This issue can be corrected either physically or by using delay to provide
proper alignment.

Electronic delay is often used in sound reinforcement applications. For example, consider an auditorium with an
under-balcony area. The audience seated underneath the balcony may not be covered well by the main
loudspeakers. In this case, supplemental loudspeakers are installed to cover the portion of the audience seated
underneath the balcony.

While the electronic audio signal arrives at both the main and under-balcony loudspeakers simultaneously, the
sound coming from these two separate loudspeaker locations would arrive at the audience underneath the balcony
at different times and sound like an echo. This is because sound travels at about 1130 feet per second (344 meters
per second), much slower than the speed of the electronic audio signal.

In this example, an electronic delay would be used on the audio signal going to the under-balcony loudspeakers.
The amount of delay would be set so that both the sound from the main loudspeakers and the under-balcony
loudspeakers arrive at the audience at the same time.

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Glossary
Delay
Delay is the retardation of a signal. In the context of audio processing, it is an adjustment of the time in which a
signal is sent to a destination, often used to compensate for the distance between loudspeakers or the differential
in processing required between multiple signals. If the delay is an unintended byproduct of signal processing, it is
usually referred to as latency.
Power Amplifiers
Identify the primary function of audio system power amplifiers.
Identify the location of a power amplifier in the audio system chain.
The amplifier is the last device used before the signal reaches the loudspeakers. Power amplifiers boost, or
amplify, electronic audio signals sufficiently to move the loudspeakers. They do this by increasing the gain (the
voltage and power) of the signal from line level to loudspeaker level.

Most amplifiers have only a power switch and input sensitivity controls. Some now include digital signal processing
and network monitoring and control.

Today, many speakers are available with built in power amplifiers. These are called "Powered Loudspeakers."

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Glossary
Amplifier
An amplifier is an electronic device for increasing the strength of electrical signals.

Potentially, the more powerful the amplifier is, the greater the amplification of the signal it can provide, and the
louder the sound that the loudspeaker can achieve.
Power amplifiers are connected to loudspeakers with larger gauge wire that we use at mic or line level. The size of
wire will depend on the distance between the power amplifier, the loudspeaker, and the current required.
Loudspeaker cabling will be unshielded and may or may not be twisted.
A common connector used for loudspeaker cabling is the Speakon® connector. Speakon® connectors are often
used for professional loudspeaker connections. They are commonly used for audiovisual staging events because
they are rugged, durable, simple to use and lock into place.

c. Audio Components Section Test

6. Loudspeakers
a. Loudspeakers – Introduction
This section includes the following topics:
Loudspeakers
Upon completion of this section, you should be able to:
Describe features, configurations, and placements of loudspeakers in order to ensure optimal sound quality.

b. Loudspeakers
Loudspeakers Introduction
For the purpose of sound reinforcement, loudspeakers are the end of the electrical signal path. The acoustic
energy that was transduced into electrical energy by the microphone is transduced back into acoustical energy by
the loudspeaker.

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Glossary
Loudspeaker
Loudspeaker is a transducer that converts electrical energy into acoustical energy.
Loudspeaker components are primarily a driver within an enclosure.

Quick Tip: A Common Element of Loudspeakers

Most loudspeakers share a common element – they have drivers mounted in an enclosure.
The suitability of a loudspeaker depends on its intended use. Uses include:
 Communication – simple public address, intercom or radio communications.
 Sound reinforcement – lectures, musical performances, theatres, etc.
 Sound reproduction – playback of prerecorded music, motion picture soundtracks, etc.

Crossovers
The audio spectrum has wavelengths and frequencies that vary dramatically. No single driver can reproduce the
entire frequency range accurately or efficiently. This is why a loudspeaker contains multiple drivers in professional
audio. A loudspeaker enclosure containing more than one frequency range of drivers is known by the different
frequency ranges being covered.
So that each driver is sent only those frequencies that it will transduce efficiently, an electrical frequency-dividing
network circuit called a crossover is used. A passive crossover (one that doesn’t require powering) would be used
to take the electrical signal coming into the loudspeaker enclosure and split it into the different frequency ranges.
Glossary
Crossover
A crossover is used to separate the audio signal into different frequency groupings and route the appropriate
material to the loudspeaker or amplifier in order to ensure that the individual loudspeaker components receive
program signals that are within their optimal frequency range. There are two types of crossovers: passive and
active.
Examples of the different drivers and frequency ranges:
 Tweeters – high frequencies
 Horns – mid to high frequencies
 Cone or midrange – midrange frequencies
 Woofers – low frequencies
 Subwoofers – lower frequencies

Loudspeaker Sensitivity
Like microphones, loudspeakers are rated based on their ability to convert one energy form into another. This
rating is called a sensitivity specification. This defines the loudspeaker’s acoustic output signal level, given a
reference input level. Put another way, sensitivity defines how efficiently a loudspeaker transduces, or converts,
electrical energy into acoustic energy.

Given the same reference electrical input level into two different loudspeakers, a more sensitive loudspeaker would
provide a higher acoustical energy output than a less sensitive loudspeaker.

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Glossary
Sensitivity Specification
Sensitivity specifications are a way to determine a device's ability to convert one form of energy into another form
of energy. It is used to define the device's efficiency in converting from one form to another.

Loudspeakers vary quite a bit when it comes to efficiency. Does this mean that lower sensitivity loudspeakers are
always of lesser quality? Not at all. Like microphones, loudspeakers are designed and chosen to meet specific
uses.
Loudspeakers are used for emergency notification, paging, speech reinforcement, and music reinforcement. They
are found in recording studios, sports arenas, concerts, houses of worship, and your home listening environment.
Very little else in AV can be found in so many different configurations and prices for so many different uses.
Here is an example of a loudspeaker sensitivity specification may look like: 88 dB SPL1 w / 1 m
This means that with a one watt input, 88 dB SPL will be measured one meter away from the loudspeaker.

Loudspeaker Frequency Response and Polar Patterns

As with directional microphones, a loudspeaker's overall frequency response will be best on-axis, directly in front of
the loudspeaker. As you move off-axis with a loudspeaker, not only will the sound be reduced, but the frequency
response will change as well. A loudspeaker with specifications showing a nominal 90 degree by 40-degree
dispersion (coverage) pattern only holds that pattern over a limited frequency range. Lower frequencies will spread
out in a more omnidirectional fashion due to their much longer wavelengths. Very large devices are required to
control the dispersion pattern at the lower frequencies.

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Glossary
Polar Plot
A polar plot is a graphical representation of the relationship between a device’s directionality and its output.
A loudspeaker polar plot. The on-axis location is at zero degrees.

Loudspeaker Impedance
Loudspeakers have a nominal impedance rating. Most are rated at four, eight, or sixteen ohms. Some are rated at
six ohms. Because impedance is frequency dependent, the impedance will not be the same over the loudspeaker’s
entire frequency range.

Glossary
Impedance
Impedance is the total opposition to current flow in an AC circuit. Like a DC circuit, an AC circuit contains
resistance but it also includes forces that oppose changes in current (inductive reactance) and voltage (capacitive
reactance). Impedance takes into account all three of these factors. Impedance is frequency dependent, measured
in ohms and is symbolized using the letter "Z."

Knowing a loudspeaker's impedance will help you find the total impedance load when you connect multiple
loudspeakers to the output of a power amplifier. This information will help you to avoid wasting power, overloading
your power amplifier, and damaging your loudspeakers. It will also give you optimum volume by reducing distortion
and noise, and avoiding uneven sound distribution.

Center Cluster and Distributed Systems

All loudspeakers have a particular coverage pattern. Consider the listening area when selecting a loudspeaker
suitable for the intended use. The audience must be able to hear clearly. The sound pressure level must also be
reasonably even throughout the listening area so that it is not too loud in some areas and too soft in others.
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Glossary
Coverage Pattern
A coverage pattern is the predictable pattern of sound energy that every loudspeaker emits. The coverage pattern
is based on the frequency of the sounds and the physical size of the loudspeaker.
Although there are many arrangements for loudspeaker placement, depending on the application and acoustic
environment, there are two general approaches to covering the enclosed listener area.
The first is the central cluster, or single source configuration, which can be a single loudspeaker or a central cluster
of loudspeakers. The central cluster is normally located directly above, and slightly in front of, the stage or
presentation area. In a central cluster, the sound is coming from one point in the room. Since you tend to look in the
direction sound is coming from, central cluster systems naturally focus your attention on something, like a pulpit in
a house of worship.

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Glossary
Central Cluster
Central cluster is a single source configuration of loudspeakers. In a central cluster, the sound is coming from one
point in the room. The central cluster is normally located directly above (on the proscenium), and slightly in front of,
the primary microphone location.

Example of a center cluster loudspeaker arrangement.

If the ceiling height is not adequate for a central cluster, or if there is no need to have a relationship between the
location of the sound source and the origin of the sound, a distributed system is often used.

Distributed systems use multiple loudspeakers that are strategically suspended overhead or located in the ceiling.
A distributed system is often used for voice reinforcement or paging applications.
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Glossary
Distributed System
Distributed loudspeaker systems are suspended overhead, often in the ceiling, at regular intervals. Due to their
even placement, they can provide even coverage throughout the listening area.
An example of a distributed system.
For permanent installations, whether central cluster or distributed, audiovisual designers will use computer software
prediction programs to make sure the audience areas will be properly covered by the loudspeakers.

c. Loudspeakers Section Test

7. Audio Signal Level Monitoring

a. Audio Signal Level Monitoring - Introduction
This section includes the following topics:
Audio Signal Level Monitoring
Upon completion of this section, you should be able to:
Monitor the flow of audio signals in order to produce quality sound and prevent unwanted noise.

b. Check Signal Levels

Once you have set up the audio signal path and know which signal levels are involved (eg. mic, line, loudspeaker),
you need to be able to monitor and adjust the signal levels at various points to make sure they are not too low or
too high.
Before turning on the power amplifier, check to make sure you are getting signals on all the audio mixer channels in
use. Make the necessary level adjustments to the microphone preamplifier and all other gain stages in the mixer
and the other audio equipment leading to the power amplifier. Checking signal levels in the audio mixer is often
done with the aid of the built-in meters and verified with headphones.
After setting the gain stages, turn the power amplifier input adjustments all the way down and turn on the power
amplifier. Slowly bring the power amplifier input adjustments up until the desired sound pressure level is reached at
the loudspeakers.
Verify through listening to the system and through signal monitoring that no signals are distorting.
These practices will help prevent signal levels that are too low, which results in poor signal-to-noise and hissing, or
signal levels that are too high, which results in distortion.

c. Audio Signal Level Monitoring

Signal level monitoring is used to ensure you don't clip the signal or add distortion. It is also important to make sure
the signal level is high enough to achieve and adequate signal-to-noise ratio without adding noise.
Signal levels that are too low decrease the system’s signal-to-noise ratio and can result in background hiss.
Two types of meters are typically used for signal monitoring. The first is the "standard" volume unit indicator is the
Volume Unit (VU) meter. The second is the Peak Program Meter (PPM).
A VU meter indicates program material levels for complex waveforms. In essence, it is a voltmeter calibrated in
decibels and referenced to a specific voltage connected across a 600 ohm load. A VU meter meets stringent
standards regarding scale, dynamic characteristics, frequency response, impedance, sensitivity, harmonic
distortion, and overload. It is commonly used to monitor broadcast signals.
When using a VU meter, set the average levels to read around -6, and the peaks around “0.”
A PPM responds much faster than a VU, showing peak levels instantaneously. It is useful for digital recording, or
when the level must not exceed 0 dB full scale (dBFS). Going above the full scale of the digital signal will cause
clipping in the signal. A PPM is often found in audio mixers.

VU stands for "volume unit," and a VU meter responds in a way that humans respond to loudness.

A PPM (Peak Program Meter) shows instantaneous peak levels and is very useful for digital recording.

d. Signal Levels
With analog, running signal levels at around 0 dBu for line level signals is often preferred and there may be some
occasions when the normal signal level exceeds 0 dBu.
With digital however, the level must never exceed 0 dBFS. dBFS is the full scale (FS) of the digital signal.
Exceeding 0 dBFS with a digital signal causes immediate distortion of the signal.
Frequently, equipment will provide at least a single LED light as some sort of indicator of the signal level condition.
Labels may include:
 Signal Present
 Overload
 Clip
Some LEDs may even change color as the signal level approaches or goes into distortion.
Notes from the Field: Signal Level Indicators
Read the equipment’s manual. It will help you make sure the equipment is operated properly and will help you
understand what a specific indicator tells you about the condition of the signal. Quite often, you will find that a red
flashing LED is indicating signal distortion, but you need to read the manual to be sure.

e. Balanced and Unbalanced Circuits

Signal processing circuitry and cables are continuously exposed to noise and interference. In fact, all electrical
circuits and the cabling used to connect them generate energy fields that can interact with other electrical circuits
and cabling, including the audiovisual circuitry and cabling. This interference and noise degrades the quality of the
audio signal and has the potential of introducing hum and buzz into our audio system. Noise introduced into a
digital signal can corrupt the data.

LO glossary element icon

Glossary
Noise
Noise is any electrical signal present in a circuit other than the desired signal.

One way to reduce the noise in a circuit or cable is to use a balanced electrical design. Electronic circuitry can be
balanced or unbalanced.

The terms “balanced” and “unbalanced” refer specifically to the impedance balance found on the two conductors as
well as the circuitry connecting those two conductors.

In a balanced design, the impedance of high side of the signal circuit is equal to lowside of the signal circuit ground.
The design of balanced circuits offers a defense mechanism against noise. This defense mechanism removes the
noise, or most of it, leaving only the intended signal. As a rule, you should use balanced components whenever you
can. The cabling used with balanced circuitry requires two signal conductors. In audio, the two signal conductors
are surrounded by a shield.

LO glossary element icon

Glossary
Balanced Circuit
A balanced circuit is a two-conductor circuit in which both conductors and all circuits connected to them have the
same impedance with respect to ground.

LO glossary element icon

Glossary
Shield
A shield is a metallic partition placed between two regions of space. It is used to control the propagation of electric
and magnetic fields from one of the regions to the other. Shields are used to contain electric and magnetic fields at
the source or to protect the receiver from electric and magnetic fields. It can be the chassis, or metallic box, that
houses an electronic device, or the metallic enclosure (eg., aluminum foil or copper braid), that surrounds a wire or
cable.
 A circuit diagram of a balanced circuit.
In an unbalanced circuit design, the equipment outputs an unbalanced signal to the cable that is connected to an
unbalanced input. As with a balanced circuit, the cable picks up noise from surrounding sources. However, with an
unbalanced circuit design, there are no noise defense mechanisms, except for the shield. So if noise gets onto the
signal conductor there is no mechanism to remove it. Additionally, if undesired noise currents are flowing on the
shield in an unbalanced circuit it actually becomes part of the problem and not one of the solutions.

Cabling used with unbalanced circuitry uses two signal conductors. One conductor carries the signal while the
second conductor, usually the cable shield, acts as the return electrical path for the circuit.

LO glossary element icon

Glossary
Unbalanced Circuit
An unbalanced circuit is a two-conductor circuit in which one conductor carries the signal and the other conductor
carries the return. The return conductor is usually the cable shield and is a low impedance connection as it is
connected to the signal ground and possibly also earth ground. The impedance of the signal circuitry is quite
different than the return circuitry, hence the impedance of the two conductors are quite different; the impedances
are unbalanced with respect to one another.

A circuit diagram of an unbalanced circuit.

With either a balanced or unbalanced circuit, the longer the cable run, the more the cabling is subjected to noise.
Balanced circuits are less prone to suffer from noise pickup than unbalanced circuits. Unbalanced lines are
extremely limited in the distance they can cover.

f. Balanced and Unbalanced Distinctions

A quick way to determine whether a piece of equipment is balanced or unbalanced is to look at its outputs or
inputs. Unbalanced audio equipment and cables typically use an RCA (phono) connector. All RCA connectors are
unbalanced. However, this is not the only distinction between the two types of circuits.

Unbalanced circuits are also called single-ended circuits. They require two conductors in the cable and connector.
The first conductor, an insulated wire, transports the signal. The second conductor is a shield around the wire used
as the electrical circuit’s return path and provides the "ground" reference for the circuit.

As with most things that offer higher quality, balanced components are more expensive due to design and
manufacturing. Like an unbalanced circuit, they require two conductors in the cable and connector. However, in this
case, both conductors are used to transport the signal. The first signal conductor carries a signal and the second
conductor carries an inverse or “mirror image” of the first signal conductor.

A balanced circuit does not require one signal on a wire with the exact opposite signal on the other wire. Instead, it
needs the impedances to be balanced.

In this case, the impedances on the two signal conductors, as well as the input and output circuitry connected to
them, are the same with respect to one another. Since the impedances are the same for the two signal conductors,
they are said to be equal or "balanced."

Balanced Shielded Twisted Pair

Although a balanced circuit only requires two conductors, with audio, you can use a foil or braided shield wrapped
around the two signal conductors. This serves as a circuit's “ground” reference. An audio balanced circuit has a
connector that requires three pins: two for the signal conductors and a third for the shield connection.

If both conductors are equally exposed to noise (the reason for the twisting in the twisted pair cable), and if the
impedances are balanced, noise is induced equally onto both conductors. So if noise 1 and noise 2 are then equal,
noise 1 - noise 2 would equal 0. The noise is cancelled.

A balanced circuit provides greater signal strength for longer distances and has less noise.

A Balanced Circuit

g. Feedback
Feedback is the "squealing" or "howling" generated between microphones and loudspeakers. Feedback occurs if a
microphone is too close to the front of a loudspeaker, or if the gain or volume has been turned up too high
somewhere in the sound system.

A sound system is an amplification system. If the microphone “hears” itself through the sound system, it goes
through the signal path where it gets amplified and then it comes out of the loudspeaker at an even louder level
than before. That’s why feedback typically gets so loud so quickly – it is a regenerative amplification loop. The
signal continues to receive additional amplification each time it goes through the system.

One way to avoid feedback is through proper microphone and loudspeaker placement. The best strategy is to place
the microphones as close to the sound source and as far from, and behind the loudspeakers as practical.
LO glossary element icon
Glossary
Feedback
In the context of audio, feedback is unwanted noise caused by the loop of an audio system's output back to its
input.

In the context of control systems, feedback is data supplied to give an indication of status, i.e., on or off. A CPU
sends out an instruction to a device, which executes the instruction and then replies back to the CPU.

infocomminterim/thumbs_up.gif
Best Practice: Avoiding Feedback
These are some best practices for avoiding feedback:
 Keep the microphone as close to the sound source as possible.
 Keep the loudspeakers in front of, and as far from, the microphones as is practical.
 Select directional microphones with polar patterns that fit the usage requirements.
 Select loudspeakers with coverage patterns that cover only the audience area.

h. Summary
In summary, best practices for controlling feedback include:
Keeping the microphone as close to the sound source as possible.
Keeping the loudspeakers in front of, and as far from, the microphones as is practical.
Select directional microphones with polar patterns that fit the usage requirements.
Select loudspeakers with coverage patterns that cover only the audience area.

i. Audio Signal Monitoring Section Test

8. Audio System Applications

a. Audio System Applications - Introduction
This section includes the following topics:
Audio System Applications
Upon completion of this section, you should be able to:
Identify types of sound reinforcement systems.

b. Sound Reinforcement
You have reviewed the electrical audio signal chain from start to finish – microphones to loudspeakers. You have
also learned about the various signal levels, the cable used, and even the types of circuits (balanced) that are
preferred for professional audio. This lesson will identify different sound reinforcement systems and their uses.
If you can’t hear something adequately without amplification, you can use microphones, audio mixers, signal
processors, power amplifiers, and loudspeakers to amplify that sound source electronically.
The term "sound reinforcement" can be broken down into the subcategories of music and speech reinforcement.
Glossary
Sound Reinforcement System
Sound reinforcement is the combination of microphones, audio mixers, signal processors, power amplifiers, and
loudspeakers that are used to electronically amplify and distribute sound.
Since musical instruments cover a good bit of the audible spectrum, music reinforcement systems tend to be full
bandwidth systems, capable of reproducing a wide frequency range with higher sound pressure levels.
Glossary
Music Reinforcement System
A music reinforcement system is a sound reinforcement system that is used to amplify and distribute sound from a
live musical performance.

Quick Tip: Sound Reinforcement Systems and Bandwidth

If you can’t adequately hear an unamplified presenter, a speech reinforcement system will help to amplify and
distribute the sound. Since the human voice has limited bandwidth, the speech reinforcement system need not be
designed for full bandwidth.

c. Mix-Minus Systems
A “mix-minus” system is a type of speech reinforcement system.
When both meeting presenters and participants need to be heard, microphones must be distributed and mixed with
each group of loudspeakers. A live microphone placed near a loudspeaker amplifies its signal and causes
feedback, so creating a stable system presents several engineering challenges.

The best approach is to create a separate sound subsystem for each loudspeaker, or group of loudspeakers (called
“loudspeaker zones”). The term "mix-minus" means that each subsystem mixes the microphone signals, minus the
microphones closest to a group of loudspeakers. These microphones would cause feedback, and, since they are
so close to that group of loudspeakers, they would not require reinforcement anyway.

You would find such systems in large boardrooms, meeting rooms, or classrooms.

Glossary
Mix-Minus System
A mix-minus system is a type of speech reinforcement system that allows both meeting presenters and participants
to be heard. Each loudspeaker is given a separate subsystem, which mixes the microphone signals, minus the
closest microphone.

A mix-minus system.

In the diagram above, "S" indicates a loudspeaker, and "M" indicates a microphone.

d. Playback and Combination Systems

A playback system is similar to a music reinforcement system in that it has a wider bandwidth, and is capable of
higher sound pressure levels. The difference between a music reinforcement system and a playback system is that
a playback system does not include microphones. It simply plays prerecorded material.
Glossary
Playback System
A playback system is a music reinforcement system that has a wide frequency response bandwidth, and is capable
of high sound pressure levels. A playback system does not include microphones; it simply plays prerecorded
material.
A single system can be used for music reinforcement, speech reinforcement, and playback. It is much easier to
include and design speech reinforcement as part of a music reinforcement system than the other way around.
Music Reinforcement System
A music reinforcement system is a sound reinforcement system that is used to amplify and distribute sound from a
live musical performance.

Sound Reinforcement System

Sound reinforcement is the combination of microphones, audio mixers, signal processors, power amplifiers, and
loudspeakers that are used to electronically amplify and distribute sound.

e. Intercom and Paging Systems

Intercoms are sound systems used for the purpose of inter-communication between people or rooms, usually, but
not always, located within the same building or complex of buildings. These are typically comprised of a central unit
with local, personal stations. They may also include multiple satellite or remote stations. The purpose is to provide
two-way communication in the same way that a telephone provides two-way communication – you can listen and
talk at the same time.
Intercoms can be very simple or very complex, analog or digital, party line or private. They can be found
everywhere from local theatre productions to Broadway, community television to all major television productions
and networks, as well as many other locations including control facilities for launching rockets.
Glossary
Intercom
An intercom is a sound system that is used for inter-communication between people or rooms. It provides
two-way communication where all participants can listen and talk at the same time.
Paging systems are for one-way communication only. They are often used for information that needs to be
communicated to a large audience. Paging systems can be simple, such as a system used for occasional
announcements at a department store, or they can be complex, as part of an ENS (Emergency Notification
System) with prerecorded announcements used during a life-threatening situation. Paging systems emphasize
intelligibility - clear one-way communication of an intended message.

Glossary
Paging System
A paging system is a sound system that is used for one-way communication only. It is often used for
communicating information to a large audience. The emphasis in a paging system is intelligibility - the clear, one-
way communication of an intended message.

f. Audio Conferencing and Videoconferencing

Audio conferencing is a way for groups of people to communicate. It is different from a telephone call in that
telephones only allow two people to communicate at a time.
Depending on the size of the conference, a central audio conferencing “pod” may fulfill audio conferencing needs.
This portable tabletop “pod” typically contains a loudspeaker and several microphones. Larger conferences may
require multiple individual microphones, installed as a part of a larger integrated audiovisual system.
Part of the equipment included in audio conferencing will include either line or acoustic echo-cancellation
technology. Larger systems may also include speech reinforcement and playback capabilities.
Quick Tip: Audio for Videoconferencing
Audio for videoconferencing is simply using audioconferencing equipment and technology to support a
videoconference.

g. Sound Masking Systems

For most types of sound systems, you will want to minimize background noise levels so that the intended message
can be communicated clearly.
Sound masking, or a “speech privacy system,” is a system that creates background noise like pink noise to reduce
clarity and create privacy. It also helps to reduce distractions from other noises. Often, an electronic noise
generator is sent to loudspeakers located in the ceiling above the drop tiles.
In a quiet, non-reverberant environment, conversations can be overheard unintentionally, invading privacy.
Mechanical or man-made noises can become disruptive to the work effort. When properly installed and balanced,
the sound masking system increases privacy and makes these noises less noticeable. Reducing noticeable noise
in the workplace can also increase efficiency.
Sound masking systems are commonly applied in medical, government, and industrial facilities.

Glossary
Speech Privacy Systems
Speech privacy system is a sound system that adds background noise to an environment to cover up human
speech and prevent privacy issues.

h. Audio Systems Summary

Much of what you need to know as an audiovisual professional revolves around sound. In this Audio section, you
learned how your hearing receives and processes sound. You read about the basics of sound propagation, sound
wave frequency, wavelength, harmonics, decibels, and the sound environment. You then learned how these factors
affect the electrical pathway used to amplify sound.
You explored the electrical audio signal chain from start to finish — from microphones to loudspeakers — and
learned about the various signal levels, cables, and the types of circuits that are preferred for professional audio.
The complete audio signal path takes acoustic energy and converts it into electrical energy so it can be routed,
processed, further amplified, and converted back into acoustical energy.

i. Audio System Applications Section Test